am d58b6cd1: am cbc90453: am 14958e21: Merge "audioflinger: fix issue with camcorder and A2DP" into ics-mr1
* commit 'd58b6cd1e3fdf3deb5147daec556fe424a568732':
diff --git a/camera/Camera.cpp b/camera/Camera.cpp
index ee458f1..b81fe86 100644
--- a/camera/Camera.cpp
+++ b/camera/Camera.cpp
@@ -27,7 +27,8 @@
#include <camera/ICameraRecordingProxyListener.h>
#include <camera/ICameraService.h>
-#include <surfaceflinger/Surface.h>
+#include <gui/ISurfaceTexture.h>
+#include <gui/Surface.h>
namespace android {
@@ -115,13 +116,13 @@
return cs->getCameraInfo(cameraId, cameraInfo);
}
-sp<Camera> Camera::connect(int cameraId)
+sp<Camera> Camera::connect(int cameraId, bool force, bool keep)
{
ALOGV("connect");
sp<Camera> c = new Camera();
const sp<ICameraService>& cs = getCameraService();
if (cs != 0) {
- c->mCamera = cs->connect(c, cameraId);
+ c->mCamera = cs->connect(c, cameraId, force, keep);
}
if (c->mCamera != 0) {
c->mCamera->asBinder()->linkToDeath(c);
diff --git a/camera/ICamera.cpp b/camera/ICamera.cpp
index 70f5dbc..8d8408c 100644
--- a/camera/ICamera.cpp
+++ b/camera/ICamera.cpp
@@ -22,6 +22,8 @@
#include <sys/types.h>
#include <binder/Parcel.h>
#include <camera/ICamera.h>
+#include <gui/ISurfaceTexture.h>
+#include <gui/Surface.h>
namespace android {
diff --git a/camera/ICameraService.cpp b/camera/ICameraService.cpp
index 85f1a29..c74298a 100644
--- a/camera/ICameraService.cpp
+++ b/camera/ICameraService.cpp
@@ -56,12 +56,15 @@
}
// connect to camera service
- virtual sp<ICamera> connect(const sp<ICameraClient>& cameraClient, int cameraId)
+ virtual sp<ICamera> connect(const sp<ICameraClient>& cameraClient, int cameraId,
+ bool force, bool keep)
{
Parcel data, reply;
data.writeInterfaceToken(ICameraService::getInterfaceDescriptor());
data.writeStrongBinder(cameraClient->asBinder());
data.writeInt32(cameraId);
+ data.writeInt32(force);
+ data.writeInt32(keep);
remote()->transact(BnCameraService::CONNECT, data, &reply);
return interface_cast<ICamera>(reply.readStrongBinder());
}
@@ -93,7 +96,10 @@
case CONNECT: {
CHECK_INTERFACE(ICameraService, data, reply);
sp<ICameraClient> cameraClient = interface_cast<ICameraClient>(data.readStrongBinder());
- sp<ICamera> camera = connect(cameraClient, data.readInt32());
+ const int cameraId = data.readInt32();
+ const int force = data.readInt32();
+ const int keep = data.readInt32();
+ sp<ICamera> camera = connect(cameraClient, cameraId, force, keep);
reply->writeStrongBinder(camera->asBinder());
return NO_ERROR;
} break;
@@ -105,4 +111,3 @@
// ----------------------------------------------------------------------------
}; // namespace android
-
diff --git a/cmds/stagefright/Android.mk b/cmds/stagefright/Android.mk
index e9642f7..30be7fa 100644
--- a/cmds/stagefright/Android.mk
+++ b/cmds/stagefright/Android.mk
@@ -7,16 +7,16 @@
SineSource.cpp
LOCAL_SHARED_LIBRARIES := \
- libstagefright libmedia libutils libbinder libstagefright_foundation \
+ libstagefright libmedia libmedia_native libutils libbinder libstagefright_foundation \
libskia libgui
LOCAL_C_INCLUDES:= \
$(JNI_H_INCLUDE) \
frameworks/base/media/libstagefright \
frameworks/base/media/libstagefright/include \
- $(TOP)/frameworks/base/include/media/stagefright/openmax \
- external/skia/include/core \
- external/skia/include/images \
+ $(TOP)/frameworks/native/include/media/openmax \
+ external/skia/include/core \
+ external/skia/include/images \
LOCAL_CFLAGS += -Wno-multichar
@@ -35,12 +35,12 @@
record.cpp
LOCAL_SHARED_LIBRARIES := \
- libstagefright liblog libutils libbinder
+ libstagefright liblog libutils libbinder libstagefright_foundation
LOCAL_C_INCLUDES:= \
$(JNI_H_INCLUDE) \
frameworks/base/media/libstagefright \
- $(TOP)/frameworks/base/include/media/stagefright/openmax
+ $(TOP)/frameworks/native/include/media/openmax
LOCAL_CFLAGS += -Wno-multichar
@@ -59,12 +59,12 @@
recordvideo.cpp
LOCAL_SHARED_LIBRARIES := \
- libstagefright liblog libutils libbinder
+ libstagefright liblog libutils libbinder libstagefright_foundation
LOCAL_C_INCLUDES:= \
$(JNI_H_INCLUDE) \
frameworks/base/media/libstagefright \
- $(TOP)/frameworks/base/include/media/stagefright/openmax
+ $(TOP)/frameworks/native/include/media/openmax
LOCAL_CFLAGS += -Wno-multichar
@@ -84,12 +84,12 @@
audioloop.cpp
LOCAL_SHARED_LIBRARIES := \
- libstagefright liblog libutils libbinder
+ libstagefright liblog libutils libbinder libstagefright_foundation
LOCAL_C_INCLUDES:= \
$(JNI_H_INCLUDE) \
frameworks/base/media/libstagefright \
- $(TOP)/frameworks/base/include/media/stagefright/openmax
+ $(TOP)/frameworks/native/include/media/openmax
LOCAL_CFLAGS += -Wno-multichar
@@ -108,12 +108,12 @@
LOCAL_SHARED_LIBRARIES := \
libstagefright liblog libutils libbinder libgui \
- libstagefright_foundation libmedia
+ libstagefright_foundation libmedia libmedia_native
LOCAL_C_INCLUDES:= \
$(JNI_H_INCLUDE) \
frameworks/base/media/libstagefright \
- $(TOP)/frameworks/base/include/media/stagefright/openmax
+ $(TOP)/frameworks/native/include/media/openmax
LOCAL_CFLAGS += -Wno-multichar
@@ -132,12 +132,12 @@
LOCAL_SHARED_LIBRARIES := \
libstagefright liblog libutils libbinder libstagefright_foundation \
- libmedia libgui libcutils libui
+ libmedia libmedia_native libgui libcutils libui
LOCAL_C_INCLUDES:= \
$(JNI_H_INCLUDE) \
frameworks/base/media/libstagefright \
- $(TOP)/frameworks/base/include/media/stagefright/openmax
+ $(TOP)/frameworks/native/include/media/openmax
LOCAL_CFLAGS += -Wno-multichar
@@ -147,4 +147,28 @@
include $(BUILD_EXECUTABLE)
+################################################################################
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES:= \
+ codec.cpp \
+ SimplePlayer.cpp \
+
+LOCAL_SHARED_LIBRARIES := \
+ libstagefright liblog libutils libbinder libstagefright_foundation \
+ libmedia libmedia_native libgui libcutils libui
+
+LOCAL_C_INCLUDES:= \
+ $(JNI_H_INCLUDE) \
+ frameworks/base/media/libstagefright \
+ $(TOP)/frameworks/native/include/media/openmax
+
+LOCAL_CFLAGS += -Wno-multichar
+
+LOCAL_MODULE_TAGS := debug
+
+LOCAL_MODULE:= codec
+
+include $(BUILD_EXECUTABLE)
diff --git a/cmds/stagefright/SimplePlayer.cpp b/cmds/stagefright/SimplePlayer.cpp
new file mode 100644
index 0000000..0cfeb3e
--- /dev/null
+++ b/cmds/stagefright/SimplePlayer.cpp
@@ -0,0 +1,645 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "SimplePlayer"
+#include <utils/Log.h>
+
+#include "SimplePlayer.h"
+
+#include <gui/SurfaceTextureClient.h>
+#include <media/AudioTrack.h>
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/MediaCodec.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/NativeWindowWrapper.h>
+#include <media/stagefright/NuMediaExtractor.h>
+
+namespace android {
+
+SimplePlayer::SimplePlayer()
+ : mState(UNINITIALIZED),
+ mDoMoreStuffGeneration(0),
+ mStartTimeRealUs(-1ll) {
+}
+
+SimplePlayer::~SimplePlayer() {
+}
+
+// static
+status_t PostAndAwaitResponse(
+ const sp<AMessage> &msg, sp<AMessage> *response) {
+ status_t err = msg->postAndAwaitResponse(response);
+
+ if (err != OK) {
+ return err;
+ }
+
+ if (!(*response)->findInt32("err", &err)) {
+ err = OK;
+ }
+
+ return err;
+}
+status_t SimplePlayer::setDataSource(const char *path) {
+ sp<AMessage> msg = new AMessage(kWhatSetDataSource, id());
+ msg->setString("path", path);
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t SimplePlayer::setSurface(const sp<ISurfaceTexture> &surfaceTexture) {
+ sp<AMessage> msg = new AMessage(kWhatSetSurface, id());
+
+ sp<SurfaceTextureClient> surfaceTextureClient;
+ if (surfaceTexture != NULL) {
+ surfaceTextureClient = new SurfaceTextureClient(surfaceTexture);
+ }
+
+ msg->setObject(
+ "native-window", new NativeWindowWrapper(surfaceTextureClient));
+
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t SimplePlayer::prepare() {
+ sp<AMessage> msg = new AMessage(kWhatPrepare, id());
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t SimplePlayer::start() {
+ sp<AMessage> msg = new AMessage(kWhatStart, id());
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t SimplePlayer::stop() {
+ sp<AMessage> msg = new AMessage(kWhatStop, id());
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t SimplePlayer::reset() {
+ sp<AMessage> msg = new AMessage(kWhatReset, id());
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+void SimplePlayer::onMessageReceived(const sp<AMessage> &msg) {
+ switch (msg->what()) {
+ case kWhatSetDataSource:
+ {
+ status_t err;
+ if (mState != UNINITIALIZED) {
+ err = INVALID_OPERATION;
+ } else {
+ CHECK(msg->findString("path", &mPath));
+ mState = UNPREPARED;
+ }
+
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", err);
+ response->postReply(replyID);
+ break;
+ }
+
+ case kWhatSetSurface:
+ {
+ status_t err;
+ if (mState != UNPREPARED) {
+ err = INVALID_OPERATION;
+ } else {
+ sp<RefBase> obj;
+ CHECK(msg->findObject("native-window", &obj));
+
+ mNativeWindow = static_cast<NativeWindowWrapper *>(obj.get());
+
+ err = OK;
+ }
+
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", err);
+ response->postReply(replyID);
+ break;
+ }
+
+ case kWhatPrepare:
+ {
+ status_t err;
+ if (mState != UNPREPARED) {
+ err = INVALID_OPERATION;
+ } else {
+ err = onPrepare();
+
+ if (err == OK) {
+ mState = STOPPED;
+ }
+ }
+
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", err);
+ response->postReply(replyID);
+ break;
+ }
+
+ case kWhatStart:
+ {
+ status_t err = OK;
+
+ if (mState == UNPREPARED) {
+ err = onPrepare();
+
+ if (err == OK) {
+ mState = STOPPED;
+ }
+ }
+
+ if (err == OK) {
+ if (mState != STOPPED) {
+ err = INVALID_OPERATION;
+ } else {
+ err = onStart();
+
+ if (err == OK) {
+ mState = STARTED;
+ }
+ }
+ }
+
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", err);
+ response->postReply(replyID);
+ break;
+ }
+
+ case kWhatStop:
+ {
+ status_t err;
+
+ if (mState != STARTED) {
+ err = INVALID_OPERATION;
+ } else {
+ err = onStop();
+
+ if (err == OK) {
+ mState = STOPPED;
+ }
+ }
+
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", err);
+ response->postReply(replyID);
+ break;
+ }
+
+ case kWhatReset:
+ {
+ status_t err = OK;
+
+ if (mState == STARTED) {
+ CHECK_EQ(onStop(), (status_t)OK);
+ mState = STOPPED;
+ }
+
+ if (mState == STOPPED) {
+ err = onReset();
+ mState = UNINITIALIZED;
+ }
+
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", err);
+ response->postReply(replyID);
+ break;
+ }
+
+ case kWhatDoMoreStuff:
+ {
+ int32_t generation;
+ CHECK(msg->findInt32("generation", &generation));
+
+ if (generation != mDoMoreStuffGeneration) {
+ break;
+ }
+
+ status_t err = onDoMoreStuff();
+
+ if (err == OK) {
+ msg->post(10000ll);
+ }
+ break;
+ }
+
+ default:
+ TRESPASS();
+ }
+}
+
+status_t SimplePlayer::onPrepare() {
+ CHECK_EQ(mState, UNPREPARED);
+
+ mExtractor = new NuMediaExtractor;
+
+ status_t err = mExtractor->setDataSource(mPath.c_str());
+
+ if (err != OK) {
+ mExtractor.clear();
+ return err;
+ }
+
+ if (mCodecLooper == NULL) {
+ mCodecLooper = new ALooper;
+ mCodecLooper->start();
+ }
+
+ bool haveAudio = false;
+ bool haveVideo = false;
+ for (size_t i = 0; i < mExtractor->countTracks(); ++i) {
+ sp<AMessage> format;
+ status_t err = mExtractor->getTrackFormat(i, &format);
+ CHECK_EQ(err, (status_t)OK);
+
+ AString mime;
+ CHECK(format->findString("mime", &mime));
+
+ if (!haveAudio && !strncasecmp(mime.c_str(), "audio/", 6)) {
+ haveAudio = true;
+ } else if (!haveVideo && !strncasecmp(mime.c_str(), "video/", 6)) {
+ haveVideo = true;
+ } else {
+ continue;
+ }
+
+ err = mExtractor->selectTrack(i);
+ CHECK_EQ(err, (status_t)OK);
+
+ CodecState *state =
+ &mStateByTrackIndex.editValueAt(
+ mStateByTrackIndex.add(i, CodecState()));
+
+ state->mNumFramesWritten = 0;
+ state->mCodec = MediaCodec::CreateByType(
+ mCodecLooper, mime.c_str(), false /* encoder */);
+
+ CHECK(state->mCodec != NULL);
+
+ err = state->mCodec->configure(
+ format, mNativeWindow->getSurfaceTextureClient(),
+ 0 /* flags */);
+
+ CHECK_EQ(err, (status_t)OK);
+
+ size_t j = 0;
+ sp<ABuffer> buffer;
+ while (format->findBuffer(StringPrintf("csd-%d", j).c_str(), &buffer)) {
+ state->mCSD.push_back(buffer);
+
+ ++j;
+ }
+ }
+
+ for (size_t i = 0; i < mStateByTrackIndex.size(); ++i) {
+ CodecState *state = &mStateByTrackIndex.editValueAt(i);
+
+ status_t err = state->mCodec->start();
+ CHECK_EQ(err, (status_t)OK);
+
+ err = state->mCodec->getInputBuffers(&state->mBuffers[0]);
+ CHECK_EQ(err, (status_t)OK);
+
+ err = state->mCodec->getOutputBuffers(&state->mBuffers[1]);
+ CHECK_EQ(err, (status_t)OK);
+
+ for (size_t j = 0; j < state->mCSD.size(); ++j) {
+ const sp<ABuffer> &srcBuffer = state->mCSD.itemAt(j);
+
+ size_t index;
+ err = state->mCodec->dequeueInputBuffer(&index, -1ll);
+ CHECK_EQ(err, (status_t)OK);
+
+ const sp<ABuffer> &dstBuffer = state->mBuffers[0].itemAt(index);
+
+ CHECK_LE(srcBuffer->size(), dstBuffer->capacity());
+ dstBuffer->setRange(0, srcBuffer->size());
+ memcpy(dstBuffer->data(), srcBuffer->data(), srcBuffer->size());
+
+ err = state->mCodec->queueInputBuffer(
+ index,
+ 0,
+ dstBuffer->size(),
+ 0ll,
+ MediaCodec::BUFFER_FLAG_CODECCONFIG);
+ CHECK_EQ(err, (status_t)OK);
+ }
+ }
+
+ return OK;
+}
+
+status_t SimplePlayer::onStart() {
+ CHECK_EQ(mState, STOPPED);
+
+ mStartTimeRealUs = -1ll;
+
+ sp<AMessage> msg = new AMessage(kWhatDoMoreStuff, id());
+ msg->setInt32("generation", ++mDoMoreStuffGeneration);
+ msg->post();
+
+ return OK;
+}
+
+status_t SimplePlayer::onStop() {
+ CHECK_EQ(mState, STARTED);
+
+ ++mDoMoreStuffGeneration;
+
+ return OK;
+}
+
+status_t SimplePlayer::onReset() {
+ CHECK_EQ(mState, STOPPED);
+
+ for (size_t i = 0; i < mStateByTrackIndex.size(); ++i) {
+ CodecState *state = &mStateByTrackIndex.editValueAt(i);
+
+ CHECK_EQ(state->mCodec->release(), (status_t)OK);
+ }
+
+ mStartTimeRealUs = -1ll;
+
+ mStateByTrackIndex.clear();
+ mCodecLooper.clear();
+ mExtractor.clear();
+ mNativeWindow.clear();
+ mPath.clear();
+
+ return OK;
+}
+
+status_t SimplePlayer::onDoMoreStuff() {
+ ALOGV("onDoMoreStuff");
+ for (size_t i = 0; i < mStateByTrackIndex.size(); ++i) {
+ CodecState *state = &mStateByTrackIndex.editValueAt(i);
+
+ status_t err;
+ do {
+ size_t index;
+ err = state->mCodec->dequeueInputBuffer(&index);
+
+ if (err == OK) {
+ ALOGV("dequeued input buffer on track %d",
+ mStateByTrackIndex.keyAt(i));
+
+ state->mAvailInputBufferIndices.push_back(index);
+ } else {
+ ALOGV("dequeueInputBuffer on track %d returned %d",
+ mStateByTrackIndex.keyAt(i), err);
+ }
+ } while (err == OK);
+
+ do {
+ BufferInfo info;
+ err = state->mCodec->dequeueOutputBuffer(
+ &info.mIndex,
+ &info.mOffset,
+ &info.mSize,
+ &info.mPresentationTimeUs,
+ &info.mFlags);
+
+ if (err == OK) {
+ ALOGV("dequeued output buffer on track %d",
+ mStateByTrackIndex.keyAt(i));
+
+ state->mAvailOutputBufferInfos.push_back(info);
+ } else if (err == INFO_FORMAT_CHANGED) {
+ err = onOutputFormatChanged(mStateByTrackIndex.keyAt(i), state);
+ CHECK_EQ(err, (status_t)OK);
+ } else if (err == INFO_OUTPUT_BUFFERS_CHANGED) {
+ err = state->mCodec->getOutputBuffers(&state->mBuffers[1]);
+ CHECK_EQ(err, (status_t)OK);
+ } else {
+ ALOGV("dequeueOutputBuffer on track %d returned %d",
+ mStateByTrackIndex.keyAt(i), err);
+ }
+ } while (err == OK
+ || err == INFO_FORMAT_CHANGED
+ || err == INFO_OUTPUT_BUFFERS_CHANGED);
+ }
+
+ for (;;) {
+ size_t trackIndex;
+ status_t err = mExtractor->getSampleTrackIndex(&trackIndex);
+
+ if (err != OK) {
+ ALOGI("encountered input EOS.");
+ break;
+ } else {
+ CodecState *state = &mStateByTrackIndex.editValueFor(trackIndex);
+
+ if (state->mAvailInputBufferIndices.empty()) {
+ break;
+ }
+
+ size_t index = *state->mAvailInputBufferIndices.begin();
+ state->mAvailInputBufferIndices.erase(
+ state->mAvailInputBufferIndices.begin());
+
+ const sp<ABuffer> &dstBuffer =
+ state->mBuffers[0].itemAt(index);
+
+ err = mExtractor->readSampleData(dstBuffer);
+ CHECK_EQ(err, (status_t)OK);
+
+ int64_t timeUs;
+ CHECK_EQ(mExtractor->getSampleTime(&timeUs), (status_t)OK);
+
+ err = state->mCodec->queueInputBuffer(
+ index,
+ dstBuffer->offset(),
+ dstBuffer->size(),
+ timeUs,
+ 0);
+ CHECK_EQ(err, (status_t)OK);
+
+ ALOGV("enqueued input data on track %d", trackIndex);
+
+ err = mExtractor->advance();
+ CHECK_EQ(err, (status_t)OK);
+ }
+ }
+
+ int64_t nowUs = ALooper::GetNowUs();
+
+ if (mStartTimeRealUs < 0ll) {
+ mStartTimeRealUs = nowUs + 1000000ll;
+ }
+
+ for (size_t i = 0; i < mStateByTrackIndex.size(); ++i) {
+ CodecState *state = &mStateByTrackIndex.editValueAt(i);
+
+ while (!state->mAvailOutputBufferInfos.empty()) {
+ BufferInfo *info = &*state->mAvailOutputBufferInfos.begin();
+
+ int64_t whenRealUs = info->mPresentationTimeUs + mStartTimeRealUs;
+ int64_t lateByUs = nowUs - whenRealUs;
+
+ if (lateByUs > -10000ll) {
+ bool release = true;
+
+ if (lateByUs > 30000ll) {
+ ALOGI("track %d buffer late by %lld us, dropping.",
+ mStateByTrackIndex.keyAt(i), lateByUs);
+ state->mCodec->releaseOutputBuffer(info->mIndex);
+ } else {
+ if (state->mAudioTrack != NULL) {
+ const sp<ABuffer> &srcBuffer =
+ state->mBuffers[1].itemAt(info->mIndex);
+
+ renderAudio(state, info, srcBuffer);
+
+ if (info->mSize > 0) {
+ release = false;
+ }
+ }
+
+ if (release) {
+ state->mCodec->renderOutputBufferAndRelease(
+ info->mIndex);
+ }
+ }
+
+ if (release) {
+ state->mAvailOutputBufferInfos.erase(
+ state->mAvailOutputBufferInfos.begin());
+
+ info = NULL;
+ } else {
+ break;
+ }
+ } else {
+ ALOGV("track %d buffer early by %lld us.",
+ mStateByTrackIndex.keyAt(i), -lateByUs);
+ break;
+ }
+ }
+ }
+
+ return OK;
+}
+
+status_t SimplePlayer::onOutputFormatChanged(
+ size_t trackIndex, CodecState *state) {
+ sp<AMessage> format;
+ status_t err = state->mCodec->getOutputFormat(&format);
+
+ if (err != OK) {
+ return err;
+ }
+
+ AString mime;
+ CHECK(format->findString("mime", &mime));
+
+ if (!strncasecmp(mime.c_str(), "audio/", 6)) {
+ int32_t channelCount;
+ int32_t sampleRate;
+ CHECK(format->findInt32("channel-count", &channelCount));
+ CHECK(format->findInt32("sample-rate", &sampleRate));
+
+ state->mAudioTrack = new AudioTrack(
+ AUDIO_STREAM_MUSIC,
+ sampleRate,
+ AUDIO_FORMAT_PCM_16_BIT,
+ audio_channel_out_mask_from_count(channelCount),
+ 0);
+
+ state->mNumFramesWritten = 0;
+ }
+
+ return OK;
+}
+
+void SimplePlayer::renderAudio(
+ CodecState *state, BufferInfo *info, const sp<ABuffer> &buffer) {
+ CHECK(state->mAudioTrack != NULL);
+
+ if (state->mAudioTrack->stopped()) {
+ state->mAudioTrack->start();
+ }
+
+ uint32_t numFramesPlayed;
+ CHECK_EQ(state->mAudioTrack->getPosition(&numFramesPlayed), (status_t)OK);
+
+ uint32_t numFramesAvailableToWrite =
+ state->mAudioTrack->frameCount()
+ - (state->mNumFramesWritten - numFramesPlayed);
+
+ size_t numBytesAvailableToWrite =
+ numFramesAvailableToWrite * state->mAudioTrack->frameSize();
+
+ size_t copy = info->mSize;
+ if (copy > numBytesAvailableToWrite) {
+ copy = numBytesAvailableToWrite;
+ }
+
+ if (copy == 0) {
+ return;
+ }
+
+ int64_t startTimeUs = ALooper::GetNowUs();
+
+ ssize_t nbytes = state->mAudioTrack->write(
+ buffer->base() + info->mOffset, copy);
+
+ CHECK_EQ(nbytes, (ssize_t)copy);
+
+ int64_t delayUs = ALooper::GetNowUs() - startTimeUs;
+
+ uint32_t numFramesWritten = nbytes / state->mAudioTrack->frameSize();
+
+ if (delayUs > 2000ll) {
+ ALOGW("AudioTrack::write took %lld us, numFramesAvailableToWrite=%u, "
+ "numFramesWritten=%u",
+ delayUs, numFramesAvailableToWrite, numFramesWritten);
+ }
+
+ info->mOffset += nbytes;
+ info->mSize -= nbytes;
+
+ state->mNumFramesWritten += numFramesWritten;
+}
+
+} // namespace android
diff --git a/cmds/stagefright/SimplePlayer.h b/cmds/stagefright/SimplePlayer.h
new file mode 100644
index 0000000..2548252
--- /dev/null
+++ b/cmds/stagefright/SimplePlayer.h
@@ -0,0 +1,109 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <media/stagefright/foundation/AHandler.h>
+#include <media/stagefright/foundation/AString.h>
+#include <utils/KeyedVector.h>
+
+namespace android {
+
+struct ABuffer;
+struct ALooper;
+struct AudioTrack;
+struct ISurfaceTexture;
+struct MediaCodec;
+struct NativeWindowWrapper;
+struct NuMediaExtractor;
+
+struct SimplePlayer : public AHandler {
+ SimplePlayer();
+
+ status_t setDataSource(const char *path);
+ status_t setSurface(const sp<ISurfaceTexture> &surfaceTexture);
+ status_t prepare();
+ status_t start();
+ status_t stop();
+ status_t reset();
+
+protected:
+ virtual ~SimplePlayer();
+
+ virtual void onMessageReceived(const sp<AMessage> &msg);
+
+private:
+ enum State {
+ UNINITIALIZED,
+ UNPREPARED,
+ STOPPED,
+ STARTED
+ };
+
+ enum {
+ kWhatSetDataSource,
+ kWhatSetSurface,
+ kWhatPrepare,
+ kWhatStart,
+ kWhatStop,
+ kWhatReset,
+ kWhatDoMoreStuff,
+ };
+
+ struct BufferInfo {
+ size_t mIndex;
+ size_t mOffset;
+ size_t mSize;
+ int64_t mPresentationTimeUs;
+ uint32_t mFlags;
+ };
+
+ struct CodecState
+ {
+ sp<MediaCodec> mCodec;
+ Vector<sp<ABuffer> > mCSD;
+ Vector<sp<ABuffer> > mBuffers[2];
+
+ List<size_t> mAvailInputBufferIndices;
+ List<BufferInfo> mAvailOutputBufferInfos;
+
+ sp<AudioTrack> mAudioTrack;
+ uint32_t mNumFramesWritten;
+ };
+
+ State mState;
+ AString mPath;
+ sp<NativeWindowWrapper> mNativeWindow;
+
+ sp<NuMediaExtractor> mExtractor;
+ sp<ALooper> mCodecLooper;
+ KeyedVector<size_t, CodecState> mStateByTrackIndex;
+ int32_t mDoMoreStuffGeneration;
+
+ int64_t mStartTimeRealUs;
+
+ status_t onPrepare();
+ status_t onStart();
+ status_t onStop();
+ status_t onReset();
+ status_t onDoMoreStuff();
+ status_t onOutputFormatChanged(size_t trackIndex, CodecState *state);
+
+ void renderAudio(
+ CodecState *state, BufferInfo *info, const sp<ABuffer> &buffer);
+
+ DISALLOW_EVIL_CONSTRUCTORS(SimplePlayer);
+};
+
+} // namespace android
diff --git a/cmds/stagefright/SineSource.cpp b/cmds/stagefright/SineSource.cpp
index 021f636..14b4306 100644
--- a/cmds/stagefright/SineSource.cpp
+++ b/cmds/stagefright/SineSource.cpp
@@ -3,7 +3,7 @@
#include <math.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
diff --git a/cmds/stagefright/audioloop.cpp b/cmds/stagefright/audioloop.cpp
index 858681f..ed7d6cb 100644
--- a/cmds/stagefright/audioloop.cpp
+++ b/cmds/stagefright/audioloop.cpp
@@ -2,10 +2,10 @@
#include <binder/ProcessState.h>
#include <media/mediarecorder.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/AMRWriter.h>
#include <media/stagefright/AudioPlayer.h>
#include <media/stagefright/AudioSource.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/OMXClient.h>
@@ -24,7 +24,7 @@
android::ProcessState::self()->startThreadPool();
OMXClient client;
- CHECK_EQ(client.connect(), OK);
+ CHECK_EQ(client.connect(), (status_t)OK);
#if 0
sp<MediaSource> source = new SineSource(kSampleRate, kNumChannels);
@@ -32,9 +32,7 @@
sp<MediaSource> source = new AudioSource(
AUDIO_SOURCE_DEFAULT,
kSampleRate,
- kNumChannels == 1
- ? AUDIO_CHANNEL_IN_MONO
- : AUDIO_CHANNEL_IN_STEREO);
+ audio_channel_in_mask_from_count(kNumChannels));
#endif
sp<MetaData> meta = new MetaData;
@@ -82,7 +80,7 @@
delete player;
player = NULL;
#elif 0
- CHECK_EQ(decoder->start(), OK);
+ CHECK_EQ(decoder->start(), (status_t)OK);
MediaBuffer *buffer;
while (decoder->read(&buffer) == OK) {
@@ -95,7 +93,7 @@
buffer = NULL;
}
- CHECK_EQ(decoder->stop(), OK);
+ CHECK_EQ(decoder->stop(), (status_t)OK);
#endif
#endif
diff --git a/cmds/stagefright/codec.cpp b/cmds/stagefright/codec.cpp
new file mode 100644
index 0000000..fea62cc
--- /dev/null
+++ b/cmds/stagefright/codec.cpp
@@ -0,0 +1,453 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "codec"
+#include <utils/Log.h>
+
+#include "SimplePlayer.h"
+
+#include <binder/ProcessState.h>
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/ALooper.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/DataSource.h>
+#include <media/stagefright/MediaCodec.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/NuMediaExtractor.h>
+#include <gui/SurfaceComposerClient.h>
+
+static void usage(const char *me) {
+ fprintf(stderr, "usage: %s [-a] use audio\n"
+ "\t\t[-v] use video\n"
+ "\t\t[-p] playback\n"
+ "\t\t[-S] allocate buffers from a surface\n", me);
+
+ exit(1);
+}
+
+namespace android {
+
+struct CodecState {
+ sp<MediaCodec> mCodec;
+ Vector<sp<ABuffer> > mCSD;
+ size_t mCSDIndex;
+ Vector<sp<ABuffer> > mInBuffers;
+ Vector<sp<ABuffer> > mOutBuffers;
+ bool mSignalledInputEOS;
+ bool mSawOutputEOS;
+ int64_t mNumBuffersDecoded;
+ int64_t mNumBytesDecoded;
+ bool mIsAudio;
+};
+
+} // namespace android
+
+static int decode(
+ const android::sp<android::ALooper> &looper,
+ const char *path,
+ bool useAudio,
+ bool useVideo,
+ const android::sp<android::Surface> &surface) {
+ using namespace android;
+
+ static int64_t kTimeout = 500ll;
+
+ sp<NuMediaExtractor> extractor = new NuMediaExtractor;
+ if (extractor->setDataSource(path) != OK) {
+ fprintf(stderr, "unable to instantiate extractor.\n");
+ return 1;
+ }
+
+ KeyedVector<size_t, CodecState> stateByTrack;
+
+ bool haveAudio = false;
+ bool haveVideo = false;
+ for (size_t i = 0; i < extractor->countTracks(); ++i) {
+ sp<AMessage> format;
+ status_t err = extractor->getTrackFormat(i, &format);
+ CHECK_EQ(err, (status_t)OK);
+
+ AString mime;
+ CHECK(format->findString("mime", &mime));
+
+ bool isAudio = !strncasecmp(mime.c_str(), "audio/", 6);
+ bool isVideo = !strncasecmp(mime.c_str(), "video/", 6);
+
+ if (useAudio && !haveAudio && isAudio) {
+ haveAudio = true;
+ } else if (useVideo && !haveVideo && isVideo) {
+ haveVideo = true;
+ } else {
+ continue;
+ }
+
+ ALOGV("selecting track %d", i);
+
+ err = extractor->selectTrack(i);
+ CHECK_EQ(err, (status_t)OK);
+
+ CodecState *state =
+ &stateByTrack.editValueAt(stateByTrack.add(i, CodecState()));
+
+ state->mNumBytesDecoded = 0;
+ state->mNumBuffersDecoded = 0;
+ state->mIsAudio = isAudio;
+
+ state->mCodec = MediaCodec::CreateByType(
+ looper, mime.c_str(), false /* encoder */);
+
+ CHECK(state->mCodec != NULL);
+
+ err = state->mCodec->configure(
+ format, isVideo ? surface : NULL, 0 /* flags */);
+
+ CHECK_EQ(err, (status_t)OK);
+
+ size_t j = 0;
+ sp<ABuffer> buffer;
+ while (format->findBuffer(StringPrintf("csd-%d", j).c_str(), &buffer)) {
+ state->mCSD.push_back(buffer);
+
+ ++j;
+ }
+
+ state->mCSDIndex = 0;
+ state->mSignalledInputEOS = false;
+ state->mSawOutputEOS = false;
+
+ ALOGV("got %d pieces of codec specific data.", state->mCSD.size());
+ }
+
+ CHECK(!stateByTrack.isEmpty());
+
+ int64_t startTimeUs = ALooper::GetNowUs();
+
+ for (size_t i = 0; i < stateByTrack.size(); ++i) {
+ CodecState *state = &stateByTrack.editValueAt(i);
+
+ sp<MediaCodec> codec = state->mCodec;
+
+ CHECK_EQ((status_t)OK, codec->start());
+
+ CHECK_EQ((status_t)OK, codec->getInputBuffers(&state->mInBuffers));
+ CHECK_EQ((status_t)OK, codec->getOutputBuffers(&state->mOutBuffers));
+
+ ALOGV("got %d input and %d output buffers",
+ state->mInBuffers.size(), state->mOutBuffers.size());
+
+ while (state->mCSDIndex < state->mCSD.size()) {
+ size_t index;
+ status_t err = codec->dequeueInputBuffer(&index, -1ll);
+ CHECK_EQ(err, (status_t)OK);
+
+ const sp<ABuffer> &srcBuffer =
+ state->mCSD.itemAt(state->mCSDIndex++);
+
+ const sp<ABuffer> &buffer = state->mInBuffers.itemAt(index);
+
+ memcpy(buffer->data(), srcBuffer->data(), srcBuffer->size());
+
+ err = codec->queueInputBuffer(
+ index,
+ 0 /* offset */,
+ srcBuffer->size(),
+ 0ll /* timeUs */,
+ MediaCodec::BUFFER_FLAG_CODECCONFIG);
+
+ CHECK_EQ(err, (status_t)OK);
+ }
+ }
+
+ bool sawInputEOS = false;
+
+ for (;;) {
+ if (!sawInputEOS) {
+ size_t trackIndex;
+ status_t err = extractor->getSampleTrackIndex(&trackIndex);
+
+ if (err != OK) {
+ ALOGV("saw input eos");
+ sawInputEOS = true;
+ } else {
+ CodecState *state = &stateByTrack.editValueFor(trackIndex);
+
+ size_t index;
+ err = state->mCodec->dequeueInputBuffer(&index, kTimeout);
+
+ if (err == OK) {
+ ALOGV("filling input buffer %d", index);
+
+ const sp<ABuffer> &buffer = state->mInBuffers.itemAt(index);
+
+ err = extractor->readSampleData(buffer);
+ CHECK_EQ(err, (status_t)OK);
+
+ int64_t timeUs;
+ err = extractor->getSampleTime(&timeUs);
+ CHECK_EQ(err, (status_t)OK);
+
+ err = state->mCodec->queueInputBuffer(
+ index,
+ 0 /* offset */,
+ buffer->size(),
+ timeUs,
+ 0 /* flags */);
+
+ CHECK_EQ(err, (status_t)OK);
+
+ extractor->advance();
+ } else {
+ CHECK_EQ(err, -EAGAIN);
+ }
+ }
+ } else {
+ for (size_t i = 0; i < stateByTrack.size(); ++i) {
+ CodecState *state = &stateByTrack.editValueAt(i);
+
+ if (!state->mSignalledInputEOS) {
+ size_t index;
+ status_t err =
+ state->mCodec->dequeueInputBuffer(&index, kTimeout);
+
+ if (err == OK) {
+ ALOGV("signalling input EOS on track %d", i);
+
+ err = state->mCodec->queueInputBuffer(
+ index,
+ 0 /* offset */,
+ 0 /* size */,
+ 0ll /* timeUs */,
+ MediaCodec::BUFFER_FLAG_EOS);
+
+ CHECK_EQ(err, (status_t)OK);
+
+ state->mSignalledInputEOS = true;
+ } else {
+ CHECK_EQ(err, -EAGAIN);
+ }
+ }
+ }
+ }
+
+ bool sawOutputEOSOnAllTracks = true;
+ for (size_t i = 0; i < stateByTrack.size(); ++i) {
+ CodecState *state = &stateByTrack.editValueAt(i);
+ if (!state->mSawOutputEOS) {
+ sawOutputEOSOnAllTracks = false;
+ break;
+ }
+ }
+
+ if (sawOutputEOSOnAllTracks) {
+ break;
+ }
+
+ for (size_t i = 0; i < stateByTrack.size(); ++i) {
+ CodecState *state = &stateByTrack.editValueAt(i);
+
+ if (state->mSawOutputEOS) {
+ continue;
+ }
+
+ size_t index;
+ size_t offset;
+ size_t size;
+ int64_t presentationTimeUs;
+ uint32_t flags;
+ status_t err = state->mCodec->dequeueOutputBuffer(
+ &index, &offset, &size, &presentationTimeUs, &flags,
+ kTimeout);
+
+ if (err == OK) {
+ ALOGV("draining output buffer %d, time = %lld us",
+ index, presentationTimeUs);
+
+ ++state->mNumBuffersDecoded;
+ state->mNumBytesDecoded += size;
+
+ err = state->mCodec->releaseOutputBuffer(index);
+ CHECK_EQ(err, (status_t)OK);
+
+ if (flags & MediaCodec::BUFFER_FLAG_EOS) {
+ ALOGV("reached EOS on output.");
+
+ state->mSawOutputEOS = true;
+ }
+ } else if (err == INFO_OUTPUT_BUFFERS_CHANGED) {
+ ALOGV("INFO_OUTPUT_BUFFERS_CHANGED");
+ CHECK_EQ((status_t)OK,
+ state->mCodec->getOutputBuffers(&state->mOutBuffers));
+
+ ALOGV("got %d output buffers", state->mOutBuffers.size());
+ } else if (err == INFO_FORMAT_CHANGED) {
+ sp<AMessage> format;
+ CHECK_EQ((status_t)OK, state->mCodec->getOutputFormat(&format));
+
+ ALOGV("INFO_FORMAT_CHANGED: %s", format->debugString().c_str());
+ } else {
+ CHECK_EQ(err, -EAGAIN);
+ }
+ }
+ }
+
+ int64_t elapsedTimeUs = ALooper::GetNowUs() - startTimeUs;
+
+ for (size_t i = 0; i < stateByTrack.size(); ++i) {
+ CodecState *state = &stateByTrack.editValueAt(i);
+
+ CHECK_EQ((status_t)OK, state->mCodec->release());
+
+ if (state->mIsAudio) {
+ printf("track %d: %lld bytes received. %.2f KB/sec\n",
+ i,
+ state->mNumBytesDecoded,
+ state->mNumBytesDecoded * 1E6 / 1024 / elapsedTimeUs);
+ } else {
+ printf("track %d: %lld frames decoded, %.2f fps. %lld bytes "
+ "received. %.2f KB/sec\n",
+ i,
+ state->mNumBuffersDecoded,
+ state->mNumBuffersDecoded * 1E6 / elapsedTimeUs,
+ state->mNumBytesDecoded,
+ state->mNumBytesDecoded * 1E6 / 1024 / elapsedTimeUs);
+ }
+ }
+
+ return 0;
+}
+
+int main(int argc, char **argv) {
+ using namespace android;
+
+ const char *me = argv[0];
+
+ bool useAudio = false;
+ bool useVideo = false;
+ bool playback = false;
+ bool useSurface = false;
+
+ int res;
+ while ((res = getopt(argc, argv, "havpS")) >= 0) {
+ switch (res) {
+ case 'a':
+ {
+ useAudio = true;
+ break;
+ }
+
+ case 'v':
+ {
+ useVideo = true;
+ break;
+ }
+
+ case 'p':
+ {
+ playback = true;
+ break;
+ }
+
+ case 'S':
+ {
+ useSurface = true;
+ break;
+ }
+
+ case '?':
+ case 'h':
+ default:
+ {
+ usage(me);
+ }
+ }
+ }
+
+ argc -= optind;
+ argv += optind;
+
+ if (argc != 1) {
+ usage(me);
+ }
+
+ if (!useAudio && !useVideo) {
+ useAudio = useVideo = true;
+ }
+
+ ProcessState::self()->startThreadPool();
+
+ DataSource::RegisterDefaultSniffers();
+
+ sp<ALooper> looper = new ALooper;
+ looper->start();
+
+ sp<SurfaceComposerClient> composerClient;
+ sp<SurfaceControl> control;
+ sp<Surface> surface;
+
+ if (playback || (useSurface && useVideo)) {
+ composerClient = new SurfaceComposerClient;
+ CHECK_EQ(composerClient->initCheck(), (status_t)OK);
+
+ ssize_t displayWidth = composerClient->getDisplayWidth(0);
+ ssize_t displayHeight = composerClient->getDisplayHeight(0);
+
+ ALOGV("display is %ld x %ld\n", displayWidth, displayHeight);
+
+ control = composerClient->createSurface(
+ String8("A Surface"),
+ 0,
+ displayWidth,
+ displayHeight,
+ PIXEL_FORMAT_RGB_565,
+ 0);
+
+ CHECK(control != NULL);
+ CHECK(control->isValid());
+
+ SurfaceComposerClient::openGlobalTransaction();
+ CHECK_EQ(control->setLayer(INT_MAX), (status_t)OK);
+ CHECK_EQ(control->show(), (status_t)OK);
+ SurfaceComposerClient::closeGlobalTransaction();
+
+ surface = control->getSurface();
+ CHECK(surface != NULL);
+ }
+
+ if (playback) {
+ sp<SimplePlayer> player = new SimplePlayer;
+ looper->registerHandler(player);
+
+ player->setDataSource(argv[0]);
+ player->setSurface(surface->getSurfaceTexture());
+ player->start();
+ sleep(60);
+ player->stop();
+ player->reset();
+ } else {
+ decode(looper, argv[0], useAudio, useVideo, surface);
+ }
+
+ if (playback || (useSurface && useVideo)) {
+ composerClient->dispose();
+ }
+
+ looper->stop();
+
+ return 0;
+}
diff --git a/cmds/stagefright/record.cpp b/cmds/stagefright/record.cpp
index 613435d..45c3f7b 100644
--- a/cmds/stagefright/record.cpp
+++ b/cmds/stagefright/record.cpp
@@ -17,11 +17,11 @@
#include "SineSource.h"
#include <binder/ProcessState.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/AudioPlayer.h>
#include <media/stagefright/CameraSource.h>
#include <media/stagefright/FileSource.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/MediaExtractor.h>
@@ -38,7 +38,7 @@
static const int32_t kAudioBitRate = 12200;
static const int64_t kDurationUs = 10000000LL; // 10 seconds
-#if 1
+#if 0
class DummySource : public MediaSource {
public:
@@ -183,7 +183,7 @@
return 1;
}
OMXClient client;
- CHECK_EQ(client.connect(), OK);
+ CHECK_EQ(client.connect(), (status_t)OK);
status_t err = OK;
@@ -231,14 +231,14 @@
sp<MPEG4Writer> writer = new MPEG4Writer("/sdcard/output.mp4");
writer->addSource(encoder);
writer->setMaxFileDuration(kDurationUs);
- CHECK_EQ(OK, writer->start());
+ CHECK_EQ((status_t)OK, writer->start());
while (!writer->reachedEOS()) {
fprintf(stderr, ".");
usleep(100000);
}
err = writer->stop();
#else
- CHECK_EQ(OK, encoder->start());
+ CHECK_EQ((status_t)OK, encoder->start());
MediaBuffer *buffer;
while (encoder->read(&buffer) == OK) {
@@ -272,7 +272,7 @@
for (int i = 0; i < 100; ++i) {
MediaBuffer *buffer;
status_t err = source->read(&buffer);
- CHECK_EQ(err, OK);
+ CHECK_EQ(err, (status_t)OK);
printf("got a frame, data=%p, size=%d\n",
buffer->data(), buffer->range_length());
@@ -299,7 +299,7 @@
android::ProcessState::self()->startThreadPool();
OMXClient client;
- CHECK_EQ(client.connect(), OK);
+ CHECK_EQ(client.connect(), (status_t)OK);
const int32_t kSampleRate = 22050;
const int32_t kNumChannels = 2;
@@ -318,7 +318,7 @@
sp<MetaData> encMeta = new MetaData;
encMeta->setCString(kKeyMIMEType,
- 1 ? MEDIA_MIMETYPE_AUDIO_AMR_WB : MEDIA_MIMETYPE_AUDIO_AAC);
+ 0 ? MEDIA_MIMETYPE_AUDIO_AMR_WB : MEDIA_MIMETYPE_AUDIO_AAC);
encMeta->setInt32(kKeySampleRate, kSampleRate);
encMeta->setInt32(kKeyChannelCount, kNumChannels);
encMeta->setInt32(kKeyMaxInputSize, 8192);
diff --git a/cmds/stagefright/recordvideo.cpp b/cmds/stagefright/recordvideo.cpp
index c402286..3bd1fe2 100644
--- a/cmds/stagefright/recordvideo.cpp
+++ b/cmds/stagefright/recordvideo.cpp
@@ -17,9 +17,9 @@
#include "SineSource.h"
#include <binder/ProcessState.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/AudioPlayer.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/MPEG4Writer.h>
@@ -243,7 +243,7 @@
}
OMXClient client;
- CHECK_EQ(client.connect(), OK);
+ CHECK_EQ(client.connect(), (status_t)OK);
status_t err = OK;
sp<MediaSource> source =
@@ -283,7 +283,7 @@
sp<MPEG4Writer> writer = new MPEG4Writer(fileName);
writer->addSource(encoder);
int64_t start = systemTime();
- CHECK_EQ(OK, writer->start());
+ CHECK_EQ((status_t)OK, writer->start());
while (!writer->reachedEOS()) {
}
err = writer->stop();
diff --git a/cmds/stagefright/sf2.cpp b/cmds/stagefright/sf2.cpp
index ae80f88..e47cdc0 100644
--- a/cmds/stagefright/sf2.cpp
+++ b/cmds/stagefright/sf2.cpp
@@ -14,6 +14,10 @@
* limitations under the License.
*/
+//#define LOG_NDEBUG 0
+#define LOG_TAG "sf2"
+#include <utils/Log.h>
+
#include <binder/ProcessState.h>
#include <media/stagefright/foundation/hexdump.h>
@@ -32,8 +36,7 @@
#include <media/stagefright/NativeWindowWrapper.h>
#include <media/stagefright/Utils.h>
-#include <surfaceflinger/ISurfaceComposer.h>
-#include <surfaceflinger/SurfaceComposerClient.h>
+#include <gui/SurfaceComposerClient.h>
#include "include/ESDS.h"
@@ -198,9 +201,7 @@
(new AMessage(kWhatSeek, id()))->post(5000000ll);
} else if (what == ACodec::kWhatOutputFormatChanged) {
- } else {
- CHECK_EQ(what, (int32_t)ACodec::kWhatShutdownCompleted);
-
+ } else if (what == ACodec::kWhatShutdownCompleted) {
mDecodeLooper->unregisterHandler(mCodec->id());
if (mDecodeLooper != looper()) {
@@ -208,6 +209,12 @@
}
looper()->stop();
+ } else if (what == ACodec::kWhatError) {
+ ALOGE("something went wrong, codec reported an error.");
+
+ printf("E\n");
+
+ (new AMessage(kWhatStop, id()))->post();
}
break;
}
@@ -360,7 +367,7 @@
buffer->meta()->setInt32("csd", true);
mCSD.push(buffer);
- msg->setObject("csd", buffer);
+ msg->setBuffer("csd", buffer);
} else if (meta->findData(kKeyESDS, &type, &data, &size)) {
ESDS esds((const char *)data, size);
CHECK_EQ(esds.InitCheck(), (status_t)OK);
@@ -410,9 +417,8 @@
return;
}
- sp<RefBase> obj;
- CHECK(msg->findObject("buffer", &obj));
- sp<ABuffer> outBuffer = static_cast<ABuffer *>(obj.get());
+ sp<ABuffer> outBuffer;
+ CHECK(msg->findBuffer("buffer", &outBuffer));
if (mCSDIndex < mCSD.size()) {
outBuffer = mCSD.editItemAt(mCSDIndex++);
@@ -511,15 +517,14 @@
}
}
- reply->setObject("buffer", outBuffer);
+ reply->setBuffer("buffer", outBuffer);
reply->post();
}
void onDrainThisBuffer(const sp<AMessage> &msg) {
- sp<RefBase> obj;
- CHECK(msg->findObject("buffer", &obj));
+ sp<ABuffer> buffer;
+ CHECK(msg->findBuffer("buffer", &buffer));
- sp<ABuffer> buffer = static_cast<ABuffer *>(obj.get());
mTotalBytesReceived += buffer->size();
sp<AMessage> reply;
diff --git a/cmds/stagefright/stagefright.cpp b/cmds/stagefright/stagefright.cpp
index 7cb8f62..dab2e0f 100644
--- a/cmds/stagefright/stagefright.cpp
+++ b/cmds/stagefright/stagefright.cpp
@@ -55,9 +55,7 @@
#include <fcntl.h>
#include <gui/SurfaceTextureClient.h>
-
-#include <surfaceflinger/ISurfaceComposer.h>
-#include <surfaceflinger/SurfaceComposerClient.h>
+#include <gui/SurfaceComposerClient.h>
using namespace android;
diff --git a/cmds/stagefright/stream.cpp b/cmds/stagefright/stream.cpp
index 0d6c738..efa1445 100644
--- a/cmds/stagefright/stream.cpp
+++ b/cmds/stagefright/stream.cpp
@@ -32,8 +32,7 @@
#include <binder/IServiceManager.h>
#include <media/IMediaPlayerService.h>
-#include <surfaceflinger/ISurfaceComposer.h>
-#include <surfaceflinger/SurfaceComposerClient.h>
+#include <gui/SurfaceComposerClient.h>
#include <fcntl.h>
diff --git a/drm/common/DrmEngineBase.cpp b/drm/common/DrmEngineBase.cpp
index 9b16c36..1c345a2 100644
--- a/drm/common/DrmEngineBase.cpp
+++ b/drm/common/DrmEngineBase.cpp
@@ -120,13 +120,23 @@
}
status_t DrmEngineBase::openDecryptSession(
- int uniqueId, DecryptHandle* decryptHandle, int fd, off64_t offset, off64_t length) {
- return onOpenDecryptSession(uniqueId, decryptHandle, fd, offset, length);
+ int uniqueId, DecryptHandle* decryptHandle,
+ int fd, off64_t offset, off64_t length, const char* mime) {
+
+ if (!mime || mime[0] == '\0') {
+ return onOpenDecryptSession(uniqueId, decryptHandle, fd, offset, length);
+ }
+
+ return onOpenDecryptSession(uniqueId, decryptHandle, fd, offset, length, mime);
}
status_t DrmEngineBase::openDecryptSession(
- int uniqueId, DecryptHandle* decryptHandle, const char* uri) {
- return onOpenDecryptSession(uniqueId, decryptHandle, uri);
+ int uniqueId, DecryptHandle* decryptHandle,
+ const char* uri, const char* mime) {
+ if (!mime || mime[0] == '\0') {
+ return onOpenDecryptSession(uniqueId, decryptHandle, uri);
+ }
+ return onOpenDecryptSession(uniqueId, decryptHandle, uri, mime);
}
status_t DrmEngineBase::closeDecryptSession(int uniqueId, DecryptHandle* decryptHandle) {
diff --git a/drm/common/IDrmManagerService.cpp b/drm/common/IDrmManagerService.cpp
index 3ed8ade..43f64f2 100644
--- a/drm/common/IDrmManagerService.cpp
+++ b/drm/common/IDrmManagerService.cpp
@@ -600,7 +600,7 @@
}
DecryptHandle* BpDrmManagerService::openDecryptSession(
- int uniqueId, int fd, off64_t offset, off64_t length) {
+ int uniqueId, int fd, off64_t offset, off64_t length, const char* mime) {
ALOGV("Entering BpDrmManagerService::openDecryptSession");
Parcel data, reply;
@@ -609,6 +609,11 @@
data.writeFileDescriptor(fd);
data.writeInt64(offset);
data.writeInt64(length);
+ String8 mimeType;
+ if (mime) {
+ mimeType = mime;
+ }
+ data.writeString8(mimeType);
remote()->transact(OPEN_DECRYPT_SESSION, data, &reply);
@@ -620,13 +625,20 @@
return handle;
}
-DecryptHandle* BpDrmManagerService::openDecryptSession(int uniqueId, const char* uri) {
- ALOGV("Entering BpDrmManagerService::openDecryptSession");
+DecryptHandle* BpDrmManagerService::openDecryptSession(
+ int uniqueId, const char* uri, const char* mime) {
+
+ ALOGV("Entering BpDrmManagerService::openDecryptSession: mime=%s", mime? mime: "NULL");
Parcel data, reply;
data.writeInterfaceToken(IDrmManagerService::getInterfaceDescriptor());
data.writeInt32(uniqueId);
data.writeString8(String8(uri));
+ String8 mimeType;
+ if (mime) {
+ mimeType = mime;
+ }
+ data.writeString8(mimeType);
remote()->transact(OPEN_DECRYPT_SESSION_FROM_URI, data, &reply);
@@ -1265,8 +1277,10 @@
const off64_t offset = data.readInt64();
const off64_t length = data.readInt64();
+ const String8 mime = data.readString8();
+
DecryptHandle* handle
- = openDecryptSession(uniqueId, fd, offset, length);
+ = openDecryptSession(uniqueId, fd, offset, length, mime.string());
if (NULL != handle) {
writeDecryptHandleToParcelData(handle, reply);
@@ -1283,8 +1297,9 @@
const int uniqueId = data.readInt32();
const String8 uri = data.readString8();
+ const String8 mime = data.readString8();
- DecryptHandle* handle = openDecryptSession(uniqueId, uri.string());
+ DecryptHandle* handle = openDecryptSession(uniqueId, uri.string(), mime.string());
if (NULL != handle) {
writeDecryptHandleToParcelData(handle, reply);
diff --git a/drm/drmserver/DrmManager.cpp b/drm/drmserver/DrmManager.cpp
index 3abf3d3..999295a 100644
--- a/drm/drmserver/DrmManager.cpp
+++ b/drm/drmserver/DrmManager.cpp
@@ -426,7 +426,9 @@
return DRM_NO_ERROR;
}
-DecryptHandle* DrmManager::openDecryptSession(int uniqueId, int fd, off64_t offset, off64_t length) {
+DecryptHandle* DrmManager::openDecryptSession(
+ int uniqueId, int fd, off64_t offset, off64_t length, const char* mime) {
+
Mutex::Autolock _l(mDecryptLock);
status_t result = DRM_ERROR_CANNOT_HANDLE;
Vector<String8> plugInIdList = mPlugInManager.getPlugInIdList();
@@ -438,7 +440,7 @@
for (unsigned int index = 0; index < plugInIdList.size(); index++) {
String8 plugInId = plugInIdList.itemAt(index);
IDrmEngine& rDrmEngine = mPlugInManager.getPlugIn(plugInId);
- result = rDrmEngine.openDecryptSession(uniqueId, handle, fd, offset, length);
+ result = rDrmEngine.openDecryptSession(uniqueId, handle, fd, offset, length, mime);
if (DRM_NO_ERROR == result) {
++mDecryptSessionId;
@@ -453,7 +455,8 @@
return handle;
}
-DecryptHandle* DrmManager::openDecryptSession(int uniqueId, const char* uri) {
+DecryptHandle* DrmManager::openDecryptSession(
+ int uniqueId, const char* uri, const char* mime) {
Mutex::Autolock _l(mDecryptLock);
status_t result = DRM_ERROR_CANNOT_HANDLE;
Vector<String8> plugInIdList = mPlugInManager.getPlugInIdList();
@@ -465,7 +468,7 @@
for (unsigned int index = 0; index < plugInIdList.size(); index++) {
String8 plugInId = plugInIdList.itemAt(index);
IDrmEngine& rDrmEngine = mPlugInManager.getPlugIn(plugInId);
- result = rDrmEngine.openDecryptSession(uniqueId, handle, uri);
+ result = rDrmEngine.openDecryptSession(uniqueId, handle, uri, mime);
if (DRM_NO_ERROR == result) {
++mDecryptSessionId;
diff --git a/drm/drmserver/DrmManagerService.cpp b/drm/drmserver/DrmManagerService.cpp
index df17ac5..8ba0203 100644
--- a/drm/drmserver/DrmManagerService.cpp
+++ b/drm/drmserver/DrmManagerService.cpp
@@ -159,12 +159,18 @@
status_t DrmManagerService::consumeRights(
int uniqueId, DecryptHandle* decryptHandle, int action, bool reserve) {
ALOGV("Entering consumeRights");
+ if (!isProtectedCallAllowed()) {
+ return DRM_ERROR_NO_PERMISSION;
+ }
return mDrmManager->consumeRights(uniqueId, decryptHandle, action, reserve);
}
status_t DrmManagerService::setPlaybackStatus(
int uniqueId, DecryptHandle* decryptHandle, int playbackStatus, int64_t position) {
ALOGV("Entering setPlaybackStatus");
+ if (!isProtectedCallAllowed()) {
+ return DRM_ERROR_NO_PERMISSION;
+ }
return mDrmManager->setPlaybackStatus(uniqueId, decryptHandle, playbackStatus, position);
}
@@ -208,20 +214,20 @@
}
DecryptHandle* DrmManagerService::openDecryptSession(
- int uniqueId, int fd, off64_t offset, off64_t length) {
+ int uniqueId, int fd, off64_t offset, off64_t length, const char* mime) {
ALOGV("Entering DrmManagerService::openDecryptSession");
if (isProtectedCallAllowed()) {
- return mDrmManager->openDecryptSession(uniqueId, fd, offset, length);
+ return mDrmManager->openDecryptSession(uniqueId, fd, offset, length, mime);
}
return NULL;
}
DecryptHandle* DrmManagerService::openDecryptSession(
- int uniqueId, const char* uri) {
+ int uniqueId, const char* uri, const char* mime) {
ALOGV("Entering DrmManagerService::openDecryptSession with uri");
if (isProtectedCallAllowed()) {
- return mDrmManager->openDecryptSession(uniqueId, uri);
+ return mDrmManager->openDecryptSession(uniqueId, uri, mime);
}
return NULL;
@@ -229,12 +235,18 @@
status_t DrmManagerService::closeDecryptSession(int uniqueId, DecryptHandle* decryptHandle) {
ALOGV("Entering closeDecryptSession");
+ if (!isProtectedCallAllowed()) {
+ return DRM_ERROR_NO_PERMISSION;
+ }
return mDrmManager->closeDecryptSession(uniqueId, decryptHandle);
}
status_t DrmManagerService::initializeDecryptUnit(int uniqueId, DecryptHandle* decryptHandle,
int decryptUnitId, const DrmBuffer* headerInfo) {
ALOGV("Entering initializeDecryptUnit");
+ if (!isProtectedCallAllowed()) {
+ return DRM_ERROR_NO_PERMISSION;
+ }
return mDrmManager->initializeDecryptUnit(uniqueId,decryptHandle, decryptUnitId, headerInfo);
}
@@ -242,18 +254,27 @@
int uniqueId, DecryptHandle* decryptHandle, int decryptUnitId,
const DrmBuffer* encBuffer, DrmBuffer** decBuffer, DrmBuffer* IV) {
ALOGV("Entering decrypt");
+ if (!isProtectedCallAllowed()) {
+ return DRM_ERROR_NO_PERMISSION;
+ }
return mDrmManager->decrypt(uniqueId, decryptHandle, decryptUnitId, encBuffer, decBuffer, IV);
}
status_t DrmManagerService::finalizeDecryptUnit(
int uniqueId, DecryptHandle* decryptHandle, int decryptUnitId) {
ALOGV("Entering finalizeDecryptUnit");
+ if (!isProtectedCallAllowed()) {
+ return DRM_ERROR_NO_PERMISSION;
+ }
return mDrmManager->finalizeDecryptUnit(uniqueId, decryptHandle, decryptUnitId);
}
ssize_t DrmManagerService::pread(int uniqueId, DecryptHandle* decryptHandle,
void* buffer, ssize_t numBytes, off64_t offset) {
ALOGV("Entering pread");
+ if (!isProtectedCallAllowed()) {
+ return DRM_ERROR_NO_PERMISSION;
+ }
return mDrmManager->pread(uniqueId, decryptHandle, buffer, numBytes, offset);
}
diff --git a/drm/drmserver/main_drmserver.cpp b/drm/drmserver/main_drmserver.cpp
index ed42818..434d561 100644
--- a/drm/drmserver/main_drmserver.cpp
+++ b/drm/drmserver/main_drmserver.cpp
@@ -14,15 +14,13 @@
* limitations under the License.
*/
-#include <sys/types.h>
-#include <unistd.h>
-#include <grp.h>
+#define LOG_TAG "drmserver"
+//#define LOG_NDEBUG 0
#include <binder/IPCThreadState.h>
#include <binder/ProcessState.h>
#include <binder/IServiceManager.h>
#include <utils/Log.h>
-#include <private/android_filesystem_config.h>
#include <DrmManagerService.h>
@@ -32,7 +30,7 @@
{
sp<ProcessState> proc(ProcessState::self());
sp<IServiceManager> sm = defaultServiceManager();
- ALOGI("ServiceManager: %p", sm.get());
+ ALOGV("ServiceManager: %p", sm.get());
DrmManagerService::instantiate();
ProcessState::self()->startThreadPool();
IPCThreadState::self()->joinThreadPool();
diff --git a/drm/libdrmframework/DrmManagerClient.cpp b/drm/libdrmframework/DrmManagerClient.cpp
index c9c0d57..8768c08 100644
--- a/drm/libdrmframework/DrmManagerClient.cpp
+++ b/drm/libdrmframework/DrmManagerClient.cpp
@@ -116,12 +116,18 @@
return mDrmManagerClientImpl->getAllSupportInfo(mUniqueId, length, drmSupportInfoArray);
}
-sp<DecryptHandle> DrmManagerClient::openDecryptSession(int fd, off64_t offset, off64_t length) {
- return mDrmManagerClientImpl->openDecryptSession(mUniqueId, fd, offset, length);
+sp<DecryptHandle> DrmManagerClient::openDecryptSession(
+ int fd, off64_t offset, off64_t length, const char* mime) {
+
+ return mDrmManagerClientImpl->openDecryptSession(
+ mUniqueId, fd, offset, length, mime);
}
-sp<DecryptHandle> DrmManagerClient::openDecryptSession(const char* uri) {
- return mDrmManagerClientImpl->openDecryptSession(mUniqueId, uri);
+sp<DecryptHandle> DrmManagerClient::openDecryptSession(
+ const char* uri, const char* mime) {
+
+ return mDrmManagerClientImpl->openDecryptSession(
+ mUniqueId, uri, mime);
}
status_t DrmManagerClient::closeDecryptSession(sp<DecryptHandle> &decryptHandle) {
diff --git a/drm/libdrmframework/DrmManagerClientImpl.cpp b/drm/libdrmframework/DrmManagerClientImpl.cpp
index b222b8f..fb0439e 100644
--- a/drm/libdrmframework/DrmManagerClientImpl.cpp
+++ b/drm/libdrmframework/DrmManagerClientImpl.cpp
@@ -255,15 +255,19 @@
}
sp<DecryptHandle> DrmManagerClientImpl::openDecryptSession(
- int uniqueId, int fd, off64_t offset, off64_t length) {
- return getDrmManagerService()->openDecryptSession(uniqueId, fd, offset, length);
+ int uniqueId, int fd, off64_t offset,
+ off64_t length, const char* mime) {
+
+ return getDrmManagerService()->openDecryptSession(
+ uniqueId, fd, offset, length, mime);
}
sp<DecryptHandle> DrmManagerClientImpl::openDecryptSession(
- int uniqueId, const char* uri) {
+ int uniqueId, const char* uri, const char* mime) {
+
DecryptHandle* handle = NULL;
if (NULL != uri) {
- handle = getDrmManagerService()->openDecryptSession(uniqueId, uri);
+ handle = getDrmManagerService()->openDecryptSession(uniqueId, uri, mime);
}
return handle;
}
diff --git a/drm/libdrmframework/include/DrmManager.h b/drm/libdrmframework/include/DrmManager.h
index ac2b946..c9167d4 100644
--- a/drm/libdrmframework/include/DrmManager.h
+++ b/drm/libdrmframework/include/DrmManager.h
@@ -111,9 +111,10 @@
status_t getAllSupportInfo(int uniqueId, int* length, DrmSupportInfo** drmSupportInfoArray);
- DecryptHandle* openDecryptSession(int uniqueId, int fd, off64_t offset, off64_t length);
+ DecryptHandle* openDecryptSession(
+ int uniqueId, int fd, off64_t offset, off64_t length, const char* mime);
- DecryptHandle* openDecryptSession(int uniqueId, const char* uri);
+ DecryptHandle* openDecryptSession(int uniqueId, const char* uri, const char* mime);
status_t closeDecryptSession(int uniqueId, DecryptHandle* decryptHandle);
diff --git a/drm/libdrmframework/include/DrmManagerClientImpl.h b/drm/libdrmframework/include/DrmManagerClientImpl.h
index e3338d9..2aa493f 100644
--- a/drm/libdrmframework/include/DrmManagerClientImpl.h
+++ b/drm/libdrmframework/include/DrmManagerClientImpl.h
@@ -300,20 +300,24 @@
* @param[in] fd File descriptor of the protected content to be decrypted
* @param[in] offset Start position of the content
* @param[in] length The length of the protected content
+ * @param[in] mime The mime type of the protected content if it is not NULL or empty
* @return
* Handle for the decryption session
*/
- sp<DecryptHandle> openDecryptSession(int uniqueId, int fd, off64_t offset, off64_t length);
+ sp<DecryptHandle> openDecryptSession(
+ int uniqueId, int fd, off64_t offset, off64_t length, const char* mime);
/**
* Open the decrypt session to decrypt the given protected content
*
* @param[in] uniqueId Unique identifier for a session
* @param[in] uri Path of the protected content to be decrypted
+ * @param[in] mime The mime type of the protected content if it is not NULL or empty
* @return
* Handle for the decryption session
*/
- sp<DecryptHandle> openDecryptSession(int uniqueId, const char* uri);
+ sp<DecryptHandle> openDecryptSession(
+ int uniqueId, const char* uri, const char* mime);
/**
* Close the decrypt session for the given handle
diff --git a/drm/libdrmframework/include/DrmManagerService.h b/drm/libdrmframework/include/DrmManagerService.h
index 9cb5804..1a8c2ae 100644
--- a/drm/libdrmframework/include/DrmManagerService.h
+++ b/drm/libdrmframework/include/DrmManagerService.h
@@ -98,9 +98,11 @@
status_t getAllSupportInfo(int uniqueId, int* length, DrmSupportInfo** drmSupportInfoArray);
- DecryptHandle* openDecryptSession(int uniqueId, int fd, off64_t offset, off64_t length);
+ DecryptHandle* openDecryptSession(
+ int uniqueId, int fd, off64_t offset, off64_t length, const char *mime);
- DecryptHandle* openDecryptSession(int uniqueId, const char* uri);
+ DecryptHandle* openDecryptSession(
+ int uniqueId, const char* uri, const char* mime);
status_t closeDecryptSession(int uniqueId, DecryptHandle* decryptHandle);
diff --git a/drm/libdrmframework/include/IDrmManagerService.h b/drm/libdrmframework/include/IDrmManagerService.h
index b9618bb..a7d21c5 100644
--- a/drm/libdrmframework/include/IDrmManagerService.h
+++ b/drm/libdrmframework/include/IDrmManagerService.h
@@ -139,9 +139,12 @@
virtual status_t getAllSupportInfo(
int uniqueId, int* length, DrmSupportInfo** drmSupportInfoArray) = 0;
- virtual DecryptHandle* openDecryptSession(int uniqueId, int fd, off64_t offset, off64_t length) = 0;
+ virtual DecryptHandle* openDecryptSession(
+ int uniqueId, int fd, off64_t offset,
+ off64_t length, const char* mime) = 0;
- virtual DecryptHandle* openDecryptSession(int uniqueId, const char* uri) = 0;
+ virtual DecryptHandle* openDecryptSession(
+ int uniqueId, const char* uri, const char* mime) = 0;
virtual status_t closeDecryptSession(int uniqueId, DecryptHandle* decryptHandle) = 0;
@@ -222,9 +225,12 @@
virtual status_t getAllSupportInfo(
int uniqueId, int* length, DrmSupportInfo** drmSupportInfoArray);
- virtual DecryptHandle* openDecryptSession(int uniqueId, int fd, off64_t offset, off64_t length);
+ virtual DecryptHandle* openDecryptSession(
+ int uniqueId, int fd, off64_t offset, off64_t length,
+ const char* mime);
- virtual DecryptHandle* openDecryptSession(int uniqueId, const char* uri);
+ virtual DecryptHandle* openDecryptSession(
+ int uniqueId, const char* uri, const char* mime);
virtual status_t closeDecryptSession(int uniqueId, DecryptHandle* decryptHandle);
diff --git a/drm/libdrmframework/plugins/common/include/DrmEngineBase.h b/drm/libdrmframework/plugins/common/include/DrmEngineBase.h
index 4a5afcf..08f6e6d 100644
--- a/drm/libdrmframework/plugins/common/include/DrmEngineBase.h
+++ b/drm/libdrmframework/plugins/common/include/DrmEngineBase.h
@@ -80,10 +80,12 @@
DrmSupportInfo* getSupportInfo(int uniqueId);
status_t openDecryptSession(
- int uniqueId, DecryptHandle* decryptHandle, int fd, off64_t offset, off64_t length);
+ int uniqueId, DecryptHandle* decryptHandle,
+ int fd, off64_t offset, off64_t length, const char* mime);
status_t openDecryptSession(
- int uniqueId, DecryptHandle* decryptHandle, const char* uri);
+ int uniqueId, DecryptHandle* decryptHandle,
+ const char* uri, const char* mime);
status_t closeDecryptSession(int uniqueId, DecryptHandle* decryptHandle);
@@ -375,7 +377,29 @@
* DRM_ERROR_CANNOT_HANDLE for failure and DRM_NO_ERROR for success
*/
virtual status_t onOpenDecryptSession(
- int uniqueId, DecryptHandle* decryptHandle, int fd, off64_t offset, off64_t length) = 0;
+ int uniqueId, DecryptHandle* decryptHandle,
+ int fd, off64_t offset, off64_t length) = 0;
+
+ /**
+ * Open the decrypt session to decrypt the given protected content
+ *
+ * @param[in] uniqueId Unique identifier for a session
+ * @param[in] decryptHandle Handle for the current decryption session
+ * @param[in] fd File descriptor of the protected content to be decrypted
+ * @param[in] offset Start position of the content
+ * @param[in] length The length of the protected content
+ * @param[in] mime Mime type of the protected content
+ * drm plugin may do some optimization since the mime type is known.
+ * @return
+ * DRM_ERROR_CANNOT_HANDLE for failure and DRM_NO_ERROR for success
+ */
+ virtual status_t onOpenDecryptSession(
+ int uniqueId, DecryptHandle* decryptHandle,
+ int fd, off64_t offset, off64_t length,
+ const char* mime) {
+
+ return DRM_ERROR_CANNOT_HANDLE;
+ }
/**
* Open the decrypt session to decrypt the given protected content
@@ -387,7 +411,26 @@
* DRM_ERROR_CANNOT_HANDLE for failure and DRM_NO_ERROR for success
*/
virtual status_t onOpenDecryptSession(
- int uniqueId, DecryptHandle* decryptHandle, const char* uri) = 0;
+ int uniqueId, DecryptHandle* decryptHandle,
+ const char* uri) = 0;
+
+ /**
+ * Open the decrypt session to decrypt the given protected content
+ *
+ * @param[in] uniqueId Unique identifier for a session
+ * @param[in] decryptHandle Handle for the current decryption session
+ * @param[in] uri Path of the protected content to be decrypted
+ * @param[in] mime Mime type of the protected content. The corresponding
+ * drm plugin may do some optimization since the mime type is known.
+ * @return
+ * DRM_ERROR_CANNOT_HANDLE for failure and DRM_NO_ERROR for success
+ */
+ virtual status_t onOpenDecryptSession(
+ int uniqueId, DecryptHandle* decryptHandle,
+ const char* uri, const char* mime) {
+
+ return DRM_ERROR_CANNOT_HANDLE;
+ }
/**
* Close the decrypt session for the given handle
diff --git a/drm/libdrmframework/plugins/common/include/IDrmEngine.h b/drm/libdrmframework/plugins/common/include/IDrmEngine.h
index 77460f6..dcf5977 100644
--- a/drm/libdrmframework/plugins/common/include/IDrmEngine.h
+++ b/drm/libdrmframework/plugins/common/include/IDrmEngine.h
@@ -320,11 +320,14 @@
* @param[in] fd File descriptor of the protected content to be decrypted
* @param[in] offset Start position of the content
* @param[in] length The length of the protected content
+ * @param[in] mime Mime type of the protected content if it is
+ * not NULL or empty
* @return
* DRM_ERROR_CANNOT_HANDLE for failure and DRM_NO_ERROR for success
*/
virtual status_t openDecryptSession(
- int uniqueId, DecryptHandle* decryptHandle, int fd, off64_t offset, off64_t length) = 0;
+ int uniqueId, DecryptHandle* decryptHandle,
+ int fd, off64_t offset, off64_t length, const char* mime) = 0;
/**
* Open the decrypt session to decrypt the given protected content
@@ -332,11 +335,14 @@
* @param[in] uniqueId Unique identifier for a session
* @param[in] decryptHandle Handle for the current decryption session
* @param[in] uri Path of the protected content to be decrypted
+ * @param[in] mime Mime type of the protected content if it is
+ * not NULL or empty
* @return
* DRM_ERROR_CANNOT_HANDLE for failure and DRM_NO_ERROR for success
*/
virtual status_t openDecryptSession(
- int uniqueId, DecryptHandle* decryptHandle, const char* uri) = 0;
+ int uniqueId, DecryptHandle* decryptHandle,
+ const char* uri, const char* mime) = 0;
/**
* Close the decrypt session for the given handle
diff --git a/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/include/FwdLockEngine.h b/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/include/FwdLockEngine.h
index 34804cf..c0e408e 100644
--- a/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/include/FwdLockEngine.h
+++ b/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/include/FwdLockEngine.h
@@ -499,6 +499,14 @@
private:
+ static const String8 Description;
+ static const String8 FileSuffixes[];
+ static const String8 MimeTypes[];
+ static bool IsFileSuffixSupported(const String8& suffix);
+ static bool IsMimeTypeSupported(const String8& mime);
+ static void AddSupportedMimeTypes(DrmSupportInfo *info);
+ static void AddSupportedFileSuffixes(DrmSupportInfo *info);
+
/**
* Session Class for Forward Lock Conversion. An object of this class is created
* for every conversion.
diff --git a/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/include/FwdLockEngineConst.h b/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/include/FwdLockEngineConst.h
deleted file mode 100644
index da95d60..0000000
--- a/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/include/FwdLockEngineConst.h
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef __FWDLOCKENGINECONST_H__
-#define __FWDLOCKENGINECONST_H__
-
-namespace android {
-
-/**
- * Constants for forward Lock Engine used for exposing engine's capabilities.
- */
-#define FWDLOCK_EXTENSION_FL ("FL")
-#define FWDLOCK_DOTEXTENSION_FL (".fl")
-#define FWDLOCK_MIMETYPE_FL ("application/x-android-drm-fl")
-
-#define FWDLOCK_EXTENSION_DM ("DM")
-#define FWDLOCK_DOTEXTENSION_DM (".dm")
-#define FWDLOCK_MIMETYPE_DM ("application/vnd.oma.drm.message")
-
-#define FWDLOCK_DESCRIPTION ("OMA V1 Forward Lock")
-
-};
-
-#endif /* __FWDLOCKENGINECONST_H__ */
diff --git a/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/src/FwdLockEngine.cpp b/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/src/FwdLockEngine.cpp
index 0273a4b..4b1b40e 100644
--- a/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/src/FwdLockEngine.cpp
+++ b/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/src/FwdLockEngine.cpp
@@ -35,7 +35,6 @@
#include "FwdLockConv.h"
#include "FwdLockFile.h"
#include "FwdLockGlue.h"
-#include "FwdLockEngineConst.h"
#include "MimeTypeUtil.h"
#undef LOG_TAG
@@ -160,6 +159,54 @@
return DRM_NO_ERROR;
}
+// make sure that lower-case letters are used.
+const String8 FwdLockEngine::FileSuffixes[] = {
+ String8(".fl"),
+ String8(".dm"),
+};
+
+// make sure that lower-case letters are used.
+const String8 FwdLockEngine::MimeTypes[] = {
+ String8("application/x-android-drm-fl"),
+ String8("application/vnd.oma.drm.message"),
+};
+
+const String8 FwdLockEngine::Description("OMA V1 Forward Lock");
+
+void FwdLockEngine::AddSupportedMimeTypes(DrmSupportInfo *info) {
+ for (size_t i = 0, n = sizeof(MimeTypes)/sizeof(MimeTypes[0]); i < n; ++i) {
+ info->addMimeType(MimeTypes[i]);
+ }
+}
+
+void FwdLockEngine::AddSupportedFileSuffixes(DrmSupportInfo *info) {
+ for (size_t i = 0, n = sizeof(FileSuffixes)/sizeof(FileSuffixes[0]); i < n; ++i) {
+ info->addFileSuffix(FileSuffixes[i]);
+ }
+}
+
+bool FwdLockEngine::IsMimeTypeSupported(const String8& mime) {
+ String8 tmp(mime);
+ tmp.toLower();
+ for (size_t i = 0, n = sizeof(MimeTypes)/sizeof(MimeTypes[0]); i < n; ++i) {
+ if (tmp == MimeTypes[i]) {
+ return true;
+ }
+ }
+ return false;
+}
+
+bool FwdLockEngine::IsFileSuffixSupported(const String8& suffix) {
+ String8 tmp(suffix);
+ tmp.toLower();
+ for (size_t i = 0, n = sizeof(FileSuffixes)/sizeof(FileSuffixes[0]); i < n; ++i) {
+ if (tmp == FileSuffixes[i]) {
+ return true;
+ }
+ }
+ return false;
+}
+
DrmSupportInfo* FwdLockEngine::onGetSupportInfo(int uniqueId) {
DrmSupportInfo* pSupportInfo = new DrmSupportInfo();
@@ -167,12 +214,9 @@
// fill all Forward Lock mimetypes and extensions
if (NULL != pSupportInfo) {
- pSupportInfo->addMimeType(String8(FWDLOCK_MIMETYPE_FL));
- pSupportInfo->addFileSuffix(String8(FWDLOCK_DOTEXTENSION_FL));
- pSupportInfo->addMimeType(String8(FWDLOCK_MIMETYPE_DM));
- pSupportInfo->addFileSuffix(String8(FWDLOCK_DOTEXTENSION_DM));
-
- pSupportInfo->setDescription(String8(FWDLOCK_DESCRIPTION));
+ AddSupportedMimeTypes(pSupportInfo);
+ AddSupportedFileSuffixes(pSupportInfo);
+ pSupportInfo->setDescription(Description);
}
return pSupportInfo;
@@ -182,14 +226,7 @@
bool result = false;
String8 extString = path.getPathExtension();
-
- extString.toLower();
-
- if ((extString == String8(FWDLOCK_DOTEXTENSION_FL)) ||
- (extString == String8(FWDLOCK_DOTEXTENSION_DM))) {
- result = true;
- }
- return result;
+ return IsFileSuffixSupported(extString);
}
DrmInfoStatus* FwdLockEngine::onProcessDrmInfo(int uniqueId, const DrmInfo* drmInfo) {
@@ -299,8 +336,6 @@
LOG_VERBOSE("FwdLockEngine::onGetDrmObjectType");
- mimeStr.toLower();
-
/* Checks whether
* 1. path and mime type both are not empty strings (meaning unavailable) else content is unknown
* 2. if one of them is empty string and if other is known then its a DRM Content Object.
@@ -308,8 +343,7 @@
* (regardless of the relation between them to make it compatible with other DRM Engines)
*/
if (((0 == path.length()) || onCanHandle(uniqueId, path)) &&
- ((0 == mimeType.length()) || ((mimeStr == String8(FWDLOCK_MIMETYPE_FL)) ||
- (mimeStr == String8(FWDLOCK_MIMETYPE_DM)))) && (mimeType != path) ) {
+ ((0 == mimeType.length()) || IsMimeTypeSupported(mimeType)) && (mimeType != path) ) {
return DrmObjectType::CONTENT;
}
diff --git a/drm/libdrmframework/plugins/forward-lock/internal-format/decoder/FwdLockFile.c b/drm/libdrmframework/plugins/forward-lock/internal-format/decoder/FwdLockFile.c
index dacf00e..365bdec 100644
--- a/drm/libdrmframework/plugins/forward-lock/internal-format/decoder/FwdLockFile.c
+++ b/drm/libdrmframework/plugins/forward-lock/internal-format/decoder/FwdLockFile.c
@@ -14,6 +14,7 @@
* limitations under the License.
*/
+#include <utils/Log.h>
#include <assert.h>
#include <errno.h>
#include <fcntl.h>
@@ -107,6 +108,7 @@
}
pthread_mutex_unlock(&sessionAcquisitionMutex);
if (i == MAX_NUM_SESSIONS) {
+ ALOGE("Too many sessions opened at the same time");
errno = ENFILE;
}
}
@@ -293,7 +295,12 @@
int FwdLockFile_open(const char *pFilename) {
int fileDesc = open(pFilename, O_RDONLY);
- if (fileDesc >= 0 && FwdLockFile_attach(fileDesc) < 0) {
+ if (fileDesc < 0) {
+ ALOGE("failed to open file '%s': %s", pFilename, strerror(errno));
+ return fileDesc;
+ }
+
+ if (FwdLockFile_attach(fileDesc) < 0) {
(void)close(fileDesc);
fileDesc = -1;
}
diff --git a/drm/libdrmframework/plugins/passthru/src/DrmPassthruPlugIn.cpp b/drm/libdrmframework/plugins/passthru/src/DrmPassthruPlugIn.cpp
index 0ffc0a7..a3eac3e 100644
--- a/drm/libdrmframework/plugins/passthru/src/DrmPassthruPlugIn.cpp
+++ b/drm/libdrmframework/plugins/passthru/src/DrmPassthruPlugIn.cpp
@@ -14,7 +14,7 @@
* limitations under the License.
*/
-#define LOG_NDEBUG 0
+//#define LOG_NDEBUG 0
#define LOG_TAG "DrmPassthruPlugIn"
#include <utils/Log.h>
@@ -58,7 +58,7 @@
DrmConstraints* DrmPassthruPlugIn::onGetConstraints(
int uniqueId, const String8* path, int action) {
- ALOGD("DrmPassthruPlugIn::onGetConstraints From Path: %d", uniqueId);
+ ALOGV("DrmPassthruPlugIn::onGetConstraints From Path: %d", uniqueId);
DrmConstraints* drmConstraints = new DrmConstraints();
String8 value("dummy_available_time");
@@ -73,7 +73,7 @@
}
DrmInfoStatus* DrmPassthruPlugIn::onProcessDrmInfo(int uniqueId, const DrmInfo* drmInfo) {
- ALOGD("DrmPassthruPlugIn::onProcessDrmInfo - Enter : %d", uniqueId);
+ ALOGV("DrmPassthruPlugIn::onProcessDrmInfo - Enter : %d", uniqueId);
DrmInfoStatus* drmInfoStatus = NULL;
if (NULL != drmInfo) {
switch (drmInfo->getInfoType()) {
@@ -102,28 +102,28 @@
}
}
}
- ALOGD("DrmPassthruPlugIn::onProcessDrmInfo - Exit");
+ ALOGV("DrmPassthruPlugIn::onProcessDrmInfo - Exit");
return drmInfoStatus;
}
status_t DrmPassthruPlugIn::onSetOnInfoListener(
int uniqueId, const IDrmEngine::OnInfoListener* infoListener) {
- ALOGD("DrmPassthruPlugIn::onSetOnInfoListener : %d", uniqueId);
+ ALOGV("DrmPassthruPlugIn::onSetOnInfoListener : %d", uniqueId);
return DRM_NO_ERROR;
}
status_t DrmPassthruPlugIn::onInitialize(int uniqueId) {
- ALOGD("DrmPassthruPlugIn::onInitialize : %d", uniqueId);
+ ALOGV("DrmPassthruPlugIn::onInitialize : %d", uniqueId);
return DRM_NO_ERROR;
}
status_t DrmPassthruPlugIn::onTerminate(int uniqueId) {
- ALOGD("DrmPassthruPlugIn::onTerminate : %d", uniqueId);
+ ALOGV("DrmPassthruPlugIn::onTerminate : %d", uniqueId);
return DRM_NO_ERROR;
}
DrmSupportInfo* DrmPassthruPlugIn::onGetSupportInfo(int uniqueId) {
- ALOGD("DrmPassthruPlugIn::onGetSupportInfo : %d", uniqueId);
+ ALOGV("DrmPassthruPlugIn::onGetSupportInfo : %d", uniqueId);
DrmSupportInfo* drmSupportInfo = new DrmSupportInfo();
// Add mimetype's
drmSupportInfo->addMimeType(String8("application/vnd.passthru.drm"));
@@ -136,12 +136,12 @@
status_t DrmPassthruPlugIn::onSaveRights(int uniqueId, const DrmRights& drmRights,
const String8& rightsPath, const String8& contentPath) {
- ALOGD("DrmPassthruPlugIn::onSaveRights : %d", uniqueId);
+ ALOGV("DrmPassthruPlugIn::onSaveRights : %d", uniqueId);
return DRM_NO_ERROR;
}
DrmInfo* DrmPassthruPlugIn::onAcquireDrmInfo(int uniqueId, const DrmInfoRequest* drmInfoRequest) {
- ALOGD("DrmPassthruPlugIn::onAcquireDrmInfo : %d", uniqueId);
+ ALOGV("DrmPassthruPlugIn::onAcquireDrmInfo : %d", uniqueId);
DrmInfo* drmInfo = NULL;
if (NULL != drmInfoRequest) {
@@ -157,65 +157,65 @@
}
bool DrmPassthruPlugIn::onCanHandle(int uniqueId, const String8& path) {
- ALOGD("DrmPassthruPlugIn::canHandle: %s ", path.string());
+ ALOGV("DrmPassthruPlugIn::canHandle: %s ", path.string());
String8 extension = path.getPathExtension();
extension.toLower();
return (String8(".passthru") == extension);
}
String8 DrmPassthruPlugIn::onGetOriginalMimeType(int uniqueId, const String8& path) {
- ALOGD("DrmPassthruPlugIn::onGetOriginalMimeType() : %d", uniqueId);
+ ALOGV("DrmPassthruPlugIn::onGetOriginalMimeType() : %d", uniqueId);
return String8("video/passthru");
}
int DrmPassthruPlugIn::onGetDrmObjectType(
int uniqueId, const String8& path, const String8& mimeType) {
- ALOGD("DrmPassthruPlugIn::onGetDrmObjectType() : %d", uniqueId);
+ ALOGV("DrmPassthruPlugIn::onGetDrmObjectType() : %d", uniqueId);
return DrmObjectType::UNKNOWN;
}
int DrmPassthruPlugIn::onCheckRightsStatus(int uniqueId, const String8& path, int action) {
- ALOGD("DrmPassthruPlugIn::onCheckRightsStatus() : %d", uniqueId);
+ ALOGV("DrmPassthruPlugIn::onCheckRightsStatus() : %d", uniqueId);
int rightsStatus = RightsStatus::RIGHTS_VALID;
return rightsStatus;
}
status_t DrmPassthruPlugIn::onConsumeRights(int uniqueId, DecryptHandle* decryptHandle,
int action, bool reserve) {
- ALOGD("DrmPassthruPlugIn::onConsumeRights() : %d", uniqueId);
+ ALOGV("DrmPassthruPlugIn::onConsumeRights() : %d", uniqueId);
return DRM_NO_ERROR;
}
status_t DrmPassthruPlugIn::onSetPlaybackStatus(int uniqueId, DecryptHandle* decryptHandle,
int playbackStatus, int64_t position) {
- ALOGD("DrmPassthruPlugIn::onSetPlaybackStatus() : %d", uniqueId);
+ ALOGV("DrmPassthruPlugIn::onSetPlaybackStatus() : %d", uniqueId);
return DRM_NO_ERROR;
}
bool DrmPassthruPlugIn::onValidateAction(int uniqueId, const String8& path,
int action, const ActionDescription& description) {
- ALOGD("DrmPassthruPlugIn::onValidateAction() : %d", uniqueId);
+ ALOGV("DrmPassthruPlugIn::onValidateAction() : %d", uniqueId);
return true;
}
status_t DrmPassthruPlugIn::onRemoveRights(int uniqueId, const String8& path) {
- ALOGD("DrmPassthruPlugIn::onRemoveRights() : %d", uniqueId);
+ ALOGV("DrmPassthruPlugIn::onRemoveRights() : %d", uniqueId);
return DRM_NO_ERROR;
}
status_t DrmPassthruPlugIn::onRemoveAllRights(int uniqueId) {
- ALOGD("DrmPassthruPlugIn::onRemoveAllRights() : %d", uniqueId);
+ ALOGV("DrmPassthruPlugIn::onRemoveAllRights() : %d", uniqueId);
return DRM_NO_ERROR;
}
status_t DrmPassthruPlugIn::onOpenConvertSession(int uniqueId, int convertId) {
- ALOGD("DrmPassthruPlugIn::onOpenConvertSession() : %d", uniqueId);
+ ALOGV("DrmPassthruPlugIn::onOpenConvertSession() : %d", uniqueId);
return DRM_NO_ERROR;
}
DrmConvertedStatus* DrmPassthruPlugIn::onConvertData(
int uniqueId, int convertId, const DrmBuffer* inputData) {
- ALOGD("DrmPassthruPlugIn::onConvertData() : %d", uniqueId);
+ ALOGV("DrmPassthruPlugIn::onConvertData() : %d", uniqueId);
DrmBuffer* convertedData = NULL;
if (NULL != inputData && 0 < inputData->length) {
@@ -229,13 +229,13 @@
}
DrmConvertedStatus* DrmPassthruPlugIn::onCloseConvertSession(int uniqueId, int convertId) {
- ALOGD("DrmPassthruPlugIn::onCloseConvertSession() : %d", uniqueId);
+ ALOGV("DrmPassthruPlugIn::onCloseConvertSession() : %d", uniqueId);
return new DrmConvertedStatus(DrmConvertedStatus::STATUS_OK, NULL, 0 /*offset*/);
}
status_t DrmPassthruPlugIn::onOpenDecryptSession(
int uniqueId, DecryptHandle* decryptHandle, int fd, off64_t offset, off64_t length) {
- ALOGD("DrmPassthruPlugIn::onOpenDecryptSession() : %d", uniqueId);
+ ALOGV("DrmPassthruPlugIn::onOpenDecryptSession() : %d", uniqueId);
#ifdef ENABLE_PASSTHRU_DECRYPTION
decryptHandle->mimeType = String8("video/passthru");
@@ -254,7 +254,7 @@
}
status_t DrmPassthruPlugIn::onCloseDecryptSession(int uniqueId, DecryptHandle* decryptHandle) {
- ALOGD("DrmPassthruPlugIn::onCloseDecryptSession() : %d", uniqueId);
+ ALOGV("DrmPassthruPlugIn::onCloseDecryptSession() : %d", uniqueId);
if (NULL != decryptHandle) {
if (NULL != decryptHandle->decryptInfo) {
delete decryptHandle->decryptInfo; decryptHandle->decryptInfo = NULL;
@@ -266,34 +266,40 @@
status_t DrmPassthruPlugIn::onInitializeDecryptUnit(int uniqueId, DecryptHandle* decryptHandle,
int decryptUnitId, const DrmBuffer* headerInfo) {
- ALOGD("DrmPassthruPlugIn::onInitializeDecryptUnit() : %d", uniqueId);
+ ALOGV("DrmPassthruPlugIn::onInitializeDecryptUnit() : %d", uniqueId);
return DRM_NO_ERROR;
}
status_t DrmPassthruPlugIn::onDecrypt(int uniqueId, DecryptHandle* decryptHandle,
int decryptUnitId, const DrmBuffer* encBuffer, DrmBuffer** decBuffer, DrmBuffer* IV) {
- ALOGD("DrmPassthruPlugIn::onDecrypt() : %d", uniqueId);
+ ALOGV("DrmPassthruPlugIn::onDecrypt() : %d", uniqueId);
/**
* As a workaround implementation passthru would copy the given
* encrypted buffer as it is to decrypted buffer. Note, decBuffer
* memory has to be allocated by the caller.
*/
if (NULL != (*decBuffer) && 0 < (*decBuffer)->length) {
- memcpy((*decBuffer)->data, encBuffer->data, encBuffer->length);
- (*decBuffer)->length = encBuffer->length;
+ if ((*decBuffer)->length >= encBuffer->length) {
+ memcpy((*decBuffer)->data, encBuffer->data, encBuffer->length);
+ (*decBuffer)->length = encBuffer->length;
+ } else {
+ ALOGE("decBuffer size (%d) too small to hold %d bytes",
+ (*decBuffer)->length, encBuffer->length);
+ return DRM_ERROR_UNKNOWN;
+ }
}
return DRM_NO_ERROR;
}
status_t DrmPassthruPlugIn::onFinalizeDecryptUnit(
int uniqueId, DecryptHandle* decryptHandle, int decryptUnitId) {
- ALOGD("DrmPassthruPlugIn::onFinalizeDecryptUnit() : %d", uniqueId);
+ ALOGV("DrmPassthruPlugIn::onFinalizeDecryptUnit() : %d", uniqueId);
return DRM_NO_ERROR;
}
ssize_t DrmPassthruPlugIn::onPread(int uniqueId, DecryptHandle* decryptHandle,
void* buffer, ssize_t numBytes, off64_t offset) {
- ALOGD("DrmPassthruPlugIn::onPread() : %d", uniqueId);
+ ALOGV("DrmPassthruPlugIn::onPread() : %d", uniqueId);
return 0;
}
diff --git a/include/camera/Camera.h b/include/camera/Camera.h
index 234e165..3fedea0 100644
--- a/include/camera/Camera.h
+++ b/include/camera/Camera.h
@@ -72,7 +72,7 @@
static int32_t getNumberOfCameras();
static status_t getCameraInfo(int cameraId,
struct CameraInfo* cameraInfo);
- static sp<Camera> connect(int cameraId);
+ static sp<Camera> connect(int cameraId, bool force, bool keep);
virtual ~Camera();
void init();
diff --git a/include/camera/CameraParameters.h b/include/camera/CameraParameters.h
deleted file mode 100644
index 7edf6b4..0000000
--- a/include/camera/CameraParameters.h
+++ /dev/null
@@ -1,666 +0,0 @@
-/*
- * Copyright (C) 2008 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_HARDWARE_CAMERA_PARAMETERS_H
-#define ANDROID_HARDWARE_CAMERA_PARAMETERS_H
-
-#include <utils/KeyedVector.h>
-#include <utils/String8.h>
-
-namespace android {
-
-struct Size {
- int width;
- int height;
-
- Size() {
- width = 0;
- height = 0;
- }
-
- Size(int w, int h) {
- width = w;
- height = h;
- }
-};
-
-class CameraParameters
-{
-public:
- CameraParameters();
- CameraParameters(const String8 ¶ms) { unflatten(params); }
- ~CameraParameters();
-
- String8 flatten() const;
- void unflatten(const String8 ¶ms);
-
- void set(const char *key, const char *value);
- void set(const char *key, int value);
- void setFloat(const char *key, float value);
- const char *get(const char *key) const;
- int getInt(const char *key) const;
- float getFloat(const char *key) const;
-
- void remove(const char *key);
-
- void setPreviewSize(int width, int height);
- void getPreviewSize(int *width, int *height) const;
- void getSupportedPreviewSizes(Vector<Size> &sizes) const;
-
- // Set the dimensions in pixels to the given width and height
- // for video frames. The given width and height must be one
- // of the supported dimensions returned from
- // getSupportedVideoSizes(). Must not be called if
- // getSupportedVideoSizes() returns an empty Vector of Size.
- void setVideoSize(int width, int height);
- // Retrieve the current dimensions (width and height)
- // in pixels for video frames, which must be one of the
- // supported dimensions returned from getSupportedVideoSizes().
- // Must not be called if getSupportedVideoSizes() returns an
- // empty Vector of Size.
- void getVideoSize(int *width, int *height) const;
- // Retrieve a Vector of supported dimensions (width and height)
- // in pixels for video frames. If sizes returned from the method
- // is empty, the camera does not support calls to setVideoSize()
- // or getVideoSize(). In adddition, it also indicates that
- // the camera only has a single output, and does not have
- // separate output for video frames and preview frame.
- void getSupportedVideoSizes(Vector<Size> &sizes) const;
- // Retrieve the preferred preview size (width and height) in pixels
- // for video recording. The given width and height must be one of
- // supported preview sizes returned from getSupportedPreviewSizes().
- // Must not be called if getSupportedVideoSizes() returns an empty
- // Vector of Size. If getSupportedVideoSizes() returns an empty
- // Vector of Size, the width and height returned from this method
- // is invalid, and is "-1x-1".
- void getPreferredPreviewSizeForVideo(int *width, int *height) const;
-
- void setPreviewFrameRate(int fps);
- int getPreviewFrameRate() const;
- void getPreviewFpsRange(int *min_fps, int *max_fps) const;
- void setPreviewFormat(const char *format);
- const char *getPreviewFormat() const;
- void setPictureSize(int width, int height);
- void getPictureSize(int *width, int *height) const;
- void getSupportedPictureSizes(Vector<Size> &sizes) const;
- void setPictureFormat(const char *format);
- const char *getPictureFormat() const;
-
- void dump() const;
- status_t dump(int fd, const Vector<String16>& args) const;
-
- // Parameter keys to communicate between camera application and driver.
- // The access (read/write, read only, or write only) is viewed from the
- // perspective of applications, not driver.
-
- // Preview frame size in pixels (width x height).
- // Example value: "480x320". Read/Write.
- static const char KEY_PREVIEW_SIZE[];
- // Supported preview frame sizes in pixels.
- // Example value: "800x600,480x320". Read only.
- static const char KEY_SUPPORTED_PREVIEW_SIZES[];
- // The current minimum and maximum preview fps. This controls the rate of
- // preview frames received (CAMERA_MSG_PREVIEW_FRAME). The minimum and
- // maximum fps must be one of the elements from
- // KEY_SUPPORTED_PREVIEW_FPS_RANGE parameter.
- // Example value: "10500,26623"
- static const char KEY_PREVIEW_FPS_RANGE[];
- // The supported preview fps (frame-per-second) ranges. Each range contains
- // a minimum fps and maximum fps. If minimum fps equals to maximum fps, the
- // camera outputs frames in fixed frame rate. If not, the camera outputs
- // frames in auto frame rate. The actual frame rate fluctuates between the
- // minimum and the maximum. The list has at least one element. The list is
- // sorted from small to large (first by maximum fps and then minimum fps).
- // Example value: "(10500,26623),(15000,26623),(30000,30000)"
- static const char KEY_SUPPORTED_PREVIEW_FPS_RANGE[];
- // The image format for preview frames. See CAMERA_MSG_PREVIEW_FRAME in
- // frameworks/base/include/camera/Camera.h.
- // Example value: "yuv420sp" or PIXEL_FORMAT_XXX constants. Read/write.
- static const char KEY_PREVIEW_FORMAT[];
- // Supported image formats for preview frames.
- // Example value: "yuv420sp,yuv422i-yuyv". Read only.
- static const char KEY_SUPPORTED_PREVIEW_FORMATS[];
- // Number of preview frames per second. This is the target frame rate. The
- // actual frame rate depends on the driver.
- // Example value: "15". Read/write.
- static const char KEY_PREVIEW_FRAME_RATE[];
- // Supported number of preview frames per second.
- // Example value: "24,15,10". Read.
- static const char KEY_SUPPORTED_PREVIEW_FRAME_RATES[];
- // The dimensions for captured pictures in pixels (width x height).
- // Example value: "1024x768". Read/write.
- static const char KEY_PICTURE_SIZE[];
- // Supported dimensions for captured pictures in pixels.
- // Example value: "2048x1536,1024x768". Read only.
- static const char KEY_SUPPORTED_PICTURE_SIZES[];
- // The image format for captured pictures. See CAMERA_MSG_COMPRESSED_IMAGE
- // in frameworks/base/include/camera/Camera.h.
- // Example value: "jpeg" or PIXEL_FORMAT_XXX constants. Read/write.
- static const char KEY_PICTURE_FORMAT[];
- // Supported image formats for captured pictures.
- // Example value: "jpeg,rgb565". Read only.
- static const char KEY_SUPPORTED_PICTURE_FORMATS[];
- // The width (in pixels) of EXIF thumbnail in Jpeg picture.
- // Example value: "512". Read/write.
- static const char KEY_JPEG_THUMBNAIL_WIDTH[];
- // The height (in pixels) of EXIF thumbnail in Jpeg picture.
- // Example value: "384". Read/write.
- static const char KEY_JPEG_THUMBNAIL_HEIGHT[];
- // Supported EXIF thumbnail sizes (width x height). 0x0 means not thumbnail
- // in EXIF.
- // Example value: "512x384,320x240,0x0". Read only.
- static const char KEY_SUPPORTED_JPEG_THUMBNAIL_SIZES[];
- // The quality of the EXIF thumbnail in Jpeg picture. The range is 1 to 100,
- // with 100 being the best.
- // Example value: "90". Read/write.
- static const char KEY_JPEG_THUMBNAIL_QUALITY[];
- // Jpeg quality of captured picture. The range is 1 to 100, with 100 being
- // the best.
- // Example value: "90". Read/write.
- static const char KEY_JPEG_QUALITY[];
- // The rotation angle in degrees relative to the orientation of the camera.
- // This affects the pictures returned from CAMERA_MSG_COMPRESSED_IMAGE. The
- // camera driver may set orientation in the EXIF header without rotating the
- // picture. Or the driver may rotate the picture and the EXIF thumbnail. If
- // the Jpeg picture is rotated, the orientation in the EXIF header will be
- // missing or 1 (row #0 is top and column #0 is left side).
- //
- // Note that the JPEG pictures of front-facing cameras are not mirrored
- // as in preview display.
- //
- // For example, suppose the natural orientation of the device is portrait.
- // The device is rotated 270 degrees clockwise, so the device orientation is
- // 270. Suppose a back-facing camera sensor is mounted in landscape and the
- // top side of the camera sensor is aligned with the right edge of the
- // display in natural orientation. So the camera orientation is 90. The
- // rotation should be set to 0 (270 + 90).
- //
- // Example value: "0" or "90" or "180" or "270". Write only.
- static const char KEY_ROTATION[];
- // GPS latitude coordinate. GPSLatitude and GPSLatitudeRef will be stored in
- // JPEG EXIF header.
- // Example value: "25.032146" or "-33.462809". Write only.
- static const char KEY_GPS_LATITUDE[];
- // GPS longitude coordinate. GPSLongitude and GPSLongitudeRef will be stored
- // in JPEG EXIF header.
- // Example value: "121.564448" or "-70.660286". Write only.
- static const char KEY_GPS_LONGITUDE[];
- // GPS altitude. GPSAltitude and GPSAltitudeRef will be stored in JPEG EXIF
- // header.
- // Example value: "21.0" or "-5". Write only.
- static const char KEY_GPS_ALTITUDE[];
- // GPS timestamp (UTC in seconds since January 1, 1970). This should be
- // stored in JPEG EXIF header.
- // Example value: "1251192757". Write only.
- static const char KEY_GPS_TIMESTAMP[];
- // GPS Processing Method
- // Example value: "GPS" or "NETWORK". Write only.
- static const char KEY_GPS_PROCESSING_METHOD[];
- // Current white balance setting.
- // Example value: "auto" or WHITE_BALANCE_XXX constants. Read/write.
- static const char KEY_WHITE_BALANCE[];
- // Supported white balance settings.
- // Example value: "auto,incandescent,daylight". Read only.
- static const char KEY_SUPPORTED_WHITE_BALANCE[];
- // Current color effect setting.
- // Example value: "none" or EFFECT_XXX constants. Read/write.
- static const char KEY_EFFECT[];
- // Supported color effect settings.
- // Example value: "none,mono,sepia". Read only.
- static const char KEY_SUPPORTED_EFFECTS[];
- // Current antibanding setting.
- // Example value: "auto" or ANTIBANDING_XXX constants. Read/write.
- static const char KEY_ANTIBANDING[];
- // Supported antibanding settings.
- // Example value: "auto,50hz,60hz,off". Read only.
- static const char KEY_SUPPORTED_ANTIBANDING[];
- // Current scene mode.
- // Example value: "auto" or SCENE_MODE_XXX constants. Read/write.
- static const char KEY_SCENE_MODE[];
- // Supported scene mode settings.
- // Example value: "auto,night,fireworks". Read only.
- static const char KEY_SUPPORTED_SCENE_MODES[];
- // Current flash mode.
- // Example value: "auto" or FLASH_MODE_XXX constants. Read/write.
- static const char KEY_FLASH_MODE[];
- // Supported flash modes.
- // Example value: "auto,on,off". Read only.
- static const char KEY_SUPPORTED_FLASH_MODES[];
- // Current focus mode. This will not be empty. Applications should call
- // CameraHardwareInterface.autoFocus to start the focus if focus mode is
- // FOCUS_MODE_AUTO or FOCUS_MODE_MACRO.
- // Example value: "auto" or FOCUS_MODE_XXX constants. Read/write.
- static const char KEY_FOCUS_MODE[];
- // Supported focus modes.
- // Example value: "auto,macro,fixed". Read only.
- static const char KEY_SUPPORTED_FOCUS_MODES[];
- // The maximum number of focus areas supported. This is the maximum length
- // of KEY_FOCUS_AREAS.
- // Example value: "0" or "2". Read only.
- static const char KEY_MAX_NUM_FOCUS_AREAS[];
- // Current focus areas.
- //
- // Before accessing this parameter, apps should check
- // KEY_MAX_NUM_FOCUS_AREAS first to know the maximum number of focus areas
- // first. If the value is 0, focus area is not supported.
- //
- // Each focus area is a five-element int array. The first four elements are
- // the rectangle of the area (left, top, right, bottom). The direction is
- // relative to the sensor orientation, that is, what the sensor sees. The
- // direction is not affected by the rotation or mirroring of
- // CAMERA_CMD_SET_DISPLAY_ORIENTATION. Coordinates range from -1000 to 1000.
- // (-1000,-1000) is the upper left point. (1000, 1000) is the lower right
- // point. The width and height of focus areas cannot be 0 or negative.
- //
- // The fifth element is the weight. Values for weight must range from 1 to
- // 1000. The weight should be interpreted as a per-pixel weight - all
- // pixels in the area have the specified weight. This means a small area
- // with the same weight as a larger area will have less influence on the
- // focusing than the larger area. Focus areas can partially overlap and the
- // driver will add the weights in the overlap region.
- //
- // A special case of single focus area (0,0,0,0,0) means driver to decide
- // the focus area. For example, the driver may use more signals to decide
- // focus areas and change them dynamically. Apps can set (0,0,0,0,0) if they
- // want the driver to decide focus areas.
- //
- // Focus areas are relative to the current field of view (KEY_ZOOM). No
- // matter what the zoom level is, (-1000,-1000) represents the top of the
- // currently visible camera frame. The focus area cannot be set to be
- // outside the current field of view, even when using zoom.
- //
- // Focus area only has effect if the current focus mode is FOCUS_MODE_AUTO,
- // FOCUS_MODE_MACRO, FOCUS_MODE_CONTINUOUS_VIDEO, or
- // FOCUS_MODE_CONTINUOUS_PICTURE.
- // Example value: "(-10,-10,0,0,300),(0,0,10,10,700)". Read/write.
- static const char KEY_FOCUS_AREAS[];
- // Focal length in millimeter.
- // Example value: "4.31". Read only.
- static const char KEY_FOCAL_LENGTH[];
- // Horizontal angle of view in degrees.
- // Example value: "54.8". Read only.
- static const char KEY_HORIZONTAL_VIEW_ANGLE[];
- // Vertical angle of view in degrees.
- // Example value: "42.5". Read only.
- static const char KEY_VERTICAL_VIEW_ANGLE[];
- // Exposure compensation index. 0 means exposure is not adjusted.
- // Example value: "0" or "5". Read/write.
- static const char KEY_EXPOSURE_COMPENSATION[];
- // The maximum exposure compensation index (>=0).
- // Example value: "6". Read only.
- static const char KEY_MAX_EXPOSURE_COMPENSATION[];
- // The minimum exposure compensation index (<=0).
- // Example value: "-6". Read only.
- static const char KEY_MIN_EXPOSURE_COMPENSATION[];
- // The exposure compensation step. Exposure compensation index multiply by
- // step eqals to EV. Ex: if exposure compensation index is 6 and step is
- // 0.3333, EV is -2.
- // Example value: "0.333333333" or "0.5". Read only.
- static const char KEY_EXPOSURE_COMPENSATION_STEP[];
- // The state of the auto-exposure lock. "true" means that
- // auto-exposure is locked to its current value and will not
- // change. "false" means the auto-exposure routine is free to
- // change exposure values. If auto-exposure is already locked,
- // setting this to true again has no effect (the driver will not
- // recalculate exposure values). Changing exposure compensation
- // settings will still affect the exposure settings while
- // auto-exposure is locked. Stopping preview or taking a still
- // image will not change the lock. In conjunction with
- // exposure compensation, this allows for capturing multi-exposure
- // brackets with known relative exposure values. Locking
- // auto-exposure after open but before the first call to
- // startPreview may result in severely over- or under-exposed
- // images. The driver will not change the AE lock after
- // auto-focus completes.
- static const char KEY_AUTO_EXPOSURE_LOCK[];
- // Whether locking the auto-exposure is supported. "true" means it is, and
- // "false" or this key not existing means it is not supported.
- static const char KEY_AUTO_EXPOSURE_LOCK_SUPPORTED[];
- // The state of the auto-white balance lock. "true" means that
- // auto-white balance is locked to its current value and will not
- // change. "false" means the auto-white balance routine is free to
- // change white balance values. If auto-white balance is already
- // locked, setting this to true again has no effect (the driver
- // will not recalculate white balance values). Stopping preview or
- // taking a still image will not change the lock. In conjunction
- // with exposure compensation, this allows for capturing
- // multi-exposure brackets with fixed white balance. Locking
- // auto-white balance after open but before the first call to
- // startPreview may result in severely incorrect color. The
- // driver will not change the AWB lock after auto-focus
- // completes.
- static const char KEY_AUTO_WHITEBALANCE_LOCK[];
- // Whether locking the auto-white balance is supported. "true"
- // means it is, and "false" or this key not existing means it is
- // not supported.
- static const char KEY_AUTO_WHITEBALANCE_LOCK_SUPPORTED[];
-
- // The maximum number of metering areas supported. This is the maximum
- // length of KEY_METERING_AREAS.
- // Example value: "0" or "2". Read only.
- static const char KEY_MAX_NUM_METERING_AREAS[];
- // Current metering areas. Camera driver uses these areas to decide
- // exposure.
- //
- // Before accessing this parameter, apps should check
- // KEY_MAX_NUM_METERING_AREAS first to know the maximum number of metering
- // areas first. If the value is 0, metering area is not supported.
- //
- // Each metering area is a rectangle with specified weight. The direction is
- // relative to the sensor orientation, that is, what the sensor sees. The
- // direction is not affected by the rotation or mirroring of
- // CAMERA_CMD_SET_DISPLAY_ORIENTATION. Coordinates of the rectangle range
- // from -1000 to 1000. (-1000, -1000) is the upper left point. (1000, 1000)
- // is the lower right point. The width and height of metering areas cannot
- // be 0 or negative.
- //
- // The fifth element is the weight. Values for weight must range from 1 to
- // 1000. The weight should be interpreted as a per-pixel weight - all
- // pixels in the area have the specified weight. This means a small area
- // with the same weight as a larger area will have less influence on the
- // metering than the larger area. Metering areas can partially overlap and
- // the driver will add the weights in the overlap region.
- //
- // A special case of all-zero single metering area means driver to decide
- // the metering area. For example, the driver may use more signals to decide
- // metering areas and change them dynamically. Apps can set all-zero if they
- // want the driver to decide metering areas.
- //
- // Metering areas are relative to the current field of view (KEY_ZOOM).
- // No matter what the zoom level is, (-1000,-1000) represents the top of the
- // currently visible camera frame. The metering area cannot be set to be
- // outside the current field of view, even when using zoom.
- //
- // No matter what metering areas are, the final exposure are compensated
- // by KEY_EXPOSURE_COMPENSATION.
- // Example value: "(-10,-10,0,0,300),(0,0,10,10,700)". Read/write.
- static const char KEY_METERING_AREAS[];
- // Current zoom value.
- // Example value: "0" or "6". Read/write.
- static const char KEY_ZOOM[];
- // Maximum zoom value.
- // Example value: "6". Read only.
- static const char KEY_MAX_ZOOM[];
- // The zoom ratios of all zoom values. The zoom ratio is in 1/100
- // increments. Ex: a zoom of 3.2x is returned as 320. The number of list
- // elements is KEY_MAX_ZOOM + 1. The first element is always 100. The last
- // element is the zoom ratio of zoom value KEY_MAX_ZOOM.
- // Example value: "100,150,200,250,300,350,400". Read only.
- static const char KEY_ZOOM_RATIOS[];
- // Whether zoom is supported. Zoom is supported if the value is "true". Zoom
- // is not supported if the value is not "true" or the key does not exist.
- // Example value: "true". Read only.
- static const char KEY_ZOOM_SUPPORTED[];
- // Whether if smooth zoom is supported. Smooth zoom is supported if the
- // value is "true". It is not supported if the value is not "true" or the
- // key does not exist.
- // See CAMERA_CMD_START_SMOOTH_ZOOM, CAMERA_CMD_STOP_SMOOTH_ZOOM, and
- // CAMERA_MSG_ZOOM in frameworks/base/include/camera/Camera.h.
- // Example value: "true". Read only.
- static const char KEY_SMOOTH_ZOOM_SUPPORTED[];
-
- // The distances (in meters) from the camera to where an object appears to
- // be in focus. The object is sharpest at the optimal focus distance. The
- // depth of field is the far focus distance minus near focus distance.
- //
- // Focus distances may change after starting auto focus, canceling auto
- // focus, or starting the preview. Applications can read this anytime to get
- // the latest focus distances. If the focus mode is FOCUS_MODE_CONTINUOUS,
- // focus distances may change from time to time.
- //
- // This is intended to estimate the distance between the camera and the
- // subject. After autofocus, the subject distance may be within near and far
- // focus distance. However, the precision depends on the camera hardware,
- // autofocus algorithm, the focus area, and the scene. The error can be
- // large and it should be only used as a reference.
- //
- // Far focus distance > optimal focus distance > near focus distance. If
- // the far focus distance is infinity, the value should be "Infinity" (case
- // sensitive). The format is three float values separated by commas. The
- // first is near focus distance. The second is optimal focus distance. The
- // third is far focus distance.
- // Example value: "0.95,1.9,Infinity" or "0.049,0.05,0.051". Read only.
- static const char KEY_FOCUS_DISTANCES[];
-
- // The current dimensions in pixels (width x height) for video frames.
- // The width and height must be one of the supported sizes retrieved
- // via KEY_SUPPORTED_VIDEO_SIZES.
- // Example value: "1280x720". Read/write.
- static const char KEY_VIDEO_SIZE[];
- // A list of the supported dimensions in pixels (width x height)
- // for video frames. See CAMERA_MSG_VIDEO_FRAME for details in
- // frameworks/base/include/camera/Camera.h.
- // Example: "176x144,1280x720". Read only.
- static const char KEY_SUPPORTED_VIDEO_SIZES[];
-
- // The maximum number of detected faces supported by hardware face
- // detection. If the value is 0, hardware face detection is not supported.
- // Example: "5". Read only
- static const char KEY_MAX_NUM_DETECTED_FACES_HW[];
-
- // The maximum number of detected faces supported by software face
- // detection. If the value is 0, software face detection is not supported.
- // Example: "5". Read only
- static const char KEY_MAX_NUM_DETECTED_FACES_SW[];
-
- // Preferred preview frame size in pixels for video recording.
- // The width and height must be one of the supported sizes retrieved
- // via KEY_SUPPORTED_PREVIEW_SIZES. This key can be used only when
- // getSupportedVideoSizes() does not return an empty Vector of Size.
- // Camcorder applications are recommended to set the preview size
- // to a value that is not larger than the preferred preview size.
- // In other words, the product of the width and height of the
- // preview size should not be larger than that of the preferred
- // preview size. In addition, we recommend to choos a preview size
- // that has the same aspect ratio as the resolution of video to be
- // recorded.
- // Example value: "800x600". Read only.
- static const char KEY_PREFERRED_PREVIEW_SIZE_FOR_VIDEO[];
-
- // The image format for video frames. See CAMERA_MSG_VIDEO_FRAME in
- // frameworks/base/include/camera/Camera.h.
- // Example value: "yuv420sp" or PIXEL_FORMAT_XXX constants. Read only.
- static const char KEY_VIDEO_FRAME_FORMAT[];
-
- // Sets the hint of the recording mode. If this is true, MediaRecorder.start
- // may be faster or has less glitches. This should be called before starting
- // the preview for the best result. But it is allowed to change the hint
- // while the preview is active. The default value is false.
- //
- // The apps can still call Camera.takePicture when the hint is true. The
- // apps can call MediaRecorder.start when the hint is false. But the
- // performance may be worse.
- // Example value: "true" or "false". Read/write.
- static const char KEY_RECORDING_HINT[];
-
- // Returns true if video snapshot is supported. That is, applications
- // can call Camera.takePicture during recording. Applications do not need to
- // call Camera.startPreview after taking a picture. The preview will be
- // still active. Other than that, taking a picture during recording is
- // identical to taking a picture normally. All settings and methods related
- // to takePicture work identically. Ex: KEY_PICTURE_SIZE,
- // KEY_SUPPORTED_PICTURE_SIZES, KEY_JPEG_QUALITY, KEY_ROTATION, and etc.
- // The picture will have an EXIF header. FLASH_MODE_AUTO and FLASH_MODE_ON
- // also still work, but the video will record the flash.
- //
- // Applications can set shutter callback as null to avoid the shutter
- // sound. It is also recommended to set raw picture and post view callbacks
- // to null to avoid the interrupt of preview display.
- //
- // Field-of-view of the recorded video may be different from that of the
- // captured pictures.
- // Example value: "true" or "false". Read only.
- static const char KEY_VIDEO_SNAPSHOT_SUPPORTED[];
-
- // The state of the video stabilization. If set to true, both the
- // preview stream and the recorded video stream are stabilized by
- // the camera. Only valid to set if KEY_VIDEO_STABILIZATION_SUPPORTED is
- // set to true.
- //
- // The value of this key can be changed any time the camera is
- // open. If preview or recording is active, it is acceptable for
- // there to be a slight video glitch when video stabilization is
- // toggled on and off.
- //
- // This only stabilizes video streams (between-frames stabilization), and
- // has no effect on still image capture.
- static const char KEY_VIDEO_STABILIZATION[];
-
- // Returns true if video stabilization is supported. That is, applications
- // can set KEY_VIDEO_STABILIZATION to true and have a stabilized preview
- // stream and record stabilized videos.
- static const char KEY_VIDEO_STABILIZATION_SUPPORTED[];
-
- // Value for KEY_ZOOM_SUPPORTED or KEY_SMOOTH_ZOOM_SUPPORTED.
- static const char TRUE[];
- static const char FALSE[];
-
- // Value for KEY_FOCUS_DISTANCES.
- static const char FOCUS_DISTANCE_INFINITY[];
-
- // Values for white balance settings.
- static const char WHITE_BALANCE_AUTO[];
- static const char WHITE_BALANCE_INCANDESCENT[];
- static const char WHITE_BALANCE_FLUORESCENT[];
- static const char WHITE_BALANCE_WARM_FLUORESCENT[];
- static const char WHITE_BALANCE_DAYLIGHT[];
- static const char WHITE_BALANCE_CLOUDY_DAYLIGHT[];
- static const char WHITE_BALANCE_TWILIGHT[];
- static const char WHITE_BALANCE_SHADE[];
-
- // Values for effect settings.
- static const char EFFECT_NONE[];
- static const char EFFECT_MONO[];
- static const char EFFECT_NEGATIVE[];
- static const char EFFECT_SOLARIZE[];
- static const char EFFECT_SEPIA[];
- static const char EFFECT_POSTERIZE[];
- static const char EFFECT_WHITEBOARD[];
- static const char EFFECT_BLACKBOARD[];
- static const char EFFECT_AQUA[];
-
- // Values for antibanding settings.
- static const char ANTIBANDING_AUTO[];
- static const char ANTIBANDING_50HZ[];
- static const char ANTIBANDING_60HZ[];
- static const char ANTIBANDING_OFF[];
-
- // Values for flash mode settings.
- // Flash will not be fired.
- static const char FLASH_MODE_OFF[];
- // Flash will be fired automatically when required. The flash may be fired
- // during preview, auto-focus, or snapshot depending on the driver.
- static const char FLASH_MODE_AUTO[];
- // Flash will always be fired during snapshot. The flash may also be
- // fired during preview or auto-focus depending on the driver.
- static const char FLASH_MODE_ON[];
- // Flash will be fired in red-eye reduction mode.
- static const char FLASH_MODE_RED_EYE[];
- // Constant emission of light during preview, auto-focus and snapshot.
- // This can also be used for video recording.
- static const char FLASH_MODE_TORCH[];
-
- // Values for scene mode settings.
- static const char SCENE_MODE_AUTO[];
- static const char SCENE_MODE_ACTION[];
- static const char SCENE_MODE_PORTRAIT[];
- static const char SCENE_MODE_LANDSCAPE[];
- static const char SCENE_MODE_NIGHT[];
- static const char SCENE_MODE_NIGHT_PORTRAIT[];
- static const char SCENE_MODE_THEATRE[];
- static const char SCENE_MODE_BEACH[];
- static const char SCENE_MODE_SNOW[];
- static const char SCENE_MODE_SUNSET[];
- static const char SCENE_MODE_STEADYPHOTO[];
- static const char SCENE_MODE_FIREWORKS[];
- static const char SCENE_MODE_SPORTS[];
- static const char SCENE_MODE_PARTY[];
- static const char SCENE_MODE_CANDLELIGHT[];
- // Applications are looking for a barcode. Camera driver will be optimized
- // for barcode reading.
- static const char SCENE_MODE_BARCODE[];
-
- // Pixel color formats for KEY_PREVIEW_FORMAT, KEY_PICTURE_FORMAT,
- // and KEY_VIDEO_FRAME_FORMAT
- static const char PIXEL_FORMAT_YUV422SP[];
- static const char PIXEL_FORMAT_YUV420SP[]; // NV21
- static const char PIXEL_FORMAT_YUV422I[]; // YUY2
- static const char PIXEL_FORMAT_YUV420P[]; // YV12
- static const char PIXEL_FORMAT_RGB565[];
- static const char PIXEL_FORMAT_RGBA8888[];
- static const char PIXEL_FORMAT_JPEG[];
- // Raw bayer format used for images, which is 10 bit precision samples
- // stored in 16 bit words. The filter pattern is RGGB.
- static const char PIXEL_FORMAT_BAYER_RGGB[];
-
- // Values for focus mode settings.
- // Auto-focus mode. Applications should call
- // CameraHardwareInterface.autoFocus to start the focus in this mode.
- static const char FOCUS_MODE_AUTO[];
- // Focus is set at infinity. Applications should not call
- // CameraHardwareInterface.autoFocus in this mode.
- static const char FOCUS_MODE_INFINITY[];
- // Macro (close-up) focus mode. Applications should call
- // CameraHardwareInterface.autoFocus to start the focus in this mode.
- static const char FOCUS_MODE_MACRO[];
- // Focus is fixed. The camera is always in this mode if the focus is not
- // adjustable. If the camera has auto-focus, this mode can fix the
- // focus, which is usually at hyperfocal distance. Applications should
- // not call CameraHardwareInterface.autoFocus in this mode.
- static const char FOCUS_MODE_FIXED[];
- // Extended depth of field (EDOF). Focusing is done digitally and
- // continuously. Applications should not call
- // CameraHardwareInterface.autoFocus in this mode.
- static const char FOCUS_MODE_EDOF[];
- // Continuous auto focus mode intended for video recording. The camera
- // continuously tries to focus. This is the best choice for video
- // recording because the focus changes smoothly . Applications still can
- // call CameraHardwareInterface.takePicture in this mode but the subject may
- // not be in focus. Auto focus starts when the parameter is set.
- //
- // Applications can call CameraHardwareInterface.autoFocus in this mode. The
- // focus callback will immediately return with a boolean that indicates
- // whether the focus is sharp or not. The focus position is locked after
- // autoFocus call. If applications want to resume the continuous focus,
- // cancelAutoFocus must be called. Restarting the preview will not resume
- // the continuous autofocus. To stop continuous focus, applications should
- // change the focus mode to other modes.
- static const char FOCUS_MODE_CONTINUOUS_VIDEO[];
- // Continuous auto focus mode intended for taking pictures. The camera
- // continuously tries to focus. The speed of focus change is more aggressive
- // than FOCUS_MODE_CONTINUOUS_VIDEO. Auto focus starts when the parameter is
- // set.
- //
- // Applications can call CameraHardwareInterface.autoFocus in this mode. If
- // the autofocus is in the middle of scanning, the focus callback will
- // return when it completes. If the autofocus is not scanning, focus
- // callback will immediately return with a boolean that indicates whether
- // the focus is sharp or not. The apps can then decide if they want to take
- // a picture immediately or to change the focus mode to auto, and run a full
- // autofocus cycle. The focus position is locked after autoFocus call. If
- // applications want to resume the continuous focus, cancelAutoFocus must be
- // called. Restarting the preview will not resume the continuous autofocus.
- // To stop continuous focus, applications should change the focus mode to
- // other modes.
- static const char FOCUS_MODE_CONTINUOUS_PICTURE[];
-
-private:
- DefaultKeyedVector<String8,String8> mMap;
-};
-
-}; // namespace android
-
-#endif
diff --git a/include/camera/ICamera.h b/include/camera/ICamera.h
index 400d7f4..3d18837 100644
--- a/include/camera/ICamera.h
+++ b/include/camera/ICamera.h
@@ -20,15 +20,15 @@
#include <utils/RefBase.h>
#include <binder/IInterface.h>
#include <binder/Parcel.h>
-#include <surfaceflinger/Surface.h>
#include <binder/IMemory.h>
#include <utils/String8.h>
#include <camera/Camera.h>
-#include <gui/ISurfaceTexture.h>
namespace android {
class ICameraClient;
+class ISurfaceTexture;
+class Surface;
class ICamera: public IInterface
{
diff --git a/include/camera/ICameraService.h b/include/camera/ICameraService.h
index 7d70c1e..97e3169 100644
--- a/include/camera/ICameraService.h
+++ b/include/camera/ICameraService.h
@@ -42,7 +42,7 @@
virtual status_t getCameraInfo(int cameraId,
struct CameraInfo* cameraInfo) = 0;
virtual sp<ICamera> connect(const sp<ICameraClient>& cameraClient,
- int cameraId) = 0;
+ int cameraId, bool force, bool keep) = 0;
};
// ----------------------------------------------------------------------------
diff --git a/include/common_time/ICommonClock.h b/include/common_time/ICommonClock.h
new file mode 100644
index 0000000..d7073f1
--- /dev/null
+++ b/include/common_time/ICommonClock.h
@@ -0,0 +1,108 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_ICOMMONCLOCK_H
+#define ANDROID_ICOMMONCLOCK_H
+
+#include <stdint.h>
+#include <linux/socket.h>
+
+#include <binder/IInterface.h>
+#include <binder/IServiceManager.h>
+
+namespace android {
+
+class ICommonClockListener : public IInterface {
+ public:
+ DECLARE_META_INTERFACE(CommonClockListener);
+
+ virtual void onTimelineChanged(uint64_t timelineID) = 0;
+};
+
+class BnCommonClockListener : public BnInterface<ICommonClockListener> {
+ public:
+ virtual status_t onTransact(uint32_t code, const Parcel& data,
+ Parcel* reply, uint32_t flags = 0);
+};
+
+class ICommonClock : public IInterface {
+ public:
+ DECLARE_META_INTERFACE(CommonClock);
+
+ // Name of the ICommonClock service registered with the service manager.
+ static const String16 kServiceName;
+
+ // a reserved invalid timeline ID
+ static const uint64_t kInvalidTimelineID;
+
+ // a reserved invalid error estimate
+ static const int32_t kErrorEstimateUnknown;
+
+ enum State {
+ // the device just came up and is trying to discover the master
+ STATE_INITIAL,
+
+ // the device is a client of a master
+ STATE_CLIENT,
+
+ // the device is acting as master
+ STATE_MASTER,
+
+ // the device has lost contact with its master and needs to participate
+ // in the election of a new master
+ STATE_RONIN,
+
+ // the device is waiting for announcement of the newly elected master
+ STATE_WAIT_FOR_ELECTION,
+ };
+
+ virtual status_t isCommonTimeValid(bool* valid, uint32_t* timelineID) = 0;
+ virtual status_t commonTimeToLocalTime(int64_t commonTime,
+ int64_t* localTime) = 0;
+ virtual status_t localTimeToCommonTime(int64_t localTime,
+ int64_t* commonTime) = 0;
+ virtual status_t getCommonTime(int64_t* commonTime) = 0;
+ virtual status_t getCommonFreq(uint64_t* freq) = 0;
+ virtual status_t getLocalTime(int64_t* localTime) = 0;
+ virtual status_t getLocalFreq(uint64_t* freq) = 0;
+ virtual status_t getEstimatedError(int32_t* estimate) = 0;
+ virtual status_t getTimelineID(uint64_t* id) = 0;
+ virtual status_t getState(State* state) = 0;
+ virtual status_t getMasterAddr(struct sockaddr_storage* addr) = 0;
+
+ virtual status_t registerListener(
+ const sp<ICommonClockListener>& listener) = 0;
+ virtual status_t unregisterListener(
+ const sp<ICommonClockListener>& listener) = 0;
+
+ // Simple helper to make it easier to connect to the CommonClock service.
+ static inline sp<ICommonClock> getInstance() {
+ sp<IBinder> binder = defaultServiceManager()->checkService(
+ ICommonClock::kServiceName);
+ sp<ICommonClock> clk = interface_cast<ICommonClock>(binder);
+ return clk;
+ }
+};
+
+class BnCommonClock : public BnInterface<ICommonClock> {
+ public:
+ virtual status_t onTransact(uint32_t code, const Parcel& data,
+ Parcel* reply, uint32_t flags = 0);
+};
+
+}; // namespace android
+
+#endif // ANDROID_ICOMMONCLOCK_H
diff --git a/include/common_time/ICommonTimeConfig.h b/include/common_time/ICommonTimeConfig.h
new file mode 100644
index 0000000..497b666
--- /dev/null
+++ b/include/common_time/ICommonTimeConfig.h
@@ -0,0 +1,73 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_ICOMMONTIMECONFIG_H
+#define ANDROID_ICOMMONTIMECONFIG_H
+
+#include <stdint.h>
+#include <linux/socket.h>
+
+#include <binder/IInterface.h>
+#include <binder/IServiceManager.h>
+
+namespace android {
+
+class String16;
+
+class ICommonTimeConfig : public IInterface {
+ public:
+ DECLARE_META_INTERFACE(CommonTimeConfig);
+
+ // Name of the ICommonTimeConfig service registered with the service
+ // manager.
+ static const String16 kServiceName;
+
+ virtual status_t getMasterElectionPriority(uint8_t *priority) = 0;
+ virtual status_t setMasterElectionPriority(uint8_t priority) = 0;
+ virtual status_t getMasterElectionEndpoint(struct sockaddr_storage *addr) = 0;
+ virtual status_t setMasterElectionEndpoint(const struct sockaddr_storage *addr) = 0;
+ virtual status_t getMasterElectionGroupId(uint64_t *id) = 0;
+ virtual status_t setMasterElectionGroupId(uint64_t id) = 0;
+ virtual status_t getInterfaceBinding(String16& ifaceName) = 0;
+ virtual status_t setInterfaceBinding(const String16& ifaceName) = 0;
+ virtual status_t getMasterAnnounceInterval(int *interval) = 0;
+ virtual status_t setMasterAnnounceInterval(int interval) = 0;
+ virtual status_t getClientSyncInterval(int *interval) = 0;
+ virtual status_t setClientSyncInterval(int interval) = 0;
+ virtual status_t getPanicThreshold(int *threshold) = 0;
+ virtual status_t setPanicThreshold(int threshold) = 0;
+ virtual status_t getAutoDisable(bool *autoDisable) = 0;
+ virtual status_t setAutoDisable(bool autoDisable) = 0;
+ virtual status_t forceNetworklessMasterMode() = 0;
+
+ // Simple helper to make it easier to connect to the CommonTimeConfig service.
+ static inline sp<ICommonTimeConfig> getInstance() {
+ sp<IBinder> binder = defaultServiceManager()->checkService(
+ ICommonTimeConfig::kServiceName);
+ sp<ICommonTimeConfig> clk = interface_cast<ICommonTimeConfig>(binder);
+ return clk;
+ }
+};
+
+class BnCommonTimeConfig : public BnInterface<ICommonTimeConfig> {
+ public:
+ virtual status_t onTransact(uint32_t code, const Parcel& data,
+ Parcel* reply, uint32_t flags = 0);
+};
+
+}; // namespace android
+
+#endif // ANDROID_ICOMMONTIMECONFIG_H
diff --git a/include/common_time/cc_helper.h b/include/common_time/cc_helper.h
new file mode 100644
index 0000000..8c4d5c0
--- /dev/null
+++ b/include/common_time/cc_helper.h
@@ -0,0 +1,72 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __CC_HELPER_H__
+#define __CC_HELPER_H__
+
+#include <stdint.h>
+#include <common_time/ICommonClock.h>
+#include <utils/threads.h>
+
+namespace android {
+
+// CCHelper is a simple wrapper class to help with centralizing access to the
+// Common Clock service and implementing lifetime managment, as well as to
+// implement a simple policy of making a basic attempt to reconnect to the
+// common clock service when things go wrong.
+//
+// On platforms which run the native common_time service in auto-disable mode,
+// the service will go into networkless mode whenever it has no active clients.
+// It tracks active clients using registered CommonClockListeners (the callback
+// interface for onTimelineChanged) since this provides a convienent death
+// handler notification for when the service's clients die unexpectedly. This
+// means that users of the common time service should really always have a
+// CommonClockListener, unless they know that the time service is not running in
+// auto disabled mode, or that there is at least one other registered listener
+// active in the system. The CCHelper makes this a little easier by sharing a
+// ref counted ICommonClock interface across all clients and automatically
+// registering and unregistering a listener whenever there are CCHelper
+// instances active in the process.
+class CCHelper {
+ public:
+ CCHelper();
+ ~CCHelper();
+
+ status_t isCommonTimeValid(bool* valid, uint32_t* timelineID);
+ status_t commonTimeToLocalTime(int64_t commonTime, int64_t* localTime);
+ status_t localTimeToCommonTime(int64_t localTime, int64_t* commonTime);
+ status_t getCommonTime(int64_t* commonTime);
+ status_t getCommonFreq(uint64_t* freq);
+ status_t getLocalTime(int64_t* localTime);
+ status_t getLocalFreq(uint64_t* freq);
+
+ private:
+ class CommonClockListener : public BnCommonClockListener {
+ public:
+ void onTimelineChanged(uint64_t timelineID);
+ };
+
+ static bool verifyClock_l();
+
+ static Mutex lock_;
+ static sp<ICommonClock> common_clock_;
+ static sp<ICommonClockListener> common_clock_listener_;
+ static uint32_t ref_count_;
+};
+
+
+} // namespace android
+#endif // __CC_HELPER_H__
diff --git a/include/common_time/local_clock.h b/include/common_time/local_clock.h
new file mode 100644
index 0000000..845d1c2
--- /dev/null
+++ b/include/common_time/local_clock.h
@@ -0,0 +1,47 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+
+#ifndef __LOCAL_CLOCK_H__
+#define __LOCAL_CLOCK_H__
+
+#include <stdint.h>
+
+#include <hardware/local_time_hal.h>
+#include <utils/Errors.h>
+#include <utils/threads.h>
+
+namespace android {
+
+class LocalClock {
+ public:
+ LocalClock();
+
+ bool initCheck();
+
+ int64_t getLocalTime();
+ uint64_t getLocalFreq();
+ status_t setLocalSlew(int16_t rate);
+ int32_t getDebugLog(struct local_time_debug_event* records,
+ int max_records);
+
+ private:
+ static Mutex dev_lock_;
+ static local_time_hw_device_t* dev_;
+};
+
+} // namespace android
+#endif // __LOCAL_CLOCK_H__
diff --git a/include/drm/DrmManagerClient.h b/include/drm/DrmManagerClient.h
index b8fe46d..c47bbfb 100644
--- a/include/drm/DrmManagerClient.h
+++ b/include/drm/DrmManagerClient.h
@@ -66,19 +66,21 @@
* @param[in] fd File descriptor of the protected content to be decrypted
* @param[in] offset Start position of the content
* @param[in] length The length of the protected content
+ * @param[in] mime Mime type of the protected content if it is not NULL or empty
* @return
* Handle for the decryption session
*/
- sp<DecryptHandle> openDecryptSession(int fd, off64_t offset, off64_t length);
+ sp<DecryptHandle> openDecryptSession(int fd, off64_t offset, off64_t length, const char* mime);
/**
* Open the decrypt session to decrypt the given protected content
*
* @param[in] uri Path of the protected content to be decrypted
+ * @param[in] mime Mime type of the protected content if it is not NULL or empty
* @return
* Handle for the decryption session
*/
- sp<DecryptHandle> openDecryptSession(const char* uri);
+ sp<DecryptHandle> openDecryptSession(const char* uri, const char* mime);
/**
* Close the decrypt session for the given handle
diff --git a/include/drm/drm_framework_common.h b/include/drm/drm_framework_common.h
index 2632cbd..637409c 100644
--- a/include/drm/drm_framework_common.h
+++ b/include/drm/drm_framework_common.h
@@ -43,6 +43,7 @@
DRM_ERROR_DECRYPT = ERROR_BASE - 5,
DRM_ERROR_CANNOT_HANDLE = ERROR_BASE - 6,
DRM_ERROR_TAMPER_DETECTED = ERROR_BASE - 7,
+ DRM_ERROR_NO_PERMISSION = ERROR_BASE - 8,
DRM_NO_ERROR = NO_ERROR
};
diff --git a/include/media/AudioEffect.h b/include/media/AudioEffect.h
index 1417416..02dfc1b 100644
--- a/include/media/AudioEffect.h
+++ b/include/media/AudioEffect.h
@@ -108,7 +108,8 @@
* Returned value
* *descriptor updated with effect descriptor
*/
- static status_t getEffectDescriptor(effect_uuid_t *uuid, effect_descriptor_t *descriptor);
+ static status_t getEffectDescriptor(const effect_uuid_t *uuid,
+ effect_descriptor_t *descriptor) /*const*/;
/*
@@ -226,8 +227,8 @@
AudioEffect(const effect_uuid_t *type,
const effect_uuid_t *uuid = NULL,
int32_t priority = 0,
- effect_callback_t cbf = 0,
- void* user = 0,
+ effect_callback_t cbf = NULL,
+ void* user = NULL,
int sessionId = 0,
audio_io_handle_t io = 0
);
@@ -238,8 +239,8 @@
AudioEffect(const char *typeStr,
const char *uuidStr = NULL,
int32_t priority = 0,
- effect_callback_t cbf = 0,
- void* user = 0,
+ effect_callback_t cbf = NULL,
+ void* user = NULL,
int sessionId = 0,
audio_io_handle_t io = 0
);
@@ -260,8 +261,8 @@
status_t set(const effect_uuid_t *type,
const effect_uuid_t *uuid = NULL,
int32_t priority = 0,
- effect_callback_t cbf = 0,
- void* user = 0,
+ effect_callback_t cbf = NULL,
+ void* user = NULL,
int sessionId = 0,
audio_io_handle_t io = 0
);
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index 605680a..7df6668 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -22,7 +22,6 @@
#include <media/IAudioFlinger.h>
#include <media/IAudioRecord.h>
-#include <media/AudioTrack.h>
#include <utils/RefBase.h>
#include <utils/Errors.h>
@@ -34,6 +33,8 @@
namespace android {
+class audio_track_cblk_t;
+
// ----------------------------------------------------------------------------
class AudioRecord
@@ -67,7 +68,7 @@
};
uint32_t flags;
int channelCount;
- int format;
+ audio_format_t format;
size_t frameCount;
size_t size;
union {
@@ -111,7 +112,7 @@
static status_t getMinFrameCount(int* frameCount,
uint32_t sampleRate,
- int format,
+ audio_format_t format,
int channelCount);
/* Constructs an uninitialized AudioRecord. No connection with
@@ -142,20 +143,22 @@
* user Context for use by the callback receiver.
*/
+ // FIXME consider removing this alias and replacing it by audio_in_acoustics_t
+ // or removing the parameter entirely if it is unused
enum record_flags {
RECORD_AGC_ENABLE = AUDIO_IN_ACOUSTICS_AGC_ENABLE,
RECORD_NS_ENABLE = AUDIO_IN_ACOUSTICS_NS_ENABLE,
RECORD_IIR_ENABLE = AUDIO_IN_ACOUSTICS_TX_IIR_ENABLE,
};
- AudioRecord(int inputSource,
+ AudioRecord(audio_source_t inputSource,
uint32_t sampleRate = 0,
- int format = 0,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
uint32_t channelMask = AUDIO_CHANNEL_IN_MONO,
int frameCount = 0,
- uint32_t flags = 0,
- callback_t cbf = 0,
- void* user = 0,
+ record_flags flags = (record_flags) 0,
+ callback_t cbf = NULL,
+ void* user = NULL,
int notificationFrames = 0,
int sessionId = 0);
@@ -174,14 +177,14 @@
* - NO_INIT: audio server or audio hardware not initialized
* - PERMISSION_DENIED: recording is not allowed for the requesting process
* */
- status_t set(int inputSource = 0,
+ status_t set(audio_source_t inputSource = AUDIO_SOURCE_DEFAULT,
uint32_t sampleRate = 0,
- int format = 0,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
uint32_t channelMask = AUDIO_CHANNEL_IN_MONO,
int frameCount = 0,
- uint32_t flags = 0,
- callback_t cbf = 0,
- void* user = 0,
+ record_flags flags = (record_flags) 0,
+ callback_t cbf = NULL,
+ void* user = NULL,
int notificationFrames = 0,
bool threadCanCallJava = false,
int sessionId = 0);
@@ -202,12 +205,12 @@
/* getters, see constructor */
- int format() const;
+ audio_format_t format() const;
int channelCount() const;
int channels() const;
uint32_t frameCount() const;
- int frameSize() const;
- int inputSource() const;
+ size_t frameSize() const;
+ audio_source_t inputSource() const;
/* After it's created the track is not active. Call start() to
@@ -224,7 +227,7 @@
/* get sample rate for this record track
*/
- uint32_t getSampleRate();
+ uint32_t getSampleRate() const;
/* Sets marker position. When record reaches the number of frames specified,
* a callback with event type EVENT_MARKER is called. Calling setMarkerPosition
@@ -241,7 +244,7 @@
* - INVALID_OPERATION: the AudioRecord has no callback installed.
*/
status_t setMarkerPosition(uint32_t marker);
- status_t getMarkerPosition(uint32_t *marker);
+ status_t getMarkerPosition(uint32_t *marker) const;
/* Sets position update period. Every time the number of frames specified has been recorded,
@@ -260,7 +263,7 @@
* - INVALID_OPERATION: the AudioRecord has no callback installed.
*/
status_t setPositionUpdatePeriod(uint32_t updatePeriod);
- status_t getPositionUpdatePeriod(uint32_t *updatePeriod);
+ status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const;
/* Gets record head position. The position is the total number of frames
@@ -274,7 +277,7 @@
* - NO_ERROR: successful operation
* - BAD_VALUE: position is NULL
*/
- status_t getPosition(uint32_t *position);
+ status_t getPosition(uint32_t *position) const;
/* returns a handle on the audio input used by this AudioRecord.
*
@@ -284,7 +287,7 @@
* Returned value:
* handle on audio hardware input
*/
- audio_io_handle_t getInput();
+ audio_io_handle_t getInput() const;
/* returns the audio session ID associated to this AudioRecord.
*
@@ -294,11 +297,11 @@
* Returned value:
* AudioRecord session ID.
*/
- int getSessionId();
+ int getSessionId() const;
/* obtains a buffer of "frameCount" frames. The buffer must be
* filled entirely. If the track is stopped, obtainBuffer() returns
- * STOPPED instead of NO_ERROR as long as there are buffers availlable,
+ * STOPPED instead of NO_ERROR as long as there are buffers available,
* at which point NO_MORE_BUFFERS is returned.
* Buffers will be returned until the pool (buffercount())
* is exhausted, at which point obtainBuffer() will either block
@@ -316,16 +319,17 @@
/* As a convenience we provide a read() interface to the audio buffer.
- * This is implemented on top of lockBuffer/unlockBuffer.
+ * This is implemented on top of obtainBuffer/releaseBuffer.
*/
ssize_t read(void* buffer, size_t size);
- /* Return the amount of input frames lost in the audio driver since the last call of this function.
- * Audio driver is expected to reset the value to 0 and restart counting upon returning the current value by this function call.
- * Such loss typically occurs when the user space process is blocked longer than the capacity of audio driver buffers.
+ /* Return the amount of input frames lost in the audio driver since the last call of this
+ * function. Audio driver is expected to reset the value to 0 and restart counting upon
+ * returning the current value by this function call. Such loss typically occurs when the
+ * user space process is blocked longer than the capacity of audio driver buffers.
* Unit: the number of input audio frames
*/
- unsigned int getInputFramesLost();
+ unsigned int getInputFramesLost() const;
private:
/* copying audio tracks is not allowed */
@@ -340,15 +344,14 @@
private:
friend class AudioRecord;
virtual bool threadLoop();
- virtual status_t readyToRun() { return NO_ERROR; }
+ virtual status_t readyToRun();
virtual void onFirstRef() {}
AudioRecord& mReceiver;
- Mutex mLock;
};
bool processAudioBuffer(const sp<ClientRecordThread>& thread);
status_t openRecord_l(uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
@@ -359,15 +362,16 @@
sp<IAudioRecord> mAudioRecord;
sp<IMemory> mCblkMemory;
sp<ClientRecordThread> mClientRecordThread;
- Mutex mLock;
+ status_t mReadyToRun;
+ mutable Mutex mLock;
+ Condition mCondition;
uint32_t mFrameCount;
audio_track_cblk_t* mCblk;
- uint32_t mFormat;
+ audio_format_t mFormat;
uint8_t mChannelCount;
- uint8_t mInputSource;
- uint8_t mReserved[2];
+ audio_source_t mInputSource;
status_t mStatus;
uint32_t mLatency;
@@ -381,10 +385,12 @@
bool mMarkerReached;
uint32_t mNewPosition;
uint32_t mUpdatePeriod;
- uint32_t mFlags;
+ record_flags mFlags;
uint32_t mChannelMask;
audio_io_handle_t mInput;
int mSessionId;
+ int mPreviousPriority; // before start()
+ int mPreviousSchedulingGroup;
};
}; // namespace android
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index 6a15f6e..cc0a594 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -55,19 +55,21 @@
static status_t getMasterMute(bool* mute);
// set/get stream volume on specified output
- static status_t setStreamVolume(int stream, float value, int output);
- static status_t getStreamVolume(int stream, float* volume, int output);
+ static status_t setStreamVolume(audio_stream_type_t stream, float value,
+ audio_io_handle_t output);
+ static status_t getStreamVolume(audio_stream_type_t stream, float* volume,
+ audio_io_handle_t output);
// mute/unmute stream
- static status_t setStreamMute(int stream, bool mute);
- static status_t getStreamMute(int stream, bool* mute);
+ static status_t setStreamMute(audio_stream_type_t stream, bool mute);
+ static status_t getStreamMute(audio_stream_type_t stream, bool* mute);
- // set audio mode in audio hardware (see audio_mode_t)
- static status_t setMode(int mode);
+ // set audio mode in audio hardware
+ static status_t setMode(audio_mode_t mode);
// returns true in *state if tracks are active on the specified stream or has been active
// in the past inPastMs milliseconds
- static status_t isStreamActive(int stream, bool *state, uint32_t inPastMs = 0);
+ static status_t isStreamActive(audio_stream_type_t stream, bool *state, uint32_t inPastMs = 0);
// set/get audio hardware parameters. The function accepts a list of parameters
// key value pairs in the form: key1=value1;key2=value2;...
@@ -83,13 +85,19 @@
static float linearToLog(int volume);
static int logToLinear(float volume);
+ static status_t getOutputSamplingRate(int* samplingRate, audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
+ static status_t getOutputFrameCount(int* frameCount, audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
+ static status_t getOutputLatency(uint32_t* latency, audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
+
+ // DEPRECATED
static status_t getOutputSamplingRate(int* samplingRate, int stream = AUDIO_STREAM_DEFAULT);
+
+ // DEPRECATED
static status_t getOutputFrameCount(int* frameCount, int stream = AUDIO_STREAM_DEFAULT);
- static status_t getOutputLatency(uint32_t* latency, int stream = AUDIO_STREAM_DEFAULT);
- static bool routedToA2dpOutput(int streamType);
+ static bool routedToA2dpOutput(audio_stream_type_t streamType);
- static status_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount,
+ static status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount,
size_t* buffSize);
static status_t setVoiceVolume(float volume);
@@ -103,7 +111,7 @@
// - BAD_VALUE: invalid parameter
// NOTE: this feature is not supported on all hardware platforms and it is
// necessary to check returned status before using the returned values.
- static status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int stream = AUDIO_STREAM_DEFAULT);
+ static status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
static unsigned int getInputFramesLost(audio_io_handle_t ioHandle);
@@ -123,12 +131,12 @@
NUM_CONFIG_EVENTS
};
- // audio output descritor used to cache output configurations in client process to avoid frequent calls
+ // audio output descriptor used to cache output configurations in client process to avoid frequent calls
// through IAudioFlinger
class OutputDescriptor {
public:
OutputDescriptor()
- : samplingRate(0), format(0), channels(0), frameCount(0), latency(0) {}
+ : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channels(0), frameCount(0), latency(0) {}
uint32_t samplingRate;
int32_t format;
@@ -142,15 +150,14 @@
//
static status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, const char *device_address);
static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device, const char *device_address);
- static status_t setPhoneState(int state);
- static status_t setRingerMode(uint32_t mode, uint32_t mask);
+ static status_t setPhoneState(audio_mode_t state);
static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config);
static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
static audio_io_handle_t getOutput(audio_stream_type_t stream,
uint32_t samplingRate = 0,
- uint32_t format = AUDIO_FORMAT_DEFAULT,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
uint32_t channels = AUDIO_CHANNEL_OUT_STEREO,
- audio_policy_output_flags_t flags = AUDIO_POLICY_OUTPUT_FLAG_INDIRECT);
+ audio_policy_output_flags_t flags = AUDIO_POLICY_OUTPUT_FLAG_NONE);
static status_t startOutput(audio_io_handle_t output,
audio_stream_type_t stream,
int session = 0);
@@ -158,9 +165,9 @@
audio_stream_type_t stream,
int session = 0);
static void releaseOutput(audio_io_handle_t output);
- static audio_io_handle_t getInput(int inputSource,
+ static audio_io_handle_t getInput(audio_source_t inputSource,
uint32_t samplingRate = 0,
- uint32_t format = AUDIO_FORMAT_DEFAULT,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
uint32_t channels = AUDIO_CHANNEL_IN_MONO,
audio_in_acoustics_t acoustics = (audio_in_acoustics_t)0,
int sessionId = 0);
@@ -170,11 +177,15 @@
static status_t initStreamVolume(audio_stream_type_t stream,
int indexMin,
int indexMax);
- static status_t setStreamVolumeIndex(audio_stream_type_t stream, int index);
- static status_t getStreamVolumeIndex(audio_stream_type_t stream, int *index);
+ static status_t setStreamVolumeIndex(audio_stream_type_t stream,
+ int index,
+ audio_devices_t device);
+ static status_t getStreamVolumeIndex(audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device);
static uint32_t getStrategyForStream(audio_stream_type_t stream);
- static uint32_t getDevicesForStream(audio_stream_type_t stream);
+ static audio_devices_t getDevicesForStream(audio_stream_type_t stream);
static audio_io_handle_t getOutputForEffect(effect_descriptor_t *desc);
static status_t registerEffect(effect_descriptor_t *desc,
@@ -207,8 +218,8 @@
// IAudioFlingerClient
// indicate a change in the configuration of an output or input: keeps the cached
- // values for output/input parameters upto date in client process
- virtual void ioConfigChanged(int event, int ioHandle, void *param2);
+ // values for output/input parameters up-to-date in client process
+ virtual void ioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2);
};
class AudioPolicyServiceClient: public IBinder::DeathRecipient
@@ -233,13 +244,13 @@
static size_t gInBuffSize;
// previous parameters for recording buffer size queries
static uint32_t gPrevInSamplingRate;
- static int gPrevInFormat;
+ static audio_format_t gPrevInFormat;
static int gPrevInChannelCount;
static sp<IAudioPolicyService> gAudioPolicyService;
// mapping between stream types and outputs
- static DefaultKeyedVector<int, audio_io_handle_t> gStreamOutputMap;
+ static DefaultKeyedVector<audio_stream_type_t, audio_io_handle_t> gStreamOutputMap;
// list of output descriptors containing cached parameters
// (sampling rate, framecount, channel count...)
static DefaultKeyedVector<audio_io_handle_t, OutputDescriptor *> gOutputs;
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 1c401e2..552e829 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -38,7 +38,7 @@
// ----------------------------------------------------------------------------
-class AudioTrack
+class AudioTrack : virtual public RefBase
{
public:
enum channel_index {
@@ -58,8 +58,8 @@
EVENT_BUFFER_END = 5 // Playback head is at the end of the buffer.
};
- /* Create Buffer on the stack and pass it to obtainBuffer()
- * and releaseBuffer().
+ /* Client should declare Buffer on the stack and pass address to obtainBuffer()
+ * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA.
*/
class Buffer
@@ -68,34 +68,39 @@
enum {
MUTE = 0x00000001
};
- uint32_t flags;
- int format;
+ uint32_t flags; // 0 or MUTE
+ audio_format_t format; // but AUDIO_FORMAT_PCM_8_BIT -> AUDIO_FORMAT_PCM_16_BIT
+ // accessed directly by WebKit ANP callback
int channelCount; // will be removed in the future, do not use
- size_t frameCount;
- size_t size;
+
+ size_t frameCount; // number of sample frames corresponding to size;
+ // on input it is the number of frames desired,
+ // on output is the number of frames actually filled
+
+ size_t size; // input/output in byte units
union {
void* raw;
- short* i16;
- int8_t* i8;
+ short* i16; // signed 16-bit
+ int8_t* i8; // unsigned 8-bit, offset by 0x80
};
};
/* As a convenience, if a callback is supplied, a handler thread
* is automatically created with the appropriate priority. This thread
- * invokes the callback when a new buffer becomes availlable or an underrun condition occurs.
+ * invokes the callback when a new buffer becomes available or various conditions occur.
* Parameters:
*
* event: type of event notified (see enum AudioTrack::event_type).
* user: Pointer to context for use by the callback receiver.
* info: Pointer to optional parameter according to event type:
* - EVENT_MORE_DATA: pointer to AudioTrack::Buffer struct. The callback must not write
- * more bytes than indicated by 'size' field and update 'size' if less bytes are
- * written.
+ * more bytes than indicated by 'size' field and update 'size' if fewer bytes are
+ * written.
* - EVENT_UNDERRUN: unused.
* - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining.
- * - EVENT_MARKER: pointer to an uin32_t containing the marker position in frames.
- * - EVENT_NEW_POS: pointer to an uin32_t containing the new position in frames.
+ * - EVENT_MARKER: pointer to an uint32_t containing the marker position in frames.
+ * - EVENT_NEW_POS: pointer to an uint32_t containing the new position in frames.
* - EVENT_BUFFER_END: unused.
*/
@@ -109,7 +114,7 @@
*/
static status_t getMinFrameCount(int* frameCount,
- int streamType =-1,
+ audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT,
uint32_t sampleRate = 0);
/* Constructs an uninitialized AudioTrack. No connection with
@@ -137,63 +142,76 @@
* flags: Reserved for future use.
* cbf: Callback function. If not null, this function is called periodically
* to request new PCM data.
+ * user: Context for use by the callback receiver.
* notificationFrames: The callback function is called each time notificationFrames PCM
- * frames have been comsumed from track input buffer.
- * user Context for use by the callback receiver.
+ * frames have been consumed from track input buffer.
+ * sessionId: Specific session ID, or zero to use default.
*/
- AudioTrack( int streamType,
+ AudioTrack( audio_stream_type_t streamType,
uint32_t sampleRate = 0,
- int format = 0,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
int channelMask = 0,
int frameCount = 0,
- uint32_t flags = 0,
+ audio_policy_output_flags_t flags = AUDIO_POLICY_OUTPUT_FLAG_NONE,
+ callback_t cbf = NULL,
+ void* user = NULL,
+ int notificationFrames = 0,
+ int sessionId = 0);
+
+ // DEPRECATED
+ explicit AudioTrack( int streamType,
+ uint32_t sampleRate = 0,
+ int format = AUDIO_FORMAT_DEFAULT,
+ int channelMask = 0,
+ int frameCount = 0,
+ uint32_t flags = (uint32_t) AUDIO_POLICY_OUTPUT_FLAG_NONE,
callback_t cbf = 0,
void* user = 0,
int notificationFrames = 0,
int sessionId = 0);
/* Creates an audio track and registers it with AudioFlinger. With this constructor,
- * The PCM data to be rendered by AudioTrack is passed in a shared memory buffer
+ * the PCM data to be rendered by AudioTrack is passed in a shared memory buffer
* identified by the argument sharedBuffer. This prototype is for static buffer playback.
- * PCM data must be present into memory before the AudioTrack is started.
- * The Write() and Flush() methods are not supported in this case.
- * It is recommented to pass a callback function to be notified of playback end by an
+ * PCM data must be present in memory before the AudioTrack is started.
+ * The write() and flush() methods are not supported in this case.
+ * It is recommended to pass a callback function to be notified of playback end by an
* EVENT_UNDERRUN event.
*/
- AudioTrack( int streamType,
+ AudioTrack( audio_stream_type_t streamType,
uint32_t sampleRate = 0,
- int format = 0,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
int channelMask = 0,
const sp<IMemory>& sharedBuffer = 0,
- uint32_t flags = 0,
- callback_t cbf = 0,
- void* user = 0,
+ audio_policy_output_flags_t flags = AUDIO_POLICY_OUTPUT_FLAG_NONE,
+ callback_t cbf = NULL,
+ void* user = NULL,
int notificationFrames = 0,
int sessionId = 0);
/* Terminates the AudioTrack and unregisters it from AudioFlinger.
- * Also destroys all resources assotiated with the AudioTrack.
+ * Also destroys all resources associated with the AudioTrack.
*/
~AudioTrack();
/* Initialize an uninitialized AudioTrack.
* Returned status (from utils/Errors.h) can be:
- * - NO_ERROR: successful intialization
- * - INVALID_OPERATION: AudioTrack is already intitialized
+ * - NO_ERROR: successful initialization
+ * - INVALID_OPERATION: AudioTrack is already initialized
* - BAD_VALUE: invalid parameter (channels, format, sampleRate...)
* - NO_INIT: audio server or audio hardware not initialized
* */
- status_t set(int streamType =-1,
+ status_t set(audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT,
uint32_t sampleRate = 0,
- int format = 0,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
int channelMask = 0,
int frameCount = 0,
- uint32_t flags = 0,
- callback_t cbf = 0,
- void* user = 0,
+ audio_policy_output_flags_t flags = AUDIO_POLICY_OUTPUT_FLAG_NONE,
+ callback_t cbf = NULL,
+ void* user = NULL,
int notificationFrames = 0,
const sp<IMemory>& sharedBuffer = 0,
bool threadCanCallJava = false,
@@ -201,25 +219,30 @@
/* Result of constructing the AudioTrack. This must be checked
- * before using any AudioTrack API (except for set()), using
+ * before using any AudioTrack API (except for set()), because using
* an uninitialized AudioTrack produces undefined results.
* See set() method above for possible return codes.
*/
status_t initCheck() const;
- /* Returns this track's latency in milliseconds.
+ /* Returns this track's estimated latency in milliseconds.
* This includes the latency due to AudioTrack buffer size, AudioMixer (if any)
* and audio hardware driver.
*/
uint32_t latency() const;
- /* getters, see constructor */
+ /* getters, see constructors and set() */
- int streamType() const;
- int format() const;
+ audio_stream_type_t streamType() const;
+ audio_format_t format() const;
int channelCount() const;
uint32_t frameCount() const;
- int frameSize() const;
+
+ /* Return channelCount * (bit depth per channel / 8).
+ * channelCount is determined from channelMask, and bit depth comes from format.
+ */
+ size_t frameSize() const;
+
sp<IMemory>& sharedBuffer();
@@ -235,8 +258,8 @@
void stop();
bool stopped() const;
- /* flush a stopped track. All pending buffers are discarded.
- * This function has no effect if the track is not stoped.
+ /* Flush a stopped track. All pending buffers are discarded.
+ * This function has no effect if the track is not stopped.
*/
void flush();
@@ -246,29 +269,28 @@
*/
void pause();
- /* mute or unmutes this track.
- * While mutted, the callback, if set, is still called.
+ /* Mute or unmute this track.
+ * While muted, the callback, if set, is still called.
*/
void mute(bool);
bool muted() const;
-
- /* set volume for this track, mostly used for games' sound effects
- * left and right volumes. Levels must be <= 1.0.
+ /* Set volume for this track, mostly used for games' sound effects
+ * left and right volumes. Levels must be >= 0.0 and <= 1.0.
*/
status_t setVolume(float left, float right);
- void getVolume(float* left, float* right);
+ void getVolume(float* left, float* right) const;
- /* set the send level for this track. An auxiliary effect should be attached
- * to the track with attachEffect(). Level must be <= 1.0.
+ /* Set the send level for this track. An auxiliary effect should be attached
+ * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0.
*/
status_t setAuxEffectSendLevel(float level);
- void getAuxEffectSendLevel(float* level);
+ void getAuxEffectSendLevel(float* level) const;
- /* set sample rate for this track, mostly used for games' sound effects
+ /* Set sample rate for this track, mostly used for games' sound effects
*/
status_t setSampleRate(int sampleRate);
- uint32_t getSampleRate();
+ uint32_t getSampleRate() const;
/* Enables looping and sets the start and end points of looping.
*
@@ -276,19 +298,17 @@
*
* loopStart: loop start expressed as the number of PCM frames played since AudioTrack start.
* loopEnd: loop end expressed as the number of PCM frames played since AudioTrack start.
- * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any pending or
- * active loop. loopCount = -1 means infinite looping.
+ * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any
+ * pending or active loop. loopCount = -1 means infinite looping.
*
* For proper operation the following condition must be respected:
* (loopEnd-loopStart) <= framecount()
*/
status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount);
- status_t getLoop(uint32_t *loopStart, uint32_t *loopEnd, int *loopCount);
-
- /* Sets marker position. When playback reaches the number of frames specified, a callback with event
- * type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker notification
- * callback.
+ /* Sets marker position. When playback reaches the number of frames specified, a callback with
+ * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker
+ * notification callback.
* If the AudioTrack has been opened with no callback function associated, the operation will fail.
*
* Parameters:
@@ -300,13 +320,13 @@
* - INVALID_OPERATION: the AudioTrack has no callback installed.
*/
status_t setMarkerPosition(uint32_t marker);
- status_t getMarkerPosition(uint32_t *marker);
+ status_t getMarkerPosition(uint32_t *marker) const;
- /* Sets position update period. Every time the number of frames specified has been played,
- * a callback with event type EVENT_NEW_POS is called.
- * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification
- * callback.
+ /* Sets position update period. Every time the number of frames specified has been played,
+ * a callback with event type EVENT_NEW_POS is called.
+ * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification
+ * callback.
* If the AudioTrack has been opened with no callback function associated, the operation will fail.
*
* Parameters:
@@ -318,14 +338,13 @@
* - INVALID_OPERATION: the AudioTrack has no callback installed.
*/
status_t setPositionUpdatePeriod(uint32_t updatePeriod);
- status_t getPositionUpdatePeriod(uint32_t *updatePeriod);
-
+ status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const;
/* Sets playback head position within AudioTrack buffer. The new position is specified
- * in number of frames.
+ * in number of frames.
* This method must be called with the AudioTrack in paused or stopped state.
- * Note that the actual position set is <position> modulo the AudioTrack buffer size in frames.
- * Therefore using this method makes sense only when playing a "static" audio buffer
+ * Note that the actual position set is <position> modulo the AudioTrack buffer size in frames.
+ * Therefore using this method makes sense only when playing a "static" audio buffer
* as opposed to streaming.
* The getPosition() method on the other hand returns the total number of frames played since
* playback start.
@@ -337,12 +356,12 @@
* Returned status (from utils/Errors.h) can be:
* - NO_ERROR: successful operation
* - INVALID_OPERATION: the AudioTrack is not stopped.
- * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack buffer
+ * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack buffer
*/
status_t setPosition(uint32_t position);
status_t getPosition(uint32_t *position);
- /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids
+ /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids
* rewriting the buffer before restarting playback after a stop.
* This method must be called with the AudioTrack in paused or stopped state.
*
@@ -352,7 +371,7 @@
*/
status_t reload();
- /* returns a handle on the audio output used by this AudioTrack.
+ /* Returns a handle on the audio output used by this AudioTrack.
*
* Parameters:
* none.
@@ -362,18 +381,17 @@
*/
audio_io_handle_t getOutput();
- /* returns the unique ID associated to this track.
+ /* Returns the unique session ID associated with this track.
*
* Parameters:
* none.
*
* Returned value:
- * AudioTrack ID.
+ * AudioTrack session ID.
*/
- int getSessionId();
+ int getSessionId() const;
-
- /* Attach track auxiliary output to specified effect. Used effectId = 0
+ /* Attach track auxiliary output to specified effect. Use effectId = 0
* to detach track from effect.
*
* Parameters:
@@ -387,29 +405,41 @@
*/
status_t attachAuxEffect(int effectId);
- /* obtains a buffer of "frameCount" frames. The buffer must be
- * filled entirely. If the track is stopped, obtainBuffer() returns
- * STOPPED instead of NO_ERROR as long as there are buffers availlable,
+ /* Obtains a buffer of "frameCount" frames. The buffer must be
+ * filled entirely, and then released with releaseBuffer().
+ * If the track is stopped, obtainBuffer() returns
+ * STOPPED instead of NO_ERROR as long as there are buffers available,
* at which point NO_MORE_BUFFERS is returned.
* Buffers will be returned until the pool (buffercount())
* is exhausted, at which point obtainBuffer() will either block
* or return WOULD_BLOCK depending on the value of the "blocking"
* parameter.
+ *
+ * Interpretation of waitCount:
+ * +n limits wait time to n * WAIT_PERIOD_MS,
+ * -1 causes an (almost) infinite wait time,
+ * 0 non-blocking.
*/
enum {
- NO_MORE_BUFFERS = 0x80000001,
+ NO_MORE_BUFFERS = 0x80000001, // same name in AudioFlinger.h, ok to be different value
STOPPED = 1
};
status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount);
+
+ /* Release a filled buffer of "frameCount" frames for AudioFlinger to process. */
void releaseBuffer(Buffer* audioBuffer);
-
/* As a convenience we provide a write() interface to the audio buffer.
- * This is implemented on top of lockBuffer/unlockBuffer. For best
- * performance
- *
+ * This is implemented on top of obtainBuffer/releaseBuffer. For best
+ * performance use callbacks. Returns actual number of bytes written >= 0,
+ * or one of the following negative status codes:
+ * INVALID_OPERATION AudioTrack is configured for shared buffer mode
+ * BAD_VALUE size is invalid
+ * STOPPED AudioTrack was stopped during the write
+ * NO_MORE_BUFFERS when obtainBuffer() returns same
+ * or any other error code returned by IAudioTrack::start() or restoreTrack_l().
*/
ssize_t write(const void* buffer, size_t size);
@@ -418,7 +448,7 @@
*/
status_t dump(int fd, const Vector<String16>& args) const;
-private:
+protected:
/* copying audio tracks is not allowed */
AudioTrack(const AudioTrack& other);
AudioTrack& operator = (const AudioTrack& other);
@@ -434,16 +464,16 @@
virtual status_t readyToRun();
virtual void onFirstRef();
AudioTrack& mReceiver;
- Mutex mLock;
};
+ // body of AudioTrackThread::threadLoop()
bool processAudioBuffer(const sp<AudioTrackThread>& thread);
- status_t createTrack_l(int streamType,
+ status_t createTrack_l(audio_stream_type_t streamType,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
- uint32_t flags,
+ audio_policy_output_flags_t flags,
const sp<IMemory>& sharedBuffer,
audio_io_handle_t output,
bool enforceFrameCount);
@@ -451,6 +481,7 @@
status_t setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount);
audio_io_handle_t getOutput_l();
status_t restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart);
+ bool stopped_l() const { return !mActive; }
sp<IAudioTrack> mAudioTrack;
sp<IMemory> mCblkMemory;
@@ -461,8 +492,8 @@
uint32_t mFrameCount;
audio_track_cblk_t* mCblk;
- uint32_t mFormat;
- uint8_t mStreamType;
+ audio_format_t mFormat;
+ audio_stream_type_t mStreamType;
uint8_t mChannelCount;
uint8_t mMuted;
uint8_t mReserved;
@@ -470,9 +501,9 @@
status_t mStatus;
uint32_t mLatency;
- volatile int32_t mActive;
+ bool mActive; // protected by mLock
- callback_t mCbf;
+ callback_t mCbf; // callback handler for events, or NULL
void* mUserData;
uint32_t mNotificationFramesReq; // requested number of frames between each notification callback
uint32_t mNotificationFramesAct; // actual number of frames between each notification callback
@@ -484,13 +515,38 @@
uint32_t mNewPosition;
uint32_t mUpdatePeriod;
bool mFlushed; // FIXME will be made obsolete by making flush() synchronous
- uint32_t mFlags;
+ audio_policy_output_flags_t mFlags;
int mSessionId;
int mAuxEffectId;
- Mutex mLock;
+ mutable Mutex mLock;
status_t mRestoreStatus;
+ bool mIsTimed;
+ int mPreviousPriority; // before start()
+ int mPreviousSchedulingGroup;
};
+class TimedAudioTrack : public AudioTrack
+{
+public:
+ TimedAudioTrack();
+
+ /* allocate a shared memory buffer that can be passed to queueTimedBuffer */
+ status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer);
+
+ /* queue a buffer obtained via allocateTimedBuffer for playback at the
+ given timestamp. PTS units a microseconds on the media time timeline.
+ The media time transform (set with setMediaTimeTransform) set by the
+ audio producer will handle converting from media time to local time
+ (perhaps going through the common time timeline in the case of
+ synchronized multiroom audio case) */
+ status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts);
+
+ /* define a transform between media time and either common time or
+ local time */
+ enum TargetTimeline {LOCAL_TIME, COMMON_TIME};
+ status_t setMediaTimeTransform(const LinearTransform& xform,
+ TargetTimeline target);
+};
}; // namespace android
diff --git a/include/media/EffectsFactoryApi.h b/include/media/EffectsFactoryApi.h
index 8ae13cc..65c26f4 100644
--- a/include/media/EffectsFactoryApi.h
+++ b/include/media/EffectsFactoryApi.h
@@ -87,7 +87,7 @@
// Description: Creates an effect engine of the specified type and returns an
// effect control interface on this engine. The function will allocate the
// resources for an instance of the requested effect engine and return
-// a handler on the effect control interface.
+// a handle on the effect control interface.
//
// Input:
// pEffectUuid: pointer to the effect uuid.
@@ -109,23 +109,23 @@
// *pHandle: updated with the effect handle.
//
////////////////////////////////////////////////////////////////////////////////
-int EffectCreate(effect_uuid_t *pEffectUuid, int32_t sessionId, int32_t ioId, effect_handle_t *pHandle);
+int EffectCreate(const effect_uuid_t *pEffectUuid, int32_t sessionId, int32_t ioId, effect_handle_t *pHandle);
////////////////////////////////////////////////////////////////////////////////
//
// Function: EffectRelease
//
-// Description: Releases the effect engine whose handler is given as argument.
+// Description: Releases the effect engine whose handle is given as argument.
// All resources allocated to this particular instance of the effect are
// released.
//
// Input:
-// handle: handler on the effect interface to be released.
+// handle: handle on the effect interface to be released.
//
// Output:
// returned value: 0 successful operation.
// -ENODEV factory failed to initialize
-// -EINVAL invalid interface handler
+// -EINVAL invalid interface handle
//
////////////////////////////////////////////////////////////////////////////////
int EffectRelease(effect_handle_t handle);
@@ -151,7 +151,7 @@
// *pDescriptor: updated with the effect descriptor.
//
////////////////////////////////////////////////////////////////////////////////
-int EffectGetDescriptor(effect_uuid_t *pEffectUuid, effect_descriptor_t *pDescriptor);
+int EffectGetDescriptor(const effect_uuid_t *pEffectUuid, effect_descriptor_t *pDescriptor);
////////////////////////////////////////////////////////////////////////////////
//
@@ -167,7 +167,7 @@
// 1 if uuid is equal to EFFECT_UUID_NULL.
//
////////////////////////////////////////////////////////////////////////////////
-int EffectIsNullUuid(effect_uuid_t *pEffectUuid);
+int EffectIsNullUuid(const effect_uuid_t *pEffectUuid);
#if __cplusplus
} // extern "C"
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index 9e3cb7f..0f39cf3 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -27,6 +27,8 @@
#include <media/IAudioTrack.h>
#include <media/IAudioRecord.h>
#include <media/IAudioFlingerClient.h>
+#include <system/audio.h>
+#include <system/audio_policy.h>
#include <hardware/audio_effect.h>
#include <media/IEffect.h>
#include <media/IEffectClient.h>
@@ -46,22 +48,23 @@
*/
virtual sp<IAudioTrack> createTrack(
pid_t pid,
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
- int output,
+ audio_io_handle_t output,
+ bool isTimed,
int *sessionId,
status_t *status) = 0;
virtual sp<IAudioRecord> openRecord(
pid_t pid,
- int input,
+ audio_io_handle_t input,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
@@ -71,11 +74,13 @@
/* query the audio hardware state. This state never changes,
* and therefore can be cached.
*/
- virtual uint32_t sampleRate(int output) const = 0;
- virtual int channelCount(int output) const = 0;
- virtual uint32_t format(int output) const = 0;
- virtual size_t frameCount(int output) const = 0;
- virtual uint32_t latency(int output) const = 0;
+ virtual uint32_t sampleRate(audio_io_handle_t output) const = 0;
+ virtual int channelCount(audio_io_handle_t output) const = 0;
+ virtual audio_format_t format(audio_io_handle_t output) const = 0;
+ virtual size_t frameCount(audio_io_handle_t output) const = 0;
+
+ // return estimated latency in milliseconds
+ virtual uint32_t latency(audio_io_handle_t output) const = 0;
/* set/get the audio hardware state. This will probably be used by
* the preference panel, mostly.
@@ -89,76 +94,83 @@
/* set/get stream type state. This will probably be used by
* the preference panel, mostly.
*/
- virtual status_t setStreamVolume(int stream, float value, int output) = 0;
- virtual status_t setStreamMute(int stream, bool muted) = 0;
+ virtual status_t setStreamVolume(audio_stream_type_t stream, float value,
+ audio_io_handle_t output) = 0;
+ virtual status_t setStreamMute(audio_stream_type_t stream, bool muted) = 0;
- virtual float streamVolume(int stream, int output) const = 0;
- virtual bool streamMute(int stream) const = 0;
+ virtual float streamVolume(audio_stream_type_t stream,
+ audio_io_handle_t output) const = 0;
+ virtual bool streamMute(audio_stream_type_t stream) const = 0;
// set audio mode
- virtual status_t setMode(int mode) = 0;
+ virtual status_t setMode(audio_mode_t mode) = 0;
// mic mute/state
virtual status_t setMicMute(bool state) = 0;
virtual bool getMicMute() const = 0;
- virtual status_t setParameters(int ioHandle, const String8& keyValuePairs) = 0;
- virtual String8 getParameters(int ioHandle, const String8& keys) = 0;
+ virtual status_t setParameters(audio_io_handle_t ioHandle,
+ const String8& keyValuePairs) = 0;
+ virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const = 0;
// register a current process for audio output change notifications
virtual void registerClient(const sp<IAudioFlingerClient>& client) = 0;
// retrieve the audio recording buffer size
- virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount) = 0;
+ virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const = 0;
- virtual int openOutput(uint32_t *pDevices,
+ virtual audio_io_handle_t openOutput(uint32_t *pDevices,
uint32_t *pSamplingRate,
- uint32_t *pFormat,
+ audio_format_t *pFormat,
uint32_t *pChannels,
uint32_t *pLatencyMs,
- uint32_t flags) = 0;
- virtual int openDuplicateOutput(int output1, int output2) = 0;
- virtual status_t closeOutput(int output) = 0;
- virtual status_t suspendOutput(int output) = 0;
- virtual status_t restoreOutput(int output) = 0;
+ audio_policy_output_flags_t flags) = 0;
+ virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
+ audio_io_handle_t output2) = 0;
+ virtual status_t closeOutput(audio_io_handle_t output) = 0;
+ virtual status_t suspendOutput(audio_io_handle_t output) = 0;
+ virtual status_t restoreOutput(audio_io_handle_t output) = 0;
- virtual int openInput(uint32_t *pDevices,
+ virtual audio_io_handle_t openInput(uint32_t *pDevices,
uint32_t *pSamplingRate,
- uint32_t *pFormat,
+ audio_format_t *pFormat,
uint32_t *pChannels,
- uint32_t acoustics) = 0;
- virtual status_t closeInput(int input) = 0;
+ audio_in_acoustics_t acoustics) = 0;
+ virtual status_t closeInput(audio_io_handle_t input) = 0;
- virtual status_t setStreamOutput(uint32_t stream, int output) = 0;
+ virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) = 0;
virtual status_t setVoiceVolume(float volume) = 0;
- virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) = 0;
+ virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
+ audio_io_handle_t output) const = 0;
- virtual unsigned int getInputFramesLost(int ioHandle) = 0;
+ virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const = 0;
virtual int newAudioSessionId() = 0;
virtual void acquireAudioSessionId(int audioSession) = 0;
virtual void releaseAudioSessionId(int audioSession) = 0;
- virtual status_t queryNumberEffects(uint32_t *numEffects) = 0;
+ virtual status_t queryNumberEffects(uint32_t *numEffects) const = 0;
- virtual status_t queryEffect(uint32_t index, effect_descriptor_t *pDescriptor) = 0;
+ virtual status_t queryEffect(uint32_t index, effect_descriptor_t *pDescriptor) const = 0;
- virtual status_t getEffectDescriptor(effect_uuid_t *pEffectUUID, effect_descriptor_t *pDescriptor) = 0;
+ virtual status_t getEffectDescriptor(const effect_uuid_t *pEffectUUID,
+ effect_descriptor_t *pDescriptor) const = 0;
virtual sp<IEffect> createEffect(pid_t pid,
effect_descriptor_t *pDesc,
const sp<IEffectClient>& client,
int32_t priority,
- int output,
+ audio_io_handle_t output,
int sessionId,
status_t *status,
int *id,
int *enabled) = 0;
- virtual status_t moveEffects(int session, int srcOutput, int dstOutput) = 0;
+ virtual status_t moveEffects(int session, audio_io_handle_t srcOutput,
+ audio_io_handle_t dstOutput) = 0;
};
diff --git a/include/media/IAudioFlingerClient.h b/include/media/IAudioFlingerClient.h
index aa0cdcf..75a9971 100644
--- a/include/media/IAudioFlingerClient.h
+++ b/include/media/IAudioFlingerClient.h
@@ -21,6 +21,7 @@
#include <utils/RefBase.h>
#include <binder/IInterface.h>
#include <utils/KeyedVector.h>
+#include <system/audio.h>
namespace android {
@@ -32,7 +33,7 @@
DECLARE_META_INTERFACE(AudioFlingerClient);
// Notifies a change of audio input/output configuration.
- virtual void ioConfigChanged(int event, int ioHandle, void *param2) = 0;
+ virtual void ioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2) = 0;
};
diff --git a/include/media/IAudioPolicyService.h b/include/media/IAudioPolicyService.h
index 9807cbe..04c927a 100644
--- a/include/media/IAudioPolicyService.h
+++ b/include/media/IAudioPolicyService.h
@@ -45,15 +45,14 @@
const char *device_address) = 0;
virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
const char *device_address) = 0;
- virtual status_t setPhoneState(int state) = 0;
- virtual status_t setRingerMode(uint32_t mode, uint32_t mask) = 0;
+ virtual status_t setPhoneState(audio_mode_t state) = 0;
virtual status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config) = 0;
virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage) = 0;
virtual audio_io_handle_t getOutput(audio_stream_type_t stream,
uint32_t samplingRate = 0,
- uint32_t format = AUDIO_FORMAT_DEFAULT,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
uint32_t channels = 0,
- audio_policy_output_flags_t flags = AUDIO_POLICY_OUTPUT_FLAG_INDIRECT) = 0;
+ audio_policy_output_flags_t flags = AUDIO_POLICY_OUTPUT_FLAG_NONE) = 0;
virtual status_t startOutput(audio_io_handle_t output,
audio_stream_type_t stream,
int session = 0) = 0;
@@ -61,9 +60,9 @@
audio_stream_type_t stream,
int session = 0) = 0;
virtual void releaseOutput(audio_io_handle_t output) = 0;
- virtual audio_io_handle_t getInput(int inputSource,
+ virtual audio_io_handle_t getInput(audio_source_t inputSource,
uint32_t samplingRate = 0,
- uint32_t format = AUDIO_FORMAT_DEFAULT,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
uint32_t channels = 0,
audio_in_acoustics_t acoustics = (audio_in_acoustics_t)0,
int audioSession = 0) = 0;
@@ -73,10 +72,14 @@
virtual status_t initStreamVolume(audio_stream_type_t stream,
int indexMin,
int indexMax) = 0;
- virtual status_t setStreamVolumeIndex(audio_stream_type_t stream, int index) = 0;
- virtual status_t getStreamVolumeIndex(audio_stream_type_t stream, int *index) = 0;
+ virtual status_t setStreamVolumeIndex(audio_stream_type_t stream,
+ int index,
+ audio_devices_t device) = 0;
+ virtual status_t getStreamVolumeIndex(audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device) = 0;
virtual uint32_t getStrategyForStream(audio_stream_type_t stream) = 0;
- virtual uint32_t getDevicesForStream(audio_stream_type_t stream) = 0;
+ virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream) = 0;
virtual audio_io_handle_t getOutputForEffect(effect_descriptor_t *desc) = 0;
virtual status_t registerEffect(effect_descriptor_t *desc,
audio_io_handle_t io,
@@ -85,7 +88,7 @@
int id) = 0;
virtual status_t unregisterEffect(int id) = 0;
virtual status_t setEffectEnabled(int id, bool enabled) = 0;
- virtual bool isStreamActive(int stream, uint32_t inPastMs = 0) const = 0;
+ virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const = 0;
virtual status_t queryDefaultPreProcessing(int audioSession,
effect_descriptor_t *descriptors,
uint32_t *count) = 0;
diff --git a/include/media/IAudioRecord.h b/include/media/IAudioRecord.h
index 46735de..089be3b 100644
--- a/include/media/IAudioRecord.h
+++ b/include/media/IAudioRecord.h
@@ -32,22 +32,23 @@
class IAudioRecord : public IInterface
{
-public:
+public:
DECLARE_META_INTERFACE(AudioRecord);
/* After it's created the track is not active. Call start() to
* make it active. If set, the callback will start being called.
+ * tid identifies the client callback thread, or 0 if not needed.
*/
- virtual status_t start() = 0;
+ virtual status_t start(pid_t tid) = 0;
/* Stop a track. If set, the callback will cease being called and
- * obtainBuffer will return an error. Buffers that are already released
+ * obtainBuffer will return an error. Buffers that are already released
* will be processed, unless flush() is called.
*/
virtual void stop() = 0;
/* get this tracks control block */
- virtual sp<IMemory> getCblk() const = 0;
+ virtual sp<IMemory> getCblk() const = 0;
};
// ----------------------------------------------------------------------------
diff --git a/include/media/IAudioTrack.h b/include/media/IAudioTrack.h
index 47d530b..577b095 100644
--- a/include/media/IAudioTrack.h
+++ b/include/media/IAudioTrack.h
@@ -24,7 +24,7 @@
#include <utils/Errors.h>
#include <binder/IInterface.h>
#include <binder/IMemory.h>
-
+#include <utils/LinearTransform.h>
namespace android {
@@ -32,33 +32,37 @@
class IAudioTrack : public IInterface
{
-public:
+public:
DECLARE_META_INTERFACE(AudioTrack);
+ /* Get this track's control block */
+ virtual sp<IMemory> getCblk() const = 0;
+
/* After it's created the track is not active. Call start() to
* make it active. If set, the callback will start being called.
+ * tid identifies the client callback thread, or 0 if not needed.
*/
- virtual status_t start() = 0;
+ virtual status_t start(pid_t tid) = 0;
/* Stop a track. If set, the callback will cease being called and
- * obtainBuffer will return an error. Buffers that are already released
- * will be processed, unless flush() is called.
+ * obtainBuffer will return an error. Buffers that are already released
+ * will continue to be processed, unless/until flush() is called.
*/
virtual void stop() = 0;
- /* flush a stopped track. All pending buffers are discarded.
- * This function has no effect if the track is not stoped.
+ /* Flush a stopped or paused track. All pending/released buffers are discarded.
+ * This function has no effect if the track is not stopped or paused.
*/
virtual void flush() = 0;
- /* mute or unmutes this track.
- * While mutted, the callback, if set, is still called.
+ /* Mute or unmute this track.
+ * While muted, the callback, if set, is still called.
*/
virtual void mute(bool) = 0;
-
+
/* Pause a track. If set, the callback will cease being called and
- * obtainBuffer will return an error. Buffers that are already released
- * will be processed, unless flush() is called.
+ * obtainBuffer will return an error. Buffers that are already released
+ * will continue to be processed, unless/until flush() is called.
*/
virtual void pause() = 0;
@@ -67,8 +71,23 @@
*/
virtual status_t attachAuxEffect(int effectId) = 0;
- /* get this tracks control block */
- virtual sp<IMemory> getCblk() const = 0;
+
+ /* Allocate a shared memory buffer suitable for holding timed audio
+ samples */
+ virtual status_t allocateTimedBuffer(size_t size,
+ sp<IMemory>* buffer) = 0;
+
+ /* Queue a buffer obtained via allocateTimedBuffer for playback at the given
+ timestamp */
+ virtual status_t queueTimedBuffer(const sp<IMemory>& buffer,
+ int64_t pts) = 0;
+
+ /* Define the linear transform that will be applied to the timestamps
+ given to queueTimedBuffer (which are expressed in media time).
+ Target specifies whether this transform converts media time to local time
+ or Tungsten time. The values for target are defined in AudioTrack.h */
+ virtual status_t setMediaTimeTransform(const LinearTransform& xform,
+ int target) = 0;
};
// ----------------------------------------------------------------------------
diff --git a/include/media/IMediaMetadataRetriever.h b/include/media/IMediaMetadataRetriever.h
index 1c1c268..6dbb2d7 100644
--- a/include/media/IMediaMetadataRetriever.h
+++ b/include/media/IMediaMetadataRetriever.h
@@ -56,4 +56,3 @@
}; // namespace android
#endif // ANDROID_IMEDIAMETADATARETRIEVER_H
-
diff --git a/include/media/IMediaPlayer.h b/include/media/IMediaPlayer.h
index e905903..00facc5 100644
--- a/include/media/IMediaPlayer.h
+++ b/include/media/IMediaPlayer.h
@@ -21,6 +21,11 @@
#include <binder/IInterface.h>
#include <binder/Parcel.h>
#include <utils/KeyedVector.h>
+#include <system/audio.h>
+
+// Fwd decl to make sure everyone agrees that the scope of struct sockaddr_in is
+// global, and not in android::
+struct sockaddr_in;
namespace android {
@@ -51,13 +56,15 @@
virtual status_t getCurrentPosition(int* msec) = 0;
virtual status_t getDuration(int* msec) = 0;
virtual status_t reset() = 0;
- virtual status_t setAudioStreamType(int type) = 0;
+ virtual status_t setAudioStreamType(audio_stream_type_t type) = 0;
virtual status_t setLooping(int loop) = 0;
virtual status_t setVolume(float leftVolume, float rightVolume) = 0;
virtual status_t setAuxEffectSendLevel(float level) = 0;
virtual status_t attachAuxEffect(int effectId) = 0;
virtual status_t setParameter(int key, const Parcel& request) = 0;
virtual status_t getParameter(int key, Parcel* reply) = 0;
+ virtual status_t setRetransmitEndpoint(const struct sockaddr_in* endpoint) = 0;
+ virtual status_t setNextPlayer(const sp<IMediaPlayer>& next) = 0;
// Invoke a generic method on the player by using opaque parcels
// for the request and reply.
diff --git a/include/media/IMediaPlayerClient.h b/include/media/IMediaPlayerClient.h
index daec1c7..8f1843e 100644
--- a/include/media/IMediaPlayerClient.h
+++ b/include/media/IMediaPlayerClient.h
@@ -45,4 +45,3 @@
}; // namespace android
#endif // ANDROID_IMEDIAPLAYERCLIENT_H
-
diff --git a/include/media/IMediaPlayerService.h b/include/media/IMediaPlayerService.h
index 93bbe13..4f46fcd 100644
--- a/include/media/IMediaPlayerService.h
+++ b/include/media/IMediaPlayerService.h
@@ -23,6 +23,7 @@
#include <utils/String8.h>
#include <binder/IInterface.h>
#include <binder/Parcel.h>
+#include <system/audio.h>
#include <media/IMediaPlayerClient.h>
#include <media/IMediaPlayer.h>
@@ -43,8 +44,8 @@
virtual sp<IMediaMetadataRetriever> createMetadataRetriever(pid_t pid) = 0;
virtual sp<IMediaPlayer> create(pid_t pid, const sp<IMediaPlayerClient>& client, int audioSessionId = 0) = 0;
- virtual sp<IMemory> decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, int* pFormat) = 0;
- virtual sp<IMemory> decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, int* pFormat) = 0;
+ virtual sp<IMemory> decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat) = 0;
+ virtual sp<IMemory> decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat) = 0;
virtual sp<IOMX> getOMX() = 0;
// codecs and audio devices usage tracking for the battery app
diff --git a/include/media/IMediaRecorderClient.h b/include/media/IMediaRecorderClient.h
index 0058ef2..e7d0229 100644
--- a/include/media/IMediaRecorderClient.h
+++ b/include/media/IMediaRecorderClient.h
@@ -45,4 +45,3 @@
}; // namespace android
#endif // ANDROID_IMEDIARECORDERCLIENT_H
-
diff --git a/include/media/IOMX.h b/include/media/IOMX.h
index c4cc947..a295e9a 100644
--- a/include/media/IOMX.h
+++ b/include/media/IOMX.h
@@ -42,10 +42,10 @@
typedef void *buffer_id;
typedef void *node_id;
- // Given the calling process' pid, returns true iff
+ // Given a node_id and the calling process' pid, returns true iff
// the implementation of the OMX interface lives in the same
// process.
- virtual bool livesLocally(pid_t pid) = 0;
+ virtual bool livesLocally(node_id node, pid_t pid) = 0;
struct ComponentInfo {
String8 mName;
diff --git a/include/media/JetPlayer.h b/include/media/JetPlayer.h
index 16764a9..38a3e44 100644
--- a/include/media/JetPlayer.h
+++ b/include/media/JetPlayer.h
@@ -22,7 +22,7 @@
#include <libsonivox/jet.h>
#include <libsonivox/eas_types.h>
-#include "AudioTrack.h"
+#include <media/AudioTrack.h>
namespace android {
@@ -40,13 +40,13 @@
static const int JET_NUMQUEUEDSEGMENT_UPDATE = 3;
static const int JET_PAUSE_UPDATE = 4;
- JetPlayer(jobject javaJetPlayer,
- int maxTracks = 32,
+ JetPlayer(jobject javaJetPlayer,
+ int maxTracks = 32,
int trackBufferSize = 1200);
~JetPlayer();
int init();
int release();
-
+
int loadFromFile(const char* url);
int loadFromFD(const int fd, const long long offset, const long long length);
int closeFile();
@@ -60,12 +60,11 @@
int clearQueue();
void setEventCallback(jetevent_callback callback);
-
+
int getMaxTracks() { return mMaxTracks; };
private:
- static int renderThread(void*);
int render();
void fireUpdateOnStatusChange();
void fireEventsFromJetQueue();
@@ -89,14 +88,36 @@
int mMaxTracks; // max number of MIDI tracks, usually 32
EAS_DATA_HANDLE mEasData;
EAS_FILE_LOCATOR mEasJetFileLoc;
- EAS_PCM* mAudioBuffer;// EAS renders the MIDI data into this buffer,
+ EAS_PCM* mAudioBuffer;// EAS renders the MIDI data into this buffer,
AudioTrack* mAudioTrack; // and we play it in this audio track
int mTrackBufferSize;
S_JET_STATUS mJetStatus;
S_JET_STATUS mPreviousJetStatus;
- char mJetFilePath[256];
+ char mJetFilePath[PATH_MAX];
+ class JetPlayerThread : public Thread {
+ public:
+ JetPlayerThread(JetPlayer *player) : mPlayer(player) {
+ }
+
+ protected:
+ virtual ~JetPlayerThread() {}
+
+ private:
+ JetPlayer *mPlayer;
+
+ bool threadLoop() {
+ int result;
+ result = mPlayer->render();
+ return false;
+ }
+
+ JetPlayerThread(const JetPlayerThread &);
+ JetPlayerThread &operator=(const JetPlayerThread &);
+ };
+
+ sp<JetPlayerThread> mThread;
}; // end class JetPlayer
diff --git a/include/media/MediaMetadataRetrieverInterface.h b/include/media/MediaMetadataRetrieverInterface.h
index 27b7e4d..ecc3b65 100644
--- a/include/media/MediaMetadataRetrieverInterface.h
+++ b/include/media/MediaMetadataRetrieverInterface.h
@@ -1,6 +1,6 @@
/*
**
-** Copyright (C) 2008 The Android Open Source Project
+** Copyright (C) 2008 The Android Open Source Project
**
** Licensed under the Apache License, Version 2.0 (the "License");
** you may not use this file except in compliance with the License.
@@ -56,4 +56,3 @@
}; // namespace android
#endif // ANDROID_MEDIAMETADATARETRIEVERINTERFACE_H
-
diff --git a/include/media/MediaPlayerInterface.h b/include/media/MediaPlayerInterface.h
index 80f43a3..d4aa233 100644
--- a/include/media/MediaPlayerInterface.h
+++ b/include/media/MediaPlayerInterface.h
@@ -29,6 +29,10 @@
#include <media/AudioSystem.h>
#include <media/Metadata.h>
+// Fwd decl to make sure everyone agrees that the scope of struct sockaddr_in is
+// global, and not in android::
+struct sockaddr_in;
+
namespace android {
class Parcel;
@@ -46,6 +50,9 @@
// The shared library with the test player is passed passed as an
// argument to the 'test:' url in the setDataSource call.
TEST_PLAYER = 5,
+
+ AAH_RX_PLAYER = 100,
+ AAH_TX_PLAYER = 101,
};
@@ -53,6 +60,8 @@
#define DEFAULT_AUDIOSINK_BUFFERSIZE 1200
#define DEFAULT_AUDIOSINK_SAMPLERATE 44100
+// when the channel mask isn't known, use the channel count to derive a mask in AudioSink::open()
+#define CHANNEL_MASK_USE_CHANNEL_ORDER 0
// callback mechanism for passing messages to MediaPlayer object
typedef void (*notify_callback_f)(void* cookie,
@@ -84,8 +93,8 @@
// If no callback is specified, use the "write" API below to submit
// audio data.
virtual status_t open(
- uint32_t sampleRate, int channelCount,
- int format=AUDIO_FORMAT_PCM_16_BIT,
+ uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
+ audio_format_t format=AUDIO_FORMAT_PCM_16_BIT,
int bufferCount=DEFAULT_AUDIOSINK_BUFFERCOUNT,
AudioCallback cb = NULL,
void *cookie = NULL) = 0;
@@ -96,6 +105,9 @@
virtual void flush() = 0;
virtual void pause() = 0;
virtual void close() = 0;
+
+ virtual status_t setPlaybackRatePermille(int32_t rate) { return INVALID_OPERATION; }
+ virtual bool needsTrailingPadding() { return true; }
};
MediaPlayerBase() : mCookie(0), mNotify(0) {}
@@ -136,6 +148,14 @@
virtual status_t setParameter(int key, const Parcel &request) = 0;
virtual status_t getParameter(int key, Parcel *reply) = 0;
+ // Right now, only the AAX TX player supports this functionality. For now,
+ // provide a default implementation which indicates a lack of support for
+ // this functionality to make life easier for all of the other media player
+ // maintainers out there.
+ virtual status_t setRetransmitEndpoint(const struct sockaddr_in* endpoint) {
+ return INVALID_OPERATION;
+ }
+
// Invoke a generic method on the player by using opaque parcels
// for the request and reply.
//
@@ -199,7 +219,7 @@
virtual ~MediaPlayerHWInterface() {}
virtual bool hardwareOutput() { return true; }
virtual status_t setVolume(float leftVolume, float rightVolume) = 0;
- virtual status_t setAudioStreamType(int streamType) = 0;
+ virtual status_t setAudioStreamType(audio_stream_type_t streamType) = 0;
};
}; // namespace android
diff --git a/include/media/MediaProfiles.h b/include/media/MediaProfiles.h
index 250f267..9fc962c 100644
--- a/include/media/MediaProfiles.h
+++ b/include/media/MediaProfiles.h
@@ -516,4 +516,3 @@
}; // namespace android
#endif // ANDROID_MEDIAPROFILES_H
-
diff --git a/include/media/MemoryLeakTrackUtil.h b/include/media/MemoryLeakTrackUtil.h
index 290b748..d2618aa 100644
--- a/include/media/MemoryLeakTrackUtil.h
+++ b/include/media/MemoryLeakTrackUtil.h
@@ -1,4 +1,3 @@
-
/*
* Copyright 2011, The Android Open Source Project
*
@@ -19,7 +18,7 @@
namespace android {
/*
- * Dump the memory adddress of the calling process to the given fd.
+ * Dump the memory address of the calling process to the given fd.
*/
extern void dumpMemoryAddresses(int fd);
diff --git a/include/media/ToneGenerator.h b/include/media/ToneGenerator.h
index 1ad1f26..df0c97e 100644
--- a/include/media/ToneGenerator.h
+++ b/include/media/ToneGenerator.h
@@ -151,10 +151,10 @@
NUM_SUP_TONES = LAST_SUP_TONE-FIRST_SUP_TONE+1
};
- ToneGenerator(int streamType, float volume, bool threadCanCallJava = false);
+ ToneGenerator(audio_stream_type_t streamType, float volume, bool threadCanCallJava = false);
~ToneGenerator();
- bool startTone(int toneType, int durationMs = -1);
+ bool startTone(tone_type toneType, int durationMs = -1);
void stopTone();
bool isInited() { return (mState == TONE_IDLE)?false:true;}
@@ -266,7 +266,7 @@
Mutex mCbkCondLock; // Mutex associated to mWaitCbkCond
Condition mWaitCbkCond; // condition enabling interface to wait for audio callback completion after a change is requested
float mVolume; // Volume applied to audio track
- int mStreamType; // Audio stream used for output
+ audio_stream_type_t mStreamType; // Audio stream used for output
unsigned int mProcessSize; // Size of audio blocks generated at a time by audioCallback() (in PCM frames).
bool initAudioTrack();
@@ -274,7 +274,7 @@
bool prepareWave();
unsigned int numWaves(unsigned int segmentIdx);
void clearWaveGens();
- int getToneForRegion(int toneType);
+ tone_type getToneForRegion(tone_type toneType);
// WaveGenerator generates a single sine wave
class WaveGenerator {
diff --git a/include/media/Visualizer.h b/include/media/Visualizer.h
index 5d2c874..60fa15b 100644
--- a/include/media/Visualizer.h
+++ b/include/media/Visualizer.h
@@ -66,8 +66,8 @@
* See AudioEffect constructor for details on parameters.
*/
Visualizer(int32_t priority = 0,
- effect_callback_t cbf = 0,
- void* user = 0,
+ effect_callback_t cbf = NULL,
+ void* user = NULL,
int sessionId = 0);
~Visualizer();
@@ -143,7 +143,7 @@
void periodicCapture();
uint32_t initCaptureSize();
- Mutex mLock;
+ Mutex mCaptureLock;
uint32_t mCaptureRate;
uint32_t mCaptureSize;
uint32_t mSampleRate;
diff --git a/include/media/mediaplayer.h b/include/media/mediaplayer.h
index e6a0cc5..a68ab4e 100644
--- a/include/media/mediaplayer.h
+++ b/include/media/mediaplayer.h
@@ -17,6 +17,8 @@
#ifndef ANDROID_MEDIAPLAYER_H
#define ANDROID_MEDIAPLAYER_H
+#include <arpa/inet.h>
+
#include <binder/IMemory.h>
#include <media/IMediaPlayerClient.h>
#include <media/IMediaPlayer.h>
@@ -94,6 +96,9 @@
enum media_info_type {
// 0xx
MEDIA_INFO_UNKNOWN = 1,
+ // The player was started because it was used as the next player for another
+ // player, which just completed playback
+ MEDIA_INFO_STARTED_AS_NEXT = 2,
// 7xx
// The video is too complex for the decoder: it can't decode frames fast
// enough. Possibly only the audio plays fine at this stage.
@@ -115,6 +120,9 @@
MEDIA_INFO_NOT_SEEKABLE = 801,
// New media metadata is available.
MEDIA_INFO_METADATA_UPDATE = 802,
+
+ //9xx
+ MEDIA_INFO_TIMED_TEXT_ERROR = 900,
};
@@ -135,9 +143,6 @@
// The same enum space is used for both set and get, in case there are future keys that
// can be both set and get. But as of now, all parameters are either set only or get only.
enum media_parameter_keys {
- KEY_PARAMETER_TIMED_TEXT_TRACK_INDEX = 1000, // set only
- KEY_PARAMETER_TIMED_TEXT_ADD_OUT_OF_BAND_SOURCE = 1001, // set only
-
// Streaming/buffering parameters
KEY_PARAMETER_CACHE_STAT_COLLECT_FREQ_MS = 1100, // set only
@@ -145,6 +150,26 @@
// audio track, or zero for error (e.g. no audio track) or unknown.
KEY_PARAMETER_AUDIO_CHANNEL_COUNT = 1200, // get only
+ // Playback rate expressed in permille (1000 is normal speed), saved as int32_t, with negative
+ // values used for rewinding or reverse playback.
+ KEY_PARAMETER_PLAYBACK_RATE_PERMILLE = 1300, // set only
+};
+
+// Keep INVOKE_ID_* in sync with MediaPlayer.java.
+enum media_player_invoke_ids {
+ INVOKE_ID_GET_TRACK_INFO = 1,
+ INVOKE_ID_ADD_EXTERNAL_SOURCE = 2,
+ INVOKE_ID_ADD_EXTERNAL_SOURCE_FD = 3,
+ INVOKE_ID_SELECT_TRACK = 4,
+ INVOKE_ID_UNSELECT_TRACK = 5,
+};
+
+// Keep MEDIA_TRACK_TYPE_* in sync with MediaPlayer.java.
+enum media_track_type {
+ MEDIA_TRACK_TYPE_UNKNOWN = 0,
+ MEDIA_TRACK_TYPE_VIDEO = 1,
+ MEDIA_TRACK_TYPE_AUDIO = 2,
+ MEDIA_TRACK_TYPE_TIMEDTEXT = 3,
};
// ----------------------------------------------------------------------------
@@ -185,13 +210,13 @@
status_t getCurrentPosition(int *msec);
status_t getDuration(int *msec);
status_t reset();
- status_t setAudioStreamType(int type);
+ status_t setAudioStreamType(audio_stream_type_t type);
status_t setLooping(int loop);
bool isLooping();
status_t setVolume(float leftVolume, float rightVolume);
void notify(int msg, int ext1, int ext2, const Parcel *obj = NULL);
- static sp<IMemory> decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, int* pFormat);
- static sp<IMemory> decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, int* pFormat);
+ static sp<IMemory> decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat);
+ static sp<IMemory> decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat);
status_t invoke(const Parcel& request, Parcel *reply);
status_t setMetadataFilter(const Parcel& filter);
status_t getMetadata(bool update_only, bool apply_filter, Parcel *metadata);
@@ -201,6 +226,8 @@
status_t attachAuxEffect(int effectId);
status_t setParameter(int key, const Parcel& request);
status_t getParameter(int key, Parcel* reply);
+ status_t setRetransmitEndpoint(const char* addrString, uint16_t port);
+ status_t setNextMediaPlayer(const sp<MediaPlayer>& player);
private:
void clear_l();
@@ -209,6 +236,7 @@
status_t getDuration_l(int *msec);
status_t attachNewPlayer(const sp<IMediaPlayer>& player);
status_t reset_l();
+ status_t doSetRetransmitEndpoint(const sp<IMediaPlayer>& player);
sp<IMediaPlayer> mPlayer;
thread_id_t mLockThreadId;
@@ -223,7 +251,7 @@
int mSeekPosition;
bool mPrepareSync;
status_t mPrepareStatus;
- int mStreamType;
+ audio_stream_type_t mStreamType;
bool mLoop;
float mLeftVolume;
float mRightVolume;
@@ -231,6 +259,8 @@
int mVideoHeight;
int mAudioSessionId;
float mSendLevel;
+ struct sockaddr_in mRetransmitEndpoint;
+ bool mRetransmitEndpointValid;
};
}; // namespace android
diff --git a/include/media/stagefright/AACWriter.h b/include/media/stagefright/AACWriter.h
index fa3ab8a..49397ee 100644
--- a/include/media/stagefright/AACWriter.h
+++ b/include/media/stagefright/AACWriter.h
@@ -34,7 +34,7 @@
virtual status_t addSource(const sp<MediaSource> &source);
virtual bool reachedEOS();
virtual status_t start(MetaData *params = NULL);
- virtual status_t stop();
+ virtual status_t stop() { return reset(); }
virtual status_t pause();
protected:
@@ -66,6 +66,7 @@
bool exceedsFileSizeLimit();
bool exceedsFileDurationLimit();
status_t writeAdtsHeader(uint32_t frameLength);
+ status_t reset();
DISALLOW_EVIL_CONSTRUCTORS(AACWriter);
};
diff --git a/include/media/stagefright/ACodec.h b/include/media/stagefright/ACodec.h
index 3963d9c..fa1a416 100644
--- a/include/media/stagefright/ACodec.h
+++ b/include/media/stagefright/ACodec.h
@@ -22,6 +22,7 @@
#include <android/native_window.h>
#include <media/IOMX.h>
#include <media/stagefright/foundation/AHierarchicalStateMachine.h>
+#include <OMX_Audio.h>
namespace android {
@@ -37,6 +38,9 @@
kWhatFlushCompleted = 'fcom',
kWhatOutputFormatChanged = 'outC',
kWhatError = 'erro',
+ kWhatComponentAllocated = 'cAll',
+ kWhatComponentConfigured = 'cCon',
+ kWhatBuffersAllocated = 'allc',
};
ACodec();
@@ -45,7 +49,11 @@
void initiateSetup(const sp<AMessage> &msg);
void signalFlush();
void signalResume();
- void initiateShutdown();
+ void initiateShutdown(bool keepComponentAllocated = false);
+
+ void initiateAllocateComponent(const sp<AMessage> &msg);
+ void initiateConfigureComponent(const sp<AMessage> &msg);
+ void initiateStart();
protected:
virtual ~ACodec();
@@ -53,6 +61,7 @@
private:
struct BaseState;
struct UninitializedState;
+ struct LoadedState;
struct LoadedToIdleState;
struct IdleToExecutingState;
struct ExecutingState;
@@ -70,6 +79,9 @@
kWhatFlush = 'flus',
kWhatResume = 'resm',
kWhatDrainDeferredMessages = 'drai',
+ kWhatAllocateComponent = 'allo',
+ kWhatConfigureComponent = 'conf',
+ kWhatStart = 'star',
};
enum {
@@ -96,6 +108,7 @@
sp<AMessage> mNotify;
sp<UninitializedState> mUninitializedState;
+ sp<LoadedState> mLoadedState;
sp<LoadedToIdleState> mLoadedToIdleState;
sp<IdleToExecutingState> mIdleToExecutingState;
sp<ExecutingState> mExecutingState;
@@ -105,6 +118,7 @@
sp<FlushingState> mFlushingState;
AString mComponentName;
+ uint32_t mQuirks;
sp<IOMX> mOMX;
IOMX::node_id mNode;
sp<MemoryDealer> mDealer[2];
@@ -118,6 +132,13 @@
List<sp<AMessage> > mDeferredQueue;
bool mSentFormat;
+ bool mIsEncoder;
+
+ bool mShutdownInProgress;
+
+ // If "mKeepComponentAllocated" we only transition back to Loaded state
+ // and do not release the component instance.
+ bool mKeepComponentAllocated;
status_t allocateBuffersOnPort(OMX_U32 portIndex);
status_t freeBuffersOnPort(OMX_U32 portIndex);
@@ -132,8 +153,8 @@
uint32_t portIndex, IOMX::buffer_id bufferID,
ssize_t *index = NULL);
- void setComponentRole(bool isEncoder, const char *mime);
- void configureCodec(const char *mime, const sp<AMessage> &msg);
+ status_t setComponentRole(bool isEncoder, const char *mime);
+ status_t configureCodec(const char *mime, const sp<AMessage> &msg);
status_t setVideoPortFormatType(
OMX_U32 portIndex,
@@ -145,20 +166,37 @@
status_t setupVideoDecoder(
const char *mime, int32_t width, int32_t height);
+ status_t setupVideoEncoder(
+ const char *mime, const sp<AMessage> &msg);
+
status_t setVideoFormatOnPort(
OMX_U32 portIndex,
int32_t width, int32_t height,
OMX_VIDEO_CODINGTYPE compressionFormat);
- status_t setupAACDecoder(int32_t numChannels, int32_t sampleRate);
- status_t setupAMRDecoder(bool isWAMR);
- status_t setupG711Decoder(int32_t numChannels);
+ status_t setupAACCodec(
+ bool encoder,
+ int32_t numChannels, int32_t sampleRate, int32_t bitRate);
+
+ status_t selectAudioPortFormat(
+ OMX_U32 portIndex, OMX_AUDIO_CODINGTYPE desiredFormat);
+
+ status_t setupAMRCodec(bool encoder, bool isWAMR, int32_t bitRate);
+ status_t setupG711Codec(bool encoder, int32_t numChannels);
status_t setupRawAudioFormat(
OMX_U32 portIndex, int32_t sampleRate, int32_t numChannels);
status_t setMinBufferSize(OMX_U32 portIndex, size_t size);
+ status_t setupMPEG4EncoderParameters(const sp<AMessage> &msg);
+ status_t setupH263EncoderParameters(const sp<AMessage> &msg);
+ status_t setupAVCEncoderParameters(const sp<AMessage> &msg);
+
+ status_t verifySupportForProfileAndLevel(int32_t profile, int32_t level);
+ status_t configureBitrate(int32_t bitrate);
+ status_t setupErrorCorrectionParameters();
+
status_t initNativeWindow();
// Returns true iff all buffers on the given port have status OWNED_BY_US.
@@ -173,7 +211,9 @@
void sendFormatChange();
- void signalError(OMX_ERRORTYPE error = OMX_ErrorUndefined);
+ void signalError(
+ OMX_ERRORTYPE error = OMX_ErrorUndefined,
+ status_t internalError = UNKNOWN_ERROR);
DISALLOW_EVIL_CONSTRUCTORS(ACodec);
};
diff --git a/include/media/stagefright/AMRWriter.h b/include/media/stagefright/AMRWriter.h
index 62d57b4..392f968 100644
--- a/include/media/stagefright/AMRWriter.h
+++ b/include/media/stagefright/AMRWriter.h
@@ -37,7 +37,7 @@
virtual status_t addSource(const sp<MediaSource> &source);
virtual bool reachedEOS();
virtual status_t start(MetaData *params = NULL);
- virtual status_t stop();
+ virtual status_t stop() { return reset(); }
virtual status_t pause();
protected:
@@ -60,6 +60,7 @@
status_t threadFunc();
bool exceedsFileSizeLimit();
bool exceedsFileDurationLimit();
+ status_t reset();
AMRWriter(const AMRWriter &);
AMRWriter &operator=(const AMRWriter &);
diff --git a/include/media/stagefright/AudioPlayer.h b/include/media/stagefright/AudioPlayer.h
index 0b79324..70c47ae 100644
--- a/include/media/stagefright/AudioPlayer.h
+++ b/include/media/stagefright/AudioPlayer.h
@@ -64,6 +64,8 @@
bool isSeeking();
bool reachedEOS(status_t *finalStatus);
+ status_t setPlaybackRatePermille(int32_t ratePermille);
+
private:
friend class VideoEditorAudioPlayer;
sp<MediaSource> mSource;
diff --git a/include/media/stagefright/AudioSource.h b/include/media/stagefright/AudioSource.h
index 19bd31b..f5466e8 100644
--- a/include/media/stagefright/AudioSource.h
+++ b/include/media/stagefright/AudioSource.h
@@ -34,13 +34,13 @@
// Note that the "channels" parameter is _not_ the number of channels,
// but a bitmask of audio_channels_t constants.
AudioSource(
- int inputSource, uint32_t sampleRate,
+ audio_source_t inputSource, uint32_t sampleRate,
uint32_t channels = AUDIO_CHANNEL_IN_MONO);
status_t initCheck() const;
virtual status_t start(MetaData *params = NULL);
- virtual status_t stop();
+ virtual status_t stop() { return reset(); }
virtual sp<MetaData> getFormat();
// Returns the maximum amplitude since last call.
@@ -95,8 +95,10 @@
int32_t startFrame, int32_t rampDurationFrames,
uint8_t *data, size_t bytes);
+ void queueInputBuffer_l(MediaBuffer *buffer, int64_t timeUs);
void releaseQueuedFrames_l();
void waitOutstandingEncodingFrames_l();
+ status_t reset();
AudioSource(const AudioSource &);
AudioSource &operator=(const AudioSource &);
diff --git a/include/media/stagefright/CameraSource.h b/include/media/stagefright/CameraSource.h
index 446720b..5a35358 100644
--- a/include/media/stagefright/CameraSource.h
+++ b/include/media/stagefright/CameraSource.h
@@ -79,7 +79,7 @@
virtual ~CameraSource();
virtual status_t start(MetaData *params = NULL);
- virtual status_t stop();
+ virtual status_t stop() { return reset(); }
virtual status_t read(
MediaBuffer **buffer, const ReadOptions *options = NULL);
@@ -163,7 +163,6 @@
bool storeMetaDataInVideoBuffers);
virtual void startCameraRecording();
- virtual void stopCameraRecording();
virtual void releaseRecordingFrame(const sp<IMemory>& frame);
// Returns true if need to skip the current frame.
@@ -220,7 +219,9 @@
status_t checkFrameRate(const CameraParameters& params,
int32_t frameRate);
+ void stopCameraRecording();
void releaseCamera();
+ status_t reset();
CameraSource(const CameraSource &);
CameraSource &operator=(const CameraSource &);
diff --git a/include/media/stagefright/CameraSourceTimeLapse.h b/include/media/stagefright/CameraSourceTimeLapse.h
index b060691..0936da2 100644
--- a/include/media/stagefright/CameraSourceTimeLapse.h
+++ b/include/media/stagefright/CameraSourceTimeLapse.h
@@ -121,9 +121,6 @@
// Wrapper over CameraSource::read() to implement quick stop.
virtual status_t read(MediaBuffer **buffer, const ReadOptions *options = NULL);
- // For video camera case, just stops the camera's video recording.
- virtual void stopCameraRecording();
-
// mSkipCurrentFrame is set to true in dataCallbackTimestamp() if the current
// frame needs to be skipped and this function just returns the value of mSkipCurrentFrame.
virtual bool skipCurrentFrame(int64_t timestampUs);
diff --git a/include/media/stagefright/DataSource.h b/include/media/stagefright/DataSource.h
index 713af92..00d583e 100644
--- a/include/media/stagefright/DataSource.h
+++ b/include/media/stagefright/DataSource.h
@@ -81,7 +81,7 @@
static void RegisterDefaultSniffers();
// for DRM
- virtual sp<DecryptHandle> DrmInitialization() {
+ virtual sp<DecryptHandle> DrmInitialization(const char *mime = NULL) {
return NULL;
}
virtual void getDrmInfo(sp<DecryptHandle> &handle, DrmManagerClient **client) {};
diff --git a/include/media/stagefright/FileSource.h b/include/media/stagefright/FileSource.h
index 6cf86dc..d994cb3 100644
--- a/include/media/stagefright/FileSource.h
+++ b/include/media/stagefright/FileSource.h
@@ -38,7 +38,7 @@
virtual status_t getSize(off64_t *size);
- virtual sp<DecryptHandle> DrmInitialization();
+ virtual sp<DecryptHandle> DrmInitialization(const char *mime);
virtual void getDrmInfo(sp<DecryptHandle> &handle, DrmManagerClient **client);
diff --git a/include/media/stagefright/HardwareAPI.h b/include/media/stagefright/HardwareAPI.h
deleted file mode 100644
index 32eed3f..0000000
--- a/include/media/stagefright/HardwareAPI.h
+++ /dev/null
@@ -1,113 +0,0 @@
-/*
- * Copyright (C) 2009 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef HARDWARE_API_H_
-
-#define HARDWARE_API_H_
-
-#include <media/stagefright/OMXPluginBase.h>
-#include <ui/android_native_buffer.h>
-#include <utils/RefBase.h>
-
-#include <OMX_Component.h>
-
-namespace android {
-
-// A pointer to this struct is passed to the OMX_SetParameter when the extension
-// index for the 'OMX.google.android.index.enableAndroidNativeBuffers' extension
-// is given.
-//
-// When Android native buffer use is disabled for a port (the default state),
-// the OMX node should operate as normal, and expect UseBuffer calls to set its
-// buffers. This is the mode that will be used when CPU access to the buffer is
-// required.
-//
-// When Android native buffer use has been enabled for a given port, the video
-// color format for the port is to be interpreted as an Android pixel format
-// rather than an OMX color format. The node should then expect to receive
-// UseAndroidNativeBuffer calls (via OMX_SetParameter) rather than UseBuffer
-// calls for that port.
-struct EnableAndroidNativeBuffersParams {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_BOOL enable;
-};
-
-// A pointer to this struct is passed to OMX_SetParameter() when the extension
-// index "OMX.google.android.index.storeMetaDataInBuffers"
-// is given.
-//
-// When meta data is stored in the video buffers passed between OMX clients
-// and OMX components, interpretation of the buffer data is up to the
-// buffer receiver, and the data may or may not be the actual video data, but
-// some information helpful for the receiver to locate the actual data.
-// The buffer receiver thus needs to know how to interpret what is stored
-// in these buffers, with mechanisms pre-determined externally. How to
-// interpret the meta data is outside of the scope of this method.
-//
-// Currently, this is specifically used to pass meta data from video source
-// (camera component, for instance) to video encoder to avoid memcpying of
-// input video frame data. To do this, bStoreMetaDta is set to OMX_TRUE.
-// If bStoreMetaData is set to false, real YUV frame data will be stored
-// in the buffers. In addition, if no OMX_SetParameter() call is made
-// with the corresponding extension index, real YUV data is stored
-// in the buffers.
-struct StoreMetaDataInBuffersParams {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_BOOL bStoreMetaData;
-};
-
-// A pointer to this struct is passed to OMX_SetParameter when the extension
-// index for the 'OMX.google.android.index.useAndroidNativeBuffer' extension is
-// given. This call will only be performed if a prior call was made with the
-// 'OMX.google.android.index.enableAndroidNativeBuffers' extension index,
-// enabling use of Android native buffers.
-struct UseAndroidNativeBufferParams {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_PTR pAppPrivate;
- OMX_BUFFERHEADERTYPE **bufferHeader;
- const sp<ANativeWindowBuffer>& nativeBuffer;
-};
-
-// A pointer to this struct is passed to OMX_GetParameter when the extension
-// index for the 'OMX.google.android.index.getAndroidNativeBufferUsage'
-// extension is given. The usage bits returned from this query will be used to
-// allocate the Gralloc buffers that get passed to the useAndroidNativeBuffer
-// command.
-struct GetAndroidNativeBufferUsageParams {
- OMX_U32 nSize; // IN
- OMX_VERSIONTYPE nVersion; // IN
- OMX_U32 nPortIndex; // IN
- OMX_U32 nUsage; // OUT
-};
-
-// An enum OMX_COLOR_FormatAndroidOpaque to indicate an opaque colorformat
-// is declared in media/stagefright/openmax/OMX_IVCommon.h
-// This will inform the encoder that the actual
-// colorformat will be relayed by the GRalloc Buffers.
-// OMX_COLOR_FormatAndroidOpaque = 0x7F000001,
-
-
-} // namespace android
-
-extern android::OMXPluginBase *createOMXPlugin();
-
-#endif // HARDWARE_API_H_
diff --git a/include/media/stagefright/MPEG2TSWriter.h b/include/media/stagefright/MPEG2TSWriter.h
index e4c1c49..a7c9ecf 100644
--- a/include/media/stagefright/MPEG2TSWriter.h
+++ b/include/media/stagefright/MPEG2TSWriter.h
@@ -37,7 +37,7 @@
virtual status_t addSource(const sp<MediaSource> &source);
virtual status_t start(MetaData *param = NULL);
- virtual status_t stop();
+ virtual status_t stop() { return reset(); }
virtual status_t pause();
virtual bool reachedEOS();
virtual status_t dump(int fd, const Vector<String16>& args);
@@ -78,6 +78,7 @@
void writeAccessUnit(int32_t sourceIndex, const sp<ABuffer> &buffer);
ssize_t internalWrite(const void *data, size_t size);
+ status_t reset();
DISALLOW_EVIL_CONSTRUCTORS(MPEG2TSWriter);
};
diff --git a/include/media/stagefright/MPEG4Writer.h b/include/media/stagefright/MPEG4Writer.h
index 77166ed..0409b30 100644
--- a/include/media/stagefright/MPEG4Writer.h
+++ b/include/media/stagefright/MPEG4Writer.h
@@ -37,7 +37,7 @@
virtual status_t addSource(const sp<MediaSource> &source);
virtual status_t start(MetaData *param = NULL);
- virtual status_t stop();
+ virtual status_t stop() { return reset(); }
virtual status_t pause();
virtual bool reachedEOS();
virtual status_t dump(int fd, const Vector<String16>& args);
@@ -184,6 +184,7 @@
void writeLongitude(int degreex10000);
void sendSessionSummary();
void release();
+ status_t reset();
MPEG4Writer(const MPEG4Writer &);
MPEG4Writer &operator=(const MPEG4Writer &);
diff --git a/include/media/stagefright/MediaCodec.h b/include/media/stagefright/MediaCodec.h
new file mode 100644
index 0000000..72ac56a
--- /dev/null
+++ b/include/media/stagefright/MediaCodec.h
@@ -0,0 +1,192 @@
+/*
+ * Copyright 2012, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef MEDIA_CODEC_H_
+
+#define MEDIA_CODEC_H_
+
+#include <gui/ISurfaceTexture.h>
+#include <media/stagefright/foundation/AHandler.h>
+#include <utils/Vector.h>
+
+namespace android {
+
+struct ABuffer;
+struct ACodec;
+struct AMessage;
+struct SoftwareRenderer;
+struct SurfaceTextureClient;
+
+struct MediaCodec : public AHandler {
+ enum ConfigureFlags {
+ CONFIGURE_FLAG_ENCODE = 1,
+ };
+
+ enum BufferFlags {
+ BUFFER_FLAG_SYNCFRAME = 1,
+ BUFFER_FLAG_CODECCONFIG = 2,
+ BUFFER_FLAG_EOS = 4,
+ };
+
+ static sp<MediaCodec> CreateByType(
+ const sp<ALooper> &looper, const char *mime, bool encoder);
+
+ static sp<MediaCodec> CreateByComponentName(
+ const sp<ALooper> &looper, const char *name);
+
+ status_t configure(
+ const sp<AMessage> &format,
+ const sp<SurfaceTextureClient> &nativeWindow,
+ uint32_t flags);
+
+ status_t start();
+
+ // Returns to a state in which the component remains allocated but
+ // unconfigured.
+ status_t stop();
+
+ // Client MUST call release before releasing final reference to this
+ // object.
+ status_t release();
+
+ status_t flush();
+
+ status_t queueInputBuffer(
+ size_t index,
+ size_t offset,
+ size_t size,
+ int64_t presentationTimeUs,
+ uint32_t flags);
+
+ status_t dequeueInputBuffer(size_t *index, int64_t timeoutUs = 0ll);
+
+ status_t dequeueOutputBuffer(
+ size_t *index,
+ size_t *offset,
+ size_t *size,
+ int64_t *presentationTimeUs,
+ uint32_t *flags,
+ int64_t timeoutUs = 0ll);
+
+ status_t renderOutputBufferAndRelease(size_t index);
+ status_t releaseOutputBuffer(size_t index);
+
+ status_t getOutputFormat(sp<AMessage> *format) const;
+
+ status_t getInputBuffers(Vector<sp<ABuffer> > *buffers) const;
+ status_t getOutputBuffers(Vector<sp<ABuffer> > *buffers) const;
+
+protected:
+ virtual ~MediaCodec();
+ virtual void onMessageReceived(const sp<AMessage> &msg);
+
+private:
+ enum State {
+ UNINITIALIZED,
+ INITIALIZING,
+ INITIALIZED,
+ CONFIGURING,
+ CONFIGURED,
+ STARTING,
+ STARTED,
+ FLUSHING,
+ STOPPING,
+ RELEASING,
+ };
+
+ enum {
+ kPortIndexInput = 0,
+ kPortIndexOutput = 1,
+ };
+
+ enum {
+ kWhatInit = 'init',
+ kWhatConfigure = 'conf',
+ kWhatStart = 'strt',
+ kWhatStop = 'stop',
+ kWhatRelease = 'rele',
+ kWhatDequeueInputBuffer = 'deqI',
+ kWhatQueueInputBuffer = 'queI',
+ kWhatDequeueOutputBuffer = 'deqO',
+ kWhatReleaseOutputBuffer = 'relO',
+ kWhatGetBuffers = 'getB',
+ kWhatFlush = 'flus',
+ kWhatGetOutputFormat = 'getO',
+ kWhatDequeueInputTimedOut = 'dITO',
+ kWhatDequeueOutputTimedOut = 'dOTO',
+ kWhatCodecNotify = 'codc',
+ };
+
+ enum {
+ kFlagIsSoftwareCodec = 1,
+ kFlagOutputFormatChanged = 2,
+ kFlagOutputBuffersChanged = 4,
+ kFlagStickyError = 8,
+ kFlagDequeueInputPending = 16,
+ kFlagDequeueOutputPending = 32,
+ };
+
+ struct BufferInfo {
+ void *mBufferID;
+ sp<ABuffer> mData;
+ sp<AMessage> mNotify;
+ bool mOwnedByClient;
+ };
+
+ State mState;
+ sp<ALooper> mLooper;
+ sp<ALooper> mCodecLooper;
+ sp<ACodec> mCodec;
+ uint32_t mReplyID;
+ uint32_t mFlags;
+ sp<SurfaceTextureClient> mNativeWindow;
+ SoftwareRenderer *mSoftRenderer;
+ sp<AMessage> mOutputFormat;
+
+ List<size_t> mAvailPortBuffers[2];
+ Vector<BufferInfo> mPortBuffers[2];
+
+ int32_t mDequeueInputTimeoutGeneration;
+ uint32_t mDequeueInputReplyID;
+
+ int32_t mDequeueOutputTimeoutGeneration;
+ uint32_t mDequeueOutputReplyID;
+
+ MediaCodec(const sp<ALooper> &looper);
+
+ static status_t PostAndAwaitResponse(
+ const sp<AMessage> &msg, sp<AMessage> *response);
+
+ status_t init(const char *name, bool nameIsType, bool encoder);
+
+ void setState(State newState);
+ void returnBuffersToCodec();
+ void returnBuffersToCodecOnPort(int32_t portIndex);
+ size_t updateBuffers(int32_t portIndex, const sp<AMessage> &msg);
+ status_t onQueueInputBuffer(const sp<AMessage> &msg);
+ status_t onReleaseOutputBuffer(const sp<AMessage> &msg);
+ ssize_t dequeuePortBuffer(int32_t portIndex);
+
+ bool handleDequeueInputBuffer(uint32_t replyID, bool newRequest = false);
+ bool handleDequeueOutputBuffer(uint32_t replyID, bool newRequest = false);
+ void cancelPendingDequeueOperations();
+
+ DISALLOW_EVIL_CONSTRUCTORS(MediaCodec);
+};
+
+} // namespace android
+
+#endif // MEDIA_CODEC_H_
diff --git a/include/media/stagefright/MediaCodecList.h b/include/media/stagefright/MediaCodecList.h
new file mode 100644
index 0000000..14dc1b8
--- /dev/null
+++ b/include/media/stagefright/MediaCodecList.h
@@ -0,0 +1,95 @@
+/*
+ * Copyright 2012, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef MEDIA_CODEC_LIST_H_
+
+#define MEDIA_CODEC_LIST_H_
+
+#include <media/stagefright/foundation/ABase.h>
+#include <media/stagefright/foundation/AString.h>
+
+#include <sys/types.h>
+#include <utils/Errors.h>
+#include <utils/KeyedVector.h>
+#include <utils/Vector.h>
+
+namespace android {
+
+struct MediaCodecList {
+ static const MediaCodecList *getInstance();
+
+ ssize_t findCodecByType(
+ const char *type, bool encoder, size_t startIndex = 0) const;
+
+ ssize_t findCodecByName(const char *name) const;
+
+ const char *getCodecName(size_t index) const;
+ bool codecHasQuirk(size_t index, const char *quirkName) const;
+
+private:
+ enum Section {
+ SECTION_TOPLEVEL,
+ SECTION_DECODERS,
+ SECTION_DECODER,
+ SECTION_ENCODERS,
+ SECTION_ENCODER,
+ };
+
+ struct CodecInfo {
+ AString mName;
+ bool mIsEncoder;
+ uint32_t mTypes;
+ uint32_t mQuirks;
+ };
+
+ static MediaCodecList *sCodecList;
+
+ status_t mInitCheck;
+ Section mCurrentSection;
+ int32_t mDepth;
+
+ Vector<CodecInfo> mCodecInfos;
+ KeyedVector<AString, size_t> mCodecQuirks;
+ KeyedVector<AString, size_t> mTypes;
+
+ MediaCodecList();
+ ~MediaCodecList();
+
+ status_t initCheck() const;
+ void parseXMLFile(FILE *file);
+
+ static void StartElementHandlerWrapper(
+ void *me, const char *name, const char **attrs);
+
+ static void EndElementHandlerWrapper(void *me, const char *name);
+
+ void startElementHandler(const char *name, const char **attrs);
+ void endElementHandler(const char *name);
+
+ status_t addMediaCodecFromAttributes(bool encoder, const char **attrs);
+ void addMediaCodec(bool encoder, const char *name, const char *type = NULL);
+
+ status_t addQuirk(const char **attrs);
+ status_t addTypeFromAttributes(const char **attrs);
+ void addType(const char *name);
+
+ DISALLOW_EVIL_CONSTRUCTORS(MediaCodecList);
+};
+
+} // namespace android
+
+#endif // MEDIA_CODEC_LIST_H_
+
diff --git a/include/media/stagefright/MediaDebug.h b/include/media/stagefright/MediaDebug.h
deleted file mode 100644
index 2ca9667..0000000
--- a/include/media/stagefright/MediaDebug.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (C) 2009 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef MEDIA_DEBUG_H_
-
-#define MEDIA_DEBUG_H_
-
-#include <cutils/log.h>
-
-#define LITERAL_TO_STRING_INTERNAL(x) #x
-#define LITERAL_TO_STRING(x) LITERAL_TO_STRING_INTERNAL(x)
-
-#define CHECK_EQ(x,y) \
- LOG_ALWAYS_FATAL_IF( \
- (x) != (y), \
- __FILE__ ":" LITERAL_TO_STRING(__LINE__) " " #x " != " #y)
-
-#define CHECK(x) \
- LOG_ALWAYS_FATAL_IF( \
- !(x), \
- __FILE__ ":" LITERAL_TO_STRING(__LINE__) " " #x)
-
-#endif // MEDIA_DEBUG_H_
diff --git a/include/media/stagefright/MediaDefs.h b/include/media/stagefright/MediaDefs.h
index 2eb259e..457d5d7 100644
--- a/include/media/stagefright/MediaDefs.h
+++ b/include/media/stagefright/MediaDefs.h
@@ -54,6 +54,7 @@
extern const char *MEDIA_MIMETYPE_CONTAINER_WVM;
extern const char *MEDIA_MIMETYPE_TEXT_3GPP;
+extern const char *MEDIA_MIMETYPE_TEXT_SUBRIP;
} // namespace android
diff --git a/include/media/stagefright/MediaErrors.h b/include/media/stagefright/MediaErrors.h
index 21d00b8..dd3bf28 100644
--- a/include/media/stagefright/MediaErrors.h
+++ b/include/media/stagefright/MediaErrors.h
@@ -40,6 +40,7 @@
// Not technically an error.
INFO_FORMAT_CHANGED = MEDIA_ERROR_BASE - 12,
INFO_DISCONTINUITY = MEDIA_ERROR_BASE - 13,
+ INFO_OUTPUT_BUFFERS_CHANGED = MEDIA_ERROR_BASE - 14,
// The following constant values should be in sync with
// drm/drm_framework_common.h
diff --git a/include/media/stagefright/MediaExtractor.h b/include/media/stagefright/MediaExtractor.h
index eb45237..94090ee 100644
--- a/include/media/stagefright/MediaExtractor.h
+++ b/include/media/stagefright/MediaExtractor.h
@@ -56,10 +56,10 @@
virtual uint32_t flags() const;
// for DRM
- virtual void setDrmFlag(bool flag) {
+ void setDrmFlag(bool flag) {
mIsDrm = flag;
};
- virtual bool getDrmFlag() {
+ bool getDrmFlag() {
return mIsDrm;
}
virtual char* getDrmTrackInfo(size_t trackID, int *len) {
diff --git a/include/media/stagefright/MediaSourceSplitter.h b/include/media/stagefright/MediaSourceSplitter.h
deleted file mode 100644
index 568f4c2..0000000
--- a/include/media/stagefright/MediaSourceSplitter.h
+++ /dev/null
@@ -1,193 +0,0 @@
-/*
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-// This class provides a way to split a single media source into multiple sources.
-// The constructor takes in the real mediaSource and createClient() can then be
-// used to create multiple sources served from this real mediaSource.
-//
-// Usage:
-// - Create MediaSourceSplitter by passing in a real mediaSource from which
-// multiple duplicate channels are needed.
-// - Create a client using createClient() and use it as any other mediaSource.
-//
-// Note that multiple clients can be created using createClient() and
-// started/stopped in any order. MediaSourceSplitter stops the real source only
-// when all clients have been stopped.
-//
-// If a new client is created/started after some existing clients have already
-// started, the new client will start getting its read frames from the current
-// time.
-
-#ifndef MEDIA_SOURCE_SPLITTER_H_
-
-#define MEDIA_SOURCE_SPLITTER_H_
-
-#include <media/stagefright/MediaSource.h>
-#include <utils/threads.h>
-#include <utils/Vector.h>
-#include <utils/RefBase.h>
-
-namespace android {
-
-class MediaBuffer;
-class MetaData;
-
-class MediaSourceSplitter : public RefBase {
-public:
- // Constructor
- // mediaSource: The real mediaSource. The class keeps a reference to it to
- // implement the various clients.
- MediaSourceSplitter(sp<MediaSource> mediaSource);
-
- ~MediaSourceSplitter();
-
- // Creates a new client of base type MediaSource. Multiple clients can be
- // created which get their data through the same real mediaSource. These
- // clients can then be used like any other MediaSource, all of which provide
- // data from the same real source.
- sp<MediaSource> createClient();
-
-private:
- // Total number of clients created through createClient().
- int32_t mNumberOfClients;
-
- // reference to the real MediaSource passed to the constructor.
- sp<MediaSource> mSource;
-
- // Stores pointer to the MediaBuffer read from the real MediaSource.
- // All clients use this to implement the read() call.
- MediaBuffer *mLastReadMediaBuffer;
-
- // Status code for read from the real MediaSource. All clients return
- // this for their read().
- status_t mLastReadStatus;
-
- // Boolean telling whether the real MediaSource has started.
- bool mSourceStarted;
-
- // List of booleans, one for each client, storing whether the corresponding
- // client's start() has been called.
- Vector<bool> mClientsStarted;
-
- // Stores the number of clients which are currently started.
- int32_t mNumberOfClientsStarted;
-
- // Since different clients call read() asynchronously, we need to keep track
- // of what data is currently read into the mLastReadMediaBuffer.
- // mCurrentReadBit stores the bit for the current read buffer. This bit
- // flips each time a new buffer is read from the source.
- // mClientsDesiredReadBit stores the bit for the next desired read buffer
- // for each client. This bit flips each time read() is completed for this
- // client.
- bool mCurrentReadBit;
- Vector<bool> mClientsDesiredReadBit;
-
- // Number of clients whose current read has been completed.
- int32_t mNumberOfCurrentReads;
-
- // Boolean telling whether the last read has been completed for all clients.
- // The variable is reset to false each time buffer is read from the real
- // source.
- bool mLastReadCompleted;
-
- // A global mutex for access to critical sections.
- Mutex mLock;
-
- // Condition variable for waiting on read from source to complete.
- Condition mReadFromSourceCondition;
-
- // Condition variable for waiting on all client's last read to complete.
- Condition mAllReadsCompleteCondition;
-
- // Functions used by Client to implement the MediaSource interface.
-
- // If the real source has not been started yet by any client, starts it.
- status_t start(int clientId, MetaData *params);
-
- // Stops the real source after all clients have called stop().
- status_t stop(int clientId);
-
- // returns the real source's getFormat().
- sp<MetaData> getFormat(int clientId);
-
- // If the client's desired buffer has already been read into
- // mLastReadMediaBuffer, points the buffer to that. Otherwise if it is the
- // master client, reads the buffer from source or else waits for the master
- // client to read the buffer and uses that.
- status_t read(int clientId,
- MediaBuffer **buffer, const MediaSource::ReadOptions *options = NULL);
-
- // Not implemented right now.
- status_t pause(int clientId);
-
- // Function which reads a buffer from the real source into
- // mLastReadMediaBuffer
- void readFromSource_lock(const MediaSource::ReadOptions *options);
-
- // Waits until read from the real source has been completed.
- // _lock means that the function should be called when the thread has already
- // obtained the lock for the mutex mLock.
- void waitForReadFromSource_lock(int32_t clientId);
-
- // Waits until all clients have read the current buffer in
- // mLastReadCompleted.
- void waitForAllClientsLastRead_lock(int32_t clientId);
-
- // Each client calls this after it completes its read(). Once all clients
- // have called this for the current buffer, the function calls
- // mAllReadsCompleteCondition.broadcast() to signal the waiting clients.
- void signalReadComplete_lock(bool readAborted);
-
- // Make these constructors private.
- MediaSourceSplitter();
- MediaSourceSplitter(const MediaSourceSplitter &);
- MediaSourceSplitter &operator=(const MediaSourceSplitter &);
-
- // This class implements the MediaSource interface. Each client stores a
- // reference to the parent MediaSourceSplitter and uses it to complete the
- // various calls.
- class Client : public MediaSource {
- public:
- // Constructor stores reference to the parent MediaSourceSplitter and it
- // client id.
- Client(sp<MediaSourceSplitter> splitter, int32_t clientId);
-
- // MediaSource interface
- virtual status_t start(MetaData *params = NULL);
-
- virtual status_t stop();
-
- virtual sp<MetaData> getFormat();
-
- virtual status_t read(
- MediaBuffer **buffer, const ReadOptions *options = NULL);
-
- virtual status_t pause();
-
- private:
- // Refernce to the parent MediaSourceSplitter
- sp<MediaSourceSplitter> mSplitter;
-
- // Id of this client.
- int32_t mClientId;
- };
-
- friend class Client;
-};
-
-} // namespace android
-
-#endif // MEDIA_SOURCE_SPLITTER_H_
diff --git a/include/media/stagefright/MetaData.h b/include/media/stagefright/MetaData.h
index 4cdee17..00b8679 100644
--- a/include/media/stagefright/MetaData.h
+++ b/include/media/stagefright/MetaData.h
@@ -43,6 +43,7 @@
kKeyStride = 'strd', // int32_t
kKeySliceHeight = 'slht', // int32_t
kKeyChannelCount = '#chn', // int32_t
+ kKeyChannelMask = 'chnm', // int32_t
kKeySampleRate = 'srte', // int32_t (audio sampling rate Hz)
kKeyFrameRate = 'frmR', // int32_t (video frame rate fps)
kKeyBitRate = 'brte', // int32_t (bps)
@@ -69,6 +70,8 @@
kKeyThumbnailTime = 'thbT', // int64_t (usecs)
kKeyTrackID = 'trID',
kKeyIsDRM = 'idrm', // int32_t (bool)
+ kKeyEncoderDelay = 'encd', // int32_t (frames)
+ kKeyEncoderPadding = 'encp', // int32_t (frames)
kKeyAlbum = 'albu', // cstring
kKeyArtist = 'arti', // cstring
diff --git a/include/media/stagefright/MetadataBufferType.h b/include/media/stagefright/MetadataBufferType.h
deleted file mode 100644
index 4eaf8ac..0000000
--- a/include/media/stagefright/MetadataBufferType.h
+++ /dev/null
@@ -1,94 +0,0 @@
-/*
- * Copyright (C) 2011 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef METADATA_BUFFER_TYPE_H
-#define METADATA_BUFFER_TYPE_H
-
-#ifdef __cplusplus
-extern "C" {
-namespace android {
-#endif
-
-/*
- * MetadataBufferType defines the type of the metadata buffers that
- * can be passed to video encoder component for encoding, via Stagefright
- * media recording framework. To see how to work with the metadata buffers
- * in media recording framework, please consult HardwareAPI.h
- *
- * The creator of metadata buffers and video encoder share common knowledge
- * on what is actually being stored in these metadata buffers, and
- * how the information can be used by the video encoder component
- * to locate the actual pixel data as the source input for video
- * encoder, plus whatever other information that is necessary. Stagefright
- * media recording framework does not need to know anything specific about the
- * metadata buffers, except for receving each individual metadata buffer
- * as the source input, making a copy of the metadata buffer, and passing the
- * copy via OpenMAX API to the video encoder component.
- *
- * The creator of the metadata buffers must ensure that the first
- * 4 bytes in every metadata buffer indicates its buffer type,
- * and the rest of the metadata buffer contains the
- * actual metadata information. When a video encoder component receives
- * a metadata buffer, it uses the first 4 bytes in that buffer to find
- * out the type of the metadata buffer, and takes action appropriate
- * to that type of metadata buffers (for instance, locate the actual
- * pixel data input and then encoding the input data to produce a
- * compressed output buffer).
- *
- * The following shows the layout of a metadata buffer,
- * where buffer type is a 4-byte field of MetadataBufferType,
- * and the payload is the metadata information.
- *
- * --------------------------------------------------------------
- * | buffer type | payload |
- * --------------------------------------------------------------
- *
- */
-typedef enum {
-
- /*
- * kMetadataBufferTypeCameraSource is used to indicate that
- * the source of the metadata buffer is the camera component.
- */
- kMetadataBufferTypeCameraSource = 0,
-
- /*
- * kMetadataBufferTypeGrallocSource is used to indicate that
- * the payload of the metadata buffers can be interpreted as
- * a buffer_handle_t.
- * So in this case,the metadata that the encoder receives
- * will have a byte stream that consists of two parts:
- * 1. First, there is an integer indicating that it is a GRAlloc
- * source (kMetadataBufferTypeGrallocSource)
- * 2. This is followed by the buffer_handle_t that is a handle to the
- * GRalloc buffer. The encoder needs to interpret this GRalloc handle
- * and encode the frames.
- * --------------------------------------------------------------
- * | kMetadataBufferTypeGrallocSource | sizeof(buffer_handle_t) |
- * --------------------------------------------------------------
- */
- kMetadataBufferTypeGrallocSource = 1,
-
- // Add more here...
-
-} MetadataBufferType;
-
-#ifdef __cplusplus
-} // namespace android
-}
-#endif
-
-#endif // METADATA_BUFFER_TYPE_H
diff --git a/include/media/stagefright/NativeWindowWrapper.h b/include/media/stagefright/NativeWindowWrapper.h
index f323cbc..97cc0ce 100644
--- a/include/media/stagefright/NativeWindowWrapper.h
+++ b/include/media/stagefright/NativeWindowWrapper.h
@@ -18,40 +18,28 @@
#define NATIVE_WINDOW_WRAPPER_H_
-#include <surfaceflinger/Surface.h>
#include <gui/SurfaceTextureClient.h>
namespace android {
-// Both Surface and SurfaceTextureClient are RefBase that implement the
-// ANativeWindow interface, but at different addresses. ANativeWindow is not
-// a RefBase but acts like one for use with sp<>. This wrapper converts a
-// Surface or SurfaceTextureClient into a single reference-counted object
-// that holds an sp reference to the underlying Surface or SurfaceTextureClient,
-// It provides a method to get the ANativeWindow.
+// SurfaceTextureClient derives from ANativeWindow which derives from multiple
+// base classes, in order to carry it in AMessages, we'll temporarily wrap it
+// into a NativeWindowWrapper.
struct NativeWindowWrapper : RefBase {
NativeWindowWrapper(
- const sp<Surface> &surface) :
- mSurface(surface) { }
-
- NativeWindowWrapper(
const sp<SurfaceTextureClient> &surfaceTextureClient) :
mSurfaceTextureClient(surfaceTextureClient) { }
sp<ANativeWindow> getNativeWindow() const {
- if (mSurface != NULL) {
- return mSurface;
- } else {
- return mSurfaceTextureClient;
- }
+ return mSurfaceTextureClient;
}
- // If needed later we can provide a method to ask what kind of native window
+ sp<SurfaceTextureClient> getSurfaceTextureClient() const {
+ return mSurfaceTextureClient;
+ }
private:
- // At most one of mSurface and mSurfaceTextureClient will be non-NULL
- const sp<Surface> mSurface;
const sp<SurfaceTextureClient> mSurfaceTextureClient;
DISALLOW_EVIL_CONSTRUCTORS(NativeWindowWrapper);
diff --git a/include/media/stagefright/NuMediaExtractor.h b/include/media/stagefright/NuMediaExtractor.h
new file mode 100644
index 0000000..96efdff
--- /dev/null
+++ b/include/media/stagefright/NuMediaExtractor.h
@@ -0,0 +1,80 @@
+/*
+ * Copyright 2012, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef NU_MEDIA_EXTRACTOR_H_
+#define NU_MEDIA_EXTRACTOR_H_
+
+#include <media/stagefright/foundation/ABase.h>
+#include <utils/Errors.h>
+#include <utils/RefBase.h>
+#include <utils/Vector.h>
+
+namespace android {
+
+struct ABuffer;
+struct AMessage;
+struct MediaBuffer;
+struct MediaExtractor;
+struct MediaSource;
+
+struct NuMediaExtractor : public RefBase {
+ NuMediaExtractor();
+
+ status_t setDataSource(const char *path);
+
+ size_t countTracks() const;
+ status_t getTrackFormat(size_t index, sp<AMessage> *format) const;
+
+ status_t selectTrack(size_t index);
+
+ status_t seekTo(int64_t timeUs);
+
+ status_t advance();
+ status_t readSampleData(const sp<ABuffer> &buffer);
+ status_t getSampleTrackIndex(size_t *trackIndex);
+ status_t getSampleTime(int64_t *sampleTimeUs);
+
+protected:
+ virtual ~NuMediaExtractor();
+
+private:
+ enum TrackFlags {
+ kIsVorbis = 1,
+ };
+
+ struct TrackInfo {
+ sp<MediaSource> mSource;
+ size_t mTrackIndex;
+ status_t mFinalResult;
+ MediaBuffer *mSample;
+ int64_t mSampleTimeUs;
+ uint32_t mFlags; // bitmask of "TrackFlags"
+ };
+
+ sp<MediaExtractor> mImpl;
+
+ Vector<TrackInfo> mSelectedTracks;
+
+ ssize_t fetchTrackSamples(int64_t seekTimeUs = -1ll);
+ void releaseTrackSamples();
+
+ DISALLOW_EVIL_CONSTRUCTORS(NuMediaExtractor);
+};
+
+} // namespace android
+
+#endif // NU_MEDIA_EXTRACTOR_H_
+
diff --git a/include/media/stagefright/OMXCodec.h b/include/media/stagefright/OMXCodec.h
index 84f8282..392ea87 100644
--- a/include/media/stagefright/OMXCodec.h
+++ b/include/media/stagefright/OMXCodec.h
@@ -26,6 +26,7 @@
namespace android {
+struct MediaCodecList;
class MemoryDealer;
struct OMXCodecObserver;
struct CodecProfileLevel;
@@ -82,12 +83,35 @@
// from MediaBufferObserver
virtual void signalBufferReturned(MediaBuffer *buffer);
+ enum Quirks {
+ kNeedsFlushBeforeDisable = 1,
+ kWantsNALFragments = 2,
+ kRequiresLoadedToIdleAfterAllocation = 4,
+ kRequiresAllocateBufferOnInputPorts = 8,
+ kRequiresFlushCompleteEmulation = 16,
+ kRequiresAllocateBufferOnOutputPorts = 32,
+ kRequiresFlushBeforeShutdown = 64,
+ kDefersOutputBufferAllocation = 128,
+ kDecoderLiesAboutNumberOfChannels = 256,
+ kInputBufferSizesAreBogus = 512,
+ kSupportsMultipleFramesPerInputBuffer = 1024,
+ kAvoidMemcopyInputRecordingFrames = 2048,
+ kRequiresLargerEncoderOutputBuffer = 4096,
+ kOutputBuffersAreUnreadable = 8192,
+ };
+
// for use by ACodec
static void findMatchingCodecs(
const char *mime,
bool createEncoder, const char *matchComponentName,
uint32_t flags,
- Vector<String8> *matchingCodecs);
+ Vector<String8> *matchingCodecs,
+ Vector<uint32_t> *matchingCodecQuirks = NULL);
+
+ static uint32_t getComponentQuirks(
+ const MediaCodecList *list, size_t index);
+
+ static bool findCodecQuirks(const char *componentName, uint32_t *quirks);
protected:
virtual ~OMXCodec();
@@ -125,23 +149,6 @@
SHUTTING_DOWN,
};
- enum Quirks {
- kNeedsFlushBeforeDisable = 1,
- kWantsNALFragments = 2,
- kRequiresLoadedToIdleAfterAllocation = 4,
- kRequiresAllocateBufferOnInputPorts = 8,
- kRequiresFlushCompleteEmulation = 16,
- kRequiresAllocateBufferOnOutputPorts = 32,
- kRequiresFlushBeforeShutdown = 64,
- kDefersOutputBufferAllocation = 128,
- kDecoderLiesAboutNumberOfChannels = 256,
- kInputBufferSizesAreBogus = 512,
- kSupportsMultipleFramesPerInputBuffer = 1024,
- kAvoidMemcopyInputRecordingFrames = 2048,
- kRequiresLargerEncoderOutputBuffer = 4096,
- kOutputBuffersAreUnreadable = 8192,
- };
-
enum BufferStatus {
OWNED_BY_US,
OWNED_BY_COMPONENT,
@@ -172,6 +179,7 @@
uint32_t mFlags;
bool mIsEncoder;
+ bool mIsVideo;
char *mMIME;
char *mComponentName;
sp<MetaData> mOutputFormat;
@@ -326,15 +334,12 @@
status_t configureCodec(const sp<MetaData> &meta);
- static uint32_t getComponentQuirks(
- const char *componentName, bool isEncoder);
-
void restorePatchedDataPointer(BufferInfo *info);
status_t applyRotation();
status_t waitForBufferFilled_l();
- int64_t retrieveDecodingTimeUs(bool isCodecSpecific);
+ int64_t getDecodingTimeUs();
status_t parseAVCCodecSpecificData(
const void *data, size_t size,
diff --git a/include/media/stagefright/OMXPluginBase.h b/include/media/stagefright/OMXPluginBase.h
deleted file mode 100644
index 2fd8e12..0000000
--- a/include/media/stagefright/OMXPluginBase.h
+++ /dev/null
@@ -1,61 +0,0 @@
-/*
- * Copyright (C) 2009 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef OMX_PLUGIN_BASE_H_
-
-#define OMX_PLUGIN_BASE_H_
-
-#include <sys/types.h>
-
-#include <OMX_Component.h>
-
-#include <utils/String8.h>
-#include <utils/Vector.h>
-
-namespace android {
-
-struct OMXComponentBase;
-
-struct OMXPluginBase {
- OMXPluginBase() {}
- virtual ~OMXPluginBase() {}
-
- virtual OMX_ERRORTYPE makeComponentInstance(
- const char *name,
- const OMX_CALLBACKTYPE *callbacks,
- OMX_PTR appData,
- OMX_COMPONENTTYPE **component) = 0;
-
- virtual OMX_ERRORTYPE destroyComponentInstance(
- OMX_COMPONENTTYPE *component) = 0;
-
- virtual OMX_ERRORTYPE enumerateComponents(
- OMX_STRING name,
- size_t size,
- OMX_U32 index) = 0;
-
- virtual OMX_ERRORTYPE getRolesOfComponent(
- const char *name,
- Vector<String8> *roles) = 0;
-
-private:
- OMXPluginBase(const OMXPluginBase &);
- OMXPluginBase &operator=(const OMXPluginBase &);
-};
-
-} // namespace android
-
-#endif // OMX_PLUGIN_BASE_H_
diff --git a/include/media/stagefright/SurfaceMediaSource.h b/include/media/stagefright/SurfaceMediaSource.h
index d0940bb..54baab6 100644
--- a/include/media/stagefright/SurfaceMediaSource.h
+++ b/include/media/stagefright/SurfaceMediaSource.h
@@ -58,7 +58,7 @@
// For the MediaSource interface for use by StageFrightRecorder:
virtual status_t start(MetaData *params = NULL);
- virtual status_t stop();
+ virtual status_t stop() { return reset(); }
virtual status_t read(
MediaBuffer **buffer, const ReadOptions *options = NULL);
virtual sp<MetaData> getFormat();
@@ -359,6 +359,8 @@
Condition mFrameAvailableCondition;
Condition mFrameCompleteCondition;
+ status_t reset();
+
// Avoid copying and equating and default constructor
DISALLOW_IMPLICIT_CONSTRUCTORS(SurfaceMediaSource);
};
diff --git a/include/media/stagefright/foundation/AMessage.h b/include/media/stagefright/foundation/AMessage.h
index 7ec54aa..e5416e4 100644
--- a/include/media/stagefright/foundation/AMessage.h
+++ b/include/media/stagefright/foundation/AMessage.h
@@ -25,6 +25,7 @@
namespace android {
+struct ABuffer;
struct AString;
struct Parcel;
@@ -50,6 +51,7 @@
void setPointer(const char *name, void *value);
void setString(const char *name, const char *s, ssize_t len = -1);
void setObject(const char *name, const sp<RefBase> &obj);
+ void setBuffer(const char *name, const sp<ABuffer> &buffer);
void setMessage(const char *name, const sp<AMessage> &obj);
void setRect(
@@ -64,6 +66,7 @@
bool findPointer(const char *name, void **value) const;
bool findString(const char *name, AString *value) const;
bool findObject(const char *name, sp<RefBase> *obj) const;
+ bool findBuffer(const char *name, sp<ABuffer> *buffer) const;
bool findMessage(const char *name, sp<AMessage> *obj) const;
bool findRect(
@@ -90,10 +93,6 @@
AString debugString(int32_t indent = 0) const;
-protected:
- virtual ~AMessage();
-
-private:
enum Type {
kTypeInt32,
kTypeInt64,
@@ -105,8 +104,16 @@
kTypeObject,
kTypeMessage,
kTypeRect,
+ kTypeBuffer,
};
+ size_t countEntries() const;
+ const char *getEntryNameAt(size_t index, Type *type) const;
+
+protected:
+ virtual ~AMessage();
+
+private:
uint32_t mWhat;
ALooper::handler_id mTarget;
@@ -131,7 +138,7 @@
};
enum {
- kMaxNumItems = 16
+ kMaxNumItems = 32
};
Item mItems[kMaxNumItems];
size_t mNumItems;
@@ -140,6 +147,9 @@
void freeItem(Item *item);
const Item *findItem(const char *name, Type type) const;
+ void setObjectInternal(
+ const char *name, const sp<RefBase> &obj, Type type);
+
DISALLOW_EVIL_CONSTRUCTORS(AMessage);
};
diff --git a/include/media/stagefright/openmax/OMX_Audio.h b/include/media/stagefright/openmax/OMX_Audio.h
deleted file mode 100644
index 3482841..0000000
--- a/include/media/stagefright/openmax/OMX_Audio.h
+++ /dev/null
@@ -1,1328 +0,0 @@
-/* ------------------------------------------------------------------
- * Copyright (C) 1998-2009 PacketVideo
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either
- * express or implied.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- * -------------------------------------------------------------------
- */
-/*
- * Copyright (c) 2008 The Khronos Group Inc.
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files (the
- * "Software"), to deal in the Software without restriction, including
- * without limitation the rights to use, copy, modify, merge, publish,
- * distribute, sublicense, and/or sell copies of the Software, and to
- * permit persons to whom the Software is furnished to do so, subject
- * to the following conditions:
- * The above copyright notice and this permission notice shall be included
- * in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS
- * OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY
- * CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
- * TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
- * SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- *
- */
-
-/** @file OMX_Audio.h - OpenMax IL version 1.1.2
- * The structures needed by Audio components to exchange
- * parameters and configuration data with the componenmilts.
- */
-
-#ifndef OMX_Audio_h
-#define OMX_Audio_h
-
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-
-/* Each OMX header must include all required header files to allow the
- * header to compile without errors. The includes below are required
- * for this header file to compile successfully
- */
-
-#include <OMX_Core.h>
-
-/** @defgroup midi MIDI
- * @ingroup audio
- */
-
-/** @defgroup effects Audio effects
- * @ingroup audio
- */
-
-/** @defgroup audio OpenMAX IL Audio Domain
- * Structures for OpenMAX IL Audio domain
- * @{
- */
-
-/** Enumeration used to define the possible audio codings.
- * If "OMX_AUDIO_CodingUnused" is selected, the coding selection must
- * be done in a vendor specific way. Since this is for an audio
- * processing element this enum is relevant. However, for another
- * type of component other enums would be in this area.
- */
-typedef enum OMX_AUDIO_CODINGTYPE {
- OMX_AUDIO_CodingUnused = 0, /**< Placeholder value when coding is N/A */
- OMX_AUDIO_CodingAutoDetect, /**< auto detection of audio format */
- OMX_AUDIO_CodingPCM, /**< Any variant of PCM coding */
- OMX_AUDIO_CodingADPCM, /**< Any variant of ADPCM encoded data */
- OMX_AUDIO_CodingAMR, /**< Any variant of AMR encoded data */
- OMX_AUDIO_CodingGSMFR, /**< Any variant of GSM fullrate (i.e. GSM610) */
- OMX_AUDIO_CodingGSMEFR, /**< Any variant of GSM Enhanced Fullrate encoded data*/
- OMX_AUDIO_CodingGSMHR, /**< Any variant of GSM Halfrate encoded data */
- OMX_AUDIO_CodingPDCFR, /**< Any variant of PDC Fullrate encoded data */
- OMX_AUDIO_CodingPDCEFR, /**< Any variant of PDC Enhanced Fullrate encoded data */
- OMX_AUDIO_CodingPDCHR, /**< Any variant of PDC Halfrate encoded data */
- OMX_AUDIO_CodingTDMAFR, /**< Any variant of TDMA Fullrate encoded data (TIA/EIA-136-420) */
- OMX_AUDIO_CodingTDMAEFR, /**< Any variant of TDMA Enhanced Fullrate encoded data (TIA/EIA-136-410) */
- OMX_AUDIO_CodingQCELP8, /**< Any variant of QCELP 8kbps encoded data */
- OMX_AUDIO_CodingQCELP13, /**< Any variant of QCELP 13kbps encoded data */
- OMX_AUDIO_CodingEVRC, /**< Any variant of EVRC encoded data */
- OMX_AUDIO_CodingSMV, /**< Any variant of SMV encoded data */
- OMX_AUDIO_CodingG711, /**< Any variant of G.711 encoded data */
- OMX_AUDIO_CodingG723, /**< Any variant of G.723 dot 1 encoded data */
- OMX_AUDIO_CodingG726, /**< Any variant of G.726 encoded data */
- OMX_AUDIO_CodingG729, /**< Any variant of G.729 encoded data */
- OMX_AUDIO_CodingAAC, /**< Any variant of AAC encoded data */
- OMX_AUDIO_CodingMP3, /**< Any variant of MP3 encoded data */
- OMX_AUDIO_CodingSBC, /**< Any variant of SBC encoded data */
- OMX_AUDIO_CodingVORBIS, /**< Any variant of VORBIS encoded data */
- OMX_AUDIO_CodingWMA, /**< Any variant of WMA encoded data */
- OMX_AUDIO_CodingRA, /**< Any variant of RA encoded data */
- OMX_AUDIO_CodingMIDI, /**< Any variant of MIDI encoded data */
- OMX_AUDIO_CodingKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_AUDIO_CodingVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_AUDIO_CodingMax = 0x7FFFFFFF
-} OMX_AUDIO_CODINGTYPE;
-
-
-/** The PortDefinition structure is used to define all of the parameters
- * necessary for the compliant component to setup an input or an output audio
- * path. If additional information is needed to define the parameters of the
- * port (such as frequency), additional structures must be sent such as the
- * OMX_AUDIO_PARAM_PCMMODETYPE structure to supply the extra parameters for the port.
- */
-typedef struct OMX_AUDIO_PORTDEFINITIONTYPE {
- OMX_STRING cMIMEType; /**< MIME type of data for the port */
- OMX_NATIVE_DEVICETYPE pNativeRender; /** < platform specific reference
- for an output device,
- otherwise this field is 0 */
- OMX_BOOL bFlagErrorConcealment; /**< Turns on error concealment if it is
- supported by the OMX component */
- OMX_AUDIO_CODINGTYPE eEncoding; /**< Type of data expected for this
- port (e.g. PCM, AMR, MP3, etc) */
-} OMX_AUDIO_PORTDEFINITIONTYPE;
-
-
-/** Port format parameter. This structure is used to enumerate
- * the various data input/output format supported by the port.
- */
-typedef struct OMX_AUDIO_PARAM_PORTFORMATTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< Indicates which port to set */
- OMX_U32 nIndex; /**< Indicates the enumeration index for the format from 0x0 to N-1 */
- OMX_AUDIO_CODINGTYPE eEncoding; /**< Type of data expected for this port (e.g. PCM, AMR, MP3, etc) */
-} OMX_AUDIO_PARAM_PORTFORMATTYPE;
-
-
-/** PCM mode type */
-typedef enum OMX_AUDIO_PCMMODETYPE {
- OMX_AUDIO_PCMModeLinear = 0, /**< Linear PCM encoded data */
- OMX_AUDIO_PCMModeALaw, /**< A law PCM encoded data (G.711) */
- OMX_AUDIO_PCMModeMULaw, /**< Mu law PCM encoded data (G.711) */
- OMX_AUDIO_PCMModeKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_AUDIO_PCMModeVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_AUDIO_PCMModeMax = 0x7FFFFFFF
-} OMX_AUDIO_PCMMODETYPE;
-
-
-typedef enum OMX_AUDIO_CHANNELTYPE {
- OMX_AUDIO_ChannelNone = 0x0, /**< Unused or empty */
- OMX_AUDIO_ChannelLF = 0x1, /**< Left front */
- OMX_AUDIO_ChannelRF = 0x2, /**< Right front */
- OMX_AUDIO_ChannelCF = 0x3, /**< Center front */
- OMX_AUDIO_ChannelLS = 0x4, /**< Left surround */
- OMX_AUDIO_ChannelRS = 0x5, /**< Right surround */
- OMX_AUDIO_ChannelLFE = 0x6, /**< Low frequency effects */
- OMX_AUDIO_ChannelCS = 0x7, /**< Back surround */
- OMX_AUDIO_ChannelLR = 0x8, /**< Left rear. */
- OMX_AUDIO_ChannelRR = 0x9, /**< Right rear. */
- OMX_AUDIO_ChannelKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_AUDIO_ChannelVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_AUDIO_ChannelMax = 0x7FFFFFFF
-} OMX_AUDIO_CHANNELTYPE;
-
-#define OMX_AUDIO_MAXCHANNELS 16 /**< maximum number distinct audio channels that a buffer may contain */
-#define OMX_MIN_PCMPAYLOAD_MSEC 5 /**< Minimum audio buffer payload size for uncompressed (PCM) audio */
-
-/** PCM format description */
-typedef struct OMX_AUDIO_PARAM_PCMMODETYPE {
- OMX_U32 nSize; /**< Size of this structure, in Bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_U32 nChannels; /**< Number of channels (e.g. 2 for stereo) */
- OMX_NUMERICALDATATYPE eNumData; /**< indicates PCM data as signed or unsigned */
- OMX_ENDIANTYPE eEndian; /**< indicates PCM data as little or big endian */
- OMX_BOOL bInterleaved; /**< True for normal interleaved data; false for
- non-interleaved data (e.g. block data) */
- OMX_U32 nBitPerSample; /**< Bit per sample */
- OMX_U32 nSamplingRate; /**< Sampling rate of the source data. Use 0 for
- variable or unknown sampling rate. */
- OMX_AUDIO_PCMMODETYPE ePCMMode; /**< PCM mode enumeration */
- OMX_AUDIO_CHANNELTYPE eChannelMapping[OMX_AUDIO_MAXCHANNELS]; /**< Slot i contains channel defined by eChannelMap[i] */
-
-} OMX_AUDIO_PARAM_PCMMODETYPE;
-
-
-/** Audio channel mode. This is used by both AAC and MP3, although the names are more appropriate
- * for the MP3. For example, JointStereo for MP3 is CouplingChannels for AAC.
- */
-typedef enum OMX_AUDIO_CHANNELMODETYPE {
- OMX_AUDIO_ChannelModeStereo = 0, /**< 2 channels, the bitrate allocation between those
- two channels changes accordingly to each channel information */
- OMX_AUDIO_ChannelModeJointStereo, /**< mode that takes advantage of what is common between
- 2 channels for higher compression gain */
- OMX_AUDIO_ChannelModeDual, /**< 2 mono-channels, each channel is encoded with half
- the bitrate of the overall bitrate */
- OMX_AUDIO_ChannelModeMono, /**< Mono channel mode */
- OMX_AUDIO_ChannelModeKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_AUDIO_ChannelModeVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_AUDIO_ChannelModeMax = 0x7FFFFFFF
-} OMX_AUDIO_CHANNELMODETYPE;
-
-
-typedef enum OMX_AUDIO_MP3STREAMFORMATTYPE {
- OMX_AUDIO_MP3StreamFormatMP1Layer3 = 0, /**< MP3 Audio MPEG 1 Layer 3 Stream format */
- OMX_AUDIO_MP3StreamFormatMP2Layer3, /**< MP3 Audio MPEG 2 Layer 3 Stream format */
- OMX_AUDIO_MP3StreamFormatMP2_5Layer3, /**< MP3 Audio MPEG2.5 Layer 3 Stream format */
- OMX_AUDIO_MP3StreamFormatKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_AUDIO_MP3StreamFormatVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_AUDIO_MP3StreamFormatMax = 0x7FFFFFFF
-} OMX_AUDIO_MP3STREAMFORMATTYPE;
-
-/** MP3 params */
-typedef struct OMX_AUDIO_PARAM_MP3TYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_U32 nChannels; /**< Number of channels */
- OMX_U32 nBitRate; /**< Bit rate of the input data. Use 0 for variable
- rate or unknown bit rates */
- OMX_U32 nSampleRate; /**< Sampling rate of the source data. Use 0 for
- variable or unknown sampling rate. */
- OMX_U32 nAudioBandWidth; /**< Audio band width (in Hz) to which an encoder should
- limit the audio signal. Use 0 to let encoder decide */
- OMX_AUDIO_CHANNELMODETYPE eChannelMode; /**< Channel mode enumeration */
- OMX_AUDIO_MP3STREAMFORMATTYPE eFormat; /**< MP3 stream format */
-} OMX_AUDIO_PARAM_MP3TYPE;
-
-
-typedef enum OMX_AUDIO_AACSTREAMFORMATTYPE {
- OMX_AUDIO_AACStreamFormatMP2ADTS = 0, /**< AAC Audio Data Transport Stream 2 format */
- OMX_AUDIO_AACStreamFormatMP4ADTS, /**< AAC Audio Data Transport Stream 4 format */
- OMX_AUDIO_AACStreamFormatMP4LOAS, /**< AAC Low Overhead Audio Stream format */
- OMX_AUDIO_AACStreamFormatMP4LATM, /**< AAC Low overhead Audio Transport Multiplex */
- OMX_AUDIO_AACStreamFormatADIF, /**< AAC Audio Data Interchange Format */
- OMX_AUDIO_AACStreamFormatMP4FF, /**< AAC inside MPEG-4/ISO File Format */
- OMX_AUDIO_AACStreamFormatRAW, /**< AAC Raw Format */
- OMX_AUDIO_AACStreamFormatKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_AUDIO_AACStreamFormatVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_AUDIO_AACStreamFormatMax = 0x7FFFFFFF
-} OMX_AUDIO_AACSTREAMFORMATTYPE;
-
-
-/** AAC mode type. Note that the term profile is used with the MPEG-2
- * standard and the term object type and profile is used with MPEG-4 */
-typedef enum OMX_AUDIO_AACPROFILETYPE{
- OMX_AUDIO_AACObjectNull = 0, /**< Null, not used */
- OMX_AUDIO_AACObjectMain = 1, /**< AAC Main object */
- OMX_AUDIO_AACObjectLC, /**< AAC Low Complexity object (AAC profile) */
- OMX_AUDIO_AACObjectSSR, /**< AAC Scalable Sample Rate object */
- OMX_AUDIO_AACObjectLTP, /**< AAC Long Term Prediction object */
- OMX_AUDIO_AACObjectHE, /**< AAC High Efficiency (object type SBR, HE-AAC profile) */
- OMX_AUDIO_AACObjectScalable, /**< AAC Scalable object */
- OMX_AUDIO_AACObjectERLC = 17, /**< ER AAC Low Complexity object (Error Resilient AAC-LC) */
- OMX_AUDIO_AACObjectLD = 23, /**< AAC Low Delay object (Error Resilient) */
- OMX_AUDIO_AACObjectHE_PS = 29, /**< AAC High Efficiency with Parametric Stereo coding (HE-AAC v2, object type PS) */
- OMX_AUDIO_AACObjectKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_AUDIO_AACObjectVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_AUDIO_AACObjectMax = 0x7FFFFFFF
-} OMX_AUDIO_AACPROFILETYPE;
-
-
-/** AAC tool usage (for nAACtools in OMX_AUDIO_PARAM_AACPROFILETYPE).
- * Required for encoder configuration and optional as decoder info output.
- * For MP3, OMX_AUDIO_CHANNELMODETYPE is sufficient. */
-#define OMX_AUDIO_AACToolNone 0x00000000 /**< no AAC tools allowed (encoder config) or active (decoder info output) */
-#define OMX_AUDIO_AACToolMS 0x00000001 /**< MS: Mid/side joint coding tool allowed or active */
-#define OMX_AUDIO_AACToolIS 0x00000002 /**< IS: Intensity stereo tool allowed or active */
-#define OMX_AUDIO_AACToolTNS 0x00000004 /**< TNS: Temporal Noise Shaping tool allowed or active */
-#define OMX_AUDIO_AACToolPNS 0x00000008 /**< PNS: MPEG-4 Perceptual Noise substitution tool allowed or active */
-#define OMX_AUDIO_AACToolLTP 0x00000010 /**< LTP: MPEG-4 Long Term Prediction tool allowed or active */
-#define OMX_AUDIO_AACToolAll 0x7FFFFFFF /**< all AAC tools allowed or active (*/
-
-/** MPEG-4 AAC error resilience (ER) tool usage (for nAACERtools in OMX_AUDIO_PARAM_AACPROFILETYPE).
- * Required for ER encoder configuration and optional as decoder info output */
-#define OMX_AUDIO_AACERNone 0x00000000 /**< no AAC ER tools allowed/used */
-#define OMX_AUDIO_AACERVCB11 0x00000001 /**< VCB11: Virtual Code Books for AAC section data */
-#define OMX_AUDIO_AACERRVLC 0x00000002 /**< RVLC: Reversible Variable Length Coding */
-#define OMX_AUDIO_AACERHCR 0x00000004 /**< HCR: Huffman Codeword Reordering */
-#define OMX_AUDIO_AACERAll 0x7FFFFFFF /**< all AAC ER tools allowed/used */
-
-
-/** AAC params */
-typedef struct OMX_AUDIO_PARAM_AACPROFILETYPE {
- OMX_U32 nSize; /**< Size of this structure, in Bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< Port that this structure applies to */
- OMX_U32 nChannels; /**< Number of channels */
- OMX_U32 nSampleRate; /**< Sampling rate of the source data. Use 0 for
- variable or unknown sampling rate. */
- OMX_U32 nBitRate; /**< Bit rate of the input data. Use 0 for variable
- rate or unknown bit rates */
- OMX_U32 nAudioBandWidth; /**< Audio band width (in Hz) to which an encoder should
- limit the audio signal. Use 0 to let encoder decide */
- OMX_U32 nFrameLength; /**< Frame length (in audio samples per channel) of the codec.
- Can be 1024 or 960 (AAC-LC), 2048 (HE-AAC), 480 or 512 (AAC-LD).
- Use 0 to let encoder decide */
- OMX_U32 nAACtools; /**< AAC tool usage */
- OMX_U32 nAACERtools; /**< MPEG-4 AAC error resilience tool usage */
- OMX_AUDIO_AACPROFILETYPE eAACProfile; /**< AAC profile enumeration */
- OMX_AUDIO_AACSTREAMFORMATTYPE eAACStreamFormat; /**< AAC stream format enumeration */
- OMX_AUDIO_CHANNELMODETYPE eChannelMode; /**< Channel mode enumeration */
-} OMX_AUDIO_PARAM_AACPROFILETYPE;
-
-
-/** VORBIS params */
-typedef struct OMX_AUDIO_PARAM_VORBISTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_U32 nChannels; /**< Number of channels */
- OMX_U32 nBitRate; /**< Bit rate of the encoded data data. Use 0 for variable
- rate or unknown bit rates. Encoding is set to the
- bitrate closest to specified value (in bps) */
- OMX_U32 nMinBitRate; /**< Sets minimum bitrate (in bps). */
- OMX_U32 nMaxBitRate; /**< Sets maximum bitrate (in bps). */
-
- OMX_U32 nSampleRate; /**< Sampling rate of the source data. Use 0 for
- variable or unknown sampling rate. */
- OMX_U32 nAudioBandWidth; /**< Audio band width (in Hz) to which an encoder should
- limit the audio signal. Use 0 to let encoder decide */
- OMX_S32 nQuality; /**< Sets encoding quality to n, between -1 (low) and 10 (high).
- In the default mode of operation, teh quality level is 3.
- Normal quality range is 0 - 10. */
- OMX_BOOL bManaged; /**< Set bitrate management mode. This turns off the
- normal VBR encoding, but allows hard or soft bitrate
- constraints to be enforced by the encoder. This mode can
- be slower, and may also be lower quality. It is
- primarily useful for streaming. */
- OMX_BOOL bDownmix; /**< Downmix input from stereo to mono (has no effect on
- non-stereo streams). Useful for lower-bitrate encoding. */
-} OMX_AUDIO_PARAM_VORBISTYPE;
-
-
-/** WMA Version */
-typedef enum OMX_AUDIO_WMAFORMATTYPE {
- OMX_AUDIO_WMAFormatUnused = 0, /**< format unused or unknown */
- OMX_AUDIO_WMAFormat7, /**< Windows Media Audio format 7 */
- OMX_AUDIO_WMAFormat8, /**< Windows Media Audio format 8 */
- OMX_AUDIO_WMAFormat9, /**< Windows Media Audio format 9 */
- OMX_AUDIO_WMAFormatKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_AUDIO_WMAFormatVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_AUDIO_WMAFormatMax = 0x7FFFFFFF
-} OMX_AUDIO_WMAFORMATTYPE;
-
-
-/** WMA Profile */
-typedef enum OMX_AUDIO_WMAPROFILETYPE {
- OMX_AUDIO_WMAProfileUnused = 0, /**< profile unused or unknown */
- OMX_AUDIO_WMAProfileL1, /**< Windows Media audio version 9 profile L1 */
- OMX_AUDIO_WMAProfileL2, /**< Windows Media audio version 9 profile L2 */
- OMX_AUDIO_WMAProfileL3, /**< Windows Media audio version 9 profile L3 */
- OMX_AUDIO_WMAProfileKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_AUDIO_WMAProfileVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_AUDIO_WMAProfileMax = 0x7FFFFFFF
-} OMX_AUDIO_WMAPROFILETYPE;
-
-
-/** WMA params */
-typedef struct OMX_AUDIO_PARAM_WMATYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_U16 nChannels; /**< Number of channels */
- OMX_U32 nBitRate; /**< Bit rate of the input data. Use 0 for variable
- rate or unknown bit rates */
- OMX_AUDIO_WMAFORMATTYPE eFormat; /**< Version of WMA stream / data */
- OMX_AUDIO_WMAPROFILETYPE eProfile; /**< Profile of WMA stream / data */
- OMX_U32 nSamplingRate; /**< Sampling rate of the source data */
- OMX_U16 nBlockAlign; /**< is the block alignment, or block size, in bytes of the audio codec */
- OMX_U16 nEncodeOptions; /**< WMA Type-specific data */
- OMX_U32 nSuperBlockAlign; /**< WMA Type-specific data */
-} OMX_AUDIO_PARAM_WMATYPE;
-
-/**
- * RealAudio format
- */
-typedef enum OMX_AUDIO_RAFORMATTYPE {
- OMX_AUDIO_RAFormatUnused = 0, /**< Format unused or unknown */
- OMX_AUDIO_RA8, /**< RealAudio 8 codec */
- OMX_AUDIO_RA9, /**< RealAudio 9 codec */
- OMX_AUDIO_RA10_AAC, /**< MPEG-4 AAC codec for bitrates of more than 128kbps */
- OMX_AUDIO_RA10_CODEC, /**< RealAudio codec for bitrates less than 128 kbps */
- OMX_AUDIO_RA10_LOSSLESS, /**< RealAudio Lossless */
- OMX_AUDIO_RA10_MULTICHANNEL, /**< RealAudio Multichannel */
- OMX_AUDIO_RA10_VOICE, /**< RealAudio Voice for bitrates below 15 kbps */
- OMX_AUDIO_RAFormatKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_AUDIO_RAFormatVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_VIDEO_RAFormatMax = 0x7FFFFFFF
-} OMX_AUDIO_RAFORMATTYPE;
-
-/** RA (Real Audio) params */
-typedef struct OMX_AUDIO_PARAM_RATYPE {
- OMX_U32 nSize; /**< Size of this structure, in Bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< Port that this structure applies to */
- OMX_U32 nChannels; /**< Number of channels */
- OMX_U32 nSamplingRate; /**< is the sampling rate of the source data */
- OMX_U32 nBitsPerFrame; /**< is the value for bits per frame */
- OMX_U32 nSamplePerFrame; /**< is the value for samples per frame */
- OMX_U32 nCouplingQuantBits; /**< is the number of coupling quantization bits in the stream */
- OMX_U32 nCouplingStartRegion; /**< is the coupling start region in the stream */
- OMX_U32 nNumRegions; /**< is the number of regions value */
- OMX_AUDIO_RAFORMATTYPE eFormat; /**< is the RealAudio audio format */
-} OMX_AUDIO_PARAM_RATYPE;
-
-
-/** SBC Allocation Method Type */
-typedef enum OMX_AUDIO_SBCALLOCMETHODTYPE {
- OMX_AUDIO_SBCAllocMethodLoudness, /**< Loudness allocation method */
- OMX_AUDIO_SBCAllocMethodSNR, /**< SNR allocation method */
- OMX_AUDIO_SBCAllocMethodKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_AUDIO_SBCAllocMethodVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_AUDIO_SBCAllocMethodMax = 0x7FFFFFFF
-} OMX_AUDIO_SBCALLOCMETHODTYPE;
-
-
-/** SBC params */
-typedef struct OMX_AUDIO_PARAM_SBCTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_U32 nChannels; /**< Number of channels */
- OMX_U32 nBitRate; /**< Bit rate of the input data. Use 0 for variable
- rate or unknown bit rates */
- OMX_U32 nSampleRate; /**< Sampling rate of the source data. Use 0 for
- variable or unknown sampling rate. */
- OMX_U32 nBlocks; /**< Number of blocks */
- OMX_U32 nSubbands; /**< Number of subbands */
- OMX_U32 nBitPool; /**< Bitpool value */
- OMX_BOOL bEnableBitrate; /**< Use bitrate value instead of bitpool */
- OMX_AUDIO_CHANNELMODETYPE eChannelMode; /**< Channel mode enumeration */
- OMX_AUDIO_SBCALLOCMETHODTYPE eSBCAllocType; /**< SBC Allocation method type */
-} OMX_AUDIO_PARAM_SBCTYPE;
-
-
-/** ADPCM stream format parameters */
-typedef struct OMX_AUDIO_PARAM_ADPCMTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_U32 nChannels; /**< Number of channels in the data stream (not
- necessarily the same as the number of channels
- to be rendered. */
- OMX_U32 nBitsPerSample; /**< Number of bits in each sample */
- OMX_U32 nSampleRate; /**< Sampling rate of the source data. Use 0 for
- variable or unknown sampling rate. */
-} OMX_AUDIO_PARAM_ADPCMTYPE;
-
-
-/** G723 rate */
-typedef enum OMX_AUDIO_G723RATE {
- OMX_AUDIO_G723ModeUnused = 0, /**< AMRNB Mode unused / unknown */
- OMX_AUDIO_G723ModeLow, /**< 5300 bps */
- OMX_AUDIO_G723ModeHigh, /**< 6300 bps */
- OMX_AUDIO_G723ModeKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_AUDIO_G723ModeVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_AUDIO_G723ModeMax = 0x7FFFFFFF
-} OMX_AUDIO_G723RATE;
-
-
-/** G723 - Sample rate must be 8 KHz */
-typedef struct OMX_AUDIO_PARAM_G723TYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_U32 nChannels; /**< Number of channels in the data stream (not
- necessarily the same as the number of channels
- to be rendered. */
- OMX_BOOL bDTX; /**< Enable Discontinuous Transmisssion */
- OMX_AUDIO_G723RATE eBitRate; /**< todo: Should this be moved to a config? */
- OMX_BOOL bHiPassFilter; /**< Enable High Pass Filter */
- OMX_BOOL bPostFilter; /**< Enable Post Filter */
-} OMX_AUDIO_PARAM_G723TYPE;
-
-
-/** ITU G726 (ADPCM) rate */
-typedef enum OMX_AUDIO_G726MODE {
- OMX_AUDIO_G726ModeUnused = 0, /**< G726 Mode unused / unknown */
- OMX_AUDIO_G726Mode16, /**< 16 kbps */
- OMX_AUDIO_G726Mode24, /**< 24 kbps */
- OMX_AUDIO_G726Mode32, /**< 32 kbps, most common rate, also G721 */
- OMX_AUDIO_G726Mode40, /**< 40 kbps */
- OMX_AUDIO_G726ModeKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_AUDIO_G726ModeVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_AUDIO_G726ModeMax = 0x7FFFFFFF
-} OMX_AUDIO_G726MODE;
-
-
-/** G.726 stream format parameters - must be at 8KHz */
-typedef struct OMX_AUDIO_PARAM_G726TYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_U32 nChannels; /**< Number of channels in the data stream (not
- necessarily the same as the number of channels
- to be rendered. */
- OMX_AUDIO_G726MODE eG726Mode;
-} OMX_AUDIO_PARAM_G726TYPE;
-
-
-/** G729 coder type */
-typedef enum OMX_AUDIO_G729TYPE {
- OMX_AUDIO_G729 = 0, /**< ITU G.729 encoded data */
- OMX_AUDIO_G729A, /**< ITU G.729 annex A encoded data */
- OMX_AUDIO_G729B, /**< ITU G.729 with annex B encoded data */
- OMX_AUDIO_G729AB, /**< ITU G.729 annexes A and B encoded data */
- OMX_AUDIO_G729KhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_AUDIO_G729VendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_AUDIO_G729Max = 0x7FFFFFFF
-} OMX_AUDIO_G729TYPE;
-
-
-/** G729 stream format parameters - fixed 6KHz sample rate */
-typedef struct OMX_AUDIO_PARAM_G729TYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_U32 nChannels; /**< Number of channels in the data stream (not
- necessarily the same as the number of channels
- to be rendered. */
- OMX_BOOL bDTX; /**< Enable Discontinuous Transmisssion */
- OMX_AUDIO_G729TYPE eBitType;
-} OMX_AUDIO_PARAM_G729TYPE;
-
-
-/** AMR Frame format */
-typedef enum OMX_AUDIO_AMRFRAMEFORMATTYPE {
- OMX_AUDIO_AMRFrameFormatConformance = 0, /**< Frame Format is AMR Conformance
- (Standard) Format */
- OMX_AUDIO_AMRFrameFormatIF1, /**< Frame Format is AMR Interface
- Format 1 */
- OMX_AUDIO_AMRFrameFormatIF2, /**< Frame Format is AMR Interface
- Format 2*/
- OMX_AUDIO_AMRFrameFormatFSF, /**< Frame Format is AMR File Storage
- Format */
- OMX_AUDIO_AMRFrameFormatRTPPayload, /**< Frame Format is AMR Real-Time
- Transport Protocol Payload Format */
- OMX_AUDIO_AMRFrameFormatITU, /**< Frame Format is ITU Format (added at Motorola request) */
- OMX_AUDIO_AMRFrameFormatKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_AUDIO_AMRFrameFormatVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_AUDIO_AMRFrameFormatMax = 0x7FFFFFFF
-} OMX_AUDIO_AMRFRAMEFORMATTYPE;
-
-
-/** AMR band mode */
-typedef enum OMX_AUDIO_AMRBANDMODETYPE {
- OMX_AUDIO_AMRBandModeUnused = 0, /**< AMRNB Mode unused / unknown */
- OMX_AUDIO_AMRBandModeNB0, /**< AMRNB Mode 0 = 4750 bps */
- OMX_AUDIO_AMRBandModeNB1, /**< AMRNB Mode 1 = 5150 bps */
- OMX_AUDIO_AMRBandModeNB2, /**< AMRNB Mode 2 = 5900 bps */
- OMX_AUDIO_AMRBandModeNB3, /**< AMRNB Mode 3 = 6700 bps */
- OMX_AUDIO_AMRBandModeNB4, /**< AMRNB Mode 4 = 7400 bps */
- OMX_AUDIO_AMRBandModeNB5, /**< AMRNB Mode 5 = 7950 bps */
- OMX_AUDIO_AMRBandModeNB6, /**< AMRNB Mode 6 = 10200 bps */
- OMX_AUDIO_AMRBandModeNB7, /**< AMRNB Mode 7 = 12200 bps */
- OMX_AUDIO_AMRBandModeWB0, /**< AMRWB Mode 0 = 6600 bps */
- OMX_AUDIO_AMRBandModeWB1, /**< AMRWB Mode 1 = 8850 bps */
- OMX_AUDIO_AMRBandModeWB2, /**< AMRWB Mode 2 = 12650 bps */
- OMX_AUDIO_AMRBandModeWB3, /**< AMRWB Mode 3 = 14250 bps */
- OMX_AUDIO_AMRBandModeWB4, /**< AMRWB Mode 4 = 15850 bps */
- OMX_AUDIO_AMRBandModeWB5, /**< AMRWB Mode 5 = 18250 bps */
- OMX_AUDIO_AMRBandModeWB6, /**< AMRWB Mode 6 = 19850 bps */
- OMX_AUDIO_AMRBandModeWB7, /**< AMRWB Mode 7 = 23050 bps */
- OMX_AUDIO_AMRBandModeWB8, /**< AMRWB Mode 8 = 23850 bps */
- OMX_AUDIO_AMRBandModeKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_AUDIO_AMRBandModeVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_AUDIO_AMRBandModeMax = 0x7FFFFFFF
-} OMX_AUDIO_AMRBANDMODETYPE;
-
-
-/** AMR Discontinuous Transmission mode */
-typedef enum OMX_AUDIO_AMRDTXMODETYPE {
- OMX_AUDIO_AMRDTXModeOff = 0, /**< AMR Discontinuous Transmission Mode is disabled */
- OMX_AUDIO_AMRDTXModeOnVAD1, /**< AMR Discontinuous Transmission Mode using
- Voice Activity Detector 1 (VAD1) is enabled */
- OMX_AUDIO_AMRDTXModeOnVAD2, /**< AMR Discontinuous Transmission Mode using
- Voice Activity Detector 2 (VAD2) is enabled */
- OMX_AUDIO_AMRDTXModeOnAuto, /**< The codec will automatically select between
- Off, VAD1 or VAD2 modes */
-
- OMX_AUDIO_AMRDTXasEFR, /**< DTX as EFR instead of AMR standard (3GPP 26.101, frame type =8,9,10) */
-
- OMX_AUDIO_AMRDTXModeKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_AUDIO_AMRDTXModeVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_AUDIO_AMRDTXModeMax = 0x7FFFFFFF
-} OMX_AUDIO_AMRDTXMODETYPE;
-
-
-/** AMR params */
-typedef struct OMX_AUDIO_PARAM_AMRTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_U32 nChannels; /**< Number of channels */
- OMX_U32 nBitRate; /**< Bit rate read only field */
- OMX_AUDIO_AMRBANDMODETYPE eAMRBandMode; /**< AMR Band Mode enumeration */
- OMX_AUDIO_AMRDTXMODETYPE eAMRDTXMode; /**< AMR DTX Mode enumeration */
- OMX_AUDIO_AMRFRAMEFORMATTYPE eAMRFrameFormat; /**< AMR frame format enumeration */
-} OMX_AUDIO_PARAM_AMRTYPE;
-
-
-/** GSM_FR (ETSI 06.10, 3GPP 46.010) stream format parameters */
-typedef struct OMX_AUDIO_PARAM_GSMFRTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_BOOL bDTX; /**< Enable Discontinuous Transmisssion */
- OMX_BOOL bHiPassFilter; /**< Enable High Pass Filter */
-} OMX_AUDIO_PARAM_GSMFRTYPE;
-
-
-/** GSM-HR (ETSI 06.20, 3GPP 46.020) stream format parameters */
-typedef struct OMX_AUDIO_PARAM_GSMHRTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_BOOL bDTX; /**< Enable Discontinuous Transmisssion */
- OMX_BOOL bHiPassFilter; /**< Enable High Pass Filter */
-} OMX_AUDIO_PARAM_GSMHRTYPE;
-
-
-/** GSM-EFR (ETSI 06.60, 3GPP 46.060) stream format parameters */
-typedef struct OMX_AUDIO_PARAM_GSMEFRTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_BOOL bDTX; /**< Enable Discontinuous Transmisssion */
- OMX_BOOL bHiPassFilter; /**< Enable High Pass Filter */
-} OMX_AUDIO_PARAM_GSMEFRTYPE;
-
-
-/** TDMA FR (TIA/EIA-136-420, VSELP 7.95kbps coder) stream format parameters */
-typedef struct OMX_AUDIO_PARAM_TDMAFRTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_U32 nChannels; /**< Number of channels in the data stream (not
- necessarily the same as the number of channels
- to be rendered. */
- OMX_BOOL bDTX; /**< Enable Discontinuous Transmisssion */
- OMX_BOOL bHiPassFilter; /**< Enable High Pass Filter */
-} OMX_AUDIO_PARAM_TDMAFRTYPE;
-
-
-/** TDMA EFR (TIA/EIA-136-410, ACELP 7.4kbps coder) stream format parameters */
-typedef struct OMX_AUDIO_PARAM_TDMAEFRTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_U32 nChannels; /**< Number of channels in the data stream (not
- necessarily the same as the number of channels
- to be rendered. */
- OMX_BOOL bDTX; /**< Enable Discontinuous Transmisssion */
- OMX_BOOL bHiPassFilter; /**< Enable High Pass Filter */
-} OMX_AUDIO_PARAM_TDMAEFRTYPE;
-
-
-/** PDC FR ( RCR-27, VSELP 6.7kbps coder) stream format parameters */
-typedef struct OMX_AUDIO_PARAM_PDCFRTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_U32 nChannels; /**< Number of channels in the data stream (not
- necessarily the same as the number of channels
- to be rendered. */
- OMX_BOOL bDTX; /**< Enable Discontinuous Transmisssion */
- OMX_BOOL bHiPassFilter; /**< Enable High Pass Filter */
-} OMX_AUDIO_PARAM_PDCFRTYPE;
-
-
-/** PDC EFR ( RCR-27, ACELP 6.7kbps coder) stream format parameters */
-typedef struct OMX_AUDIO_PARAM_PDCEFRTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_U32 nChannels; /**< Number of channels in the data stream (not
- necessarily the same as the number of channels
- to be rendered. */
- OMX_BOOL bDTX; /**< Enable Discontinuous Transmisssion */
- OMX_BOOL bHiPassFilter; /**< Enable High Pass Filter */
-} OMX_AUDIO_PARAM_PDCEFRTYPE;
-
-/** PDC HR ( RCR-27, PSI-CELP 3.45kbps coder) stream format parameters */
-typedef struct OMX_AUDIO_PARAM_PDCHRTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_U32 nChannels; /**< Number of channels in the data stream (not
- necessarily the same as the number of channels
- to be rendered. */
- OMX_BOOL bDTX; /**< Enable Discontinuous Transmisssion */
- OMX_BOOL bHiPassFilter; /**< Enable High Pass Filter */
-} OMX_AUDIO_PARAM_PDCHRTYPE;
-
-
-/** CDMA Rate types */
-typedef enum OMX_AUDIO_CDMARATETYPE {
- OMX_AUDIO_CDMARateBlank = 0, /**< CDMA encoded frame is blank */
- OMX_AUDIO_CDMARateFull, /**< CDMA encoded frame in full rate */
- OMX_AUDIO_CDMARateHalf, /**< CDMA encoded frame in half rate */
- OMX_AUDIO_CDMARateQuarter, /**< CDMA encoded frame in quarter rate */
- OMX_AUDIO_CDMARateEighth, /**< CDMA encoded frame in eighth rate (DTX)*/
- OMX_AUDIO_CDMARateErasure, /**< CDMA erasure frame */
- OMX_AUDIO_CDMARateKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_AUDIO_CDMARateVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_AUDIO_CDMARateMax = 0x7FFFFFFF
-} OMX_AUDIO_CDMARATETYPE;
-
-
-/** QCELP8 (TIA/EIA-96, up to 8kbps coder) stream format parameters */
-typedef struct OMX_AUDIO_PARAM_QCELP8TYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_U32 nChannels; /**< Number of channels in the data stream (not
- necessarily the same as the number of channels
- to be rendered. */
- OMX_U32 nBitRate; /**< Bit rate of the input data. Use 0 for variable
- rate or unknown bit rates */
- OMX_AUDIO_CDMARATETYPE eCDMARate; /**< Frame rate */
- OMX_U32 nMinBitRate; /**< minmal rate for the encoder = 1,2,3,4, default = 1 */
- OMX_U32 nMaxBitRate; /**< maximal rate for the encoder = 1,2,3,4, default = 4 */
-} OMX_AUDIO_PARAM_QCELP8TYPE;
-
-
-/** QCELP13 ( CDMA, EIA/TIA-733, 13.3kbps coder) stream format parameters */
-typedef struct OMX_AUDIO_PARAM_QCELP13TYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_U32 nChannels; /**< Number of channels in the data stream (not
- necessarily the same as the number of channels
- to be rendered. */
- OMX_AUDIO_CDMARATETYPE eCDMARate; /**< Frame rate */
- OMX_U32 nMinBitRate; /**< minmal rate for the encoder = 1,2,3,4, default = 1 */
- OMX_U32 nMaxBitRate; /**< maximal rate for the encoder = 1,2,3,4, default = 4 */
-} OMX_AUDIO_PARAM_QCELP13TYPE;
-
-
-/** EVRC ( CDMA, EIA/TIA-127, RCELP up to 8.55kbps coder) stream format parameters */
-typedef struct OMX_AUDIO_PARAM_EVRCTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_U32 nChannels; /**< Number of channels in the data stream (not
- necessarily the same as the number of channels
- to be rendered. */
- OMX_AUDIO_CDMARATETYPE eCDMARate; /**< actual Frame rate */
- OMX_BOOL bRATE_REDUCon; /**< RATE_REDUCtion is requested for this frame */
- OMX_U32 nMinBitRate; /**< minmal rate for the encoder = 1,2,3,4, default = 1 */
- OMX_U32 nMaxBitRate; /**< maximal rate for the encoder = 1,2,3,4, default = 4 */
- OMX_BOOL bHiPassFilter; /**< Enable encoder's High Pass Filter */
- OMX_BOOL bNoiseSuppressor; /**< Enable encoder's noise suppressor pre-processing */
- OMX_BOOL bPostFilter; /**< Enable decoder's post Filter */
-} OMX_AUDIO_PARAM_EVRCTYPE;
-
-
-/** SMV ( up to 8.55kbps coder) stream format parameters */
-typedef struct OMX_AUDIO_PARAM_SMVTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_U32 nChannels; /**< Number of channels in the data stream (not
- necessarily the same as the number of channels
- to be rendered. */
- OMX_AUDIO_CDMARATETYPE eCDMARate; /**< Frame rate */
- OMX_BOOL bRATE_REDUCon; /**< RATE_REDUCtion is requested for this frame */
- OMX_U32 nMinBitRate; /**< minmal rate for the encoder = 1,2,3,4, default = 1 ??*/
- OMX_U32 nMaxBitRate; /**< maximal rate for the encoder = 1,2,3,4, default = 4 ??*/
- OMX_BOOL bHiPassFilter; /**< Enable encoder's High Pass Filter ??*/
- OMX_BOOL bNoiseSuppressor; /**< Enable encoder's noise suppressor pre-processing */
- OMX_BOOL bPostFilter; /**< Enable decoder's post Filter ??*/
-} OMX_AUDIO_PARAM_SMVTYPE;
-
-
-/** MIDI Format
- * @ingroup midi
- */
-typedef enum OMX_AUDIO_MIDIFORMATTYPE
-{
- OMX_AUDIO_MIDIFormatUnknown = 0, /**< MIDI Format unknown or don't care */
- OMX_AUDIO_MIDIFormatSMF0, /**< Standard MIDI File Type 0 */
- OMX_AUDIO_MIDIFormatSMF1, /**< Standard MIDI File Type 1 */
- OMX_AUDIO_MIDIFormatSMF2, /**< Standard MIDI File Type 2 */
- OMX_AUDIO_MIDIFormatSPMIDI, /**< SP-MIDI */
- OMX_AUDIO_MIDIFormatXMF0, /**< eXtensible Music Format type 0 */
- OMX_AUDIO_MIDIFormatXMF1, /**< eXtensible Music Format type 1 */
- OMX_AUDIO_MIDIFormatMobileXMF, /**< Mobile XMF (eXtensible Music Format type 2) */
- OMX_AUDIO_MIDIFormatKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_AUDIO_MIDIFormatVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_AUDIO_MIDIFormatMax = 0x7FFFFFFF
-} OMX_AUDIO_MIDIFORMATTYPE;
-
-
-/** MIDI params
- * @ingroup midi
- */
-typedef struct OMX_AUDIO_PARAM_MIDITYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_U32 nFileSize; /**< size of the MIDI file in bytes, where the entire
- MIDI file passed in, otherwise if 0x0, the MIDI data
- is merged and streamed (instead of passed as an
- entire MIDI file) */
- OMX_BU32 sMaxPolyphony; /**< Specifies the maximum simultaneous polyphonic
- voices. A value of zero indicates that the default
- polyphony of the device is used */
- OMX_BOOL bLoadDefaultSound; /**< Whether to load default sound
- bank at initialization */
- OMX_AUDIO_MIDIFORMATTYPE eMidiFormat; /**< Version of the MIDI file */
-} OMX_AUDIO_PARAM_MIDITYPE;
-
-
-/** Type of the MIDI sound bank
- * @ingroup midi
- */
-typedef enum OMX_AUDIO_MIDISOUNDBANKTYPE {
- OMX_AUDIO_MIDISoundBankUnused = 0, /**< unused/unknown soundbank type */
- OMX_AUDIO_MIDISoundBankDLS1, /**< DLS version 1 */
- OMX_AUDIO_MIDISoundBankDLS2, /**< DLS version 2 */
- OMX_AUDIO_MIDISoundBankMobileDLSBase, /**< Mobile DLS, using the base functionality */
- OMX_AUDIO_MIDISoundBankMobileDLSPlusOptions, /**< Mobile DLS, using the specification-defined optional feature set */
- OMX_AUDIO_MIDISoundBankKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_AUDIO_MIDISoundBankVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_AUDIO_MIDISoundBankMax = 0x7FFFFFFF
-} OMX_AUDIO_MIDISOUNDBANKTYPE;
-
-
-/** Bank Layout describes how bank MSB & LSB are used in the DLS instrument definitions sound bank
- * @ingroup midi
- */
-typedef enum OMX_AUDIO_MIDISOUNDBANKLAYOUTTYPE {
- OMX_AUDIO_MIDISoundBankLayoutUnused = 0, /**< unused/unknown soundbank type */
- OMX_AUDIO_MIDISoundBankLayoutGM, /**< GS layout (based on bank MSB 0x00) */
- OMX_AUDIO_MIDISoundBankLayoutGM2, /**< General MIDI 2 layout (using MSB 0x78/0x79, LSB 0x00) */
- OMX_AUDIO_MIDISoundBankLayoutUser, /**< Does not conform to any bank numbering standards */
- OMX_AUDIO_MIDISoundBankLayoutKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_AUDIO_MIDISoundBankLayoutVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_AUDIO_MIDISoundBankLayoutMax = 0x7FFFFFFF
-} OMX_AUDIO_MIDISOUNDBANKLAYOUTTYPE;
-
-
-/** MIDI params to load/unload user soundbank
- * @ingroup midi
- */
-typedef struct OMX_AUDIO_PARAM_MIDILOADUSERSOUNDTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_U32 nDLSIndex; /**< DLS file index to be loaded */
- OMX_U32 nDLSSize; /**< Size in bytes */
- OMX_PTR pDLSData; /**< Pointer to DLS file data */
- OMX_AUDIO_MIDISOUNDBANKTYPE eMidiSoundBank; /**< Midi sound bank type enumeration */
- OMX_AUDIO_MIDISOUNDBANKLAYOUTTYPE eMidiSoundBankLayout; /**< Midi sound bank layout enumeration */
-} OMX_AUDIO_PARAM_MIDILOADUSERSOUNDTYPE;
-
-
-/** Structure for Live MIDI events and MIP messages.
- * (MIP = Maximum Instantaneous Polyphony; part of the SP-MIDI standard.)
- * @ingroup midi
- */
-typedef struct OMX_AUDIO_CONFIG_MIDIIMMEDIATEEVENTTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< Port that this structure applies to */
- OMX_U32 nMidiEventSize; /**< Size of immediate MIDI events or MIP message in bytes */
- OMX_U8 nMidiEvents[1]; /**< MIDI event array to be rendered immediately, or an
- array for the MIP message buffer, where the size is
- indicated by nMidiEventSize */
-} OMX_AUDIO_CONFIG_MIDIIMMEDIATEEVENTTYPE;
-
-
-/** MIDI sound bank/ program pair in a given channel
- * @ingroup midi
- */
-typedef struct OMX_AUDIO_CONFIG_MIDISOUNDBANKPROGRAMTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< Port that this structure applies to */
- OMX_U32 nChannel; /**< Valid channel values range from 1 to 16 */
- OMX_U16 nIDProgram; /**< Valid program ID range is 1 to 128 */
- OMX_U16 nIDSoundBank; /**< Sound bank ID */
- OMX_U32 nUserSoundBankIndex;/**< User soundbank index, easier to access soundbanks
- by index if multiple banks are present */
-} OMX_AUDIO_CONFIG_MIDISOUNDBANKPROGRAMTYPE;
-
-
-/** MIDI control
- * @ingroup midi
- */
-typedef struct OMX_AUDIO_CONFIG_MIDICONTROLTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_BS32 sPitchTransposition; /**< Pitch transposition in semitones, stored as Q22.10
- format based on JAVA MMAPI (JSR-135) requirement */
- OMX_BU32 sPlayBackRate; /**< Relative playback rate, stored as Q14.17 fixed-point
- number based on JSR-135 requirement */
- OMX_BU32 sTempo ; /**< Tempo in beats per minute (BPM), stored as Q22.10
- fixed-point number based on JSR-135 requirement */
- OMX_U32 nMaxPolyphony; /**< Specifies the maximum simultaneous polyphonic
- voices. A value of zero indicates that the default
- polyphony of the device is used */
- OMX_U32 nNumRepeat; /**< Number of times to repeat playback */
- OMX_U32 nStopTime; /**< Time in milliseconds to indicate when playback
- will stop automatically. Set to zero if not used */
- OMX_U16 nChannelMuteMask; /**< 16 bit mask for channel mute status */
- OMX_U16 nChannelSoloMask; /**< 16 bit mask for channel solo status */
- OMX_U32 nTrack0031MuteMask; /**< 32 bit mask for track mute status. Note: This is for tracks 0-31 */
- OMX_U32 nTrack3263MuteMask; /**< 32 bit mask for track mute status. Note: This is for tracks 32-63 */
- OMX_U32 nTrack0031SoloMask; /**< 32 bit mask for track solo status. Note: This is for tracks 0-31 */
- OMX_U32 nTrack3263SoloMask; /**< 32 bit mask for track solo status. Note: This is for tracks 32-63 */
-
-} OMX_AUDIO_CONFIG_MIDICONTROLTYPE;
-
-
-/** MIDI Playback States
- * @ingroup midi
- */
-typedef enum OMX_AUDIO_MIDIPLAYBACKSTATETYPE {
- OMX_AUDIO_MIDIPlayBackStateUnknown = 0, /**< Unknown state or state does not map to
- other defined states */
- OMX_AUDIO_MIDIPlayBackStateClosedEngaged, /**< No MIDI resource is currently open.
- The MIDI engine is currently processing
- MIDI events. */
- OMX_AUDIO_MIDIPlayBackStateParsing, /**< A MIDI resource is open and is being
- primed. The MIDI engine is currently
- processing MIDI events. */
- OMX_AUDIO_MIDIPlayBackStateOpenEngaged, /**< A MIDI resource is open and primed but
- not playing. The MIDI engine is currently
- processing MIDI events. The transition to
- this state is only possible from the
- OMX_AUDIO_MIDIPlayBackStatePlaying state,
- when the 'playback head' reaches the end
- of media data or the playback stops due
- to stop time set.*/
- OMX_AUDIO_MIDIPlayBackStatePlaying, /**< A MIDI resource is open and currently
- playing. The MIDI engine is currently
- processing MIDI events.*/
- OMX_AUDIO_MIDIPlayBackStatePlayingPartially, /**< Best-effort playback due to SP-MIDI/DLS
- resource constraints */
- OMX_AUDIO_MIDIPlayBackStatePlayingSilently, /**< Due to system resource constraints and
- SP-MIDI content constraints, there is
- no audible MIDI content during playback
- currently. The situation may change if
- resources are freed later.*/
- OMX_AUDIO_MIDIPlayBackStateKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_AUDIO_MIDIPlayBackStateVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_AUDIO_MIDIPlayBackStateMax = 0x7FFFFFFF
-} OMX_AUDIO_MIDIPLAYBACKSTATETYPE;
-
-
-/** MIDI status
- * @ingroup midi
- */
-typedef struct OMX_AUDIO_CONFIG_MIDISTATUSTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_U16 nNumTracks; /**< Number of MIDI tracks in the file, read only field.
- NOTE: May not return a meaningful value until the entire
- file is parsed and buffered. */
- OMX_U32 nDuration; /**< The length of the currently open MIDI resource
- in milliseconds. NOTE: May not return a meaningful value
- until the entire file is parsed and buffered. */
- OMX_U32 nPosition; /**< Current Position of the MIDI resource being played
- in milliseconds */
- OMX_BOOL bVibra; /**< Does Vibra track exist? NOTE: May not return a meaningful
- value until the entire file is parsed and buffered. */
- OMX_U32 nNumMetaEvents; /**< Total number of MIDI Meta Events in the currently
- open MIDI resource. NOTE: May not return a meaningful value
- until the entire file is parsed and buffered. */
- OMX_U32 nNumActiveVoices; /**< Number of active voices in the currently playing
- MIDI resource. NOTE: May not return a meaningful value until
- the entire file is parsed and buffered. */
- OMX_AUDIO_MIDIPLAYBACKSTATETYPE eMIDIPlayBackState; /**< MIDI playback state enumeration, read only field */
-} OMX_AUDIO_CONFIG_MIDISTATUSTYPE;
-
-
-/** MIDI Meta Event structure one per Meta Event.
- * MIDI Meta Events are like audio metadata, except that they are interspersed
- * with the MIDI content throughout the file and are not localized in the header.
- * As such, it is necessary to retrieve information about these Meta Events from
- * the engine, as it encounters these Meta Events within the MIDI content.
- * For example, SMF files can have up to 14 types of MIDI Meta Events (copyright,
- * author, default tempo, etc.) scattered throughout the file.
- * @ingroup midi
- */
-typedef struct OMX_AUDIO_CONFIG_MIDIMETAEVENTTYPE{
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_U32 nIndex; /**< Index of Meta Event */
- OMX_U8 nMetaEventType; /**< Meta Event Type, 7bits (i.e. 0 - 127) */
- OMX_U32 nMetaEventSize; /**< size of the Meta Event in bytes */
- OMX_U32 nTrack; /**< track number for the meta event */
- OMX_U32 nPosition; /**< Position of the meta-event in milliseconds */
-} OMX_AUDIO_CONFIG_MIDIMETAEVENTTYPE;
-
-
-/** MIDI Meta Event Data structure - one per Meta Event.
- * @ingroup midi
- */
-typedef struct OMX_AUDIO_CONFIG_MIDIMETAEVENTDATATYPE{
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_U32 nIndex; /**< Index of Meta Event */
- OMX_U32 nMetaEventSize; /**< size of the Meta Event in bytes */
- OMX_U8 nData[1]; /**< array of one or more bytes of meta data
- as indicated by the nMetaEventSize field */
-} OMX_AUDIO_CONFIG__MIDIMETAEVENTDATATYPE;
-
-
-/** Audio Volume adjustment for a port */
-typedef struct OMX_AUDIO_CONFIG_VOLUMETYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< Port index indicating which port to
- set. Select the input port to set
- just that port's volume. Select the
- output port to adjust the master
- volume. */
- OMX_BOOL bLinear; /**< Is the volume to be set in linear (0.100)
- or logarithmic scale (mB) */
- OMX_BS32 sVolume; /**< Volume linear setting in the 0..100 range, OR
- Volume logarithmic setting for this port. The values
- for volume are in mB (millibels = 1/100 dB) relative
- to a gain of 1 (e.g. the output is the same as the
- input level). Values are in mB from nMax
- (maximum volume) to nMin mB (typically negative).
- Since the volume is "voltage"
- and not a "power", it takes a setting of
- -600 mB to decrease the volume by 1/2. If
- a component cannot accurately set the
- volume to the requested value, it must
- set the volume to the closest value BELOW
- the requested value. When getting the
- volume setting, the current actual volume
- must be returned. */
-} OMX_AUDIO_CONFIG_VOLUMETYPE;
-
-
-/** Audio Volume adjustment for a channel */
-typedef struct OMX_AUDIO_CONFIG_CHANNELVOLUMETYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< Port index indicating which port to
- set. Select the input port to set
- just that port's volume. Select the
- output port to adjust the master
- volume. */
- OMX_U32 nChannel; /**< channel to select from 0 to N-1,
- using OMX_ALL to apply volume settings
- to all channels */
- OMX_BOOL bLinear; /**< Is the volume to be set in linear (0.100) or
- logarithmic scale (mB) */
- OMX_BS32 sVolume; /**< Volume linear setting in the 0..100 range, OR
- Volume logarithmic setting for this port.
- The values for volume are in mB
- (millibels = 1/100 dB) relative to a gain
- of 1 (e.g. the output is the same as the
- input level). Values are in mB from nMax
- (maximum volume) to nMin mB (typically negative).
- Since the volume is "voltage"
- and not a "power", it takes a setting of
- -600 mB to decrease the volume by 1/2. If
- a component cannot accurately set the
- volume to the requested value, it must
- set the volume to the closest value BELOW
- the requested value. When getting the
- volume setting, the current actual volume
- must be returned. */
- OMX_BOOL bIsMIDI; /**< TRUE if nChannel refers to a MIDI channel,
- FALSE otherwise */
-} OMX_AUDIO_CONFIG_CHANNELVOLUMETYPE;
-
-
-/** Audio balance setting */
-typedef struct OMX_AUDIO_CONFIG_BALANCETYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< Port index indicating which port to
- set. Select the input port to set
- just that port's balance. Select the
- output port to adjust the master
- balance. */
- OMX_S32 nBalance; /**< balance setting for this port
- (-100 to 100, where -100 indicates
- all left, and no right */
-} OMX_AUDIO_CONFIG_BALANCETYPE;
-
-
-/** Audio Port mute */
-typedef struct OMX_AUDIO_CONFIG_MUTETYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< Port index indicating which port to
- set. Select the input port to set
- just that port's mute. Select the
- output port to adjust the master
- mute. */
- OMX_BOOL bMute; /**< Mute setting for this port */
-} OMX_AUDIO_CONFIG_MUTETYPE;
-
-
-/** Audio Channel mute */
-typedef struct OMX_AUDIO_CONFIG_CHANNELMUTETYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_U32 nChannel; /**< channel to select from 0 to N-1,
- using OMX_ALL to apply mute settings
- to all channels */
- OMX_BOOL bMute; /**< Mute setting for this channel */
- OMX_BOOL bIsMIDI; /**< TRUE if nChannel refers to a MIDI channel,
- FALSE otherwise */
-} OMX_AUDIO_CONFIG_CHANNELMUTETYPE;
-
-
-
-/** Enable / Disable for loudness control, which boosts bass and to a
- * smaller extent high end frequencies to compensate for hearing
- * ability at the extreme ends of the audio spectrum
- */
-typedef struct OMX_AUDIO_CONFIG_LOUDNESSTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_BOOL bLoudness; /**< Enable/disable for loudness */
-} OMX_AUDIO_CONFIG_LOUDNESSTYPE;
-
-
-/** Enable / Disable for bass, which controls low frequencies
- */
-typedef struct OMX_AUDIO_CONFIG_BASSTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_BOOL bEnable; /**< Enable/disable for bass control */
- OMX_S32 nBass; /**< bass setting for the port, as a
- continuous value from -100 to 100
- (0 means no change in bass level)*/
-} OMX_AUDIO_CONFIG_BASSTYPE;
-
-
-/** Enable / Disable for treble, which controls high frequencies tones
- */
-typedef struct OMX_AUDIO_CONFIG_TREBLETYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_BOOL bEnable; /**< Enable/disable for treble control */
- OMX_S32 nTreble; /**< treble setting for the port, as a
- continuous value from -100 to 100
- (0 means no change in treble level) */
-} OMX_AUDIO_CONFIG_TREBLETYPE;
-
-
-/** An equalizer is typically used for two reasons: to compensate for an
- * sub-optimal frequency response of a system to make it sound more natural
- * or to create intentionally some unnatural coloring to the sound to create
- * an effect.
- * @ingroup effects
- */
-typedef struct OMX_AUDIO_CONFIG_EQUALIZERTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_BOOL bEnable; /**< Enable/disable for equalizer */
- OMX_BU32 sBandIndex; /**< Band number to be set. Upper Limit is
- N-1, where N is the number of bands, lower limit is 0 */
- OMX_BU32 sCenterFreq; /**< Center frequecies in Hz. This is a
- read only element and is used to determine
- the lower, center and upper frequency of
- this band. */
- OMX_BS32 sBandLevel; /**< band level in millibels */
-} OMX_AUDIO_CONFIG_EQUALIZERTYPE;
-
-
-/** Stereo widening mode type
- * @ingroup effects
- */
-typedef enum OMX_AUDIO_STEREOWIDENINGTYPE {
- OMX_AUDIO_StereoWideningHeadphones, /**< Stereo widening for loudspeakers */
- OMX_AUDIO_StereoWideningLoudspeakers, /**< Stereo widening for closely spaced loudspeakers */
- OMX_AUDIO_StereoWideningKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_AUDIO_StereoWideningVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_AUDIO_StereoWideningMax = 0x7FFFFFFF
-} OMX_AUDIO_STEREOWIDENINGTYPE;
-
-
-/** Control for stereo widening, which is a special 2-channel
- * case of the audio virtualizer effect. For example, for 5.1-channel
- * output, it translates to virtual surround sound.
- * @ingroup effects
- */
-typedef struct OMX_AUDIO_CONFIG_STEREOWIDENINGTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_BOOL bEnable; /**< Enable/disable for stereo widening control */
- OMX_AUDIO_STEREOWIDENINGTYPE eWideningType; /**< Stereo widening algorithm type */
- OMX_U32 nStereoWidening; /**< stereo widening setting for the port,
- as a continuous value from 0 to 100 */
-} OMX_AUDIO_CONFIG_STEREOWIDENINGTYPE;
-
-
-/** The chorus effect (or ``choralizer'') is any signal processor which makes
- * one sound source (such as a voice) sound like many such sources singing
- * (or playing) in unison. Since performance in unison is never exact, chorus
- * effects simulate this by making independently modified copies of the input
- * signal. Modifications may include (1) delay, (2) frequency shift, and
- * (3) amplitude modulation.
- * @ingroup effects
- */
-typedef struct OMX_AUDIO_CONFIG_CHORUSTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_BOOL bEnable; /**< Enable/disable for chorus */
- OMX_BU32 sDelay; /**< average delay in milliseconds */
- OMX_BU32 sModulationRate; /**< rate of modulation in millihertz */
- OMX_U32 nModulationDepth; /**< depth of modulation as a percentage of
- delay (i.e. 0 to 100) */
- OMX_BU32 nFeedback; /**< Feedback from chorus output to input in percentage */
-} OMX_AUDIO_CONFIG_CHORUSTYPE;
-
-
-/** Reverberation is part of the reflected sound that follows the early
- * reflections. In a typical room, this consists of a dense succession of
- * echoes whose energy decays exponentially. The reverberation effect structure
- * as defined here includes both (early) reflections as well as (late) reverberations.
- * @ingroup effects
- */
-typedef struct OMX_AUDIO_CONFIG_REVERBERATIONTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_BOOL bEnable; /**< Enable/disable for reverberation control */
- OMX_BS32 sRoomLevel; /**< Intensity level for the whole room effect
- (i.e. both early reflections and late
- reverberation) in millibels */
- OMX_BS32 sRoomHighFreqLevel; /**< Attenuation at high frequencies
- relative to the intensity at low
- frequencies in millibels */
- OMX_BS32 sReflectionsLevel; /**< Intensity level of early reflections
- (relative to room value), in millibels */
- OMX_BU32 sReflectionsDelay; /**< Delay time of the first reflection relative
- to the direct path, in milliseconds */
- OMX_BS32 sReverbLevel; /**< Intensity level of late reverberation
- relative to room level, in millibels */
- OMX_BU32 sReverbDelay; /**< Time delay from the first early reflection
- to the beginning of the late reverberation
- section, in milliseconds */
- OMX_BU32 sDecayTime; /**< Late reverberation decay time at low
- frequencies, in milliseconds */
- OMX_BU32 nDecayHighFreqRatio; /**< Ratio of high frequency decay time relative
- to low frequency decay time in percent */
- OMX_U32 nDensity; /**< Modal density in the late reverberation decay,
- in percent (i.e. 0 - 100) */
- OMX_U32 nDiffusion; /**< Echo density in the late reverberation decay,
- in percent (i.e. 0 - 100) */
- OMX_BU32 sReferenceHighFreq; /**< Reference high frequency in Hertz. This is
- the frequency used as the reference for all
- the high-frequency settings above */
-
-} OMX_AUDIO_CONFIG_REVERBERATIONTYPE;
-
-
-/** Possible settings for the Echo Cancelation structure to use
- * @ingroup effects
- */
-typedef enum OMX_AUDIO_ECHOCANTYPE {
- OMX_AUDIO_EchoCanOff = 0, /**< Echo Cancellation is disabled */
- OMX_AUDIO_EchoCanNormal, /**< Echo Cancellation normal operation -
- echo from plastics and face */
- OMX_AUDIO_EchoCanHFree, /**< Echo Cancellation optimized for
- Hands Free operation */
- OMX_AUDIO_EchoCanCarKit, /**< Echo Cancellation optimized for
- Car Kit (longer echo) */
- OMX_AUDIO_EchoCanKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_AUDIO_EchoCanVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_AUDIO_EchoCanMax = 0x7FFFFFFF
-} OMX_AUDIO_ECHOCANTYPE;
-
-
-/** Enable / Disable for echo cancelation, which removes undesired echo's
- * from the audio
- * @ingroup effects
- */
-typedef struct OMX_AUDIO_CONFIG_ECHOCANCELATIONTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_AUDIO_ECHOCANTYPE eEchoCancelation; /**< Echo cancelation settings */
-} OMX_AUDIO_CONFIG_ECHOCANCELATIONTYPE;
-
-
-/** Enable / Disable for noise reduction, which undesired noise from
- * the audio
- * @ingroup effects
- */
-typedef struct OMX_AUDIO_CONFIG_NOISEREDUCTIONTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_BOOL bNoiseReduction; /**< Enable/disable for noise reduction */
-} OMX_AUDIO_CONFIG_NOISEREDUCTIONTYPE;
-
-/** @} */
-
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
-#endif
-/* File EOF */
-
diff --git a/include/media/stagefright/openmax/OMX_Component.h b/include/media/stagefright/openmax/OMX_Component.h
deleted file mode 100644
index b5b784e..0000000
--- a/include/media/stagefright/openmax/OMX_Component.h
+++ /dev/null
@@ -1,596 +0,0 @@
-/* ------------------------------------------------------------------
- * Copyright (C) 1998-2009 PacketVideo
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either
- * express or implied.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- * -------------------------------------------------------------------
- */
-/*
- * Copyright (c) 2008 The Khronos Group Inc.
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files (the
- * "Software"), to deal in the Software without restriction, including
- * without limitation the rights to use, copy, modify, merge, publish,
- * distribute, sublicense, and/or sell copies of the Software, and to
- * permit persons to whom the Software is furnished to do so, subject
- * to the following conditions:
- * The above copyright notice and this permission notice shall be included
- * in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS
- * OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY
- * CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
- * TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
- * SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- *
- */
-
-/** OMX_Component.h - OpenMax IL version 1.1.2
- * The OMX_Component header file contains the definitions used to define
- * the public interface of a component. This header file is intended to
- * be used by both the application and the component.
- */
-
-#ifndef OMX_Component_h
-#define OMX_Component_h
-
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-
-
-/* Each OMX header must include all required header files to allow the
- * header to compile without errors. The includes below are required
- * for this header file to compile successfully
- */
-
-#include <OMX_Audio.h>
-#include <OMX_Video.h>
-#include <OMX_Image.h>
-#include <OMX_Other.h>
-
-/** @ingroup comp */
-typedef enum OMX_PORTDOMAINTYPE {
- OMX_PortDomainAudio,
- OMX_PortDomainVideo,
- OMX_PortDomainImage,
- OMX_PortDomainOther,
- OMX_PortDomainKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_PortDomainVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_PortDomainMax = 0x7ffffff
-} OMX_PORTDOMAINTYPE;
-
-/** @ingroup comp */
-typedef struct OMX_PARAM_PORTDEFINITIONTYPE {
- OMX_U32 nSize; /**< Size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< Port number the structure applies to */
- OMX_DIRTYPE eDir; /**< Direction (input or output) of this port */
- OMX_U32 nBufferCountActual; /**< The actual number of buffers allocated on this port */
- OMX_U32 nBufferCountMin; /**< The minimum number of buffers this port requires */
- OMX_U32 nBufferSize; /**< Size, in bytes, for buffers to be used for this channel */
- OMX_BOOL bEnabled; /**< Ports default to enabled and are enabled/disabled by
- OMX_CommandPortEnable/OMX_CommandPortDisable.
- When disabled a port is unpopulated. A disabled port
- is not populated with buffers on a transition to IDLE. */
- OMX_BOOL bPopulated; /**< Port is populated with all of its buffers as indicated by
- nBufferCountActual. A disabled port is always unpopulated.
- An enabled port is populated on a transition to OMX_StateIdle
- and unpopulated on a transition to loaded. */
- OMX_PORTDOMAINTYPE eDomain; /**< Domain of the port. Determines the contents of metadata below. */
- union {
- OMX_AUDIO_PORTDEFINITIONTYPE audio;
- OMX_VIDEO_PORTDEFINITIONTYPE video;
- OMX_IMAGE_PORTDEFINITIONTYPE image;
- OMX_OTHER_PORTDEFINITIONTYPE other;
- } format;
- OMX_BOOL bBuffersContiguous;
- OMX_U32 nBufferAlignment;
-} OMX_PARAM_PORTDEFINITIONTYPE;
-
-/** @ingroup comp */
-typedef struct OMX_PARAM_U32TYPE {
- OMX_U32 nSize; /**< Size of this structure, in Bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_U32 nU32; /**< U32 value */
-} OMX_PARAM_U32TYPE;
-
-/** @ingroup rpm */
-typedef enum OMX_SUSPENSIONPOLICYTYPE {
- OMX_SuspensionDisabled, /**< No suspension; v1.0 behavior */
- OMX_SuspensionEnabled, /**< Suspension allowed */
- OMX_SuspensionPolicyKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_SuspensionPolicyStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_SuspensionPolicyMax = 0x7fffffff
-} OMX_SUSPENSIONPOLICYTYPE;
-
-/** @ingroup rpm */
-typedef struct OMX_PARAM_SUSPENSIONPOLICYTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_SUSPENSIONPOLICYTYPE ePolicy;
-} OMX_PARAM_SUSPENSIONPOLICYTYPE;
-
-/** @ingroup rpm */
-typedef enum OMX_SUSPENSIONTYPE {
- OMX_NotSuspended, /**< component is not suspended */
- OMX_Suspended, /**< component is suspended */
- OMX_SuspensionKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_SuspensionVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_SuspendMax = 0x7FFFFFFF
-} OMX_SUSPENSIONTYPE;
-
-/** @ingroup rpm */
-typedef struct OMX_PARAM_SUSPENSIONTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_SUSPENSIONTYPE eType;
-} OMX_PARAM_SUSPENSIONTYPE ;
-
-typedef struct OMX_CONFIG_BOOLEANTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_BOOL bEnabled;
-} OMX_CONFIG_BOOLEANTYPE;
-
-/* Parameter specifying the content uri to use. */
-/** @ingroup cp */
-typedef struct OMX_PARAM_CONTENTURITYPE
-{
- OMX_U32 nSize; /**< size of the structure in bytes, including
- actual URI name */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U8 contentURI[1]; /**< The URI name */
-} OMX_PARAM_CONTENTURITYPE;
-
-/* Parameter specifying the pipe to use. */
-/** @ingroup cp */
-typedef struct OMX_PARAM_CONTENTPIPETYPE
-{
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_HANDLETYPE hPipe; /**< The pipe handle*/
-} OMX_PARAM_CONTENTPIPETYPE;
-
-/** @ingroup rpm */
-typedef struct OMX_RESOURCECONCEALMENTTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_BOOL bResourceConcealmentForbidden; /**< disallow the use of resource concealment
- methods (like degrading algorithm quality to
- lower resource consumption or functional bypass)
- on a component as a resolution to resource conflicts. */
-} OMX_RESOURCECONCEALMENTTYPE;
-
-
-/** @ingroup metadata */
-typedef enum OMX_METADATACHARSETTYPE {
- OMX_MetadataCharsetUnknown = 0,
- OMX_MetadataCharsetASCII,
- OMX_MetadataCharsetBinary,
- OMX_MetadataCharsetCodePage1252,
- OMX_MetadataCharsetUTF8,
- OMX_MetadataCharsetJavaConformantUTF8,
- OMX_MetadataCharsetUTF7,
- OMX_MetadataCharsetImapUTF7,
- OMX_MetadataCharsetUTF16LE,
- OMX_MetadataCharsetUTF16BE,
- OMX_MetadataCharsetGB12345,
- OMX_MetadataCharsetHZGB2312,
- OMX_MetadataCharsetGB2312,
- OMX_MetadataCharsetGB18030,
- OMX_MetadataCharsetGBK,
- OMX_MetadataCharsetBig5,
- OMX_MetadataCharsetISO88591,
- OMX_MetadataCharsetISO88592,
- OMX_MetadataCharsetISO88593,
- OMX_MetadataCharsetISO88594,
- OMX_MetadataCharsetISO88595,
- OMX_MetadataCharsetISO88596,
- OMX_MetadataCharsetISO88597,
- OMX_MetadataCharsetISO88598,
- OMX_MetadataCharsetISO88599,
- OMX_MetadataCharsetISO885910,
- OMX_MetadataCharsetISO885913,
- OMX_MetadataCharsetISO885914,
- OMX_MetadataCharsetISO885915,
- OMX_MetadataCharsetShiftJIS,
- OMX_MetadataCharsetISO2022JP,
- OMX_MetadataCharsetISO2022JP1,
- OMX_MetadataCharsetISOEUCJP,
- OMX_MetadataCharsetSMS7Bit,
- OMX_MetadataCharsetKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_MetadataCharsetVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_MetadataCharsetTypeMax= 0x7FFFFFFF
-} OMX_METADATACHARSETTYPE;
-
-/** @ingroup metadata */
-typedef enum OMX_METADATASCOPETYPE
-{
- OMX_MetadataScopeAllLevels,
- OMX_MetadataScopeTopLevel,
- OMX_MetadataScopePortLevel,
- OMX_MetadataScopeNodeLevel,
- OMX_MetadataScopeKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_MetadataScopeVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_MetadataScopeTypeMax = 0x7fffffff
-} OMX_METADATASCOPETYPE;
-
-/** @ingroup metadata */
-typedef enum OMX_METADATASEARCHMODETYPE
-{
- OMX_MetadataSearchValueSizeByIndex,
- OMX_MetadataSearchItemByIndex,
- OMX_MetadataSearchNextItemByKey,
- OMX_MetadataSearchKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_MetadataSearchVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_MetadataSearchTypeMax = 0x7fffffff
-} OMX_METADATASEARCHMODETYPE;
-/** @ingroup metadata */
-typedef struct OMX_CONFIG_METADATAITEMCOUNTTYPE
-{
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_METADATASCOPETYPE eScopeMode;
- OMX_U32 nScopeSpecifier;
- OMX_U32 nMetadataItemCount;
-} OMX_CONFIG_METADATAITEMCOUNTTYPE;
-
-/** @ingroup metadata */
-typedef struct OMX_CONFIG_METADATAITEMTYPE
-{
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_METADATASCOPETYPE eScopeMode;
- OMX_U32 nScopeSpecifier;
- OMX_U32 nMetadataItemIndex;
- OMX_METADATASEARCHMODETYPE eSearchMode;
- OMX_METADATACHARSETTYPE eKeyCharset;
- OMX_U8 nKeySizeUsed;
- OMX_U8 nKey[128];
- OMX_METADATACHARSETTYPE eValueCharset;
- OMX_STRING sLanguageCountry;
- OMX_U32 nValueMaxSize;
- OMX_U32 nValueSizeUsed;
- OMX_U8 nValue[1];
-} OMX_CONFIG_METADATAITEMTYPE;
-
-/* @ingroup metadata */
-typedef struct OMX_CONFIG_CONTAINERNODECOUNTTYPE
-{
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_BOOL bAllKeys;
- OMX_U32 nParentNodeID;
- OMX_U32 nNumNodes;
-} OMX_CONFIG_CONTAINERNODECOUNTTYPE;
-
-/** @ingroup metadata */
-typedef struct OMX_CONFIG_CONTAINERNODEIDTYPE
-{
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_BOOL bAllKeys;
- OMX_U32 nParentNodeID;
- OMX_U32 nNodeIndex;
- OMX_U32 nNodeID;
- OMX_STRING cNodeName;
- OMX_BOOL bIsLeafType;
-} OMX_CONFIG_CONTAINERNODEIDTYPE;
-
-/** @ingroup metadata */
-typedef struct OMX_PARAM_METADATAFILTERTYPE
-{
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_BOOL bAllKeys; /* if true then this structure refers to all keys and
- * the three key fields below are ignored */
- OMX_METADATACHARSETTYPE eKeyCharset;
- OMX_U32 nKeySizeUsed;
- OMX_U8 nKey [128];
- OMX_U32 nLanguageCountrySizeUsed;
- OMX_U8 nLanguageCountry[128];
- OMX_BOOL bEnabled; /* if true then key is part of filter (e.g.
- * retained for query later). If false then
- * key is not part of filter */
-} OMX_PARAM_METADATAFILTERTYPE;
-
-/** The OMX_HANDLETYPE structure defines the component handle. The component
- * handle is used to access all of the component's public methods and also
- * contains pointers to the component's private data area. The component
- * handle is initialized by the OMX core (with help from the component)
- * during the process of loading the component. After the component is
- * successfully loaded, the application can safely access any of the
- * component's public functions (although some may return an error because
- * the state is inappropriate for the access).
- *
- * @ingroup comp
- */
-typedef struct OMX_COMPONENTTYPE
-{
- /** The size of this structure, in bytes. It is the responsibility
- of the allocator of this structure to fill in this value. Since
- this structure is allocated by the GetHandle function, this
- function will fill in this value. */
- OMX_U32 nSize;
-
- /** nVersion is the version of the OMX specification that the structure
- is built against. It is the responsibility of the creator of this
- structure to initialize this value and every user of this structure
- should verify that it knows how to use the exact version of
- this structure found herein. */
- OMX_VERSIONTYPE nVersion;
-
- /** pComponentPrivate is a pointer to the component private data area.
- This member is allocated and initialized by the component when the
- component is first loaded. The application should not access this
- data area. */
- OMX_PTR pComponentPrivate;
-
- /** pApplicationPrivate is a pointer that is a parameter to the
- OMX_GetHandle method, and contains an application private value
- provided by the IL client. This application private data is
- returned to the IL Client by OMX in all callbacks */
- OMX_PTR pApplicationPrivate;
-
- /** refer to OMX_GetComponentVersion in OMX_core.h or the OMX IL
- specification for details on the GetComponentVersion method.
- */
- OMX_ERRORTYPE (*GetComponentVersion)(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_OUT OMX_STRING pComponentName,
- OMX_OUT OMX_VERSIONTYPE* pComponentVersion,
- OMX_OUT OMX_VERSIONTYPE* pSpecVersion,
- OMX_OUT OMX_UUIDTYPE* pComponentUUID);
-
- /** refer to OMX_SendCommand in OMX_core.h or the OMX IL
- specification for details on the SendCommand method.
- */
- OMX_ERRORTYPE (*SendCommand)(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_IN OMX_COMMANDTYPE Cmd,
- OMX_IN OMX_U32 nParam1,
- OMX_IN OMX_PTR pCmdData);
-
- /** refer to OMX_GetParameter in OMX_core.h or the OMX IL
- specification for details on the GetParameter method.
- */
- OMX_ERRORTYPE (*GetParameter)(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_IN OMX_INDEXTYPE nParamIndex,
- OMX_INOUT OMX_PTR pComponentParameterStructure);
-
-
- /** refer to OMX_SetParameter in OMX_core.h or the OMX IL
- specification for details on the SetParameter method.
- */
- OMX_ERRORTYPE (*SetParameter)(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_IN OMX_INDEXTYPE nIndex,
- OMX_IN OMX_PTR pComponentParameterStructure);
-
-
- /** refer to OMX_GetConfig in OMX_core.h or the OMX IL
- specification for details on the GetConfig method.
- */
- OMX_ERRORTYPE (*GetConfig)(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_IN OMX_INDEXTYPE nIndex,
- OMX_INOUT OMX_PTR pComponentConfigStructure);
-
-
- /** refer to OMX_SetConfig in OMX_core.h or the OMX IL
- specification for details on the SetConfig method.
- */
- OMX_ERRORTYPE (*SetConfig)(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_IN OMX_INDEXTYPE nIndex,
- OMX_IN OMX_PTR pComponentConfigStructure);
-
-
- /** refer to OMX_GetExtensionIndex in OMX_core.h or the OMX IL
- specification for details on the GetExtensionIndex method.
- */
- OMX_ERRORTYPE (*GetExtensionIndex)(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_IN OMX_STRING cParameterName,
- OMX_OUT OMX_INDEXTYPE* pIndexType);
-
-
- /** refer to OMX_GetState in OMX_core.h or the OMX IL
- specification for details on the GetState method.
- */
- OMX_ERRORTYPE (*GetState)(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_OUT OMX_STATETYPE* pState);
-
-
- /** The ComponentTunnelRequest method will interact with another OMX
- component to determine if tunneling is possible and to setup the
- tunneling. The return codes for this method can be used to
- determine if tunneling is not possible, or if tunneling is not
- supported.
-
- Base profile components (i.e. non-interop) do not support this
- method and should return OMX_ErrorNotImplemented
-
- The interop profile component MUST support tunneling to another
- interop profile component with a compatible port parameters.
- A component may also support proprietary communication.
-
- If proprietary communication is supported the negotiation of
- proprietary communication is done outside of OMX in a vendor
- specific way. It is only required that the proper result be
- returned and the details of how the setup is done is left
- to the component implementation.
-
- When this method is invoked when nPort in an output port, the
- component will:
- 1. Populate the pTunnelSetup structure with the output port's
- requirements and constraints for the tunnel.
-
- When this method is invoked when nPort in an input port, the
- component will:
- 1. Query the necessary parameters from the output port to
- determine if the ports are compatible for tunneling
- 2. If the ports are compatible, the component should store
- the tunnel step provided by the output port
- 3. Determine which port (either input or output) is the buffer
- supplier, and call OMX_SetParameter on the output port to
- indicate this selection.
-
- The component will return from this call within 5 msec.
-
- @param [in] hComp
- Handle of the component to be accessed. This is the component
- handle returned by the call to the OMX_GetHandle method.
- @param [in] nPort
- nPort is used to select the port on the component to be used
- for tunneling.
- @param [in] hTunneledComp
- Handle of the component to tunnel with. This is the component
- handle returned by the call to the OMX_GetHandle method. When
- this parameter is 0x0 the component should setup the port for
- communication with the application / IL Client.
- @param [in] nPortOutput
- nPortOutput is used indicate the port the component should
- tunnel with.
- @param [in] pTunnelSetup
- Pointer to the tunnel setup structure. When nPort is an output port
- the component should populate the fields of this structure. When
- When nPort is an input port the component should review the setup
- provided by the component with the output port.
- @return OMX_ERRORTYPE
- If the command successfully executes, the return code will be
- OMX_ErrorNone. Otherwise the appropriate OMX error will be returned.
- @ingroup tun
- */
-
- OMX_ERRORTYPE (*ComponentTunnelRequest)(
- OMX_IN OMX_HANDLETYPE hComp,
- OMX_IN OMX_U32 nPort,
- OMX_IN OMX_HANDLETYPE hTunneledComp,
- OMX_IN OMX_U32 nTunneledPort,
- OMX_INOUT OMX_TUNNELSETUPTYPE* pTunnelSetup);
-
- /** refer to OMX_UseBuffer in OMX_core.h or the OMX IL
- specification for details on the UseBuffer method.
- @ingroup buf
- */
- OMX_ERRORTYPE (*UseBuffer)(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_INOUT OMX_BUFFERHEADERTYPE** ppBufferHdr,
- OMX_IN OMX_U32 nPortIndex,
- OMX_IN OMX_PTR pAppPrivate,
- OMX_IN OMX_U32 nSizeBytes,
- OMX_IN OMX_U8* pBuffer);
-
- /** refer to OMX_AllocateBuffer in OMX_core.h or the OMX IL
- specification for details on the AllocateBuffer method.
- @ingroup buf
- */
- OMX_ERRORTYPE (*AllocateBuffer)(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_INOUT OMX_BUFFERHEADERTYPE** ppBuffer,
- OMX_IN OMX_U32 nPortIndex,
- OMX_IN OMX_PTR pAppPrivate,
- OMX_IN OMX_U32 nSizeBytes);
-
- /** refer to OMX_FreeBuffer in OMX_core.h or the OMX IL
- specification for details on the FreeBuffer method.
- @ingroup buf
- */
- OMX_ERRORTYPE (*FreeBuffer)(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_IN OMX_U32 nPortIndex,
- OMX_IN OMX_BUFFERHEADERTYPE* pBuffer);
-
- /** refer to OMX_EmptyThisBuffer in OMX_core.h or the OMX IL
- specification for details on the EmptyThisBuffer method.
- @ingroup buf
- */
- OMX_ERRORTYPE (*EmptyThisBuffer)(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_IN OMX_BUFFERHEADERTYPE* pBuffer);
-
- /** refer to OMX_FillThisBuffer in OMX_core.h or the OMX IL
- specification for details on the FillThisBuffer method.
- @ingroup buf
- */
- OMX_ERRORTYPE (*FillThisBuffer)(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_IN OMX_BUFFERHEADERTYPE* pBuffer);
-
- /** The SetCallbacks method is used by the core to specify the callback
- structure from the application to the component. This is a blocking
- call. The component will return from this call within 5 msec.
- @param [in] hComponent
- Handle of the component to be accessed. This is the component
- handle returned by the call to the GetHandle function.
- @param [in] pCallbacks
- pointer to an OMX_CALLBACKTYPE structure used to provide the
- callback information to the component
- @param [in] pAppData
- pointer to an application defined value. It is anticipated that
- the application will pass a pointer to a data structure or a "this
- pointer" in this area to allow the callback (in the application)
- to determine the context of the call
- @return OMX_ERRORTYPE
- If the command successfully executes, the return code will be
- OMX_ErrorNone. Otherwise the appropriate OMX error will be returned.
- */
- OMX_ERRORTYPE (*SetCallbacks)(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_IN OMX_CALLBACKTYPE* pCallbacks,
- OMX_IN OMX_PTR pAppData);
-
- /** ComponentDeInit method is used to deinitialize the component
- providing a means to free any resources allocated at component
- initialization. NOTE: After this call the component handle is
- not valid for further use.
- @param [in] hComponent
- Handle of the component to be accessed. This is the component
- handle returned by the call to the GetHandle function.
- @return OMX_ERRORTYPE
- If the command successfully executes, the return code will be
- OMX_ErrorNone. Otherwise the appropriate OMX error will be returned.
- */
- OMX_ERRORTYPE (*ComponentDeInit)(
- OMX_IN OMX_HANDLETYPE hComponent);
-
- /** @ingroup buf */
- OMX_ERRORTYPE (*UseEGLImage)(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_INOUT OMX_BUFFERHEADERTYPE** ppBufferHdr,
- OMX_IN OMX_U32 nPortIndex,
- OMX_IN OMX_PTR pAppPrivate,
- OMX_IN void* eglImage);
-
- OMX_ERRORTYPE (*ComponentRoleEnum)(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_OUT OMX_U8 *cRole,
- OMX_IN OMX_U32 nIndex);
-
-} OMX_COMPONENTTYPE;
-
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
-#endif
-/* File EOF */
diff --git a/include/media/stagefright/openmax/OMX_ContentPipe.h b/include/media/stagefright/openmax/OMX_ContentPipe.h
deleted file mode 100644
index ee9e4db..0000000
--- a/include/media/stagefright/openmax/OMX_ContentPipe.h
+++ /dev/null
@@ -1,212 +0,0 @@
-/* ------------------------------------------------------------------
- * Copyright (C) 1998-2009 PacketVideo
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either
- * express or implied.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- * -------------------------------------------------------------------
- */
-/*
- * Copyright (c) 2008 The Khronos Group Inc.
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files (the
- * "Software"), to deal in the Software without restriction, including
- * without limitation the rights to use, copy, modify, merge, publish,
- * distribute, sublicense, and/or sell copies of the Software, and to
- * permit persons to whom the Software is furnished to do so, subject
- * to the following conditions:
- * The above copyright notice and this permission notice shall be included
- * in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS
- * OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY
- * CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
- * TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
- * SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- *
- */
-
-/** OMX_ContentPipe.h - OpenMax IL version 1.1.2
- * The OMX_ContentPipe header file contains the definitions used to define
- * the public interface for content piples. This header file is intended to
- * be used by the component.
- */
-
-#ifndef OMX_CONTENTPIPE_H
-#define OMX_CONTENTPIPE_H
-
-#ifndef KD_EACCES
-/* OpenKODE error codes. CPResult values may be zero (indicating success
- or one of the following values) */
-#define KD_EACCES (1)
-#define KD_EADDRINUSE (2)
-#define KD_EAGAIN (5)
-#define KD_EBADF (7)
-#define KD_EBUSY (8)
-#define KD_ECONNREFUSED (9)
-#define KD_ECONNRESET (10)
-#define KD_EDEADLK (11)
-#define KD_EDESTADDRREQ (12)
-#define KD_ERANGE (35)
-#define KD_EEXIST (13)
-#define KD_EFBIG (14)
-#define KD_EHOSTUNREACH (15)
-#define KD_EINVAL (17)
-#define KD_EIO (18)
-#define KD_EISCONN (20)
-#define KD_EISDIR (21)
-#define KD_EMFILE (22)
-#define KD_ENAMETOOLONG (23)
-#define KD_ENOENT (24)
-#define KD_ENOMEM (25)
-#define KD_ENOSPC (26)
-#define KD_ENOSYS (27)
-#define KD_ENOTCONN (28)
-#define KD_EPERM (33)
-#define KD_ETIMEDOUT (36)
-#define KD_EILSEQ (19)
-#endif
-
-/** Map types from OMX standard types only here so interface is as generic as possible. */
-typedef OMX_U32 CPresult;
-typedef char * CPstring;
-typedef void * CPhandle;
-typedef OMX_U32 CPuint;
-typedef OMX_S32 CPint;
-typedef char CPbyte;
-typedef OMX_BOOL CPbool;
-
-/** enumeration of origin types used in the CP_PIPETYPE's Seek function
- * @ingroup cp
- */
-typedef enum CP_ORIGINTYPE {
- CP_OriginBegin,
- CP_OriginCur,
- CP_OriginEnd,
- CP_OriginKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- CP_OriginVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- CP_OriginMax = 0X7FFFFFFF
-} CP_ORIGINTYPE;
-
-/** enumeration of contact access types used in the CP_PIPETYPE's Open function
- * @ingroup cp
- */
-typedef enum CP_ACCESSTYPE {
- CP_AccessRead,
- CP_AccessWrite,
- CP_AccessReadWrite ,
- CP_AccessKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- CP_AccessVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- CP_AccessMax = 0X7FFFFFFF
-} CP_ACCESSTYPE;
-
-/** enumeration of results returned by the CP_PIPETYPE's CheckAvailableBytes function
- * @ingroup cp
- */
-typedef enum CP_CHECKBYTESRESULTTYPE
-{
- CP_CheckBytesOk, /**< There are at least the request number
- of bytes available */
- CP_CheckBytesNotReady, /**< The pipe is still retrieving bytes
- and presently lacks sufficient bytes.
- Client will be called when they are
- sufficient bytes are available. */
- CP_CheckBytesInsufficientBytes , /**< The pipe has retrieved all bytes
- but those available are less than those
- requested */
- CP_CheckBytesAtEndOfStream, /**< The pipe has reached the end of stream
- and no more bytes are available. */
- CP_CheckBytesOutOfBuffers, /**< All read/write buffers are currently in use. */
- CP_CheckBytesKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- CP_CheckBytesVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- CP_CheckBytesMax = 0X7FFFFFFF
-} CP_CHECKBYTESRESULTTYPE;
-
-/** enumeration of content pipe events sent to the client callback.
- * @ingroup cp
- */
-typedef enum CP_EVENTTYPE{
- CP_BytesAvailable, /** bytes requested in a CheckAvailableBytes call are now available*/
- CP_Overflow, /** enumeration of content pipe events sent to the client callback*/
- CP_PipeDisconnected , /** enumeration of content pipe events sent to the client callback*/
- CP_EventKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- CP_EventVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- CP_EventMax = 0X7FFFFFFF
-} CP_EVENTTYPE;
-
-/** content pipe definition
- * @ingroup cp
- */
-typedef struct CP_PIPETYPE
-{
- /** Open a content stream for reading or writing. */
- CPresult (*Open)( CPhandle* hContent, CPstring szURI, CP_ACCESSTYPE eAccess );
-
- /** Close a content stream. */
- CPresult (*Close)( CPhandle hContent );
-
- /** Create a content source and open it for writing. */
- CPresult (*Create)( CPhandle *hContent, CPstring szURI );
-
- /** Check the that specified number of bytes are available for reading or writing (depending on access type).*/
- CPresult (*CheckAvailableBytes)( CPhandle hContent, CPuint nBytesRequested, CP_CHECKBYTESRESULTTYPE *eResult );
-
- /** Seek to certain position in the content relative to the specified origin. */
- CPresult (*SetPosition)( CPhandle hContent, CPint nOffset, CP_ORIGINTYPE eOrigin);
-
- /** Retrieve the current position relative to the start of the content. */
- CPresult (*GetPosition)( CPhandle hContent, CPuint *pPosition);
-
- /** Retrieve data of the specified size from the content stream (advance content pointer by size of data).
- Note: pipe client provides pointer. This function is appropriate for small high frequency reads. */
- CPresult (*Read)( CPhandle hContent, CPbyte *pData, CPuint nSize);
-
- /** Retrieve a buffer allocated by the pipe that contains the requested number of bytes.
- Buffer contains the next block of bytes, as specified by nSize, of the content. nSize also
- returns the size of the block actually read. Content pointer advances the by the returned size.
- Note: pipe provides pointer. This function is appropriate for large reads. The client must call
- ReleaseReadBuffer when done with buffer.
-
- In some cases the requested block may not reside in contiguous memory within the
- pipe implementation. For instance if the pipe leverages a circular buffer then the requested
- block may straddle the boundary of the circular buffer. By default a pipe implementation
- performs a copy in this case to provide the block to the pipe client in one contiguous buffer.
- If, however, the client sets bForbidCopy, then the pipe returns only those bytes preceding the memory
- boundary. Here the client may retrieve the data in segments over successive calls. */
- CPresult (*ReadBuffer)( CPhandle hContent, CPbyte **ppBuffer, CPuint *nSize, CPbool bForbidCopy);
-
- /** Release a buffer obtained by ReadBuffer back to the pipe. */
- CPresult (*ReleaseReadBuffer)(CPhandle hContent, CPbyte *pBuffer);
-
- /** Write data of the specified size to the content (advance content pointer by size of data).
- Note: pipe client provides pointer. This function is appropriate for small high frequency writes. */
- CPresult (*Write)( CPhandle hContent, CPbyte *data, CPuint nSize);
-
- /** Retrieve a buffer allocated by the pipe used to write data to the content.
- Client will fill buffer with output data. Note: pipe provides pointer. This function is appropriate
- for large writes. The client must call WriteBuffer when done it has filled the buffer with data.*/
- CPresult (*GetWriteBuffer)( CPhandle hContent, CPbyte **ppBuffer, CPuint nSize);
-
- /** Deliver a buffer obtained via GetWriteBuffer to the pipe. Pipe will write the
- the contents of the buffer to content and advance content pointer by the size of the buffer */
- CPresult (*WriteBuffer)( CPhandle hContent, CPbyte *pBuffer, CPuint nFilledSize);
-
- /** Register a per-handle client callback with the content pipe. */
- CPresult (*RegisterCallback)( CPhandle hContent, CPresult (*ClientCallback)(CP_EVENTTYPE eEvent, CPuint iParam));
-
-} CP_PIPETYPE;
-
-#endif
-
diff --git a/include/media/stagefright/openmax/OMX_Core.h b/include/media/stagefright/openmax/OMX_Core.h
deleted file mode 100644
index 9fb0f6f..0000000
--- a/include/media/stagefright/openmax/OMX_Core.h
+++ /dev/null
@@ -1,1448 +0,0 @@
-/* ------------------------------------------------------------------
- * Copyright (C) 1998-2009 PacketVideo
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either
- * express or implied.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- * -------------------------------------------------------------------
- */
-/*
- * Copyright (c) 2008 The Khronos Group Inc.
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files (the
- * "Software"), to deal in the Software without restriction, including
- * without limitation the rights to use, copy, modify, merge, publish,
- * distribute, sublicense, and/or sell copies of the Software, and to
- * permit persons to whom the Software is furnished to do so, subject
- * to the following conditions:
- * The above copyright notice and this permission notice shall be included
- * in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS
- * OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY
- * CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
- * TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
- * SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- *
- */
-
-/** OMX_Core.h - OpenMax IL version 1.1.2
- * The OMX_Core header file contains the definitions used by both the
- * application and the component to access common items.
- */
-
-#ifndef OMX_Core_h
-#define OMX_Core_h
-
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-
-/* Each OMX header shall include all required header files to allow the
- * header to compile without errors. The includes below are required
- * for this header file to compile successfully
- */
-
-#include <OMX_Index.h>
-
-
-/** The OMX_COMMANDTYPE enumeration is used to specify the action in the
- * OMX_SendCommand macro.
- * @ingroup core
- */
-typedef enum OMX_COMMANDTYPE
-{
- OMX_CommandStateSet, /**< Change the component state */
- OMX_CommandFlush, /**< Flush the data queue(s) of a component */
- OMX_CommandPortDisable, /**< Disable a port on a component. */
- OMX_CommandPortEnable, /**< Enable a port on a component. */
- OMX_CommandMarkBuffer, /**< Mark a component/buffer for observation */
- OMX_CommandKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_CommandVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_CommandMax = 0X7FFFFFFF
-} OMX_COMMANDTYPE;
-
-
-
-/** The OMX_STATETYPE enumeration is used to indicate or change the component
- * state. This enumeration reflects the current state of the component when
- * used with the OMX_GetState macro or becomes the parameter in a state change
- * command when used with the OMX_SendCommand macro.
- *
- * The component will be in the Loaded state after the component is initially
- * loaded into memory. In the Loaded state, the component is not allowed to
- * allocate or hold resources other than to build it's internal parameter
- * and configuration tables. The application will send one or more
- * SetParameters/GetParameters and SetConfig/GetConfig commands to the
- * component and the component will record each of these parameter and
- * configuration changes for use later. When the application sends the
- * Idle command, the component will acquire the resources needed for the
- * specified configuration and will transition to the idle state if the
- * allocation is successful. If the component cannot successfully
- * transition to the idle state for any reason, the state of the component
- * shall be fully rolled back to the Loaded state (e.g. all allocated
- * resources shall be released). When the component receives the command
- * to go to the Executing state, it shall begin processing buffers by
- * sending all input buffers it holds to the application. While
- * the component is in the Idle state, the application may also send the
- * Pause command. If the component receives the pause command while in the
- * Idle state, the component shall send all input buffers it holds to the
- * application, but shall not begin processing buffers. This will allow the
- * application to prefill buffers.
- *
- * @ingroup comp
- */
-
-typedef enum OMX_STATETYPE
-{
- OMX_StateInvalid, /**< component has detected that it's internal data
- structures are corrupted to the point that
- it cannot determine it's state properly */
- OMX_StateLoaded, /**< component has been loaded but has not completed
- initialization. The OMX_SetParameter macro
- and the OMX_GetParameter macro are the only
- valid macros allowed to be sent to the
- component in this state. */
- OMX_StateIdle, /**< component initialization has been completed
- successfully and the component is ready to
- to start. */
- OMX_StateExecuting, /**< component has accepted the start command and
- is processing data (if data is available) */
- OMX_StatePause, /**< component has received pause command */
- OMX_StateWaitForResources, /**< component is waiting for resources, either after
- preemption or before it gets the resources requested.
- See specification for complete details. */
- OMX_StateKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_StateVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_StateMax = 0X7FFFFFFF
-} OMX_STATETYPE;
-
-/** The OMX_ERRORTYPE enumeration defines the standard OMX Errors. These
- * errors should cover most of the common failure cases. However,
- * vendors are free to add additional error messages of their own as
- * long as they follow these rules:
- * 1. Vendor error messages shall be in the range of 0x90000000 to
- * 0x9000FFFF.
- * 2. Vendor error messages shall be defined in a header file provided
- * with the component. No error messages are allowed that are
- * not defined.
- */
-typedef enum OMX_ERRORTYPE
-{
- OMX_ErrorNone = 0,
-
- /** There were insufficient resources to perform the requested operation */
- OMX_ErrorInsufficientResources = (OMX_S32) 0x80001000,
-
- /** There was an error, but the cause of the error could not be determined */
- OMX_ErrorUndefined = (OMX_S32) 0x80001001,
-
- /** The component name string was not valid */
- OMX_ErrorInvalidComponentName = (OMX_S32) 0x80001002,
-
- /** No component with the specified name string was found */
- OMX_ErrorComponentNotFound = (OMX_S32) 0x80001003,
-
- /** The component specified did not have a "OMX_ComponentInit" or
- "OMX_ComponentDeInit entry point */
- OMX_ErrorInvalidComponent = (OMX_S32) 0x80001004,
-
- /** One or more parameters were not valid */
- OMX_ErrorBadParameter = (OMX_S32) 0x80001005,
-
- /** The requested function is not implemented */
- OMX_ErrorNotImplemented = (OMX_S32) 0x80001006,
-
- /** The buffer was emptied before the next buffer was ready */
- OMX_ErrorUnderflow = (OMX_S32) 0x80001007,
-
- /** The buffer was not available when it was needed */
- OMX_ErrorOverflow = (OMX_S32) 0x80001008,
-
- /** The hardware failed to respond as expected */
- OMX_ErrorHardware = (OMX_S32) 0x80001009,
-
- /** The component is in the state OMX_StateInvalid */
- OMX_ErrorInvalidState = (OMX_S32) 0x8000100A,
-
- /** Stream is found to be corrupt */
- OMX_ErrorStreamCorrupt = (OMX_S32) 0x8000100B,
-
- /** Ports being connected are not compatible */
- OMX_ErrorPortsNotCompatible = (OMX_S32) 0x8000100C,
-
- /** Resources allocated to an idle component have been
- lost resulting in the component returning to the loaded state */
- OMX_ErrorResourcesLost = (OMX_S32) 0x8000100D,
-
- /** No more indicies can be enumerated */
- OMX_ErrorNoMore = (OMX_S32) 0x8000100E,
-
- /** The component detected a version mismatch */
- OMX_ErrorVersionMismatch = (OMX_S32) 0x8000100F,
-
- /** The component is not ready to return data at this time */
- OMX_ErrorNotReady = (OMX_S32) 0x80001010,
-
- /** There was a timeout that occurred */
- OMX_ErrorTimeout = (OMX_S32) 0x80001011,
-
- /** This error occurs when trying to transition into the state you are already in */
- OMX_ErrorSameState = (OMX_S32) 0x80001012,
-
- /** Resources allocated to an executing or paused component have been
- preempted, causing the component to return to the idle state */
- OMX_ErrorResourcesPreempted = (OMX_S32) 0x80001013,
-
- /** A non-supplier port sends this error to the IL client (via the EventHandler callback)
- during the allocation of buffers (on a transition from the LOADED to the IDLE state or
- on a port restart) when it deems that it has waited an unusually long time for the supplier
- to send it an allocated buffer via a UseBuffer call. */
- OMX_ErrorPortUnresponsiveDuringAllocation = (OMX_S32) 0x80001014,
-
- /** A non-supplier port sends this error to the IL client (via the EventHandler callback)
- during the deallocation of buffers (on a transition from the IDLE to LOADED state or
- on a port stop) when it deems that it has waited an unusually long time for the supplier
- to request the deallocation of a buffer header via a FreeBuffer call. */
- OMX_ErrorPortUnresponsiveDuringDeallocation = (OMX_S32) 0x80001015,
-
- /** A supplier port sends this error to the IL client (via the EventHandler callback)
- during the stopping of a port (either on a transition from the IDLE to LOADED
- state or a port stop) when it deems that it has waited an unusually long time for
- the non-supplier to return a buffer via an EmptyThisBuffer or FillThisBuffer call. */
- OMX_ErrorPortUnresponsiveDuringStop = (OMX_S32) 0x80001016,
-
- /** Attempting a state transtion that is not allowed */
- OMX_ErrorIncorrectStateTransition = (OMX_S32) 0x80001017,
-
- /* Attempting a command that is not allowed during the present state. */
- OMX_ErrorIncorrectStateOperation = (OMX_S32) 0x80001018,
-
- /** The values encapsulated in the parameter or config structure are not supported. */
- OMX_ErrorUnsupportedSetting = (OMX_S32) 0x80001019,
-
- /** The parameter or config indicated by the given index is not supported. */
- OMX_ErrorUnsupportedIndex = (OMX_S32) 0x8000101A,
-
- /** The port index supplied is incorrect. */
- OMX_ErrorBadPortIndex = (OMX_S32) 0x8000101B,
-
- /** The port has lost one or more of its buffers and it thus unpopulated. */
- OMX_ErrorPortUnpopulated = (OMX_S32) 0x8000101C,
-
- /** Component suspended due to temporary loss of resources */
- OMX_ErrorComponentSuspended = (OMX_S32) 0x8000101D,
-
- /** Component suspended due to an inability to acquire dynamic resources */
- OMX_ErrorDynamicResourcesUnavailable = (OMX_S32) 0x8000101E,
-
- /** When the macroblock error reporting is enabled the component returns new error
- for every frame that has errors */
- OMX_ErrorMbErrorsInFrame = (OMX_S32) 0x8000101F,
-
- /** A component reports this error when it cannot parse or determine the format of an input stream. */
- OMX_ErrorFormatNotDetected = (OMX_S32) 0x80001020,
-
- /** The content open operation failed. */
- OMX_ErrorContentPipeOpenFailed = (OMX_S32) 0x80001021,
-
- /** The content creation operation failed. */
- OMX_ErrorContentPipeCreationFailed = (OMX_S32) 0x80001022,
-
- /** Separate table information is being used */
- OMX_ErrorSeperateTablesUsed = (OMX_S32) 0x80001023,
-
- /** Tunneling is unsupported by the component*/
- OMX_ErrorTunnelingUnsupported = (OMX_S32) 0x80001024,
-
- OMX_ErrorKhronosExtensions = (OMX_S32)0x8F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_ErrorVendorStartUnused = (OMX_S32)0x90000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_ErrorMax = 0x7FFFFFFF
-} OMX_ERRORTYPE;
-
-/** @ingroup core */
-typedef OMX_ERRORTYPE (* OMX_COMPONENTINITTYPE)(OMX_IN OMX_HANDLETYPE hComponent);
-
-/** @ingroup core */
-typedef struct OMX_COMPONENTREGISTERTYPE
-{
- const char * pName; /* Component name, 128 byte limit (including '\0') applies */
- OMX_COMPONENTINITTYPE pInitialize; /* Component instance initialization function */
-} OMX_COMPONENTREGISTERTYPE;
-
-/** @ingroup core */
-extern OMX_COMPONENTREGISTERTYPE OMX_ComponentRegistered[];
-
-/** @ingroup rpm */
-typedef struct OMX_PRIORITYMGMTTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nGroupPriority; /**< Priority of the component group */
- OMX_U32 nGroupID; /**< ID of the component group */
-} OMX_PRIORITYMGMTTYPE;
-
-/* Component name and Role names are limited to 128 characters including the terminating '\0'. */
-#define OMX_MAX_STRINGNAME_SIZE 128
-
-/** @ingroup comp */
-typedef struct OMX_PARAM_COMPONENTROLETYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U8 cRole[OMX_MAX_STRINGNAME_SIZE]; /**< name of standard component which defines component role */
-} OMX_PARAM_COMPONENTROLETYPE;
-
-/** End of Stream Buffer Flag:
- *
- * A component sets EOS when it has no more data to emit on a particular
- * output port. Thus an output port shall set EOS on the last buffer it
- * emits. A component's determination of when an output port should
- * cease sending data is implemenation specific.
- * @ingroup buf
- */
-
-#define OMX_BUFFERFLAG_EOS 0x00000001
-
-/** Start Time Buffer Flag:
- *
- * The source of a stream (e.g. a demux component) sets the STARTTIME
- * flag on the buffer that contains the starting timestamp for the
- * stream. The starting timestamp corresponds to the first data that
- * should be displayed at startup or after a seek.
- * The first timestamp of the stream is not necessarily the start time.
- * For instance, in the case of a seek to a particular video frame,
- * the target frame may be an interframe. Thus the first buffer of
- * the stream will be the intra-frame preceding the target frame and
- * the starttime will occur with the target frame (with any other
- * required frames required to reconstruct the target intervening).
- *
- * The STARTTIME flag is directly associated with the buffer's
- * timestamp ' thus its association to buffer data and its
- * propagation is identical to the timestamp's.
- *
- * When a Sync Component client receives a buffer with the
- * STARTTIME flag it shall perform a SetConfig on its sync port
- * using OMX_ConfigTimeClientStartTime and passing the buffer's
- * timestamp.
- *
- * @ingroup buf
- */
-
-#define OMX_BUFFERFLAG_STARTTIME 0x00000002
-
-
-
-/** Decode Only Buffer Flag:
- *
- * The source of a stream (e.g. a demux component) sets the DECODEONLY
- * flag on any buffer that should shall be decoded but should not be
- * displayed. This flag is used, for instance, when a source seeks to
- * a target interframe that requires the decode of frames preceding the
- * target to facilitate the target's reconstruction. In this case the
- * source would emit the frames preceding the target downstream
- * but mark them as decode only.
- *
- * The DECODEONLY is associated with buffer data and propagated in a
- * manner identical to the buffer timestamp.
- *
- * A component that renders data should ignore all buffers with
- * the DECODEONLY flag set.
- *
- * @ingroup buf
- */
-
-#define OMX_BUFFERFLAG_DECODEONLY 0x00000004
-
-
-/* Data Corrupt Flag: This flag is set when the IL client believes the data in the associated buffer is corrupt
- * @ingroup buf
- */
-
-#define OMX_BUFFERFLAG_DATACORRUPT 0x00000008
-
-/* End of Frame: The buffer contains exactly one end of frame and no data
- * occurs after the end of frame. This flag is an optional hint. The absence
- * of this flag does not imply the absence of an end of frame within the buffer.
- * @ingroup buf
-*/
-#define OMX_BUFFERFLAG_ENDOFFRAME 0x00000010
-
-/* Sync Frame Flag: This flag is set when the buffer content contains a coded sync frame '
- * a frame that has no dependency on any other frame information
- * @ingroup buf
- */
-#define OMX_BUFFERFLAG_SYNCFRAME 0x00000020
-
-/* Extra data present flag: there is extra data appended to the data stream
- * residing in the buffer
- * @ingroup buf
- */
-#define OMX_BUFFERFLAG_EXTRADATA 0x00000040
-
-/** Codec Config Buffer Flag:
-* OMX_BUFFERFLAG_CODECCONFIG is an optional flag that is set by an
-* output port when all bytes in the buffer form part or all of a set of
-* codec specific configuration data. Examples include SPS/PPS nal units
-* for OMX_VIDEO_CodingAVC or AudioSpecificConfig data for
-* OMX_AUDIO_CodingAAC. Any component that for a given stream sets
-* OMX_BUFFERFLAG_CODECCONFIG shall not mix codec configuration bytes
-* with frame data in the same buffer, and shall send all buffers
-* containing codec configuration bytes before any buffers containing
-* frame data that those configurations bytes describe.
-* If the stream format for a particular codec has a frame specific
-* header at the start of each frame, for example OMX_AUDIO_CodingMP3 or
-* OMX_AUDIO_CodingAAC in ADTS mode, then these shall be presented as
-* normal without setting OMX_BUFFERFLAG_CODECCONFIG.
- * @ingroup buf
- */
-#define OMX_BUFFERFLAG_CODECCONFIG 0x00000080
-
-
-
-/** @ingroup buf */
-typedef struct OMX_BUFFERHEADERTYPE
-{
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U8* pBuffer; /**< Pointer to actual block of memory
- that is acting as the buffer */
- OMX_U32 nAllocLen; /**< size of the buffer allocated, in bytes */
- OMX_U32 nFilledLen; /**< number of bytes currently in the
- buffer */
- OMX_U32 nOffset; /**< start offset of valid data in bytes from
- the start of the buffer */
- OMX_PTR pAppPrivate; /**< pointer to any data the application
- wants to associate with this buffer */
- OMX_PTR pPlatformPrivate; /**< pointer to any data the platform
- wants to associate with this buffer */
- OMX_PTR pInputPortPrivate; /**< pointer to any data the input port
- wants to associate with this buffer */
- OMX_PTR pOutputPortPrivate; /**< pointer to any data the output port
- wants to associate with this buffer */
- OMX_HANDLETYPE hMarkTargetComponent; /**< The component that will generate a
- mark event upon processing this buffer. */
- OMX_PTR pMarkData; /**< Application specific data associated with
- the mark sent on a mark event to disambiguate
- this mark from others. */
- OMX_U32 nTickCount; /**< Optional entry that the component and
- application can update with a tick count
- when they access the component. This
- value should be in microseconds. Since
- this is a value relative to an arbitrary
- starting point, this value cannot be used
- to determine absolute time. This is an
- optional entry and not all components
- will update it.*/
- OMX_TICKS nTimeStamp; /**< Timestamp corresponding to the sample
- starting at the first logical sample
- boundary in the buffer. Timestamps of
- successive samples within the buffer may
- be inferred by adding the duration of the
- of the preceding buffer to the timestamp
- of the preceding buffer.*/
- OMX_U32 nFlags; /**< buffer specific flags */
- OMX_U32 nOutputPortIndex; /**< The index of the output port (if any) using
- this buffer */
- OMX_U32 nInputPortIndex; /**< The index of the input port (if any) using
- this buffer */
-} OMX_BUFFERHEADERTYPE;
-
-/** The OMX_EXTRADATATYPE enumeration is used to define the
- * possible extra data payload types.
- * NB: this enum is binary backwards compatible with the previous
- * OMX_EXTRADATA_QUANT define. This should be replaced with
- * OMX_ExtraDataQuantization.
- */
-typedef enum OMX_EXTRADATATYPE
-{
- OMX_ExtraDataNone = 0, /**< Indicates that no more extra data sections follow */
- OMX_ExtraDataQuantization, /**< The data payload contains quantization data */
- OMX_ExtraDataKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_ExtraDataVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_ExtraDataMax = 0x7FFFFFFF
-} OMX_EXTRADATATYPE;
-
-
-typedef struct OMX_OTHER_EXTRADATATYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_EXTRADATATYPE eType; /* Extra Data type */
- OMX_U32 nDataSize; /* Size of the supporting data to follow */
- OMX_U8 data[1]; /* Supporting data hint */
-} OMX_OTHER_EXTRADATATYPE;
-
-/** @ingroup comp */
-typedef struct OMX_PORT_PARAM_TYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPorts; /**< The number of ports for this component */
- OMX_U32 nStartPortNumber; /** first port number for this type of port */
-} OMX_PORT_PARAM_TYPE;
-
-/** @ingroup comp */
-typedef enum OMX_EVENTTYPE
-{
- OMX_EventCmdComplete, /**< component has sucessfully completed a command */
- OMX_EventError, /**< component has detected an error condition */
- OMX_EventMark, /**< component has detected a buffer mark */
- OMX_EventPortSettingsChanged, /**< component is reported a port settings change */
- OMX_EventBufferFlag, /**< component has detected an EOS */
- OMX_EventResourcesAcquired, /**< component has been granted resources and is
- automatically starting the state change from
- OMX_StateWaitForResources to OMX_StateIdle. */
- OMX_EventComponentResumed, /**< Component resumed due to reacquisition of resources */
- OMX_EventDynamicResourcesAvailable, /**< Component has acquired previously unavailable dynamic resources */
- OMX_EventPortFormatDetected, /**< Component has detected a supported format. */
- OMX_EventKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_EventVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_EventMax = 0x7FFFFFFF
-} OMX_EVENTTYPE;
-
-typedef struct OMX_CALLBACKTYPE
-{
- /** The EventHandler method is used to notify the application when an
- event of interest occurs. Events are defined in the OMX_EVENTTYPE
- enumeration. Please see that enumeration for details of what will
- be returned for each type of event. Callbacks should not return
- an error to the component, so if an error occurs, the application
- shall handle it internally. This is a blocking call.
-
- The application should return from this call within 5 msec to avoid
- blocking the component for an excessively long period of time.
-
- @param hComponent
- handle of the component to access. This is the component
- handle returned by the call to the GetHandle function.
- @param pAppData
- pointer to an application defined value that was provided in the
- pAppData parameter to the OMX_GetHandle method for the component.
- This application defined value is provided so that the application
- can have a component specific context when receiving the callback.
- @param eEvent
- Event that the component wants to notify the application about.
- @param nData1
- nData will be the OMX_ERRORTYPE for an error event and will be
- an OMX_COMMANDTYPE for a command complete event and OMX_INDEXTYPE for a OMX_PortSettingsChanged event.
- @param nData2
- nData2 will hold further information related to the event. Can be OMX_STATETYPE for
- a OMX_CommandStateSet command or port index for a OMX_PortSettingsChanged event.
- Default value is 0 if not used. )
- @param pEventData
- Pointer to additional event-specific data (see spec for meaning).
- */
-
- OMX_ERRORTYPE (*EventHandler)(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_IN OMX_PTR pAppData,
- OMX_IN OMX_EVENTTYPE eEvent,
- OMX_IN OMX_U32 nData1,
- OMX_IN OMX_U32 nData2,
- OMX_IN OMX_PTR pEventData);
-
- /** The EmptyBufferDone method is used to return emptied buffers from an
- input port back to the application for reuse. This is a blocking call
- so the application should not attempt to refill the buffers during this
- call, but should queue them and refill them in another thread. There
- is no error return, so the application shall handle any errors generated
- internally.
-
- The application should return from this call within 5 msec.
-
- @param hComponent
- handle of the component to access. This is the component
- handle returned by the call to the GetHandle function.
- @param pAppData
- pointer to an application defined value that was provided in the
- pAppData parameter to the OMX_GetHandle method for the component.
- This application defined value is provided so that the application
- can have a component specific context when receiving the callback.
- @param pBuffer
- pointer to an OMX_BUFFERHEADERTYPE structure allocated with UseBuffer
- or AllocateBuffer indicating the buffer that was emptied.
- @ingroup buf
- */
- OMX_ERRORTYPE (*EmptyBufferDone)(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_IN OMX_PTR pAppData,
- OMX_IN OMX_BUFFERHEADERTYPE* pBuffer);
-
- /** The FillBufferDone method is used to return filled buffers from an
- output port back to the application for emptying and then reuse.
- This is a blocking call so the application should not attempt to
- empty the buffers during this call, but should queue the buffers
- and empty them in another thread. There is no error return, so
- the application shall handle any errors generated internally. The
- application shall also update the buffer header to indicate the
- number of bytes placed into the buffer.
-
- The application should return from this call within 5 msec.
-
- @param hComponent
- handle of the component to access. This is the component
- handle returned by the call to the GetHandle function.
- @param pAppData
- pointer to an application defined value that was provided in the
- pAppData parameter to the OMX_GetHandle method for the component.
- This application defined value is provided so that the application
- can have a component specific context when receiving the callback.
- @param pBuffer
- pointer to an OMX_BUFFERHEADERTYPE structure allocated with UseBuffer
- or AllocateBuffer indicating the buffer that was filled.
- @ingroup buf
- */
- OMX_ERRORTYPE (*FillBufferDone)(
- OMX_OUT OMX_HANDLETYPE hComponent,
- OMX_OUT OMX_PTR pAppData,
- OMX_OUT OMX_BUFFERHEADERTYPE* pBuffer);
-
-} OMX_CALLBACKTYPE;
-
-/** The OMX_BUFFERSUPPLIERTYPE enumeration is used to dictate port supplier
- preference when tunneling between two ports.
- @ingroup tun buf
-*/
-typedef enum OMX_BUFFERSUPPLIERTYPE
-{
- OMX_BufferSupplyUnspecified = 0x0, /**< port supplying the buffers is unspecified,
- or don't care */
- OMX_BufferSupplyInput, /**< input port supplies the buffers */
- OMX_BufferSupplyOutput, /**< output port supplies the buffers */
- OMX_BufferSupplyKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_BufferSupplyVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_BufferSupplyMax = 0x7FFFFFFF
-} OMX_BUFFERSUPPLIERTYPE;
-
-
-/** buffer supplier parameter
- * @ingroup tun
- */
-typedef struct OMX_PARAM_BUFFERSUPPLIERTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_BUFFERSUPPLIERTYPE eBufferSupplier; /**< buffer supplier */
-} OMX_PARAM_BUFFERSUPPLIERTYPE;
-
-
-/**< indicates that buffers received by an input port of a tunnel
- may not modify the data in the buffers
- @ingroup tun
- */
-#define OMX_PORTTUNNELFLAG_READONLY 0x00000001
-
-
-/** The OMX_TUNNELSETUPTYPE structure is used to pass data from an output
- port to an input port as part the two ComponentTunnelRequest calls
- resulting from a OMX_SetupTunnel call from the IL Client.
- @ingroup tun
- */
-typedef struct OMX_TUNNELSETUPTYPE
-{
- OMX_U32 nTunnelFlags; /**< bit flags for tunneling */
- OMX_BUFFERSUPPLIERTYPE eSupplier; /**< supplier preference */
-} OMX_TUNNELSETUPTYPE;
-
-/* OMX Component headers is included to enable the core to use
- macros for functions into the component for OMX release 1.0.
- Developers should not access any structures or data from within
- the component header directly */
-/* TO BE REMOVED - #include <OMX_Component.h> */
-
-/** GetComponentVersion will return information about the component.
- This is a blocking call. This macro will go directly from the
- application to the component (via a core macro). The
- component will return from this call within 5 msec.
- @param [in] hComponent
- handle of component to execute the command
- @param [out] pComponentName
- pointer to an empty string of length 128 bytes. The component
- will write its name into this string. The name will be
- terminated by a single zero byte. The name of a component will
- be 127 bytes or less to leave room for the trailing zero byte.
- An example of a valid component name is "OMX.ABC.ChannelMixer\0".
- @param [out] pComponentVersion
- pointer to an OMX Version structure that the component will fill
- in. The component will fill in a value that indicates the
- component version. NOTE: the component version is NOT the same
- as the OMX Specification version (found in all structures). The
- component version is defined by the vendor of the component and
- its value is entirely up to the component vendor.
- @param [out] pSpecVersion
- pointer to an OMX Version structure that the component will fill
- in. The SpecVersion is the version of the specification that the
- component was built against. Please note that this value may or
- may not match the structure's version. For example, if the
- component was built against the 2.0 specification, but the
- application (which creates the structure is built against the
- 1.0 specification the versions would be different.
- @param [out] pComponentUUID
- pointer to the UUID of the component which will be filled in by
- the component. The UUID is a unique identifier that is set at
- RUN time for the component and is unique to each instantion of
- the component.
- @return OMX_ERRORTYPE
- If the command successfully executes, the return code will be
- OMX_ErrorNone. Otherwise the appropriate OMX error will be returned.
- @ingroup comp
- */
-#define OMX_GetComponentVersion( \
- hComponent, \
- pComponentName, \
- pComponentVersion, \
- pSpecVersion, \
- pComponentUUID) \
- ((OMX_COMPONENTTYPE*)hComponent)->GetComponentVersion( \
- hComponent, \
- pComponentName, \
- pComponentVersion, \
- pSpecVersion, \
- pComponentUUID) /* Macro End */
-
-
-/** Send a command to the component. This call is a non-blocking call.
- The component should check the parameters and then queue the command
- to the component thread to be executed. The component thread shall
- send the EventHandler() callback at the conclusion of the command.
- This macro will go directly from the application to the component (via
- a core macro). The component will return from this call within 5 msec.
-
- When the command is "OMX_CommandStateSet" the component will queue a
- state transition to the new state idenfied in nParam.
-
- When the command is "OMX_CommandFlush", to flush a port's buffer queues,
- the command will force the component to return all buffers NOT CURRENTLY
- BEING PROCESSED to the application, in the order in which the buffers
- were received.
-
- When the command is "OMX_CommandPortDisable" or
- "OMX_CommandPortEnable", the component's port (given by the value of
- nParam) will be stopped or restarted.
-
- When the command "OMX_CommandMarkBuffer" is used to mark a buffer, the
- pCmdData will point to a OMX_MARKTYPE structure containing the component
- handle of the component to examine the buffer chain for the mark. nParam1
- contains the index of the port on which the buffer mark is applied.
-
- Specification text for more details.
-
- @param [in] hComponent
- handle of component to execute the command
- @param [in] Cmd
- Command for the component to execute
- @param [in] nParam
- Parameter for the command to be executed. When Cmd has the value
- OMX_CommandStateSet, value is a member of OMX_STATETYPE. When Cmd has
- the value OMX_CommandFlush, value of nParam indicates which port(s)
- to flush. -1 is used to flush all ports a single port index will
- only flush that port. When Cmd has the value "OMX_CommandPortDisable"
- or "OMX_CommandPortEnable", the component's port is given by
- the value of nParam. When Cmd has the value "OMX_CommandMarkBuffer"
- the components pot is given by the value of nParam.
- @param [in] pCmdData
- Parameter pointing to the OMX_MARKTYPE structure when Cmd has the value
- "OMX_CommandMarkBuffer".
- @return OMX_ERRORTYPE
- If the command successfully executes, the return code will be
- OMX_ErrorNone. Otherwise the appropriate OMX error will be returned.
- @ingroup comp
- */
-#define OMX_SendCommand( \
- hComponent, \
- Cmd, \
- nParam, \
- pCmdData) \
- ((OMX_COMPONENTTYPE*)hComponent)->SendCommand( \
- hComponent, \
- Cmd, \
- nParam, \
- pCmdData) /* Macro End */
-
-
-/** The OMX_GetParameter macro will get one of the current parameter
- settings from the component. This macro cannot only be invoked when
- the component is in the OMX_StateInvalid state. The nParamIndex
- parameter is used to indicate which structure is being requested from
- the component. The application shall allocate the correct structure
- and shall fill in the structure size and version information before
- invoking this macro. When the parameter applies to a port, the
- caller shall fill in the appropriate nPortIndex value indicating the
- port on which the parameter applies. If the component has not had
- any settings changed, then the component should return a set of
- valid DEFAULT parameters for the component. This is a blocking
- call.
-
- The component should return from this call within 20 msec.
-
- @param [in] hComponent
- Handle of the component to be accessed. This is the component
- handle returned by the call to the OMX_GetHandle function.
- @param [in] nParamIndex
- Index of the structure to be filled. This value is from the
- OMX_INDEXTYPE enumeration.
- @param [in,out] pComponentParameterStructure
- Pointer to application allocated structure to be filled by the
- component.
- @return OMX_ERRORTYPE
- If the command successfully executes, the return code will be
- OMX_ErrorNone. Otherwise the appropriate OMX error will be returned.
- @ingroup comp
- */
-#define OMX_GetParameter( \
- hComponent, \
- nParamIndex, \
- pComponentParameterStructure) \
- ((OMX_COMPONENTTYPE*)hComponent)->GetParameter( \
- hComponent, \
- nParamIndex, \
- pComponentParameterStructure) /* Macro End */
-
-
-/** The OMX_SetParameter macro will send an initialization parameter
- structure to a component. Each structure shall be sent one at a time,
- in a separate invocation of the macro. This macro can only be
- invoked when the component is in the OMX_StateLoaded state, or the
- port is disabled (when the parameter applies to a port). The
- nParamIndex parameter is used to indicate which structure is being
- passed to the component. The application shall allocate the
- correct structure and shall fill in the structure size and version
- information (as well as the actual data) before invoking this macro.
- The application is free to dispose of this structure after the call
- as the component is required to copy any data it shall retain. This
- is a blocking call.
-
- The component should return from this call within 20 msec.
-
- @param [in] hComponent
- Handle of the component to be accessed. This is the component
- handle returned by the call to the OMX_GetHandle function.
- @param [in] nIndex
- Index of the structure to be sent. This value is from the
- OMX_INDEXTYPE enumeration.
- @param [in] pComponentParameterStructure
- pointer to application allocated structure to be used for
- initialization by the component.
- @return OMX_ERRORTYPE
- If the command successfully executes, the return code will be
- OMX_ErrorNone. Otherwise the appropriate OMX error will be returned.
- @ingroup comp
- */
-#define OMX_SetParameter( \
- hComponent, \
- nParamIndex, \
- pComponentParameterStructure) \
- ((OMX_COMPONENTTYPE*)hComponent)->SetParameter( \
- hComponent, \
- nParamIndex, \
- pComponentParameterStructure) /* Macro End */
-
-
-/** The OMX_GetConfig macro will get one of the configuration structures
- from a component. This macro can be invoked anytime after the
- component has been loaded. The nParamIndex call parameter is used to
- indicate which structure is being requested from the component. The
- application shall allocate the correct structure and shall fill in the
- structure size and version information before invoking this macro.
- If the component has not had this configuration parameter sent before,
- then the component should return a set of valid DEFAULT values for the
- component. This is a blocking call.
-
- The component should return from this call within 5 msec.
-
- @param [in] hComponent
- Handle of the component to be accessed. This is the component
- handle returned by the call to the OMX_GetHandle function.
- @param [in] nIndex
- Index of the structure to be filled. This value is from the
- OMX_INDEXTYPE enumeration.
- @param [in,out] pComponentConfigStructure
- pointer to application allocated structure to be filled by the
- component.
- @return OMX_ERRORTYPE
- If the command successfully executes, the return code will be
- OMX_ErrorNone. Otherwise the appropriate OMX error will be returned.
- @ingroup comp
-*/
-#define OMX_GetConfig( \
- hComponent, \
- nConfigIndex, \
- pComponentConfigStructure) \
- ((OMX_COMPONENTTYPE*)hComponent)->GetConfig( \
- hComponent, \
- nConfigIndex, \
- pComponentConfigStructure) /* Macro End */
-
-
-/** The OMX_SetConfig macro will send one of the configuration
- structures to a component. Each structure shall be sent one at a time,
- each in a separate invocation of the macro. This macro can be invoked
- anytime after the component has been loaded. The application shall
- allocate the correct structure and shall fill in the structure size
- and version information (as well as the actual data) before invoking
- this macro. The application is free to dispose of this structure after
- the call as the component is required to copy any data it shall retain.
- This is a blocking call.
-
- The component should return from this call within 5 msec.
-
- @param [in] hComponent
- Handle of the component to be accessed. This is the component
- handle returned by the call to the OMX_GetHandle function.
- @param [in] nConfigIndex
- Index of the structure to be sent. This value is from the
- OMX_INDEXTYPE enumeration above.
- @param [in] pComponentConfigStructure
- pointer to application allocated structure to be used for
- initialization by the component.
- @return OMX_ERRORTYPE
- If the command successfully executes, the return code will be
- OMX_ErrorNone. Otherwise the appropriate OMX error will be returned.
- @ingroup comp
- */
-#define OMX_SetConfig( \
- hComponent, \
- nConfigIndex, \
- pComponentConfigStructure) \
- ((OMX_COMPONENTTYPE*)hComponent)->SetConfig( \
- hComponent, \
- nConfigIndex, \
- pComponentConfigStructure) /* Macro End */
-
-
-/** The OMX_GetExtensionIndex macro will invoke a component to translate
- a vendor specific configuration or parameter string into an OMX
- structure index. There is no requirement for the vendor to support
- this command for the indexes already found in the OMX_INDEXTYPE
- enumeration (this is done to save space in small components). The
- component shall support all vendor supplied extension indexes not found
- in the master OMX_INDEXTYPE enumeration. This is a blocking call.
-
- The component should return from this call within 5 msec.
-
- @param [in] hComponent
- Handle of the component to be accessed. This is the component
- handle returned by the call to the GetHandle function.
- @param [in] cParameterName
- OMX_STRING that shall be less than 128 characters long including
- the trailing null byte. This is the string that will get
- translated by the component into a configuration index.
- @param [out] pIndexType
- a pointer to a OMX_INDEXTYPE to receive the index value.
- @return OMX_ERRORTYPE
- If the command successfully executes, the return code will be
- OMX_ErrorNone. Otherwise the appropriate OMX error will be returned.
- @ingroup comp
- */
-#define OMX_GetExtensionIndex( \
- hComponent, \
- cParameterName, \
- pIndexType) \
- ((OMX_COMPONENTTYPE*)hComponent)->GetExtensionIndex( \
- hComponent, \
- cParameterName, \
- pIndexType) /* Macro End */
-
-
-/** The OMX_GetState macro will invoke the component to get the current
- state of the component and place the state value into the location
- pointed to by pState.
-
- The component should return from this call within 5 msec.
-
- @param [in] hComponent
- Handle of the component to be accessed. This is the component
- handle returned by the call to the OMX_GetHandle function.
- @param [out] pState
- pointer to the location to receive the state. The value returned
- is one of the OMX_STATETYPE members
- @return OMX_ERRORTYPE
- If the command successfully executes, the return code will be
- OMX_ErrorNone. Otherwise the appropriate OMX error will be returned.
- @ingroup comp
- */
-#define OMX_GetState( \
- hComponent, \
- pState) \
- ((OMX_COMPONENTTYPE*)hComponent)->GetState( \
- hComponent, \
- pState) /* Macro End */
-
-
-/** The OMX_UseBuffer macro will request that the component use
- a buffer (and allocate its own buffer header) already allocated
- by another component, or by the IL Client. This is a blocking
- call.
-
- The component should return from this call within 20 msec.
-
- @param [in] hComponent
- Handle of the component to be accessed. This is the component
- handle returned by the call to the OMX_GetHandle function.
- @param [out] ppBuffer
- pointer to an OMX_BUFFERHEADERTYPE structure used to receive the
- pointer to the buffer header
- @return OMX_ERRORTYPE
- If the command successfully executes, the return code will be
- OMX_ErrorNone. Otherwise the appropriate OMX error will be returned.
- @ingroup comp buf
- */
-
-#define OMX_UseBuffer( \
- hComponent, \
- ppBufferHdr, \
- nPortIndex, \
- pAppPrivate, \
- nSizeBytes, \
- pBuffer) \
- ((OMX_COMPONENTTYPE*)hComponent)->UseBuffer( \
- hComponent, \
- ppBufferHdr, \
- nPortIndex, \
- pAppPrivate, \
- nSizeBytes, \
- pBuffer)
-
-
-/** The OMX_AllocateBuffer macro will request that the component allocate
- a new buffer and buffer header. The component will allocate the
- buffer and the buffer header and return a pointer to the buffer
- header. This is a blocking call.
-
- The component should return from this call within 5 msec.
-
- @param [in] hComponent
- Handle of the component to be accessed. This is the component
- handle returned by the call to the OMX_GetHandle function.
- @param [out] ppBuffer
- pointer to an OMX_BUFFERHEADERTYPE structure used to receive
- the pointer to the buffer header
- @param [in] nPortIndex
- nPortIndex is used to select the port on the component the buffer will
- be used with. The port can be found by using the nPortIndex
- value as an index into the Port Definition array of the component.
- @param [in] pAppPrivate
- pAppPrivate is used to initialize the pAppPrivate member of the
- buffer header structure.
- @param [in] nSizeBytes
- size of the buffer to allocate. Used when bAllocateNew is true.
- @return OMX_ERRORTYPE
- If the command successfully executes, the return code will be
- OMX_ErrorNone. Otherwise the appropriate OMX error will be returned.
- @ingroup comp buf
- */
-#define OMX_AllocateBuffer( \
- hComponent, \
- ppBuffer, \
- nPortIndex, \
- pAppPrivate, \
- nSizeBytes) \
- ((OMX_COMPONENTTYPE*)hComponent)->AllocateBuffer( \
- hComponent, \
- ppBuffer, \
- nPortIndex, \
- pAppPrivate, \
- nSizeBytes) /* Macro End */
-
-
-/** The OMX_FreeBuffer macro will release a buffer header from the component
- which was allocated using either OMX_AllocateBuffer or OMX_UseBuffer. If
- the component allocated the buffer (see the OMX_UseBuffer macro) then
- the component shall free the buffer and buffer header. This is a
- blocking call.
-
- The component should return from this call within 20 msec.
-
- @param [in] hComponent
- Handle of the component to be accessed. This is the component
- handle returned by the call to the OMX_GetHandle function.
- @param [in] nPortIndex
- nPortIndex is used to select the port on the component the buffer will
- be used with.
- @param [in] pBuffer
- pointer to an OMX_BUFFERHEADERTYPE structure allocated with UseBuffer
- or AllocateBuffer.
- @return OMX_ERRORTYPE
- If the command successfully executes, the return code will be
- OMX_ErrorNone. Otherwise the appropriate OMX error will be returned.
- @ingroup comp buf
- */
-#define OMX_FreeBuffer( \
- hComponent, \
- nPortIndex, \
- pBuffer) \
- ((OMX_COMPONENTTYPE*)hComponent)->FreeBuffer( \
- hComponent, \
- nPortIndex, \
- pBuffer) /* Macro End */
-
-
-/** The OMX_EmptyThisBuffer macro will send a buffer full of data to an
- input port of a component. The buffer will be emptied by the component
- and returned to the application via the EmptyBufferDone call back.
- This is a non-blocking call in that the component will record the buffer
- and return immediately and then empty the buffer, later, at the proper
- time. As expected, this macro may be invoked only while the component
- is in the OMX_StateExecuting. If nPortIndex does not specify an input
- port, the component shall return an error.
-
- The component should return from this call within 5 msec.
-
- @param [in] hComponent
- Handle of the component to be accessed. This is the component
- handle returned by the call to the OMX_GetHandle function.
- @param [in] pBuffer
- pointer to an OMX_BUFFERHEADERTYPE structure allocated with UseBuffer
- or AllocateBuffer.
- @return OMX_ERRORTYPE
- If the command successfully executes, the return code will be
- OMX_ErrorNone. Otherwise the appropriate OMX error will be returned.
- @ingroup comp buf
- */
-#define OMX_EmptyThisBuffer( \
- hComponent, \
- pBuffer) \
- ((OMX_COMPONENTTYPE*)hComponent)->EmptyThisBuffer( \
- hComponent, \
- pBuffer) /* Macro End */
-
-
-/** The OMX_FillThisBuffer macro will send an empty buffer to an
- output port of a component. The buffer will be filled by the component
- and returned to the application via the FillBufferDone call back.
- This is a non-blocking call in that the component will record the buffer
- and return immediately and then fill the buffer, later, at the proper
- time. As expected, this macro may be invoked only while the component
- is in the OMX_ExecutingState. If nPortIndex does not specify an output
- port, the component shall return an error.
-
- The component should return from this call within 5 msec.
-
- @param [in] hComponent
- Handle of the component to be accessed. This is the component
- handle returned by the call to the OMX_GetHandle function.
- @param [in] pBuffer
- pointer to an OMX_BUFFERHEADERTYPE structure allocated with UseBuffer
- or AllocateBuffer.
- @return OMX_ERRORTYPE
- If the command successfully executes, the return code will be
- OMX_ErrorNone. Otherwise the appropriate OMX error will be returned.
- @ingroup comp buf
- */
-#define OMX_FillThisBuffer( \
- hComponent, \
- pBuffer) \
- ((OMX_COMPONENTTYPE*)hComponent)->FillThisBuffer( \
- hComponent, \
- pBuffer) /* Macro End */
-
-
-
-/** The OMX_UseEGLImage macro will request that the component use
- a EGLImage provided by EGL (and allocate its own buffer header)
- This is a blocking call.
-
- The component should return from this call within 20 msec.
-
- @param [in] hComponent
- Handle of the component to be accessed. This is the component
- handle returned by the call to the OMX_GetHandle function.
- @param [out] ppBuffer
- pointer to an OMX_BUFFERHEADERTYPE structure used to receive the
- pointer to the buffer header. Note that the memory location used
- for this buffer is NOT visible to the IL Client.
- @param [in] nPortIndex
- nPortIndex is used to select the port on the component the buffer will
- be used with. The port can be found by using the nPortIndex
- value as an index into the Port Definition array of the component.
- @param [in] pAppPrivate
- pAppPrivate is used to initialize the pAppPrivate member of the
- buffer header structure.
- @param [in] eglImage
- eglImage contains the handle of the EGLImage to use as a buffer on the
- specified port. The component is expected to validate properties of
- the EGLImage against the configuration of the port to ensure the component
- can use the EGLImage as a buffer.
- @return OMX_ERRORTYPE
- If the command successfully executes, the return code will be
- OMX_ErrorNone. Otherwise the appropriate OMX error will be returned.
- @ingroup comp buf
- */
-#define OMX_UseEGLImage( \
- hComponent, \
- ppBufferHdr, \
- nPortIndex, \
- pAppPrivate, \
- eglImage) \
- ((OMX_COMPONENTTYPE*)hComponent)->UseEGLImage( \
- hComponent, \
- ppBufferHdr, \
- nPortIndex, \
- pAppPrivate, \
- eglImage)
-
-/** The OMX_Init method is used to initialize the OMX core. It shall be the
- first call made into OMX and it should only be executed one time without
- an interviening OMX_Deinit call.
-
- The core should return from this call within 20 msec.
-
- @return OMX_ERRORTYPE
- If the command successfully executes, the return code will be
- OMX_ErrorNone. Otherwise the appropriate OMX error will be returned.
- @ingroup core
- */
-OMX_API OMX_ERRORTYPE OMX_APIENTRY OMX_Init(void);
-
-
-/** The OMX_Deinit method is used to deinitialize the OMX core. It shall be
- the last call made into OMX. In the event that the core determines that
- thare are components loaded when this call is made, the core may return
- with an error rather than try to unload the components.
-
- The core should return from this call within 20 msec.
-
- @return OMX_ERRORTYPE
- If the command successfully executes, the return code will be
- OMX_ErrorNone. Otherwise the appropriate OMX error will be returned.
- @ingroup core
- */
-OMX_API OMX_ERRORTYPE OMX_APIENTRY OMX_Deinit(void);
-
-
-/** The OMX_ComponentNameEnum method will enumerate through all the names of
- recognised valid components in the system. This function is provided
- as a means to detect all the components in the system run-time. There is
- no strict ordering to the enumeration order of component names, although
- each name will only be enumerated once. If the OMX core supports run-time
- installation of new components, it is only requried to detect newly
- installed components when the first call to enumerate component names
- is made (i.e. when nIndex is 0x0).
-
- The core should return from this call in 20 msec.
-
- @param [out] cComponentName
- pointer to a null terminated string with the component name. The
- names of the components are strings less than 127 bytes in length
- plus the trailing null for a maximum size of 128 bytes. An example
- of a valid component name is "OMX.TI.AUDIO.DSP.MIXER\0". Names are
- assigned by the vendor, but shall start with "OMX." and then have
- the Vendor designation next.
- @param [in] nNameLength
- number of characters in the cComponentName string. With all
- component name strings restricted to less than 128 characters
- (including the trailing null) it is recomended that the caller
- provide a input string for the cComponentName of 128 characters.
- @param [in] nIndex
- number containing the enumeration index for the component.
- Multiple calls to OMX_ComponentNameEnum with increasing values
- of nIndex will enumerate through the component names in the
- system until OMX_ErrorNoMore is returned. The value of nIndex
- is 0 to (N-1), where N is the number of valid installed components
- in the system.
- @return OMX_ERRORTYPE
- If the command successfully executes, the return code will be
- OMX_ErrorNone. When the value of nIndex exceeds the number of
- components in the system minus 1, OMX_ErrorNoMore will be
- returned. Otherwise the appropriate OMX error will be returned.
- @ingroup core
- */
-OMX_API OMX_ERRORTYPE OMX_APIENTRY OMX_ComponentNameEnum(
- OMX_OUT OMX_STRING cComponentName,
- OMX_IN OMX_U32 nNameLength,
- OMX_IN OMX_U32 nIndex);
-
-
-/** The OMX_GetHandle method will locate the component specified by the
- component name given, load that component into memory and then invoke
- the component's methods to create an instance of the component.
-
- The core should return from this call within 20 msec.
-
- @param [out] pHandle
- pointer to an OMX_HANDLETYPE pointer to be filled in by this method.
- @param [in] cComponentName
- pointer to a null terminated string with the component name. The
- names of the components are strings less than 127 bytes in length
- plus the trailing null for a maximum size of 128 bytes. An example
- of a valid component name is "OMX.TI.AUDIO.DSP.MIXER\0". Names are
- assigned by the vendor, but shall start with "OMX." and then have
- the Vendor designation next.
- @param [in] pAppData
- pointer to an application defined value that will be returned
- during callbacks so that the application can identify the source
- of the callback.
- @param [in] pCallBacks
- pointer to a OMX_CALLBACKTYPE structure that will be passed to the
- component to initialize it with.
- @return OMX_ERRORTYPE
- If the command successfully executes, the return code will be
- OMX_ErrorNone. Otherwise the appropriate OMX error will be returned.
- @ingroup core
- */
-OMX_API OMX_ERRORTYPE OMX_APIENTRY OMX_GetHandle(
- OMX_OUT OMX_HANDLETYPE* pHandle,
- OMX_IN OMX_STRING cComponentName,
- OMX_IN OMX_PTR pAppData,
- OMX_IN OMX_CALLBACKTYPE* pCallBacks);
-
-
-/** The OMX_FreeHandle method will free a handle allocated by the OMX_GetHandle
- method. If the component reference count goes to zero, the component will
- be unloaded from memory.
-
- The core should return from this call within 20 msec when the component is
- in the OMX_StateLoaded state.
-
- @param [in] hComponent
- Handle of the component to be accessed. This is the component
- handle returned by the call to the GetHandle function.
- @return OMX_ERRORTYPE
- If the command successfully executes, the return code will be
- OMX_ErrorNone. Otherwise the appropriate OMX error will be returned.
- @ingroup core
- */
-OMX_API OMX_ERRORTYPE OMX_APIENTRY OMX_FreeHandle(
- OMX_IN OMX_HANDLETYPE hComponent);
-
-
-
-/** The OMX_SetupTunnel method will handle the necessary calls to the components
- to setup the specified tunnel the two components. NOTE: This is
- an actual method (not a #define macro). This method will make calls into
- the component ComponentTunnelRequest method to do the actual tunnel
- connection.
-
- The ComponentTunnelRequest method on both components will be called.
- This method shall not be called unless the component is in the
- OMX_StateLoaded state except when the ports used for the tunnel are
- disabled. In this case, the component may be in the OMX_StateExecuting,
- OMX_StatePause, or OMX_StateIdle states.
-
- The core should return from this call within 20 msec.
-
- @param [in] hOutput
- Handle of the component to be accessed. Also this is the handle
- of the component whose port, specified in the nPortOutput parameter
- will be used the source for the tunnel. This is the component handle
- returned by the call to the OMX_GetHandle function. There is a
- requirement that hOutput be the source for the data when
- tunelling (i.e. nPortOutput is an output port). If 0x0, the component
- specified in hInput will have it's port specified in nPortInput
- setup for communication with the application / IL client.
- @param [in] nPortOutput
- nPortOutput is used to select the source port on component to be
- used in the tunnel.
- @param [in] hInput
- This is the component to setup the tunnel with. This is the handle
- of the component whose port, specified in the nPortInput parameter
- will be used the destination for the tunnel. This is the component handle
- returned by the call to the OMX_GetHandle function. There is a
- requirement that hInput be the destination for the data when
- tunelling (i.e. nPortInut is an input port). If 0x0, the component
- specified in hOutput will have it's port specified in nPortPOutput
- setup for communication with the application / IL client.
- @param [in] nPortInput
- nPortInput is used to select the destination port on component to be
- used in the tunnel.
- @return OMX_ERRORTYPE
- If the command successfully executes, the return code will be
- OMX_ErrorNone. Otherwise the appropriate OMX error will be returned.
- When OMX_ErrorNotImplemented is returned, one or both components is
- a non-interop component and does not support tunneling.
-
- On failure, the ports of both components are setup for communication
- with the application / IL Client.
- @ingroup core tun
- */
-OMX_API OMX_ERRORTYPE OMX_APIENTRY OMX_SetupTunnel(
- OMX_IN OMX_HANDLETYPE hOutput,
- OMX_IN OMX_U32 nPortOutput,
- OMX_IN OMX_HANDLETYPE hInput,
- OMX_IN OMX_U32 nPortInput);
-
-/** @ingroup cp */
-OMX_API OMX_ERRORTYPE OMX_GetContentPipe(
- OMX_OUT OMX_HANDLETYPE *hPipe,
- OMX_IN OMX_STRING szURI);
-
-/** The OMX_GetComponentsOfRole method will return the number of components that support the given
- role and (if the compNames field is non-NULL) the names of those components. The call will fail if
- an insufficiently sized array of names is supplied. To ensure the array is sufficiently sized the
- client should:
- * first call this function with the compNames field NULL to determine the number of component names
- * second call this function with the compNames field pointing to an array of names allocated
- according to the number returned by the first call.
-
- The core should return from this call within 5 msec.
-
- @param [in] role
- This is generic standard component name consisting only of component class
- name and the type within that class (e.g. 'audio_decoder.aac').
- @param [inout] pNumComps
- This is used both as input and output.
-
- If compNames is NULL, the input is ignored and the output specifies how many components support
- the given role.
-
- If compNames is not NULL, on input it bounds the size of the input structure and
- on output, it specifies the number of components string names listed within the compNames parameter.
- @param [inout] compNames
- If NULL this field is ignored. If non-NULL this points to an array of 128-byte strings which accepts
- a list of the names of all physical components that implement the specified standard component name.
- Each name is NULL terminated. numComps indicates the number of names.
- @ingroup core
- */
-OMX_API OMX_ERRORTYPE OMX_GetComponentsOfRole (
- OMX_IN OMX_STRING role,
- OMX_INOUT OMX_U32 *pNumComps,
- OMX_INOUT OMX_U8 **compNames);
-
-/** The OMX_GetRolesOfComponent method will return the number of roles supported by the given
- component and (if the roles field is non-NULL) the names of those roles. The call will fail if
- an insufficiently sized array of names is supplied. To ensure the array is sufficiently sized the
- client should:
- * first call this function with the roles field NULL to determine the number of role names
- * second call this function with the roles field pointing to an array of names allocated
- according to the number returned by the first call.
-
- The core should return from this call within 5 msec.
-
- @param [in] compName
- This is the name of the component being queried about.
- @param [inout] pNumRoles
- This is used both as input and output.
-
- If roles is NULL, the input is ignored and the output specifies how many roles the component supports.
-
- If compNames is not NULL, on input it bounds the size of the input structure and
- on output, it specifies the number of roles string names listed within the roles parameter.
- @param [out] roles
- If NULL this field is ignored. If non-NULL this points to an array of 128-byte strings
- which accepts a list of the names of all standard components roles implemented on the
- specified component name. numComps indicates the number of names.
- @ingroup core
- */
-OMX_API OMX_ERRORTYPE OMX_GetRolesOfComponent (
- OMX_IN OMX_STRING compName,
- OMX_INOUT OMX_U32 *pNumRoles,
- OMX_OUT OMX_U8 **roles);
-
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
-#endif
-/* File EOF */
-
diff --git a/include/media/stagefright/openmax/OMX_IVCommon.h b/include/media/stagefright/openmax/OMX_IVCommon.h
deleted file mode 100644
index 8bb4ded..0000000
--- a/include/media/stagefright/openmax/OMX_IVCommon.h
+++ /dev/null
@@ -1,947 +0,0 @@
-/* ------------------------------------------------------------------
- * Copyright (C) 1998-2009 PacketVideo
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either
- * express or implied.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- * -------------------------------------------------------------------
- */
-/**
- * Copyright (c) 2008 The Khronos Group Inc.
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files (the
- * "Software"), to deal in the Software without restriction, including
- * without limitation the rights to use, copy, modify, merge, publish,
- * distribute, sublicense, and/or sell copies of the Software, and to
- * permit persons to whom the Software is furnished to do so, subject
- * to the following conditions:
- * The above copyright notice and this permission notice shall be included
- * in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS
- * OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY
- * CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
- * TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
- * SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- *
- */
-
-/**
- * @file OMX_IVCommon.h - OpenMax IL version 1.1.2
- * The structures needed by Video and Image components to exchange
- * parameters and configuration data with the components.
- */
-#ifndef OMX_IVCommon_h
-#define OMX_IVCommon_h
-
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-/**
- * Each OMX header must include all required header files to allow the header
- * to compile without errors. The includes below are required for this header
- * file to compile successfully
- */
-
-#include <OMX_Core.h>
-
-/** @defgroup iv OpenMAX IL Imaging and Video Domain
- * Common structures for OpenMAX IL Imaging and Video domains
- * @{
- */
-
-
-/**
- * Enumeration defining possible uncompressed image/video formats.
- *
- * ENUMS:
- * Unused : Placeholder value when format is N/A
- * Monochrome : black and white
- * 8bitRGB332 : Red 7:5, Green 4:2, Blue 1:0
- * 12bitRGB444 : Red 11:8, Green 7:4, Blue 3:0
- * 16bitARGB4444 : Alpha 15:12, Red 11:8, Green 7:4, Blue 3:0
- * 16bitARGB1555 : Alpha 15, Red 14:10, Green 9:5, Blue 4:0
- * 16bitRGB565 : Red 15:11, Green 10:5, Blue 4:0
- * 16bitBGR565 : Blue 15:11, Green 10:5, Red 4:0
- * 18bitRGB666 : Red 17:12, Green 11:6, Blue 5:0
- * 18bitARGB1665 : Alpha 17, Red 16:11, Green 10:5, Blue 4:0
- * 19bitARGB1666 : Alpha 18, Red 17:12, Green 11:6, Blue 5:0
- * 24bitRGB888 : Red 24:16, Green 15:8, Blue 7:0
- * 24bitBGR888 : Blue 24:16, Green 15:8, Red 7:0
- * 24bitARGB1887 : Alpha 23, Red 22:15, Green 14:7, Blue 6:0
- * 25bitARGB1888 : Alpha 24, Red 23:16, Green 15:8, Blue 7:0
- * 32bitBGRA8888 : Blue 31:24, Green 23:16, Red 15:8, Alpha 7:0
- * 32bitARGB8888 : Alpha 31:24, Red 23:16, Green 15:8, Blue 7:0
- * YUV411Planar : U,Y are subsampled by a factor of 4 horizontally
- * YUV411PackedPlanar : packed per payload in planar slices
- * YUV420Planar : Three arrays Y,U,V.
- * YUV420PackedPlanar : packed per payload in planar slices
- * YUV420SemiPlanar : Two arrays, one is all Y, the other is U and V
- * YUV422Planar : Three arrays Y,U,V.
- * YUV422PackedPlanar : packed per payload in planar slices
- * YUV422SemiPlanar : Two arrays, one is all Y, the other is U and V
- * YCbYCr : Organized as 16bit YUYV (i.e. YCbYCr)
- * YCrYCb : Organized as 16bit YVYU (i.e. YCrYCb)
- * CbYCrY : Organized as 16bit UYVY (i.e. CbYCrY)
- * CrYCbY : Organized as 16bit VYUY (i.e. CrYCbY)
- * YUV444Interleaved : Each pixel contains equal parts YUV
- * RawBayer8bit : SMIA camera output format
- * RawBayer10bit : SMIA camera output format
- * RawBayer8bitcompressed : SMIA camera output format
- */
-typedef enum OMX_COLOR_FORMATTYPE {
- OMX_COLOR_FormatUnused,
- OMX_COLOR_FormatMonochrome,
- OMX_COLOR_Format8bitRGB332,
- OMX_COLOR_Format12bitRGB444,
- OMX_COLOR_Format16bitARGB4444,
- OMX_COLOR_Format16bitARGB1555,
- OMX_COLOR_Format16bitRGB565,
- OMX_COLOR_Format16bitBGR565,
- OMX_COLOR_Format18bitRGB666,
- OMX_COLOR_Format18bitARGB1665,
- OMX_COLOR_Format19bitARGB1666,
- OMX_COLOR_Format24bitRGB888,
- OMX_COLOR_Format24bitBGR888,
- OMX_COLOR_Format24bitARGB1887,
- OMX_COLOR_Format25bitARGB1888,
- OMX_COLOR_Format32bitBGRA8888,
- OMX_COLOR_Format32bitARGB8888,
- OMX_COLOR_FormatYUV411Planar,
- OMX_COLOR_FormatYUV411PackedPlanar,
- OMX_COLOR_FormatYUV420Planar,
- OMX_COLOR_FormatYUV420PackedPlanar,
- OMX_COLOR_FormatYUV420SemiPlanar,
- OMX_COLOR_FormatYUV422Planar,
- OMX_COLOR_FormatYUV422PackedPlanar,
- OMX_COLOR_FormatYUV422SemiPlanar,
- OMX_COLOR_FormatYCbYCr,
- OMX_COLOR_FormatYCrYCb,
- OMX_COLOR_FormatCbYCrY,
- OMX_COLOR_FormatCrYCbY,
- OMX_COLOR_FormatYUV444Interleaved,
- OMX_COLOR_FormatRawBayer8bit,
- OMX_COLOR_FormatRawBayer10bit,
- OMX_COLOR_FormatRawBayer8bitcompressed,
- OMX_COLOR_FormatL2,
- OMX_COLOR_FormatL4,
- OMX_COLOR_FormatL8,
- OMX_COLOR_FormatL16,
- OMX_COLOR_FormatL24,
- OMX_COLOR_FormatL32,
- OMX_COLOR_FormatYUV420PackedSemiPlanar,
- OMX_COLOR_FormatYUV422PackedSemiPlanar,
- OMX_COLOR_Format18BitBGR666,
- OMX_COLOR_Format24BitARGB6666,
- OMX_COLOR_Format24BitABGR6666,
- OMX_COLOR_FormatKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_COLOR_FormatVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- /**<Reserved android opaque colorformat. Tells the encoder that
- * the actual colorformat will be relayed by the
- * Gralloc Buffers.
- * FIXME: In the process of reserving some enum values for
- * Android-specific OMX IL colorformats. Change this enum to
- * an acceptable range once that is done.
- * */
- OMX_COLOR_FormatAndroidOpaque = 0x7F000789,
- OMX_TI_COLOR_FormatYUV420PackedSemiPlanar = 0x7F000100,
- OMX_QCOM_COLOR_FormatYVU420SemiPlanar = 0x7FA30C00,
- OMX_COLOR_FormatMax = 0x7FFFFFFF
-} OMX_COLOR_FORMATTYPE;
-
-
-/**
- * Defines the matrix for conversion from RGB to YUV or vice versa.
- * iColorMatrix should be initialized with the fixed point values
- * used in converting between formats.
- */
-typedef struct OMX_CONFIG_COLORCONVERSIONTYPE {
- OMX_U32 nSize; /**< Size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version info */
- OMX_U32 nPortIndex; /**< Port that this struct applies to */
- OMX_S32 xColorMatrix[3][3]; /**< Stored in signed Q16 format */
- OMX_S32 xColorOffset[4]; /**< Stored in signed Q16 format */
-}OMX_CONFIG_COLORCONVERSIONTYPE;
-
-
-/**
- * Structure defining percent to scale each frame dimension. For example:
- * To make the width 50% larger, use fWidth = 1.5 and to make the width
- * 1/2 the original size, use fWidth = 0.5
- */
-typedef struct OMX_CONFIG_SCALEFACTORTYPE {
- OMX_U32 nSize; /**< Size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version info */
- OMX_U32 nPortIndex; /**< Port that this struct applies to */
- OMX_S32 xWidth; /**< Fixed point value stored as Q16 */
- OMX_S32 xHeight; /**< Fixed point value stored as Q16 */
-}OMX_CONFIG_SCALEFACTORTYPE;
-
-
-/**
- * Enumeration of possible image filter types
- */
-typedef enum OMX_IMAGEFILTERTYPE {
- OMX_ImageFilterNone,
- OMX_ImageFilterNoise,
- OMX_ImageFilterEmboss,
- OMX_ImageFilterNegative,
- OMX_ImageFilterSketch,
- OMX_ImageFilterOilPaint,
- OMX_ImageFilterHatch,
- OMX_ImageFilterGpen,
- OMX_ImageFilterAntialias,
- OMX_ImageFilterDeRing,
- OMX_ImageFilterSolarize,
- OMX_ImageFilterKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_ImageFilterVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_ImageFilterMax = 0x7FFFFFFF
-} OMX_IMAGEFILTERTYPE;
-
-
-/**
- * Image filter configuration
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * eImageFilter : Image filter type enumeration
- */
-typedef struct OMX_CONFIG_IMAGEFILTERTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_IMAGEFILTERTYPE eImageFilter;
-} OMX_CONFIG_IMAGEFILTERTYPE;
-
-
-/**
- * Customized U and V for color enhancement
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * bColorEnhancement : Enable/disable color enhancement
- * nCustomizedU : Practical values: 16-240, range: 0-255, value set for
- * U component
- * nCustomizedV : Practical values: 16-240, range: 0-255, value set for
- * V component
- */
-typedef struct OMX_CONFIG_COLORENHANCEMENTTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_BOOL bColorEnhancement;
- OMX_U8 nCustomizedU;
- OMX_U8 nCustomizedV;
-} OMX_CONFIG_COLORENHANCEMENTTYPE;
-
-
-/**
- * Define color key and color key mask
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * nARGBColor : 32bit Alpha, Red, Green, Blue Color
- * nARGBMask : 32bit Mask for Alpha, Red, Green, Blue channels
- */
-typedef struct OMX_CONFIG_COLORKEYTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_U32 nARGBColor;
- OMX_U32 nARGBMask;
-} OMX_CONFIG_COLORKEYTYPE;
-
-
-/**
- * List of color blend types for pre/post processing
- *
- * ENUMS:
- * None : No color blending present
- * AlphaConstant : Function is (alpha_constant * src) +
- * (1 - alpha_constant) * dst)
- * AlphaPerPixel : Function is (alpha * src) + (1 - alpha) * dst)
- * Alternate : Function is alternating pixels from src and dst
- * And : Function is (src & dst)
- * Or : Function is (src | dst)
- * Invert : Function is ~src
- */
-typedef enum OMX_COLORBLENDTYPE {
- OMX_ColorBlendNone,
- OMX_ColorBlendAlphaConstant,
- OMX_ColorBlendAlphaPerPixel,
- OMX_ColorBlendAlternate,
- OMX_ColorBlendAnd,
- OMX_ColorBlendOr,
- OMX_ColorBlendInvert,
- OMX_ColorBlendKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_ColorBlendVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_ColorBlendMax = 0x7FFFFFFF
-} OMX_COLORBLENDTYPE;
-
-
-/**
- * Color blend configuration
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * nRGBAlphaConstant : Constant global alpha values when global alpha is used
- * eColorBlend : Color blend type enumeration
- */
-typedef struct OMX_CONFIG_COLORBLENDTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_U32 nRGBAlphaConstant;
- OMX_COLORBLENDTYPE eColorBlend;
-} OMX_CONFIG_COLORBLENDTYPE;
-
-
-/**
- * Hold frame dimension
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * nWidth : Frame width in pixels
- * nHeight : Frame height in pixels
- */
-typedef struct OMX_FRAMESIZETYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_U32 nWidth;
- OMX_U32 nHeight;
-} OMX_FRAMESIZETYPE;
-
-
-/**
- * Rotation configuration
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * nRotation : +/- integer rotation value
- */
-typedef struct OMX_CONFIG_ROTATIONTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_S32 nRotation;
-} OMX_CONFIG_ROTATIONTYPE;
-
-
-/**
- * Possible mirroring directions for pre/post processing
- *
- * ENUMS:
- * None : No mirroring
- * Vertical : Vertical mirroring, flip on X axis
- * Horizontal : Horizontal mirroring, flip on Y axis
- * Both : Both vertical and horizontal mirroring
- */
-typedef enum OMX_MIRRORTYPE {
- OMX_MirrorNone = 0,
- OMX_MirrorVertical,
- OMX_MirrorHorizontal,
- OMX_MirrorBoth,
- OMX_MirrorKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_MirrorVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_MirrorMax = 0x7FFFFFFF
-} OMX_MIRRORTYPE;
-
-
-/**
- * Mirroring configuration
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * eMirror : Mirror type enumeration
- */
-typedef struct OMX_CONFIG_MIRRORTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_MIRRORTYPE eMirror;
-} OMX_CONFIG_MIRRORTYPE;
-
-
-/**
- * Position information only
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * nX : X coordinate for the point
- * nY : Y coordinate for the point
- */
-typedef struct OMX_CONFIG_POINTTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_S32 nX;
- OMX_S32 nY;
-} OMX_CONFIG_POINTTYPE;
-
-
-/**
- * Frame size plus position
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * nLeft : X Coordinate of the top left corner of the rectangle
- * nTop : Y Coordinate of the top left corner of the rectangle
- * nWidth : Width of the rectangle
- * nHeight : Height of the rectangle
- */
-typedef struct OMX_CONFIG_RECTTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_S32 nLeft;
- OMX_S32 nTop;
- OMX_U32 nWidth;
- OMX_U32 nHeight;
-} OMX_CONFIG_RECTTYPE;
-
-
-/**
- * Deblocking state; it is required to be set up before starting the codec
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * bDeblocking : Enable/disable deblocking mode
- */
-typedef struct OMX_PARAM_DEBLOCKINGTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_BOOL bDeblocking;
-} OMX_PARAM_DEBLOCKINGTYPE;
-
-
-/**
- * Stabilization state
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * bStab : Enable/disable frame stabilization state
- */
-typedef struct OMX_CONFIG_FRAMESTABTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_BOOL bStab;
-} OMX_CONFIG_FRAMESTABTYPE;
-
-
-/**
- * White Balance control type
- *
- * STRUCT MEMBERS:
- * SunLight : Referenced in JSR-234
- * Flash : Optimal for device's integrated flash
- */
-typedef enum OMX_WHITEBALCONTROLTYPE {
- OMX_WhiteBalControlOff = 0,
- OMX_WhiteBalControlAuto,
- OMX_WhiteBalControlSunLight,
- OMX_WhiteBalControlCloudy,
- OMX_WhiteBalControlShade,
- OMX_WhiteBalControlTungsten,
- OMX_WhiteBalControlFluorescent,
- OMX_WhiteBalControlIncandescent,
- OMX_WhiteBalControlFlash,
- OMX_WhiteBalControlHorizon,
- OMX_WhiteBalControlKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_WhiteBalControlVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_WhiteBalControlMax = 0x7FFFFFFF
-} OMX_WHITEBALCONTROLTYPE;
-
-
-/**
- * White Balance control configuration
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * eWhiteBalControl : White balance enumeration
- */
-typedef struct OMX_CONFIG_WHITEBALCONTROLTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_WHITEBALCONTROLTYPE eWhiteBalControl;
-} OMX_CONFIG_WHITEBALCONTROLTYPE;
-
-
-/**
- * Exposure control type
- */
-typedef enum OMX_EXPOSURECONTROLTYPE {
- OMX_ExposureControlOff = 0,
- OMX_ExposureControlAuto,
- OMX_ExposureControlNight,
- OMX_ExposureControlBackLight,
- OMX_ExposureControlSpotLight,
- OMX_ExposureControlSports,
- OMX_ExposureControlSnow,
- OMX_ExposureControlBeach,
- OMX_ExposureControlLargeAperture,
- OMX_ExposureControlSmallApperture,
- OMX_ExposureControlKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_ExposureControlVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_ExposureControlMax = 0x7FFFFFFF
-} OMX_EXPOSURECONTROLTYPE;
-
-
-/**
- * White Balance control configuration
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * eExposureControl : Exposure control enumeration
- */
-typedef struct OMX_CONFIG_EXPOSURECONTROLTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_EXPOSURECONTROLTYPE eExposureControl;
-} OMX_CONFIG_EXPOSURECONTROLTYPE;
-
-
-/**
- * Defines sensor supported mode.
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * nFrameRate : Single shot mode is indicated by a 0
- * bOneShot : Enable for single shot, disable for streaming
- * sFrameSize : Framesize
- */
-typedef struct OMX_PARAM_SENSORMODETYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_U32 nFrameRate;
- OMX_BOOL bOneShot;
- OMX_FRAMESIZETYPE sFrameSize;
-} OMX_PARAM_SENSORMODETYPE;
-
-
-/**
- * Defines contrast level
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * nContrast : Values allowed for contrast -100 to 100, zero means no change
- */
-typedef struct OMX_CONFIG_CONTRASTTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_S32 nContrast;
-} OMX_CONFIG_CONTRASTTYPE;
-
-
-/**
- * Defines brightness level
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * nBrightness : 0-100%
- */
-typedef struct OMX_CONFIG_BRIGHTNESSTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_U32 nBrightness;
-} OMX_CONFIG_BRIGHTNESSTYPE;
-
-
-/**
- * Defines backlight level configuration for a video sink, e.g. LCD panel
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * nBacklight : Values allowed for backlight 0-100%
- * nTimeout : Number of milliseconds before backlight automatically turns
- * off. A value of 0x0 disables backight timeout
- */
-typedef struct OMX_CONFIG_BACKLIGHTTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_U32 nBacklight;
- OMX_U32 nTimeout;
-} OMX_CONFIG_BACKLIGHTTYPE;
-
-
-/**
- * Defines setting for Gamma
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * nGamma : Values allowed for gamma -100 to 100, zero means no change
- */
-typedef struct OMX_CONFIG_GAMMATYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_S32 nGamma;
-} OMX_CONFIG_GAMMATYPE;
-
-
-/**
- * Define for setting saturation
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * nSaturation : Values allowed for saturation -100 to 100, zero means
- * no change
- */
-typedef struct OMX_CONFIG_SATURATIONTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_S32 nSaturation;
-} OMX_CONFIG_SATURATIONTYPE;
-
-
-/**
- * Define for setting Lightness
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * nLightness : Values allowed for lightness -100 to 100, zero means no
- * change
- */
-typedef struct OMX_CONFIG_LIGHTNESSTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_S32 nLightness;
-} OMX_CONFIG_LIGHTNESSTYPE;
-
-
-/**
- * Plane blend configuration
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Index of input port associated with the plane.
- * nDepth : Depth of the plane in relation to the screen. Higher
- * numbered depths are "behind" lower number depths.
- * This number defaults to the Port Index number.
- * nAlpha : Transparency blending component for the entire plane.
- * See blending modes for more detail.
- */
-typedef struct OMX_CONFIG_PLANEBLENDTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_U32 nDepth;
- OMX_U32 nAlpha;
-} OMX_CONFIG_PLANEBLENDTYPE;
-
-
-/**
- * Define interlace type
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * bEnable : Enable control variable for this functionality
- * (see below)
- * nInterleavePortIndex : Index of input or output port associated with
- * the interleaved plane.
- * pPlanarPortIndexes[4] : Index of input or output planar ports.
- */
-typedef struct OMX_PARAM_INTERLEAVETYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_BOOL bEnable;
- OMX_U32 nInterleavePortIndex;
-} OMX_PARAM_INTERLEAVETYPE;
-
-
-/**
- * Defines the picture effect used for an input picture
- */
-typedef enum OMX_TRANSITIONEFFECTTYPE {
- OMX_EffectNone,
- OMX_EffectFadeFromBlack,
- OMX_EffectFadeToBlack,
- OMX_EffectUnspecifiedThroughConstantColor,
- OMX_EffectDissolve,
- OMX_EffectWipe,
- OMX_EffectUnspecifiedMixOfTwoScenes,
- OMX_EffectKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_EffectVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_EffectMax = 0x7FFFFFFF
-} OMX_TRANSITIONEFFECTTYPE;
-
-
-/**
- * Structure used to configure current transition effect
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * eEffect : Effect to enable
- */
-typedef struct OMX_CONFIG_TRANSITIONEFFECTTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_TRANSITIONEFFECTTYPE eEffect;
-} OMX_CONFIG_TRANSITIONEFFECTTYPE;
-
-
-/**
- * Defines possible data unit types for encoded video data. The data unit
- * types are used both for encoded video input for playback as well as
- * encoded video output from recording.
- */
-typedef enum OMX_DATAUNITTYPE {
- OMX_DataUnitCodedPicture,
- OMX_DataUnitVideoSegment,
- OMX_DataUnitSeveralSegments,
- OMX_DataUnitArbitraryStreamSection,
- OMX_DataUnitKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_DataUnitVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_DataUnitMax = 0x7FFFFFFF
-} OMX_DATAUNITTYPE;
-
-
-/**
- * Defines possible encapsulation types for coded video data unit. The
- * encapsulation information is used both for encoded video input for
- * playback as well as encoded video output from recording.
- */
-typedef enum OMX_DATAUNITENCAPSULATIONTYPE {
- OMX_DataEncapsulationElementaryStream,
- OMX_DataEncapsulationGenericPayload,
- OMX_DataEncapsulationRtpPayload,
- OMX_DataEncapsulationKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_DataEncapsulationVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_DataEncapsulationMax = 0x7FFFFFFF
-} OMX_DATAUNITENCAPSULATIONTYPE;
-
-
-/**
- * Structure used to configure the type of being decoded/encoded
- */
-typedef struct OMX_PARAM_DATAUNITTYPE {
- OMX_U32 nSize; /**< Size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< Port that this structure applies to */
- OMX_DATAUNITTYPE eUnitType;
- OMX_DATAUNITENCAPSULATIONTYPE eEncapsulationType;
-} OMX_PARAM_DATAUNITTYPE;
-
-
-/**
- * Defines dither types
- */
-typedef enum OMX_DITHERTYPE {
- OMX_DitherNone,
- OMX_DitherOrdered,
- OMX_DitherErrorDiffusion,
- OMX_DitherOther,
- OMX_DitherKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_DitherVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_DitherMax = 0x7FFFFFFF
-} OMX_DITHERTYPE;
-
-
-/**
- * Structure used to configure current type of dithering
- */
-typedef struct OMX_CONFIG_DITHERTYPE {
- OMX_U32 nSize; /**< Size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< Port that this structure applies to */
- OMX_DITHERTYPE eDither; /**< Type of dithering to use */
-} OMX_CONFIG_DITHERTYPE;
-
-typedef struct OMX_CONFIG_CAPTUREMODETYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex; /**< Port that this structure applies to */
- OMX_BOOL bContinuous; /**< If true then ignore frame rate and emit capture
- * data as fast as possible (otherwise obey port's frame rate). */
- OMX_BOOL bFrameLimited; /**< If true then terminate capture after the port emits the
- * specified number of frames (otherwise the port does not
- * terminate the capture until instructed to do so by the client).
- * Even if set, the client may manually terminate the capture prior
- * to reaching the limit. */
- OMX_U32 nFrameLimit; /**< Limit on number of frames emitted during a capture (only
- * valid if bFrameLimited is set). */
-} OMX_CONFIG_CAPTUREMODETYPE;
-
-typedef enum OMX_METERINGTYPE {
-
- OMX_MeteringModeAverage, /**< Center-weighted average metering. */
- OMX_MeteringModeSpot, /**< Spot (partial) metering. */
- OMX_MeteringModeMatrix, /**< Matrix or evaluative metering. */
-
- OMX_MeteringKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_MeteringVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_EVModeMax = 0x7fffffff
-} OMX_METERINGTYPE;
-
-typedef struct OMX_CONFIG_EXPOSUREVALUETYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_METERINGTYPE eMetering;
- OMX_S32 xEVCompensation; /**< Fixed point value stored as Q16 */
- OMX_U32 nApertureFNumber; /**< e.g. nApertureFNumber = 2 implies "f/2" - Q16 format */
- OMX_BOOL bAutoAperture; /**< Whether aperture number is defined automatically */
- OMX_U32 nShutterSpeedMsec; /**< Shutterspeed in milliseconds */
- OMX_BOOL bAutoShutterSpeed; /**< Whether shutter speed is defined automatically */
- OMX_U32 nSensitivity; /**< e.g. nSensitivity = 100 implies "ISO 100" */
- OMX_BOOL bAutoSensitivity; /**< Whether sensitivity is defined automatically */
-} OMX_CONFIG_EXPOSUREVALUETYPE;
-
-/**
- * Focus region configuration
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * bCenter : Use center region as focus region of interest
- * bLeft : Use left region as focus region of interest
- * bRight : Use right region as focus region of interest
- * bTop : Use top region as focus region of interest
- * bBottom : Use bottom region as focus region of interest
- * bTopLeft : Use top left region as focus region of interest
- * bTopRight : Use top right region as focus region of interest
- * bBottomLeft : Use bottom left region as focus region of interest
- * bBottomRight : Use bottom right region as focus region of interest
- */
-typedef struct OMX_CONFIG_FOCUSREGIONTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_BOOL bCenter;
- OMX_BOOL bLeft;
- OMX_BOOL bRight;
- OMX_BOOL bTop;
- OMX_BOOL bBottom;
- OMX_BOOL bTopLeft;
- OMX_BOOL bTopRight;
- OMX_BOOL bBottomLeft;
- OMX_BOOL bBottomRight;
-} OMX_CONFIG_FOCUSREGIONTYPE;
-
-/**
- * Focus Status type
- */
-typedef enum OMX_FOCUSSTATUSTYPE {
- OMX_FocusStatusOff = 0,
- OMX_FocusStatusRequest,
- OMX_FocusStatusReached,
- OMX_FocusStatusUnableToReach,
- OMX_FocusStatusLost,
- OMX_FocusStatusKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_FocusStatusVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_FocusStatusMax = 0x7FFFFFFF
-} OMX_FOCUSSTATUSTYPE;
-
-/**
- * Focus status configuration
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * eFocusStatus : Specifies the focus status
- * bCenterStatus : Use center region as focus region of interest
- * bLeftStatus : Use left region as focus region of interest
- * bRightStatus : Use right region as focus region of interest
- * bTopStatus : Use top region as focus region of interest
- * bBottomStatus : Use bottom region as focus region of interest
- * bTopLeftStatus : Use top left region as focus region of interest
- * bTopRightStatus : Use top right region as focus region of interest
- * bBottomLeftStatus : Use bottom left region as focus region of interest
- * bBottomRightStatus : Use bottom right region as focus region of interest
- */
-typedef struct OMX_PARAM_FOCUSSTATUSTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_FOCUSSTATUSTYPE eFocusStatus;
- OMX_BOOL bCenterStatus;
- OMX_BOOL bLeftStatus;
- OMX_BOOL bRightStatus;
- OMX_BOOL bTopStatus;
- OMX_BOOL bBottomStatus;
- OMX_BOOL bTopLeftStatus;
- OMX_BOOL bTopRightStatus;
- OMX_BOOL bBottomLeftStatus;
- OMX_BOOL bBottomRightStatus;
-} OMX_PARAM_FOCUSSTATUSTYPE;
-
-/** @} */
-
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
-#endif
-/* File EOF */
diff --git a/include/media/stagefright/openmax/OMX_Image.h b/include/media/stagefright/openmax/OMX_Image.h
deleted file mode 100644
index 42e39ec..0000000
--- a/include/media/stagefright/openmax/OMX_Image.h
+++ /dev/null
@@ -1,345 +0,0 @@
-/* ------------------------------------------------------------------
- * Copyright (C) 1998-2009 PacketVideo
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either
- * express or implied.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- * -------------------------------------------------------------------
- */
-/**
- * Copyright (c) 2008 The Khronos Group Inc.
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files (the
- * "Software"), to deal in the Software without restriction, including
- * without limitation the rights to use, copy, modify, merge, publish,
- * distribute, sublicense, and/or sell copies of the Software, and to
- * permit persons to whom the Software is furnished to do so, subject
- * to the following conditions:
- * The above copyright notice and this permission notice shall be included
- * in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS
- * OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY
- * CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
- * TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
- * SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- */
-
-/**
- * @file OMX_Image.h - OpenMax IL version 1.1.2
- * The structures needed by Image components to exchange parameters and
- * configuration data with the components.
- */
-#ifndef OMX_Image_h
-#define OMX_Image_h
-
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-
-/**
- * Each OMX header must include all required header files to allow the
- * header to compile without errors. The includes below are required
- * for this header file to compile successfully
- */
-
-#include <OMX_IVCommon.h>
-
-/** @defgroup imaging OpenMAX IL Imaging Domain
- * @ingroup iv
- * Structures for OpenMAX IL Imaging domain
- * @{
- */
-
-/**
- * Enumeration used to define the possible image compression coding.
- */
-typedef enum OMX_IMAGE_CODINGTYPE {
- OMX_IMAGE_CodingUnused, /**< Value when format is N/A */
- OMX_IMAGE_CodingAutoDetect, /**< Auto detection of image format */
- OMX_IMAGE_CodingJPEG, /**< JPEG/JFIF image format */
- OMX_IMAGE_CodingJPEG2K, /**< JPEG 2000 image format */
- OMX_IMAGE_CodingEXIF, /**< EXIF image format */
- OMX_IMAGE_CodingTIFF, /**< TIFF image format */
- OMX_IMAGE_CodingGIF, /**< Graphics image format */
- OMX_IMAGE_CodingPNG, /**< PNG image format */
- OMX_IMAGE_CodingLZW, /**< LZW image format */
- OMX_IMAGE_CodingBMP, /**< Windows Bitmap format */
- OMX_IMAGE_CodingKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_IMAGE_CodingVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_IMAGE_CodingMax = 0x7FFFFFFF
-} OMX_IMAGE_CODINGTYPE;
-
-
-/**
- * Data structure used to define an image path. The number of image paths
- * for input and output will vary by type of the image component.
- *
- * Input (aka Source) : Zero Inputs, one Output,
- * Splitter : One Input, 2 or more Outputs,
- * Processing Element : One Input, one output,
- * Mixer : 2 or more inputs, one output,
- * Output (aka Sink) : One Input, zero outputs.
- *
- * The PortDefinition structure is used to define all of the parameters
- * necessary for the compliant component to setup an input or an output
- * image path. If additional vendor specific data is required, it should
- * be transmitted to the component using the CustomCommand function.
- * Compliant components will prepopulate this structure with optimal
- * values during the OMX_GetParameter() command.
- *
- * STRUCT MEMBERS:
- * cMIMEType : MIME type of data for the port
- * pNativeRender : Platform specific reference for a display if a
- * sync, otherwise this field is 0
- * nFrameWidth : Width of frame to be used on port if
- * uncompressed format is used. Use 0 for
- * unknown, don't care or variable
- * nFrameHeight : Height of frame to be used on port if
- * uncompressed format is used. Use 0 for
- * unknown, don't care or variable
- * nStride : Number of bytes per span of an image (i.e.
- * indicates the number of bytes to get from
- * span N to span N+1, where negative stride
- * indicates the image is bottom up
- * nSliceHeight : Height used when encoding in slices
- * bFlagErrorConcealment : Turns on error concealment if it is supported by
- * the OMX component
- * eCompressionFormat : Compression format used in this instance of
- * the component. When OMX_IMAGE_CodingUnused is
- * specified, eColorFormat is valid
- * eColorFormat : Decompressed format used by this component
- * pNativeWindow : Platform specific reference for a window object if a
- * display sink , otherwise this field is 0x0.
- */
-typedef struct OMX_IMAGE_PORTDEFINITIONTYPE {
- OMX_STRING cMIMEType;
- OMX_NATIVE_DEVICETYPE pNativeRender;
- OMX_U32 nFrameWidth;
- OMX_U32 nFrameHeight;
- OMX_S32 nStride;
- OMX_U32 nSliceHeight;
- OMX_BOOL bFlagErrorConcealment;
- OMX_IMAGE_CODINGTYPE eCompressionFormat;
- OMX_COLOR_FORMATTYPE eColorFormat;
- OMX_NATIVE_WINDOWTYPE pNativeWindow;
-} OMX_IMAGE_PORTDEFINITIONTYPE;
-
-
-/**
- * Port format parameter. This structure is used to enumerate the various
- * data input/output format supported by the port.
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Indicates which port to set
- * nIndex : Indicates the enumeration index for the format from
- * 0x0 to N-1
- * eCompressionFormat : Compression format used in this instance of the
- * component. When OMX_IMAGE_CodingUnused is specified,
- * eColorFormat is valid
- * eColorFormat : Decompressed format used by this component
- */
-typedef struct OMX_IMAGE_PARAM_PORTFORMATTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_U32 nIndex;
- OMX_IMAGE_CODINGTYPE eCompressionFormat;
- OMX_COLOR_FORMATTYPE eColorFormat;
-} OMX_IMAGE_PARAM_PORTFORMATTYPE;
-
-
-/**
- * Flash control type
- *
- * ENUMS
- * Torch : Flash forced constantly on
- */
-typedef enum OMX_IMAGE_FLASHCONTROLTYPE {
- OMX_IMAGE_FlashControlOn = 0,
- OMX_IMAGE_FlashControlOff,
- OMX_IMAGE_FlashControlAuto,
- OMX_IMAGE_FlashControlRedEyeReduction,
- OMX_IMAGE_FlashControlFillin,
- OMX_IMAGE_FlashControlTorch,
- OMX_IMAGE_FlashControlKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_IMAGE_FlashControlVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_IMAGE_FlashControlMax = 0x7FFFFFFF
-} OMX_IMAGE_FLASHCONTROLTYPE;
-
-
-/**
- * Flash control configuration
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * eFlashControl : Flash control type
- */
-typedef struct OMX_IMAGE_PARAM_FLASHCONTROLTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_IMAGE_FLASHCONTROLTYPE eFlashControl;
-} OMX_IMAGE_PARAM_FLASHCONTROLTYPE;
-
-
-/**
- * Focus control type
- */
-typedef enum OMX_IMAGE_FOCUSCONTROLTYPE {
- OMX_IMAGE_FocusControlOn = 0,
- OMX_IMAGE_FocusControlOff,
- OMX_IMAGE_FocusControlAuto,
- OMX_IMAGE_FocusControlAutoLock,
- OMX_IMAGE_FocusControlKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_IMAGE_FocusControlVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_IMAGE_FocusControlMax = 0x7FFFFFFF
-} OMX_IMAGE_FOCUSCONTROLTYPE;
-
-
-/**
- * Focus control configuration
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * eFocusControl : Focus control
- * nFocusSteps : Focus can take on values from 0 mm to infinity.
- * Interest is only in number of steps over this range.
- * nFocusStepIndex : Current focus step index
- */
-typedef struct OMX_IMAGE_CONFIG_FOCUSCONTROLTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_IMAGE_FOCUSCONTROLTYPE eFocusControl;
- OMX_U32 nFocusSteps;
- OMX_U32 nFocusStepIndex;
-} OMX_IMAGE_CONFIG_FOCUSCONTROLTYPE;
-
-
-/**
- * Q Factor for JPEG compression, which controls the tradeoff between image
- * quality and size. Q Factor provides a more simple means of controlling
- * JPEG compression quality, without directly programming Quantization
- * tables for chroma and luma
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * nQFactor : JPEG Q factor value in the range of 1-100. A factor of 1
- * produces the smallest, worst quality images, and a factor
- * of 100 produces the largest, best quality images. A
- * typical default is 75 for small good quality images
- */
-typedef struct OMX_IMAGE_PARAM_QFACTORTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_U32 nQFactor;
-} OMX_IMAGE_PARAM_QFACTORTYPE;
-
-/**
- * Quantization table type
- */
-
-typedef enum OMX_IMAGE_QUANTIZATIONTABLETYPE {
- OMX_IMAGE_QuantizationTableLuma = 0,
- OMX_IMAGE_QuantizationTableChroma,
- OMX_IMAGE_QuantizationTableChromaCb,
- OMX_IMAGE_QuantizationTableChromaCr,
- OMX_IMAGE_QuantizationTableKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_IMAGE_QuantizationTableVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_IMAGE_QuantizationTableMax = 0x7FFFFFFF
-} OMX_IMAGE_QUANTIZATIONTABLETYPE;
-
-/**
- * JPEG quantization tables are used to determine DCT compression for
- * YUV data, as an alternative to specifying Q factor, providing exact
- * control of compression
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * eQuantizationTable : Quantization table type
- * nQuantizationMatrix[64] : JPEG quantization table of coefficients stored
- * in increasing columns then by rows of data (i.e.
- * row 1, ... row 8). Quantization values are in
- * the range 0-255 and stored in linear order
- * (i.e. the component will zig-zag the
- * quantization table data if required internally)
- */
-typedef struct OMX_IMAGE_PARAM_QUANTIZATIONTABLETYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_IMAGE_QUANTIZATIONTABLETYPE eQuantizationTable;
- OMX_U8 nQuantizationMatrix[64];
-} OMX_IMAGE_PARAM_QUANTIZATIONTABLETYPE;
-
-
-/**
- * Huffman table type, the same Huffman table is applied for chroma and
- * luma component
- */
-typedef enum OMX_IMAGE_HUFFMANTABLETYPE {
- OMX_IMAGE_HuffmanTableAC = 0,
- OMX_IMAGE_HuffmanTableDC,
- OMX_IMAGE_HuffmanTableACLuma,
- OMX_IMAGE_HuffmanTableACChroma,
- OMX_IMAGE_HuffmanTableDCLuma,
- OMX_IMAGE_HuffmanTableDCChroma,
- OMX_IMAGE_HuffmanTableKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_IMAGE_HuffmanTableVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_IMAGE_HuffmanTableMax = 0x7FFFFFFF
-} OMX_IMAGE_HUFFMANTABLETYPE;
-
-/**
- * JPEG Huffman table
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * eHuffmanTable : Huffman table type
- * nNumberOfHuffmanCodeOfLength[16] : 0-16, number of Huffman codes of each
- * possible length
- * nHuffmanTable[256] : 0-255, the size used for AC and DC
- * HuffmanTable are 16 and 162
- */
-typedef struct OMX_IMAGE_PARAM_HUFFMANTTABLETYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_IMAGE_HUFFMANTABLETYPE eHuffmanTable;
- OMX_U8 nNumberOfHuffmanCodeOfLength[16];
- OMX_U8 nHuffmanTable[256];
-}OMX_IMAGE_PARAM_HUFFMANTTABLETYPE;
-
-/** @} */
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
-#endif
-/* File EOF */
diff --git a/include/media/stagefright/openmax/OMX_Index.h b/include/media/stagefright/openmax/OMX_Index.h
deleted file mode 100644
index c0b8d92..0000000
--- a/include/media/stagefright/openmax/OMX_Index.h
+++ /dev/null
@@ -1,275 +0,0 @@
-/* ------------------------------------------------------------------
- * Copyright (C) 1998-2009 PacketVideo
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either
- * express or implied.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- * -------------------------------------------------------------------
- */
-/*
- * Copyright (c) 2008 The Khronos Group Inc.
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files (the
- * "Software"), to deal in the Software without restriction, including
- * without limitation the rights to use, copy, modify, merge, publish,
- * distribute, sublicense, and/or sell copies of the Software, and to
- * permit persons to whom the Software is furnished to do so, subject
- * to the following conditions:
- * The above copyright notice and this permission notice shall be included
- * in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS
- * OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY
- * CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
- * TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
- * SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- *
- */
-
-/** @file OMX_Index.h - OpenMax IL version 1.1.2
- * The OMX_Index header file contains the definitions for both applications
- * and components .
- */
-
-
-#ifndef OMX_Index_h
-#define OMX_Index_h
-
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-
-/* Each OMX header must include all required header files to allow the
- * header to compile without errors. The includes below are required
- * for this header file to compile successfully
- */
-#include <OMX_Types.h>
-
-
-/** The OMX_INDEXTYPE enumeration is used to select a structure when either
- * getting or setting parameters and/or configuration data. Each entry in
- * this enumeration maps to an OMX specified structure. When the
- * OMX_GetParameter, OMX_SetParameter, OMX_GetConfig or OMX_SetConfig methods
- * are used, the second parameter will always be an entry from this enumeration
- * and the third entry will be the structure shown in the comments for the entry.
- * For example, if the application is initializing a cropping function, the
- * OMX_SetConfig command would have OMX_IndexConfigCommonInputCrop as the second parameter
- * and would send a pointer to an initialized OMX_RECTTYPE structure as the
- * third parameter.
- *
- * The enumeration entries named with the OMX_Config prefix are sent using
- * the OMX_SetConfig command and the enumeration entries named with the
- * OMX_PARAM_ prefix are sent using the OMX_SetParameter command.
- */
-typedef enum OMX_INDEXTYPE {
-
- OMX_IndexComponentStartUnused = 0x01000000,
- OMX_IndexParamPriorityMgmt, /**< reference: OMX_PRIORITYMGMTTYPE */
- OMX_IndexParamAudioInit, /**< reference: OMX_PORT_PARAM_TYPE */
- OMX_IndexParamImageInit, /**< reference: OMX_PORT_PARAM_TYPE */
- OMX_IndexParamVideoInit, /**< reference: OMX_PORT_PARAM_TYPE */
- OMX_IndexParamOtherInit, /**< reference: OMX_PORT_PARAM_TYPE */
- OMX_IndexParamNumAvailableStreams, /**< reference: OMX_PARAM_U32TYPE */
- OMX_IndexParamActiveStream, /**< reference: OMX_PARAM_U32TYPE */
- OMX_IndexParamSuspensionPolicy, /**< reference: OMX_PARAM_SUSPENSIONPOLICYTYPE */
- OMX_IndexParamComponentSuspended, /**< reference: OMX_PARAM_SUSPENSIONTYPE */
- OMX_IndexConfigCapturing, /**< reference: OMX_CONFIG_BOOLEANTYPE */
- OMX_IndexConfigCaptureMode, /**< reference: OMX_CONFIG_CAPTUREMODETYPE */
- OMX_IndexAutoPauseAfterCapture, /**< reference: OMX_CONFIG_BOOLEANTYPE */
- OMX_IndexParamContentURI, /**< reference: OMX_PARAM_CONTENTURITYPE */
- OMX_IndexParamCustomContentPipe, /**< reference: OMX_PARAM_CONTENTPIPETYPE */
- OMX_IndexParamDisableResourceConcealment, /**< reference: OMX_RESOURCECONCEALMENTTYPE */
- OMX_IndexConfigMetadataItemCount, /**< reference: OMX_CONFIG_METADATAITEMCOUNTTYPE */
- OMX_IndexConfigContainerNodeCount, /**< reference: OMX_CONFIG_CONTAINERNODECOUNTTYPE */
- OMX_IndexConfigMetadataItem, /**< reference: OMX_CONFIG_METADATAITEMTYPE */
- OMX_IndexConfigCounterNodeID, /**< reference: OMX_CONFIG_CONTAINERNODEIDTYPE */
- OMX_IndexParamMetadataFilterType, /**< reference: OMX_PARAM_METADATAFILTERTYPE */
- OMX_IndexParamMetadataKeyFilter, /**< reference: OMX_PARAM_METADATAFILTERTYPE */
- OMX_IndexConfigPriorityMgmt, /**< reference: OMX_PRIORITYMGMTTYPE */
- OMX_IndexParamStandardComponentRole, /**< reference: OMX_PARAM_COMPONENTROLETYPE */
-
- OMX_IndexPortStartUnused = 0x02000000,
- OMX_IndexParamPortDefinition, /**< reference: OMX_PARAM_PORTDEFINITIONTYPE */
- OMX_IndexParamCompBufferSupplier, /**< reference: OMX_PARAM_BUFFERSUPPLIERTYPE */
- OMX_IndexReservedStartUnused = 0x03000000,
-
- /* Audio parameters and configurations */
- OMX_IndexAudioStartUnused = 0x04000000,
- OMX_IndexParamAudioPortFormat, /**< reference: OMX_AUDIO_PARAM_PORTFORMATTYPE */
- OMX_IndexParamAudioPcm, /**< reference: OMX_AUDIO_PARAM_PCMMODETYPE */
- OMX_IndexParamAudioAac, /**< reference: OMX_AUDIO_PARAM_AACPROFILETYPE */
- OMX_IndexParamAudioRa, /**< reference: OMX_AUDIO_PARAM_RATYPE */
- OMX_IndexParamAudioMp3, /**< reference: OMX_AUDIO_PARAM_MP3TYPE */
- OMX_IndexParamAudioAdpcm, /**< reference: OMX_AUDIO_PARAM_ADPCMTYPE */
- OMX_IndexParamAudioG723, /**< reference: OMX_AUDIO_PARAM_G723TYPE */
- OMX_IndexParamAudioG729, /**< reference: OMX_AUDIO_PARAM_G729TYPE */
- OMX_IndexParamAudioAmr, /**< reference: OMX_AUDIO_PARAM_AMRTYPE */
- OMX_IndexParamAudioWma, /**< reference: OMX_AUDIO_PARAM_WMATYPE */
- OMX_IndexParamAudioSbc, /**< reference: OMX_AUDIO_PARAM_SBCTYPE */
- OMX_IndexParamAudioMidi, /**< reference: OMX_AUDIO_PARAM_MIDITYPE */
- OMX_IndexParamAudioGsm_FR, /**< reference: OMX_AUDIO_PARAM_GSMFRTYPE */
- OMX_IndexParamAudioMidiLoadUserSound, /**< reference: OMX_AUDIO_PARAM_MIDILOADUSERSOUNDTYPE */
- OMX_IndexParamAudioG726, /**< reference: OMX_AUDIO_PARAM_G726TYPE */
- OMX_IndexParamAudioGsm_EFR, /**< reference: OMX_AUDIO_PARAM_GSMEFRTYPE */
- OMX_IndexParamAudioGsm_HR, /**< reference: OMX_AUDIO_PARAM_GSMHRTYPE */
- OMX_IndexParamAudioPdc_FR, /**< reference: OMX_AUDIO_PARAM_PDCFRTYPE */
- OMX_IndexParamAudioPdc_EFR, /**< reference: OMX_AUDIO_PARAM_PDCEFRTYPE */
- OMX_IndexParamAudioPdc_HR, /**< reference: OMX_AUDIO_PARAM_PDCHRTYPE */
- OMX_IndexParamAudioTdma_FR, /**< reference: OMX_AUDIO_PARAM_TDMAFRTYPE */
- OMX_IndexParamAudioTdma_EFR, /**< reference: OMX_AUDIO_PARAM_TDMAEFRTYPE */
- OMX_IndexParamAudioQcelp8, /**< reference: OMX_AUDIO_PARAM_QCELP8TYPE */
- OMX_IndexParamAudioQcelp13, /**< reference: OMX_AUDIO_PARAM_QCELP13TYPE */
- OMX_IndexParamAudioEvrc, /**< reference: OMX_AUDIO_PARAM_EVRCTYPE */
- OMX_IndexParamAudioSmv, /**< reference: OMX_AUDIO_PARAM_SMVTYPE */
- OMX_IndexParamAudioVorbis, /**< reference: OMX_AUDIO_PARAM_VORBISTYPE */
-
- OMX_IndexConfigAudioMidiImmediateEvent, /**< reference: OMX_AUDIO_CONFIG_MIDIIMMEDIATEEVENTTYPE */
- OMX_IndexConfigAudioMidiControl, /**< reference: OMX_AUDIO_CONFIG_MIDICONTROLTYPE */
- OMX_IndexConfigAudioMidiSoundBankProgram, /**< reference: OMX_AUDIO_CONFIG_MIDISOUNDBANKPROGRAMTYPE */
- OMX_IndexConfigAudioMidiStatus, /**< reference: OMX_AUDIO_CONFIG_MIDISTATUSTYPE */
- OMX_IndexConfigAudioMidiMetaEvent, /**< reference: OMX_AUDIO_CONFIG_MIDIMETAEVENTTYPE */
- OMX_IndexConfigAudioMidiMetaEventData, /**< reference: OMX_AUDIO_CONFIG_MIDIMETAEVENTDATATYPE */
- OMX_IndexConfigAudioVolume, /**< reference: OMX_AUDIO_CONFIG_VOLUMETYPE */
- OMX_IndexConfigAudioBalance, /**< reference: OMX_AUDIO_CONFIG_BALANCETYPE */
- OMX_IndexConfigAudioChannelMute, /**< reference: OMX_AUDIO_CONFIG_CHANNELMUTETYPE */
- OMX_IndexConfigAudioMute, /**< reference: OMX_AUDIO_CONFIG_MUTETYPE */
- OMX_IndexConfigAudioLoudness, /**< reference: OMX_AUDIO_CONFIG_LOUDNESSTYPE */
- OMX_IndexConfigAudioEchoCancelation, /**< reference: OMX_AUDIO_CONFIG_ECHOCANCELATIONTYPE */
- OMX_IndexConfigAudioNoiseReduction, /**< reference: OMX_AUDIO_CONFIG_NOISEREDUCTIONTYPE */
- OMX_IndexConfigAudioBass, /**< reference: OMX_AUDIO_CONFIG_BASSTYPE */
- OMX_IndexConfigAudioTreble, /**< reference: OMX_AUDIO_CONFIG_TREBLETYPE */
- OMX_IndexConfigAudioStereoWidening, /**< reference: OMX_AUDIO_CONFIG_STEREOWIDENINGTYPE */
- OMX_IndexConfigAudioChorus, /**< reference: OMX_AUDIO_CONFIG_CHORUSTYPE */
- OMX_IndexConfigAudioEqualizer, /**< reference: OMX_AUDIO_CONFIG_EQUALIZERTYPE */
- OMX_IndexConfigAudioReverberation, /**< reference: OMX_AUDIO_CONFIG_REVERBERATIONTYPE */
- OMX_IndexConfigAudioChannelVolume, /**< reference: OMX_AUDIO_CONFIG_CHANNELVOLUMETYPE */
-
- /* Image specific parameters and configurations */
- OMX_IndexImageStartUnused = 0x05000000,
- OMX_IndexParamImagePortFormat, /**< reference: OMX_IMAGE_PARAM_PORTFORMATTYPE */
- OMX_IndexParamFlashControl, /**< reference: OMX_IMAGE_PARAM_FLASHCONTROLTYPE */
- OMX_IndexConfigFocusControl, /**< reference: OMX_IMAGE_CONFIG_FOCUSCONTROLTYPE */
- OMX_IndexParamQFactor, /**< reference: OMX_IMAGE_PARAM_QFACTORTYPE */
- OMX_IndexParamQuantizationTable, /**< reference: OMX_IMAGE_PARAM_QUANTIZATIONTABLETYPE */
- OMX_IndexParamHuffmanTable, /**< reference: OMX_IMAGE_PARAM_HUFFMANTTABLETYPE */
- OMX_IndexConfigFlashControl, /**< reference: OMX_IMAGE_PARAM_FLASHCONTROLTYPE */
-
- /* Video specific parameters and configurations */
- OMX_IndexVideoStartUnused = 0x06000000,
- OMX_IndexParamVideoPortFormat, /**< reference: OMX_VIDEO_PARAM_PORTFORMATTYPE */
- OMX_IndexParamVideoQuantization, /**< reference: OMX_VIDEO_PARAM_QUANTIZATIONTYPE */
- OMX_IndexParamVideoFastUpdate, /**< reference: OMX_VIDEO_PARAM_VIDEOFASTUPDATETYPE */
- OMX_IndexParamVideoBitrate, /**< reference: OMX_VIDEO_PARAM_BITRATETYPE */
- OMX_IndexParamVideoMotionVector, /**< reference: OMX_VIDEO_PARAM_MOTIONVECTORTYPE */
- OMX_IndexParamVideoIntraRefresh, /**< reference: OMX_VIDEO_PARAM_INTRAREFRESHTYPE */
- OMX_IndexParamVideoErrorCorrection, /**< reference: OMX_VIDEO_PARAM_ERRORCORRECTIONTYPE */
- OMX_IndexParamVideoVBSMC, /**< reference: OMX_VIDEO_PARAM_VBSMCTYPE */
- OMX_IndexParamVideoMpeg2, /**< reference: OMX_VIDEO_PARAM_MPEG2TYPE */
- OMX_IndexParamVideoMpeg4, /**< reference: OMX_VIDEO_PARAM_MPEG4TYPE */
- OMX_IndexParamVideoWmv, /**< reference: OMX_VIDEO_PARAM_WMVTYPE */
- OMX_IndexParamVideoRv, /**< reference: OMX_VIDEO_PARAM_RVTYPE */
- OMX_IndexParamVideoAvc, /**< reference: OMX_VIDEO_PARAM_AVCTYPE */
- OMX_IndexParamVideoH263, /**< reference: OMX_VIDEO_PARAM_H263TYPE */
- OMX_IndexParamVideoProfileLevelQuerySupported, /**< reference: OMX_VIDEO_PARAM_PROFILELEVELTYPE */
- OMX_IndexParamVideoProfileLevelCurrent, /**< reference: OMX_VIDEO_PARAM_PROFILELEVELTYPE */
- OMX_IndexConfigVideoBitrate, /**< reference: OMX_VIDEO_CONFIG_BITRATETYPE */
- OMX_IndexConfigVideoFramerate, /**< reference: OMX_CONFIG_FRAMERATETYPE */
- OMX_IndexConfigVideoIntraVOPRefresh, /**< reference: OMX_CONFIG_INTRAREFRESHVOPTYPE */
- OMX_IndexConfigVideoIntraMBRefresh, /**< reference: OMX_CONFIG_MACROBLOCKERRORMAPTYPE */
- OMX_IndexConfigVideoMBErrorReporting, /**< reference: OMX_CONFIG_MBERRORREPORTINGTYPE */
- OMX_IndexParamVideoMacroblocksPerFrame, /**< reference: OMX_PARAM_MACROBLOCKSTYPE */
- OMX_IndexConfigVideoMacroBlockErrorMap, /**< reference: OMX_CONFIG_MACROBLOCKERRORMAPTYPE */
- OMX_IndexParamVideoSliceFMO, /**< reference: OMX_VIDEO_PARAM_AVCSLICEFMO */
- OMX_IndexConfigVideoAVCIntraPeriod, /**< reference: OMX_VIDEO_CONFIG_AVCINTRAPERIOD */
- OMX_IndexConfigVideoNalSize, /**< reference: OMX_VIDEO_CONFIG_NALSIZE */
-
- /* Image & Video common Configurations */
- OMX_IndexCommonStartUnused = 0x07000000,
- OMX_IndexParamCommonDeblocking, /**< reference: OMX_PARAM_DEBLOCKINGTYPE */
- OMX_IndexParamCommonSensorMode, /**< reference: OMX_PARAM_SENSORMODETYPE */
- OMX_IndexParamCommonInterleave, /**< reference: OMX_PARAM_INTERLEAVETYPE */
- OMX_IndexConfigCommonColorFormatConversion, /**< reference: OMX_CONFIG_COLORCONVERSIONTYPE */
- OMX_IndexConfigCommonScale, /**< reference: OMX_CONFIG_SCALEFACTORTYPE */
- OMX_IndexConfigCommonImageFilter, /**< reference: OMX_CONFIG_IMAGEFILTERTYPE */
- OMX_IndexConfigCommonColorEnhancement, /**< reference: OMX_CONFIG_COLORENHANCEMENTTYPE */
- OMX_IndexConfigCommonColorKey, /**< reference: OMX_CONFIG_COLORKEYTYPE */
- OMX_IndexConfigCommonColorBlend, /**< reference: OMX_CONFIG_COLORBLENDTYPE */
- OMX_IndexConfigCommonFrameStabilisation,/**< reference: OMX_CONFIG_FRAMESTABTYPE */
- OMX_IndexConfigCommonRotate, /**< reference: OMX_CONFIG_ROTATIONTYPE */
- OMX_IndexConfigCommonMirror, /**< reference: OMX_CONFIG_MIRRORTYPE */
- OMX_IndexConfigCommonOutputPosition, /**< reference: OMX_CONFIG_POINTTYPE */
- OMX_IndexConfigCommonInputCrop, /**< reference: OMX_CONFIG_RECTTYPE */
- OMX_IndexConfigCommonOutputCrop, /**< reference: OMX_CONFIG_RECTTYPE */
- OMX_IndexConfigCommonDigitalZoom, /**< reference: OMX_CONFIG_SCALEFACTORTYPE */
- OMX_IndexConfigCommonOpticalZoom, /**< reference: OMX_CONFIG_SCALEFACTORTYPE*/
- OMX_IndexConfigCommonWhiteBalance, /**< reference: OMX_CONFIG_WHITEBALCONTROLTYPE */
- OMX_IndexConfigCommonExposure, /**< reference: OMX_CONFIG_EXPOSURECONTROLTYPE */
- OMX_IndexConfigCommonContrast, /**< reference: OMX_CONFIG_CONTRASTTYPE */
- OMX_IndexConfigCommonBrightness, /**< reference: OMX_CONFIG_BRIGHTNESSTYPE */
- OMX_IndexConfigCommonBacklight, /**< reference: OMX_CONFIG_BACKLIGHTTYPE */
- OMX_IndexConfigCommonGamma, /**< reference: OMX_CONFIG_GAMMATYPE */
- OMX_IndexConfigCommonSaturation, /**< reference: OMX_CONFIG_SATURATIONTYPE */
- OMX_IndexConfigCommonLightness, /**< reference: OMX_CONFIG_LIGHTNESSTYPE */
- OMX_IndexConfigCommonExclusionRect, /**< reference: OMX_CONFIG_RECTTYPE */
- OMX_IndexConfigCommonDithering, /**< reference: OMX_CONFIG_DITHERTYPE */
- OMX_IndexConfigCommonPlaneBlend, /**< reference: OMX_CONFIG_PLANEBLENDTYPE */
- OMX_IndexConfigCommonExposureValue, /**< reference: OMX_CONFIG_EXPOSUREVALUETYPE */
- OMX_IndexConfigCommonOutputSize, /**< reference: OMX_FRAMESIZETYPE */
- OMX_IndexParamCommonExtraQuantData, /**< reference: OMX_OTHER_EXTRADATATYPE */
- OMX_IndexConfigCommonFocusRegion, /**< reference: OMX_CONFIG_FOCUSREGIONTYPE */
- OMX_IndexConfigCommonFocusStatus, /**< reference: OMX_PARAM_FOCUSSTATUSTYPE */
- OMX_IndexConfigCommonTransitionEffect, /**< reference: OMX_CONFIG_TRANSITIONEFFECTTYPE */
-
- /* Reserved Configuration range */
- OMX_IndexOtherStartUnused = 0x08000000,
- OMX_IndexParamOtherPortFormat, /**< reference: OMX_OTHER_PARAM_PORTFORMATTYPE */
- OMX_IndexConfigOtherPower, /**< reference: OMX_OTHER_CONFIG_POWERTYPE */
- OMX_IndexConfigOtherStats, /**< reference: OMX_OTHER_CONFIG_STATSTYPE */
-
-
- /* Reserved Time range */
- OMX_IndexTimeStartUnused = 0x09000000,
- OMX_IndexConfigTimeScale, /**< reference: OMX_TIME_CONFIG_SCALETYPE */
- OMX_IndexConfigTimeClockState, /**< reference: OMX_TIME_CONFIG_CLOCKSTATETYPE */
- OMX_IndexConfigTimeActiveRefClock, /**< reference: OMX_TIME_CONFIG_ACTIVEREFCLOCKTYPE */
- OMX_IndexConfigTimeCurrentMediaTime, /**< reference: OMX_TIME_CONFIG_TIMESTAMPTYPE (read only) */
- OMX_IndexConfigTimeCurrentWallTime, /**< reference: OMX_TIME_CONFIG_TIMESTAMPTYPE (read only) */
- OMX_IndexConfigTimeCurrentAudioReference, /**< reference: OMX_TIME_CONFIG_TIMESTAMPTYPE (write only) */
- OMX_IndexConfigTimeCurrentVideoReference, /**< reference: OMX_TIME_CONFIG_TIMESTAMPTYPE (write only) */
- OMX_IndexConfigTimeMediaTimeRequest, /**< reference: OMX_TIME_CONFIG_MEDIATIMEREQUESTTYPE (write only) */
- OMX_IndexConfigTimeClientStartTime, /**<reference: OMX_TIME_CONFIG_TIMESTAMPTYPE (write only) */
- OMX_IndexConfigTimePosition, /**< reference: OMX_TIME_CONFIG_TIMESTAMPTYPE */
- OMX_IndexConfigTimeSeekMode, /**< reference: OMX_TIME_CONFIG_SEEKMODETYPE */
-
-
- OMX_IndexKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- /* Vendor specific area */
- OMX_IndexVendorStartUnused = 0x7F000000,
- /* Vendor specific structures should be in the range of 0x7F000000
- to 0x7FFFFFFE. This range is not broken out by vendor, so
- private indexes are not guaranteed unique and therefore should
- only be sent to the appropriate component. */
-
- OMX_IndexMax = 0x7FFFFFFF
-
-} OMX_INDEXTYPE;
-
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
-#endif
-/* File EOF */
diff --git a/include/media/stagefright/openmax/OMX_Other.h b/include/media/stagefright/openmax/OMX_Other.h
deleted file mode 100644
index efbce83..0000000
--- a/include/media/stagefright/openmax/OMX_Other.h
+++ /dev/null
@@ -1,354 +0,0 @@
-/* ------------------------------------------------------------------
- * Copyright (C) 1998-2009 PacketVideo
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either
- * express or implied.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- * -------------------------------------------------------------------
- */
-/*
- * Copyright (c) 2008 The Khronos Group Inc.
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files (the
- * "Software"), to deal in the Software without restriction, including
- * without limitation the rights to use, copy, modify, merge, publish,
- * distribute, sublicense, and/or sell copies of the Software, and to
- * permit persons to whom the Software is furnished to do so, subject
- * to the following conditions:
- * The above copyright notice and this permission notice shall be included
- * in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS
- * OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY
- * CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
- * TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
- * SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- *
- */
-
-/** @file OMX_Other.h - OpenMax IL version 1.1.2
- * The structures needed by Other components to exchange
- * parameters and configuration data with the components.
- */
-
-#ifndef OMX_Other_h
-#define OMX_Other_h
-
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-
-/* Each OMX header must include all required header files to allow the
- * header to compile without errors. The includes below are required
- * for this header file to compile successfully
- */
-
-#include <OMX_Core.h>
-
-
-/**
- * Enumeration of possible data types which match to multiple domains or no
- * domain at all. For types which are vendor specific, a value above
- * OMX_OTHER_VENDORTSTART should be used.
- */
-typedef enum OMX_OTHER_FORMATTYPE {
- OMX_OTHER_FormatTime = 0, /**< Transmission of various timestamps, elapsed time,
- time deltas, etc */
- OMX_OTHER_FormatPower, /**< Perhaps used for enabling/disabling power
- management, setting clocks? */
- OMX_OTHER_FormatStats, /**< Could be things such as frame rate, frames
- dropped, etc */
- OMX_OTHER_FormatBinary, /**< Arbitrary binary data */
- OMX_OTHER_FormatVendorReserved = 1000, /**< Starting value for vendor specific
- formats */
-
- OMX_OTHER_FormatKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_OTHER_FormatVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_OTHER_FormatMax = 0x7FFFFFFF
-} OMX_OTHER_FORMATTYPE;
-
-/**
- * Enumeration of seek modes.
- */
-typedef enum OMX_TIME_SEEKMODETYPE {
- OMX_TIME_SeekModeFast = 0, /**< Prefer seeking to an approximation
- * of the requested seek position over
- * the actual seek position if it
- * results in a faster seek. */
- OMX_TIME_SeekModeAccurate, /**< Prefer seeking to the actual seek
- * position over an approximation
- * of the requested seek position even
- * if it results in a slower seek. */
- OMX_TIME_SeekModeKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_TIME_SeekModeVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_TIME_SeekModeMax = 0x7FFFFFFF
-} OMX_TIME_SEEKMODETYPE;
-
-/* Structure representing the seekmode of the component */
-typedef struct OMX_TIME_CONFIG_SEEKMODETYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_TIME_SEEKMODETYPE eType; /**< The seek mode */
-} OMX_TIME_CONFIG_SEEKMODETYPE;
-
-/** Structure representing a time stamp used with the following configs
- * on the Clock Component (CC):
- *
- * OMX_IndexConfigTimeCurrentWallTime: query of the CCÂ’s current wall
- * time
- * OMX_IndexConfigTimeCurrentMediaTime: query of the CCÂ’s current media
- * time
- * OMX_IndexConfigTimeCurrentAudioReference and
- * OMX_IndexConfigTimeCurrentVideoReference: audio/video reference
- * clock sending SC its reference time
- * OMX_IndexConfigTimeClientStartTime: a Clock Component client sends
- * this structure to the Clock Component via a SetConfig on its
- * client port when it receives a buffer with
- * OMX_BUFFERFLAG_STARTTIME set. It must use the timestamp
- * specified by that buffer for nStartTimestamp.
- *
- * ItÂ’s also used with the following config on components in general:
- *
- * OMX_IndexConfigTimePosition: IL client querying component position
- * (GetConfig) or commanding a component to seek to the given location
- * (SetConfig)
- */
-typedef struct OMX_TIME_CONFIG_TIMESTAMPTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version
- * information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_TICKS nTimestamp; /**< timestamp .*/
-} OMX_TIME_CONFIG_TIMESTAMPTYPE;
-
-/** Enumeration of possible reference clocks to the media time. */
-typedef enum OMX_TIME_UPDATETYPE {
- OMX_TIME_UpdateRequestFulfillment, /**< Update is the fulfillment of a media time request. */
- OMX_TIME_UpdateScaleChanged, /**< Update was generated because the scale chagned. */
- OMX_TIME_UpdateClockStateChanged, /**< Update was generated because the clock state changed. */
- OMX_TIME_UpdateKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_TIME_UpdateVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_TIME_UpdateMax = 0x7FFFFFFF
-} OMX_TIME_UPDATETYPE;
-
-/** Enumeration of possible reference clocks to the media time. */
-typedef enum OMX_TIME_REFCLOCKTYPE {
- OMX_TIME_RefClockNone, /**< Use no references. */
- OMX_TIME_RefClockAudio, /**< Use references sent through OMX_IndexConfigTimeCurrentAudioReference */
- OMX_TIME_RefClockVideo, /**< Use references sent through OMX_IndexConfigTimeCurrentVideoReference */
- OMX_TIME_RefClockKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_TIME_RefClockVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_TIME_RefClockMax = 0x7FFFFFFF
-} OMX_TIME_REFCLOCKTYPE;
-
-/** Enumeration of clock states. */
-typedef enum OMX_TIME_CLOCKSTATE {
- OMX_TIME_ClockStateRunning, /**< Clock running. */
- OMX_TIME_ClockStateWaitingForStartTime, /**< Clock waiting until the
- * prescribed clients emit their
- * start time. */
- OMX_TIME_ClockStateStopped, /**< Clock stopped. */
- OMX_TIME_ClockStateKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_TIME_ClockStateVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_TIME_ClockStateMax = 0x7FFFFFFF
-} OMX_TIME_CLOCKSTATE;
-
-/** Structure representing a media time request to the clock component.
- *
- * A client component sends this structure to the Clock Component via a SetConfig
- * on its client port to specify a media timestamp the Clock Component
- * should emit. The Clock Component should fulfill the request by sending a
- * OMX_TIME_MEDIATIMETYPE when its media clock matches the requested
- * timestamp.
- *
- * The client may require a media time request be fulfilled slightly
- * earlier than the media time specified. In this case the client specifies
- * an offset which is equal to the difference between wall time corresponding
- * to the requested media time and the wall time when it will be
- * fulfilled.
- *
- * A client component may uses these requests and the OMX_TIME_MEDIATIMETYPE to
- * time events according to timestamps. If a client must perform an operation O at
- * a time T (e.g. deliver a video frame at its corresponding timestamp), it makes a
- * media time request at T (perhaps specifying an offset to ensure the request fulfillment
- * is a little early). When the clock component passes the resulting OMX_TIME_MEDIATIMETYPE
- * structure back to the client component, the client may perform operation O (perhaps having
- * to wait a slight amount more time itself as specified by the return values).
- */
-
-typedef struct OMX_TIME_CONFIG_MEDIATIMEREQUESTTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< port that this structure applies to */
- OMX_PTR pClientPrivate; /**< Client private data to disabiguate this media time
- * from others (e.g. the number of the frame to deliver).
- * Duplicated in the media time structure that fulfills
- * this request. A value of zero is reserved for time scale
- * updates. */
- OMX_TICKS nMediaTimestamp; /**< Media timestamp requested.*/
- OMX_TICKS nOffset; /**< Amount of wall clock time by which this
- * request should be fulfilled early */
-} OMX_TIME_CONFIG_MEDIATIMEREQUESTTYPE;
-
-/**< Structure sent from the clock component client either when fulfilling
- * a media time request or when the time scale has changed.
- *
- * In the former case the Clock Component fills this structure and times its emission
- * to a client component (via the client port) according to the corresponding media
- * time request sent by the client. The Clock Component should time the emission to occur
- * when the requested timestamp matches the Clock Component's media time but also the
- * prescribed offset early.
- *
- * Upon scale changes the clock component clears the nClientPrivate data, sends the current
- * media time and sets the nScale to the new scale via the client port. It emits a
- * OMX_TIME_MEDIATIMETYPE to all clients independent of any requests. This allows clients to
- * alter processing to accomodate scaling. For instance a video component might skip inter-frames
- * in the case of extreme fastforward. Likewise an audio component might add or remove samples
- * from an audio frame to scale audio data.
- *
- * It is expected that some clock components may not be able to fulfill requests
- * at exactly the prescribed time. This is acceptable so long as the request is
- * fulfilled at least as early as described and not later. This structure provides
- * fields the client may use to wait for the remaining time.
- *
- * The client may use either the nOffset or nWallTimeAtMedia fields to determine the
- * wall time until the nMediaTimestamp actually occurs. In the latter case the
- * client can get a more accurate value for offset by getting the current wall
- * from the cloc component and subtracting it from nWallTimeAtMedia.
- */
-
-typedef struct OMX_TIME_MEDIATIMETYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nClientPrivate; /**< Client private data to disabiguate this media time
- * from others. Copied from the media time request.
- * A value of zero is reserved for time scale updates. */
- OMX_TIME_UPDATETYPE eUpdateType; /**< Reason for the update */
- OMX_TICKS nMediaTimestamp; /**< Media time requested. If no media time was
- * requested then this is the current media time. */
- OMX_TICKS nOffset; /**< Amount of wall clock time by which this
- * request was actually fulfilled early */
-
- OMX_TICKS nWallTimeAtMediaTime; /**< Wall time corresponding to nMediaTimeStamp.
- * A client may compare this value to current
- * media time obtained from the Clock Component to determine
- * the wall time until the media timestamp is really
- * current. */
- OMX_S32 xScale; /**< Current media time scale in Q16 format. */
- OMX_TIME_CLOCKSTATE eState; /* Seeking Change. Added 7/12.*/
- /**< State of the media time. */
-} OMX_TIME_MEDIATIMETYPE;
-
-/** Structure representing the current media time scale factor. Applicable only to clock
- * component, other components see scale changes via OMX_TIME_MEDIATIMETYPE buffers sent via
- * the clock component client ports. Upon recieving this config the clock component changes
- * the rate by which the media time increases or decreases effectively implementing trick modes.
- */
-typedef struct OMX_TIME_CONFIG_SCALETYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_S32 xScale; /**< This is a value in Q16 format which is used for
- * scaling the media time */
-} OMX_TIME_CONFIG_SCALETYPE;
-
-/** Bits used to identify a clock port. Used in OMX_TIME_CONFIG_CLOCKSTATETYPEÂ’s nWaitMask field */
-#define OMX_CLOCKPORT0 0x00000001
-#define OMX_CLOCKPORT1 0x00000002
-#define OMX_CLOCKPORT2 0x00000004
-#define OMX_CLOCKPORT3 0x00000008
-#define OMX_CLOCKPORT4 0x00000010
-#define OMX_CLOCKPORT5 0x00000020
-#define OMX_CLOCKPORT6 0x00000040
-#define OMX_CLOCKPORT7 0x00000080
-
-/** Structure representing the current mode of the media clock.
- * IL Client uses this config to change or query the mode of the
- * media clock of the clock component. Applicable only to clock
- * component.
- *
- * On a SetConfig if eState is OMX_TIME_ClockStateRunning media time
- * starts immediately at the prescribed start time. If
- * OMX_TIME_ClockStateWaitingForStartTime the Clock Component ignores
- * the given nStartTime and waits for all clients specified in the
- * nWaitMask to send starttimes (via
- * OMX_IndexConfigTimeClientStartTime). The Clock Component then starts
- * the media clock using the earliest start time supplied. */
-typedef struct OMX_TIME_CONFIG_CLOCKSTATETYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version
- * information */
- OMX_TIME_CLOCKSTATE eState; /**< State of the media time. */
- OMX_TICKS nStartTime; /**< Start time of the media time. */
- OMX_TICKS nOffset; /**< Time to offset the media time by
- * (e.g. preroll). Media time will be
- * reported to be nOffset ticks earlier.
- */
- OMX_U32 nWaitMask; /**< Mask of OMX_CLOCKPORT values. */
-} OMX_TIME_CONFIG_CLOCKSTATETYPE;
-
-/** Structure representing the reference clock currently being used to
- * compute media time. IL client uses this config to change or query the
- * clock component's active reference clock */
-typedef struct OMX_TIME_CONFIG_ACTIVEREFCLOCKTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_TIME_REFCLOCKTYPE eClock; /**< Reference clock used to compute media time */
-} OMX_TIME_CONFIG_ACTIVEREFCLOCKTYPE;
-
-/** Descriptor for setting specifics of power type.
- * Note: this structure is listed for backwards compatibility. */
-typedef struct OMX_OTHER_CONFIG_POWERTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_BOOL bEnablePM; /**< Flag to enable Power Management */
-} OMX_OTHER_CONFIG_POWERTYPE;
-
-
-/** Descriptor for setting specifics of stats type.
- * Note: this structure is listed for backwards compatibility. */
-typedef struct OMX_OTHER_CONFIG_STATSTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- /* what goes here */
-} OMX_OTHER_CONFIG_STATSTYPE;
-
-
-/**
- * The PortDefinition structure is used to define all of the parameters
- * necessary for the compliant component to setup an input or an output other
- * path.
- */
-typedef struct OMX_OTHER_PORTDEFINITIONTYPE {
- OMX_OTHER_FORMATTYPE eFormat; /**< Type of data expected for this channel */
-} OMX_OTHER_PORTDEFINITIONTYPE;
-
-/** Port format parameter. This structure is used to enumerate
- * the various data input/output format supported by the port.
- */
-typedef struct OMX_OTHER_PARAM_PORTFORMATTYPE {
- OMX_U32 nSize; /**< size of the structure in bytes */
- OMX_VERSIONTYPE nVersion; /**< OMX specification version information */
- OMX_U32 nPortIndex; /**< Indicates which port to set */
- OMX_U32 nIndex; /**< Indicates the enumeration index for the format from 0x0 to N-1 */
- OMX_OTHER_FORMATTYPE eFormat; /**< Type of data expected for this channel */
-} OMX_OTHER_PARAM_PORTFORMATTYPE;
-
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
-#endif
-/* File EOF */
diff --git a/include/media/stagefright/openmax/OMX_Types.h b/include/media/stagefright/openmax/OMX_Types.h
deleted file mode 100644
index 03fd4bc..0000000
--- a/include/media/stagefright/openmax/OMX_Types.h
+++ /dev/null
@@ -1,365 +0,0 @@
-/* ------------------------------------------------------------------
- * Copyright (C) 1998-2009 PacketVideo
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either
- * express or implied.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- * -------------------------------------------------------------------
- */
-/*
- * Copyright (c) 2008 The Khronos Group Inc.
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files (the
- * "Software"), to deal in the Software without restriction, including
- * without limitation the rights to use, copy, modify, merge, publish,
- * distribute, sublicense, and/or sell copies of the Software, and to
- * permit persons to whom the Software is furnished to do so, subject
- * to the following conditions:
- * The above copyright notice and this permission notice shall be included
- * in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS
- * OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY
- * CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
- * TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
- * SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- *
- */
-
-/** OMX_Types.h - OpenMax IL version 1.1.2
- * The OMX_Types header file contains the primitive type definitions used by
- * the core, the application and the component. This file may need to be
- * modified to be used on systems that do not have "char" set to 8 bits,
- * "short" set to 16 bits and "long" set to 32 bits.
- */
-
-#ifndef OMX_Types_h
-#define OMX_Types_h
-
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-/** The OMX_API and OMX_APIENTRY are platform specific definitions used
- * to declare OMX function prototypes. They are modified to meet the
- * requirements for a particular platform */
-#ifdef __SYMBIAN32__
-# ifdef __OMX_EXPORTS
-# define OMX_API __declspec(dllexport)
-# else
-# ifdef _WIN32
-# define OMX_API __declspec(dllexport)
-# else
-# define OMX_API __declspec(dllimport)
-# endif
-# endif
-#else
-# ifdef _WIN32
-# ifdef __OMX_EXPORTS
-# define OMX_API __declspec(dllexport)
-# else
-//# define OMX_API __declspec(dllimport)
-#define OMX_API
-# endif
-# else
-# ifdef __OMX_EXPORTS
-# define OMX_API
-# else
-# define OMX_API extern
-# endif
-# endif
-#endif
-
-#ifndef OMX_APIENTRY
-#define OMX_APIENTRY
-#endif
-
-/** OMX_IN is used to identify inputs to an OMX function. This designation
- will also be used in the case of a pointer that points to a parameter
- that is used as an output. */
-#ifndef OMX_IN
-#define OMX_IN
-#endif
-
-/** OMX_OUT is used to identify outputs from an OMX function. This
- designation will also be used in the case of a pointer that points
- to a parameter that is used as an input. */
-#ifndef OMX_OUT
-#define OMX_OUT
-#endif
-
-
-/** OMX_INOUT is used to identify parameters that may be either inputs or
- outputs from an OMX function at the same time. This designation will
- also be used in the case of a pointer that points to a parameter that
- is used both as an input and an output. */
-#ifndef OMX_INOUT
-#define OMX_INOUT
-#endif
-
-/** OMX_ALL is used to as a wildcard to select all entities of the same type
- * when specifying the index, or referring to a object by an index. (i.e.
- * use OMX_ALL to indicate all N channels). When used as a port index
- * for a config or parameter this OMX_ALL denotes that the config or
- * parameter applies to the entire component not just one port. */
-#define OMX_ALL 0xFFFFFFFF
-
-/** In the following we define groups that help building doxygen documentation */
-
-/** @defgroup core OpenMAX IL core
- * Functions and structure related to the OMX IL core
- */
-
- /** @defgroup comp OpenMAX IL component
- * Functions and structure related to the OMX IL component
- */
-
-/** @defgroup rpm Resource and Policy Management
- * Structures for resource and policy management of components
- */
-
-/** @defgroup buf Buffer Management
- * Buffer handling functions and structures
- */
-
-/** @defgroup tun Tunneling
- * @ingroup core comp
- * Structures and functions to manage tunnels among component ports
- */
-
-/** @defgroup cp Content Pipes
- * @ingroup core
- */
-
- /** @defgroup metadata Metadata handling
- *
- */
-
-/** OMX_U8 is an 8 bit unsigned quantity that is byte aligned */
-typedef unsigned char OMX_U8;
-
-/** OMX_S8 is an 8 bit signed quantity that is byte aligned */
-typedef signed char OMX_S8;
-
-/** OMX_U16 is a 16 bit unsigned quantity that is 16 bit word aligned */
-typedef unsigned short OMX_U16;
-
-/** OMX_S16 is a 16 bit signed quantity that is 16 bit word aligned */
-typedef signed short OMX_S16;
-
-/** OMX_U32 is a 32 bit unsigned quantity that is 32 bit word aligned */
-typedef unsigned long OMX_U32;
-
-/** OMX_S32 is a 32 bit signed quantity that is 32 bit word aligned */
-typedef signed long OMX_S32;
-
-
-/* Users with compilers that cannot accept the "long long" designation should
- define the OMX_SKIP64BIT macro. It should be noted that this may cause
- some components to fail to compile if the component was written to require
- 64 bit integral types. However, these components would NOT compile anyway
- since the compiler does not support the way the component was written.
-*/
-#ifndef OMX_SKIP64BIT
-#ifdef __SYMBIAN32__
-/** OMX_U64 is a 64 bit unsigned quantity that is 64 bit word aligned */
-typedef unsigned long long OMX_U64;
-
-/** OMX_S64 is a 64 bit signed quantity that is 64 bit word aligned */
-typedef signed long long OMX_S64;
-
-#elif defined(WIN32)
-
-/** OMX_U64 is a 64 bit unsigned quantity that is 64 bit word aligned */
-typedef unsigned __int64 OMX_U64;
-
-/** OMX_S64 is a 64 bit signed quantity that is 64 bit word aligned */
-typedef signed __int64 OMX_S64;
-
-#else /* WIN32 */
-
-/** OMX_U64 is a 64 bit unsigned quantity that is 64 bit word aligned */
-typedef unsigned long long OMX_U64;
-
-/** OMX_S64 is a 64 bit signed quantity that is 64 bit word aligned */
-typedef signed long long OMX_S64;
-
-#endif /* WIN32 */
-#endif
-
-
-/** The OMX_BOOL type is intended to be used to represent a true or a false
- value when passing parameters to and from the OMX core and components. The
- OMX_BOOL is a 32 bit quantity and is aligned on a 32 bit word boundary.
- */
-typedef enum OMX_BOOL {
- OMX_FALSE = 0,
- OMX_TRUE = !OMX_FALSE,
- OMX_BOOL_MAX = 0x7FFFFFFF
-} OMX_BOOL;
-
-/** The OMX_PTR type is intended to be used to pass pointers between the OMX
- applications and the OMX Core and components. This is a 32 bit pointer and
- is aligned on a 32 bit boundary.
- */
-typedef void* OMX_PTR;
-
-/** The OMX_STRING type is intended to be used to pass "C" type strings between
- the application and the core and component. The OMX_STRING type is a 32
- bit pointer to a zero terminated string. The pointer is word aligned and
- the string is byte aligned.
- */
-typedef char* OMX_STRING;
-
-/** The OMX_BYTE type is intended to be used to pass arrays of bytes such as
- buffers between the application and the component and core. The OMX_BYTE
- type is a 32 bit pointer to a zero terminated string. The pointer is word
- aligned and the string is byte aligned.
- */
-typedef unsigned char* OMX_BYTE;
-
-/** OMX_UUIDTYPE is a very long unique identifier to uniquely identify
- at runtime. This identifier should be generated by a component in a way
- that guarantees that every instance of the identifier running on the system
- is unique. */
-typedef unsigned char OMX_UUIDTYPE[128];
-
-/** The OMX_DIRTYPE enumeration is used to indicate if a port is an input or
- an output port. This enumeration is common across all component types.
- */
-typedef enum OMX_DIRTYPE
-{
- OMX_DirInput, /**< Port is an input port */
- OMX_DirOutput, /**< Port is an output port */
- OMX_DirMax = 0x7FFFFFFF
-} OMX_DIRTYPE;
-
-/** The OMX_ENDIANTYPE enumeration is used to indicate the bit ordering
- for numerical data (i.e. big endian, or little endian).
- */
-typedef enum OMX_ENDIANTYPE
-{
- OMX_EndianBig, /**< big endian */
- OMX_EndianLittle, /**< little endian */
- OMX_EndianMax = 0x7FFFFFFF
-} OMX_ENDIANTYPE;
-
-
-/** The OMX_NUMERICALDATATYPE enumeration is used to indicate if data
- is signed or unsigned
- */
-typedef enum OMX_NUMERICALDATATYPE
-{
- OMX_NumericalDataSigned, /**< signed data */
- OMX_NumericalDataUnsigned, /**< unsigned data */
- OMX_NumercialDataMax = 0x7FFFFFFF
-} OMX_NUMERICALDATATYPE;
-
-
-/** Unsigned bounded value type */
-typedef struct OMX_BU32 {
- OMX_U32 nValue; /**< actual value */
- OMX_U32 nMin; /**< minimum for value (i.e. nValue >= nMin) */
- OMX_U32 nMax; /**< maximum for value (i.e. nValue <= nMax) */
-} OMX_BU32;
-
-
-/** Signed bounded value type */
-typedef struct OMX_BS32 {
- OMX_S32 nValue; /**< actual value */
- OMX_S32 nMin; /**< minimum for value (i.e. nValue >= nMin) */
- OMX_S32 nMax; /**< maximum for value (i.e. nValue <= nMax) */
-} OMX_BS32;
-
-
-/** Structure representing some time or duration in microseconds. This structure
- * must be interpreted as a signed 64 bit value. The quantity is signed to accommodate
- * negative deltas and preroll scenarios. The quantity is represented in microseconds
- * to accomodate high resolution timestamps (e.g. DVD presentation timestamps based
- * on a 90kHz clock) and to allow more accurate and synchronized delivery (e.g.
- * individual audio samples delivered at 192 kHz). The quantity is 64 bit to
- * accommodate a large dynamic range (signed 32 bit values would allow only for plus
- * or minus 35 minutes).
- *
- * Implementations with limited precision may convert the signed 64 bit value to
- * a signed 32 bit value internally but risk loss of precision.
- */
-#ifndef OMX_SKIP64BIT
-typedef OMX_S64 OMX_TICKS;
-#else
-typedef struct OMX_TICKS
-{
- OMX_U32 nLowPart; /** low bits of the signed 64 bit tick value */
- OMX_U32 nHighPart; /** high bits of the signed 64 bit tick value */
-} OMX_TICKS;
-#endif
-#define OMX_TICKS_PER_SECOND 1000000
-
-/** Define the public interface for the OMX Handle. The core will not use
- this value internally, but the application should only use this value.
- */
-typedef void* OMX_HANDLETYPE;
-
-typedef struct OMX_MARKTYPE
-{
- OMX_HANDLETYPE hMarkTargetComponent; /**< The component that will
- generate a mark event upon
- processing the mark. */
- OMX_PTR pMarkData; /**< Application specific data associated with
- the mark sent on a mark event to disambiguate
- this mark from others. */
-} OMX_MARKTYPE;
-
-
-/** OMX_NATIVE_DEVICETYPE is used to map a OMX video port to the
- * platform & operating specific object used to reference the display
- * or can be used by a audio port for native audio rendering */
-typedef void* OMX_NATIVE_DEVICETYPE;
-
-/** OMX_NATIVE_WINDOWTYPE is used to map a OMX video port to the
- * platform & operating specific object used to reference the window */
-typedef void* OMX_NATIVE_WINDOWTYPE;
-
-/** The OMX_VERSIONTYPE union is used to specify the version for
- a structure or component. For a component, the version is entirely
- specified by the component vendor. Components doing the same function
- from different vendors may or may not have the same version. For
- structures, the version shall be set by the entity that allocates the
- structure. For structures specified in the OMX 1.1 specification, the
- value of the version shall be set to 1.1.0.0 in all cases. Access to the
- OMX_VERSIONTYPE can be by a single 32 bit access (e.g. by nVersion) or
- by accessing one of the structure elements to, for example, check only
- the Major revision.
- */
-typedef union OMX_VERSIONTYPE
-{
- struct
- {
- OMX_U8 nVersionMajor; /**< Major version accessor element */
- OMX_U8 nVersionMinor; /**< Minor version accessor element */
- OMX_U8 nRevision; /**< Revision version accessor element */
- OMX_U8 nStep; /**< Step version accessor element */
- } s;
- OMX_U32 nVersion; /**< 32 bit value to make accessing the
- version easily done in a single word
- size copy/compare operation */
-} OMX_VERSIONTYPE;
-
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
-#endif
-/* File EOF */
diff --git a/include/media/stagefright/openmax/OMX_Video.h b/include/media/stagefright/openmax/OMX_Video.h
deleted file mode 100644
index 4f8485d..0000000
--- a/include/media/stagefright/openmax/OMX_Video.h
+++ /dev/null
@@ -1,1078 +0,0 @@
-/* ------------------------------------------------------------------
- * Copyright (C) 1998-2009 PacketVideo
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either
- * express or implied.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- * -------------------------------------------------------------------
- */
-/**
- * Copyright (c) 2008 The Khronos Group Inc.
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files (the
- * "Software"), to deal in the Software without restriction, including
- * without limitation the rights to use, copy, modify, merge, publish,
- * distribute, sublicense, and/or sell copies of the Software, and to
- * permit persons to whom the Software is furnished to do so, subject
- * to the following conditions:
- * The above copyright notice and this permission notice shall be included
- * in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS
- * OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY
- * CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
- * TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
- * SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- *
- */
-
-/**
- * @file OMX_Video.h - OpenMax IL version 1.1.2
- * The structures is needed by Video components to exchange parameters
- * and configuration data with OMX components.
- */
-#ifndef OMX_Video_h
-#define OMX_Video_h
-
-/** @defgroup video OpenMAX IL Video Domain
- * @ingroup iv
- * Structures for OpenMAX IL Video domain
- * @{
- */
-
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-
-/**
- * Each OMX header must include all required header files to allow the
- * header to compile without errors. The includes below are required
- * for this header file to compile successfully
- */
-
-#include <OMX_IVCommon.h>
-
-
-/**
- * Enumeration used to define the possible video compression codings.
- * NOTE: This essentially refers to file extensions. If the coding is
- * being used to specify the ENCODE type, then additional work
- * must be done to configure the exact flavor of the compression
- * to be used. For decode cases where the user application can
- * not differentiate between MPEG-4 and H.264 bit streams, it is
- * up to the codec to handle this.
- */
-typedef enum OMX_VIDEO_CODINGTYPE {
- OMX_VIDEO_CodingUnused, /**< Value when coding is N/A */
- OMX_VIDEO_CodingAutoDetect, /**< Autodetection of coding type */
- OMX_VIDEO_CodingMPEG2, /**< AKA: H.262 */
- OMX_VIDEO_CodingH263, /**< H.263 */
- OMX_VIDEO_CodingMPEG4, /**< MPEG-4 */
- OMX_VIDEO_CodingWMV, /**< all versions of Windows Media Video */
- OMX_VIDEO_CodingRV, /**< all versions of Real Video */
- OMX_VIDEO_CodingAVC, /**< H.264/AVC */
- OMX_VIDEO_CodingMJPEG, /**< Motion JPEG */
- OMX_VIDEO_CodingVPX, /**< Google VPX, formerly known as On2 VP8 */
- OMX_VIDEO_CodingKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_VIDEO_CodingVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_VIDEO_CodingMax = 0x7FFFFFFF
-} OMX_VIDEO_CODINGTYPE;
-
-
-/**
- * Data structure used to define a video path. The number of Video paths for
- * input and output will vary by type of the Video component.
- *
- * Input (aka Source) : zero Inputs, one Output,
- * Splitter : one Input, 2 or more Outputs,
- * Processing Element : one Input, one output,
- * Mixer : 2 or more inputs, one output,
- * Output (aka Sink) : one Input, zero outputs.
- *
- * The PortDefinition structure is used to define all of the parameters
- * necessary for the compliant component to setup an input or an output video
- * path. If additional vendor specific data is required, it should be
- * transmitted to the component using the CustomCommand function. Compliant
- * components will prepopulate this structure with optimal values during the
- * GetDefaultInitParams command.
- *
- * STRUCT MEMBERS:
- * cMIMEType : MIME type of data for the port
- * pNativeRender : Platform specific reference for a display if a
- * sync, otherwise this field is 0
- * nFrameWidth : Width of frame to be used on channel if
- * uncompressed format is used. Use 0 for unknown,
- * don't care or variable
- * nFrameHeight : Height of frame to be used on channel if
- * uncompressed format is used. Use 0 for unknown,
- * don't care or variable
- * nStride : Number of bytes per span of an image
- * (i.e. indicates the number of bytes to get
- * from span N to span N+1, where negative stride
- * indicates the image is bottom up
- * nSliceHeight : Height used when encoding in slices
- * nBitrate : Bit rate of frame to be used on channel if
- * compressed format is used. Use 0 for unknown,
- * don't care or variable
- * xFramerate : Frame rate to be used on channel if uncompressed
- * format is used. Use 0 for unknown, don't care or
- * variable. Units are Q16 frames per second.
- * bFlagErrorConcealment : Turns on error concealment if it is supported by
- * the OMX component
- * eCompressionFormat : Compression format used in this instance of the
- * component. When OMX_VIDEO_CodingUnused is
- * specified, eColorFormat is used
- * eColorFormat : Decompressed format used by this component
- * pNativeWindow : Platform specific reference for a window object if a
- * display sink , otherwise this field is 0x0.
- */
-typedef struct OMX_VIDEO_PORTDEFINITIONTYPE {
- OMX_STRING cMIMEType;
- OMX_NATIVE_DEVICETYPE pNativeRender;
- OMX_U32 nFrameWidth;
- OMX_U32 nFrameHeight;
- OMX_S32 nStride;
- OMX_U32 nSliceHeight;
- OMX_U32 nBitrate;
- OMX_U32 xFramerate;
- OMX_BOOL bFlagErrorConcealment;
- OMX_VIDEO_CODINGTYPE eCompressionFormat;
- OMX_COLOR_FORMATTYPE eColorFormat;
- OMX_NATIVE_WINDOWTYPE pNativeWindow;
-} OMX_VIDEO_PORTDEFINITIONTYPE;
-
-/**
- * Port format parameter. This structure is used to enumerate the various
- * data input/output format supported by the port.
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Indicates which port to set
- * nIndex : Indicates the enumeration index for the format from
- * 0x0 to N-1
- * eCompressionFormat : Compression format used in this instance of the
- * component. When OMX_VIDEO_CodingUnused is specified,
- * eColorFormat is used
- * eColorFormat : Decompressed format used by this component
- * xFrameRate : Indicates the video frame rate in Q16 format
- */
-typedef struct OMX_VIDEO_PARAM_PORTFORMATTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_U32 nIndex;
- OMX_VIDEO_CODINGTYPE eCompressionFormat;
- OMX_COLOR_FORMATTYPE eColorFormat;
- OMX_U32 xFramerate;
-} OMX_VIDEO_PARAM_PORTFORMATTYPE;
-
-
-/**
- * This is a structure for configuring video compression quantization
- * parameter values. Codecs may support different QP values for different
- * frame types.
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version info
- * nPortIndex : Port that this structure applies to
- * nQpI : QP value to use for index frames
- * nQpP : QP value to use for P frames
- * nQpB : QP values to use for bidirectional frames
- */
-typedef struct OMX_VIDEO_PARAM_QUANTIZATIONTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_U32 nQpI;
- OMX_U32 nQpP;
- OMX_U32 nQpB;
-} OMX_VIDEO_PARAM_QUANTIZATIONTYPE;
-
-
-/**
- * Structure for configuration of video fast update parameters.
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version info
- * nPortIndex : Port that this structure applies to
- * bEnableVFU : Enable/Disable video fast update
- * nFirstGOB : Specifies the number of the first macroblock row
- * nFirstMB : specifies the first MB relative to the specified first GOB
- * nNumMBs : Specifies the number of MBs to be refreshed from nFirstGOB
- * and nFirstMB
- */
-typedef struct OMX_VIDEO_PARAM_VIDEOFASTUPDATETYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_BOOL bEnableVFU;
- OMX_U32 nFirstGOB;
- OMX_U32 nFirstMB;
- OMX_U32 nNumMBs;
-} OMX_VIDEO_PARAM_VIDEOFASTUPDATETYPE;
-
-
-/**
- * Enumeration of possible bitrate control types
- */
-typedef enum OMX_VIDEO_CONTROLRATETYPE {
- OMX_Video_ControlRateDisable,
- OMX_Video_ControlRateVariable,
- OMX_Video_ControlRateConstant,
- OMX_Video_ControlRateVariableSkipFrames,
- OMX_Video_ControlRateConstantSkipFrames,
- OMX_Video_ControlRateKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_Video_ControlRateVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_Video_ControlRateMax = 0x7FFFFFFF
-} OMX_VIDEO_CONTROLRATETYPE;
-
-
-/**
- * Structure for configuring bitrate mode of a codec.
- *
- * STRUCT MEMBERS:
- * nSize : Size of the struct in bytes
- * nVersion : OMX spec version info
- * nPortIndex : Port that this struct applies to
- * eControlRate : Control rate type enum
- * nTargetBitrate : Target bitrate to encode with
- */
-typedef struct OMX_VIDEO_PARAM_BITRATETYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_VIDEO_CONTROLRATETYPE eControlRate;
- OMX_U32 nTargetBitrate;
-} OMX_VIDEO_PARAM_BITRATETYPE;
-
-
-/**
- * Enumeration of possible motion vector (MV) types
- */
-typedef enum OMX_VIDEO_MOTIONVECTORTYPE {
- OMX_Video_MotionVectorPixel,
- OMX_Video_MotionVectorHalfPel,
- OMX_Video_MotionVectorQuarterPel,
- OMX_Video_MotionVectorEighthPel,
- OMX_Video_MotionVectorKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_Video_MotionVectorVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_Video_MotionVectorMax = 0x7FFFFFFF
-} OMX_VIDEO_MOTIONVECTORTYPE;
-
-
-/**
- * Structure for configuring the number of motion vectors used as well
- * as their accuracy.
- *
- * STRUCT MEMBERS:
- * nSize : Size of the struct in bytes
- * nVersion : OMX spec version info
- * nPortIndex : port that this structure applies to
- * eAccuracy : Enumerated MV accuracy
- * bUnrestrictedMVs : Allow unrestricted MVs
- * bFourMV : Allow use of 4 MVs
- * sXSearchRange : Search range in horizontal direction for MVs
- * sYSearchRange : Search range in vertical direction for MVs
- */
-typedef struct OMX_VIDEO_PARAM_MOTIONVECTORTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_VIDEO_MOTIONVECTORTYPE eAccuracy;
- OMX_BOOL bUnrestrictedMVs;
- OMX_BOOL bFourMV;
- OMX_S32 sXSearchRange;
- OMX_S32 sYSearchRange;
-} OMX_VIDEO_PARAM_MOTIONVECTORTYPE;
-
-
-/**
- * Enumeration of possible methods to use for Intra Refresh
- */
-typedef enum OMX_VIDEO_INTRAREFRESHTYPE {
- OMX_VIDEO_IntraRefreshCyclic,
- OMX_VIDEO_IntraRefreshAdaptive,
- OMX_VIDEO_IntraRefreshBoth,
- OMX_VIDEO_IntraRefreshKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_VIDEO_IntraRefreshVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_VIDEO_IntraRefreshMax = 0x7FFFFFFF
-} OMX_VIDEO_INTRAREFRESHTYPE;
-
-
-/**
- * Structure for configuring intra refresh mode
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * eRefreshMode : Cyclic, Adaptive, or Both
- * nAirMBs : Number of intra macroblocks to refresh in a frame when
- * AIR is enabled
- * nAirRef : Number of times a motion marked macroblock has to be
- * intra coded
- * nCirMBs : Number of consecutive macroblocks to be coded as "intra"
- * when CIR is enabled
- */
-typedef struct OMX_VIDEO_PARAM_INTRAREFRESHTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_VIDEO_INTRAREFRESHTYPE eRefreshMode;
- OMX_U32 nAirMBs;
- OMX_U32 nAirRef;
- OMX_U32 nCirMBs;
-} OMX_VIDEO_PARAM_INTRAREFRESHTYPE;
-
-
-/**
- * Structure for enabling various error correction methods for video
- * compression.
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * bEnableHEC : Enable/disable header extension codes (HEC)
- * bEnableResync : Enable/disable resynchronization markers
- * nResynchMarkerSpacing : Resynch markers interval (in bits) to be
- * applied in the stream
- * bEnableDataPartitioning : Enable/disable data partitioning
- * bEnableRVLC : Enable/disable reversible variable length
- * coding
- */
-typedef struct OMX_VIDEO_PARAM_ERRORCORRECTIONTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_BOOL bEnableHEC;
- OMX_BOOL bEnableResync;
- OMX_U32 nResynchMarkerSpacing;
- OMX_BOOL bEnableDataPartitioning;
- OMX_BOOL bEnableRVLC;
-} OMX_VIDEO_PARAM_ERRORCORRECTIONTYPE;
-
-
-/**
- * Configuration of variable block-size motion compensation (VBSMC)
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * b16x16 : Enable inter block search 16x16
- * b16x8 : Enable inter block search 16x8
- * b8x16 : Enable inter block search 8x16
- * b8x8 : Enable inter block search 8x8
- * b8x4 : Enable inter block search 8x4
- * b4x8 : Enable inter block search 4x8
- * b4x4 : Enable inter block search 4x4
- */
-typedef struct OMX_VIDEO_PARAM_VBSMCTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_BOOL b16x16;
- OMX_BOOL b16x8;
- OMX_BOOL b8x16;
- OMX_BOOL b8x8;
- OMX_BOOL b8x4;
- OMX_BOOL b4x8;
- OMX_BOOL b4x4;
-} OMX_VIDEO_PARAM_VBSMCTYPE;
-
-
-/**
- * H.263 profile types, each profile indicates support for various
- * performance bounds and different annexes.
- *
- * ENUMS:
- * Baseline : Baseline Profile: H.263 (V1), no optional modes
- * H320 Coding : H.320 Coding Efficiency Backward Compatibility
- * Profile: H.263+ (V2), includes annexes I, J, L.4
- * and T
- * BackwardCompatible : Backward Compatibility Profile: H.263 (V1),
- * includes annex F
- * ISWV2 : Interactive Streaming Wireless Profile: H.263+
- * (V2), includes annexes I, J, K and T
- * ISWV3 : Interactive Streaming Wireless Profile: H.263++
- * (V3), includes profile 3 and annexes V and W.6.3.8
- * HighCompression : Conversational High Compression Profile: H.263++
- * (V3), includes profiles 1 & 2 and annexes D and U
- * Internet : Conversational Internet Profile: H.263++ (V3),
- * includes profile 5 and annex K
- * Interlace : Conversational Interlace Profile: H.263++ (V3),
- * includes profile 5 and annex W.6.3.11
- * HighLatency : High Latency Profile: H.263++ (V3), includes
- * profile 6 and annexes O.1 and P.5
- */
-typedef enum OMX_VIDEO_H263PROFILETYPE {
- OMX_VIDEO_H263ProfileBaseline = 0x01,
- OMX_VIDEO_H263ProfileH320Coding = 0x02,
- OMX_VIDEO_H263ProfileBackwardCompatible = 0x04,
- OMX_VIDEO_H263ProfileISWV2 = 0x08,
- OMX_VIDEO_H263ProfileISWV3 = 0x10,
- OMX_VIDEO_H263ProfileHighCompression = 0x20,
- OMX_VIDEO_H263ProfileInternet = 0x40,
- OMX_VIDEO_H263ProfileInterlace = 0x80,
- OMX_VIDEO_H263ProfileHighLatency = 0x100,
- OMX_VIDEO_H263ProfileKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_VIDEO_H263ProfileVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_VIDEO_H263ProfileMax = 0x7FFFFFFF
-} OMX_VIDEO_H263PROFILETYPE;
-
-
-/**
- * H.263 level types, each level indicates support for various frame sizes,
- * bit rates, decoder frame rates.
- */
-typedef enum OMX_VIDEO_H263LEVELTYPE {
- OMX_VIDEO_H263Level10 = 0x01,
- OMX_VIDEO_H263Level20 = 0x02,
- OMX_VIDEO_H263Level30 = 0x04,
- OMX_VIDEO_H263Level40 = 0x08,
- OMX_VIDEO_H263Level45 = 0x10,
- OMX_VIDEO_H263Level50 = 0x20,
- OMX_VIDEO_H263Level60 = 0x40,
- OMX_VIDEO_H263Level70 = 0x80,
- OMX_VIDEO_H263LevelKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_VIDEO_H263LevelVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_VIDEO_H263LevelMax = 0x7FFFFFFF
-} OMX_VIDEO_H263LEVELTYPE;
-
-
-/**
- * Specifies the picture type. These values should be OR'd to signal all
- * pictures types which are allowed.
- *
- * ENUMS:
- * Generic Picture Types: I, P and B
- * H.263 Specific Picture Types: SI and SP
- * H.264 Specific Picture Types: EI and EP
- * MPEG-4 Specific Picture Types: S
- */
-typedef enum OMX_VIDEO_PICTURETYPE {
- OMX_VIDEO_PictureTypeI = 0x01,
- OMX_VIDEO_PictureTypeP = 0x02,
- OMX_VIDEO_PictureTypeB = 0x04,
- OMX_VIDEO_PictureTypeSI = 0x08,
- OMX_VIDEO_PictureTypeSP = 0x10,
- OMX_VIDEO_PictureTypeEI = 0x11,
- OMX_VIDEO_PictureTypeEP = 0x12,
- OMX_VIDEO_PictureTypeS = 0x14,
- OMX_VIDEO_PictureTypeKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_VIDEO_PictureTypeVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_VIDEO_PictureTypeMax = 0x7FFFFFFF
-} OMX_VIDEO_PICTURETYPE;
-
-
-/**
- * H.263 Params
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * nPFrames : Number of P frames between each I frame
- * nBFrames : Number of B frames between each I frame
- * eProfile : H.263 profile(s) to use
- * eLevel : H.263 level(s) to use
- * bPLUSPTYPEAllowed : Indicating that it is allowed to use PLUSPTYPE
- * (specified in the 1998 version of H.263) to
- * indicate custom picture sizes or clock
- * frequencies
- * nAllowedPictureTypes : Specifies the picture types allowed in the
- * bitstream
- * bForceRoundingTypeToZero : value of the RTYPE bit (bit 6 of MPPTYPE) is
- * not constrained. It is recommended to change
- * the value of the RTYPE bit for each reference
- * picture in error-free communication
- * nPictureHeaderRepetition : Specifies the frequency of picture header
- * repetition
- * nGOBHeaderInterval : Specifies the interval of non-empty GOB
- * headers in units of GOBs
- */
-typedef struct OMX_VIDEO_PARAM_H263TYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_U32 nPFrames;
- OMX_U32 nBFrames;
- OMX_VIDEO_H263PROFILETYPE eProfile;
- OMX_VIDEO_H263LEVELTYPE eLevel;
- OMX_BOOL bPLUSPTYPEAllowed;
- OMX_U32 nAllowedPictureTypes;
- OMX_BOOL bForceRoundingTypeToZero;
- OMX_U32 nPictureHeaderRepetition;
- OMX_U32 nGOBHeaderInterval;
-} OMX_VIDEO_PARAM_H263TYPE;
-
-
-/**
- * MPEG-2 profile types, each profile indicates support for various
- * performance bounds and different annexes.
- */
-typedef enum OMX_VIDEO_MPEG2PROFILETYPE {
- OMX_VIDEO_MPEG2ProfileSimple = 0, /**< Simple Profile */
- OMX_VIDEO_MPEG2ProfileMain, /**< Main Profile */
- OMX_VIDEO_MPEG2Profile422, /**< 4:2:2 Profile */
- OMX_VIDEO_MPEG2ProfileSNR, /**< SNR Profile */
- OMX_VIDEO_MPEG2ProfileSpatial, /**< Spatial Profile */
- OMX_VIDEO_MPEG2ProfileHigh, /**< High Profile */
- OMX_VIDEO_MPEG2ProfileKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_VIDEO_MPEG2ProfileVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_VIDEO_MPEG2ProfileMax = 0x7FFFFFFF
-} OMX_VIDEO_MPEG2PROFILETYPE;
-
-
-/**
- * MPEG-2 level types, each level indicates support for various frame
- * sizes, bit rates, decoder frame rates. No need
- */
-typedef enum OMX_VIDEO_MPEG2LEVELTYPE {
- OMX_VIDEO_MPEG2LevelLL = 0, /**< Low Level */
- OMX_VIDEO_MPEG2LevelML, /**< Main Level */
- OMX_VIDEO_MPEG2LevelH14, /**< High 1440 */
- OMX_VIDEO_MPEG2LevelHL, /**< High Level */
- OMX_VIDEO_MPEG2LevelKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_VIDEO_MPEG2LevelVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_VIDEO_MPEG2LevelMax = 0x7FFFFFFF
-} OMX_VIDEO_MPEG2LEVELTYPE;
-
-
-/**
- * MPEG-2 params
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * nPFrames : Number of P frames between each I frame
- * nBFrames : Number of B frames between each I frame
- * eProfile : MPEG-2 profile(s) to use
- * eLevel : MPEG-2 levels(s) to use
- */
-typedef struct OMX_VIDEO_PARAM_MPEG2TYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_U32 nPFrames;
- OMX_U32 nBFrames;
- OMX_VIDEO_MPEG2PROFILETYPE eProfile;
- OMX_VIDEO_MPEG2LEVELTYPE eLevel;
-} OMX_VIDEO_PARAM_MPEG2TYPE;
-
-
-/**
- * MPEG-4 profile types, each profile indicates support for various
- * performance bounds and different annexes.
- *
- * ENUMS:
- * - Simple Profile, Levels 1-3
- * - Simple Scalable Profile, Levels 1-2
- * - Core Profile, Levels 1-2
- * - Main Profile, Levels 2-4
- * - N-bit Profile, Level 2
- * - Scalable Texture Profile, Level 1
- * - Simple Face Animation Profile, Levels 1-2
- * - Simple Face and Body Animation (FBA) Profile, Levels 1-2
- * - Basic Animated Texture Profile, Levels 1-2
- * - Hybrid Profile, Levels 1-2
- * - Advanced Real Time Simple Profiles, Levels 1-4
- * - Core Scalable Profile, Levels 1-3
- * - Advanced Coding Efficiency Profile, Levels 1-4
- * - Advanced Core Profile, Levels 1-2
- * - Advanced Scalable Texture, Levels 2-3
- */
-typedef enum OMX_VIDEO_MPEG4PROFILETYPE {
- OMX_VIDEO_MPEG4ProfileSimple = 0x01,
- OMX_VIDEO_MPEG4ProfileSimpleScalable = 0x02,
- OMX_VIDEO_MPEG4ProfileCore = 0x04,
- OMX_VIDEO_MPEG4ProfileMain = 0x08,
- OMX_VIDEO_MPEG4ProfileNbit = 0x10,
- OMX_VIDEO_MPEG4ProfileScalableTexture = 0x20,
- OMX_VIDEO_MPEG4ProfileSimpleFace = 0x40,
- OMX_VIDEO_MPEG4ProfileSimpleFBA = 0x80,
- OMX_VIDEO_MPEG4ProfileBasicAnimated = 0x100,
- OMX_VIDEO_MPEG4ProfileHybrid = 0x200,
- OMX_VIDEO_MPEG4ProfileAdvancedRealTime = 0x400,
- OMX_VIDEO_MPEG4ProfileCoreScalable = 0x800,
- OMX_VIDEO_MPEG4ProfileAdvancedCoding = 0x1000,
- OMX_VIDEO_MPEG4ProfileAdvancedCore = 0x2000,
- OMX_VIDEO_MPEG4ProfileAdvancedScalable = 0x4000,
- OMX_VIDEO_MPEG4ProfileAdvancedSimple = 0x8000,
- OMX_VIDEO_MPEG4ProfileKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_VIDEO_MPEG4ProfileVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_VIDEO_MPEG4ProfileMax = 0x7FFFFFFF
-} OMX_VIDEO_MPEG4PROFILETYPE;
-
-
-/**
- * MPEG-4 level types, each level indicates support for various frame
- * sizes, bit rates, decoder frame rates. No need
- */
-typedef enum OMX_VIDEO_MPEG4LEVELTYPE {
- OMX_VIDEO_MPEG4Level0 = 0x01, /**< Level 0 */
- OMX_VIDEO_MPEG4Level0b = 0x02, /**< Level 0b */
- OMX_VIDEO_MPEG4Level1 = 0x04, /**< Level 1 */
- OMX_VIDEO_MPEG4Level2 = 0x08, /**< Level 2 */
- OMX_VIDEO_MPEG4Level3 = 0x10, /**< Level 3 */
- OMX_VIDEO_MPEG4Level4 = 0x20, /**< Level 4 */
- OMX_VIDEO_MPEG4Level4a = 0x40, /**< Level 4a */
- OMX_VIDEO_MPEG4Level5 = 0x80, /**< Level 5 */
- OMX_VIDEO_MPEG4LevelKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_VIDEO_MPEG4LevelVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_VIDEO_MPEG4LevelMax = 0x7FFFFFFF
-} OMX_VIDEO_MPEG4LEVELTYPE;
-
-
-/**
- * MPEG-4 configuration. This structure handles configuration options
- * which are specific to MPEG4 algorithms
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * nSliceHeaderSpacing : Number of macroblocks between slice header (H263+
- * Annex K). Put zero if not used
- * bSVH : Enable Short Video Header mode
- * bGov : Flag to enable GOV
- * nPFrames : Number of P frames between each I frame (also called
- * GOV period)
- * nBFrames : Number of B frames between each I frame
- * nIDCVLCThreshold : Value of intra DC VLC threshold
- * bACPred : Flag to use ac prediction
- * nMaxPacketSize : Maximum size of packet in bytes.
- * nTimeIncRes : Used to pass VOP time increment resolution for MPEG4.
- * Interpreted as described in MPEG4 standard.
- * eProfile : MPEG-4 profile(s) to use.
- * eLevel : MPEG-4 level(s) to use.
- * nAllowedPictureTypes : Specifies the picture types allowed in the bitstream
- * nHeaderExtension : Specifies the number of consecutive video packet
- * headers within a VOP
- * bReversibleVLC : Specifies whether reversible variable length coding
- * is in use
- */
-typedef struct OMX_VIDEO_PARAM_MPEG4TYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_U32 nSliceHeaderSpacing;
- OMX_BOOL bSVH;
- OMX_BOOL bGov;
- OMX_U32 nPFrames;
- OMX_U32 nBFrames;
- OMX_U32 nIDCVLCThreshold;
- OMX_BOOL bACPred;
- OMX_U32 nMaxPacketSize;
- OMX_U32 nTimeIncRes;
- OMX_VIDEO_MPEG4PROFILETYPE eProfile;
- OMX_VIDEO_MPEG4LEVELTYPE eLevel;
- OMX_U32 nAllowedPictureTypes;
- OMX_U32 nHeaderExtension;
- OMX_BOOL bReversibleVLC;
-} OMX_VIDEO_PARAM_MPEG4TYPE;
-
-
-/**
- * WMV Versions
- */
-typedef enum OMX_VIDEO_WMVFORMATTYPE {
- OMX_VIDEO_WMVFormatUnused = 0x01, /**< Format unused or unknown */
- OMX_VIDEO_WMVFormat7 = 0x02, /**< Windows Media Video format 7 */
- OMX_VIDEO_WMVFormat8 = 0x04, /**< Windows Media Video format 8 */
- OMX_VIDEO_WMVFormat9 = 0x08, /**< Windows Media Video format 9 */
- OMX_VIDEO_WMFFormatKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_VIDEO_WMFFormatVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_VIDEO_WMVFormatMax = 0x7FFFFFFF
-} OMX_VIDEO_WMVFORMATTYPE;
-
-
-/**
- * WMV Params
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * eFormat : Version of WMV stream / data
- */
-typedef struct OMX_VIDEO_PARAM_WMVTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_VIDEO_WMVFORMATTYPE eFormat;
-} OMX_VIDEO_PARAM_WMVTYPE;
-
-
-/**
- * Real Video Version
- */
-typedef enum OMX_VIDEO_RVFORMATTYPE {
- OMX_VIDEO_RVFormatUnused = 0, /**< Format unused or unknown */
- OMX_VIDEO_RVFormat8, /**< Real Video format 8 */
- OMX_VIDEO_RVFormat9, /**< Real Video format 9 */
- OMX_VIDEO_RVFormatG2, /**< Real Video Format G2 */
- OMX_VIDEO_RVFormatKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_VIDEO_RVFormatVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_VIDEO_RVFormatMax = 0x7FFFFFFF
-} OMX_VIDEO_RVFORMATTYPE;
-
-
-/**
- * Real Video Params
- *
- * STUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * eFormat : Version of RV stream / data
- * nBitsPerPixel : Bits per pixel coded in the frame
- * nPaddedWidth : Padded width in pixel of a video frame
- * nPaddedHeight : Padded Height in pixels of a video frame
- * nFrameRate : Rate of video in frames per second
- * nBitstreamFlags : Flags which internal information about the bitstream
- * nBitstreamVersion : Bitstream version
- * nMaxEncodeFrameSize: Max encoded frame size
- * bEnablePostFilter : Turn on/off post filter
- * bEnableTemporalInterpolation : Turn on/off temporal interpolation
- * bEnableLatencyMode : When enabled, the decoder does not display a decoded
- * frame until it has detected that no enhancement layer
- * frames or dependent B frames will be coming. This
- * detection usually occurs when a subsequent non-B
- * frame is encountered
- */
-typedef struct OMX_VIDEO_PARAM_RVTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_VIDEO_RVFORMATTYPE eFormat;
- OMX_U16 nBitsPerPixel;
- OMX_U16 nPaddedWidth;
- OMX_U16 nPaddedHeight;
- OMX_U32 nFrameRate;
- OMX_U32 nBitstreamFlags;
- OMX_U32 nBitstreamVersion;
- OMX_U32 nMaxEncodeFrameSize;
- OMX_BOOL bEnablePostFilter;
- OMX_BOOL bEnableTemporalInterpolation;
- OMX_BOOL bEnableLatencyMode;
-} OMX_VIDEO_PARAM_RVTYPE;
-
-
-/**
- * AVC profile types, each profile indicates support for various
- * performance bounds and different annexes.
- */
-typedef enum OMX_VIDEO_AVCPROFILETYPE {
- OMX_VIDEO_AVCProfileBaseline = 0x01, /**< Baseline profile */
- OMX_VIDEO_AVCProfileMain = 0x02, /**< Main profile */
- OMX_VIDEO_AVCProfileExtended = 0x04, /**< Extended profile */
- OMX_VIDEO_AVCProfileHigh = 0x08, /**< High profile */
- OMX_VIDEO_AVCProfileHigh10 = 0x10, /**< High 10 profile */
- OMX_VIDEO_AVCProfileHigh422 = 0x20, /**< High 4:2:2 profile */
- OMX_VIDEO_AVCProfileHigh444 = 0x40, /**< High 4:4:4 profile */
- OMX_VIDEO_AVCProfileKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_VIDEO_AVCProfileVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_VIDEO_AVCProfileMax = 0x7FFFFFFF
-} OMX_VIDEO_AVCPROFILETYPE;
-
-
-/**
- * AVC level types, each level indicates support for various frame sizes,
- * bit rates, decoder frame rates. No need
- */
-typedef enum OMX_VIDEO_AVCLEVELTYPE {
- OMX_VIDEO_AVCLevel1 = 0x01, /**< Level 1 */
- OMX_VIDEO_AVCLevel1b = 0x02, /**< Level 1b */
- OMX_VIDEO_AVCLevel11 = 0x04, /**< Level 1.1 */
- OMX_VIDEO_AVCLevel12 = 0x08, /**< Level 1.2 */
- OMX_VIDEO_AVCLevel13 = 0x10, /**< Level 1.3 */
- OMX_VIDEO_AVCLevel2 = 0x20, /**< Level 2 */
- OMX_VIDEO_AVCLevel21 = 0x40, /**< Level 2.1 */
- OMX_VIDEO_AVCLevel22 = 0x80, /**< Level 2.2 */
- OMX_VIDEO_AVCLevel3 = 0x100, /**< Level 3 */
- OMX_VIDEO_AVCLevel31 = 0x200, /**< Level 3.1 */
- OMX_VIDEO_AVCLevel32 = 0x400, /**< Level 3.2 */
- OMX_VIDEO_AVCLevel4 = 0x800, /**< Level 4 */
- OMX_VIDEO_AVCLevel41 = 0x1000, /**< Level 4.1 */
- OMX_VIDEO_AVCLevel42 = 0x2000, /**< Level 4.2 */
- OMX_VIDEO_AVCLevel5 = 0x4000, /**< Level 5 */
- OMX_VIDEO_AVCLevel51 = 0x8000, /**< Level 5.1 */
- OMX_VIDEO_AVCLevelKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_VIDEO_AVCLevelVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_VIDEO_AVCLevelMax = 0x7FFFFFFF
-} OMX_VIDEO_AVCLEVELTYPE;
-
-
-/**
- * AVC loop filter modes
- *
- * OMX_VIDEO_AVCLoopFilterEnable : Enable
- * OMX_VIDEO_AVCLoopFilterDisable : Disable
- * OMX_VIDEO_AVCLoopFilterDisableSliceBoundary : Disabled on slice boundaries
- */
-typedef enum OMX_VIDEO_AVCLOOPFILTERTYPE {
- OMX_VIDEO_AVCLoopFilterEnable = 0,
- OMX_VIDEO_AVCLoopFilterDisable,
- OMX_VIDEO_AVCLoopFilterDisableSliceBoundary,
- OMX_VIDEO_AVCLoopFilterKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_VIDEO_AVCLoopFilterVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_VIDEO_AVCLoopFilterMax = 0x7FFFFFFF
-} OMX_VIDEO_AVCLOOPFILTERTYPE;
-
-
-/**
- * AVC params
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * nSliceHeaderSpacing : Number of macroblocks between slice header, put
- * zero if not used
- * nPFrames : Number of P frames between each I frame
- * nBFrames : Number of B frames between each I frame
- * bUseHadamard : Enable/disable Hadamard transform
- * nRefFrames : Max number of reference frames to use for inter
- * motion search (1-16)
- * nRefIdxTrailing : Pic param set ref frame index (index into ref
- * frame buffer of trailing frames list), B frame
- * support
- * nRefIdxForward : Pic param set ref frame index (index into ref
- * frame buffer of forward frames list), B frame
- * support
- * bEnableUEP : Enable/disable unequal error protection. This
- * is only valid of data partitioning is enabled.
- * bEnableFMO : Enable/disable flexible macroblock ordering
- * bEnableASO : Enable/disable arbitrary slice ordering
- * bEnableRS : Enable/disable sending of redundant slices
- * eProfile : AVC profile(s) to use
- * eLevel : AVC level(s) to use
- * nAllowedPictureTypes : Specifies the picture types allowed in the
- * bitstream
- * bFrameMBsOnly : specifies that every coded picture of the
- * coded video sequence is a coded frame
- * containing only frame macroblocks
- * bMBAFF : Enable/disable switching between frame and
- * field macroblocks within a picture
- * bEntropyCodingCABAC : Entropy decoding method to be applied for the
- * syntax elements for which two descriptors appear
- * in the syntax tables
- * bWeightedPPrediction : Enable/disable weighted prediction shall not
- * be applied to P and SP slices
- * nWeightedBipredicitonMode : Default weighted prediction is applied to B
- * slices
- * bconstIpred : Enable/disable intra prediction
- * bDirect8x8Inference : Specifies the method used in the derivation
- * process for luma motion vectors for B_Skip,
- * B_Direct_16x16 and B_Direct_8x8 as specified
- * in subclause 8.4.1.2 of the AVC spec
- * bDirectSpatialTemporal : Flag indicating spatial or temporal direct
- * mode used in B slice coding (related to
- * bDirect8x8Inference) . Spatial direct mode is
- * more common and should be the default.
- * nCabacInitIdx : Index used to init CABAC contexts
- * eLoopFilterMode : Enable/disable loop filter
- */
-typedef struct OMX_VIDEO_PARAM_AVCTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_U32 nSliceHeaderSpacing;
- OMX_U32 nPFrames;
- OMX_U32 nBFrames;
- OMX_BOOL bUseHadamard;
- OMX_U32 nRefFrames;
- OMX_U32 nRefIdx10ActiveMinus1;
- OMX_U32 nRefIdx11ActiveMinus1;
- OMX_BOOL bEnableUEP;
- OMX_BOOL bEnableFMO;
- OMX_BOOL bEnableASO;
- OMX_BOOL bEnableRS;
- OMX_VIDEO_AVCPROFILETYPE eProfile;
- OMX_VIDEO_AVCLEVELTYPE eLevel;
- OMX_U32 nAllowedPictureTypes;
- OMX_BOOL bFrameMBsOnly;
- OMX_BOOL bMBAFF;
- OMX_BOOL bEntropyCodingCABAC;
- OMX_BOOL bWeightedPPrediction;
- OMX_U32 nWeightedBipredicitonMode;
- OMX_BOOL bconstIpred ;
- OMX_BOOL bDirect8x8Inference;
- OMX_BOOL bDirectSpatialTemporal;
- OMX_U32 nCabacInitIdc;
- OMX_VIDEO_AVCLOOPFILTERTYPE eLoopFilterMode;
-} OMX_VIDEO_PARAM_AVCTYPE;
-
-typedef struct OMX_VIDEO_PARAM_PROFILELEVELTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_U32 eProfile; /**< type is OMX_VIDEO_AVCPROFILETYPE, OMX_VIDEO_H263PROFILETYPE,
- or OMX_VIDEO_MPEG4PROFILETYPE depending on context */
- OMX_U32 eLevel; /**< type is OMX_VIDEO_AVCLEVELTYPE, OMX_VIDEO_H263LEVELTYPE,
- or OMX_VIDEO_MPEG4PROFILETYPE depending on context */
- OMX_U32 nProfileIndex; /**< Used to query for individual profile support information,
- This parameter is valid only for
- OMX_IndexParamVideoProfileLevelQuerySupported index,
- For all other indices this parameter is to be ignored. */
-} OMX_VIDEO_PARAM_PROFILELEVELTYPE;
-
-/**
- * Structure for dynamically configuring bitrate mode of a codec.
- *
- * STRUCT MEMBERS:
- * nSize : Size of the struct in bytes
- * nVersion : OMX spec version info
- * nPortIndex : Port that this struct applies to
- * nEncodeBitrate : Target average bitrate to be generated in bps
- */
-typedef struct OMX_VIDEO_CONFIG_BITRATETYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_U32 nEncodeBitrate;
-} OMX_VIDEO_CONFIG_BITRATETYPE;
-
-/**
- * Defines Encoder Frame Rate setting
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * xEncodeFramerate : Encoding framerate represented in Q16 format
- */
-typedef struct OMX_CONFIG_FRAMERATETYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_U32 xEncodeFramerate; /* Q16 format */
-} OMX_CONFIG_FRAMERATETYPE;
-
-typedef struct OMX_CONFIG_INTRAREFRESHVOPTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_BOOL IntraRefreshVOP;
-} OMX_CONFIG_INTRAREFRESHVOPTYPE;
-
-typedef struct OMX_CONFIG_MACROBLOCKERRORMAPTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_U32 nErrMapSize; /* Size of the Error Map in bytes */
- OMX_U8 ErrMap[1]; /* Error map hint */
-} OMX_CONFIG_MACROBLOCKERRORMAPTYPE;
-
-typedef struct OMX_CONFIG_MBERRORREPORTINGTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_BOOL bEnabled;
-} OMX_CONFIG_MBERRORREPORTINGTYPE;
-
-typedef struct OMX_PARAM_MACROBLOCKSTYPE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_U32 nMacroblocks;
-} OMX_PARAM_MACROBLOCKSTYPE;
-
-/**
- * AVC Slice Mode modes
- *
- * OMX_VIDEO_SLICEMODE_AVCDefault : Normal frame encoding, one slice per frame
- * OMX_VIDEO_SLICEMODE_AVCMBSlice : NAL mode, number of MBs per frame
- * OMX_VIDEO_SLICEMODE_AVCByteSlice : NAL mode, number of bytes per frame
- */
-typedef enum OMX_VIDEO_AVCSLICEMODETYPE {
- OMX_VIDEO_SLICEMODE_AVCDefault = 0,
- OMX_VIDEO_SLICEMODE_AVCMBSlice,
- OMX_VIDEO_SLICEMODE_AVCByteSlice,
- OMX_VIDEO_SLICEMODE_AVCKhronosExtensions = 0x6F000000, /**< Reserved region for introducing Khronos Standard Extensions */
- OMX_VIDEO_SLICEMODE_AVCVendorStartUnused = 0x7F000000, /**< Reserved region for introducing Vendor Extensions */
- OMX_VIDEO_SLICEMODE_AVCLevelMax = 0x7FFFFFFF
-} OMX_VIDEO_AVCSLICEMODETYPE;
-
-/**
- * AVC FMO Slice Mode Params
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * nNumSliceGroups : Specifies the number of slice groups
- * nSliceGroupMapType : Specifies the type of slice groups
- * eSliceMode : Specifies the type of slice
- */
-typedef struct OMX_VIDEO_PARAM_AVCSLICEFMO {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_U8 nNumSliceGroups;
- OMX_U8 nSliceGroupMapType;
- OMX_VIDEO_AVCSLICEMODETYPE eSliceMode;
-} OMX_VIDEO_PARAM_AVCSLICEFMO;
-
-/**
- * AVC IDR Period Configs
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * nIDRPeriod : Specifies periodicity of IDR frames
- * nPFrames : Specifies internal of coding Intra frames
- */
-typedef struct OMX_VIDEO_CONFIG_AVCINTRAPERIOD {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_U32 nIDRPeriod;
- OMX_U32 nPFrames;
-} OMX_VIDEO_CONFIG_AVCINTRAPERIOD;
-
-/**
- * AVC NAL Size Configs
- *
- * STRUCT MEMBERS:
- * nSize : Size of the structure in bytes
- * nVersion : OMX specification version information
- * nPortIndex : Port that this structure applies to
- * nNaluBytes : Specifies the NAL unit size
- */
-typedef struct OMX_VIDEO_CONFIG_NALSIZE {
- OMX_U32 nSize;
- OMX_VERSIONTYPE nVersion;
- OMX_U32 nPortIndex;
- OMX_U32 nNaluBytes;
-} OMX_VIDEO_CONFIG_NALSIZE;
-
-/** @} */
-
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
-#endif
-/* File EOF */
-
diff --git a/include/media/stagefright/timedtext/TimedTextDriver.h b/include/media/stagefright/timedtext/TimedTextDriver.h
new file mode 100644
index 0000000..b9752df
--- /dev/null
+++ b/include/media/stagefright/timedtext/TimedTextDriver.h
@@ -0,0 +1,81 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef TIMED_TEXT_DRIVER_H_
+#define TIMED_TEXT_DRIVER_H_
+
+#include <media/stagefright/foundation/ABase.h> // for DISALLOW_* macro
+#include <utils/Errors.h> // for status_t
+#include <utils/RefBase.h>
+#include <utils/threads.h>
+
+namespace android {
+
+class ALooper;
+class MediaPlayerBase;
+class MediaSource;
+class Parcel;
+class TimedTextPlayer;
+class TimedTextSource;
+
+class TimedTextDriver {
+public:
+ TimedTextDriver(const wp<MediaPlayerBase> &listener);
+
+ ~TimedTextDriver();
+
+ status_t start();
+ status_t pause();
+ status_t selectTrack(int32_t index);
+ status_t unselectTrack(int32_t index);
+
+ status_t seekToAsync(int64_t timeUs);
+
+ status_t addInBandTextSource(const sp<MediaSource>& source);
+ status_t addOutOfBandTextSource(const char *uri, const char *mimeType);
+ // Caller owns the file desriptor and caller is responsible for closing it.
+ status_t addOutOfBandTextSource(
+ int fd, off64_t offset, size_t length, const char *mimeType);
+
+ void getTrackInfo(Parcel *parcel);
+
+private:
+ Mutex mLock;
+
+ enum State {
+ UNINITIALIZED,
+ PLAYING,
+ PAUSED,
+ };
+
+ sp<ALooper> mLooper;
+ sp<TimedTextPlayer> mPlayer;
+ wp<MediaPlayerBase> mListener;
+
+ // Variables to be guarded by mLock.
+ State mState;
+ int32_t mCurrentTrackIndex;
+ Vector<sp<TimedTextSource> > mTextSourceVector;
+ // -- End of variables to be guarded by mLock
+
+ status_t selectTrack_l(int32_t index);
+
+ DISALLOW_EVIL_CONSTRUCTORS(TimedTextDriver);
+};
+
+} // namespace android
+
+#endif // TIMED_TEXT_DRIVER_H_
diff --git a/include/private/binder/Static.h b/include/private/binder/Static.h
deleted file mode 100644
index 5b0f9fc..0000000
--- a/include/private/binder/Static.h
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- * Copyright (C) 2008 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-// All static variables go here, to control initialization and
-// destruction order in the library.
-
-#include <utils/threads.h>
-
-#include <binder/IBinder.h>
-#include <binder/IMemory.h>
-#include <binder/ProcessState.h>
-#include <binder/IPermissionController.h>
-#include <binder/IServiceManager.h>
-
-namespace android {
-
-// For ProcessState.cpp
-extern Mutex gProcessMutex;
-extern sp<ProcessState> gProcess;
-
-// For ServiceManager.cpp
-extern Mutex gDefaultServiceManagerLock;
-extern sp<IServiceManager> gDefaultServiceManager;
-extern sp<IPermissionController> gPermissionController;
-
-} // namespace android
diff --git a/include/private/binder/binder_module.h b/include/private/binder/binder_module.h
deleted file mode 100644
index a8dd64f..0000000
--- a/include/private/binder/binder_module.h
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (C) 2008 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef _BINDER_MODULE_H_
-#define _BINDER_MODULE_H_
-
-#ifdef __cplusplus
-namespace android {
-#endif
-
-/* obtain structures and constants from the kernel header */
-
-#include <sys/ioctl.h>
-#include <linux/binder.h>
-
-#ifdef __cplusplus
-} // namespace android
-#endif
-
-#endif // _BINDER_MODULE_H_
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 20abd51..af2db93 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -54,50 +54,99 @@
#define CBLK_RESTORED_ON 0x0040 // track has been restored after invalidation
#define CBLK_RESTORED_OFF 0x0040 // by AudioFlinger
+// Important: do not add any virtual methods, including ~
struct audio_track_cblk_t
{
// The data members are grouped so that members accessed frequently and in the same context
// are in the same line of data cache.
- Mutex lock;
- Condition cv;
+ Mutex lock; // sizeof(int)
+ Condition cv; // sizeof(int)
+
+ // next 4 are offsets within "buffers"
volatile uint32_t user;
volatile uint32_t server;
uint32_t userBase;
uint32_t serverBase;
+
+ // if there is a shared buffer, "buffers" is the value of pointer() for the shared
+ // buffer, otherwise "buffers" points immediately after the control block
void* buffers;
uint32_t frameCount;
+
// Cache line boundary
+
uint32_t loopStart;
- uint32_t loopEnd;
- int loopCount;
- volatile union {
- uint16_t volume[2];
- uint32_t volumeLR;
- };
+ uint32_t loopEnd; // read-only for server, read/write for client
+ int loopCount; // read/write for client
+
+ // Channel volumes are fixed point U4.12, so 0x1000 means 1.0.
+ // Left channel is in [0:15], right channel is in [16:31].
+ // Always read and write the combined pair atomically.
+ // For AudioTrack only, not used by AudioRecord.
+private:
+ uint32_t mVolumeLR;
+public:
+
uint32_t sampleRate;
+
// NOTE: audio_track_cblk_t::frameSize is not equal to AudioTrack::frameSize() for
// 8 bit PCM data: in this case, mCblk->frameSize is based on a sample size of
// 16 bit because data is converted to 16 bit before being stored in buffer
- uint8_t frameSize;
+ // read-only for client, server writes once at initialization and is then read-only
+ uint8_t frameSize; // would normally be size_t, but 8 bits is plenty
+
+ // never used
uint8_t pad1;
+
+ // used by client only
uint16_t bufferTimeoutMs; // Maximum cumulated timeout before restarting audioflinger
- uint16_t waitTimeMs; // Cumulated wait time
- uint16_t sendLevel;
+ uint16_t waitTimeMs; // Cumulated wait time, used by client only
+private:
+ // client write-only, server read-only
+ uint16_t mSendLevel; // Fixed point U4.12 so 0x1000 means 1.0
+public:
volatile int32_t flags;
// Cache line boundary (32 bytes)
+ // Since the control block is always located in shared memory, this constructor
+ // is only used for placement new(). It is never used for regular new() or stack.
audio_track_cblk_t();
- uint32_t stepUser(uint32_t frameCount);
- bool stepServer(uint32_t frameCount);
+ uint32_t stepUser(uint32_t frameCount); // called by client only, where
+ // client includes regular AudioTrack and AudioFlinger::PlaybackThread::OutputTrack
+ bool stepServer(uint32_t frameCount); // called by server only
void* buffer(uint32_t offset) const;
uint32_t framesAvailable();
uint32_t framesAvailable_l();
- uint32_t framesReady();
+ uint32_t framesReady(); // called by server only
bool tryLock();
+
+ // No barriers on the following operations, so the ordering of loads/stores
+ // with respect to other parameters is UNPREDICTABLE. That's considered safe.
+
+ // for AudioTrack client only, caller must limit to 0.0 <= sendLevel <= 1.0
+ void setSendLevel(float sendLevel) {
+ mSendLevel = uint16_t(sendLevel * 0x1000);
+ }
+
+ // for AudioFlinger only; the return value must be validated by the caller
+ uint16_t getSendLevel_U4_12() const {
+ return mSendLevel;
+ }
+
+ // for AudioTrack client only, caller must limit to 0 <= volumeLR <= 0x10001000
+ void setVolumeLR(uint32_t volumeLR) {
+ mVolumeLR = volumeLR;
+ }
+
+ // for AudioFlinger only; the return value must be validated by the caller
+ uint32_t getVolumeLR() const {
+ return mVolumeLR;
+ }
+
};
diff --git a/include/private/media/VideoFrame.h b/include/private/media/VideoFrame.h
index 3aff0c6..0ecc348 100644
--- a/include/private/media/VideoFrame.h
+++ b/include/private/media/VideoFrame.h
@@ -26,7 +26,7 @@
namespace android {
// A simple buffer to hold binary data
-class MediaAlbumArt
+class MediaAlbumArt
{
public:
MediaAlbumArt(): mSize(0), mData(0) {}
@@ -57,9 +57,9 @@
fclose(in);
}
- MediaAlbumArt(const MediaAlbumArt& copy) {
- mSize = copy.mSize;
- mData = NULL; // initialize it first
+ MediaAlbumArt(const MediaAlbumArt& copy) {
+ mSize = copy.mSize;
+ mData = NULL; // initialize it first
if (mSize > 0 && copy.mData != NULL) {
mData = new uint8_t[copy.mSize];
if (mData != NULL) {
@@ -89,7 +89,7 @@
{
public:
VideoFrame(): mWidth(0), mHeight(0), mDisplayWidth(0), mDisplayHeight(0), mSize(0), mData(0) {}
-
+
VideoFrame(const VideoFrame& copy) {
mWidth = copy.mWidth;
mHeight = copy.mHeight;
diff --git a/include/private/opengles/gl_context.h b/include/private/opengles/gl_context.h
deleted file mode 100644
index 6b1fa77..0000000
--- a/include/private/opengles/gl_context.h
+++ /dev/null
@@ -1,640 +0,0 @@
-/*
- * Copyright (C) 2006 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_OPENGLES_CONTEXT_H
-#define ANDROID_OPENGLES_CONTEXT_H
-
-#include <stdint.h>
-#include <stddef.h>
-#include <sys/types.h>
-#include <pthread.h>
-#ifdef HAVE_ANDROID_OS
-#include <bionic_tls.h>
-#endif
-
-#include <private/pixelflinger/ggl_context.h>
-#include <hardware/gralloc.h>
-
-#include <GLES/gl.h>
-#include <GLES/glext.h>
-
-namespace android {
-
-
-const unsigned int OGLES_NUM_COMPRESSED_TEXTURE_FORMATS = 10
-#ifdef GL_OES_compressed_ETC1_RGB8_texture
- + 1
-#endif
- ;
-
-class EGLTextureObject;
-class EGLSurfaceManager;
-class EGLBufferObjectManager;
-
-namespace gl {
-
-struct ogles_context_t;
-struct matrixx_t;
-struct transform_t;
-struct buffer_t;
-
-ogles_context_t* getGlContext();
-
-template<typename T>
-static inline void swap(T& a, T& b) {
- T t(a); a = b; b = t;
-}
-template<typename T>
-inline T max(T a, T b) {
- return a<b ? b : a;
-}
-template<typename T>
-inline T max(T a, T b, T c) {
- return max(a, max(b, c));
-}
-template<typename T>
-inline T min(T a, T b) {
- return a<b ? a : b;
-}
-template<typename T>
-inline T min(T a, T b, T c) {
- return min(a, min(b, c));
-}
-template<typename T>
-inline T min(T a, T b, T c, T d) {
- return min(min(a,b), min(c,d));
-}
-
-// ----------------------------------------------------------------------------
-// vertices
-// ----------------------------------------------------------------------------
-
-struct vec3_t {
- union {
- struct { GLfixed x, y, z; };
- struct { GLfixed r, g, b; };
- struct { GLfixed S, T, R; };
- GLfixed v[3];
- };
-};
-
-struct vec4_t {
- union {
- struct { GLfixed x, y, z, w; };
- struct { GLfixed r, g, b, a; };
- struct { GLfixed S, T, R, Q; };
- GLfixed v[4];
- };
-};
-
-struct vertex_t {
- enum {
- // these constant matter for our clipping
- CLIP_L = 0x0001, // clipping flags
- CLIP_R = 0x0002,
- CLIP_B = 0x0004,
- CLIP_T = 0x0008,
- CLIP_N = 0x0010,
- CLIP_F = 0x0020,
-
- EYE = 0x0040,
- RESERVED = 0x0080,
-
- USER_CLIP_0 = 0x0100, // user clipping flags
- USER_CLIP_1 = 0x0200,
- USER_CLIP_2 = 0x0400,
- USER_CLIP_3 = 0x0800,
- USER_CLIP_4 = 0x1000,
- USER_CLIP_5 = 0x2000,
-
- LIT = 0x4000, // lighting has been applied
- TT = 0x8000, // texture coords transformed
-
- FRUSTUM_CLIP_ALL= 0x003F,
- USER_CLIP_ALL = 0x3F00,
- CLIP_ALL = 0x3F3F,
- };
-
- // the fields below are arranged to minimize d-cache usage
- // we group together, by cache-line, the fields most likely to be used
-
- union {
- vec4_t obj;
- vec4_t eye;
- };
- vec4_t clip;
-
- uint32_t flags;
- size_t index; // cache tag, and vertex index
- GLfixed fog;
- uint8_t locked;
- uint8_t mru;
- uint8_t reserved[2];
- vec4_t window;
-
- vec4_t color;
- vec4_t texture[GGL_TEXTURE_UNIT_COUNT];
- uint32_t reserved1[4];
-
- inline void clear() {
- flags = index = locked = mru = 0;
- }
-};
-
-struct point_size_t {
- GGLcoord size;
- GLboolean smooth;
-};
-
-struct line_width_t {
- GGLcoord width;
- GLboolean smooth;
-};
-
-struct polygon_offset_t {
- GLfixed factor;
- GLfixed units;
- GLboolean enable;
-};
-
-// ----------------------------------------------------------------------------
-// arrays
-// ----------------------------------------------------------------------------
-
-struct array_t {
- typedef void (*fetcher_t)(ogles_context_t*, GLfixed*, const GLvoid*);
- fetcher_t fetch;
- GLvoid const* physical_pointer;
- GLint size;
- GLsizei stride;
- GLvoid const* pointer;
- buffer_t const* bo;
- uint16_t type;
- GLboolean enable;
- GLboolean pad;
- GLsizei bounds;
- void init(GLint, GLenum, GLsizei, const GLvoid *, const buffer_t*, GLsizei);
- inline void resolve();
- inline const GLubyte* element(GLint i) const {
- return (const GLubyte*)physical_pointer + i * stride;
- }
-};
-
-struct array_machine_t {
- array_t vertex;
- array_t normal;
- array_t color;
- array_t texture[GGL_TEXTURE_UNIT_COUNT];
- uint8_t activeTexture;
- uint8_t tmu;
- uint16_t cull;
- uint32_t flags;
- GLenum indicesType;
- buffer_t const* array_buffer;
- buffer_t const* element_array_buffer;
-
- void (*compileElements)(ogles_context_t*, vertex_t*, GLint, GLsizei);
- void (*compileElement)(ogles_context_t*, vertex_t*, GLint);
-
- void (*mvp_transform)(transform_t const*, vec4_t*, vec4_t const*);
- void (*mv_transform)(transform_t const*, vec4_t*, vec4_t const*);
- void (*tex_transform[2])(transform_t const*, vec4_t*, vec4_t const*);
- void (*perspective)(ogles_context_t*c, vertex_t* v);
- void (*clipVertex)(ogles_context_t* c, vertex_t* nv,
- GGLfixed t, const vertex_t* s, const vertex_t* p);
- void (*clipEye)(ogles_context_t* c, vertex_t* nv,
- GGLfixed t, const vertex_t* s, const vertex_t* p);
-};
-
-struct vertex_cache_t {
- enum {
- // must be at least 4
- // 3 vertice for triangles
- // or 2 + 2 for indexed triangles w/ cache contention
- VERTEX_BUFFER_SIZE = 8,
- // must be a power of two and at least 3
- VERTEX_CACHE_SIZE = 64, // 8 KB
-
- INDEX_BITS = 16,
- INDEX_MASK = ((1LU<<INDEX_BITS)-1),
- INDEX_SEQ = 1LU<<INDEX_BITS,
- };
- vertex_t* vBuffer;
- vertex_t* vCache;
- uint32_t sequence;
- void* base;
- uint32_t total;
- uint32_t misses;
- int64_t startTime;
- void init();
- void uninit();
- void clear();
- void dump_stats(GLenum mode);
-};
-
-// ----------------------------------------------------------------------------
-// fog
-// ----------------------------------------------------------------------------
-
-struct fog_t {
- GLfixed density;
- GLfixed start;
- GLfixed end;
- GLfixed invEndMinusStart;
- GLenum mode;
- GLfixed (*fog)(ogles_context_t* c, GLfixed z);
-};
-
-// ----------------------------------------------------------------------------
-// user clip planes
-// ----------------------------------------------------------------------------
-
-const unsigned int OGLES_MAX_CLIP_PLANES = 6;
-
-struct clip_plane_t {
- vec4_t equation;
-};
-
-struct user_clip_planes_t {
- clip_plane_t plane[OGLES_MAX_CLIP_PLANES];
- uint32_t enable;
-};
-
-// ----------------------------------------------------------------------------
-// lighting
-// ----------------------------------------------------------------------------
-
-const unsigned int OGLES_MAX_LIGHTS = 8;
-
-struct light_t {
- vec4_t ambient;
- vec4_t diffuse;
- vec4_t specular;
- vec4_t implicitAmbient;
- vec4_t implicitDiffuse;
- vec4_t implicitSpecular;
- vec4_t position; // position in eye space
- vec4_t objPosition;
- vec4_t normalizedObjPosition;
- vec4_t spotDir;
- vec4_t normalizedSpotDir;
- GLfixed spotExp;
- GLfixed spotCutoff;
- GLfixed spotCutoffCosine;
- GLfixed attenuation[3];
- GLfixed rConstAttenuation;
- GLboolean enable;
-};
-
-struct material_t {
- vec4_t ambient;
- vec4_t diffuse;
- vec4_t specular;
- vec4_t emission;
- GLfixed shininess;
-};
-
-struct light_model_t {
- vec4_t ambient;
- GLboolean twoSide;
-};
-
-struct color_material_t {
- GLenum face;
- GLenum mode;
- GLboolean enable;
-};
-
-struct lighting_t {
- light_t lights[OGLES_MAX_LIGHTS];
- material_t front;
- light_model_t lightModel;
- color_material_t colorMaterial;
- vec4_t implicitSceneEmissionAndAmbient;
- vec4_t objViewer;
- uint32_t enabledLights;
- GLboolean enable;
- GLenum shadeModel;
- typedef void (*light_fct_t)(ogles_context_t*, vertex_t*);
- void (*lightVertex)(ogles_context_t* c, vertex_t* v);
- void (*lightTriangle)(ogles_context_t* c,
- vertex_t* v0, vertex_t* v1, vertex_t* v2);
-};
-
-struct culling_t {
- GLenum cullFace;
- GLenum frontFace;
- GLboolean enable;
-};
-
-// ----------------------------------------------------------------------------
-// textures
-// ----------------------------------------------------------------------------
-
-struct texture_unit_t {
- GLuint name;
- EGLTextureObject* texture;
- uint8_t dirty;
-};
-
-struct texture_state_t
-{
- texture_unit_t tmu[GGL_TEXTURE_UNIT_COUNT];
- int active; // active tmu
- EGLTextureObject* defaultTexture;
- GGLContext* ggl;
- uint8_t packAlignment;
- uint8_t unpackAlignment;
-};
-
-// ----------------------------------------------------------------------------
-// transformation and matrices
-// ----------------------------------------------------------------------------
-
-struct matrixf_t;
-
-struct matrixx_t {
- GLfixed m[16];
- void load(const matrixf_t& rhs);
-};
-
-struct matrix_stack_t;
-
-
-struct matrixf_t {
- void loadIdentity();
- void load(const matrixf_t& rhs);
-
- inline GLfloat* editElements() { return m; }
- inline GLfloat const* elements() const { return m; }
-
- void set(const GLfixed* rhs);
- void set(const GLfloat* rhs);
-
- static void multiply(matrixf_t& r,
- const matrixf_t& lhs, const matrixf_t& rhs);
-
- void dump(const char* what);
-
-private:
- friend struct matrix_stack_t;
- GLfloat m[16];
- void load(const GLfixed* rhs);
- void load(const GLfloat* rhs);
- void multiply(const matrixf_t& rhs);
- void translate(GLfloat x, GLfloat y, GLfloat z);
- void scale(GLfloat x, GLfloat y, GLfloat z);
- void rotate(GLfloat a, GLfloat x, GLfloat y, GLfloat z);
-};
-
-enum {
- OP_IDENTITY = 0x00,
- OP_TRANSLATE = 0x01,
- OP_UNIFORM_SCALE = 0x02,
- OP_SCALE = 0x05,
- OP_ROTATE = 0x08,
- OP_SKEW = 0x10,
- OP_ALL = 0x1F
-};
-
-struct transform_t {
- enum {
- FLAGS_2D_PROJECTION = 0x1
- };
- matrixx_t matrix;
- uint32_t flags;
- uint32_t ops;
-
- union {
- struct {
- void (*point2)(transform_t const* t, vec4_t*, vec4_t const*);
- void (*point3)(transform_t const* t, vec4_t*, vec4_t const*);
- void (*point4)(transform_t const* t, vec4_t*, vec4_t const*);
- };
- void (*pointv[3])(transform_t const* t, vec4_t*, vec4_t const*);
- };
-
- void loadIdentity();
- void picker();
- void dump(const char* what);
-};
-
-struct mvui_transform_t : public transform_t
-{
- void picker();
-};
-
-struct matrix_stack_t {
- enum {
- DO_PICKER = 0x1,
- DO_FLOAT_TO_FIXED = 0x2
- };
- transform_t transform;
- uint8_t maxDepth;
- uint8_t depth;
- uint8_t dirty;
- uint8_t reserved;
- matrixf_t *stack;
- uint8_t *ops;
- void init(int depth);
- void uninit();
- void loadIdentity();
- void load(const GLfixed* rhs);
- void load(const GLfloat* rhs);
- void multiply(const matrixf_t& rhs);
- void translate(GLfloat x, GLfloat y, GLfloat z);
- void scale(GLfloat x, GLfloat y, GLfloat z);
- void rotate(GLfloat a, GLfloat x, GLfloat y, GLfloat z);
- GLint push();
- GLint pop();
- void validate();
- matrixf_t& top() { return stack[depth]; }
- const matrixf_t& top() const { return stack[depth]; }
- uint32_t top_ops() const { return ops[depth]; }
- inline bool isRigidBody() const {
- return !(ops[depth] & ~(OP_TRANSLATE|OP_UNIFORM_SCALE|OP_ROTATE));
- }
-};
-
-struct vp_transform_t {
- transform_t transform;
- matrixf_t matrix;
- GLfloat zNear;
- GLfloat zFar;
- void loadIdentity();
-};
-
-struct transform_state_t {
- enum {
- MODELVIEW = 0x01,
- PROJECTION = 0x02,
- VIEWPORT = 0x04,
- TEXTURE = 0x08,
- MVUI = 0x10,
- MVIT = 0x20,
- MVP = 0x40,
- };
- matrix_stack_t *current;
- matrix_stack_t modelview;
- matrix_stack_t projection;
- matrix_stack_t texture[GGL_TEXTURE_UNIT_COUNT];
-
- // modelview * projection
- transform_t mvp __attribute__((aligned(32)));
- // viewport transformation
- vp_transform_t vpt __attribute__((aligned(32)));
- // same for 4-D vertices
- transform_t mvp4;
- // full modelview inverse transpose
- transform_t mvit4;
- // upper 3x3 of mv-inverse-transpose (for normals)
- mvui_transform_t mvui;
-
- GLenum matrixMode;
- GLenum rescaleNormals;
- uint32_t dirty;
- void invalidate();
- void update_mvp();
- void update_mvit();
- void update_mvui();
-};
-
-struct viewport_t {
- GLint x;
- GLint y;
- GLsizei w;
- GLsizei h;
- struct {
- GLint x;
- GLint y;
- } surfaceport;
- struct {
- GLint x;
- GLint y;
- GLsizei w;
- GLsizei h;
- } scissor;
-};
-
-// ----------------------------------------------------------------------------
-// Lerping
-// ----------------------------------------------------------------------------
-
-struct compute_iterators_t
-{
- void initTriangle(
- vertex_t const* v0,
- vertex_t const* v1,
- vertex_t const* v2);
-
- void initLine(
- vertex_t const* v0,
- vertex_t const* v1);
-
- inline void initLerp(vertex_t const* v0, uint32_t enables);
-
- int iteratorsScale(int32_t it[3],
- int32_t c0, int32_t c1, int32_t c2) const;
-
- void iterators1616(GGLfixed it[3],
- GGLfixed c0, GGLfixed c1, GGLfixed c2) const;
-
- void iterators0032(int32_t it[3],
- int32_t c0, int32_t c1, int32_t c2) const;
-
- void iterators0032(int64_t it[3],
- int32_t c0, int32_t c1, int32_t c2) const;
-
- GGLcoord area() const { return m_area; }
-
-private:
- // don't change order of members here -- used by iterators.S
- GGLcoord m_dx01, m_dy10, m_dx20, m_dy02;
- GGLcoord m_x0, m_y0;
- GGLcoord m_area;
- uint8_t m_scale;
- uint8_t m_area_scale;
- uint8_t m_reserved[2];
-
-};
-
-// ----------------------------------------------------------------------------
-// state
-// ----------------------------------------------------------------------------
-
-#ifdef HAVE_ANDROID_OS
- // We have a dedicated TLS slot in bionic
- inline void setGlThreadSpecific(ogles_context_t *value) {
- ((uint32_t *)__get_tls())[TLS_SLOT_OPENGL] = (uint32_t)value;
- }
- inline ogles_context_t* getGlThreadSpecific() {
- return (ogles_context_t *)(((unsigned *)__get_tls())[TLS_SLOT_OPENGL]);
- }
-#else
- extern pthread_key_t gGLKey;
- inline void setGlThreadSpecific(ogles_context_t *value) {
- pthread_setspecific(gGLKey, value);
- }
- inline ogles_context_t* getGlThreadSpecific() {
- return static_cast<ogles_context_t*>(pthread_getspecific(gGLKey));
- }
-#endif
-
-
-struct prims_t {
- typedef ogles_context_t* GL;
- void (*renderPoint)(GL, vertex_t*);
- void (*renderLine)(GL, vertex_t*, vertex_t*);
- void (*renderTriangle)(GL, vertex_t*, vertex_t*, vertex_t*);
-};
-
-struct ogles_context_t {
- context_t rasterizer;
- array_machine_t arrays __attribute__((aligned(32)));
- texture_state_t textures;
- transform_state_t transforms;
- vertex_cache_t vc;
- prims_t prims;
- culling_t cull;
- lighting_t lighting;
- user_clip_planes_t clipPlanes;
- compute_iterators_t lerp; __attribute__((aligned(32)));
- vertex_t current;
- vec4_t currentColorClamped;
- vec3_t currentNormal;
- viewport_t viewport;
- point_size_t point;
- line_width_t line;
- polygon_offset_t polygonOffset;
- fog_t fog;
- uint32_t perspective : 1;
- uint32_t transformTextures : 1;
- EGLSurfaceManager* surfaceManager;
- EGLBufferObjectManager* bufferObjectManager;
-
- GLenum error;
-
- static inline ogles_context_t* get() {
- return getGlThreadSpecific();
- }
-
-};
-
-}; // namespace gl
-}; // namespace android
-
-#endif // ANDROID_OPENGLES_CONTEXT_H
-
diff --git a/include/private/surfaceflinger/LayerState.h b/include/private/surfaceflinger/LayerState.h
deleted file mode 100644
index 3eb5c99..0000000
--- a/include/private/surfaceflinger/LayerState.h
+++ /dev/null
@@ -1,82 +0,0 @@
-/*
- * Copyright (C) 2008 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_SF_LAYER_STATE_H
-#define ANDROID_SF_LAYER_STATE_H
-
-#include <stdint.h>
-#include <sys/types.h>
-
-#include <utils/Errors.h>
-
-#include <ui/Region.h>
-
-#include <surfaceflinger/ISurface.h>
-
-namespace android {
-
-class Parcel;
-class ISurfaceComposerClient;
-
-struct layer_state_t {
-
- layer_state_t()
- : surface(0), what(0),
- x(0), y(0), z(0), w(0), h(0),
- alpha(0), tint(0), flags(0), mask(0),
- reserved(0)
- {
- matrix.dsdx = matrix.dtdy = 1.0f;
- matrix.dsdy = matrix.dtdx = 0.0f;
- }
-
- status_t write(Parcel& output) const;
- status_t read(const Parcel& input);
-
- struct matrix22_t {
- float dsdx;
- float dtdx;
- float dsdy;
- float dtdy;
- };
- SurfaceID surface;
- uint32_t what;
- float x;
- float y;
- uint32_t z;
- uint32_t w;
- uint32_t h;
- float alpha;
- uint32_t tint;
- uint8_t flags;
- uint8_t mask;
- uint8_t reserved;
- matrix22_t matrix;
- // non POD must be last. see write/read
- Region transparentRegion;
-};
-
-struct ComposerState {
- sp<ISurfaceComposerClient> client;
- layer_state_t state;
- status_t write(Parcel& output) const;
- status_t read(const Parcel& input);
-};
-
-}; // namespace android
-
-#endif // ANDROID_SF_LAYER_STATE_H
-
diff --git a/include/private/surfaceflinger/SharedBufferStack.h b/include/private/surfaceflinger/SharedBufferStack.h
deleted file mode 100644
index 0da03d1..0000000
--- a/include/private/surfaceflinger/SharedBufferStack.h
+++ /dev/null
@@ -1,59 +0,0 @@
-/*
- * Copyright (C) 2007 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_SF_SHARED_BUFFER_STACK_H
-#define ANDROID_SF_SHARED_BUFFER_STACK_H
-
-#include <stdint.h>
-#include <sys/types.h>
-
-#include <utils/Debug.h>
-
-namespace android {
-// ---------------------------------------------------------------------------
-
-#define NUM_DISPLAY_MAX 4
-
-struct display_cblk_t
-{
- uint16_t w;
- uint16_t h;
- uint8_t format;
- uint8_t orientation;
- uint8_t reserved[2];
- float fps;
- float density;
- float xdpi;
- float ydpi;
- uint32_t pad[2];
-};
-
-struct surface_flinger_cblk_t // 4KB max
-{
- uint8_t connected;
- uint8_t reserved[3];
- uint32_t pad[7];
- display_cblk_t displays[NUM_DISPLAY_MAX];
-};
-
-// ---------------------------------------------------------------------------
-
-COMPILE_TIME_ASSERT(sizeof(surface_flinger_cblk_t) <= 4096)
-
-// ---------------------------------------------------------------------------
-}; // namespace android
-
-#endif /* ANDROID_SF_SHARED_BUFFER_STACK_H */
diff --git a/include/private/ui/RegionHelper.h b/include/private/ui/RegionHelper.h
deleted file mode 100644
index 8d76533..0000000
--- a/include/private/ui/RegionHelper.h
+++ /dev/null
@@ -1,284 +0,0 @@
-/*
- * Copyright (C) 2009 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_UI_PRIVATE_REGION_HELPER_H
-#define ANDROID_UI_PRIVATE_REGION_HELPER_H
-
-#include <stdint.h>
-#include <sys/types.h>
-
-namespace android {
-// ----------------------------------------------------------------------------
-
-template<typename RECT>
-class region_operator
-{
- typedef typename RECT::value_type TYPE;
- static const TYPE max_value = 0x7FFFFFF;
-
-public:
- /*
- * Common boolean operations:
- * value is computed as 0b101 op 0b110
- * other boolean operation are possible, simply compute
- * their corresponding value with the above formulae and use
- * it when instantiating a region_operator.
- */
- static const uint32_t LHS = 0x5; // 0b101
- static const uint32_t RHS = 0x6; // 0b110
- enum {
- op_nand = LHS & ~RHS,
- op_and = LHS & RHS,
- op_or = LHS | RHS,
- op_xor = LHS ^ RHS
- };
-
- struct region {
- RECT const* rects;
- size_t count;
- TYPE dx;
- TYPE dy;
- inline region(const region& rhs)
- : rects(rhs.rects), count(rhs.count), dx(rhs.dx), dy(rhs.dy) { }
- inline region(RECT const* r, size_t c)
- : rects(r), count(c), dx(), dy() { }
- inline region(RECT const* r, size_t c, TYPE dx, TYPE dy)
- : rects(r), count(c), dx(dx), dy(dy) { }
- };
-
- class region_rasterizer {
- friend class region_operator;
- virtual void operator()(const RECT& rect) = 0;
- public:
- virtual ~region_rasterizer() { };
- };
-
- inline region_operator(int op, const region& lhs, const region& rhs)
- : op_mask(op), spanner(lhs, rhs)
- {
- }
-
- void operator()(region_rasterizer& rasterizer) {
- RECT current;
- do {
- SpannerInner spannerInner(spanner.lhs, spanner.rhs);
- int inside = spanner.next(current.top, current.bottom);
- spannerInner.prepare(inside);
- do {
- TYPE left, right;
- int inside = spannerInner.next(current.left, current.right);
- if ((op_mask >> inside) & 1) {
- if (current.left < current.right &&
- current.top < current.bottom) {
- rasterizer(current);
- }
- }
- } while(!spannerInner.isDone());
- } while(!spanner.isDone());
- }
-
-private:
- uint32_t op_mask;
-
- class SpannerBase
- {
- public:
- enum {
- lhs_before_rhs = 0,
- lhs_after_rhs = 1,
- lhs_coincide_rhs = 2
- };
-
- protected:
- TYPE lhs_head;
- TYPE lhs_tail;
- TYPE rhs_head;
- TYPE rhs_tail;
-
- inline int next(TYPE& head, TYPE& tail,
- bool& more_lhs, bool& more_rhs)
- {
- int inside;
- more_lhs = false;
- more_rhs = false;
- if (lhs_head < rhs_head) {
- inside = lhs_before_rhs;
- head = lhs_head;
- if (lhs_tail <= rhs_head) {
- tail = lhs_tail;
- more_lhs = true;
- } else {
- lhs_head = rhs_head;
- tail = rhs_head;
- }
- } else if (rhs_head < lhs_head) {
- inside = lhs_after_rhs;
- head = rhs_head;
- if (rhs_tail <= lhs_head) {
- tail = rhs_tail;
- more_rhs = true;
- } else {
- rhs_head = lhs_head;
- tail = lhs_head;
- }
- } else {
- inside = lhs_coincide_rhs;
- head = lhs_head;
- if (lhs_tail <= rhs_tail) {
- tail = rhs_head = lhs_tail;
- more_lhs = true;
- }
- if (rhs_tail <= lhs_tail) {
- tail = lhs_head = rhs_tail;
- more_rhs = true;
- }
- }
- return inside;
- }
- };
-
- class Spanner : protected SpannerBase
- {
- friend class region_operator;
- region lhs;
- region rhs;
-
- public:
- inline Spanner(const region& lhs, const region& rhs)
- : lhs(lhs), rhs(rhs)
- {
- SpannerBase::lhs_head = lhs.rects->top + lhs.dy;
- SpannerBase::lhs_tail = lhs.rects->bottom + lhs.dy;
- SpannerBase::rhs_head = rhs.rects->top + rhs.dy;
- SpannerBase::rhs_tail = rhs.rects->bottom + rhs.dy;
- }
-
- inline bool isDone() const {
- return !rhs.count && !lhs.count;
- }
-
- inline int next(TYPE& top, TYPE& bottom)
- {
- bool more_lhs = false;
- bool more_rhs = false;
- int inside = SpannerBase::next(top, bottom, more_lhs, more_rhs);
- if (more_lhs) {
- advance(lhs, SpannerBase::lhs_head, SpannerBase::lhs_tail);
- }
- if (more_rhs) {
- advance(rhs, SpannerBase::rhs_head, SpannerBase::rhs_tail);
- }
- return inside;
- }
-
- private:
- static inline
- void advance(region& reg, TYPE& aTop, TYPE& aBottom) {
- // got to next span
- size_t count = reg.count;
- RECT const * rects = reg.rects;
- RECT const * const end = rects + count;
- const int top = rects->top;
- while (rects != end && rects->top == top) {
- rects++;
- count--;
- }
- if (rects != end) {
- aTop = rects->top + reg.dy;
- aBottom = rects->bottom + reg.dy;
- } else {
- aTop = max_value;
- aBottom = max_value;
- }
- reg.rects = rects;
- reg.count = count;
- }
- };
-
- class SpannerInner : protected SpannerBase
- {
- region lhs;
- region rhs;
-
- public:
- inline SpannerInner(const region& lhs, const region& rhs)
- : lhs(lhs), rhs(rhs)
- {
- }
-
- inline void prepare(int inside) {
- if (inside == SpannerBase::lhs_before_rhs) {
- SpannerBase::lhs_head = lhs.rects->left + lhs.dx;
- SpannerBase::lhs_tail = lhs.rects->right + lhs.dx;
- SpannerBase::rhs_head = max_value;
- SpannerBase::rhs_tail = max_value;
- } else if (inside == SpannerBase::lhs_after_rhs) {
- SpannerBase::lhs_head = max_value;
- SpannerBase::lhs_tail = max_value;
- SpannerBase::rhs_head = rhs.rects->left + rhs.dx;
- SpannerBase::rhs_tail = rhs.rects->right + rhs.dx;
- } else {
- SpannerBase::lhs_head = lhs.rects->left + lhs.dx;
- SpannerBase::lhs_tail = lhs.rects->right + lhs.dx;
- SpannerBase::rhs_head = rhs.rects->left + rhs.dx;
- SpannerBase::rhs_tail = rhs.rects->right + rhs.dx;
- }
- }
-
- inline bool isDone() const {
- return SpannerBase::lhs_head == max_value &&
- SpannerBase::rhs_head == max_value;
- }
-
- inline int next(TYPE& left, TYPE& right)
- {
- bool more_lhs = false;
- bool more_rhs = false;
- int inside = SpannerBase::next(left, right, more_lhs, more_rhs);
- if (more_lhs) {
- advance(lhs, SpannerBase::lhs_head, SpannerBase::lhs_tail);
- }
- if (more_rhs) {
- advance(rhs, SpannerBase::rhs_head, SpannerBase::rhs_tail);
- }
- return inside;
- }
-
- private:
- static inline
- void advance(region& reg, TYPE& left, TYPE& right) {
- if (reg.rects && reg.count) {
- const int cur_span_top = reg.rects->top;
- reg.rects++;
- reg.count--;
- if (!reg.count || reg.rects->top != cur_span_top) {
- left = max_value;
- right = max_value;
- } else {
- left = reg.rects->left + reg.dx;
- right = reg.rects->right + reg.dx;
- }
- }
- }
- };
-
- Spanner spanner;
-};
-
-// ----------------------------------------------------------------------------
-};
-
-#endif /* ANDROID_UI_PRIVATE_REGION_HELPER_H */
diff --git a/include/private/ui/android_natives_priv.h b/include/private/ui/android_natives_priv.h
deleted file mode 100644
index 6b9f524..0000000
--- a/include/private/ui/android_natives_priv.h
+++ /dev/null
@@ -1,17 +0,0 @@
-/*
- * Copyright (C) 2009 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include <ui/android_native_buffer.h>
diff --git a/include/private/utils/Static.h b/include/private/utils/Static.h
deleted file mode 100644
index d95ae0d..0000000
--- a/include/private/utils/Static.h
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- * Copyright (C) 2008 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-// All static variables go here, to control initialization and
-// destruction order in the library.
-
-#include <utils/threads.h>
-#include <utils/KeyedVector.h>
-
-namespace android {
-// For TextStream.cpp
-extern Vector<int32_t> gTextBuffers;
-
-// For String8.cpp
-extern void initialize_string8();
-extern void terminate_string8();
-
-// For String16.cpp
-extern void initialize_string16();
-extern void terminate_string16();
-
-} // namespace android
diff --git a/media/common_time/Android.mk b/media/common_time/Android.mk
new file mode 100644
index 0000000..526f17b
--- /dev/null
+++ b/media/common_time/Android.mk
@@ -0,0 +1,21 @@
+LOCAL_PATH:= $(call my-dir)
+#
+# libcommon_time_client
+# (binder marshalers for ICommonClock as well as common clock and local clock
+# helper code)
+#
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE := libcommon_time_client
+LOCAL_MODULE_TAGS := optional
+LOCAL_SRC_FILES := cc_helper.cpp \
+ local_clock.cpp \
+ ICommonClock.cpp \
+ ICommonTimeConfig.cpp \
+ utils.cpp
+LOCAL_SHARED_LIBRARIES := libbinder \
+ libhardware \
+ libutils
+
+include $(BUILD_SHARED_LIBRARY)
diff --git a/media/common_time/ICommonClock.cpp b/media/common_time/ICommonClock.cpp
new file mode 100644
index 0000000..28b43ac
--- /dev/null
+++ b/media/common_time/ICommonClock.cpp
@@ -0,0 +1,432 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+#include <linux/socket.h>
+
+#include <common_time/ICommonClock.h>
+#include <binder/Parcel.h>
+
+#include "utils.h"
+
+namespace android {
+
+/***** ICommonClock *****/
+
+enum {
+ IS_COMMON_TIME_VALID = IBinder::FIRST_CALL_TRANSACTION,
+ COMMON_TIME_TO_LOCAL_TIME,
+ LOCAL_TIME_TO_COMMON_TIME,
+ GET_COMMON_TIME,
+ GET_COMMON_FREQ,
+ GET_LOCAL_TIME,
+ GET_LOCAL_FREQ,
+ GET_ESTIMATED_ERROR,
+ GET_TIMELINE_ID,
+ GET_STATE,
+ GET_MASTER_ADDRESS,
+ REGISTER_LISTENER,
+ UNREGISTER_LISTENER,
+};
+
+const String16 ICommonClock::kServiceName("common_time.clock");
+const uint64_t ICommonClock::kInvalidTimelineID = 0;
+const int32_t ICommonClock::kErrorEstimateUnknown = 0x7FFFFFFF;
+
+class BpCommonClock : public BpInterface<ICommonClock>
+{
+ public:
+ BpCommonClock(const sp<IBinder>& impl)
+ : BpInterface<ICommonClock>(impl) {}
+
+ virtual status_t isCommonTimeValid(bool* valid, uint32_t* timelineID) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonClock::getInterfaceDescriptor());
+ status_t status = remote()->transact(IS_COMMON_TIME_VALID,
+ data,
+ &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *valid = reply.readInt32();
+ *timelineID = reply.readInt32();
+ }
+ }
+ return status;
+ }
+
+ virtual status_t commonTimeToLocalTime(int64_t commonTime,
+ int64_t* localTime) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonClock::getInterfaceDescriptor());
+ data.writeInt64(commonTime);
+ status_t status = remote()->transact(COMMON_TIME_TO_LOCAL_TIME,
+ data, &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *localTime = reply.readInt64();
+ }
+ }
+ return status;
+ }
+
+ virtual status_t localTimeToCommonTime(int64_t localTime,
+ int64_t* commonTime) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonClock::getInterfaceDescriptor());
+ data.writeInt64(localTime);
+ status_t status = remote()->transact(LOCAL_TIME_TO_COMMON_TIME,
+ data, &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *commonTime = reply.readInt64();
+ }
+ }
+ return status;
+ }
+
+ virtual status_t getCommonTime(int64_t* commonTime) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonClock::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_COMMON_TIME, data, &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *commonTime = reply.readInt64();
+ }
+ }
+ return status;
+ }
+
+ virtual status_t getCommonFreq(uint64_t* freq) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonClock::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_COMMON_FREQ, data, &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *freq = reply.readInt64();
+ }
+ }
+ return status;
+ }
+
+ virtual status_t getLocalTime(int64_t* localTime) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonClock::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_LOCAL_TIME, data, &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *localTime = reply.readInt64();
+ }
+ }
+ return status;
+ }
+
+ virtual status_t getLocalFreq(uint64_t* freq) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonClock::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_LOCAL_FREQ, data, &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *freq = reply.readInt64();
+ }
+ }
+ return status;
+ }
+
+ virtual status_t getEstimatedError(int32_t* estimate) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonClock::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_ESTIMATED_ERROR, data, &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *estimate = reply.readInt32();
+ }
+ }
+ return status;
+ }
+
+ virtual status_t getTimelineID(uint64_t* id) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonClock::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_TIMELINE_ID, data, &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *id = static_cast<uint64_t>(reply.readInt64());
+ }
+ }
+ return status;
+ }
+
+ virtual status_t getState(State* state) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonClock::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_STATE, data, &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *state = static_cast<State>(reply.readInt32());
+ }
+ }
+ return status;
+ }
+
+ virtual status_t getMasterAddr(struct sockaddr_storage* addr) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonClock::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_MASTER_ADDRESS, data, &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK)
+ deserializeSockaddr(&reply, addr);
+ }
+ return status;
+ }
+
+ virtual status_t registerListener(
+ const sp<ICommonClockListener>& listener) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonClock::getInterfaceDescriptor());
+ data.writeStrongBinder(listener->asBinder());
+
+ status_t status = remote()->transact(REGISTER_LISTENER, data, &reply);
+
+ if (status == OK) {
+ status = reply.readInt32();
+ }
+
+ return status;
+ }
+
+ virtual status_t unregisterListener(
+ const sp<ICommonClockListener>& listener) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonClock::getInterfaceDescriptor());
+ data.writeStrongBinder(listener->asBinder());
+ status_t status = remote()->transact(UNREGISTER_LISTENER, data, &reply);
+
+ if (status == OK) {
+ status = reply.readInt32();
+ }
+
+ return status;
+ }
+};
+
+IMPLEMENT_META_INTERFACE(CommonClock, "android.os.ICommonClock");
+
+status_t BnCommonClock::onTransact(uint32_t code,
+ const Parcel& data,
+ Parcel* reply,
+ uint32_t flags) {
+ switch(code) {
+ case IS_COMMON_TIME_VALID: {
+ CHECK_INTERFACE(ICommonClock, data, reply);
+ bool valid;
+ uint32_t timelineID;
+ status_t status = isCommonTimeValid(&valid, &timelineID);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt32(valid);
+ reply->writeInt32(timelineID);
+ }
+ return OK;
+ } break;
+
+ case COMMON_TIME_TO_LOCAL_TIME: {
+ CHECK_INTERFACE(ICommonClock, data, reply);
+ int64_t commonTime = data.readInt64();
+ int64_t localTime;
+ status_t status = commonTimeToLocalTime(commonTime, &localTime);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt64(localTime);
+ }
+ return OK;
+ } break;
+
+ case LOCAL_TIME_TO_COMMON_TIME: {
+ CHECK_INTERFACE(ICommonClock, data, reply);
+ int64_t localTime = data.readInt64();
+ int64_t commonTime;
+ status_t status = localTimeToCommonTime(localTime, &commonTime);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt64(commonTime);
+ }
+ return OK;
+ } break;
+
+ case GET_COMMON_TIME: {
+ CHECK_INTERFACE(ICommonClock, data, reply);
+ int64_t commonTime;
+ status_t status = getCommonTime(&commonTime);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt64(commonTime);
+ }
+ return OK;
+ } break;
+
+ case GET_COMMON_FREQ: {
+ CHECK_INTERFACE(ICommonClock, data, reply);
+ uint64_t freq;
+ status_t status = getCommonFreq(&freq);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt64(freq);
+ }
+ return OK;
+ } break;
+
+ case GET_LOCAL_TIME: {
+ CHECK_INTERFACE(ICommonClock, data, reply);
+ int64_t localTime;
+ status_t status = getLocalTime(&localTime);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt64(localTime);
+ }
+ return OK;
+ } break;
+
+ case GET_LOCAL_FREQ: {
+ CHECK_INTERFACE(ICommonClock, data, reply);
+ uint64_t freq;
+ status_t status = getLocalFreq(&freq);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt64(freq);
+ }
+ return OK;
+ } break;
+
+ case GET_ESTIMATED_ERROR: {
+ CHECK_INTERFACE(ICommonClock, data, reply);
+ int32_t error;
+ status_t status = getEstimatedError(&error);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt32(error);
+ }
+ return OK;
+ } break;
+
+ case GET_TIMELINE_ID: {
+ CHECK_INTERFACE(ICommonClock, data, reply);
+ uint64_t id;
+ status_t status = getTimelineID(&id);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt64(static_cast<int64_t>(id));
+ }
+ return OK;
+ } break;
+
+ case GET_STATE: {
+ CHECK_INTERFACE(ICommonClock, data, reply);
+ State state;
+ status_t status = getState(&state);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt32(static_cast<int32_t>(state));
+ }
+ return OK;
+ } break;
+
+ case GET_MASTER_ADDRESS: {
+ CHECK_INTERFACE(ICommonClock, data, reply);
+ struct sockaddr_storage addr;
+ status_t status = getMasterAddr(&addr);
+
+ if ((status == OK) && !canSerializeSockaddr(&addr)) {
+ status = UNKNOWN_ERROR;
+ }
+
+ reply->writeInt32(status);
+
+ if (status == OK) {
+ serializeSockaddr(reply, &addr);
+ }
+
+ return OK;
+ } break;
+
+ case REGISTER_LISTENER: {
+ CHECK_INTERFACE(ICommonClock, data, reply);
+ sp<ICommonClockListener> listener =
+ interface_cast<ICommonClockListener>(data.readStrongBinder());
+ status_t status = registerListener(listener);
+ reply->writeInt32(status);
+ return OK;
+ } break;
+
+ case UNREGISTER_LISTENER: {
+ CHECK_INTERFACE(ICommonClock, data, reply);
+ sp<ICommonClockListener> listener =
+ interface_cast<ICommonClockListener>(data.readStrongBinder());
+ status_t status = unregisterListener(listener);
+ reply->writeInt32(status);
+ return OK;
+ } break;
+ }
+ return BBinder::onTransact(code, data, reply, flags);
+}
+
+/***** ICommonClockListener *****/
+
+enum {
+ ON_TIMELINE_CHANGED = IBinder::FIRST_CALL_TRANSACTION,
+};
+
+class BpCommonClockListener : public BpInterface<ICommonClockListener>
+{
+ public:
+ BpCommonClockListener(const sp<IBinder>& impl)
+ : BpInterface<ICommonClockListener>(impl) {}
+
+ virtual void onTimelineChanged(uint64_t timelineID) {
+ Parcel data, reply;
+ data.writeInterfaceToken(
+ ICommonClockListener::getInterfaceDescriptor());
+ data.writeInt64(timelineID);
+ remote()->transact(ON_TIMELINE_CHANGED, data, &reply);
+ }
+};
+
+IMPLEMENT_META_INTERFACE(CommonClockListener,
+ "android.os.ICommonClockListener");
+
+status_t BnCommonClockListener::onTransact(
+ uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) {
+ switch(code) {
+ case ON_TIMELINE_CHANGED: {
+ CHECK_INTERFACE(ICommonClockListener, data, reply);
+ uint32_t timelineID = data.readInt64();
+ onTimelineChanged(timelineID);
+ return NO_ERROR;
+ } break;
+ }
+
+ return BBinder::onTransact(code, data, reply, flags);
+}
+
+}; // namespace android
diff --git a/media/common_time/ICommonTimeConfig.cpp b/media/common_time/ICommonTimeConfig.cpp
new file mode 100644
index 0000000..8eb37cb
--- /dev/null
+++ b/media/common_time/ICommonTimeConfig.cpp
@@ -0,0 +1,508 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+#include <linux/socket.h>
+
+#include <common_time/ICommonTimeConfig.h>
+#include <binder/Parcel.h>
+
+#include "utils.h"
+
+namespace android {
+
+/***** ICommonTimeConfig *****/
+
+enum {
+ GET_MASTER_ELECTION_PRIORITY = IBinder::FIRST_CALL_TRANSACTION,
+ SET_MASTER_ELECTION_PRIORITY,
+ GET_MASTER_ELECTION_ENDPOINT,
+ SET_MASTER_ELECTION_ENDPOINT,
+ GET_MASTER_ELECTION_GROUP_ID,
+ SET_MASTER_ELECTION_GROUP_ID,
+ GET_INTERFACE_BINDING,
+ SET_INTERFACE_BINDING,
+ GET_MASTER_ANNOUNCE_INTERVAL,
+ SET_MASTER_ANNOUNCE_INTERVAL,
+ GET_CLIENT_SYNC_INTERVAL,
+ SET_CLIENT_SYNC_INTERVAL,
+ GET_PANIC_THRESHOLD,
+ SET_PANIC_THRESHOLD,
+ GET_AUTO_DISABLE,
+ SET_AUTO_DISABLE,
+ FORCE_NETWORKLESS_MASTER_MODE,
+};
+
+const String16 ICommonTimeConfig::kServiceName("common_time.config");
+
+class BpCommonTimeConfig : public BpInterface<ICommonTimeConfig>
+{
+ public:
+ BpCommonTimeConfig(const sp<IBinder>& impl)
+ : BpInterface<ICommonTimeConfig>(impl) {}
+
+ virtual status_t getMasterElectionPriority(uint8_t *priority) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_MASTER_ELECTION_PRIORITY,
+ data,
+ &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *priority = static_cast<uint8_t>(reply.readInt32());
+ }
+ }
+
+ return status;
+ }
+
+ virtual status_t setMasterElectionPriority(uint8_t priority) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ data.writeInt32(static_cast<int32_t>(priority));
+ status_t status = remote()->transact(SET_MASTER_ELECTION_PRIORITY,
+ data,
+ &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ }
+
+ return status;
+ }
+
+ virtual status_t getMasterElectionEndpoint(struct sockaddr_storage *addr) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_MASTER_ELECTION_ENDPOINT,
+ data,
+ &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ deserializeSockaddr(&reply, addr);
+ }
+ }
+
+ return status;
+ }
+
+ virtual status_t setMasterElectionEndpoint(
+ const struct sockaddr_storage *addr) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ if (!canSerializeSockaddr(addr))
+ return BAD_VALUE;
+ if (NULL == addr) {
+ data.writeInt32(0);
+ } else {
+ data.writeInt32(1);
+ serializeSockaddr(&data, addr);
+ }
+ status_t status = remote()->transact(SET_MASTER_ELECTION_ENDPOINT,
+ data,
+ &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ }
+
+ return status;
+ }
+
+ virtual status_t getMasterElectionGroupId(uint64_t *id) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_MASTER_ELECTION_GROUP_ID,
+ data,
+ &reply);
+
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *id = static_cast<uint64_t>(reply.readInt64());
+ }
+ }
+
+ return status;
+ }
+
+ virtual status_t setMasterElectionGroupId(uint64_t id) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ data.writeInt64(id);
+ status_t status = remote()->transact(SET_MASTER_ELECTION_GROUP_ID,
+ data,
+ &reply);
+
+ if (status == OK) {
+ status = reply.readInt32();
+ }
+
+ return status;
+ }
+
+ virtual status_t getInterfaceBinding(String16& ifaceName) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_INTERFACE_BINDING,
+ data,
+ &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ ifaceName = reply.readString16();
+ }
+ }
+
+ return status;
+ }
+
+ virtual status_t setInterfaceBinding(const String16& ifaceName) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ data.writeString16(ifaceName);
+ status_t status = remote()->transact(SET_INTERFACE_BINDING,
+ data,
+ &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ }
+
+ return status;
+ }
+
+ virtual status_t getMasterAnnounceInterval(int *interval) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_MASTER_ANNOUNCE_INTERVAL,
+ data,
+ &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *interval = reply.readInt32();
+ }
+ }
+
+ return status;
+ }
+
+ virtual status_t setMasterAnnounceInterval(int interval) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ data.writeInt32(interval);
+ status_t status = remote()->transact(SET_MASTER_ANNOUNCE_INTERVAL,
+ data,
+ &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ }
+
+ return status;
+ }
+
+ virtual status_t getClientSyncInterval(int *interval) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_CLIENT_SYNC_INTERVAL,
+ data,
+ &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *interval = reply.readInt32();
+ }
+ }
+
+ return status;
+ }
+
+ virtual status_t setClientSyncInterval(int interval) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ data.writeInt32(interval);
+ status_t status = remote()->transact(SET_CLIENT_SYNC_INTERVAL,
+ data,
+ &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ }
+
+ return status;
+ }
+
+ virtual status_t getPanicThreshold(int *threshold) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_PANIC_THRESHOLD,
+ data,
+ &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *threshold = reply.readInt32();
+ }
+ }
+
+ return status;
+ }
+
+ virtual status_t setPanicThreshold(int threshold) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ data.writeInt32(threshold);
+ status_t status = remote()->transact(SET_PANIC_THRESHOLD,
+ data,
+ &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ }
+
+ return status;
+ }
+
+ virtual status_t getAutoDisable(bool *autoDisable) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_AUTO_DISABLE,
+ data,
+ &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *autoDisable = (0 != reply.readInt32());
+ }
+ }
+
+ return status;
+ }
+
+ virtual status_t setAutoDisable(bool autoDisable) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ data.writeInt32(autoDisable ? 1 : 0);
+ status_t status = remote()->transact(SET_AUTO_DISABLE,
+ data,
+ &reply);
+
+ if (status == OK) {
+ status = reply.readInt32();
+ }
+
+ return status;
+ }
+
+ virtual status_t forceNetworklessMasterMode() {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ status_t status = remote()->transact(FORCE_NETWORKLESS_MASTER_MODE,
+ data,
+ &reply);
+
+ if (status == OK) {
+ status = reply.readInt32();
+ }
+
+ return status;
+ }
+};
+
+IMPLEMENT_META_INTERFACE(CommonTimeConfig, "android.os.ICommonTimeConfig");
+
+status_t BnCommonTimeConfig::onTransact(uint32_t code,
+ const Parcel& data,
+ Parcel* reply,
+ uint32_t flags) {
+ switch(code) {
+ case GET_MASTER_ELECTION_PRIORITY: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ uint8_t priority;
+ status_t status = getMasterElectionPriority(&priority);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt32(static_cast<int32_t>(priority));
+ }
+ return OK;
+ } break;
+
+ case SET_MASTER_ELECTION_PRIORITY: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ uint8_t priority = static_cast<uint8_t>(data.readInt32());
+ status_t status = setMasterElectionPriority(priority);
+ reply->writeInt32(status);
+ return OK;
+ } break;
+
+ case GET_MASTER_ELECTION_ENDPOINT: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ struct sockaddr_storage addr;
+ status_t status = getMasterElectionEndpoint(&addr);
+
+ if ((status == OK) && !canSerializeSockaddr(&addr)) {
+ status = UNKNOWN_ERROR;
+ }
+
+ reply->writeInt32(status);
+
+ if (status == OK) {
+ serializeSockaddr(reply, &addr);
+ }
+
+ return OK;
+ } break;
+
+ case SET_MASTER_ELECTION_ENDPOINT: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ struct sockaddr_storage addr;
+ int hasAddr = data.readInt32();
+
+ status_t status;
+ if (hasAddr) {
+ deserializeSockaddr(&data, &addr);
+ status = setMasterElectionEndpoint(&addr);
+ } else {
+ status = setMasterElectionEndpoint(&addr);
+ }
+
+ reply->writeInt32(status);
+ return OK;
+ } break;
+
+ case GET_MASTER_ELECTION_GROUP_ID: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ uint64_t id;
+ status_t status = getMasterElectionGroupId(&id);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt64(id);
+ }
+ return OK;
+ } break;
+
+ case SET_MASTER_ELECTION_GROUP_ID: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ uint64_t id = static_cast<uint64_t>(data.readInt64());
+ status_t status = setMasterElectionGroupId(id);
+ reply->writeInt32(status);
+ return OK;
+ } break;
+
+ case GET_INTERFACE_BINDING: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ String16 ret;
+ status_t status = getInterfaceBinding(ret);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeString16(ret);
+ }
+ return OK;
+ } break;
+
+ case SET_INTERFACE_BINDING: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ String16 ifaceName;
+ ifaceName = data.readString16();
+ status_t status = setInterfaceBinding(ifaceName);
+ reply->writeInt32(status);
+ return OK;
+ } break;
+
+ case GET_MASTER_ANNOUNCE_INTERVAL: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ int interval;
+ status_t status = getMasterAnnounceInterval(&interval);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt32(interval);
+ }
+ return OK;
+ } break;
+
+ case SET_MASTER_ANNOUNCE_INTERVAL: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ int interval = data.readInt32();
+ status_t status = setMasterAnnounceInterval(interval);
+ reply->writeInt32(status);
+ return OK;
+ } break;
+
+ case GET_CLIENT_SYNC_INTERVAL: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ int interval;
+ status_t status = getClientSyncInterval(&interval);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt32(interval);
+ }
+ return OK;
+ } break;
+
+ case SET_CLIENT_SYNC_INTERVAL: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ int interval = data.readInt32();
+ status_t status = setClientSyncInterval(interval);
+ reply->writeInt32(status);
+ return OK;
+ } break;
+
+ case GET_PANIC_THRESHOLD: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ int threshold;
+ status_t status = getPanicThreshold(&threshold);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt32(threshold);
+ }
+ return OK;
+ } break;
+
+ case SET_PANIC_THRESHOLD: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ int threshold = data.readInt32();
+ status_t status = setPanicThreshold(threshold);
+ reply->writeInt32(status);
+ return OK;
+ } break;
+
+ case GET_AUTO_DISABLE: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ bool autoDisable;
+ status_t status = getAutoDisable(&autoDisable);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt32(autoDisable ? 1 : 0);
+ }
+ return OK;
+ } break;
+
+ case SET_AUTO_DISABLE: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ bool autoDisable = (0 != data.readInt32());
+ status_t status = setAutoDisable(autoDisable);
+ reply->writeInt32(status);
+ return OK;
+ } break;
+
+ case FORCE_NETWORKLESS_MASTER_MODE: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ status_t status = forceNetworklessMasterMode();
+ reply->writeInt32(status);
+ return OK;
+ } break;
+ }
+ return BBinder::onTransact(code, data, reply, flags);
+}
+
+}; // namespace android
+
diff --git a/media/common_time/cc_helper.cpp b/media/common_time/cc_helper.cpp
new file mode 100644
index 0000000..8d8556c
--- /dev/null
+++ b/media/common_time/cc_helper.cpp
@@ -0,0 +1,129 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <stdint.h>
+
+#include <common_time/cc_helper.h>
+#include <common_time/ICommonClock.h>
+#include <utils/threads.h>
+
+namespace android {
+
+Mutex CCHelper::lock_;
+sp<ICommonClock> CCHelper::common_clock_;
+sp<ICommonClockListener> CCHelper::common_clock_listener_;
+uint32_t CCHelper::ref_count_ = 0;
+
+bool CCHelper::verifyClock_l() {
+ bool ret = false;
+
+ if (common_clock_ == NULL) {
+ common_clock_ = ICommonClock::getInstance();
+ if (common_clock_ == NULL)
+ goto bailout;
+ }
+
+ if (ref_count_ > 0) {
+ if (common_clock_listener_ == NULL) {
+ common_clock_listener_ = new CommonClockListener();
+ if (common_clock_listener_ == NULL)
+ goto bailout;
+
+ if (OK != common_clock_->registerListener(common_clock_listener_))
+ goto bailout;
+ }
+ }
+
+ ret = true;
+
+bailout:
+ if (!ret) {
+ common_clock_listener_ = NULL;
+ common_clock_ = NULL;
+ }
+ return ret;
+}
+
+CCHelper::CCHelper() {
+ Mutex::Autolock lock(&lock_);
+ ref_count_++;
+ verifyClock_l();
+}
+
+CCHelper::~CCHelper() {
+ Mutex::Autolock lock(&lock_);
+
+ assert(ref_count_ > 0);
+ ref_count_--;
+
+ // If we were the last CCHelper instance in the system, and we had
+ // previously register a listener, unregister it now so that the common time
+ // service has the chance to go into auto-disabled mode.
+ if (!ref_count_ &&
+ (common_clock_ != NULL) &&
+ (common_clock_listener_ != NULL)) {
+ common_clock_->unregisterListener(common_clock_listener_);
+ common_clock_listener_ = NULL;
+ }
+}
+
+void CCHelper::CommonClockListener::onTimelineChanged(uint64_t timelineID) {
+ // do nothing; listener is only really used as a token so the server can
+ // find out when clients die.
+}
+
+// Helper methods which attempts to make calls to the common time binder
+// service. If the first attempt fails with DEAD_OBJECT, the helpers will
+// attempt to make a connection to the service again (assuming that the process
+// hosting the service had crashed and the client proxy we are holding is dead)
+// If the second attempt fails, or no connection can be made, the we let the
+// error propagate up the stack and let the caller deal with the situation as
+// best they can.
+#define CCHELPER_METHOD(decl, call) \
+ status_t CCHelper::decl { \
+ Mutex::Autolock lock(&lock_); \
+ \
+ if (!verifyClock_l()) \
+ return DEAD_OBJECT; \
+ \
+ status_t status = common_clock_->call; \
+ if (DEAD_OBJECT == status) { \
+ if (!verifyClock_l()) \
+ return DEAD_OBJECT; \
+ status = common_clock_->call; \
+ } \
+ \
+ return status; \
+ }
+
+#define VERIFY_CLOCK()
+
+CCHELPER_METHOD(isCommonTimeValid(bool* valid, uint32_t* timelineID),
+ isCommonTimeValid(valid, timelineID))
+CCHELPER_METHOD(commonTimeToLocalTime(int64_t commonTime, int64_t* localTime),
+ commonTimeToLocalTime(commonTime, localTime))
+CCHELPER_METHOD(localTimeToCommonTime(int64_t localTime, int64_t* commonTime),
+ localTimeToCommonTime(localTime, commonTime))
+CCHELPER_METHOD(getCommonTime(int64_t* commonTime),
+ getCommonTime(commonTime))
+CCHELPER_METHOD(getCommonFreq(uint64_t* freq),
+ getCommonFreq(freq))
+CCHELPER_METHOD(getLocalTime(int64_t* localTime),
+ getLocalTime(localTime))
+CCHELPER_METHOD(getLocalFreq(uint64_t* freq),
+ getLocalFreq(freq))
+
+} // namespace android
diff --git a/media/common_time/local_clock.cpp b/media/common_time/local_clock.cpp
new file mode 100644
index 0000000..a7c61fc
--- /dev/null
+++ b/media/common_time/local_clock.cpp
@@ -0,0 +1,92 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "common_time"
+#include <utils/Log.h>
+
+#include <assert.h>
+#include <stdint.h>
+
+#include <common_time/local_clock.h>
+#include <hardware/hardware.h>
+#include <hardware/local_time_hal.h>
+#include <utils/Errors.h>
+#include <utils/threads.h>
+
+namespace android {
+
+Mutex LocalClock::dev_lock_;
+local_time_hw_device_t* LocalClock::dev_ = NULL;
+
+LocalClock::LocalClock() {
+ int res;
+ const hw_module_t* mod;
+
+ AutoMutex lock(&dev_lock_);
+
+ if (dev_ != NULL)
+ return;
+
+ res = hw_get_module_by_class(LOCAL_TIME_HARDWARE_MODULE_ID, NULL, &mod);
+ if (res) {
+ ALOGE("Failed to open local time HAL module (res = %d)", res);
+ } else {
+ res = local_time_hw_device_open(mod, &dev_);
+ if (res) {
+ ALOGE("Failed to open local time HAL device (res = %d)", res);
+ dev_ = NULL;
+ }
+ }
+}
+
+bool LocalClock::initCheck() {
+ return (NULL != dev_);
+}
+
+int64_t LocalClock::getLocalTime() {
+ assert(NULL != dev_);
+ assert(NULL != dev_->get_local_time);
+
+ return dev_->get_local_time(dev_);
+}
+
+uint64_t LocalClock::getLocalFreq() {
+ assert(NULL != dev_);
+ assert(NULL != dev_->get_local_freq);
+
+ return dev_->get_local_freq(dev_);
+}
+
+status_t LocalClock::setLocalSlew(int16_t rate) {
+ assert(NULL != dev_);
+
+ if (!dev_->set_local_slew)
+ return INVALID_OPERATION;
+
+ return static_cast<status_t>(dev_->set_local_slew(dev_, rate));
+}
+
+int32_t LocalClock::getDebugLog(struct local_time_debug_event* records,
+ int max_records) {
+ assert(NULL != dev_);
+
+ if (!dev_->get_debug_log)
+ return INVALID_OPERATION;
+
+ return dev_->get_debug_log(dev_, records, max_records);
+}
+
+} // namespace android
diff --git a/media/common_time/utils.cpp b/media/common_time/utils.cpp
new file mode 100644
index 0000000..6539171
--- /dev/null
+++ b/media/common_time/utils.cpp
@@ -0,0 +1,89 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <arpa/inet.h>
+#include <linux/socket.h>
+
+#include <binder/Parcel.h>
+
+namespace android {
+
+bool canSerializeSockaddr(const struct sockaddr_storage* addr) {
+ switch (addr->ss_family) {
+ case AF_INET:
+ case AF_INET6:
+ return true;
+ default:
+ return false;
+ }
+}
+
+void serializeSockaddr(Parcel* p, const struct sockaddr_storage* addr) {
+ switch (addr->ss_family) {
+ case AF_INET: {
+ const struct sockaddr_in* s =
+ reinterpret_cast<const struct sockaddr_in*>(addr);
+ p->writeInt32(AF_INET);
+ p->writeInt32(ntohl(s->sin_addr.s_addr));
+ p->writeInt32(static_cast<int32_t>(ntohs(s->sin_port)));
+ } break;
+
+ case AF_INET6: {
+ const struct sockaddr_in6* s =
+ reinterpret_cast<const struct sockaddr_in6*>(addr);
+ const int32_t* a =
+ reinterpret_cast<const int32_t*>(s->sin6_addr.s6_addr);
+ p->writeInt32(AF_INET6);
+ p->writeInt32(ntohl(a[0]));
+ p->writeInt32(ntohl(a[1]));
+ p->writeInt32(ntohl(a[2]));
+ p->writeInt32(ntohl(a[3]));
+ p->writeInt32(static_cast<int32_t>(ntohs(s->sin6_port)));
+ p->writeInt32(ntohl(s->sin6_flowinfo));
+ p->writeInt32(ntohl(s->sin6_scope_id));
+ } break;
+ }
+}
+
+void deserializeSockaddr(const Parcel* p, struct sockaddr_storage* addr) {
+ memset(addr, 0, sizeof(addr));
+
+ addr->ss_family = p->readInt32();
+ switch(addr->ss_family) {
+ case AF_INET: {
+ struct sockaddr_in* s =
+ reinterpret_cast<struct sockaddr_in*>(addr);
+ s->sin_addr.s_addr = htonl(p->readInt32());
+ s->sin_port = htons(static_cast<uint16_t>(p->readInt32()));
+ } break;
+
+ case AF_INET6: {
+ struct sockaddr_in6* s =
+ reinterpret_cast<struct sockaddr_in6*>(addr);
+ int32_t* a = reinterpret_cast<int32_t*>(s->sin6_addr.s6_addr);
+
+ a[0] = htonl(p->readInt32());
+ a[1] = htonl(p->readInt32());
+ a[2] = htonl(p->readInt32());
+ a[3] = htonl(p->readInt32());
+ s->sin6_port = htons(static_cast<uint16_t>(p->readInt32()));
+ s->sin6_flowinfo = htonl(p->readInt32());
+ s->sin6_scope_id = htonl(p->readInt32());
+ } break;
+ }
+}
+
+} // namespace android
diff --git a/media/common_time/utils.h b/media/common_time/utils.h
new file mode 100644
index 0000000..ce79d0d
--- /dev/null
+++ b/media/common_time/utils.h
@@ -0,0 +1,34 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_LIBCOMMONCLOCK_UTILS_H
+#define ANDROID_LIBCOMMONCLOCK_UTILS_H
+
+#include <linux/socket.h>
+
+#include <binder/Parcel.h>
+#include <utils/Errors.h>
+
+namespace android {
+
+extern bool canSerializeSockaddr(const struct sockaddr_storage* addr);
+extern void serializeSockaddr(Parcel* p, const struct sockaddr_storage* addr);
+extern status_t deserializeSockaddr(const Parcel* p,
+ struct sockaddr_storage* addr);
+
+}; // namespace android
+
+#endif // ANDROID_LIBCOMMONCLOCK_UTILS_H
diff --git a/media/libaah_rtp/Android.mk b/media/libaah_rtp/Android.mk
new file mode 100644
index 0000000..6c927ba
--- /dev/null
+++ b/media/libaah_rtp/Android.mk
@@ -0,0 +1,40 @@
+LOCAL_PATH:= $(call my-dir)
+#
+# libaah_rtp
+#
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE := libaah_rtp
+LOCAL_MODULE_TAGS := optional
+
+LOCAL_SRC_FILES := \
+ aah_decoder_pump.cpp \
+ aah_rx_player.cpp \
+ aah_rx_player_core.cpp \
+ aah_rx_player_ring_buffer.cpp \
+ aah_rx_player_substream.cpp \
+ aah_tx_packet.cpp \
+ aah_tx_player.cpp \
+ aah_tx_sender.cpp \
+ pipe_event.cpp
+
+LOCAL_C_INCLUDES := \
+ frameworks/base/include \
+ frameworks/base/media \
+ frameworks/base/media/libstagefright \
+ frameworks/native/include/media/openmax
+
+LOCAL_SHARED_LIBRARIES := \
+ libcommon_time_client \
+ libbinder \
+ libmedia \
+ libmedia_native \
+ libstagefright \
+ libstagefright_foundation \
+ libutils
+
+LOCAL_LDLIBS := \
+ -lpthread
+
+include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libaah_rtp/aah_decoder_pump.cpp b/media/libaah_rtp/aah_decoder_pump.cpp
new file mode 100644
index 0000000..bebba54
--- /dev/null
+++ b/media/libaah_rtp/aah_decoder_pump.cpp
@@ -0,0 +1,519 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "LibAAH_RTP"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
+#include <poll.h>
+#include <pthread.h>
+
+#include <common_time/cc_helper.h>
+#include <media/AudioSystem.h>
+#include <media/AudioTrack.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/OMXClient.h>
+#include <media/stagefright/OMXCodec.h>
+#include <media/stagefright/Utils.h>
+#include <utils/Timers.h>
+#include <utils/threads.h>
+
+#include "aah_decoder_pump.h"
+
+namespace android {
+
+static const long long kLongDecodeErrorThreshold = 1000000ll;
+static const uint32_t kMaxLongErrorsBeforeFatal = 3;
+static const uint32_t kMaxErrorsBeforeFatal = 60;
+
+AAH_DecoderPump::AAH_DecoderPump(OMXClient& omx)
+ : omx_(omx)
+ , thread_status_(OK)
+ , renderer_(NULL)
+ , last_queued_pts_valid_(false)
+ , last_queued_pts_(0)
+ , last_ts_transform_valid_(false)
+ , last_volume_(0xFF) {
+ thread_ = new ThreadWrapper(this);
+}
+
+AAH_DecoderPump::~AAH_DecoderPump() {
+ shutdown();
+}
+
+status_t AAH_DecoderPump::initCheck() {
+ if (thread_ == NULL) {
+ ALOGE("Failed to allocate thread");
+ return NO_MEMORY;
+ }
+
+ return OK;
+}
+
+status_t AAH_DecoderPump::queueForDecode(MediaBuffer* buf) {
+ if (NULL == buf) {
+ return BAD_VALUE;
+ }
+
+ if (OK != thread_status_) {
+ return thread_status_;
+ }
+
+ { // Explicit scope for AutoMutex pattern.
+ AutoMutex lock(&thread_lock_);
+ in_queue_.push_back(buf);
+ }
+
+ thread_cond_.signal();
+
+ return OK;
+}
+
+void AAH_DecoderPump::queueToRenderer(MediaBuffer* decoded_sample) {
+ Mutex::Autolock lock(&render_lock_);
+ sp<MetaData> meta;
+ int64_t ts;
+ status_t res;
+
+ // Fetch the metadata and make sure the sample has a timestamp. We
+ // cannot render samples which are missing PTSs.
+ meta = decoded_sample->meta_data();
+ if ((meta == NULL) || (!meta->findInt64(kKeyTime, &ts))) {
+ ALOGV("Decoded sample missing timestamp, cannot render.");
+ CHECK(false);
+ } else {
+ // If we currently are not holding on to a renderer, go ahead and
+ // make one now.
+ if (NULL == renderer_) {
+ renderer_ = new TimedAudioTrack();
+ if (NULL != renderer_) {
+ int frameCount;
+ AudioTrack::getMinFrameCount(&frameCount,
+ AUDIO_STREAM_DEFAULT,
+ static_cast<int>(format_sample_rate_));
+ audio_channel_mask_t ch_format =
+ audio_channel_out_mask_from_count(format_channels_);
+
+ res = renderer_->set(AUDIO_STREAM_DEFAULT,
+ format_sample_rate_,
+ AUDIO_FORMAT_PCM_16_BIT,
+ ch_format,
+ frameCount);
+ if (res != OK) {
+ ALOGE("Failed to setup audio renderer. (res = %d)", res);
+ delete renderer_;
+ renderer_ = NULL;
+ } else {
+ CHECK(last_ts_transform_valid_);
+
+ res = renderer_->setMediaTimeTransform(
+ last_ts_transform_, TimedAudioTrack::COMMON_TIME);
+ if (res != NO_ERROR) {
+ ALOGE("Failed to set media time transform on AudioTrack"
+ " (res = %d)", res);
+ delete renderer_;
+ renderer_ = NULL;
+ } else {
+ float volume = static_cast<float>(last_volume_)
+ / 255.0f;
+ if (renderer_->setVolume(volume, volume) != OK) {
+ ALOGW("%s: setVolume failed", __FUNCTION__);
+ }
+
+ renderer_->start();
+ }
+ }
+ } else {
+ ALOGE("Failed to allocate AudioTrack to use as a renderer.");
+ }
+ }
+
+ if (NULL != renderer_) {
+ uint8_t* decoded_data =
+ reinterpret_cast<uint8_t*>(decoded_sample->data());
+ uint32_t decoded_amt = decoded_sample->range_length();
+ decoded_data += decoded_sample->range_offset();
+
+ sp<IMemory> pcm_payload;
+ res = renderer_->allocateTimedBuffer(decoded_amt, &pcm_payload);
+ if (res != OK) {
+ ALOGE("Failed to allocate %d byte audio track buffer."
+ " (res = %d)", decoded_amt, res);
+ } else {
+ memcpy(pcm_payload->pointer(), decoded_data, decoded_amt);
+
+ res = renderer_->queueTimedBuffer(pcm_payload, ts);
+ if (res != OK) {
+ ALOGE("Failed to queue %d byte audio track buffer with"
+ " media PTS %lld. (res = %d)", decoded_amt, ts, res);
+ } else {
+ last_queued_pts_valid_ = true;
+ last_queued_pts_ = ts;
+ }
+ }
+
+ } else {
+ ALOGE("No renderer, dropping audio payload.");
+ }
+ }
+}
+
+void AAH_DecoderPump::stopAndCleanupRenderer() {
+ if (NULL == renderer_) {
+ return;
+ }
+
+ renderer_->stop();
+ delete renderer_;
+ renderer_ = NULL;
+}
+
+void AAH_DecoderPump::setRenderTSTransform(const LinearTransform& trans) {
+ Mutex::Autolock lock(&render_lock_);
+
+ if (last_ts_transform_valid_ && !memcmp(&trans,
+ &last_ts_transform_,
+ sizeof(trans))) {
+ return;
+ }
+
+ last_ts_transform_ = trans;
+ last_ts_transform_valid_ = true;
+
+ if (NULL != renderer_) {
+ status_t res = renderer_->setMediaTimeTransform(
+ last_ts_transform_, TimedAudioTrack::COMMON_TIME);
+ if (res != NO_ERROR) {
+ ALOGE("Failed to set media time transform on AudioTrack"
+ " (res = %d)", res);
+ }
+ }
+}
+
+void AAH_DecoderPump::setRenderVolume(uint8_t volume) {
+ Mutex::Autolock lock(&render_lock_);
+
+ if (volume == last_volume_) {
+ return;
+ }
+
+ last_volume_ = volume;
+ if (renderer_ != NULL) {
+ float volume = static_cast<float>(last_volume_) / 255.0f;
+ if (renderer_->setVolume(volume, volume) != OK) {
+ ALOGW("%s: setVolume failed", __FUNCTION__);
+ }
+ }
+}
+
+// isAboutToUnderflow is something of a hack used to figure out when it might be
+// time to give up on trying to fill in a gap in the RTP sequence and simply
+// move on with a discontinuity. If we had perfect knowledge of when we were
+// going to underflow, it would not be a hack, but unfortunately we do not.
+// Right now, we just take the PTS of the last sample queued, and check to see
+// if its presentation time is within kAboutToUnderflowThreshold from now. If
+// it is, then we say that we are about to underflow. This decision is based on
+// two (possibly invalid) assumptions.
+//
+// 1) The transmitter is leading the clock by more than
+// kAboutToUnderflowThreshold.
+// 2) The delta between the PTS of the last sample queued and the next sample
+// is less than the transmitter's clock lead amount.
+//
+// Right now, the default transmitter lead time is 1 second, which is a pretty
+// large number and greater than the 50mSec that kAboutToUnderflowThreshold is
+// currently set to. This should satisfy assumption #1 for now, but changes to
+// the transmitter clock lead time could effect this.
+//
+// For non-sparse streams with a homogeneous sample rate (the vast majority of
+// streams in the world), the delta between any two adjacent PTSs will always be
+// the homogeneous sample period. It is very uncommon to see a sample period
+// greater than the 1 second clock lead we are currently using, and you
+// certainly will not see it in an MP3 file which should satisfy assumption #2.
+// Sparse audio streams (where no audio is transmitted for long periods of
+// silence) and extremely low framerate video stream (like an MPEG-2 slideshow
+// or the video stream for a pay TV audio channel) are examples of streams which
+// might violate assumption #2.
+bool AAH_DecoderPump::isAboutToUnderflow(int64_t threshold) {
+ Mutex::Autolock lock(&render_lock_);
+
+ // If we have never queued anything to the decoder, we really don't know if
+ // we are going to underflow or not.
+ if (!last_queued_pts_valid_ || !last_ts_transform_valid_) {
+ return false;
+ }
+
+ // Don't have access to Common Time? If so, then things are Very Bad
+ // elsewhere in the system; it pretty much does not matter what we do here.
+ // Since we cannot really tell if we are about to underflow or not, its
+ // probably best to assume that we are not and proceed accordingly.
+ int64_t tt_now;
+ if (OK != cc_helper_.getCommonTime(&tt_now)) {
+ return false;
+ }
+
+ // Transform from media time to common time.
+ int64_t last_queued_pts_tt;
+ if (!last_ts_transform_.doForwardTransform(last_queued_pts_,
+ &last_queued_pts_tt)) {
+ return false;
+ }
+
+ // Check to see if we are underflowing.
+ return ((tt_now + threshold - last_queued_pts_tt) > 0);
+}
+
+void* AAH_DecoderPump::workThread() {
+ // No need to lock when accessing decoder_ from the thread. The
+ // implementation of init and shutdown ensure that other threads never touch
+ // decoder_ while the work thread is running.
+ CHECK(decoder_ != NULL);
+ CHECK(format_ != NULL);
+
+ // Start the decoder and note its result code. If something goes horribly
+ // wrong, callers of queueForDecode and getOutput will be able to detect
+ // that the thread encountered a fatal error and shut down by examining
+ // thread_status_.
+ thread_status_ = decoder_->start(format_.get());
+ if (OK != thread_status_) {
+ ALOGE("AAH_DecoderPump's work thread failed to start decoder"
+ " (res = %d)", thread_status_);
+ return NULL;
+ }
+
+ DurationTimer decode_timer;
+ uint32_t consecutive_long_errors = 0;
+ uint32_t consecutive_errors = 0;
+
+ while (!thread_->exitPending()) {
+ status_t res;
+ MediaBuffer* bufOut = NULL;
+
+ decode_timer.start();
+ res = decoder_->read(&bufOut);
+ decode_timer.stop();
+
+ if (res == INFO_FORMAT_CHANGED) {
+ // Format has changed. Destroy our current renderer so that a new
+ // one can be created during queueToRenderer with the proper format.
+ //
+ // TODO : In order to transition seamlessly, we should change this
+ // to put the old renderer in a queue to play out completely before
+ // we destroy it. We can still create a new renderer, the timed
+ // nature of the renderer should ensure a seamless splice.
+ stopAndCleanupRenderer();
+ res = OK;
+ }
+
+ // Try to be a little nuanced in our handling of actual decode errors.
+ // Errors could happen because of minor stream corruption or because of
+ // transient resource limitations. In these cases, we would rather drop
+ // a little bit of output and ride out the unpleasantness then throw up
+ // our hands and abort everything.
+ //
+ // OTOH - When things are really bad (like we have a non-transient
+ // resource or bookkeeping issue, or the stream being fed to us is just
+ // complete and total garbage) we really want to terminate playback and
+ // raise an error condition all the way up to the application level so
+ // they can deal with it.
+ //
+ // Unfortunately, the error codes returned by the decoder can be a
+ // little non-specific. For example, if an OMXCodec times out
+ // attempting to obtain an output buffer, the error we get back is a
+ // generic -1. Try to distinguish between this resource timeout error
+ // and ES corruption error by timing how long the decode operation
+ // takes. Maintain accounting for both errors and "long errors". If we
+ // get more than a certain number consecutive errors of either type,
+ // consider it fatal and shutdown (which will cause the error to
+ // propagate all of the way up to the application level). The threshold
+ // for "long errors" is deliberately much lower than that of normal
+ // decode errors, both because of how long they take to happen and
+ // because they generally indicate resource limitation errors which are
+ // unlikely to go away in pathologically bad cases (in contrast to
+ // stream corruption errors which might happen 20 times in a row and
+ // then be suddenly OK again)
+ if (res != OK) {
+ consecutive_errors++;
+ if (decode_timer.durationUsecs() >= kLongDecodeErrorThreshold)
+ consecutive_long_errors++;
+
+ CHECK(NULL == bufOut);
+
+ ALOGW("%s: Failed to decode data (res = %d)",
+ __PRETTY_FUNCTION__, res);
+
+ if ((consecutive_errors >= kMaxErrorsBeforeFatal) ||
+ (consecutive_long_errors >= kMaxLongErrorsBeforeFatal)) {
+ ALOGE("%s: Maximum decode error threshold has been reached."
+ " There have been %d consecutive decode errors, and %d"
+ " consecutive decode operations which resulted in errors"
+ " and took more than %lld uSec to process. The last"
+ " decode operation took %lld uSec.",
+ __PRETTY_FUNCTION__,
+ consecutive_errors, consecutive_long_errors,
+ kLongDecodeErrorThreshold, decode_timer.durationUsecs());
+ thread_status_ = res;
+ break;
+ }
+
+ continue;
+ }
+
+ if (NULL == bufOut) {
+ ALOGW("%s: Successful decode, but no buffer produced",
+ __PRETTY_FUNCTION__);
+ continue;
+ }
+
+ // Successful decode (with actual output produced). Clear the error
+ // counters.
+ consecutive_errors = 0;
+ consecutive_long_errors = 0;
+
+ queueToRenderer(bufOut);
+ bufOut->release();
+ }
+
+ decoder_->stop();
+ stopAndCleanupRenderer();
+
+ return NULL;
+}
+
+status_t AAH_DecoderPump::init(const sp<MetaData>& params) {
+ Mutex::Autolock lock(&init_lock_);
+
+ if (decoder_ != NULL) {
+ // already inited
+ return OK;
+ }
+
+ if (params == NULL) {
+ return BAD_VALUE;
+ }
+
+ if (!params->findInt32(kKeyChannelCount, &format_channels_)) {
+ return BAD_VALUE;
+ }
+
+ if (!params->findInt32(kKeySampleRate, &format_sample_rate_)) {
+ return BAD_VALUE;
+ }
+
+ CHECK(OK == thread_status_);
+ CHECK(decoder_ == NULL);
+
+ status_t ret_val = UNKNOWN_ERROR;
+
+ // Cache the format and attempt to create the decoder.
+ format_ = params;
+ decoder_ = OMXCodec::Create(
+ omx_.interface(), // IOMX Handle
+ format_, // Metadata for substream (indicates codec)
+ false, // Make a decoder, not an encoder
+ sp<MediaSource>(this)); // We will be the source for this codec.
+
+ if (decoder_ == NULL) {
+ ALOGE("Failed to allocate decoder in %s", __PRETTY_FUNCTION__);
+ goto bailout;
+ }
+
+ // Fire up the pump thread. It will take care of starting and stopping the
+ // decoder.
+ ret_val = thread_->run("aah_decode_pump", ANDROID_PRIORITY_AUDIO);
+ if (OK != ret_val) {
+ ALOGE("Failed to start work thread in %s (res = %d)",
+ __PRETTY_FUNCTION__, ret_val);
+ goto bailout;
+ }
+
+bailout:
+ if (OK != ret_val) {
+ decoder_ = NULL;
+ format_ = NULL;
+ }
+
+ return OK;
+}
+
+status_t AAH_DecoderPump::shutdown() {
+ Mutex::Autolock lock(&init_lock_);
+ return shutdown_l();
+}
+
+status_t AAH_DecoderPump::shutdown_l() {
+ thread_->requestExit();
+ thread_cond_.signal();
+ thread_->requestExitAndWait();
+
+ for (MBQueue::iterator iter = in_queue_.begin();
+ iter != in_queue_.end();
+ ++iter) {
+ (*iter)->release();
+ }
+ in_queue_.clear();
+
+ last_queued_pts_valid_ = false;
+ last_ts_transform_valid_ = false;
+ last_volume_ = 0xFF;
+ thread_status_ = OK;
+
+ decoder_ = NULL;
+ format_ = NULL;
+
+ return OK;
+}
+
+status_t AAH_DecoderPump::read(MediaBuffer **buffer,
+ const ReadOptions *options) {
+ if (!buffer) {
+ return BAD_VALUE;
+ }
+
+ *buffer = NULL;
+
+ // While its not time to shut down, and we have no data to process, wait.
+ AutoMutex lock(&thread_lock_);
+ while (!thread_->exitPending() && in_queue_.empty())
+ thread_cond_.wait(thread_lock_);
+
+ // At this point, if its not time to shutdown then we must have something to
+ // process. Go ahead and pop the front of the queue for processing.
+ if (!thread_->exitPending()) {
+ CHECK(!in_queue_.empty());
+
+ *buffer = *(in_queue_.begin());
+ in_queue_.erase(in_queue_.begin());
+ }
+
+ // If we managed to get a buffer, then everything must be OK. If not, then
+ // we must be shutting down.
+ return (NULL == *buffer) ? INVALID_OPERATION : OK;
+}
+
+AAH_DecoderPump::ThreadWrapper::ThreadWrapper(AAH_DecoderPump* owner)
+ : Thread(false /* canCallJava*/ )
+ , owner_(owner) {
+}
+
+bool AAH_DecoderPump::ThreadWrapper::threadLoop() {
+ CHECK(NULL != owner_);
+ owner_->workThread();
+ return false;
+}
+
+} // namespace android
diff --git a/media/libaah_rtp/aah_decoder_pump.h b/media/libaah_rtp/aah_decoder_pump.h
new file mode 100644
index 0000000..4d57e49
--- /dev/null
+++ b/media/libaah_rtp/aah_decoder_pump.h
@@ -0,0 +1,107 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __DECODER_PUMP_H__
+#define __DECODER_PUMP_H__
+
+#include <pthread.h>
+
+#include <common_time/cc_helper.h>
+#include <media/stagefright/MediaSource.h>
+#include <utils/LinearTransform.h>
+#include <utils/List.h>
+#include <utils/threads.h>
+
+namespace android {
+
+class MetaData;
+class OMXClient;
+class TimedAudioTrack;
+
+class AAH_DecoderPump : public MediaSource {
+ public:
+ explicit AAH_DecoderPump(OMXClient& omx);
+ status_t initCheck();
+
+ status_t queueForDecode(MediaBuffer* buf);
+
+ status_t init(const sp<MetaData>& params);
+ status_t shutdown();
+
+ void setRenderTSTransform(const LinearTransform& trans);
+ void setRenderVolume(uint8_t volume);
+ bool isAboutToUnderflow(int64_t threshold);
+ bool getStatus() const { return thread_status_; }
+
+ // MediaSource methods
+ virtual status_t start(MetaData *params) { return OK; }
+ virtual sp<MetaData> getFormat() { return format_; }
+ virtual status_t stop() { return OK; }
+ virtual status_t read(MediaBuffer **buffer,
+ const ReadOptions *options);
+
+ protected:
+ virtual ~AAH_DecoderPump();
+
+ private:
+ class ThreadWrapper : public Thread {
+ public:
+ friend class AAH_DecoderPump;
+ explicit ThreadWrapper(AAH_DecoderPump* owner);
+
+ private:
+ virtual bool threadLoop();
+ AAH_DecoderPump* owner_;
+
+ DISALLOW_EVIL_CONSTRUCTORS(ThreadWrapper);
+ };
+
+ void* workThread();
+ virtual status_t shutdown_l();
+ void queueToRenderer(MediaBuffer* decoded_sample);
+ void stopAndCleanupRenderer();
+
+ sp<MetaData> format_;
+ int32_t format_channels_; // channel count, not channel mask
+ int32_t format_sample_rate_;
+
+ sp<MediaSource> decoder_;
+ OMXClient& omx_;
+ Mutex init_lock_;
+
+ sp<ThreadWrapper> thread_;
+ Condition thread_cond_;
+ Mutex thread_lock_;
+ status_t thread_status_;
+
+ Mutex render_lock_;
+ TimedAudioTrack* renderer_;
+ bool last_queued_pts_valid_;
+ int64_t last_queued_pts_;
+ bool last_ts_transform_valid_;
+ LinearTransform last_ts_transform_;
+ uint8_t last_volume_;
+ CCHelper cc_helper_;
+
+ // protected by the thread_lock_
+ typedef List<MediaBuffer*> MBQueue;
+ MBQueue in_queue_;
+
+ DISALLOW_EVIL_CONSTRUCTORS(AAH_DecoderPump);
+};
+
+} // namespace android
+#endif // __DECODER_PUMP_H__
diff --git a/media/libaah_rtp/aah_rx_player.cpp b/media/libaah_rtp/aah_rx_player.cpp
new file mode 100644
index 0000000..9dd79fd
--- /dev/null
+++ b/media/libaah_rtp/aah_rx_player.cpp
@@ -0,0 +1,288 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "LibAAH_RTP"
+//#define LOG_NDEBUG 0
+
+#include <binder/IServiceManager.h>
+#include <media/MediaPlayerInterface.h>
+#include <utils/Log.h>
+
+#include "aah_rx_player.h"
+
+namespace android {
+
+const uint32_t AAH_RXPlayer::kRTPRingBufferSize = 1 << 10;
+
+sp<MediaPlayerBase> createAAH_RXPlayer() {
+ sp<MediaPlayerBase> ret = new AAH_RXPlayer();
+ return ret;
+}
+
+AAH_RXPlayer::AAH_RXPlayer()
+ : ring_buffer_(kRTPRingBufferSize)
+ , substreams_(NULL) {
+ thread_wrapper_ = new ThreadWrapper(*this);
+
+ is_playing_ = false;
+ multicast_joined_ = false;
+ transmitter_known_ = false;
+ current_epoch_known_ = false;
+ data_source_set_ = false;
+ sock_fd_ = -1;
+
+ substreams_.setCapacity(4);
+
+ memset(&listen_addr_, 0, sizeof(listen_addr_));
+ memset(&transmitter_addr_, 0, sizeof(transmitter_addr_));
+
+ fetchAudioFlinger();
+}
+
+AAH_RXPlayer::~AAH_RXPlayer() {
+ reset_l();
+ CHECK(substreams_.size() == 0);
+ omx_.disconnect();
+}
+
+status_t AAH_RXPlayer::initCheck() {
+ if (thread_wrapper_ == NULL) {
+ ALOGE("Failed to allocate thread wrapper!");
+ return NO_MEMORY;
+ }
+
+ if (!ring_buffer_.initCheck()) {
+ ALOGE("Failed to allocate reassembly ring buffer!");
+ return NO_MEMORY;
+ }
+
+ // Check for the presense of the common time service by attempting to query
+ // for CommonTime's frequency. If we get an error back, we cannot talk to
+ // the service at all and should abort now.
+ status_t res;
+ uint64_t freq;
+ res = cc_helper_.getCommonFreq(&freq);
+ if (OK != res) {
+ ALOGE("Failed to connect to common time service!");
+ return res;
+ }
+
+ return omx_.connect();
+}
+
+status_t AAH_RXPlayer::setDataSource(
+ const char *url,
+ const KeyedVector<String8, String8> *headers) {
+ AutoMutex api_lock(&api_lock_);
+ uint32_t a, b, c, d;
+ uint16_t port;
+
+ if (data_source_set_) {
+ return INVALID_OPERATION;
+ }
+
+ if (NULL == url) {
+ return BAD_VALUE;
+ }
+
+ if (5 != sscanf(url, "%*[^:/]://%u.%u.%u.%u:%hu", &a, &b, &c, &d, &port)) {
+ ALOGE("Failed to parse URL \"%s\"", url);
+ return BAD_VALUE;
+ }
+
+ if ((a > 255) || (b > 255) || (c > 255) || (d > 255) || (port == 0)) {
+ ALOGE("Bad multicast address \"%s\"", url);
+ return BAD_VALUE;
+ }
+
+ ALOGI("setDataSource :: %u.%u.%u.%u:%hu", a, b, c, d, port);
+
+ a = (a << 24) | (b << 16) | (c << 8) | d;
+
+ memset(&listen_addr_, 0, sizeof(listen_addr_));
+ listen_addr_.sin_family = AF_INET;
+ listen_addr_.sin_port = htons(port);
+ listen_addr_.sin_addr.s_addr = htonl(a);
+ data_source_set_ = true;
+
+ return OK;
+}
+
+status_t AAH_RXPlayer::setDataSource(int fd, int64_t offset, int64_t length) {
+ return INVALID_OPERATION;
+}
+
+status_t AAH_RXPlayer::setVideoSurface(const sp<Surface>& surface) {
+ return OK;
+}
+
+status_t AAH_RXPlayer::setVideoSurfaceTexture(
+ const sp<ISurfaceTexture>& surfaceTexture) {
+ return OK;
+}
+
+status_t AAH_RXPlayer::prepare() {
+ return OK;
+}
+
+status_t AAH_RXPlayer::prepareAsync() {
+ sendEvent(MEDIA_PREPARED);
+ return OK;
+}
+
+status_t AAH_RXPlayer::start() {
+ AutoMutex api_lock(&api_lock_);
+
+ if (is_playing_) {
+ return OK;
+ }
+
+ status_t res = startWorkThread();
+ is_playing_ = (res == OK);
+ return res;
+}
+
+status_t AAH_RXPlayer::stop() {
+ return pause();
+}
+
+status_t AAH_RXPlayer::pause() {
+ AutoMutex api_lock(&api_lock_);
+ stopWorkThread();
+ CHECK(sock_fd_ < 0);
+ is_playing_ = false;
+ return OK;
+}
+
+bool AAH_RXPlayer::isPlaying() {
+ AutoMutex api_lock(&api_lock_);
+ return is_playing_;
+}
+
+status_t AAH_RXPlayer::seekTo(int msec) {
+ sendEvent(MEDIA_SEEK_COMPLETE);
+ return OK;
+}
+
+status_t AAH_RXPlayer::getCurrentPosition(int *msec) {
+ if (NULL != msec) {
+ *msec = 0;
+ }
+ return OK;
+}
+
+status_t AAH_RXPlayer::getDuration(int *msec) {
+ if (NULL != msec) {
+ *msec = 1;
+ }
+ return OK;
+}
+
+status_t AAH_RXPlayer::reset() {
+ AutoMutex api_lock(&api_lock_);
+ reset_l();
+ return OK;
+}
+
+void AAH_RXPlayer::reset_l() {
+ stopWorkThread();
+ CHECK(sock_fd_ < 0);
+ CHECK(!multicast_joined_);
+ is_playing_ = false;
+ data_source_set_ = false;
+ transmitter_known_ = false;
+ memset(&listen_addr_, 0, sizeof(listen_addr_));
+}
+
+status_t AAH_RXPlayer::setLooping(int loop) {
+ return OK;
+}
+
+player_type AAH_RXPlayer::playerType() {
+ return AAH_RX_PLAYER;
+}
+
+status_t AAH_RXPlayer::setParameter(int key, const Parcel &request) {
+ return ERROR_UNSUPPORTED;
+}
+
+status_t AAH_RXPlayer::getParameter(int key, Parcel *reply) {
+ return ERROR_UNSUPPORTED;
+}
+
+status_t AAH_RXPlayer::invoke(const Parcel& request, Parcel *reply) {
+ if (!reply) {
+ return BAD_VALUE;
+ }
+
+ int32_t magic;
+ status_t err = request.readInt32(&magic);
+ if (err != OK) {
+ reply->writeInt32(err);
+ return OK;
+ }
+
+ if (magic != 0x12345) {
+ reply->writeInt32(BAD_VALUE);
+ return OK;
+ }
+
+ int32_t methodID;
+ err = request.readInt32(&methodID);
+ if (err != OK) {
+ reply->writeInt32(err);
+ return OK;
+ }
+
+ switch (methodID) {
+ // Get Volume
+ case INVOKE_GET_MASTER_VOLUME: {
+ if (audio_flinger_ != NULL) {
+ reply->writeInt32(OK);
+ reply->writeFloat(audio_flinger_->masterVolume());
+ } else {
+ reply->writeInt32(UNKNOWN_ERROR);
+ }
+ } break;
+
+ // Set Volume
+ case INVOKE_SET_MASTER_VOLUME: {
+ float targetVol = request.readFloat();
+ reply->writeInt32(audio_flinger_->setMasterVolume(targetVol));
+ } break;
+
+ default: return BAD_VALUE;
+ }
+
+ return OK;
+}
+
+void AAH_RXPlayer::fetchAudioFlinger() {
+ if (audio_flinger_ == NULL) {
+ sp<IServiceManager> sm = defaultServiceManager();
+ sp<IBinder> binder;
+ binder = sm->getService(String16("media.audio_flinger"));
+
+ if (binder == NULL) {
+ ALOGW("AAH_RXPlayer failed to fetch handle to audio flinger."
+ " Master volume control will not be possible.");
+ }
+
+ audio_flinger_ = interface_cast<IAudioFlinger>(binder);
+ }
+}
+
+} // namespace android
diff --git a/media/libaah_rtp/aah_rx_player.h b/media/libaah_rtp/aah_rx_player.h
new file mode 100644
index 0000000..ba5617e
--- /dev/null
+++ b/media/libaah_rtp/aah_rx_player.h
@@ -0,0 +1,318 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __AAH_RX_PLAYER_H__
+#define __AAH_RX_PLAYER_H__
+
+#include <common_time/cc_helper.h>
+#include <media/MediaPlayerInterface.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/MediaBuffer.h>
+#include <media/stagefright/MediaSource.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/OMXClient.h>
+#include <netinet/in.h>
+#include <utils/KeyedVector.h>
+#include <utils/LinearTransform.h>
+#include <utils/threads.h>
+
+#include "aah_decoder_pump.h"
+#include "pipe_event.h"
+
+namespace android {
+
+class AAH_RXPlayer : public MediaPlayerInterface {
+ public:
+ AAH_RXPlayer();
+
+ virtual status_t initCheck();
+ virtual status_t setDataSource(const char *url,
+ const KeyedVector<String8, String8>*
+ headers);
+ virtual status_t setDataSource(int fd, int64_t offset, int64_t length);
+ virtual status_t setVideoSurface(const sp<Surface>& surface);
+ virtual status_t setVideoSurfaceTexture(const sp<ISurfaceTexture>&
+ surfaceTexture);
+ virtual status_t prepare();
+ virtual status_t prepareAsync();
+ virtual status_t start();
+ virtual status_t stop();
+ virtual status_t pause();
+ virtual bool isPlaying();
+ virtual status_t seekTo(int msec);
+ virtual status_t getCurrentPosition(int *msec);
+ virtual status_t getDuration(int *msec);
+ virtual status_t reset();
+ virtual status_t setLooping(int loop);
+ virtual player_type playerType();
+ virtual status_t setParameter(int key, const Parcel &request);
+ virtual status_t getParameter(int key, Parcel *reply);
+ virtual status_t invoke(const Parcel& request, Parcel *reply);
+
+ protected:
+ virtual ~AAH_RXPlayer();
+
+ private:
+ class ThreadWrapper : public Thread {
+ public:
+ friend class AAH_RXPlayer;
+ explicit ThreadWrapper(AAH_RXPlayer& player)
+ : Thread(false /* canCallJava */ )
+ , player_(player) { }
+
+ virtual bool threadLoop() { return player_.threadLoop(); }
+
+ private:
+ AAH_RXPlayer& player_;
+
+ DISALLOW_EVIL_CONSTRUCTORS(ThreadWrapper);
+ };
+
+#pragma pack(push, 1)
+ // PacketBuffers are structures used by the RX ring buffer. The ring buffer
+ // is a ring of pointers to PacketBuffer structures which act as variable
+ // length byte arrays and hold the contents of received UDP packets. Rather
+ // than make this a structure which hold a length and a pointer to another
+ // allocated structure (which would require two allocations), this struct
+ // uses a structure overlay pattern where allocation for the byte array
+ // consists of allocating (arrayLen + sizeof(ssize_t)) bytes of data from
+ // whatever pool/heap the packet buffer pulls from, and then overlaying the
+ // packed PacketBuffer structure on top of the allocation. The one-byte
+ // array at the end of the structure serves as an offset to the the data
+ // portion of the allocation; packet buffers are never allocated on the
+ // stack or using the new operator. Instead, the static allocate-byte-array
+ // and destroy methods handle the allocate and overlay pattern. They also
+ // allow for a potential future optimization where instead of just
+ // allocating blocks from the process global heap and overlaying, the
+ // allocator is replaced with a different implementation (private heap,
+ // free-list, circular buffer, etc) which reduces potential heap
+ // fragmentation issues which might arise from the frequent allocation and
+ // destruction of the received UDP traffic.
+ struct PacketBuffer {
+ ssize_t length_;
+ uint8_t data_[1];
+
+ // TODO : consider changing this to be some form of ring buffer or free
+ // pool system instead of just using the heap in order to avoid heap
+ // fragmentation.
+ static PacketBuffer* allocate(ssize_t length);
+ static void destroy(PacketBuffer* pb);
+
+ private:
+ // Force people to use allocate/destroy instead of new/delete.
+ PacketBuffer() { }
+ ~PacketBuffer() { }
+ };
+
+ struct RetransRequest {
+ uint32_t magic_;
+ uint32_t mcast_ip_;
+ uint16_t mcast_port_;
+ uint16_t start_seq_;
+ uint16_t end_seq_;
+ };
+#pragma pack(pop)
+
+ enum GapStatus {
+ kGS_NoGap = 0,
+ kGS_NormalGap,
+ kGS_FastStartGap,
+ };
+
+ struct SeqNoGap {
+ uint16_t start_seq_;
+ uint16_t end_seq_;
+ };
+
+ class RXRingBuffer {
+ public:
+ explicit RXRingBuffer(uint32_t capacity);
+ ~RXRingBuffer();
+
+ bool initCheck() const { return (ring_ != NULL); }
+ void reset();
+
+ // Push a packet buffer with a given sequence number into the ring
+ // buffer. pushBuffer will always consume the buffer pushed to it,
+ // either destroying it because it was a duplicate or overflow, or
+ // holding on to it in the ring. Callers should not hold any references
+ // to PacketBuffers after they have been pushed to the ring. Returns
+ // false in the case of a serious error (such as ring overflow).
+ // Callers should consider resetting the pipeline entirely in the event
+ // of a serious error.
+ bool pushBuffer(PacketBuffer* buf, uint16_t seq);
+
+ // Fetch the next buffer in the RTP sequence. Returns NULL if there is
+ // no buffer to fetch. If a non-NULL PacketBuffer is returned,
+ // is_discon will be set to indicate whether or not this PacketBuffer is
+ // discontiuous with any previously returned packet buffers. Packet
+ // buffers returned by fetchBuffer are the caller's responsibility; they
+ // must be certain to destroy the buffers when they are done.
+ PacketBuffer* fetchBuffer(bool* is_discon);
+
+ // Returns true and fills out the gap structure if the read pointer of
+ // the ring buffer is currently pointing to a gap which would stall a
+ // fetchBuffer operation. Returns false if the read pointer is not
+ // pointing to a gap in the sequence currently.
+ GapStatus fetchCurrentGap(SeqNoGap* gap);
+
+ // Causes the read pointer to skip over any portion of a gap indicated
+ // by nak. If nak is NULL, any gap currently blocking the read pointer
+ // will be completely skipped. If any portion of a gap is skipped, the
+ // next successful read from fetch buffer will indicate a discontinuity.
+ void processNAK(const SeqNoGap* nak = NULL);
+
+ // Compute the number of milliseconds until the inactivity timer for
+ // this RTP stream. Returns -1 if there is no active timeout, or 0 if
+ // the system has already timed out.
+ int computeInactivityTimeout();
+
+ private:
+ Mutex lock_;
+ PacketBuffer** ring_;
+ uint32_t capacity_;
+ uint32_t rd_;
+ uint32_t wr_;
+
+ uint16_t rd_seq_;
+ bool rd_seq_known_;
+ bool waiting_for_fast_start_;
+ bool fetched_first_packet_;
+
+ uint64_t rtp_activity_timeout_;
+ bool rtp_activity_timeout_valid_;
+
+ DISALLOW_EVIL_CONSTRUCTORS(RXRingBuffer);
+ };
+
+ class Substream : public virtual RefBase {
+ public:
+ Substream(uint32_t ssrc, OMXClient& omx);
+
+ void cleanupBufferInProgress();
+ void shutdown();
+ void processPayloadStart(uint8_t* buf,
+ uint32_t amt,
+ int32_t ts_lower);
+ void processPayloadCont (uint8_t* buf,
+ uint32_t amt);
+ void processTSTransform(const LinearTransform& trans);
+
+ bool isAboutToUnderflow();
+ uint32_t getSSRC() const { return ssrc_; }
+ uint16_t getProgramID() const { return (ssrc_ >> 5) & 0x1F; }
+ status_t getStatus() const { return status_; }
+
+ protected:
+ virtual ~Substream();
+
+ private:
+ void cleanupDecoder();
+ bool shouldAbort(const char* log_tag);
+ void processCompletedBuffer();
+ bool setupSubstreamMeta();
+ bool setupMP3SubstreamMeta();
+ bool setupAACSubstreamMeta();
+ bool setupSubstreamType(uint8_t substream_type,
+ uint8_t codec_type);
+
+ uint32_t ssrc_;
+ bool waiting_for_rap_;
+ status_t status_;
+
+ bool substream_details_known_;
+ uint8_t substream_type_;
+ uint8_t codec_type_;
+ const char* codec_mime_type_;
+ sp<MetaData> substream_meta_;
+
+ MediaBuffer* buffer_in_progress_;
+ uint32_t expected_buffer_size_;
+ uint32_t buffer_filled_;
+
+ Vector<uint8_t> aux_data_in_progress_;
+ uint32_t aux_data_expected_size_;
+
+ sp<AAH_DecoderPump> decoder_;
+
+ static int64_t kAboutToUnderflowThreshold;
+
+ DISALLOW_EVIL_CONSTRUCTORS(Substream);
+ };
+
+ typedef DefaultKeyedVector< uint32_t, sp<Substream> > SubstreamVec;
+
+ status_t startWorkThread();
+ void stopWorkThread();
+ virtual bool threadLoop();
+ bool setupSocket();
+ void cleanupSocket();
+ void resetPipeline();
+ void reset_l();
+ bool processRX(PacketBuffer* pb);
+ void processRingBuffer();
+ void processCommandPacket(PacketBuffer* pb);
+ bool processGaps();
+ int computeNextGapRetransmitTimeout();
+ void fetchAudioFlinger();
+
+ PipeEvent wakeup_work_thread_evt_;
+ sp<ThreadWrapper> thread_wrapper_;
+ Mutex api_lock_;
+ bool is_playing_;
+ bool data_source_set_;
+
+ struct sockaddr_in listen_addr_;
+ int sock_fd_;
+ bool multicast_joined_;
+
+ struct sockaddr_in transmitter_addr_;
+ bool transmitter_known_;
+
+ uint32_t current_epoch_;
+ bool current_epoch_known_;
+
+ SeqNoGap current_gap_;
+ GapStatus current_gap_status_;
+ uint64_t next_retrans_req_time_;
+
+ RXRingBuffer ring_buffer_;
+ SubstreamVec substreams_;
+ OMXClient omx_;
+ CCHelper cc_helper_;
+
+ // Connection to audio flinger used to hack a path to setMasterVolume.
+ sp<IAudioFlinger> audio_flinger_;
+
+ static const uint32_t kRTPRingBufferSize;
+ static const uint32_t kRetransRequestMagic;
+ static const uint32_t kFastStartRequestMagic;
+ static const uint32_t kRetransNAKMagic;
+ static const uint32_t kGapRerequestTimeoutUSec;
+ static const uint32_t kFastStartTimeoutUSec;
+ static const uint32_t kRTPActivityTimeoutUSec;
+
+ static const uint32_t INVOKE_GET_MASTER_VOLUME = 3;
+ static const uint32_t INVOKE_SET_MASTER_VOLUME = 4;
+
+ static uint64_t monotonicUSecNow();
+
+ DISALLOW_EVIL_CONSTRUCTORS(AAH_RXPlayer);
+};
+
+} // namespace android
+
+#endif // __AAH_RX_PLAYER_H__
diff --git a/media/libaah_rtp/aah_rx_player_core.cpp b/media/libaah_rtp/aah_rx_player_core.cpp
new file mode 100644
index 0000000..d6b31fd
--- /dev/null
+++ b/media/libaah_rtp/aah_rx_player_core.cpp
@@ -0,0 +1,809 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "LibAAH_RTP"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
+#include <fcntl.h>
+#include <poll.h>
+#include <sys/socket.h>
+#include <time.h>
+#include <utils/misc.h>
+
+#include <media/stagefright/Utils.h>
+
+#include "aah_rx_player.h"
+#include "aah_tx_packet.h"
+
+namespace android {
+
+const uint32_t AAH_RXPlayer::kRetransRequestMagic =
+ FOURCC('T','r','e','q');
+const uint32_t AAH_RXPlayer::kRetransNAKMagic =
+ FOURCC('T','n','a','k');
+const uint32_t AAH_RXPlayer::kFastStartRequestMagic =
+ FOURCC('T','f','s','t');
+const uint32_t AAH_RXPlayer::kGapRerequestTimeoutUSec = 75000;
+const uint32_t AAH_RXPlayer::kFastStartTimeoutUSec = 800000;
+const uint32_t AAH_RXPlayer::kRTPActivityTimeoutUSec = 10000000;
+
+static inline int16_t fetchInt16(uint8_t* data) {
+ return static_cast<int16_t>(U16_AT(data));
+}
+
+static inline int32_t fetchInt32(uint8_t* data) {
+ return static_cast<int32_t>(U32_AT(data));
+}
+
+static inline int64_t fetchInt64(uint8_t* data) {
+ return static_cast<int64_t>(U64_AT(data));
+}
+
+uint64_t AAH_RXPlayer::monotonicUSecNow() {
+ struct timespec now;
+ int res = clock_gettime(CLOCK_MONOTONIC, &now);
+ CHECK(res >= 0);
+
+ uint64_t ret = static_cast<uint64_t>(now.tv_sec) * 1000000;
+ ret += now.tv_nsec / 1000;
+
+ return ret;
+}
+
+status_t AAH_RXPlayer::startWorkThread() {
+ status_t res;
+ stopWorkThread();
+ res = thread_wrapper_->run("TRX_Player", PRIORITY_AUDIO);
+
+ if (res != OK) {
+ ALOGE("Failed to start work thread (res = %d)", res);
+ }
+
+ return res;
+}
+
+void AAH_RXPlayer::stopWorkThread() {
+ thread_wrapper_->requestExit(); // set the exit pending flag
+ wakeup_work_thread_evt_.setEvent();
+
+ status_t res;
+ res = thread_wrapper_->requestExitAndWait(); // block until thread exit.
+ if (res != OK) {
+ ALOGE("Failed to stop work thread (res = %d)", res);
+ }
+
+ wakeup_work_thread_evt_.clearPendingEvents();
+}
+
+void AAH_RXPlayer::cleanupSocket() {
+ if (sock_fd_ >= 0) {
+ if (multicast_joined_) {
+ int res;
+ struct ip_mreq mreq;
+ mreq.imr_multiaddr = listen_addr_.sin_addr;
+ mreq.imr_interface.s_addr = htonl(INADDR_ANY);
+ res = setsockopt(sock_fd_,
+ IPPROTO_IP,
+ IP_DROP_MEMBERSHIP,
+ &mreq, sizeof(mreq));
+ if (res < 0) {
+ ALOGW("Failed to leave multicast group. (%d, %d)", res, errno);
+ }
+ multicast_joined_ = false;
+ }
+
+ close(sock_fd_);
+ sock_fd_ = -1;
+ }
+
+ resetPipeline();
+}
+
+void AAH_RXPlayer::resetPipeline() {
+ ring_buffer_.reset();
+
+ // Explicitly shudown all of the active substreams, then call clear out the
+ // collection. Failure to clear out a substream can result in its decoder
+ // holding a reference to itself and therefor not going away when the
+ // collection is cleared.
+ for (size_t i = 0; i < substreams_.size(); ++i)
+ substreams_.valueAt(i)->shutdown();
+
+ substreams_.clear();
+
+ current_gap_status_ = kGS_NoGap;
+}
+
+bool AAH_RXPlayer::setupSocket() {
+ long flags;
+ int res, buf_size;
+ socklen_t opt_size;
+
+ cleanupSocket();
+ CHECK(sock_fd_ < 0);
+
+ // Make the socket
+ sock_fd_ = socket(AF_INET, SOCK_DGRAM, IPPROTO_UDP);
+ if (sock_fd_ < 0) {
+ ALOGE("Failed to create listen socket (errno %d)", errno);
+ goto bailout;
+ }
+
+ // Set non-blocking operation
+ flags = fcntl(sock_fd_, F_GETFL);
+ res = fcntl(sock_fd_, F_SETFL, flags | O_NONBLOCK);
+ if (res < 0) {
+ ALOGE("Failed to set socket (%d) to non-blocking mode (errno %d)",
+ sock_fd_, errno);
+ goto bailout;
+ }
+
+ // Bind to our port
+ struct sockaddr_in bind_addr;
+ memset(&bind_addr, 0, sizeof(bind_addr));
+ bind_addr.sin_family = AF_INET;
+ bind_addr.sin_addr.s_addr = INADDR_ANY;
+ bind_addr.sin_port = listen_addr_.sin_port;
+ res = bind(sock_fd_,
+ reinterpret_cast<const sockaddr*>(&bind_addr),
+ sizeof(bind_addr));
+ if (res < 0) {
+ uint32_t a = ntohl(bind_addr.sin_addr.s_addr);
+ uint16_t p = ntohs(bind_addr.sin_port);
+ ALOGE("Failed to bind socket (%d) to %d.%d.%d.%d:%hd. (errno %d)",
+ sock_fd_,
+ (a >> 24) & 0xFF,
+ (a >> 16) & 0xFF,
+ (a >> 8) & 0xFF,
+ (a ) & 0xFF,
+ p,
+ errno);
+
+ goto bailout;
+ }
+
+ buf_size = 1 << 16; // 64k
+ res = setsockopt(sock_fd_,
+ SOL_SOCKET, SO_RCVBUF,
+ &buf_size, sizeof(buf_size));
+ if (res < 0) {
+ ALOGW("Failed to increase socket buffer size to %d. (errno %d)",
+ buf_size, errno);
+ }
+
+ buf_size = 0;
+ opt_size = sizeof(buf_size);
+ res = getsockopt(sock_fd_,
+ SOL_SOCKET, SO_RCVBUF,
+ &buf_size, &opt_size);
+ if (res < 0) {
+ ALOGW("Failed to fetch socket buffer size. (errno %d)", errno);
+ } else {
+ ALOGI("RX socket buffer size is now %d bytes", buf_size);
+ }
+
+ if (listen_addr_.sin_addr.s_addr) {
+ // Join the multicast group and we should be good to go.
+ struct ip_mreq mreq;
+ mreq.imr_multiaddr = listen_addr_.sin_addr;
+ mreq.imr_interface.s_addr = htonl(INADDR_ANY);
+ res = setsockopt(sock_fd_,
+ IPPROTO_IP,
+ IP_ADD_MEMBERSHIP,
+ &mreq, sizeof(mreq));
+ if (res < 0) {
+ ALOGE("Failed to join multicast group. (errno %d)", errno);
+ goto bailout;
+ }
+ multicast_joined_ = true;
+ }
+
+ return true;
+
+bailout:
+ cleanupSocket();
+ return false;
+}
+
+bool AAH_RXPlayer::threadLoop() {
+ struct pollfd poll_fds[2];
+ bool process_more_right_now = false;
+
+ if (!setupSocket()) {
+ sendEvent(MEDIA_ERROR);
+ goto bailout;
+ }
+
+ while (!thread_wrapper_->exitPending()) {
+ // Step 1: Wait until there is something to do.
+ int gap_timeout = computeNextGapRetransmitTimeout();
+ int ring_timeout = ring_buffer_.computeInactivityTimeout();
+ int timeout = -1;
+
+ if (!ring_timeout) {
+ ALOGW("RTP inactivity timeout reached, resetting pipeline.");
+ resetPipeline();
+ timeout = gap_timeout;
+ } else {
+ if (gap_timeout < 0) {
+ timeout = ring_timeout;
+ } else if (ring_timeout < 0) {
+ timeout = gap_timeout;
+ } else {
+ timeout = (gap_timeout < ring_timeout) ? gap_timeout
+ : ring_timeout;
+ }
+ }
+
+ if ((0 != timeout) && (!process_more_right_now)) {
+ // Set up the events to wait on. Start with the wakeup pipe.
+ memset(&poll_fds, 0, sizeof(poll_fds));
+ poll_fds[0].fd = wakeup_work_thread_evt_.getWakeupHandle();
+ poll_fds[0].events = POLLIN;
+
+ // Add the RX socket.
+ poll_fds[1].fd = sock_fd_;
+ poll_fds[1].events = POLLIN;
+
+ // Wait for something interesing to happen.
+ int poll_res = poll(poll_fds, NELEM(poll_fds), timeout);
+ if (poll_res < 0) {
+ ALOGE("Fatal error (%d,%d) while waiting on events",
+ poll_res, errno);
+ sendEvent(MEDIA_ERROR);
+ goto bailout;
+ }
+ }
+
+ if (thread_wrapper_->exitPending()) {
+ break;
+ }
+
+ wakeup_work_thread_evt_.clearPendingEvents();
+ process_more_right_now = false;
+
+ // Step 2: Do we have data waiting in the socket? If so, drain the
+ // socket moving valid RTP information into the ring buffer to be
+ // processed.
+ if (poll_fds[1].revents) {
+ struct sockaddr_in from;
+ socklen_t from_len;
+
+ ssize_t res = 0;
+ while (!thread_wrapper_->exitPending()) {
+ // Check the size of any pending packet.
+ res = recv(sock_fd_, NULL, 0, MSG_PEEK | MSG_TRUNC);
+
+ // Error?
+ if (res < 0) {
+ // If the error is anything other than would block,
+ // something has gone very wrong.
+ if ((errno != EAGAIN) && (errno != EWOULDBLOCK)) {
+ ALOGE("Fatal socket error during recvfrom (%d, %d)",
+ (int)res, errno);
+ goto bailout;
+ }
+
+ // Socket is out of data, just break out of processing and
+ // wait for more.
+ break;
+ }
+
+ // Allocate a payload.
+ PacketBuffer* pb = PacketBuffer::allocate(res);
+ if (NULL == pb) {
+ ALOGE("Fatal error, failed to allocate packet buffer of"
+ " length %u", static_cast<uint32_t>(res));
+ goto bailout;
+ }
+
+ // Fetch the data.
+ from_len = sizeof(from);
+ res = recvfrom(sock_fd_, pb->data_, pb->length_, 0,
+ reinterpret_cast<struct sockaddr*>(&from),
+ &from_len);
+ if (res != pb->length_) {
+ ALOGE("Fatal error, fetched packet length (%d) does not"
+ " match peeked packet length (%u). This should never"
+ " happen. (errno = %d)",
+ static_cast<int>(res),
+ static_cast<uint32_t>(pb->length_),
+ errno);
+ }
+
+ bool drop_packet = false;
+ if (transmitter_known_) {
+ if (from.sin_addr.s_addr !=
+ transmitter_addr_.sin_addr.s_addr) {
+ uint32_t a = ntohl(from.sin_addr.s_addr);
+ uint16_t p = ntohs(from.sin_port);
+ ALOGV("Dropping packet from unknown transmitter"
+ " %u.%u.%u.%u:%hu",
+ ((a >> 24) & 0xFF),
+ ((a >> 16) & 0xFF),
+ ((a >> 8) & 0xFF),
+ ( a & 0xFF),
+ p);
+
+ drop_packet = true;
+ } else {
+ transmitter_addr_.sin_port = from.sin_port;
+ }
+ } else {
+ memcpy(&transmitter_addr_, &from, sizeof(from));
+ transmitter_known_ = true;
+ }
+
+ if (!drop_packet) {
+ bool serious_error = !processRX(pb);
+
+ if (serious_error) {
+ // Something went "seriously wrong". Currently, the
+ // only trigger for this should be a ring buffer
+ // overflow. The current failsafe behavior for when
+ // something goes seriously wrong is to just reset the
+ // pipeline. The system should behave as if this
+ // AAH_RXPlayer was just set up for the first time.
+ ALOGE("Something just went seriously wrong with the"
+ " pipeline. Resetting.");
+ resetPipeline();
+ }
+ } else {
+ PacketBuffer::destroy(pb);
+ }
+ }
+ }
+
+ // Step 3: Process any data we mave have accumulated in the ring buffer
+ // so far.
+ if (!thread_wrapper_->exitPending()) {
+ processRingBuffer();
+ }
+
+ // Step 4: At this point in time, the ring buffer should either be
+ // empty, or stalled in front of a gap caused by some dropped packets.
+ // Check on the current gap situation and deal with it in an appropriate
+ // fashion. If processGaps returns true, it means that it has given up
+ // on a gap and that we should try to process some more data
+ // immediately.
+ if (!thread_wrapper_->exitPending()) {
+ process_more_right_now = processGaps();
+ }
+
+ // Step 5: Check for fatal errors. If any of our substreams has
+ // encountered a fatal, unrecoverable, error, then propagate the error
+ // up to user level and shut down.
+ for (size_t i = 0; i < substreams_.size(); ++i) {
+ status_t status;
+ CHECK(substreams_.valueAt(i) != NULL);
+
+ status = substreams_.valueAt(i)->getStatus();
+ if (OK != status) {
+ ALOGE("Substream index %d has encountered an unrecoverable"
+ " error (%d). Signalling application level and shutting"
+ " down.", i, status);
+ sendEvent(MEDIA_ERROR);
+ goto bailout;
+ }
+ }
+ }
+
+bailout:
+ cleanupSocket();
+ return false;
+}
+
+bool AAH_RXPlayer::processRX(PacketBuffer* pb) {
+ CHECK(NULL != pb);
+
+ uint8_t* data = pb->data_;
+ ssize_t amt = pb->length_;
+ uint32_t nak_magic;
+ uint16_t seq_no;
+ uint32_t epoch;
+
+ // Every packet either starts with an RTP header which is at least 12 bytes
+ // long or is a retry NAK which is 14 bytes long. If there are fewer than
+ // 12 bytes here, this cannot be a proper RTP packet.
+ if (amt < 12) {
+ ALOGV("Dropping packet, too short to contain RTP header (%u bytes)",
+ static_cast<uint32_t>(amt));
+ goto drop_packet;
+ }
+
+ // Check to see if this is the special case of a NAK packet.
+ nak_magic = ntohl(*(reinterpret_cast<uint32_t*>(data)));
+ if (nak_magic == kRetransNAKMagic) {
+ // Looks like a NAK packet; make sure its long enough.
+
+ if (amt < static_cast<ssize_t>(sizeof(RetransRequest))) {
+ ALOGV("Dropping packet, too short to contain NAK payload"
+ " (%u bytes)", static_cast<uint32_t>(amt));
+ goto drop_packet;
+ }
+
+ SeqNoGap gap;
+ RetransRequest* rtr = reinterpret_cast<RetransRequest*>(data);
+ gap.start_seq_ = ntohs(rtr->start_seq_);
+ gap.end_seq_ = ntohs(rtr->end_seq_);
+
+ ALOGV("Process NAK for gap at [%hu, %hu]",
+ gap.start_seq_, gap.end_seq_);
+ ring_buffer_.processNAK(&gap);
+
+ return true;
+ }
+
+ // According to the TRTP spec, version should be 2, padding should be 0,
+ // extension should be 0 and CSRCCnt should be 0. If any of these tests
+ // fail, we chuck the packet.
+ if (data[0] != 0x80) {
+ ALOGV("Dropping packet, bad V/P/X/CSRCCnt field (0x%02x)",
+ data[0]);
+ goto drop_packet;
+ }
+
+ // Check the payload type. For TRTP, it should always be 100.
+ if ((data[1] & 0x7F) != 100) {
+ ALOGV("Dropping packet, bad payload type. (%u)",
+ data[1] & 0x7F);
+ goto drop_packet;
+ }
+
+ // Check whether the transmitter has begun a new epoch.
+ epoch = (U32_AT(data + 8) >> 10) & 0x3FFFFF;
+ if (current_epoch_known_) {
+ if (epoch != current_epoch_) {
+ ALOGV("%s: new epoch %u", __PRETTY_FUNCTION__, epoch);
+ current_epoch_ = epoch;
+ resetPipeline();
+ }
+ } else {
+ current_epoch_ = epoch;
+ current_epoch_known_ = true;
+ }
+
+ // Extract the sequence number and hand the packet off to the ring buffer
+ // for dropped packet detection and later processing.
+ seq_no = U16_AT(data + 2);
+ return ring_buffer_.pushBuffer(pb, seq_no);
+
+drop_packet:
+ PacketBuffer::destroy(pb);
+ return true;
+}
+
+void AAH_RXPlayer::processRingBuffer() {
+ PacketBuffer* pb;
+ bool is_discon;
+ sp<Substream> substream;
+ LinearTransform trans;
+ bool foundTrans = false;
+
+ while (NULL != (pb = ring_buffer_.fetchBuffer(&is_discon))) {
+ if (is_discon) {
+ // Abort all partially assembled payloads.
+ for (size_t i = 0; i < substreams_.size(); ++i) {
+ CHECK(substreams_.valueAt(i) != NULL);
+ substreams_.valueAt(i)->cleanupBufferInProgress();
+ }
+ }
+
+ uint8_t* data = pb->data_;
+ ssize_t amt = pb->length_;
+
+ // Should not have any non-RTP packets in the ring buffer. RTP packets
+ // must be at least 12 bytes long.
+ CHECK(amt >= 12);
+
+ // Extract the marker bit and the SSRC field.
+ bool marker = (data[1] & 0x80) != 0;
+ uint32_t ssrc = U32_AT(data + 8);
+
+ // Is this the start of a new TRTP payload? If so, the marker bit
+ // should be set and there are some things we should be checking for.
+ if (marker) {
+ // TRTP headers need to have at least a byte for version, a byte for
+ // payload type and flags, and 4 bytes for length.
+ if (amt < 18) {
+ ALOGV("Dropping packet, too short to contain TRTP header"
+ " (%u bytes)", static_cast<uint32_t>(amt));
+ goto process_next_packet;
+ }
+
+ // Check the TRTP version and extract the payload type/flags.
+ uint8_t trtp_version = data[12];
+ uint8_t payload_type = (data[13] >> 4) & 0xF;
+ uint8_t trtp_flags = data[13] & 0xF;
+
+ if (1 != trtp_version) {
+ ALOGV("Dropping packet, bad trtp version %hhu", trtp_version);
+ goto process_next_packet;
+ }
+
+ // Is there a timestamp transformation present on this packet? If
+ // so, extract it and pass it to the appropriate substreams.
+ if (trtp_flags & 0x02) {
+ ssize_t offset = 18 + ((trtp_flags & 0x01) ? 4 : 0);
+ if (amt < (offset + 24)) {
+ ALOGV("Dropping packet, too short to contain TRTP Timestamp"
+ " Transformation (%u bytes)",
+ static_cast<uint32_t>(amt));
+ goto process_next_packet;
+ }
+
+ trans.a_zero = fetchInt64(data + offset);
+ trans.b_zero = fetchInt64(data + offset + 16);
+ trans.a_to_b_numer = static_cast<int32_t>(
+ fetchInt32 (data + offset + 8));
+ trans.a_to_b_denom = U32_AT(data + offset + 12);
+ foundTrans = true;
+
+ uint32_t program_id = (ssrc >> 5) & 0x1F;
+ for (size_t i = 0; i < substreams_.size(); ++i) {
+ sp<Substream> iter = substreams_.valueAt(i);
+ CHECK(iter != NULL);
+
+ if (iter->getProgramID() == program_id) {
+ iter->processTSTransform(trans);
+ }
+ }
+ }
+
+ // Is this a command packet? If so, its not necessarily associate
+ // with one particular substream. Just give it to the command
+ // packet handler and then move on.
+ if (4 == payload_type) {
+ processCommandPacket(pb);
+ goto process_next_packet;
+ }
+ }
+
+ // If we got to here, then we are a normal packet. Find (or allocate)
+ // the substream we belong to and send the packet off to be processed.
+ substream = substreams_.valueFor(ssrc);
+ if (substream == NULL) {
+ substream = new Substream(ssrc, omx_);
+ if (substream == NULL) {
+ ALOGE("Failed to allocate substream for SSRC 0x%08x", ssrc);
+ goto process_next_packet;
+ }
+ substreams_.add(ssrc, substream);
+
+ if (foundTrans) {
+ substream->processTSTransform(trans);
+ }
+ }
+
+ CHECK(substream != NULL);
+
+ if (marker) {
+ // Start of a new TRTP payload for this substream. Extract the
+ // lower 32 bits of the timestamp and hand the buffer to the
+ // substream for processing.
+ uint32_t ts_lower = U32_AT(data + 4);
+ substream->processPayloadStart(data + 12, amt - 12, ts_lower);
+ } else {
+ // Continuation of an existing TRTP payload. Just hand it off to
+ // the substream for processing.
+ substream->processPayloadCont(data + 12, amt - 12);
+ }
+
+process_next_packet:
+ PacketBuffer::destroy(pb);
+ } // end of main processing while loop.
+}
+
+void AAH_RXPlayer::processCommandPacket(PacketBuffer* pb) {
+ CHECK(NULL != pb);
+
+ uint8_t* data = pb->data_;
+ ssize_t amt = pb->length_;
+
+ // verify that this packet meets the minimum length of a command packet
+ if (amt < 20) {
+ return;
+ }
+
+ uint8_t trtp_version = data[12];
+ uint8_t trtp_flags = data[13] & 0xF;
+
+ if (1 != trtp_version) {
+ ALOGV("Dropping packet, bad trtp version %hhu", trtp_version);
+ return;
+ }
+
+ // calculate the start of the command payload
+ ssize_t offset = 18;
+ if (trtp_flags & 0x01) {
+ // timestamp is present (4 bytes)
+ offset += 4;
+ }
+ if (trtp_flags & 0x02) {
+ // transform is present (24 bytes)
+ offset += 24;
+ }
+
+ // the packet must contain 2 bytes of command payload beyond the TRTP header
+ if (amt < offset + 2) {
+ return;
+ }
+
+ uint16_t command_id = U16_AT(data + offset);
+
+ switch (command_id) {
+ case TRTPControlPacket::kCommandNop:
+ break;
+
+ case TRTPControlPacket::kCommandEOS:
+ case TRTPControlPacket::kCommandFlush: {
+ uint16_t program_id = (U32_AT(data + 8) >> 5) & 0x1F;
+ ALOGI("*** %s flushing program_id=%d",
+ __PRETTY_FUNCTION__, program_id);
+
+ Vector<uint32_t> substreams_to_remove;
+ for (size_t i = 0; i < substreams_.size(); ++i) {
+ sp<Substream> iter = substreams_.valueAt(i);
+ if (iter->getProgramID() == program_id) {
+ iter->shutdown();
+ substreams_to_remove.add(iter->getSSRC());
+ }
+ }
+
+ for (size_t i = 0; i < substreams_to_remove.size(); ++i) {
+ substreams_.removeItem(substreams_to_remove[i]);
+ }
+ } break;
+ }
+}
+
+bool AAH_RXPlayer::processGaps() {
+ // Deal with the current gap situation. Specifically...
+ //
+ // 1) If a new gap has shown up, send a retransmit request to the
+ // transmitter.
+ // 2) If a gap we were working on has had a packet in the middle or at
+ // the end filled in, send another retransmit request for the begining
+ // portion of the gap. TRTP was designed for LANs where packet
+ // re-ordering is very unlikely; so see the middle or end of a gap
+ // filled in before the begining is an almost certain indication that
+ // a retransmission packet was also dropped.
+ // 3) If we have been working on a gap for a while and it still has not
+ // been filled in, send another retransmit request.
+ // 4) If the are no more gaps in the ring, clear the current_gap_status_
+ // flag to indicate that all is well again.
+
+ // Start by fetching the active gap status.
+ SeqNoGap gap;
+ bool send_retransmit_request = false;
+ bool ret_val = false;
+ GapStatus gap_status;
+ if (kGS_NoGap != (gap_status = ring_buffer_.fetchCurrentGap(&gap))) {
+ // Note: checking for a change in the end sequence number should cover
+ // moving on to an entirely new gap for case #1 as well as resending the
+ // begining of a gap range for case #2.
+ send_retransmit_request = (kGS_NoGap == current_gap_status_) ||
+ (current_gap_.end_seq_ != gap.end_seq_);
+
+ // If this is the same gap we have been working on, and it has timed
+ // out, then check to see if our substreams are about to underflow. If
+ // so, instead of sending another retransmit request, just give up on
+ // this gap and move on.
+ if (!send_retransmit_request &&
+ (kGS_NoGap != current_gap_status_) &&
+ (0 == computeNextGapRetransmitTimeout())) {
+
+ // If out current gap is the fast-start gap, don't bother to skip it
+ // because substreams look like the are about to underflow.
+ if ((kGS_FastStartGap != gap_status) ||
+ (current_gap_.end_seq_ != gap.end_seq_)) {
+ for (size_t i = 0; i < substreams_.size(); ++i) {
+ if (substreams_.valueAt(i)->isAboutToUnderflow()) {
+ ALOGV("About to underflow, giving up on gap [%hu, %hu]",
+ gap.start_seq_, gap.end_seq_);
+ ring_buffer_.processNAK();
+ current_gap_status_ = kGS_NoGap;
+ return true;
+ }
+ }
+ }
+
+ // Looks like no one is about to underflow. Just go ahead and send
+ // the request.
+ send_retransmit_request = true;
+ }
+ } else {
+ current_gap_status_ = kGS_NoGap;
+ }
+
+ if (send_retransmit_request) {
+ // If we have been working on a fast start, and it is still not filled
+ // in, even after the extended retransmit time out, give up and skip it.
+ // The system should fall back into its normal slow-start behavior.
+ if ((kGS_FastStartGap == current_gap_status_) &&
+ (current_gap_.end_seq_ == gap.end_seq_)) {
+ ALOGV("Fast start is taking forever; giving up.");
+ ring_buffer_.processNAK();
+ current_gap_status_ = kGS_NoGap;
+ return true;
+ }
+
+ // Send the request.
+ RetransRequest req;
+ uint32_t magic = (kGS_FastStartGap == gap_status)
+ ? kFastStartRequestMagic
+ : kRetransRequestMagic;
+ req.magic_ = htonl(magic);
+ req.mcast_ip_ = listen_addr_.sin_addr.s_addr;
+ req.mcast_port_ = listen_addr_.sin_port;
+ req.start_seq_ = htons(gap.start_seq_);
+ req.end_seq_ = htons(gap.end_seq_);
+
+ {
+ uint32_t a = ntohl(transmitter_addr_.sin_addr.s_addr);
+ uint16_t p = ntohs(transmitter_addr_.sin_port);
+ ALOGV("Sending to transmitter %u.%u.%u.%u:%hu",
+ ((a >> 24) & 0xFF),
+ ((a >> 16) & 0xFF),
+ ((a >> 8) & 0xFF),
+ ( a & 0xFF),
+ p);
+ }
+
+ int res = sendto(sock_fd_, &req, sizeof(req), 0,
+ reinterpret_cast<struct sockaddr*>(&transmitter_addr_),
+ sizeof(transmitter_addr_));
+ if (res < 0) {
+ ALOGE("Error when sending retransmit request (%d)", errno);
+ } else {
+ ALOGV("%s request for range [%hu, %hu] sent",
+ (kGS_FastStartGap == gap_status) ? "Fast Start"
+ : "Retransmit",
+ gap.start_seq_, gap.end_seq_);
+ }
+
+ // Update the current gap info.
+ current_gap_ = gap;
+ current_gap_status_ = gap_status;
+ next_retrans_req_time_ = monotonicUSecNow() +
+ ((kGS_FastStartGap == current_gap_status_)
+ ? kFastStartTimeoutUSec
+ : kGapRerequestTimeoutUSec);
+ }
+
+ return false;
+}
+
+// Compute when its time to send the next gap retransmission in milliseconds.
+// Returns < 0 for an infinite timeout (no gap) and 0 if its time to retransmit
+// right now.
+int AAH_RXPlayer::computeNextGapRetransmitTimeout() {
+ if (kGS_NoGap == current_gap_status_) {
+ return -1;
+ }
+
+ int64_t timeout_delta = next_retrans_req_time_ - monotonicUSecNow();
+
+ timeout_delta /= 1000;
+ if (timeout_delta <= 0) {
+ return 0;
+ }
+
+ return static_cast<uint32_t>(timeout_delta);
+}
+
+} // namespace android
diff --git a/media/libaah_rtp/aah_rx_player_ring_buffer.cpp b/media/libaah_rtp/aah_rx_player_ring_buffer.cpp
new file mode 100644
index 0000000..779405e
--- /dev/null
+++ b/media/libaah_rtp/aah_rx_player_ring_buffer.cpp
@@ -0,0 +1,366 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "LibAAH_RTP"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
+#include "aah_rx_player.h"
+
+namespace android {
+
+AAH_RXPlayer::RXRingBuffer::RXRingBuffer(uint32_t capacity) {
+ capacity_ = capacity;
+ rd_ = wr_ = 0;
+ ring_ = new PacketBuffer*[capacity];
+ memset(ring_, 0, sizeof(PacketBuffer*) * capacity);
+ reset();
+}
+
+AAH_RXPlayer::RXRingBuffer::~RXRingBuffer() {
+ reset();
+ delete[] ring_;
+}
+
+void AAH_RXPlayer::RXRingBuffer::reset() {
+ AutoMutex lock(&lock_);
+
+ if (NULL != ring_) {
+ while (rd_ != wr_) {
+ CHECK(rd_ < capacity_);
+ if (NULL != ring_[rd_]) {
+ PacketBuffer::destroy(ring_[rd_]);
+ ring_[rd_] = NULL;
+ }
+ rd_ = (rd_ + 1) % capacity_;
+ }
+ }
+
+ rd_ = wr_ = 0;
+ rd_seq_known_ = false;
+ waiting_for_fast_start_ = true;
+ fetched_first_packet_ = false;
+ rtp_activity_timeout_valid_ = false;
+}
+
+bool AAH_RXPlayer::RXRingBuffer::pushBuffer(PacketBuffer* buf,
+ uint16_t seq) {
+ AutoMutex lock(&lock_);
+ CHECK(NULL != ring_);
+ CHECK(NULL != buf);
+
+ rtp_activity_timeout_valid_ = true;
+ rtp_activity_timeout_ = monotonicUSecNow() + kRTPActivityTimeoutUSec;
+
+ // If the ring buffer is totally reset (we have never received a single
+ // payload) then we don't know the rd sequence number and this should be
+ // simple. We just store the payload, advance the wr pointer and record the
+ // initial sequence number.
+ if (!rd_seq_known_) {
+ CHECK(rd_ == wr_);
+ CHECK(NULL == ring_[wr_]);
+ CHECK(wr_ < capacity_);
+
+ ring_[wr_] = buf;
+ wr_ = (wr_ + 1) % capacity_;
+ rd_seq_ = seq;
+ rd_seq_known_ = true;
+ return true;
+ }
+
+ // Compute the seqence number of this payload and of the write pointer,
+ // normalized around the read pointer. IOW - transform the payload seq no
+ // and the wr pointer seq no into a space where the rd pointer seq no is
+ // zero. This will define 4 cases we can consider...
+ //
+ // 1) norm_seq == norm_wr_seq
+ // This payload is contiguous with the last. All is good.
+ //
+ // 2) ((norm_seq < norm_wr_seq) && (norm_seq >= norm_rd_seq)
+ // aka ((norm_seq < norm_wr_seq) && (norm_seq >= 0)
+ // This payload is in the past, in the unprocessed region of the ring
+ // buffer. It is probably a retransmit intended to fill in a dropped
+ // payload; it may be a duplicate.
+ //
+ // 3) ((norm_seq - norm_wr_seq) & 0x8000) != 0
+ // This payload is in the past compared to the write pointer (or so very
+ // far in the future that it has wrapped the seq no space), but not in
+ // the unprocessed region of the ring buffer. This could be a duplicate
+ // retransmit; we just drop these payloads unless we are waiting for our
+ // first fast start packet. If we are waiting for fast start, than this
+ // packet is probably the first packet of the fast start retransmission.
+ // If it will fit in the buffer, back up the read pointer to its position
+ // and clear the fast start flag, otherwise just drop it.
+ //
+ // 4) ((norm_seq - norm_wr_seq) & 0x8000) == 0
+ // This payload which is ahead of the next write pointer. This indicates
+ // that we have missed some payloads and need to request a retransmit.
+ // If norm_seq >= (capacity - 1), then the gap is so large that it would
+ // overflow the ring buffer and we should probably start to panic.
+
+ uint16_t norm_wr_seq = ((wr_ + capacity_ - rd_) % capacity_);
+ uint16_t norm_seq = seq - rd_seq_;
+
+ // Check for overflow first.
+ if ((!(norm_seq & 0x8000)) && (norm_seq >= (capacity_ - 1))) {
+ ALOGW("Ring buffer overflow; cap = %u, [rd, wr] = [%hu, %hu],"
+ " seq = %hu", capacity_, rd_seq_, norm_wr_seq + rd_seq_, seq);
+ PacketBuffer::destroy(buf);
+ return false;
+ }
+
+ // Check for case #1
+ if (norm_seq == norm_wr_seq) {
+ CHECK(wr_ < capacity_);
+ CHECK(NULL == ring_[wr_]);
+
+ ring_[wr_] = buf;
+ wr_ = (wr_ + 1) % capacity_;
+
+ CHECK(wr_ != rd_);
+ return true;
+ }
+
+ // Check case #2
+ uint32_t ring_pos = (rd_ + norm_seq) % capacity_;
+ if ((norm_seq < norm_wr_seq) && (!(norm_seq & 0x8000))) {
+ // Do we already have a payload for this slot? If so, then this looks
+ // like a duplicate retransmit. Just ignore it.
+ if (NULL != ring_[ring_pos]) {
+ ALOGD("RXed duplicate retransmit, seq = %hu", seq);
+ PacketBuffer::destroy(buf);
+ } else {
+ // Looks like we were missing this payload. Go ahead and store it.
+ ring_[ring_pos] = buf;
+ }
+
+ return true;
+ }
+
+ // Check case #3
+ if ((norm_seq - norm_wr_seq) & 0x8000) {
+ if (!waiting_for_fast_start_) {
+ ALOGD("RXed duplicate retransmit from before rd pointer, seq = %hu",
+ seq);
+ PacketBuffer::destroy(buf);
+ } else {
+ // Looks like a fast start fill-in. Go ahead and store it, assuming
+ // that we can fit it in the buffer.
+ uint32_t implied_ring_size = static_cast<uint32_t>(norm_wr_seq)
+ + (rd_seq_ - seq);
+
+ if (implied_ring_size >= (capacity_ - 1)) {
+ ALOGD("RXed what looks like a fast start packet (seq = %hu),"
+ " but packet is too far in the past to fit into the ring"
+ " buffer. Dropping.", seq);
+ PacketBuffer::destroy(buf);
+ } else {
+ ring_pos = (rd_ + capacity_ + seq - rd_seq_) % capacity_;
+ rd_seq_ = seq;
+ rd_ = ring_pos;
+ waiting_for_fast_start_ = false;
+
+ CHECK(ring_pos < capacity_);
+ CHECK(NULL == ring_[ring_pos]);
+ ring_[ring_pos] = buf;
+ }
+
+ }
+ return true;
+ }
+
+ // Must be in case #4 with no overflow. This packet fits in the current
+ // ring buffer, but is discontiuguous. Advance the write pointer leaving a
+ // gap behind.
+ uint32_t gap_len = (ring_pos + capacity_ - wr_) % capacity_;
+ ALOGD("Drop detected; %u packets, seq_range [%hu, %hu]",
+ gap_len,
+ rd_seq_ + norm_wr_seq,
+ rd_seq_ + norm_wr_seq + gap_len - 1);
+
+ CHECK(NULL == ring_[ring_pos]);
+ ring_[ring_pos] = buf;
+ wr_ = (ring_pos + 1) % capacity_;
+ CHECK(wr_ != rd_);
+
+ return true;
+}
+
+AAH_RXPlayer::PacketBuffer*
+AAH_RXPlayer::RXRingBuffer::fetchBuffer(bool* is_discon) {
+ AutoMutex lock(&lock_);
+ CHECK(NULL != ring_);
+ CHECK(NULL != is_discon);
+
+ // If the read seqence number is not known, then this ring buffer has not
+ // received a packet since being reset and there cannot be any packets to
+ // return. If we are still waiting for the first fast start packet to show
+ // up, we don't want to let any buffer be consumed yet because we expect to
+ // see a packet before the initial read sequence number show up shortly.
+ if (!rd_seq_known_ || waiting_for_fast_start_) {
+ *is_discon = false;
+ return NULL;
+ }
+
+ PacketBuffer* ret = NULL;
+ *is_discon = !fetched_first_packet_;
+
+ while ((rd_ != wr_) && (NULL == ret)) {
+ CHECK(rd_ < capacity_);
+
+ // If we hit a gap, stall and do not advance the read pointer. Let the
+ // higher level code deal with requesting retries and/or deciding to
+ // skip the current gap.
+ ret = ring_[rd_];
+ if (NULL == ret) {
+ break;
+ }
+
+ ring_[rd_] = NULL;
+ rd_ = (rd_ + 1) % capacity_;
+ ++rd_seq_;
+ }
+
+ if (NULL != ret) {
+ fetched_first_packet_ = true;
+ }
+
+ return ret;
+}
+
+AAH_RXPlayer::GapStatus
+AAH_RXPlayer::RXRingBuffer::fetchCurrentGap(SeqNoGap* gap) {
+ AutoMutex lock(&lock_);
+ CHECK(NULL != ring_);
+ CHECK(NULL != gap);
+
+ // If the read seqence number is not known, then this ring buffer has not
+ // received a packet since being reset and there cannot be any gaps.
+ if (!rd_seq_known_) {
+ return kGS_NoGap;
+ }
+
+ // If we are waiting for fast start, then the current gap is a fast start
+ // gap and it includes all packets before the read sequence number.
+ if (waiting_for_fast_start_) {
+ gap->start_seq_ =
+ gap->end_seq_ = rd_seq_ - 1;
+ return kGS_FastStartGap;
+ }
+
+ // If rd == wr, then the buffer is empty and there cannot be any gaps.
+ if (rd_ == wr_) {
+ return kGS_NoGap;
+ }
+
+ // If rd_ is currently pointing at an unprocessed packet, then there is no
+ // current gap.
+ CHECK(rd_ < capacity_);
+ if (NULL != ring_[rd_]) {
+ return kGS_NoGap;
+ }
+
+ // Looks like there must be a gap here. The start of the gap is the current
+ // rd sequence number, all we need to do now is determine its length in
+ // order to compute the end sequence number.
+ gap->start_seq_ = rd_seq_;
+ uint16_t end = rd_seq_;
+ uint32_t tmp = (rd_ + 1) % capacity_;
+ while ((tmp != wr_) && (NULL == ring_[tmp])) {
+ ++end;
+ tmp = (tmp + 1) % capacity_;
+ }
+ gap->end_seq_ = end;
+
+ return kGS_NormalGap;
+}
+
+void AAH_RXPlayer::RXRingBuffer::processNAK(const SeqNoGap* nak) {
+ AutoMutex lock(&lock_);
+ CHECK(NULL != ring_);
+
+ // If we were waiting for our first fast start fill-in packet, and we
+ // received a NAK, then apparantly we are not getting our fast start. Just
+ // clear the waiting flag and go back to normal behavior.
+ if (waiting_for_fast_start_) {
+ waiting_for_fast_start_ = false;
+ }
+
+ // If we have not received a packet since last reset, or there is no data in
+ // the ring, then there is nothing to skip.
+ if ((!rd_seq_known_) || (rd_ == wr_)) {
+ return;
+ }
+
+ // If rd_ is currently pointing at an unprocessed packet, then there is no
+ // gap to skip.
+ CHECK(rd_ < capacity_);
+ if (NULL != ring_[rd_]) {
+ return;
+ }
+
+ // Looks like there must be a gap here. Advance rd until we have passed
+ // over the portion of it indicated by nak (or all of the gap if nak is
+ // NULL). Then reset fetched_first_packet_ so that the next read will show
+ // up as being discontiguous.
+ uint16_t seq_after_gap = (NULL == nak) ? 0 : nak->end_seq_ + 1;
+ while ((rd_ != wr_) &&
+ (NULL == ring_[rd_]) &&
+ ((NULL == nak) || (seq_after_gap != rd_seq_))) {
+ rd_ = (rd_ + 1) % capacity_;
+ ++rd_seq_;
+ }
+ fetched_first_packet_ = false;
+}
+
+int AAH_RXPlayer::RXRingBuffer::computeInactivityTimeout() {
+ AutoMutex lock(&lock_);
+
+ if (!rtp_activity_timeout_valid_) {
+ return -1;
+ }
+
+ uint64_t now = monotonicUSecNow();
+ if (rtp_activity_timeout_ <= now) {
+ return 0;
+ }
+
+ return (rtp_activity_timeout_ - now) / 1000;
+}
+
+AAH_RXPlayer::PacketBuffer*
+AAH_RXPlayer::PacketBuffer::allocate(ssize_t length) {
+ if (length <= 0) {
+ return NULL;
+ }
+
+ uint32_t alloc_len = sizeof(PacketBuffer) + length;
+ PacketBuffer* ret = reinterpret_cast<PacketBuffer*>(
+ new uint8_t[alloc_len]);
+
+ if (NULL != ret) {
+ ret->length_ = length;
+ }
+
+ return ret;
+}
+
+void AAH_RXPlayer::PacketBuffer::destroy(PacketBuffer* pb) {
+ uint8_t* kill_me = reinterpret_cast<uint8_t*>(pb);
+ delete[] kill_me;
+}
+
+} // namespace android
diff --git a/media/libaah_rtp/aah_rx_player_substream.cpp b/media/libaah_rtp/aah_rx_player_substream.cpp
new file mode 100644
index 0000000..18b0e2b
--- /dev/null
+++ b/media/libaah_rtp/aah_rx_player_substream.cpp
@@ -0,0 +1,677 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "LibAAH_RTP"
+//#define LOG_NDEBUG 0
+
+#include <utils/Log.h>
+
+#include <include/avc_utils.h>
+#include <media/stagefright/MediaBuffer.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/OMXCodec.h>
+#include <media/stagefright/Utils.h>
+
+#include "aah_rx_player.h"
+#include "aah_tx_packet.h"
+
+inline uint32_t min(uint32_t a, uint32_t b) {
+ return (a < b ? a : b);
+}
+
+namespace android {
+
+int64_t AAH_RXPlayer::Substream::kAboutToUnderflowThreshold =
+ 50ull * 1000;
+
+AAH_RXPlayer::Substream::Substream(uint32_t ssrc, OMXClient& omx) {
+ ssrc_ = ssrc;
+ substream_details_known_ = false;
+ buffer_in_progress_ = NULL;
+ status_ = OK;
+ codec_mime_type_ = "";
+
+ decoder_ = new AAH_DecoderPump(omx);
+ if (decoder_ == NULL) {
+ ALOGE("%s failed to allocate decoder pump!", __PRETTY_FUNCTION__);
+ }
+ if (OK != decoder_->initCheck()) {
+ ALOGE("%s failed to initialize decoder pump!", __PRETTY_FUNCTION__);
+ }
+
+ // cleanupBufferInProgress will reset most of the internal state variables.
+ // Just need to make sure that buffer_in_progress_ is NULL before calling.
+ cleanupBufferInProgress();
+}
+
+AAH_RXPlayer::Substream::~Substream() {
+ shutdown();
+}
+
+void AAH_RXPlayer::Substream::shutdown() {
+ substream_meta_ = NULL;
+ status_ = OK;
+ cleanupBufferInProgress();
+ cleanupDecoder();
+}
+
+void AAH_RXPlayer::Substream::cleanupBufferInProgress() {
+ if (NULL != buffer_in_progress_) {
+ buffer_in_progress_->release();
+ buffer_in_progress_ = NULL;
+ }
+
+ expected_buffer_size_ = 0;
+ buffer_filled_ = 0;
+ waiting_for_rap_ = true;
+
+ aux_data_in_progress_.clear();
+ aux_data_expected_size_ = 0;
+}
+
+void AAH_RXPlayer::Substream::cleanupDecoder() {
+ if (decoder_ != NULL) {
+ decoder_->shutdown();
+ }
+}
+
+bool AAH_RXPlayer::Substream::shouldAbort(const char* log_tag) {
+ // If we have already encountered a fatal error, do nothing. We are just
+ // waiting for our owner to shut us down now.
+ if (OK != status_) {
+ ALOGV("Skipping %s, substream has encountered fatal error (%d).",
+ log_tag, status_);
+ return true;
+ }
+
+ return false;
+}
+
+void AAH_RXPlayer::Substream::processPayloadStart(uint8_t* buf,
+ uint32_t amt,
+ int32_t ts_lower) {
+ uint32_t min_length = 6;
+
+ if (shouldAbort(__PRETTY_FUNCTION__)) {
+ return;
+ }
+
+ // Do we have a buffer in progress already? If so, abort the buffer. In
+ // theory, this should never happen. If there were a discontinutity in the
+ // stream, the discon in the seq_nos at the RTP level should have already
+ // triggered a cleanup of the buffer in progress. To see a problem at this
+ // level is an indication either of a bug in the transmitter, or some form
+ // of terrible corruption/tampering on the wire.
+ if (NULL != buffer_in_progress_) {
+ ALOGE("processPayloadStart is aborting payload already in progress.");
+ cleanupBufferInProgress();
+ }
+
+ // Parse enough of the header to know where we stand. Since this is a
+ // payload start, it should begin with a TRTP header which has to be at
+ // least 6 bytes long.
+ if (amt < min_length) {
+ ALOGV("Discarding payload too short to contain TRTP header (len = %u)",
+ amt);
+ return;
+ }
+
+ // Check the TRTP version number.
+ if (0x01 != buf[0]) {
+ ALOGV("Unexpected TRTP version (%u) in header. Expected %u.",
+ buf[0], 1);
+ return;
+ }
+
+ // Extract the substream type field and make sure its one we understand (and
+ // one that does not conflict with any previously received substream type.
+ uint8_t header_type = (buf[1] >> 4) & 0xF;
+ switch (header_type) {
+ case TRTPPacket::kHeaderTypeAudio:
+ // Audio, yay! Just break. We understand audio payloads.
+ break;
+ case TRTPPacket::kHeaderTypeVideo:
+ ALOGV("RXed packet with unhandled TRTP header type (Video).");
+ return;
+ case TRTPPacket::kHeaderTypeSubpicture:
+ ALOGV("RXed packet with unhandled TRTP header type (Subpicture).");
+ return;
+ case TRTPPacket::kHeaderTypeControl:
+ ALOGV("RXed packet with unhandled TRTP header type (Control).");
+ return;
+ default:
+ ALOGV("RXed packet with unhandled TRTP header type (%u).",
+ header_type);
+ return;
+ }
+
+ if (substream_details_known_ && (header_type != substream_type_)) {
+ ALOGV("RXed TRTP Payload for SSRC=0x%08x where header type (%u) does"
+ " not match previously received header type (%u)",
+ ssrc_, header_type, substream_type_);
+ return;
+ }
+
+ // Check the flags to see if there is another 32 bits of timestamp present.
+ uint32_t trtp_header_len = 6;
+ bool ts_valid = buf[1] & TRTPPacket::kFlag_TSValid;
+ if (ts_valid) {
+ min_length += 4;
+ trtp_header_len += 4;
+ if (amt < min_length) {
+ ALOGV("Discarding payload too short to contain TRTP timestamp"
+ " (len = %u)", amt);
+ return;
+ }
+ }
+
+ // Extract the TRTP length field and sanity check it.
+ uint32_t trtp_len = U32_AT(buf + 2);
+ if (trtp_len < min_length) {
+ ALOGV("TRTP length (%u) is too short to be valid. Must be at least %u"
+ " bytes.", trtp_len, min_length);
+ return;
+ }
+
+ // Extract the rest of the timestamp field if valid.
+ int64_t ts = 0;
+ uint32_t parse_offset = 6;
+ if (ts_valid) {
+ uint32_t ts_upper = U32_AT(buf + parse_offset);
+ parse_offset += 4;
+ ts = (static_cast<int64_t>(ts_upper) << 32) | ts_lower;
+ }
+
+ // Check the flags to see if there is another 24 bytes of timestamp
+ // transformation present.
+ if (buf[1] & TRTPPacket::kFlag_TSTransformPresent) {
+ min_length += 24;
+ parse_offset += 24;
+ trtp_header_len += 24;
+ if (amt < min_length) {
+ ALOGV("Discarding payload too short to contain TRTP timestamp"
+ " transformation (len = %u)", amt);
+ return;
+ }
+ }
+
+ // TODO : break the parsing into individual parsers for the different
+ // payload types (audio, video, etc).
+ //
+ // At this point in time, we know that this is audio. Go ahead and parse
+ // the basic header, check the codec type, and find the payload portion of
+ // the packet.
+ min_length += 3;
+ if (trtp_len < min_length) {
+ ALOGV("TRTP length (%u) is too short to be a valid audio payload. Must"
+ " be at least %u bytes.", trtp_len, min_length);
+ return;
+ }
+
+ if (amt < min_length) {
+ ALOGV("TRTP porttion of RTP payload (%u bytes) too small to contain"
+ " entire TRTP header. TRTP does not currently support"
+ " fragmenting TRTP headers across RTP payloads", amt);
+ return;
+ }
+
+ uint8_t codec_type = buf[parse_offset ];
+ uint8_t flags = buf[parse_offset + 1];
+ uint8_t volume = buf[parse_offset + 2];
+ parse_offset += 3;
+ trtp_header_len += 3;
+
+ if (!setupSubstreamType(header_type, codec_type)) {
+ return;
+ }
+
+ if (decoder_ != NULL) {
+ decoder_->setRenderVolume(volume);
+ }
+
+ if (waiting_for_rap_ && !(flags & TRTPAudioPacket::kFlag_RandomAccessPoint)) {
+ ALOGV("Dropping non-RAP TRTP Audio Payload while waiting for RAP.");
+ return;
+ }
+
+ // Check for the presence of codec aux data.
+ if (flags & TRTPAudioPacket::kFlag_AuxLengthPresent) {
+ min_length += 4;
+ trtp_header_len += 4;
+
+ if (trtp_len < min_length) {
+ ALOGV("TRTP length (%u) is too short to be a valid audio payload. "
+ "Must be at least %u bytes.", trtp_len, min_length);
+ return;
+ }
+
+ if (amt < min_length) {
+ ALOGV("TRTP porttion of RTP payload (%u bytes) too small to contain"
+ " entire TRTP header. TRTP does not currently support"
+ " fragmenting TRTP headers across RTP payloads", amt);
+ return;
+ }
+
+ aux_data_expected_size_ = U32_AT(buf + parse_offset);
+ aux_data_in_progress_.clear();
+ if (aux_data_in_progress_.capacity() < aux_data_expected_size_) {
+ aux_data_in_progress_.setCapacity(aux_data_expected_size_);
+ }
+ } else {
+ aux_data_expected_size_ = 0;
+ }
+
+ if ((aux_data_expected_size_ + trtp_header_len) > trtp_len) {
+ ALOGV("Expected codec aux data length (%u) and TRTP header overhead"
+ " (%u) too large for total TRTP payload length (%u).",
+ aux_data_expected_size_, trtp_header_len, trtp_len);
+ return;
+ }
+
+ // OK - everything left is just payload. Compute the payload size, start
+ // the buffer in progress and pack as much payload as we can into it. If
+ // the payload is finished once we are done, go ahead and send the payload
+ // to the decoder.
+ expected_buffer_size_ = trtp_len
+ - trtp_header_len
+ - aux_data_expected_size_;
+ if (!expected_buffer_size_) {
+ ALOGV("Dropping TRTP Audio Payload with 0 Access Unit length");
+ return;
+ }
+
+ CHECK(amt >= trtp_header_len);
+ uint32_t todo = amt - trtp_header_len;
+ if ((expected_buffer_size_ + aux_data_expected_size_) < todo) {
+ ALOGV("Extra data (%u > %u) present in initial TRTP Audio Payload;"
+ " dropping payload.", todo,
+ expected_buffer_size_ + aux_data_expected_size_);
+ return;
+ }
+
+ buffer_filled_ = 0;
+ buffer_in_progress_ = new MediaBuffer(expected_buffer_size_);
+ if ((NULL == buffer_in_progress_) ||
+ (NULL == buffer_in_progress_->data())) {
+ ALOGV("Failed to allocate MediaBuffer of length %u",
+ expected_buffer_size_);
+ cleanupBufferInProgress();
+ return;
+ }
+
+ sp<MetaData> meta = buffer_in_progress_->meta_data();
+ if (meta == NULL) {
+ ALOGV("Missing metadata structure in allocated MediaBuffer; dropping"
+ " payload");
+ cleanupBufferInProgress();
+ return;
+ }
+
+ meta->setCString(kKeyMIMEType, codec_mime_type_);
+ if (ts_valid) {
+ meta->setInt64(kKeyTime, ts);
+ }
+
+ // Skip over the header we have already extracted.
+ amt -= trtp_header_len;
+ buf += trtp_header_len;
+
+ // Extract as much of the expected aux data as we can.
+ todo = min(aux_data_expected_size_, amt);
+ if (todo) {
+ aux_data_in_progress_.appendArray(buf, todo);
+ buf += todo;
+ amt -= todo;
+ }
+
+ // Extract as much of the expected payload as we can.
+ todo = min(expected_buffer_size_, amt);
+ if (todo > 0) {
+ uint8_t* tgt =
+ reinterpret_cast<uint8_t*>(buffer_in_progress_->data());
+ memcpy(tgt, buf, todo);
+ buffer_filled_ = amt;
+ buf += todo;
+ amt -= todo;
+ }
+
+ if (buffer_filled_ >= expected_buffer_size_) {
+ processCompletedBuffer();
+ }
+}
+
+void AAH_RXPlayer::Substream::processPayloadCont(uint8_t* buf,
+ uint32_t amt) {
+ if (shouldAbort(__PRETTY_FUNCTION__)) {
+ return;
+ }
+
+ if (NULL == buffer_in_progress_) {
+ ALOGV("TRTP Receiver skipping payload continuation; no buffer currently"
+ " in progress.");
+ return;
+ }
+
+ CHECK(aux_data_in_progress_.size() <= aux_data_expected_size_);
+ uint32_t aux_left = aux_data_expected_size_ - aux_data_in_progress_.size();
+ if (aux_left) {
+ uint32_t todo = min(aux_left, amt);
+ aux_data_in_progress_.appendArray(buf, todo);
+ amt -= todo;
+ buf += todo;
+
+ if (!amt)
+ return;
+ }
+
+ CHECK(buffer_filled_ < expected_buffer_size_);
+ uint32_t buffer_left = expected_buffer_size_ - buffer_filled_;
+ if (amt > buffer_left) {
+ ALOGV("Extra data (%u > %u) present in continued TRTP Audio Payload;"
+ " dropping payload.", amt, buffer_left);
+ cleanupBufferInProgress();
+ return;
+ }
+
+ if (amt > 0) {
+ uint8_t* tgt =
+ reinterpret_cast<uint8_t*>(buffer_in_progress_->data());
+ memcpy(tgt + buffer_filled_, buf, amt);
+ buffer_filled_ += amt;
+ }
+
+ if (buffer_filled_ >= expected_buffer_size_) {
+ processCompletedBuffer();
+ }
+}
+
+void AAH_RXPlayer::Substream::processCompletedBuffer() {
+ status_t res;
+
+ CHECK(NULL != buffer_in_progress_);
+
+ if (decoder_ == NULL) {
+ ALOGV("Dropping complete buffer, no decoder pump allocated");
+ goto bailout;
+ }
+
+ // Make sure our metadata used to initialize the decoder has been properly
+ // set up.
+ if (!setupSubstreamMeta())
+ goto bailout;
+
+ // If our decoder has not be set up, do so now.
+ res = decoder_->init(substream_meta_);
+ if (OK != res) {
+ ALOGE("Failed to init decoder (res = %d)", res);
+ cleanupDecoder();
+ substream_meta_ = NULL;
+ goto bailout;
+ }
+
+ // Queue the payload for decode.
+ res = decoder_->queueForDecode(buffer_in_progress_);
+
+ if (res != OK) {
+ ALOGD("Failed to queue payload for decode, resetting decoder pump!"
+ " (res = %d)", res);
+ status_ = res;
+ cleanupDecoder();
+ cleanupBufferInProgress();
+ }
+
+ // NULL out buffer_in_progress before calling the cleanup helper.
+ //
+ // MediaBuffers use something of a hybrid ref-counting pattern which prevent
+ // the AAH_DecoderPump's input queue from adding their own reference to the
+ // MediaBuffer. MediaBuffers start life with a reference count of 0, as
+ // well as an observer which starts as NULL. Before being given an
+ // observer, the ref count cannot be allowed to become non-zero as it will
+ // cause calls to release() to assert. Basically, before a MediaBuffer has
+ // an observer, they behave like non-ref counted obects where release()
+ // serves the roll of delete. After a MediaBuffer has an observer, they
+ // become more like ref counted objects where add ref and release can be
+ // used, and when the ref count hits zero, the MediaBuffer is handed off to
+ // the observer.
+ //
+ // Given all of this, when we give the buffer to the decoder pump to wait in
+ // the to-be-processed queue, the decoder cannot add a ref to the buffer as
+ // it would in a traditional ref counting system. Instead it needs to
+ // "steal" the non-existent ref. In the case of queue failure, we need to
+ // make certain to release this non-existent reference so that the buffer is
+ // cleaned up during the cleanupBufferInProgress helper. In the case of a
+ // successful queue operation, we need to make certain that the
+ // cleanupBufferInProgress helper does not release the buffer since it needs
+ // to remain alive in the queue. We acomplish this by NULLing out the
+ // buffer pointer before calling the cleanup helper.
+ buffer_in_progress_ = NULL;
+
+bailout:
+ cleanupBufferInProgress();
+}
+
+bool AAH_RXPlayer::Substream::setupSubstreamMeta() {
+ switch (codec_type_) {
+ case TRTPAudioPacket::kCodecMPEG1Audio:
+ codec_mime_type_ = MEDIA_MIMETYPE_AUDIO_MPEG;
+ return setupMP3SubstreamMeta();
+
+ case TRTPAudioPacket::kCodecAACAudio:
+ codec_mime_type_ = MEDIA_MIMETYPE_AUDIO_AAC;
+ return setupAACSubstreamMeta();
+
+ default:
+ ALOGV("Failed to setup substream metadata for unsupported codec"
+ " type (%u)", codec_type_);
+ break;
+ }
+
+ return false;
+}
+
+bool AAH_RXPlayer::Substream::setupMP3SubstreamMeta() {
+ const uint8_t* buffer_data = NULL;
+ int sample_rate;
+ int channel_count;
+ size_t frame_size;
+ status_t res;
+
+ buffer_data = reinterpret_cast<const uint8_t*>(buffer_in_progress_->data());
+ if (buffer_in_progress_->size() < 4) {
+ ALOGV("MP3 payload too short to contain header, dropping payload.");
+ return false;
+ }
+
+ // Extract the channel count and the sample rate from the MP3 header. The
+ // stagefright MP3 requires that these be delivered before decoing can
+ // begin.
+ if (!GetMPEGAudioFrameSize(U32_AT(buffer_data),
+ &frame_size,
+ &sample_rate,
+ &channel_count,
+ NULL,
+ NULL)) {
+ ALOGV("Failed to parse MP3 header in payload, droping payload.");
+ return false;
+ }
+
+
+ // Make sure that our substream metadata is set up properly. If there has
+ // been a format change, be sure to reset the underlying decoder. In
+ // stagefright, it seems like the only way to do this is to destroy and
+ // recreate the decoder.
+ if (substream_meta_ == NULL) {
+ substream_meta_ = new MetaData();
+
+ if (substream_meta_ == NULL) {
+ ALOGE("Failed to allocate MetaData structure for MP3 substream");
+ return false;
+ }
+
+ substream_meta_->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_MPEG);
+ substream_meta_->setInt32 (kKeyChannelCount, channel_count);
+ substream_meta_->setInt32 (kKeySampleRate, sample_rate);
+ } else {
+ int32_t prev_sample_rate;
+ int32_t prev_channel_count;
+ substream_meta_->findInt32(kKeySampleRate, &prev_sample_rate);
+ substream_meta_->findInt32(kKeyChannelCount, &prev_channel_count);
+
+ if ((prev_channel_count != channel_count) ||
+ (prev_sample_rate != sample_rate)) {
+ ALOGW("MP3 format change detected, forcing decoder reset.");
+ cleanupDecoder();
+
+ substream_meta_->setInt32(kKeyChannelCount, channel_count);
+ substream_meta_->setInt32(kKeySampleRate, sample_rate);
+ }
+ }
+
+ return true;
+}
+
+bool AAH_RXPlayer::Substream::setupAACSubstreamMeta() {
+ int32_t sample_rate, channel_cnt;
+ static const size_t overhead = sizeof(sample_rate)
+ + sizeof(channel_cnt);
+
+ if (aux_data_in_progress_.size() < overhead) {
+ ALOGE("Not enough aux data (%u) to initialize AAC substream decoder",
+ aux_data_in_progress_.size());
+ return false;
+ }
+
+ const uint8_t* aux_data = aux_data_in_progress_.array();
+ size_t aux_data_size = aux_data_in_progress_.size();
+ sample_rate = U32_AT(aux_data);
+ channel_cnt = U32_AT(aux_data + sizeof(sample_rate));
+
+ const uint8_t* esds_data = NULL;
+ size_t esds_data_size = 0;
+ if (aux_data_size > overhead) {
+ esds_data = aux_data + overhead;
+ esds_data_size = aux_data_size - overhead;
+ }
+
+ // Do we already have metadata? If so, has it changed at all? If not, then
+ // there should be nothing else to do. Otherwise, release our old stream
+ // metadata and make new metadata.
+ if (substream_meta_ != NULL) {
+ uint32_t type;
+ const void* data;
+ size_t size;
+ int32_t prev_sample_rate;
+ int32_t prev_channel_count;
+ bool res;
+
+ res = substream_meta_->findInt32(kKeySampleRate, &prev_sample_rate);
+ CHECK(res);
+ res = substream_meta_->findInt32(kKeyChannelCount, &prev_channel_count);
+ CHECK(res);
+
+ // If nothing has changed about the codec aux data (esds, sample rate,
+ // channel count), then we can just do nothing and get out. Otherwise,
+ // we will need to reset the decoder and make a new metadata object to
+ // deal with the format change.
+ bool hasData = (esds_data != NULL);
+ bool hadData = substream_meta_->findData(kKeyESDS, &type, &data, &size);
+ bool esds_change = (hadData != hasData);
+
+ if (!esds_change && hasData)
+ esds_change = ((size != esds_data_size) ||
+ memcmp(data, esds_data, size));
+
+ if (!esds_change &&
+ (prev_sample_rate == sample_rate) &&
+ (prev_channel_count == channel_cnt)) {
+ return true; // no change, just get out.
+ }
+
+ ALOGW("AAC format change detected, forcing decoder reset.");
+ cleanupDecoder();
+ substream_meta_ = NULL;
+ }
+
+ CHECK(substream_meta_ == NULL);
+
+ substream_meta_ = new MetaData();
+ if (substream_meta_ == NULL) {
+ ALOGE("Failed to allocate MetaData structure for AAC substream");
+ return false;
+ }
+
+ substream_meta_->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_AAC);
+ substream_meta_->setInt32 (kKeySampleRate, sample_rate);
+ substream_meta_->setInt32 (kKeyChannelCount, channel_cnt);
+
+ if (esds_data) {
+ substream_meta_->setData(kKeyESDS, kTypeESDS,
+ esds_data, esds_data_size);
+ }
+
+ return true;
+}
+
+void AAH_RXPlayer::Substream::processTSTransform(const LinearTransform& trans) {
+ if (decoder_ != NULL) {
+ decoder_->setRenderTSTransform(trans);
+ }
+}
+
+bool AAH_RXPlayer::Substream::isAboutToUnderflow() {
+ if (decoder_ == NULL) {
+ return false;
+ }
+
+ return decoder_->isAboutToUnderflow(kAboutToUnderflowThreshold);
+}
+
+bool AAH_RXPlayer::Substream::setupSubstreamType(uint8_t substream_type,
+ uint8_t codec_type) {
+ // Sanity check the codec type. Right now we only support MP3 and AAC.
+ // Also check for conflicts with previously delivered codec types.
+ if (substream_details_known_) {
+ if (codec_type != codec_type_) {
+ ALOGV("RXed TRTP Payload for SSRC=0x%08x where codec type (%u) does"
+ " not match previously received codec type (%u)",
+ ssrc_, codec_type, codec_type_);
+ return false;
+ }
+
+ return true;
+ }
+
+ switch (codec_type) {
+ // MP3 and AAC are all we support right now.
+ case TRTPAudioPacket::kCodecMPEG1Audio:
+ case TRTPAudioPacket::kCodecAACAudio:
+ break;
+
+ default:
+ ALOGV("RXed TRTP Audio Payload for SSRC=0x%08x with unsupported"
+ " codec type (%u)", ssrc_, codec_type);
+ return false;
+ }
+
+ substream_type_ = substream_type;
+ codec_type_ = codec_type;
+ substream_details_known_ = true;
+
+ return true;
+}
+
+} // namespace android
diff --git a/media/libaah_rtp/aah_tx_packet.cpp b/media/libaah_rtp/aah_tx_packet.cpp
new file mode 100644
index 0000000..4cd6e47
--- /dev/null
+++ b/media/libaah_rtp/aah_tx_packet.cpp
@@ -0,0 +1,344 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "LibAAH_RTP"
+#include <utils/Log.h>
+
+#include <arpa/inet.h>
+#include <string.h>
+
+#include <media/stagefright/foundation/ADebug.h>
+
+#include "aah_tx_packet.h"
+
+namespace android {
+
+const int TRTPPacket::kRTPHeaderLen;
+const uint32_t TRTPPacket::kTRTPEpochMask;
+
+TRTPPacket::~TRTPPacket() {
+ delete mPacket;
+}
+
+/*** TRTP packet properties ***/
+
+void TRTPPacket::setSeqNumber(uint16_t val) {
+ mSeqNumber = val;
+
+ if (mIsPacked) {
+ const int kTRTPSeqNumberOffset = 2;
+ uint16_t* buf = reinterpret_cast<uint16_t*>(
+ mPacket + kTRTPSeqNumberOffset);
+ *buf = htons(mSeqNumber);
+ }
+}
+
+uint16_t TRTPPacket::getSeqNumber() const {
+ return mSeqNumber;
+}
+
+void TRTPPacket::setPTS(int64_t val) {
+ CHECK(!mIsPacked);
+ mPTS = val;
+ mPTSValid = true;
+}
+
+int64_t TRTPPacket::getPTS() const {
+ return mPTS;
+}
+
+void TRTPPacket::setEpoch(uint32_t val) {
+ mEpoch = val;
+
+ if (mIsPacked) {
+ const int kTRTPEpochOffset = 8;
+ uint32_t* buf = reinterpret_cast<uint32_t*>(
+ mPacket + kTRTPEpochOffset);
+ uint32_t val = ntohl(*buf);
+ val &= ~(kTRTPEpochMask << kTRTPEpochShift);
+ val |= (mEpoch & kTRTPEpochMask) << kTRTPEpochShift;
+ *buf = htonl(val);
+ }
+}
+
+void TRTPPacket::setProgramID(uint16_t val) {
+ CHECK(!mIsPacked);
+ mProgramID = val;
+}
+
+void TRTPPacket::setSubstreamID(uint16_t val) {
+ CHECK(!mIsPacked);
+ mSubstreamID = val;
+}
+
+
+void TRTPPacket::setClockTransform(const LinearTransform& trans) {
+ CHECK(!mIsPacked);
+ mClockTranform = trans;
+ mClockTranformValid = true;
+}
+
+uint8_t* TRTPPacket::getPacket() const {
+ CHECK(mIsPacked);
+ return mPacket;
+}
+
+int TRTPPacket::getPacketLen() const {
+ CHECK(mIsPacked);
+ return mPacketLen;
+}
+
+void TRTPPacket::setExpireTime(nsecs_t val) {
+ CHECK(!mIsPacked);
+ mExpireTime = val;
+}
+
+nsecs_t TRTPPacket::getExpireTime() const {
+ return mExpireTime;
+}
+
+/*** TRTP audio packet properties ***/
+
+void TRTPAudioPacket::setCodecType(TRTPAudioCodecType val) {
+ CHECK(!mIsPacked);
+ mCodecType = val;
+}
+
+void TRTPAudioPacket::setRandomAccessPoint(bool val) {
+ CHECK(!mIsPacked);
+ mRandomAccessPoint = val;
+}
+
+void TRTPAudioPacket::setDropable(bool val) {
+ CHECK(!mIsPacked);
+ mDropable = val;
+}
+
+void TRTPAudioPacket::setDiscontinuity(bool val) {
+ CHECK(!mIsPacked);
+ mDiscontinuity = val;
+}
+
+void TRTPAudioPacket::setEndOfStream(bool val) {
+ CHECK(!mIsPacked);
+ mEndOfStream = val;
+}
+
+void TRTPAudioPacket::setVolume(uint8_t val) {
+ CHECK(!mIsPacked);
+ mVolume = val;
+}
+
+void TRTPAudioPacket::setAccessUnitData(const void* data, size_t len) {
+ CHECK(!mIsPacked);
+ mAccessUnitData = data;
+ mAccessUnitLen = len;
+}
+
+void TRTPAudioPacket::setAuxData(const void* data, size_t len) {
+ CHECK(!mIsPacked);
+ mAuxData = data;
+ mAuxDataLen = len;
+}
+
+/*** TRTP control packet properties ***/
+
+void TRTPControlPacket::setCommandID(TRTPCommandID val) {
+ CHECK(!mIsPacked);
+ mCommandID = val;
+}
+
+/*** TRTP packet serializers ***/
+
+void TRTPPacket::writeU8(uint8_t*& buf, uint8_t val) {
+ *buf = val;
+ buf++;
+}
+
+void TRTPPacket::writeU16(uint8_t*& buf, uint16_t val) {
+ *reinterpret_cast<uint16_t*>(buf) = htons(val);
+ buf += 2;
+}
+
+void TRTPPacket::writeU32(uint8_t*& buf, uint32_t val) {
+ *reinterpret_cast<uint32_t*>(buf) = htonl(val);
+ buf += 4;
+}
+
+void TRTPPacket::writeU64(uint8_t*& buf, uint64_t val) {
+ buf[0] = static_cast<uint8_t>(val >> 56);
+ buf[1] = static_cast<uint8_t>(val >> 48);
+ buf[2] = static_cast<uint8_t>(val >> 40);
+ buf[3] = static_cast<uint8_t>(val >> 32);
+ buf[4] = static_cast<uint8_t>(val >> 24);
+ buf[5] = static_cast<uint8_t>(val >> 16);
+ buf[6] = static_cast<uint8_t>(val >> 8);
+ buf[7] = static_cast<uint8_t>(val);
+ buf += 8;
+}
+
+void TRTPPacket::writeTRTPHeader(uint8_t*& buf,
+ bool isFirstFragment,
+ int totalPacketLen) {
+ // RTP header
+ writeU8(buf,
+ ((mVersion & 0x03) << 6) |
+ (static_cast<int>(mPadding) << 5) |
+ (static_cast<int>(mExtension) << 4) |
+ (mCsrcCount & 0x0F));
+ writeU8(buf,
+ (static_cast<int>(isFirstFragment) << 7) |
+ (mPayloadType & 0x7F));
+ writeU16(buf, mSeqNumber);
+ if (isFirstFragment && mPTSValid) {
+ writeU32(buf, mPTS & 0xFFFFFFFF);
+ } else {
+ writeU32(buf, 0);
+ }
+ writeU32(buf,
+ ((mEpoch & kTRTPEpochMask) << kTRTPEpochShift) |
+ ((mProgramID & 0x1F) << 5) |
+ (mSubstreamID & 0x1F));
+
+ // TRTP header
+ writeU8(buf, mTRTPVersion);
+ writeU8(buf,
+ ((mTRTPHeaderType & 0x0F) << 4) |
+ (mClockTranformValid ? 0x02 : 0x00) |
+ (mPTSValid ? 0x01 : 0x00));
+ writeU32(buf, totalPacketLen - kRTPHeaderLen);
+ if (mPTSValid) {
+ writeU32(buf, mPTS >> 32);
+ }
+
+ if (mClockTranformValid) {
+ writeU64(buf, mClockTranform.a_zero);
+ writeU32(buf, mClockTranform.a_to_b_numer);
+ writeU32(buf, mClockTranform.a_to_b_denom);
+ writeU64(buf, mClockTranform.b_zero);
+ }
+}
+
+bool TRTPAudioPacket::pack() {
+ if (mIsPacked) {
+ return false;
+ }
+
+ int packetLen = kRTPHeaderLen +
+ mAuxDataLen +
+ mAccessUnitLen +
+ TRTPHeaderLen();
+
+ // TODO : support multiple fragments
+ const int kMaxUDPPayloadLen = 65507;
+ if (packetLen > kMaxUDPPayloadLen) {
+ return false;
+ }
+
+ mPacket = new uint8_t[packetLen];
+ if (!mPacket) {
+ return false;
+ }
+
+ mPacketLen = packetLen;
+
+ uint8_t* cur = mPacket;
+ bool hasAux = mAuxData && mAuxDataLen;
+ uint8_t flags = (static_cast<int>(hasAux) << 4) |
+ (static_cast<int>(mRandomAccessPoint) << 3) |
+ (static_cast<int>(mDropable) << 2) |
+ (static_cast<int>(mDiscontinuity) << 1) |
+ (static_cast<int>(mEndOfStream));
+
+ writeTRTPHeader(cur, true, packetLen);
+ writeU8(cur, mCodecType);
+ writeU8(cur, flags);
+ writeU8(cur, mVolume);
+
+ if (hasAux) {
+ writeU32(cur, mAuxDataLen);
+ memcpy(cur, mAuxData, mAuxDataLen);
+ cur += mAuxDataLen;
+ }
+
+ memcpy(cur, mAccessUnitData, mAccessUnitLen);
+
+ mIsPacked = true;
+ return true;
+}
+
+int TRTPPacket::TRTPHeaderLen() const {
+ // 6 bytes for version, payload type, flags and length. An additional 4 if
+ // there are upper timestamp bits present and another 24 if there is a clock
+ // transformation present.
+ return 6 +
+ (mClockTranformValid ? 24 : 0) +
+ (mPTSValid ? 4 : 0);
+}
+
+int TRTPAudioPacket::TRTPHeaderLen() const {
+ // TRTPPacket::TRTPHeaderLen() for the base TRTPHeader. 3 bytes for audio's
+ // codec type, flags and volume field. Another 5 bytes if the codec type is
+ // PCM and we are sending sample rate/channel count. as well as however long
+ // the aux data (if present) is.
+
+ int pcmParamLength;
+ switch(mCodecType) {
+ case kCodecPCMBigEndian:
+ case kCodecPCMLittleEndian:
+ pcmParamLength = 5;
+ break;
+
+ default:
+ pcmParamLength = 0;
+ break;
+ }
+
+
+ int auxDataLenField = (NULL != mAuxData) ? sizeof(uint32_t) : 0;
+ return TRTPPacket::TRTPHeaderLen() +
+ 3 +
+ auxDataLenField +
+ pcmParamLength;
+}
+
+bool TRTPControlPacket::pack() {
+ if (mIsPacked) {
+ return false;
+ }
+
+ // command packets contain a 2-byte command ID
+ int packetLen = kRTPHeaderLen +
+ TRTPHeaderLen() +
+ 2;
+
+ mPacket = new uint8_t[packetLen];
+ if (!mPacket) {
+ return false;
+ }
+
+ mPacketLen = packetLen;
+
+ uint8_t* cur = mPacket;
+
+ writeTRTPHeader(cur, true, packetLen);
+ writeU16(cur, mCommandID);
+
+ mIsPacked = true;
+ return true;
+}
+
+} // namespace android
diff --git a/media/libaah_rtp/aah_tx_packet.h b/media/libaah_rtp/aah_tx_packet.h
new file mode 100644
index 0000000..7f78ea0
--- /dev/null
+++ b/media/libaah_rtp/aah_tx_packet.h
@@ -0,0 +1,213 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __AAH_TX_PACKET_H__
+#define __AAH_TX_PACKET_H__
+
+#include <media/stagefright/foundation/ABase.h>
+#include <utils/LinearTransform.h>
+#include <utils/RefBase.h>
+#include <utils/Timers.h>
+
+namespace android {
+
+class TRTPPacket : public RefBase {
+ public:
+ enum TRTPHeaderType {
+ kHeaderTypeAudio = 1,
+ kHeaderTypeVideo = 2,
+ kHeaderTypeSubpicture = 3,
+ kHeaderTypeControl = 4,
+ };
+
+ enum TRTPPayloadFlags {
+ kFlag_TSTransformPresent = 0x02,
+ kFlag_TSValid = 0x01,
+ };
+
+ protected:
+ TRTPPacket(TRTPHeaderType headerType)
+ : mIsPacked(false)
+ , mVersion(2)
+ , mPadding(false)
+ , mExtension(false)
+ , mCsrcCount(0)
+ , mPayloadType(100)
+ , mSeqNumber(0)
+ , mPTSValid(false)
+ , mPTS(0)
+ , mEpoch(0)
+ , mProgramID(0)
+ , mSubstreamID(0)
+ , mClockTranformValid(false)
+ , mTRTPVersion(1)
+ , mTRTPLength(0)
+ , mTRTPHeaderType(headerType)
+ , mPacket(NULL)
+ , mPacketLen(0) { }
+
+ public:
+ virtual ~TRTPPacket();
+
+ void setSeqNumber(uint16_t val);
+ uint16_t getSeqNumber() const;
+
+ void setPTS(int64_t val);
+ int64_t getPTS() const;
+
+ void setEpoch(uint32_t val);
+ void setProgramID(uint16_t val);
+ void setSubstreamID(uint16_t val);
+ void setClockTransform(const LinearTransform& trans);
+
+ uint8_t* getPacket() const;
+ int getPacketLen() const;
+
+ void setExpireTime(nsecs_t val);
+ nsecs_t getExpireTime() const;
+
+ virtual bool pack() = 0;
+
+ // mask for the number of bits in a TRTP epoch
+ static const uint32_t kTRTPEpochMask = (1 << 22) - 1;
+ static const int kTRTPEpochShift = 10;
+
+ protected:
+ static const int kRTPHeaderLen = 12;
+ virtual int TRTPHeaderLen() const;
+
+ void writeTRTPHeader(uint8_t*& buf,
+ bool isFirstFragment,
+ int totalPacketLen);
+
+ void writeU8(uint8_t*& buf, uint8_t val);
+ void writeU16(uint8_t*& buf, uint16_t val);
+ void writeU32(uint8_t*& buf, uint32_t val);
+ void writeU64(uint8_t*& buf, uint64_t val);
+
+ bool mIsPacked;
+
+ uint8_t mVersion;
+ bool mPadding;
+ bool mExtension;
+ uint8_t mCsrcCount;
+ uint8_t mPayloadType;
+ uint16_t mSeqNumber;
+ bool mPTSValid;
+ int64_t mPTS;
+ uint32_t mEpoch;
+ uint16_t mProgramID;
+ uint16_t mSubstreamID;
+ LinearTransform mClockTranform;
+ bool mClockTranformValid;
+ uint8_t mTRTPVersion;
+ uint32_t mTRTPLength;
+ TRTPHeaderType mTRTPHeaderType;
+
+ uint8_t* mPacket;
+ int mPacketLen;
+
+ nsecs_t mExpireTime;
+
+ DISALLOW_EVIL_CONSTRUCTORS(TRTPPacket);
+};
+
+class TRTPAudioPacket : public TRTPPacket {
+ public:
+ enum AudioPayloadFlags {
+ kFlag_AuxLengthPresent = 0x10,
+ kFlag_RandomAccessPoint = 0x08,
+ kFlag_Dropable = 0x04,
+ kFlag_Discontinuity = 0x02,
+ kFlag_EndOfStream = 0x01,
+ };
+
+ TRTPAudioPacket()
+ : TRTPPacket(kHeaderTypeAudio)
+ , mCodecType(kCodecInvalid)
+ , mRandomAccessPoint(false)
+ , mDropable(false)
+ , mDiscontinuity(false)
+ , mEndOfStream(false)
+ , mVolume(0)
+ , mAccessUnitData(NULL)
+ , mAccessUnitLen(0)
+ , mAuxData(NULL)
+ , mAuxDataLen(0) { }
+
+ enum TRTPAudioCodecType {
+ kCodecInvalid = 0,
+ kCodecPCMBigEndian = 1,
+ kCodecPCMLittleEndian = 2,
+ kCodecMPEG1Audio = 3,
+ kCodecAACAudio = 4,
+ };
+
+ void setCodecType(TRTPAudioCodecType val);
+ void setRandomAccessPoint(bool val);
+ void setDropable(bool val);
+ void setDiscontinuity(bool val);
+ void setEndOfStream(bool val);
+ void setVolume(uint8_t val);
+ void setAccessUnitData(const void* data, size_t len);
+ void setAuxData(const void* data, size_t len);
+
+ virtual bool pack();
+
+ protected:
+ virtual int TRTPHeaderLen() const;
+
+ private:
+ TRTPAudioCodecType mCodecType;
+ bool mRandomAccessPoint;
+ bool mDropable;
+ bool mDiscontinuity;
+ bool mEndOfStream;
+ uint8_t mVolume;
+
+ const void* mAccessUnitData;
+ size_t mAccessUnitLen;
+ const void* mAuxData;
+ size_t mAuxDataLen;
+
+ DISALLOW_EVIL_CONSTRUCTORS(TRTPAudioPacket);
+};
+
+class TRTPControlPacket : public TRTPPacket {
+ public:
+ TRTPControlPacket()
+ : TRTPPacket(kHeaderTypeControl)
+ , mCommandID(kCommandNop) {}
+
+ enum TRTPCommandID {
+ kCommandNop = 1,
+ kCommandFlush = 2,
+ kCommandEOS = 3,
+ };
+
+ void setCommandID(TRTPCommandID val);
+
+ virtual bool pack();
+
+ private:
+ TRTPCommandID mCommandID;
+
+ DISALLOW_EVIL_CONSTRUCTORS(TRTPControlPacket);
+};
+
+} // namespace android
+
+#endif // __AAH_TX_PLAYER_H__
diff --git a/media/libaah_rtp/aah_tx_player.cpp b/media/libaah_rtp/aah_tx_player.cpp
new file mode 100644
index 0000000..974805b
--- /dev/null
+++ b/media/libaah_rtp/aah_tx_player.cpp
@@ -0,0 +1,1177 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "LibAAH_RTP"
+#include <utils/Log.h>
+
+#define __STDC_FORMAT_MACROS
+#include <inttypes.h>
+#include <netdb.h>
+#include <netinet/ip.h>
+
+#include <common_time/cc_helper.h>
+#include <media/IMediaPlayer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/FileSource.h>
+#include <media/stagefright/MediaBuffer.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MetaData.h>
+#include <utils/Timers.h>
+
+#include "aah_tx_packet.h"
+#include "aah_tx_player.h"
+
+namespace android {
+
+static int64_t kLowWaterMarkUs = 2000000ll; // 2secs
+static int64_t kHighWaterMarkUs = 10000000ll; // 10secs
+static const size_t kLowWaterMarkBytes = 40000;
+static const size_t kHighWaterMarkBytes = 200000;
+
+// When we start up, how much lead time should we put on the first access unit?
+static const int64_t kAAHStartupLeadTimeUs = 300000LL;
+
+// How much time do we attempt to lead the clock by in steady state?
+static const int64_t kAAHBufferTimeUs = 1000000LL;
+
+// how long do we keep data in our retransmit buffer after sending it.
+const int64_t AAH_TXPlayer::kAAHRetryKeepAroundTimeNs =
+ kAAHBufferTimeUs * 1100;
+
+sp<MediaPlayerBase> createAAH_TXPlayer() {
+ sp<MediaPlayerBase> ret = new AAH_TXPlayer();
+ return ret;
+}
+
+template <typename T> static T clamp(T val, T min, T max) {
+ if (val < min) {
+ return min;
+ } else if (val > max) {
+ return max;
+ } else {
+ return val;
+ }
+}
+
+struct AAH_TXEvent : public TimedEventQueue::Event {
+ AAH_TXEvent(AAH_TXPlayer *player,
+ void (AAH_TXPlayer::*method)()) : mPlayer(player)
+ , mMethod(method) {}
+
+ protected:
+ virtual ~AAH_TXEvent() {}
+
+ virtual void fire(TimedEventQueue *queue, int64_t /* now_us */) {
+ (mPlayer->*mMethod)();
+ }
+
+ private:
+ AAH_TXPlayer *mPlayer;
+ void (AAH_TXPlayer::*mMethod)();
+
+ AAH_TXEvent(const AAH_TXEvent &);
+ AAH_TXEvent& operator=(const AAH_TXEvent &);
+};
+
+AAH_TXPlayer::AAH_TXPlayer()
+ : mQueueStarted(false)
+ , mFlags(0)
+ , mExtractorFlags(0) {
+ DataSource::RegisterDefaultSniffers();
+
+ mBufferingEvent = new AAH_TXEvent(this, &AAH_TXPlayer::onBufferingUpdate);
+ mBufferingEventPending = false;
+
+ mPumpAudioEvent = new AAH_TXEvent(this, &AAH_TXPlayer::onPumpAudio);
+ mPumpAudioEventPending = false;
+
+ mAudioCodecData = NULL;
+
+ reset_l();
+}
+
+AAH_TXPlayer::~AAH_TXPlayer() {
+ if (mQueueStarted) {
+ mQueue.stop();
+ }
+
+ reset_l();
+}
+
+void AAH_TXPlayer::cancelPlayerEvents(bool keepBufferingGoing) {
+ if (!keepBufferingGoing) {
+ mQueue.cancelEvent(mBufferingEvent->eventID());
+ mBufferingEventPending = false;
+
+ mQueue.cancelEvent(mPumpAudioEvent->eventID());
+ mPumpAudioEventPending = false;
+ }
+}
+
+status_t AAH_TXPlayer::initCheck() {
+ // Check for the presense of the common time service by attempting to query
+ // for CommonTime's frequency. If we get an error back, we cannot talk to
+ // the service at all and should abort now.
+ status_t res;
+ uint64_t freq;
+ res = mCCHelper.getCommonFreq(&freq);
+ if (OK != res) {
+ ALOGE("Failed to connect to common time service! (res %d)", res);
+ return res;
+ }
+
+ return OK;
+}
+
+status_t AAH_TXPlayer::setDataSource(
+ const char *url,
+ const KeyedVector<String8, String8> *headers) {
+ Mutex::Autolock autoLock(mLock);
+ return setDataSource_l(url, headers);
+}
+
+status_t AAH_TXPlayer::setDataSource_l(
+ const char *url,
+ const KeyedVector<String8, String8> *headers) {
+ reset_l();
+
+ mUri.setTo(url);
+
+ if (headers) {
+ mUriHeaders = *headers;
+
+ ssize_t index = mUriHeaders.indexOfKey(String8("x-hide-urls-from-log"));
+ if (index >= 0) {
+ // Browser is in "incognito" mode, suppress logging URLs.
+
+ // This isn't something that should be passed to the server.
+ mUriHeaders.removeItemsAt(index);
+
+ mFlags |= INCOGNITO;
+ }
+ }
+
+ // The URL may optionally contain a "#" character followed by a Skyjam
+ // cookie. Ideally the cookie header should just be passed in the headers
+ // argument, but the Java API for supplying headers is apparently not yet
+ // exposed in the SDK used by application developers.
+ const char kSkyjamCookieDelimiter = '#';
+ char* skyjamCookie = strrchr(mUri.string(), kSkyjamCookieDelimiter);
+ if (skyjamCookie) {
+ skyjamCookie++;
+ mUriHeaders.add(String8("Cookie"), String8(skyjamCookie));
+ mUri = String8(mUri.string(), skyjamCookie - mUri.string());
+ }
+
+ return OK;
+}
+
+status_t AAH_TXPlayer::setDataSource(int fd, int64_t offset, int64_t length) {
+ Mutex::Autolock autoLock(mLock);
+
+ reset_l();
+
+ sp<DataSource> dataSource = new FileSource(dup(fd), offset, length);
+
+ status_t err = dataSource->initCheck();
+
+ if (err != OK) {
+ return err;
+ }
+
+ mFileSource = dataSource;
+
+ sp<MediaExtractor> extractor = MediaExtractor::Create(dataSource);
+
+ if (extractor == NULL) {
+ return UNKNOWN_ERROR;
+ }
+
+ return setDataSource_l(extractor);
+}
+
+status_t AAH_TXPlayer::setVideoSurface(const sp<Surface>& surface) {
+ return OK;
+}
+
+status_t AAH_TXPlayer::setVideoSurfaceTexture(
+ const sp<ISurfaceTexture>& surfaceTexture) {
+ return OK;
+}
+
+status_t AAH_TXPlayer::prepare() {
+ return INVALID_OPERATION;
+}
+
+status_t AAH_TXPlayer::prepareAsync() {
+ Mutex::Autolock autoLock(mLock);
+
+ return prepareAsync_l();
+}
+
+status_t AAH_TXPlayer::prepareAsync_l() {
+ if (mFlags & PREPARING) {
+ return UNKNOWN_ERROR; // async prepare already pending
+ }
+
+ mAAH_Sender = AAH_TXSender::GetInstance();
+ if (mAAH_Sender == NULL) {
+ return NO_MEMORY;
+ }
+
+ if (!mQueueStarted) {
+ mQueue.start();
+ mQueueStarted = true;
+ }
+
+ mFlags |= PREPARING;
+ mAsyncPrepareEvent = new AAH_TXEvent(
+ this, &AAH_TXPlayer::onPrepareAsyncEvent);
+
+ mQueue.postEvent(mAsyncPrepareEvent);
+
+ return OK;
+}
+
+status_t AAH_TXPlayer::finishSetDataSource_l() {
+ sp<DataSource> dataSource;
+
+ if (!strncasecmp("http://", mUri.string(), 7) ||
+ !strncasecmp("https://", mUri.string(), 8)) {
+
+ mConnectingDataSource = HTTPBase::Create(
+ (mFlags & INCOGNITO)
+ ? HTTPBase::kFlagIncognito
+ : 0);
+
+ mLock.unlock();
+ status_t err = mConnectingDataSource->connect(mUri, &mUriHeaders);
+ mLock.lock();
+
+ if (err != OK) {
+ mConnectingDataSource.clear();
+
+ ALOGI("mConnectingDataSource->connect() returned %d", err);
+ return err;
+ }
+
+ mCachedSource = new NuCachedSource2(mConnectingDataSource);
+ mConnectingDataSource.clear();
+
+ dataSource = mCachedSource;
+
+ // We're going to prefill the cache before trying to instantiate
+ // the extractor below, as the latter is an operation that otherwise
+ // could block on the datasource for a significant amount of time.
+ // During that time we'd be unable to abort the preparation phase
+ // without this prefill.
+
+ mLock.unlock();
+
+ for (;;) {
+ status_t finalStatus;
+ size_t cachedDataRemaining =
+ mCachedSource->approxDataRemaining(&finalStatus);
+
+ if (finalStatus != OK ||
+ cachedDataRemaining >= kHighWaterMarkBytes ||
+ (mFlags & PREPARE_CANCELLED)) {
+ break;
+ }
+
+ usleep(200000);
+ }
+
+ mLock.lock();
+
+ if (mFlags & PREPARE_CANCELLED) {
+ ALOGI("Prepare cancelled while waiting for initial cache fill.");
+ return UNKNOWN_ERROR;
+ }
+ } else {
+ dataSource = DataSource::CreateFromURI(mUri.string(), &mUriHeaders);
+ }
+
+ if (dataSource == NULL) {
+ return UNKNOWN_ERROR;
+ }
+
+ sp<MediaExtractor> extractor = MediaExtractor::Create(dataSource);
+
+ if (extractor == NULL) {
+ return UNKNOWN_ERROR;
+ }
+
+ return setDataSource_l(extractor);
+}
+
+status_t AAH_TXPlayer::setDataSource_l(const sp<MediaExtractor> &extractor) {
+ // Attempt to approximate overall stream bitrate by summing all
+ // tracks' individual bitrates, if not all of them advertise bitrate,
+ // we have to fail.
+
+ int64_t totalBitRate = 0;
+
+ for (size_t i = 0; i < extractor->countTracks(); ++i) {
+ sp<MetaData> meta = extractor->getTrackMetaData(i);
+
+ int32_t bitrate;
+ if (!meta->findInt32(kKeyBitRate, &bitrate)) {
+ totalBitRate = -1;
+ break;
+ }
+
+ totalBitRate += bitrate;
+ }
+
+ mBitrate = totalBitRate;
+
+ ALOGV("mBitrate = %lld bits/sec", mBitrate);
+
+ bool haveAudio = false;
+ for (size_t i = 0; i < extractor->countTracks(); ++i) {
+ sp<MetaData> meta = extractor->getTrackMetaData(i);
+
+ const char *mime;
+ CHECK(meta->findCString(kKeyMIMEType, &mime));
+
+ if (!strncasecmp(mime, "audio/", 6)) {
+ mAudioSource = extractor->getTrack(i);
+ CHECK(mAudioSource != NULL);
+ haveAudio = true;
+ break;
+ }
+ }
+
+ if (!haveAudio) {
+ return UNKNOWN_ERROR;
+ }
+
+ mExtractorFlags = extractor->flags();
+
+ return OK;
+}
+
+void AAH_TXPlayer::abortPrepare(status_t err) {
+ CHECK(err != OK);
+
+ notifyListener_l(MEDIA_ERROR, MEDIA_ERROR_UNKNOWN, err);
+
+ mPrepareResult = err;
+ mFlags &= ~(PREPARING|PREPARE_CANCELLED|PREPARING_CONNECTED);
+ mPreparedCondition.broadcast();
+}
+
+void AAH_TXPlayer::onPrepareAsyncEvent() {
+ Mutex::Autolock autoLock(mLock);
+
+ if (mFlags & PREPARE_CANCELLED) {
+ ALOGI("prepare was cancelled before doing anything");
+ abortPrepare(UNKNOWN_ERROR);
+ return;
+ }
+
+ if (mUri.size() > 0) {
+ status_t err = finishSetDataSource_l();
+
+ if (err != OK) {
+ abortPrepare(err);
+ return;
+ }
+ }
+
+ mAudioFormat = mAudioSource->getFormat();
+ if (!mAudioFormat->findInt64(kKeyDuration, &mDurationUs))
+ mDurationUs = 1;
+
+ const char* mime_type = NULL;
+ if (!mAudioFormat->findCString(kKeyMIMEType, &mime_type)) {
+ ALOGE("Failed to find audio substream MIME type during prepare.");
+ abortPrepare(BAD_VALUE);
+ return;
+ }
+
+ if (!strcmp(mime_type, MEDIA_MIMETYPE_AUDIO_MPEG)) {
+ mAudioCodec = TRTPAudioPacket::kCodecMPEG1Audio;
+ } else
+ if (!strcmp(mime_type, MEDIA_MIMETYPE_AUDIO_AAC)) {
+ mAudioCodec = TRTPAudioPacket::kCodecAACAudio;
+
+ uint32_t type;
+ int32_t sample_rate;
+ int32_t channel_count;
+ const void* esds_data;
+ size_t esds_len;
+
+ if (!mAudioFormat->findInt32(kKeySampleRate, &sample_rate)) {
+ ALOGE("Failed to find sample rate for AAC substream.");
+ abortPrepare(BAD_VALUE);
+ return;
+ }
+
+ if (!mAudioFormat->findInt32(kKeyChannelCount, &channel_count)) {
+ ALOGE("Failed to find channel count for AAC substream.");
+ abortPrepare(BAD_VALUE);
+ return;
+ }
+
+ if (!mAudioFormat->findData(kKeyESDS, &type, &esds_data, &esds_len)) {
+ ALOGE("Failed to find codec init data for AAC substream.");
+ abortPrepare(BAD_VALUE);
+ return;
+ }
+
+ CHECK(NULL == mAudioCodecData);
+ mAudioCodecDataSize = esds_len
+ + sizeof(sample_rate)
+ + sizeof(channel_count);
+ mAudioCodecData = new uint8_t[mAudioCodecDataSize];
+ if (NULL == mAudioCodecData) {
+ ALOGE("Failed to allocate %u bytes for AAC substream codec aux"
+ " data.", mAudioCodecDataSize);
+ mAudioCodecDataSize = 0;
+ abortPrepare(BAD_VALUE);
+ return;
+ }
+
+ uint8_t* tmp = mAudioCodecData;
+ tmp[0] = static_cast<uint8_t>((sample_rate >> 24) & 0xFF);
+ tmp[1] = static_cast<uint8_t>((sample_rate >> 16) & 0xFF);
+ tmp[2] = static_cast<uint8_t>((sample_rate >> 8) & 0xFF);
+ tmp[3] = static_cast<uint8_t>((sample_rate ) & 0xFF);
+ tmp[4] = static_cast<uint8_t>((channel_count >> 24) & 0xFF);
+ tmp[5] = static_cast<uint8_t>((channel_count >> 16) & 0xFF);
+ tmp[6] = static_cast<uint8_t>((channel_count >> 8) & 0xFF);
+ tmp[7] = static_cast<uint8_t>((channel_count ) & 0xFF);
+
+ memcpy(tmp + 8, esds_data, esds_len);
+ } else {
+ ALOGE("Unsupported MIME type \"%s\" in audio substream", mime_type);
+ abortPrepare(BAD_VALUE);
+ return;
+ }
+
+ status_t err = mAudioSource->start();
+ if (err != OK) {
+ ALOGI("failed to start audio source, err=%d", err);
+ abortPrepare(err);
+ return;
+ }
+
+ mFlags |= PREPARING_CONNECTED;
+
+ if (mCachedSource != NULL) {
+ postBufferingEvent_l();
+ } else {
+ finishAsyncPrepare_l();
+ }
+}
+
+void AAH_TXPlayer::finishAsyncPrepare_l() {
+ notifyListener_l(MEDIA_PREPARED);
+
+ mPrepareResult = OK;
+ mFlags &= ~(PREPARING|PREPARE_CANCELLED|PREPARING_CONNECTED);
+ mFlags |= PREPARED;
+ mPreparedCondition.broadcast();
+}
+
+status_t AAH_TXPlayer::start() {
+ Mutex::Autolock autoLock(mLock);
+
+ mFlags &= ~CACHE_UNDERRUN;
+
+ return play_l();
+}
+
+status_t AAH_TXPlayer::play_l() {
+ if (mFlags & PLAYING) {
+ return OK;
+ }
+
+ if (!(mFlags & PREPARED)) {
+ return INVALID_OPERATION;
+ }
+
+ {
+ Mutex::Autolock lock(mEndpointLock);
+ if (!mEndpointValid) {
+ return INVALID_OPERATION;
+ }
+ if (!mEndpointRegistered) {
+ mProgramID = mAAH_Sender->registerEndpoint(mEndpoint);
+ mEndpointRegistered = true;
+ }
+ }
+
+ mFlags |= PLAYING;
+
+ updateClockTransform_l(false);
+
+ postPumpAudioEvent_l(-1);
+
+ return OK;
+}
+
+status_t AAH_TXPlayer::stop() {
+ status_t ret = pause();
+ sendEOS_l();
+ return ret;
+}
+
+status_t AAH_TXPlayer::pause() {
+ Mutex::Autolock autoLock(mLock);
+
+ mFlags &= ~CACHE_UNDERRUN;
+
+ return pause_l();
+}
+
+status_t AAH_TXPlayer::pause_l(bool doClockUpdate) {
+ if (!(mFlags & PLAYING)) {
+ return OK;
+ }
+
+ cancelPlayerEvents(true /* keepBufferingGoing */);
+
+ mFlags &= ~PLAYING;
+
+ if (doClockUpdate) {
+ updateClockTransform_l(true);
+ }
+
+ return OK;
+}
+
+void AAH_TXPlayer::updateClockTransform_l(bool pause) {
+ // record the new pause status so that onPumpAudio knows what rate to apply
+ // when it initializes the transform
+ mPlayRateIsPaused = pause;
+
+ // if we haven't yet established a valid clock transform, then we can't
+ // do anything here
+ if (!mCurrentClockTransformValid) {
+ return;
+ }
+
+ // sample the current common time
+ int64_t commonTimeNow;
+ if (OK != mCCHelper.getCommonTime(&commonTimeNow)) {
+ ALOGE("updateClockTransform_l get common time failed");
+ mCurrentClockTransformValid = false;
+ return;
+ }
+
+ // convert the current common time to media time using the old
+ // transform
+ int64_t mediaTimeNow;
+ if (!mCurrentClockTransform.doReverseTransform(
+ commonTimeNow, &mediaTimeNow)) {
+ ALOGE("updateClockTransform_l reverse transform failed");
+ mCurrentClockTransformValid = false;
+ return;
+ }
+
+ // calculate a new transform that preserves the old transform's
+ // result for the current time
+ mCurrentClockTransform.a_zero = mediaTimeNow;
+ mCurrentClockTransform.b_zero = commonTimeNow;
+ mCurrentClockTransform.a_to_b_numer = 1;
+ mCurrentClockTransform.a_to_b_denom = pause ? 0 : 1;
+
+ // send a packet announcing the new transform
+ sp<TRTPControlPacket> packet = new TRTPControlPacket();
+ packet->setClockTransform(mCurrentClockTransform);
+ packet->setCommandID(TRTPControlPacket::kCommandNop);
+ queuePacketToSender_l(packet);
+}
+
+void AAH_TXPlayer::sendEOS_l() {
+ sp<TRTPControlPacket> packet = new TRTPControlPacket();
+ packet->setCommandID(TRTPControlPacket::kCommandEOS);
+ queuePacketToSender_l(packet);
+}
+
+bool AAH_TXPlayer::isPlaying() {
+ return (mFlags & PLAYING) || (mFlags & CACHE_UNDERRUN);
+}
+
+status_t AAH_TXPlayer::seekTo(int msec) {
+ if (mExtractorFlags & MediaExtractor::CAN_SEEK) {
+ Mutex::Autolock autoLock(mLock);
+ return seekTo_l(static_cast<int64_t>(msec) * 1000);
+ }
+
+ notifyListener_l(MEDIA_SEEK_COMPLETE);
+ return OK;
+}
+
+status_t AAH_TXPlayer::seekTo_l(int64_t timeUs) {
+ mIsSeeking = true;
+ mSeekTimeUs = timeUs;
+
+ mCurrentClockTransformValid = false;
+ mLastQueuedMediaTimePTSValid = false;
+
+ // send a flush command packet
+ sp<TRTPControlPacket> packet = new TRTPControlPacket();
+ packet->setCommandID(TRTPControlPacket::kCommandFlush);
+ queuePacketToSender_l(packet);
+
+ return OK;
+}
+
+status_t AAH_TXPlayer::getCurrentPosition(int *msec) {
+ if (!msec) {
+ return BAD_VALUE;
+ }
+
+ Mutex::Autolock lock(mLock);
+
+ int position;
+
+ if (mIsSeeking) {
+ position = mSeekTimeUs / 1000;
+ } else if (mCurrentClockTransformValid) {
+ // sample the current common time
+ int64_t commonTimeNow;
+ if (OK != mCCHelper.getCommonTime(&commonTimeNow)) {
+ ALOGE("getCurrentPosition get common time failed");
+ return INVALID_OPERATION;
+ }
+
+ int64_t mediaTimeNow;
+ if (!mCurrentClockTransform.doReverseTransform(commonTimeNow,
+ &mediaTimeNow)) {
+ ALOGE("getCurrentPosition reverse transform failed");
+ return INVALID_OPERATION;
+ }
+
+ position = static_cast<int>(mediaTimeNow / 1000);
+ } else {
+ position = 0;
+ }
+
+ int duration;
+ if (getDuration_l(&duration) == OK) {
+ *msec = clamp(position, 0, duration);
+ } else {
+ *msec = (position >= 0) ? position : 0;
+ }
+
+ return OK;
+}
+
+status_t AAH_TXPlayer::getDuration(int* msec) {
+ if (!msec) {
+ return BAD_VALUE;
+ }
+
+ Mutex::Autolock lock(mLock);
+
+ return getDuration_l(msec);
+}
+
+status_t AAH_TXPlayer::getDuration_l(int* msec) {
+ if (mDurationUs < 0) {
+ return UNKNOWN_ERROR;
+ }
+
+ *msec = (mDurationUs + 500) / 1000;
+
+ return OK;
+}
+
+status_t AAH_TXPlayer::reset() {
+ Mutex::Autolock autoLock(mLock);
+ reset_l();
+ return OK;
+}
+
+void AAH_TXPlayer::reset_l() {
+ if (mFlags & PREPARING) {
+ mFlags |= PREPARE_CANCELLED;
+ if (mConnectingDataSource != NULL) {
+ ALOGI("interrupting the connection process");
+ mConnectingDataSource->disconnect();
+ }
+
+ if (mFlags & PREPARING_CONNECTED) {
+ // We are basically done preparing, we're just buffering
+ // enough data to start playback, we can safely interrupt that.
+ finishAsyncPrepare_l();
+ }
+ }
+
+ while (mFlags & PREPARING) {
+ mPreparedCondition.wait(mLock);
+ }
+
+ cancelPlayerEvents();
+
+ sendEOS_l();
+
+ mCachedSource.clear();
+
+ if (mAudioSource != NULL) {
+ mAudioSource->stop();
+ }
+ mAudioSource.clear();
+ mAudioCodec = TRTPAudioPacket::kCodecInvalid;
+ mAudioFormat = NULL;
+ delete[] mAudioCodecData;
+ mAudioCodecData = NULL;
+ mAudioCodecDataSize = 0;
+
+ mFlags = 0;
+ mExtractorFlags = 0;
+
+ mDurationUs = -1;
+ mIsSeeking = false;
+ mSeekTimeUs = 0;
+
+ mUri.setTo("");
+ mUriHeaders.clear();
+
+ mFileSource.clear();
+
+ mBitrate = -1;
+
+ {
+ Mutex::Autolock lock(mEndpointLock);
+ if (mAAH_Sender != NULL && mEndpointRegistered) {
+ mAAH_Sender->unregisterEndpoint(mEndpoint);
+ }
+ mEndpointRegistered = false;
+ mEndpointValid = false;
+ }
+
+ mProgramID = 0;
+
+ mAAH_Sender.clear();
+ mLastQueuedMediaTimePTSValid = false;
+ mCurrentClockTransformValid = false;
+ mPlayRateIsPaused = false;
+
+ mTRTPVolume = 255;
+}
+
+status_t AAH_TXPlayer::setLooping(int loop) {
+ return OK;
+}
+
+player_type AAH_TXPlayer::playerType() {
+ return AAH_TX_PLAYER;
+}
+
+status_t AAH_TXPlayer::setParameter(int key, const Parcel &request) {
+ return ERROR_UNSUPPORTED;
+}
+
+status_t AAH_TXPlayer::getParameter(int key, Parcel *reply) {
+ return ERROR_UNSUPPORTED;
+}
+
+status_t AAH_TXPlayer::invoke(const Parcel& request, Parcel *reply) {
+ return INVALID_OPERATION;
+}
+
+status_t AAH_TXPlayer::getMetadata(const media::Metadata::Filter& ids,
+ Parcel* records) {
+ using media::Metadata;
+
+ Metadata metadata(records);
+
+ metadata.appendBool(Metadata::kPauseAvailable, true);
+ metadata.appendBool(Metadata::kSeekBackwardAvailable, false);
+ metadata.appendBool(Metadata::kSeekForwardAvailable, false);
+ metadata.appendBool(Metadata::kSeekAvailable, false);
+
+ return OK;
+}
+
+status_t AAH_TXPlayer::setVolume(float leftVolume, float rightVolume) {
+ if (leftVolume != rightVolume) {
+ ALOGE("%s does not support per channel volume: %f, %f",
+ __PRETTY_FUNCTION__, leftVolume, rightVolume);
+ }
+
+ float volume = clamp(leftVolume, 0.0f, 1.0f);
+
+ Mutex::Autolock lock(mLock);
+ mTRTPVolume = static_cast<uint8_t>((leftVolume * 255.0) + 0.5);
+
+ return OK;
+}
+
+status_t AAH_TXPlayer::setAudioStreamType(audio_stream_type_t streamType) {
+ return OK;
+}
+
+status_t AAH_TXPlayer::setRetransmitEndpoint(
+ const struct sockaddr_in* endpoint) {
+ Mutex::Autolock lock(mLock);
+
+ if (NULL == endpoint)
+ return BAD_VALUE;
+
+ // Once the endpoint has been registered, it may not be changed.
+ if (mEndpointRegistered)
+ return INVALID_OPERATION;
+
+ mEndpoint.addr = endpoint->sin_addr.s_addr;
+ mEndpoint.port = endpoint->sin_port;
+ mEndpointValid = true;
+
+ return OK;
+}
+
+void AAH_TXPlayer::notifyListener_l(int msg, int ext1, int ext2) {
+ sendEvent(msg, ext1, ext2);
+}
+
+bool AAH_TXPlayer::getBitrate_l(int64_t *bitrate) {
+ off64_t size;
+ if (mDurationUs >= 0 &&
+ mCachedSource != NULL &&
+ mCachedSource->getSize(&size) == OK) {
+ *bitrate = size * 8000000ll / mDurationUs; // in bits/sec
+ return true;
+ }
+
+ if (mBitrate >= 0) {
+ *bitrate = mBitrate;
+ return true;
+ }
+
+ *bitrate = 0;
+
+ return false;
+}
+
+// Returns true iff cached duration is available/applicable.
+bool AAH_TXPlayer::getCachedDuration_l(int64_t *durationUs, bool *eos) {
+ int64_t bitrate;
+
+ if (mCachedSource != NULL && getBitrate_l(&bitrate)) {
+ status_t finalStatus;
+ size_t cachedDataRemaining = mCachedSource->approxDataRemaining(
+ &finalStatus);
+ *durationUs = cachedDataRemaining * 8000000ll / bitrate;
+ *eos = (finalStatus != OK);
+ return true;
+ }
+
+ return false;
+}
+
+void AAH_TXPlayer::ensureCacheIsFetching_l() {
+ if (mCachedSource != NULL) {
+ mCachedSource->resumeFetchingIfNecessary();
+ }
+}
+
+void AAH_TXPlayer::postBufferingEvent_l() {
+ if (mBufferingEventPending) {
+ return;
+ }
+ mBufferingEventPending = true;
+ mQueue.postEventWithDelay(mBufferingEvent, 1000000ll);
+}
+
+void AAH_TXPlayer::postPumpAudioEvent_l(int64_t delayUs) {
+ if (mPumpAudioEventPending) {
+ return;
+ }
+ mPumpAudioEventPending = true;
+ mQueue.postEventWithDelay(mPumpAudioEvent, delayUs < 0 ? 10000 : delayUs);
+}
+
+void AAH_TXPlayer::onBufferingUpdate() {
+ Mutex::Autolock autoLock(mLock);
+ if (!mBufferingEventPending) {
+ return;
+ }
+ mBufferingEventPending = false;
+
+ if (mCachedSource != NULL) {
+ status_t finalStatus;
+ size_t cachedDataRemaining = mCachedSource->approxDataRemaining(
+ &finalStatus);
+ bool eos = (finalStatus != OK);
+
+ if (eos) {
+ if (finalStatus == ERROR_END_OF_STREAM) {
+ notifyListener_l(MEDIA_BUFFERING_UPDATE, 100);
+ }
+ if (mFlags & PREPARING) {
+ ALOGV("cache has reached EOS, prepare is done.");
+ finishAsyncPrepare_l();
+ }
+ } else {
+ int64_t bitrate;
+ if (getBitrate_l(&bitrate)) {
+ size_t cachedSize = mCachedSource->cachedSize();
+ int64_t cachedDurationUs = cachedSize * 8000000ll / bitrate;
+
+ int percentage = (100.0 * (double) cachedDurationUs)
+ / mDurationUs;
+ if (percentage > 100) {
+ percentage = 100;
+ }
+
+ notifyListener_l(MEDIA_BUFFERING_UPDATE, percentage);
+ } else {
+ // We don't know the bitrate of the stream, use absolute size
+ // limits to maintain the cache.
+
+ if ((mFlags & PLAYING) &&
+ !eos &&
+ (cachedDataRemaining < kLowWaterMarkBytes)) {
+ ALOGI("cache is running low (< %d) , pausing.",
+ kLowWaterMarkBytes);
+ mFlags |= CACHE_UNDERRUN;
+ pause_l();
+ ensureCacheIsFetching_l();
+ notifyListener_l(MEDIA_INFO, MEDIA_INFO_BUFFERING_START);
+ } else if (eos || cachedDataRemaining > kHighWaterMarkBytes) {
+ if (mFlags & CACHE_UNDERRUN) {
+ ALOGI("cache has filled up (> %d), resuming.",
+ kHighWaterMarkBytes);
+ mFlags &= ~CACHE_UNDERRUN;
+ play_l();
+ notifyListener_l(MEDIA_INFO, MEDIA_INFO_BUFFERING_END);
+ } else if (mFlags & PREPARING) {
+ ALOGV("cache has filled up (> %d), prepare is done",
+ kHighWaterMarkBytes);
+ finishAsyncPrepare_l();
+ }
+ }
+ }
+ }
+ }
+
+ int64_t cachedDurationUs;
+ bool eos;
+ if (getCachedDuration_l(&cachedDurationUs, &eos)) {
+ ALOGV("cachedDurationUs = %.2f secs, eos=%d",
+ cachedDurationUs / 1E6, eos);
+
+ if ((mFlags & PLAYING) &&
+ !eos &&
+ (cachedDurationUs < kLowWaterMarkUs)) {
+ ALOGI("cache is running low (%.2f secs) , pausing.",
+ cachedDurationUs / 1E6);
+ mFlags |= CACHE_UNDERRUN;
+ pause_l();
+ ensureCacheIsFetching_l();
+ notifyListener_l(MEDIA_INFO, MEDIA_INFO_BUFFERING_START);
+ } else if (eos || cachedDurationUs > kHighWaterMarkUs) {
+ if (mFlags & CACHE_UNDERRUN) {
+ ALOGI("cache has filled up (%.2f secs), resuming.",
+ cachedDurationUs / 1E6);
+ mFlags &= ~CACHE_UNDERRUN;
+ play_l();
+ notifyListener_l(MEDIA_INFO, MEDIA_INFO_BUFFERING_END);
+ } else if (mFlags & PREPARING) {
+ ALOGV("cache has filled up (%.2f secs), prepare is done",
+ cachedDurationUs / 1E6);
+ finishAsyncPrepare_l();
+ }
+ }
+ }
+
+ postBufferingEvent_l();
+}
+
+void AAH_TXPlayer::onPumpAudio() {
+ while (true) {
+ Mutex::Autolock autoLock(mLock);
+ // If this flag is clear, its because someone has externally canceled
+ // this pump operation (probably because we a resetting/shutting down).
+ // Get out immediately, do not reschedule ourselves.
+ if (!mPumpAudioEventPending) {
+ return;
+ }
+
+ // Start by checking if there is still work to be doing. If we have
+ // never queued a payload (so we don't know what the last queued PTS is)
+ // or we have never established a MediaTime->CommonTime transformation,
+ // then we have work to do (one time through this loop should establish
+ // both). Otherwise, we want to keep a fixed amt of presentation time
+ // worth of data buffered. If we cannot get common time (service is
+ // unavailable, or common time is undefined)) then we don't have a lot
+ // of good options here. For now, signal an error up to the app level
+ // and shut down the transmission pump.
+ int64_t commonTimeNow;
+ if (OK != mCCHelper.getCommonTime(&commonTimeNow)) {
+ // Failed to get common time; either the service is down or common
+ // time is not synced. Raise an error and shutdown the player.
+ ALOGE("*** Cannot pump audio, unable to fetch common time."
+ " Shutting down.");
+ notifyListener_l(MEDIA_ERROR, MEDIA_ERROR_UNKNOWN, UNKNOWN_ERROR);
+ mPumpAudioEventPending = false;
+ break;
+ }
+
+ if (mCurrentClockTransformValid && mLastQueuedMediaTimePTSValid) {
+ int64_t mediaTimeNow;
+ bool conversionResult = mCurrentClockTransform.doReverseTransform(
+ commonTimeNow,
+ &mediaTimeNow);
+ CHECK(conversionResult);
+
+ if ((mediaTimeNow +
+ kAAHBufferTimeUs -
+ mLastQueuedMediaTimePTS) <= 0) {
+ break;
+ }
+ }
+
+ MediaSource::ReadOptions options;
+ if (mIsSeeking) {
+ options.setSeekTo(mSeekTimeUs);
+ }
+
+ MediaBuffer* mediaBuffer;
+ status_t err = mAudioSource->read(&mediaBuffer, &options);
+ if (err != NO_ERROR) {
+ if (err == ERROR_END_OF_STREAM) {
+ ALOGI("*** %s reached end of stream", __PRETTY_FUNCTION__);
+ notifyListener_l(MEDIA_BUFFERING_UPDATE, 100);
+ notifyListener_l(MEDIA_PLAYBACK_COMPLETE);
+ pause_l(false);
+ sendEOS_l();
+ } else {
+ ALOGE("*** %s read failed err=%d", __PRETTY_FUNCTION__, err);
+ }
+ return;
+ }
+
+ if (mIsSeeking) {
+ mIsSeeking = false;
+ notifyListener_l(MEDIA_SEEK_COMPLETE);
+ }
+
+ uint8_t* data = (static_cast<uint8_t*>(mediaBuffer->data()) +
+ mediaBuffer->range_offset());
+ ALOGV("*** %s got media buffer data=[%02hhx %02hhx %02hhx %02hhx]"
+ " offset=%d length=%d", __PRETTY_FUNCTION__,
+ data[0], data[1], data[2], data[3],
+ mediaBuffer->range_offset(), mediaBuffer->range_length());
+
+ int64_t mediaTimeUs;
+ CHECK(mediaBuffer->meta_data()->findInt64(kKeyTime, &mediaTimeUs));
+ ALOGV("*** timeUs=%lld", mediaTimeUs);
+
+ if (!mCurrentClockTransformValid) {
+ if (OK == mCCHelper.getCommonTime(&commonTimeNow)) {
+ mCurrentClockTransform.a_zero = mediaTimeUs;
+ mCurrentClockTransform.b_zero = commonTimeNow +
+ kAAHStartupLeadTimeUs;
+ mCurrentClockTransform.a_to_b_numer = 1;
+ mCurrentClockTransform.a_to_b_denom = mPlayRateIsPaused ? 0 : 1;
+ mCurrentClockTransformValid = true;
+ } else {
+ // Failed to get common time; either the service is down or
+ // common time is not synced. Raise an error and shutdown the
+ // player.
+ ALOGE("*** Cannot begin transmission, unable to fetch common"
+ " time. Dropping sample with pts=%lld", mediaTimeUs);
+ notifyListener_l(MEDIA_ERROR,
+ MEDIA_ERROR_UNKNOWN,
+ UNKNOWN_ERROR);
+ mPumpAudioEventPending = false;
+ break;
+ }
+ }
+
+ ALOGV("*** transmitting packet with pts=%lld", mediaTimeUs);
+
+ sp<TRTPAudioPacket> packet = new TRTPAudioPacket();
+ packet->setPTS(mediaTimeUs);
+ packet->setSubstreamID(1);
+
+ packet->setCodecType(mAudioCodec);
+ packet->setVolume(mTRTPVolume);
+ // TODO : introduce a throttle for this so we can control the
+ // frequency with which transforms get sent.
+ packet->setClockTransform(mCurrentClockTransform);
+ packet->setAccessUnitData(data, mediaBuffer->range_length());
+
+ // TODO : while its pretty much universally true that audio ES payloads
+ // are all RAPs across all codecs, it might be a good idea to throttle
+ // the frequency with which we send codec out of band data to the RXers.
+ // If/when we do, we need to flag only those payloads which have
+ // required out of band data attached to them as RAPs.
+ packet->setRandomAccessPoint(true);
+
+ if (mAudioCodecData && mAudioCodecDataSize) {
+ packet->setAuxData(mAudioCodecData, mAudioCodecDataSize);
+ }
+
+ queuePacketToSender_l(packet);
+ mediaBuffer->release();
+
+ mLastQueuedMediaTimePTSValid = true;
+ mLastQueuedMediaTimePTS = mediaTimeUs;
+ }
+
+ { // Explicit scope for the autolock pattern.
+ Mutex::Autolock autoLock(mLock);
+
+ // If someone externally has cleared this flag, its because we should be
+ // shutting down. Do not reschedule ourselves.
+ if (!mPumpAudioEventPending) {
+ return;
+ }
+
+ // Looks like no one canceled us explicitly. Clear our flag and post a
+ // new event to ourselves.
+ mPumpAudioEventPending = false;
+ postPumpAudioEvent_l(10000);
+ }
+}
+
+void AAH_TXPlayer::queuePacketToSender_l(const sp<TRTPPacket>& packet) {
+ if (mAAH_Sender == NULL) {
+ return;
+ }
+
+ sp<AMessage> message = new AMessage(AAH_TXSender::kWhatSendPacket,
+ mAAH_Sender->handlerID());
+
+ {
+ Mutex::Autolock lock(mEndpointLock);
+ if (!mEndpointValid) {
+ return;
+ }
+
+ message->setInt32(AAH_TXSender::kSendPacketIPAddr, mEndpoint.addr);
+ message->setInt32(AAH_TXSender::kSendPacketPort, mEndpoint.port);
+ }
+
+ packet->setProgramID(mProgramID);
+ packet->setExpireTime(systemTime() + kAAHRetryKeepAroundTimeNs);
+ packet->pack();
+
+ message->setObject(AAH_TXSender::kSendPacketTRTPPacket, packet);
+
+ message->post();
+}
+
+} // namespace android
diff --git a/media/libaah_rtp/aah_tx_player.h b/media/libaah_rtp/aah_tx_player.h
new file mode 100644
index 0000000..2e4b1f7
--- /dev/null
+++ b/media/libaah_rtp/aah_tx_player.h
@@ -0,0 +1,176 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __AAH_TX_PLAYER_H__
+#define __AAH_TX_PLAYER_H__
+
+#include <common_time/cc_helper.h>
+#include <libstagefright/include/HTTPBase.h>
+#include <libstagefright/include/NuCachedSource2.h>
+#include <libstagefright/include/TimedEventQueue.h>
+#include <media/MediaPlayerInterface.h>
+#include <media/stagefright/MediaExtractor.h>
+#include <media/stagefright/MediaSource.h>
+#include <utils/LinearTransform.h>
+#include <utils/String8.h>
+#include <utils/threads.h>
+
+#include "aah_tx_sender.h"
+
+namespace android {
+
+class AAH_TXPlayer : public MediaPlayerHWInterface {
+ public:
+ AAH_TXPlayer();
+
+ virtual status_t initCheck();
+ virtual status_t setDataSource(const char *url,
+ const KeyedVector<String8, String8>*
+ headers);
+ virtual status_t setDataSource(int fd, int64_t offset, int64_t length);
+ virtual status_t setVideoSurface(const sp<Surface>& surface);
+ virtual status_t setVideoSurfaceTexture(const sp<ISurfaceTexture>&
+ surfaceTexture);
+ virtual status_t prepare();
+ virtual status_t prepareAsync();
+ virtual status_t start();
+ virtual status_t stop();
+ virtual status_t pause();
+ virtual bool isPlaying();
+ virtual status_t seekTo(int msec);
+ virtual status_t getCurrentPosition(int *msec);
+ virtual status_t getDuration(int *msec);
+ virtual status_t reset();
+ virtual status_t setLooping(int loop);
+ virtual player_type playerType();
+ virtual status_t setParameter(int key, const Parcel &request);
+ virtual status_t getParameter(int key, Parcel *reply);
+ virtual status_t invoke(const Parcel& request, Parcel *reply);
+ virtual status_t getMetadata(const media::Metadata::Filter& ids,
+ Parcel* records);
+ virtual status_t setVolume(float leftVolume, float rightVolume);
+ virtual status_t setAudioStreamType(audio_stream_type_t streamType);
+ virtual status_t setRetransmitEndpoint(
+ const struct sockaddr_in* endpoint);
+
+ static const int64_t kAAHRetryKeepAroundTimeNs;
+
+ protected:
+ virtual ~AAH_TXPlayer();
+
+ private:
+ friend struct AwesomeEvent;
+
+ enum {
+ PLAYING = 1,
+ PREPARING = 8,
+ PREPARED = 16,
+ PREPARE_CANCELLED = 64,
+ CACHE_UNDERRUN = 128,
+
+ // We are basically done preparing but are currently buffering
+ // sufficient data to begin playback and finish the preparation
+ // phase for good.
+ PREPARING_CONNECTED = 2048,
+
+ INCOGNITO = 32768,
+ };
+
+ status_t setDataSource_l(const char *url,
+ const KeyedVector<String8, String8> *headers);
+ status_t setDataSource_l(const sp<MediaExtractor>& extractor);
+ status_t finishSetDataSource_l();
+ status_t prepareAsync_l();
+ void onPrepareAsyncEvent();
+ void finishAsyncPrepare_l();
+ void abortPrepare(status_t err);
+ status_t play_l();
+ status_t pause_l(bool doClockUpdate = true);
+ status_t seekTo_l(int64_t timeUs);
+ void updateClockTransform_l(bool pause);
+ void sendEOS_l();
+ void cancelPlayerEvents(bool keepBufferingGoing = false);
+ void reset_l();
+ void notifyListener_l(int msg, int ext1 = 0, int ext2 = 0);
+ bool getBitrate_l(int64_t* bitrate);
+ status_t getDuration_l(int* msec);
+ bool getCachedDuration_l(int64_t* durationUs, bool* eos);
+ void ensureCacheIsFetching_l();
+ void postBufferingEvent_l();
+ void postPumpAudioEvent_l(int64_t delayUs);
+ void onBufferingUpdate();
+ void onPumpAudio();
+ void queuePacketToSender_l(const sp<TRTPPacket>& packet);
+
+ Mutex mLock;
+
+ TimedEventQueue mQueue;
+ bool mQueueStarted;
+
+ sp<TimedEventQueue::Event> mBufferingEvent;
+ bool mBufferingEventPending;
+
+ uint32_t mFlags;
+ uint32_t mExtractorFlags;
+
+ String8 mUri;
+ KeyedVector<String8, String8> mUriHeaders;
+
+ sp<DataSource> mFileSource;
+
+ sp<TimedEventQueue::Event> mAsyncPrepareEvent;
+ Condition mPreparedCondition;
+ status_t mPrepareResult;
+
+ bool mIsSeeking;
+ int64_t mSeekTimeUs;
+
+ sp<TimedEventQueue::Event> mPumpAudioEvent;
+ bool mPumpAudioEventPending;
+
+ sp<HTTPBase> mConnectingDataSource;
+ sp<NuCachedSource2> mCachedSource;
+
+ sp<MediaSource> mAudioSource;
+ TRTPAudioPacket::TRTPAudioCodecType mAudioCodec;
+ sp<MetaData> mAudioFormat;
+ uint8_t* mAudioCodecData;
+ size_t mAudioCodecDataSize;
+
+ int64_t mDurationUs;
+ int64_t mBitrate;
+
+ sp<AAH_TXSender> mAAH_Sender;
+ LinearTransform mCurrentClockTransform;
+ bool mCurrentClockTransformValid;
+ int64_t mLastQueuedMediaTimePTS;
+ bool mLastQueuedMediaTimePTSValid;
+ bool mPlayRateIsPaused;
+ CCHelper mCCHelper;
+
+ Mutex mEndpointLock;
+ AAH_TXSender::Endpoint mEndpoint;
+ bool mEndpointValid;
+ bool mEndpointRegistered;
+ uint16_t mProgramID;
+ uint8_t mTRTPVolume;
+
+ DISALLOW_EVIL_CONSTRUCTORS(AAH_TXPlayer);
+};
+
+} // namespace android
+
+#endif // __AAH_TX_PLAYER_H__
diff --git a/media/libaah_rtp/aah_tx_sender.cpp b/media/libaah_rtp/aah_tx_sender.cpp
new file mode 100644
index 0000000..08e32d2
--- /dev/null
+++ b/media/libaah_rtp/aah_tx_sender.cpp
@@ -0,0 +1,603 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "LibAAH_RTP"
+#include <media/stagefright/foundation/ADebug.h>
+
+#include <netinet/in.h>
+#include <poll.h>
+#include <sys/types.h>
+#include <sys/socket.h>
+#include <unistd.h>
+
+#include <media/stagefright/foundation/AMessage.h>
+#include <utils/misc.h>
+
+#include "aah_tx_player.h"
+#include "aah_tx_sender.h"
+
+namespace android {
+
+const char* AAH_TXSender::kSendPacketIPAddr = "ipaddr";
+const char* AAH_TXSender::kSendPacketPort = "port";
+const char* AAH_TXSender::kSendPacketTRTPPacket = "trtp";
+
+const int AAH_TXSender::kRetryTrimIntervalUs = 100000;
+const int AAH_TXSender::kHeartbeatIntervalUs = 1000000;
+const int AAH_TXSender::kRetryBufferCapacity = 100;
+const nsecs_t AAH_TXSender::kHeartbeatTimeout = 600ull * 1000000000ull;
+
+Mutex AAH_TXSender::sLock;
+wp<AAH_TXSender> AAH_TXSender::sInstance;
+uint32_t AAH_TXSender::sNextEpoch;
+bool AAH_TXSender::sNextEpochValid = false;
+
+AAH_TXSender::AAH_TXSender() : mSocket(-1) {
+ mLastSentPacketTime = systemTime();
+}
+
+sp<AAH_TXSender> AAH_TXSender::GetInstance() {
+ Mutex::Autolock autoLock(sLock);
+
+ sp<AAH_TXSender> sender = sInstance.promote();
+
+ if (sender == NULL) {
+ sender = new AAH_TXSender();
+ if (sender == NULL) {
+ return NULL;
+ }
+
+ sender->mLooper = new ALooper();
+ if (sender->mLooper == NULL) {
+ return NULL;
+ }
+
+ sender->mReflector = new AHandlerReflector<AAH_TXSender>(sender.get());
+ if (sender->mReflector == NULL) {
+ return NULL;
+ }
+
+ sender->mSocket = socket(AF_INET, SOCK_DGRAM, IPPROTO_UDP);
+ if (sender->mSocket == -1) {
+ ALOGW("%s unable to create socket", __PRETTY_FUNCTION__);
+ return NULL;
+ }
+
+ struct sockaddr_in bind_addr;
+ memset(&bind_addr, 0, sizeof(bind_addr));
+ bind_addr.sin_family = AF_INET;
+ if (bind(sender->mSocket,
+ reinterpret_cast<const sockaddr*>(&bind_addr),
+ sizeof(bind_addr)) < 0) {
+ ALOGW("%s unable to bind socket (errno %d)",
+ __PRETTY_FUNCTION__, errno);
+ return NULL;
+ }
+
+ sender->mRetryReceiver = new RetryReceiver(sender.get());
+ if (sender->mRetryReceiver == NULL) {
+ return NULL;
+ }
+
+ sender->mLooper->setName("AAH_TXSender");
+ sender->mLooper->registerHandler(sender->mReflector);
+ sender->mLooper->start(false, false, PRIORITY_AUDIO);
+
+ if (sender->mRetryReceiver->run("AAH_TXSenderRetry", PRIORITY_AUDIO)
+ != OK) {
+ ALOGW("%s unable to start retry thread", __PRETTY_FUNCTION__);
+ return NULL;
+ }
+
+ sInstance = sender;
+ }
+
+ return sender;
+}
+
+AAH_TXSender::~AAH_TXSender() {
+ mLooper->stop();
+ mLooper->unregisterHandler(mReflector->id());
+
+ if (mRetryReceiver != NULL) {
+ mRetryReceiver->requestExit();
+ mRetryReceiver->mWakeupEvent.setEvent();
+ if (mRetryReceiver->requestExitAndWait() != OK) {
+ ALOGW("%s shutdown of retry receiver failed", __PRETTY_FUNCTION__);
+ }
+ mRetryReceiver->mSender = NULL;
+ mRetryReceiver.clear();
+ }
+
+ if (mSocket != -1) {
+ close(mSocket);
+ }
+}
+
+// Return the next epoch number usable for a newly instantiated endpoint.
+uint32_t AAH_TXSender::getNextEpoch() {
+ Mutex::Autolock autoLock(sLock);
+
+ if (sNextEpochValid) {
+ sNextEpoch = (sNextEpoch + 1) & TRTPPacket::kTRTPEpochMask;
+ } else {
+ sNextEpoch = ns2ms(systemTime()) & TRTPPacket::kTRTPEpochMask;
+ sNextEpochValid = true;
+ }
+
+ return sNextEpoch;
+}
+
+// Notify the sender that a player has started sending to this endpoint.
+// Returns a program ID for use by the calling player.
+uint16_t AAH_TXSender::registerEndpoint(const Endpoint& endpoint) {
+ Mutex::Autolock lock(mEndpointLock);
+
+ EndpointState* eps = mEndpointMap.valueFor(endpoint);
+ if (eps) {
+ eps->playerRefCount++;
+ } else {
+ eps = new EndpointState(getNextEpoch());
+ mEndpointMap.add(endpoint, eps);
+ }
+
+ // if this is the first registered endpoint, then send a message to start
+ // trimming retry buffers and a message to start sending heartbeats.
+ if (mEndpointMap.size() == 1) {
+ sp<AMessage> trimMessage = new AMessage(kWhatTrimRetryBuffers,
+ handlerID());
+ trimMessage->post(kRetryTrimIntervalUs);
+
+ sp<AMessage> heartbeatMessage = new AMessage(kWhatSendHeartbeats,
+ handlerID());
+ heartbeatMessage->post(kHeartbeatIntervalUs);
+ }
+
+ eps->nextProgramID++;
+ return eps->nextProgramID;
+}
+
+// Notify the sender that a player has ceased sending to this endpoint.
+// An endpoint's state can not be deleted until all of the endpoint's
+// registered players have called unregisterEndpoint.
+void AAH_TXSender::unregisterEndpoint(const Endpoint& endpoint) {
+ Mutex::Autolock lock(mEndpointLock);
+
+ EndpointState* eps = mEndpointMap.valueFor(endpoint);
+ if (eps) {
+ eps->playerRefCount--;
+ CHECK(eps->playerRefCount >= 0);
+ }
+}
+
+void AAH_TXSender::onMessageReceived(const sp<AMessage>& msg) {
+ switch (msg->what()) {
+ case kWhatSendPacket:
+ onSendPacket(msg);
+ break;
+
+ case kWhatTrimRetryBuffers:
+ trimRetryBuffers();
+ break;
+
+ case kWhatSendHeartbeats:
+ sendHeartbeats();
+ break;
+
+ default:
+ TRESPASS();
+ break;
+ }
+}
+
+void AAH_TXSender::onSendPacket(const sp<AMessage>& msg) {
+ sp<RefBase> obj;
+ CHECK(msg->findObject(kSendPacketTRTPPacket, &obj));
+ sp<TRTPPacket> packet = static_cast<TRTPPacket*>(obj.get());
+
+ uint32_t ipAddr;
+ CHECK(msg->findInt32(kSendPacketIPAddr,
+ reinterpret_cast<int32_t*>(&ipAddr)));
+
+ int32_t port32;
+ CHECK(msg->findInt32(kSendPacketPort, &port32));
+ uint16_t port = port32;
+
+ Mutex::Autolock lock(mEndpointLock);
+ doSendPacket_l(packet, Endpoint(ipAddr, port));
+ mLastSentPacketTime = systemTime();
+}
+
+void AAH_TXSender::doSendPacket_l(const sp<TRTPPacket>& packet,
+ const Endpoint& endpoint) {
+ EndpointState* eps = mEndpointMap.valueFor(endpoint);
+ if (!eps) {
+ // the endpoint state has disappeared, so the player that sent this
+ // packet must be dead.
+ return;
+ }
+
+ // assign the packet's sequence number
+ packet->setEpoch(eps->epoch);
+ packet->setSeqNumber(eps->trtpSeqNumber++);
+
+ // add the packet to the retry buffer
+ RetryBuffer& retry = eps->retry;
+ retry.push_back(packet);
+
+ // send the packet
+ struct sockaddr_in addr;
+ memset(&addr, 0, sizeof(addr));
+ addr.sin_family = AF_INET;
+ addr.sin_addr.s_addr = endpoint.addr;
+ addr.sin_port = endpoint.port;
+
+ ssize_t result = sendto(mSocket,
+ packet->getPacket(),
+ packet->getPacketLen(),
+ 0,
+ (const struct sockaddr *) &addr,
+ sizeof(addr));
+ if (result == -1) {
+ ALOGW("%s sendto failed", __PRETTY_FUNCTION__);
+ }
+}
+
+void AAH_TXSender::trimRetryBuffers() {
+ Mutex::Autolock lock(mEndpointLock);
+
+ nsecs_t localTimeNow = systemTime();
+
+ Vector<Endpoint> endpointsToRemove;
+
+ for (size_t i = 0; i < mEndpointMap.size(); i++) {
+ EndpointState* eps = mEndpointMap.editValueAt(i);
+ RetryBuffer& retry = eps->retry;
+
+ while (!retry.isEmpty()) {
+ if (retry[0]->getExpireTime() < localTimeNow) {
+ retry.pop_front();
+ } else {
+ break;
+ }
+ }
+
+ if (retry.isEmpty() && eps->playerRefCount == 0) {
+ endpointsToRemove.add(mEndpointMap.keyAt(i));
+ }
+ }
+
+ // remove the state for any endpoints that are no longer in use
+ for (size_t i = 0; i < endpointsToRemove.size(); i++) {
+ Endpoint& e = endpointsToRemove.editItemAt(i);
+ ALOGD("*** %s removing endpoint addr=%08x",
+ __PRETTY_FUNCTION__, e.addr);
+ size_t index = mEndpointMap.indexOfKey(e);
+ delete mEndpointMap.valueAt(index);
+ mEndpointMap.removeItemsAt(index);
+ }
+
+ // schedule the next trim
+ if (mEndpointMap.size()) {
+ sp<AMessage> trimMessage = new AMessage(kWhatTrimRetryBuffers,
+ handlerID());
+ trimMessage->post(kRetryTrimIntervalUs);
+ }
+}
+
+void AAH_TXSender::sendHeartbeats() {
+ Mutex::Autolock lock(mEndpointLock);
+
+ if (shouldSendHeartbeats_l()) {
+ for (size_t i = 0; i < mEndpointMap.size(); i++) {
+ EndpointState* eps = mEndpointMap.editValueAt(i);
+ const Endpoint& ep = mEndpointMap.keyAt(i);
+
+ sp<TRTPControlPacket> packet = new TRTPControlPacket();
+ packet->setCommandID(TRTPControlPacket::kCommandNop);
+
+ packet->setExpireTime(systemTime() +
+ AAH_TXPlayer::kAAHRetryKeepAroundTimeNs);
+ packet->pack();
+
+ doSendPacket_l(packet, ep);
+ }
+ }
+
+ // schedule the next heartbeat
+ if (mEndpointMap.size()) {
+ sp<AMessage> heartbeatMessage = new AMessage(kWhatSendHeartbeats,
+ handlerID());
+ heartbeatMessage->post(kHeartbeatIntervalUs);
+ }
+}
+
+bool AAH_TXSender::shouldSendHeartbeats_l() {
+ // assert(holding endpoint lock)
+ return (systemTime() < (mLastSentPacketTime + kHeartbeatTimeout));
+}
+
+// Receiver
+
+// initial 4-byte ID of a retry request packet
+const uint32_t AAH_TXSender::RetryReceiver::kRetryRequestID = 'Treq';
+
+// initial 4-byte ID of a retry NAK packet
+const uint32_t AAH_TXSender::RetryReceiver::kRetryNakID = 'Tnak';
+
+// initial 4-byte ID of a fast start request packet
+const uint32_t AAH_TXSender::RetryReceiver::kFastStartRequestID = 'Tfst';
+
+AAH_TXSender::RetryReceiver::RetryReceiver(AAH_TXSender* sender)
+ : Thread(false),
+ mSender(sender) {}
+
+ AAH_TXSender::RetryReceiver::~RetryReceiver() {
+ mWakeupEvent.clearPendingEvents();
+ }
+
+// Returns true if val is within the interval bounded inclusively by
+// start and end. Also handles the case where there is a rollover of the
+// range between start and end.
+template <typename T>
+static inline bool withinIntervalWithRollover(T val, T start, T end) {
+ return ((start <= end && val >= start && val <= end) ||
+ (start > end && (val >= start || val <= end)));
+}
+
+bool AAH_TXSender::RetryReceiver::threadLoop() {
+ struct pollfd pollFds[2];
+ pollFds[0].fd = mSender->mSocket;
+ pollFds[0].events = POLLIN;
+ pollFds[0].revents = 0;
+ pollFds[1].fd = mWakeupEvent.getWakeupHandle();
+ pollFds[1].events = POLLIN;
+ pollFds[1].revents = 0;
+
+ int pollResult = poll(pollFds, NELEM(pollFds), -1);
+ if (pollResult == -1) {
+ ALOGE("%s poll failed", __PRETTY_FUNCTION__);
+ return false;
+ }
+
+ if (exitPending()) {
+ ALOGI("*** %s exiting", __PRETTY_FUNCTION__);
+ return false;
+ }
+
+ if (pollFds[0].revents) {
+ handleRetryRequest();
+ }
+
+ return true;
+}
+
+void AAH_TXSender::RetryReceiver::handleRetryRequest() {
+ ALOGV("*** RX %s start", __PRETTY_FUNCTION__);
+
+ RetryPacket request;
+ struct sockaddr requestSrcAddr;
+ socklen_t requestSrcAddrLen = sizeof(requestSrcAddr);
+
+ ssize_t result = recvfrom(mSender->mSocket, &request, sizeof(request), 0,
+ &requestSrcAddr, &requestSrcAddrLen);
+ if (result == -1) {
+ ALOGE("%s recvfrom failed, errno=%d", __PRETTY_FUNCTION__, errno);
+ return;
+ }
+
+ if (static_cast<size_t>(result) < sizeof(RetryPacket)) {
+ ALOGW("%s short packet received", __PRETTY_FUNCTION__);
+ return;
+ }
+
+ uint32_t host_request_id = ntohl(request.id);
+ if ((host_request_id != kRetryRequestID) &&
+ (host_request_id != kFastStartRequestID)) {
+ ALOGW("%s received retry request with bogus ID (%08x)",
+ __PRETTY_FUNCTION__, host_request_id);
+ return;
+ }
+
+ Endpoint endpoint(request.endpointIP, request.endpointPort);
+
+ Mutex::Autolock lock(mSender->mEndpointLock);
+
+ EndpointState* eps = mSender->mEndpointMap.valueFor(endpoint);
+
+ if (eps == NULL || eps->retry.isEmpty()) {
+ // we have no retry buffer or an empty retry buffer for this endpoint,
+ // so NAK the entire request
+ RetryPacket nak = request;
+ nak.id = htonl(kRetryNakID);
+ result = sendto(mSender->mSocket, &nak, sizeof(nak), 0,
+ &requestSrcAddr, requestSrcAddrLen);
+ if (result == -1) {
+ ALOGW("%s sendto failed", __PRETTY_FUNCTION__);
+ }
+ return;
+ }
+
+ RetryBuffer& retry = eps->retry;
+
+ uint16_t startSeq = ntohs(request.seqStart);
+ uint16_t endSeq = ntohs(request.seqEnd);
+
+ uint16_t retryFirstSeq = retry[0]->getSeqNumber();
+ uint16_t retryLastSeq = retry[retry.size() - 1]->getSeqNumber();
+
+ // If this is a fast start, then force the start of the retry to match the
+ // start of the retransmit ring buffer (unless the end of the retransmit
+ // ring buffer is already past the point of fast start)
+ if ((host_request_id == kFastStartRequestID) &&
+ !((startSeq - retryFirstSeq) & 0x8000)) {
+ startSeq = retryFirstSeq;
+ }
+
+ int startIndex;
+ if (withinIntervalWithRollover(startSeq, retryFirstSeq, retryLastSeq)) {
+ startIndex = static_cast<uint16_t>(startSeq - retryFirstSeq);
+ } else {
+ startIndex = -1;
+ }
+
+ int endIndex;
+ if (withinIntervalWithRollover(endSeq, retryFirstSeq, retryLastSeq)) {
+ endIndex = static_cast<uint16_t>(endSeq - retryFirstSeq);
+ } else {
+ endIndex = -1;
+ }
+
+ if (startIndex == -1 && endIndex == -1) {
+ // no part of the request range is found in the retry buffer
+ RetryPacket nak = request;
+ nak.id = htonl(kRetryNakID);
+ result = sendto(mSender->mSocket, &nak, sizeof(nak), 0,
+ &requestSrcAddr, requestSrcAddrLen);
+ if (result == -1) {
+ ALOGW("%s sendto failed", __PRETTY_FUNCTION__);
+ }
+ return;
+ }
+
+ if (startIndex == -1) {
+ // NAK a subrange at the front of the request range
+ RetryPacket nak = request;
+ nak.id = htonl(kRetryNakID);
+ nak.seqEnd = htons(retryFirstSeq - 1);
+ result = sendto(mSender->mSocket, &nak, sizeof(nak), 0,
+ &requestSrcAddr, requestSrcAddrLen);
+ if (result == -1) {
+ ALOGW("%s sendto failed", __PRETTY_FUNCTION__);
+ }
+
+ startIndex = 0;
+ } else if (endIndex == -1) {
+ // NAK a subrange at the back of the request range
+ RetryPacket nak = request;
+ nak.id = htonl(kRetryNakID);
+ nak.seqStart = htons(retryLastSeq + 1);
+ result = sendto(mSender->mSocket, &nak, sizeof(nak), 0,
+ &requestSrcAddr, requestSrcAddrLen);
+ if (result == -1) {
+ ALOGW("%s sendto failed", __PRETTY_FUNCTION__);
+ }
+
+ endIndex = retry.size() - 1;
+ }
+
+ // send the retry packets
+ for (int i = startIndex; i <= endIndex; i++) {
+ const sp<TRTPPacket>& replyPacket = retry[i];
+
+ result = sendto(mSender->mSocket,
+ replyPacket->getPacket(),
+ replyPacket->getPacketLen(),
+ 0,
+ &requestSrcAddr,
+ requestSrcAddrLen);
+
+ if (result == -1) {
+ ALOGW("%s sendto failed", __PRETTY_FUNCTION__);
+ }
+ }
+}
+
+// Endpoint
+
+AAH_TXSender::Endpoint::Endpoint()
+ : addr(0)
+ , port(0) { }
+
+AAH_TXSender::Endpoint::Endpoint(uint32_t a, uint16_t p)
+ : addr(a)
+ , port(p) {}
+
+bool AAH_TXSender::Endpoint::operator<(const Endpoint& other) const {
+ return ((addr < other.addr) ||
+ (addr == other.addr && port < other.port));
+}
+
+// EndpointState
+
+AAH_TXSender::EndpointState::EndpointState(uint32_t _epoch)
+ : retry(kRetryBufferCapacity)
+ , playerRefCount(1)
+ , trtpSeqNumber(0)
+ , nextProgramID(0)
+ , epoch(_epoch) { }
+
+// CircularBuffer
+
+template <typename T>
+CircularBuffer<T>::CircularBuffer(size_t capacity)
+ : mCapacity(capacity)
+ , mHead(0)
+ , mTail(0)
+ , mFillCount(0) {
+ mBuffer = new T[capacity];
+}
+
+template <typename T>
+CircularBuffer<T>::~CircularBuffer() {
+ delete [] mBuffer;
+}
+
+template <typename T>
+void CircularBuffer<T>::push_back(const T& item) {
+ if (this->isFull()) {
+ this->pop_front();
+ }
+ mBuffer[mHead] = item;
+ mHead = (mHead + 1) % mCapacity;
+ mFillCount++;
+}
+
+template <typename T>
+void CircularBuffer<T>::pop_front() {
+ CHECK(!isEmpty());
+ mBuffer[mTail] = T();
+ mTail = (mTail + 1) % mCapacity;
+ mFillCount--;
+}
+
+template <typename T>
+size_t CircularBuffer<T>::size() const {
+ return mFillCount;
+}
+
+template <typename T>
+bool CircularBuffer<T>::isFull() const {
+ return (mFillCount == mCapacity);
+}
+
+template <typename T>
+bool CircularBuffer<T>::isEmpty() const {
+ return (mFillCount == 0);
+}
+
+template <typename T>
+const T& CircularBuffer<T>::itemAt(size_t index) const {
+ CHECK(index < mFillCount);
+ return mBuffer[(mTail + index) % mCapacity];
+}
+
+template <typename T>
+const T& CircularBuffer<T>::operator[](size_t index) const {
+ return itemAt(index);
+}
+
+} // namespace android
diff --git a/media/libaah_rtp/aah_tx_sender.h b/media/libaah_rtp/aah_tx_sender.h
new file mode 100644
index 0000000..74206c4
--- /dev/null
+++ b/media/libaah_rtp/aah_tx_sender.h
@@ -0,0 +1,162 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __AAH_TX_SENDER_H__
+#define __AAH_TX_SENDER_H__
+
+#include <media/stagefright/foundation/ALooper.h>
+#include <media/stagefright/foundation/AHandlerReflector.h>
+#include <utils/RefBase.h>
+#include <utils/threads.h>
+
+#include "aah_tx_packet.h"
+#include "pipe_event.h"
+
+namespace android {
+
+template <typename T> class CircularBuffer {
+ public:
+ CircularBuffer(size_t capacity);
+ ~CircularBuffer();
+ void push_back(const T& item);;
+ void pop_front();
+ size_t size() const;
+ bool isFull() const;
+ bool isEmpty() const;
+ const T& itemAt(size_t index) const;
+ const T& operator[](size_t index) const;
+
+ private:
+ T* mBuffer;
+ size_t mCapacity;
+ size_t mHead;
+ size_t mTail;
+ size_t mFillCount;
+
+ DISALLOW_EVIL_CONSTRUCTORS(CircularBuffer);
+};
+
+class AAH_TXSender : public virtual RefBase {
+ public:
+ ~AAH_TXSender();
+
+ static sp<AAH_TXSender> GetInstance();
+
+ ALooper::handler_id handlerID() { return mReflector->id(); }
+
+ // an IP address and port
+ struct Endpoint {
+ Endpoint();
+ Endpoint(uint32_t a, uint16_t p);
+ bool operator<(const Endpoint& other) const;
+
+ uint32_t addr;
+ uint16_t port;
+ };
+
+ uint16_t registerEndpoint(const Endpoint& endpoint);
+ void unregisterEndpoint(const Endpoint& endpoint);
+
+ enum {
+ kWhatSendPacket,
+ kWhatTrimRetryBuffers,
+ kWhatSendHeartbeats,
+ };
+
+ // fields for SendPacket messages
+ static const char* kSendPacketIPAddr;
+ static const char* kSendPacketPort;
+ static const char* kSendPacketTRTPPacket;
+
+ private:
+ AAH_TXSender();
+
+ static Mutex sLock;
+ static wp<AAH_TXSender> sInstance;
+ static uint32_t sNextEpoch;
+ static bool sNextEpochValid;
+
+ static uint32_t getNextEpoch();
+
+ typedef CircularBuffer<sp<TRTPPacket> > RetryBuffer;
+
+ // state maintained on a per-endpoint basis
+ struct EndpointState {
+ EndpointState(uint32_t epoch);
+ RetryBuffer retry;
+ int playerRefCount;
+ uint16_t trtpSeqNumber;
+ uint16_t nextProgramID;
+ uint32_t epoch;
+ };
+
+ friend class AHandlerReflector<AAH_TXSender>;
+ void onMessageReceived(const sp<AMessage>& msg);
+ void onSendPacket(const sp<AMessage>& msg);
+ void doSendPacket_l(const sp<TRTPPacket>& packet,
+ const Endpoint& endpoint);
+ void trimRetryBuffers();
+ void sendHeartbeats();
+ bool shouldSendHeartbeats_l();
+
+ sp<ALooper> mLooper;
+ sp<AHandlerReflector<AAH_TXSender> > mReflector;
+
+ int mSocket;
+ nsecs_t mLastSentPacketTime;
+
+ DefaultKeyedVector<Endpoint, EndpointState*> mEndpointMap;
+ Mutex mEndpointLock;
+
+ static const int kRetryTrimIntervalUs;
+ static const int kHeartbeatIntervalUs;
+ static const int kRetryBufferCapacity;
+ static const nsecs_t kHeartbeatTimeout;
+
+ class RetryReceiver : public Thread {
+ private:
+ friend class AAH_TXSender;
+
+ RetryReceiver(AAH_TXSender* sender);
+ virtual ~RetryReceiver();
+ virtual bool threadLoop();
+ void handleRetryRequest();
+
+ static const int kMaxReceiverPacketLen;
+ static const uint32_t kRetryRequestID;
+ static const uint32_t kFastStartRequestID;
+ static const uint32_t kRetryNakID;
+
+ AAH_TXSender* mSender;
+ PipeEvent mWakeupEvent;
+ };
+
+ sp<RetryReceiver> mRetryReceiver;
+
+ DISALLOW_EVIL_CONSTRUCTORS(AAH_TXSender);
+};
+
+struct RetryPacket {
+ uint32_t id;
+ uint32_t endpointIP;
+ uint16_t endpointPort;
+ uint16_t seqStart;
+ uint16_t seqEnd;
+} __attribute__((packed));
+
+} // namespace android
+
+#endif // __AAH_TX_SENDER_H__
diff --git a/media/libaah_rtp/pipe_event.cpp b/media/libaah_rtp/pipe_event.cpp
new file mode 100644
index 0000000..b8e6960
--- /dev/null
+++ b/media/libaah_rtp/pipe_event.cpp
@@ -0,0 +1,86 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "LibAAH_RTP"
+#include <utils/Log.h>
+
+#include <errno.h>
+#include <fcntl.h>
+#include <poll.h>
+#include <unistd.h>
+
+#include "pipe_event.h"
+
+namespace android {
+
+PipeEvent::PipeEvent() {
+ pipe_[0] = -1;
+ pipe_[1] = -1;
+
+ // Create the pipe.
+ if (pipe(pipe_) >= 0) {
+ // Set non-blocking mode on the read side of the pipe so we can
+ // easily drain it whenever we wakeup.
+ fcntl(pipe_[0], F_SETFL, O_NONBLOCK);
+ } else {
+ ALOGE("Failed to create pipe event %d %d %d",
+ pipe_[0], pipe_[1], errno);
+ pipe_[0] = -1;
+ pipe_[1] = -1;
+ }
+}
+
+PipeEvent::~PipeEvent() {
+ if (pipe_[0] >= 0) {
+ close(pipe_[0]);
+ }
+
+ if (pipe_[1] >= 0) {
+ close(pipe_[1]);
+ }
+}
+
+void PipeEvent::clearPendingEvents() {
+ char drain_buffer[16];
+ while (read(pipe_[0], drain_buffer, sizeof(drain_buffer)) > 0) {
+ // No body.
+ }
+}
+
+bool PipeEvent::wait(int timeout) {
+ struct pollfd wait_fd;
+
+ wait_fd.fd = getWakeupHandle();
+ wait_fd.events = POLLIN;
+ wait_fd.revents = 0;
+
+ int res = poll(&wait_fd, 1, timeout);
+
+ if (res < 0) {
+ ALOGE("Wait error in PipeEvent; sleeping to prevent overload!");
+ usleep(1000);
+ }
+
+ return (res > 0);
+}
+
+void PipeEvent::setEvent() {
+ char foo = 'q';
+ write(pipe_[1], &foo, 1);
+}
+
+} // namespace android
+
diff --git a/media/libaah_rtp/pipe_event.h b/media/libaah_rtp/pipe_event.h
new file mode 100644
index 0000000..e53b0fd
--- /dev/null
+++ b/media/libaah_rtp/pipe_event.h
@@ -0,0 +1,51 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __PIPE_EVENT_H__
+#define __PIPE_EVENT_H__
+
+#include <media/stagefright/foundation/ABase.h>
+
+namespace android {
+
+class PipeEvent {
+ public:
+ PipeEvent();
+ ~PipeEvent();
+
+ bool initCheck() const {
+ return ((pipe_[0] >= 0) && (pipe_[1] >= 0));
+ }
+
+ int getWakeupHandle() const { return pipe_[0]; }
+
+ // block until the event fires; returns true if the event fired and false if
+ // the wait timed out. Timeout is expressed in milliseconds; negative
+ // values mean wait forever.
+ bool wait(int timeout = -1);
+
+ void clearPendingEvents();
+ void setEvent();
+
+ private:
+ int pipe_[2];
+
+ DISALLOW_EVIL_CONSTRUCTORS(PipeEvent);
+};
+
+} // namespace android
+
+#endif // __PIPE_EVENT_H__
diff --git a/media/libeffects/data/audio_effects.conf b/media/libeffects/data/audio_effects.conf
index b8fa487..ce25bc8 100644
--- a/media/libeffects/data/audio_effects.conf
+++ b/media/libeffects/data/audio_effects.conf
@@ -50,11 +50,11 @@
}
volume {
library bundle
- uuid 119341a0-8469-11df-81f9- 0002a5d5c51b
+ uuid 119341a0-8469-11df-81f9-0002a5d5c51b
}
reverb_env_aux {
library reverb
- uuid 4a387fc0-8ab3-11df-8bad- 0002a5d5c51b
+ uuid 4a387fc0-8ab3-11df-8bad-0002a5d5c51b
}
reverb_env_ins {
library reverb
diff --git a/media/libeffects/downmix/Android.mk b/media/libeffects/downmix/Android.mk
new file mode 100644
index 0000000..95ca6fd
--- /dev/null
+++ b/media/libeffects/downmix/Android.mk
@@ -0,0 +1,28 @@
+LOCAL_PATH:= $(call my-dir)
+
+# Multichannel downmix effect library
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES:= \
+ EffectDownmix.c
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils
+
+LOCAL_MODULE:= libdownmix
+
+LOCAL_MODULE_TAGS := optional
+
+LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
+
+ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
+LOCAL_LDLIBS += -ldl
+endif
+
+LOCAL_C_INCLUDES := \
+ $(call include-path-for, audio-effects) \
+ $(call include-path-for, audio-utils)
+
+LOCAL_PRELINK_MODULE := false
+
+include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libeffects/downmix/EffectDownmix.c b/media/libeffects/downmix/EffectDownmix.c
new file mode 100644
index 0000000..a325172
--- /dev/null
+++ b/media/libeffects/downmix/EffectDownmix.c
@@ -0,0 +1,889 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "EffectDownmix"
+#define LOG_NDEBUG 0
+#include <cutils/log.h>
+#include <stdlib.h>
+#include <string.h>
+#include <stdbool.h>
+#include "EffectDownmix.h"
+
+#define MINUS_3_DB_IN_Q19_12 2896 // -3dB = 0.707 * 2^12 = 2896
+
+// effect_handle_t interface implementation for downmix effect
+const struct effect_interface_s gDownmixInterface = {
+ Downmix_Process,
+ Downmix_Command,
+ Downmix_GetDescriptor,
+ NULL /* no process_reverse function, no reference stream needed */
+};
+
+audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM = {
+ tag : AUDIO_EFFECT_LIBRARY_TAG,
+ version : EFFECT_LIBRARY_API_VERSION,
+ name : "Downmix Library",
+ implementor : "The Android Open Source Project",
+ query_num_effects : DownmixLib_QueryNumberEffects,
+ query_effect : DownmixLib_QueryEffect,
+ create_effect : DownmixLib_Create,
+ release_effect : DownmixLib_Release,
+ get_descriptor : DownmixLib_GetDescriptor,
+};
+
+
+// AOSP insert downmix UUID: 93f04452-e4fe-41cc-91f9-e475b6d1d69f
+static const effect_descriptor_t gDownmixDescriptor = {
+ EFFECT_UIID_DOWNMIX__, //type
+ {0x93f04452, 0xe4fe, 0x41cc, 0x91f9, {0xe4, 0x75, 0xb6, 0xd1, 0xd6, 0x9f}}, // uuid
+ EFFECT_CONTROL_API_VERSION,
+ EFFECT_FLAG_TYPE_INSERT | EFFECT_FLAG_INSERT_FIRST,
+ 0, //FIXME what value should be reported? // cpu load
+ 0, //FIXME what value should be reported? // memory usage
+ "Multichannel Downmix To Stereo", // human readable effect name
+ "The Android Open Source Project" // human readable effect implementor name
+};
+
+// gDescriptors contains pointers to all defined effect descriptor in this library
+static const effect_descriptor_t * const gDescriptors[] = {
+ &gDownmixDescriptor
+};
+
+// number of effects in this library
+const int kNbEffects = sizeof(gDescriptors) / sizeof(const effect_descriptor_t *);
+
+
+/*----------------------------------------------------------------------------
+ * Effect API implementation
+ *--------------------------------------------------------------------------*/
+
+/*--- Effect Library Interface Implementation ---*/
+
+int32_t DownmixLib_QueryNumberEffects(uint32_t *pNumEffects) {
+ ALOGV("DownmixLib_QueryNumberEffects()");
+ *pNumEffects = kNbEffects;
+ return 0;
+}
+
+int32_t DownmixLib_QueryEffect(uint32_t index, effect_descriptor_t *pDescriptor) {
+ ALOGV("DownmixLib_QueryEffect() index=%d", index);
+ if (pDescriptor == NULL) {
+ return -EINVAL;
+ }
+ if (index >= (uint32_t)kNbEffects) {
+ return -EINVAL;
+ }
+ memcpy(pDescriptor, gDescriptors[index], sizeof(effect_descriptor_t));
+ return 0;
+}
+
+
+int32_t DownmixLib_Create(const effect_uuid_t *uuid,
+ int32_t sessionId,
+ int32_t ioId,
+ effect_handle_t *pHandle) {
+ int ret;
+ int i;
+ downmix_module_t *module;
+ const effect_descriptor_t *desc;
+
+ ALOGV("DownmixLib_Create()");
+
+ if (pHandle == NULL || uuid == NULL) {
+ return -EINVAL;
+ }
+
+ for (i = 0 ; i < kNbEffects ; i++) {
+ desc = gDescriptors[i];
+ if (memcmp(uuid, &desc->uuid, sizeof(effect_uuid_t)) == 0) {
+ break;
+ }
+ }
+
+ if (i == kNbEffects) {
+ return -ENOENT;
+ }
+
+ module = malloc(sizeof(downmix_module_t));
+
+ module->itfe = &gDownmixInterface;
+
+ module->context.state = DOWNMIX_STATE_UNINITIALIZED;
+
+ ret = Downmix_Init(module);
+ if (ret < 0) {
+ ALOGW("DownmixLib_Create() init failed");
+ free(module);
+ return ret;
+ }
+
+ *pHandle = (effect_handle_t) module;
+
+ ALOGV("DownmixLib_Create() %p , size %d", module, sizeof(downmix_module_t));
+
+ return 0;
+}
+
+
+int32_t DownmixLib_Release(effect_handle_t handle) {
+ downmix_module_t *pDwmModule = (downmix_module_t *)handle;
+
+ ALOGV("DownmixLib_Release() %p", handle);
+ if (handle == NULL) {
+ return -EINVAL;
+ }
+
+ pDwmModule->context.state = DOWNMIX_STATE_UNINITIALIZED;
+
+ free(pDwmModule);
+ return 0;
+}
+
+
+int32_t DownmixLib_GetDescriptor(const effect_uuid_t *uuid, effect_descriptor_t *pDescriptor) {
+ ALOGV("DownmixLib_GetDescriptor()");
+ int i;
+
+ if (pDescriptor == NULL || uuid == NULL){
+ ALOGE("DownmixLib_Create() called with NULL pointer");
+ return -EINVAL;
+ }
+ ALOGV("DownmixLib_GetDescriptor() nb effects=%d", kNbEffects);
+ for (i = 0; i < kNbEffects; i++) {
+ ALOGV("DownmixLib_GetDescriptor() i=%d", i);
+ if (memcmp(uuid, &gDescriptors[i]->uuid, sizeof(effect_uuid_t)) == 0) {
+ memcpy(pDescriptor, gDescriptors[i], sizeof(effect_descriptor_t));
+ ALOGV("EffectGetDescriptor - UUID matched downmix type %d, UUID = %x",
+ i, gDescriptors[i]->uuid.timeLow);
+ return 0;
+ }
+ }
+
+ return -EINVAL;
+}
+
+
+/*--- Effect Control Interface Implementation ---*/
+
+static int Downmix_Process(effect_handle_t self,
+ audio_buffer_t *inBuffer, audio_buffer_t *outBuffer) {
+
+ downmix_object_t *pDownmixer;
+ int16_t *pSrc, *pDst;
+ downmix_module_t *pDwmModule = (downmix_module_t *)self;
+
+ if (pDwmModule == NULL) {
+ return -EINVAL;
+ }
+
+ if (inBuffer == NULL || inBuffer->raw == NULL ||
+ outBuffer == NULL || outBuffer->raw == NULL ||
+ inBuffer->frameCount != outBuffer->frameCount) {
+ return -EINVAL;
+ }
+
+ pDownmixer = (downmix_object_t*) &pDwmModule->context;
+
+ if (pDownmixer->state == DOWNMIX_STATE_UNINITIALIZED) {
+ ALOGE("Downmix_Process error: trying to use an uninitialized downmixer");
+ return -EINVAL;
+ } else if (pDownmixer->state == DOWNMIX_STATE_INITIALIZED) {
+ ALOGE("Downmix_Process error: trying to use a non-configured downmixer");
+ return -ENODATA;
+ }
+
+ pSrc = inBuffer->s16;
+ pDst = outBuffer->s16;
+ size_t numFrames = outBuffer->frameCount;
+
+ const bool accumulate =
+ (pDwmModule->config.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE);
+
+ switch(pDownmixer->type) {
+
+ case DOWNMIX_TYPE_STRIP:
+ if (accumulate) {
+ while (numFrames) {
+ pDst[0] = clamp16(pDst[0] + pSrc[0]);
+ pDst[1] = clamp16(pDst[1] + pSrc[1]);
+ pSrc += pDownmixer->input_channel_count;
+ pDst += 2;
+ numFrames--;
+ }
+ } else {
+ while (numFrames) {
+ pDst[0] = pSrc[0];
+ pDst[1] = pSrc[1];
+ pSrc += pDownmixer->input_channel_count;
+ pDst += 2;
+ numFrames--;
+ }
+ }
+ break;
+
+ case DOWNMIX_TYPE_FOLD:
+ // optimize for the common formats
+ switch(pDwmModule->config.inputCfg.channels) {
+ case AUDIO_CHANNEL_OUT_QUAD:
+ Downmix_foldFromQuad(pSrc, pDst, numFrames, accumulate);
+ break;
+ case AUDIO_CHANNEL_OUT_SURROUND:
+ Downmix_foldFromSurround(pSrc, pDst, numFrames, accumulate);
+ break;
+ case AUDIO_CHANNEL_OUT_5POINT1:
+ Downmix_foldFrom5Point1(pSrc, pDst, numFrames, accumulate);
+ break;
+ case AUDIO_CHANNEL_OUT_7POINT1:
+ Downmix_foldFrom7Point1(pSrc, pDst, numFrames, accumulate);
+ break;
+ default:
+ // FIXME implement generic downmix
+ ALOGE("Multichannel configurations other than quad, 4.0, 5.1 and 7.1 are not supported");
+ break;
+ }
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+
+static int Downmix_Command(effect_handle_t self, uint32_t cmdCode, uint32_t cmdSize,
+ void *pCmdData, uint32_t *replySize, void *pReplyData) {
+
+ downmix_module_t *pDwmModule = (downmix_module_t *) self;
+ downmix_object_t *pDownmixer;
+ int retsize;
+
+ if (pDwmModule == NULL || pDwmModule->context.state == DOWNMIX_STATE_UNINITIALIZED) {
+ return -EINVAL;
+ }
+
+ pDownmixer = (downmix_object_t*) &pDwmModule->context;
+
+ ALOGV("Downmix_Command command %d cmdSize %d",cmdCode, cmdSize);
+
+ switch (cmdCode) {
+ case EFFECT_CMD_INIT:
+ if (pReplyData == NULL || *replySize != sizeof(int)) {
+ return -EINVAL;
+ }
+ *(int *) pReplyData = Downmix_Init(pDwmModule);
+ break;
+
+ case EFFECT_CMD_SET_CONFIG:
+ if (pCmdData == NULL || cmdSize != sizeof(effect_config_t)
+ || pReplyData == NULL || *replySize != sizeof(int)) {
+ return -EINVAL;
+ }
+ *(int *) pReplyData = Downmix_Configure(pDwmModule,
+ (effect_config_t *)pCmdData, false);
+ break;
+
+ case EFFECT_CMD_RESET:
+ Downmix_Reset(pDownmixer, false);
+ break;
+
+ case EFFECT_CMD_GET_PARAM:
+ ALOGV("Downmix_Command EFFECT_CMD_GET_PARAM pCmdData %p, *replySize %d, pReplyData: %p",
+ pCmdData, *replySize, pReplyData);
+ if (pCmdData == NULL || cmdSize < (int)(sizeof(effect_param_t) + sizeof(int32_t)) ||
+ pReplyData == NULL ||
+ *replySize < (int) sizeof(effect_param_t) + 2 * sizeof(int32_t)) {
+ return -EINVAL;
+ }
+ effect_param_t *rep = (effect_param_t *) pReplyData;
+ memcpy(pReplyData, pCmdData, sizeof(effect_param_t) + sizeof(int32_t));
+ ALOGV("Downmix_Command EFFECT_CMD_GET_PARAM param %d, replySize %d",
+ *(int32_t *)rep->data, rep->vsize);
+ rep->status = Downmix_getParameter(pDownmixer, *(int32_t *)rep->data, &rep->vsize,
+ rep->data + sizeof(int32_t));
+ *replySize = sizeof(effect_param_t) + sizeof(int32_t) + rep->vsize;
+ break;
+
+ case EFFECT_CMD_SET_PARAM:
+ ALOGV("Downmix_Command EFFECT_CMD_SET_PARAM cmdSize %d pCmdData %p, *replySize %d, " \
+ "pReplyData %p", cmdSize, pCmdData, *replySize, pReplyData);
+ if (pCmdData == NULL || (cmdSize < (int)(sizeof(effect_param_t) + sizeof(int32_t)))
+ || pReplyData == NULL || *replySize != (int)sizeof(int32_t)) {
+ return -EINVAL;
+ }
+ effect_param_t *cmd = (effect_param_t *) pCmdData;
+ *(int *)pReplyData = Downmix_setParameter(pDownmixer, *(int32_t *)cmd->data,
+ cmd->vsize, cmd->data + sizeof(int32_t));
+ break;
+
+ case EFFECT_CMD_SET_PARAM_DEFERRED:
+ //FIXME implement
+ ALOGW("Downmix_Command command EFFECT_CMD_SET_PARAM_DEFERRED not supported, FIXME");
+ break;
+
+ case EFFECT_CMD_SET_PARAM_COMMIT:
+ //FIXME implement
+ ALOGW("Downmix_Command command EFFECT_CMD_SET_PARAM_COMMIT not supported, FIXME");
+ break;
+
+ case EFFECT_CMD_ENABLE:
+ if (pReplyData == NULL || *replySize != sizeof(int)) {
+ return -EINVAL;
+ }
+ if (pDownmixer->state != DOWNMIX_STATE_INITIALIZED) {
+ return -ENOSYS;
+ }
+ pDownmixer->state = DOWNMIX_STATE_ACTIVE;
+ ALOGV("EFFECT_CMD_ENABLE() OK");
+ *(int *)pReplyData = 0;
+ break;
+
+ case EFFECT_CMD_DISABLE:
+ if (pReplyData == NULL || *replySize != sizeof(int)) {
+ return -EINVAL;
+ }
+ if (pDownmixer->state != DOWNMIX_STATE_ACTIVE) {
+ return -ENOSYS;
+ }
+ pDownmixer->state = DOWNMIX_STATE_INITIALIZED;
+ ALOGV("EFFECT_CMD_DISABLE() OK");
+ *(int *)pReplyData = 0;
+ break;
+
+ case EFFECT_CMD_SET_DEVICE:
+ if (pCmdData == NULL || cmdSize != (int)sizeof(uint32_t)) {
+ return -EINVAL;
+ }
+ // FIXME change type if playing on headset vs speaker
+ ALOGV("Downmix_Command EFFECT_CMD_SET_DEVICE: 0x%08x", *(uint32_t *)pCmdData);
+ break;
+
+ case EFFECT_CMD_SET_VOLUME: {
+ // audio output is always stereo => 2 channel volumes
+ if (pCmdData == NULL || cmdSize != (int)sizeof(uint32_t) * 2) {
+ return -EINVAL;
+ }
+ // FIXME change volume
+ ALOGW("Downmix_Command command EFFECT_CMD_SET_VOLUME not supported, FIXME");
+ float left = (float)(*(uint32_t *)pCmdData) / (1 << 24);
+ float right = (float)(*((uint32_t *)pCmdData + 1)) / (1 << 24);
+ ALOGV("Downmix_Command EFFECT_CMD_SET_VOLUME: left %f, right %f ", left, right);
+ break;
+ }
+
+ case EFFECT_CMD_SET_AUDIO_MODE:
+ if (pCmdData == NULL || cmdSize != (int)sizeof(uint32_t)) {
+ return -EINVAL;
+ }
+ ALOGV("Downmix_Command EFFECT_CMD_SET_AUDIO_MODE: %d", *(uint32_t *)pCmdData);
+ break;
+
+ case EFFECT_CMD_SET_CONFIG_REVERSE:
+ case EFFECT_CMD_SET_INPUT_DEVICE:
+ // these commands are ignored by a downmix effect
+ break;
+
+ default:
+ ALOGW("Downmix_Command invalid command %d",cmdCode);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+
+int Downmix_GetDescriptor(effect_handle_t self, effect_descriptor_t *pDescriptor)
+{
+ downmix_module_t *pDwnmxModule = (downmix_module_t *) self;
+
+ if (pDwnmxModule == NULL ||
+ pDwnmxModule->context.state == DOWNMIX_STATE_UNINITIALIZED) {
+ return -EINVAL;
+ }
+
+ memcpy(pDescriptor, &gDownmixDescriptor, sizeof(effect_descriptor_t));
+
+ return 0;
+}
+
+
+/*----------------------------------------------------------------------------
+ * Downmix internal functions
+ *--------------------------------------------------------------------------*/
+
+/*----------------------------------------------------------------------------
+ * Downmix_Init()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Initialize downmix context and apply default parameters
+ *
+ * Inputs:
+ * pDwmModule pointer to downmix effect module
+ *
+ * Outputs:
+ *
+ * Returns:
+ * 0 indicates success
+ *
+ * Side Effects:
+ * updates:
+ * pDwmModule->context.type
+ * pDwmModule->context.apply_volume_correction
+ * pDwmModule->config.inputCfg
+ * pDwmModule->config.outputCfg
+ * pDwmModule->config.inputCfg.samplingRate
+ * pDwmModule->config.outputCfg.samplingRate
+ * pDwmModule->context.state
+ * doesn't set:
+ * pDwmModule->itfe
+ *
+ *----------------------------------------------------------------------------
+ */
+
+int Downmix_Init(downmix_module_t *pDwmModule) {
+
+ ALOGV("Downmix_Init module %p", pDwmModule);
+ int ret = 0;
+
+ memset(&pDwmModule->context, 0, sizeof(downmix_object_t));
+
+ pDwmModule->config.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
+ pDwmModule->config.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
+ pDwmModule->config.inputCfg.channels = AUDIO_CHANNEL_OUT_7POINT1;
+ pDwmModule->config.inputCfg.bufferProvider.getBuffer = NULL;
+ pDwmModule->config.inputCfg.bufferProvider.releaseBuffer = NULL;
+ pDwmModule->config.inputCfg.bufferProvider.cookie = NULL;
+ pDwmModule->config.inputCfg.mask = EFFECT_CONFIG_ALL;
+
+ pDwmModule->config.inputCfg.samplingRate = 44100;
+ pDwmModule->config.outputCfg.samplingRate = pDwmModule->config.inputCfg.samplingRate;
+
+ // set a default value for the access mode, but should be overwritten by caller
+ pDwmModule->config.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE;
+ pDwmModule->config.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
+ pDwmModule->config.outputCfg.channels = AUDIO_CHANNEL_OUT_STEREO;
+ pDwmModule->config.outputCfg.bufferProvider.getBuffer = NULL;
+ pDwmModule->config.outputCfg.bufferProvider.releaseBuffer = NULL;
+ pDwmModule->config.outputCfg.bufferProvider.cookie = NULL;
+ pDwmModule->config.outputCfg.mask = EFFECT_CONFIG_ALL;
+
+ ret = Downmix_Configure(pDwmModule, &pDwmModule->config, true);
+ if (ret != 0) {
+ ALOGV("Downmix_Init error %d on module %p", ret, pDwmModule);
+ } else {
+ pDwmModule->context.state = DOWNMIX_STATE_INITIALIZED;
+ }
+
+ return ret;
+}
+
+
+/*----------------------------------------------------------------------------
+ * Downmix_Configure()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Set input and output audio configuration.
+ *
+ * Inputs:
+ * pDwmModule pointer to downmix effect module
+ * pConfig pointer to effect_config_t structure containing input
+ * and output audio parameters configuration
+ * init true if called from init function
+ *
+ * Outputs:
+ *
+ * Returns:
+ * 0 indicates success
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+ */
+
+int Downmix_Configure(downmix_module_t *pDwmModule, effect_config_t *pConfig, bool init) {
+
+ downmix_object_t *pDownmixer = &pDwmModule->context;
+
+ // Check configuration compatibility with build options, and effect capabilities
+ if (pConfig->inputCfg.samplingRate != pConfig->outputCfg.samplingRate
+ || pConfig->outputCfg.channels != DOWNMIX_OUTPUT_CHANNELS
+ || pConfig->inputCfg.format != AUDIO_FORMAT_PCM_16_BIT
+ || pConfig->outputCfg.format != AUDIO_FORMAT_PCM_16_BIT) {
+ ALOGE("Downmix_Configure error: invalid config");
+ return -EINVAL;
+ }
+
+ memcpy(&pDwmModule->config, pConfig, sizeof(effect_config_t));
+
+ if (init) {
+ pDownmixer->type = DOWNMIX_TYPE_FOLD;
+ pDownmixer->apply_volume_correction = false;
+ pDownmixer->input_channel_count = 8; // matches default input of AUDIO_CHANNEL_OUT_7POINT1
+ } else {
+ // when configuring the effect, do not allow a blank channel mask
+ if (pConfig->inputCfg.channels == 0) {
+ ALOGE("Downmix_Configure error: input channel mask can't be 0");
+ return -EINVAL;
+ }
+ pDownmixer->input_channel_count = popcount(pConfig->inputCfg.channels);
+ }
+
+ Downmix_Reset(pDownmixer, init);
+
+ return 0;
+}
+
+
+/*----------------------------------------------------------------------------
+ * Downmix_Reset()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Reset internal states.
+ *
+ * Inputs:
+ * pDownmixer pointer to downmix context
+ * init true if called from init function
+ *
+ * Outputs:
+*
+ * Returns:
+ * 0 indicates success
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+ */
+
+int Downmix_Reset(downmix_object_t *pDownmixer, bool init) {
+ // nothing to do here
+ return 0;
+}
+
+
+/*----------------------------------------------------------------------------
+ * Downmix_setParameter()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Set a Downmix parameter
+ *
+ * Inputs:
+ * pDownmixer handle to instance data
+ * param parameter
+ * pValue pointer to parameter value
+ * size value size
+ *
+ * Outputs:
+ *
+ * Returns:
+ * 0 indicates success
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+ */
+int Downmix_setParameter(downmix_object_t *pDownmixer, int32_t param, size_t size, void *pValue) {
+
+ int16_t value16;
+ ALOGV("Downmix_setParameter, context %p, param %d, value16 %d, value32 %d",
+ pDownmixer, param, *(int16_t *)pValue, *(int32_t *)pValue);
+
+ switch (param) {
+
+ case DOWNMIX_PARAM_TYPE:
+ if (size != sizeof(downmix_type_t)) {
+ ALOGE("Downmix_setParameter(DOWNMIX_PARAM_TYPE) invalid size %d, should be %d",
+ size, sizeof(downmix_type_t));
+ return -EINVAL;
+ }
+ value16 = *(int16_t *)pValue;
+ ALOGV("set DOWNMIX_PARAM_TYPE, type %d", value16);
+ if (!((value16 > DOWNMIX_TYPE_INVALID) && (value16 < DOWNMIX_TYPE_LAST))) {
+ ALOGE("Downmix_setParameter invalid DOWNMIX_PARAM_TYPE value %d", value16);
+ return -EINVAL;
+ } else {
+ pDownmixer->type = (downmix_type_t) value16;
+ break;
+
+ default:
+ ALOGE("Downmix_setParameter unknown parameter %d", param);
+ return -EINVAL;
+ }
+}
+
+ return 0;
+} /* end Downmix_setParameter */
+
+
+/*----------------------------------------------------------------------------
+ * Downmix_getParameter()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Get a Downmix parameter
+ *
+ * Inputs:
+ * pDownmixer handle to instance data
+ * param parameter
+ * pValue pointer to variable to hold retrieved value
+ * pSize pointer to value size: maximum size as input
+ *
+ * Outputs:
+ * *pValue updated with parameter value
+ * *pSize updated with actual value size
+ *
+ * Returns:
+ * 0 indicates success
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+ */
+int Downmix_getParameter(downmix_object_t *pDownmixer, int32_t param, size_t *pSize, void *pValue) {
+ int16_t *pValue16;
+
+ switch (param) {
+
+ case DOWNMIX_PARAM_TYPE:
+ if (*pSize < sizeof(int16_t)) {
+ ALOGE("Downmix_getParameter invalid parameter size %d for DOWNMIX_PARAM_TYPE", *pSize);
+ return -EINVAL;
+ }
+ pValue16 = (int16_t *)pValue;
+ *pValue16 = (int16_t) pDownmixer->type;
+ *pSize = sizeof(int16_t);
+ ALOGV("Downmix_getParameter DOWNMIX_PARAM_TYPE is %d", *pValue16);
+ break;
+
+ default:
+ ALOGE("Downmix_getParameter unknown parameter %d", param);
+ return -EINVAL;
+ }
+
+ return 0;
+} /* end Downmix_getParameter */
+
+
+/*----------------------------------------------------------------------------
+ * Downmix_foldFromQuad()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * downmix a quad signal to stereo
+ *
+ * Inputs:
+ * pSrc quad audio samples to downmix
+ * numFrames the number of quad frames to downmix
+ *
+ * Outputs:
+ * pDst downmixed stereo audio samples
+ *
+ *----------------------------------------------------------------------------
+ */
+void Downmix_foldFromQuad(int16_t *pSrc, int16_t*pDst, size_t numFrames, bool accumulate) {
+ // sample at index 0 is FL
+ // sample at index 1 is FR
+ // sample at index 2 is RL
+ // sample at index 3 is RR
+ if (accumulate) {
+ while (numFrames) {
+ // FL + RL
+ pDst[0] = clamp16(pDst[0] + pSrc[0] + pSrc[2]);
+ // FR + RR
+ pDst[1] = clamp16(pDst[1] + pSrc[1] + pSrc[3]);
+ pSrc += 4;
+ pDst += 2;
+ numFrames--;
+ }
+ } else { // same code as above but without adding and clamping pDst[i] to itself
+ while (numFrames) {
+ // FL + RL
+ pDst[0] = clamp16(pSrc[0] + pSrc[2]);
+ // FR + RR
+ pDst[1] = clamp16(pSrc[1] + pSrc[3]);
+ pSrc += 4;
+ pDst += 2;
+ numFrames--;
+ }
+ }
+}
+
+
+/*----------------------------------------------------------------------------
+ * Downmix_foldFromSurround()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * downmix a "surround sound" (mono rear) signal to stereo
+ *
+ * Inputs:
+ * pSrc surround signal to downmix
+ * numFrames the number of surround frames to downmix
+ *
+ * Outputs:
+ * pDst downmixed stereo audio samples
+ *
+ *----------------------------------------------------------------------------
+ */
+void Downmix_foldFromSurround(int16_t *pSrc, int16_t*pDst, size_t numFrames, bool accumulate) {
+ int32_t lt, rt, centerPlusRearContrib; // samples in Q19.12 format
+ // sample at index 0 is FL
+ // sample at index 1 is FR
+ // sample at index 2 is FC
+ // sample at index 3 is RC
+ if (accumulate) {
+ while (numFrames) {
+ // centerPlusRearContrib = FC(-3dB) + RC(-3dB)
+ centerPlusRearContrib = (pSrc[2] * MINUS_3_DB_IN_Q19_12) + (pSrc[3] * MINUS_3_DB_IN_Q19_12);
+ // FL + centerPlusRearContrib
+ lt = (pSrc[0] << 12) + centerPlusRearContrib;
+ // FR + centerPlusRearContrib
+ rt = (pSrc[1] << 12) + centerPlusRearContrib;
+ pDst[0] = clamp16(pDst[0] + (lt >> 12));
+ pDst[1] = clamp16(pDst[1] + (rt >> 12));
+ pSrc += 4;
+ pDst += 2;
+ numFrames--;
+ }
+ } else { // same code as above but without adding and clamping pDst[i] to itself
+ while (numFrames) {
+ // centerPlusRearContrib = FC(-3dB) + RC(-3dB)
+ centerPlusRearContrib = (pSrc[2] * MINUS_3_DB_IN_Q19_12) + (pSrc[3] * MINUS_3_DB_IN_Q19_12);
+ // FL + centerPlusRearContrib
+ lt = (pSrc[0] << 12) + centerPlusRearContrib;
+ // FR + centerPlusRearContrib
+ rt = (pSrc[1] << 12) + centerPlusRearContrib;
+ pDst[0] = clamp16(lt >> 12);
+ pDst[1] = clamp16(rt >> 12);
+ pSrc += 4;
+ pDst += 2;
+ numFrames--;
+ }
+ }
+}
+
+
+/*----------------------------------------------------------------------------
+ * Downmix_foldFrom5Point1()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * downmix a 5.1 signal to stereo
+ *
+ * Inputs:
+ * pSrc 5.1 audio samples to downmix
+ * numFrames the number of 5.1 frames to downmix
+ *
+ * Outputs:
+ * pDst downmixed stereo audio samples
+ *
+ *----------------------------------------------------------------------------
+ */
+void Downmix_foldFrom5Point1(int16_t *pSrc, int16_t*pDst, size_t numFrames, bool accumulate) {
+ int32_t lt, rt, centerPlusLfeContrib; // samples in Q19.12 format
+ // sample at index 0 is FL
+ // sample at index 1 is FR
+ // sample at index 2 is FC
+ // sample at index 3 is LFE
+ // sample at index 4 is RL
+ // sample at index 5 is RR
+ if (accumulate) {
+ while (numFrames) {
+ // centerPlusLfeContrib = FC(-3dB) + LFE(-3dB)
+ centerPlusLfeContrib = (pSrc[2] * MINUS_3_DB_IN_Q19_12)
+ + (pSrc[3] * MINUS_3_DB_IN_Q19_12);
+ // FL + centerPlusLfeContrib + RL
+ lt = (pSrc[0] << 12) + centerPlusLfeContrib + (pSrc[4] << 12);
+ // FR + centerPlusLfeContrib + RR
+ rt = (pSrc[1] << 12) + centerPlusLfeContrib + (pSrc[5] << 12);
+ pDst[0] = clamp16(pDst[0] + (lt >> 12));
+ pDst[1] = clamp16(pDst[1] + (rt >> 12));
+ pSrc += 6;
+ pDst += 2;
+ numFrames--;
+ }
+ } else { // same code as above but without adding and clamping pDst[i] to itself
+ while (numFrames) {
+ // centerPlusLfeContrib = FC(-3dB) + LFE(-3dB)
+ centerPlusLfeContrib = (pSrc[2] * MINUS_3_DB_IN_Q19_12)
+ + (pSrc[3] * MINUS_3_DB_IN_Q19_12);
+ // FL + centerPlusLfeContrib + RL
+ lt = (pSrc[0] << 12) + centerPlusLfeContrib + (pSrc[4] << 12);
+ // FR + centerPlusLfeContrib + RR
+ rt = (pSrc[1] << 12) + centerPlusLfeContrib + (pSrc[5] << 12);
+ pDst[0] = clamp16(lt >> 12);
+ pDst[1] = clamp16(rt >> 12);
+ pSrc += 6;
+ pDst += 2;
+ numFrames--;
+ }
+ }
+}
+
+
+/*----------------------------------------------------------------------------
+ * Downmix_foldFrom7Point1()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * downmix a 7.1 signal to stereo
+ *
+ * Inputs:
+ * pSrc 7.1 audio samples to downmix
+ * numFrames the number of 7.1 frames to downmix
+ *
+ * Outputs:
+ * pDst downmixed stereo audio samples
+ *
+ *----------------------------------------------------------------------------
+ */
+void Downmix_foldFrom7Point1(int16_t *pSrc, int16_t*pDst, size_t numFrames, bool accumulate) {
+ int32_t lt, rt, centerPlusLfeContrib; // samples in Q19.12 format
+ // sample at index 0 is FL
+ // sample at index 1 is FR
+ // sample at index 2 is FC
+ // sample at index 3 is LFE
+ // sample at index 4 is RL
+ // sample at index 5 is RR
+ // sample at index 6 is SL
+ // sample at index 7 is SR
+ if (accumulate) {
+ while (numFrames) {
+ // centerPlusLfeContrib = FC(-3dB) + LFE(-3dB)
+ centerPlusLfeContrib = (pSrc[2] * MINUS_3_DB_IN_Q19_12)
+ + (pSrc[3] * MINUS_3_DB_IN_Q19_12);
+ // FL + centerPlusLfeContrib + SL + RL
+ lt = (pSrc[0] << 12) + centerPlusLfeContrib + (pSrc[6] << 12) + (pSrc[4] << 12);
+ // FR + centerPlusLfeContrib + SR + RR
+ rt = (pSrc[1] << 12) + centerPlusLfeContrib + (pSrc[7] << 12) + (pSrc[5] << 12);
+ pDst[0] = clamp16(lt >> 12);
+ pDst[1] = clamp16(rt >> 12);
+ pSrc += 8;
+ pDst += 2;
+ numFrames--;
+ }
+ } else { // same code as above but without adding and clamping pDst[i] to itself
+ while (numFrames) {
+ // centerPlusLfeContrib = FC(-3dB) + LFE(-3dB)
+ centerPlusLfeContrib = (pSrc[2] * MINUS_3_DB_IN_Q19_12)
+ + (pSrc[3] * MINUS_3_DB_IN_Q19_12);
+ // FL + centerPlusLfeContrib + SL + RL
+ lt = (pSrc[0] << 12) + centerPlusLfeContrib + (pSrc[6] << 12) + (pSrc[4] << 12);
+ // FR + centerPlusLfeContrib + SR + RR
+ rt = (pSrc[1] << 12) + centerPlusLfeContrib + (pSrc[7] << 12) + (pSrc[5] << 12);
+ pDst[0] = clamp16(pDst[0] + (lt >> 12));
+ pDst[1] = clamp16(pDst[1] + (rt >> 12));
+ pSrc += 8;
+ pDst += 2;
+ numFrames--;
+ }
+ }
+}
+
diff --git a/media/libeffects/downmix/EffectDownmix.h b/media/libeffects/downmix/EffectDownmix.h
new file mode 100644
index 0000000..4176b5a
--- /dev/null
+++ b/media/libeffects/downmix/EffectDownmix.h
@@ -0,0 +1,96 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_EFFECTDOWNMIX_H_
+#define ANDROID_EFFECTDOWNMIX_H_
+
+#include <audio_effects/effect_downmix.h>
+#include <audio_utils/primitives.h>
+#include <system/audio.h>
+
+/*------------------------------------
+ * definitions
+ *------------------------------------
+*/
+
+#define DOWNMIX_OUTPUT_CHANNELS AUDIO_CHANNEL_OUT_STEREO
+
+typedef enum {
+ DOWNMIX_STATE_UNINITIALIZED,
+ DOWNMIX_STATE_INITIALIZED,
+ DOWNMIX_STATE_ACTIVE,
+} downmix_state_t;
+
+/* parameters for each downmixer */
+typedef struct {
+ downmix_state_t state;
+ downmix_type_t type;
+ bool apply_volume_correction;
+ uint8_t input_channel_count;
+} downmix_object_t;
+
+
+typedef struct downmix_module_s {
+ const struct effect_interface_s *itfe;
+ effect_config_t config;
+ downmix_object_t context;
+} downmix_module_t;
+
+
+/*------------------------------------
+ * Effect API
+ *------------------------------------
+*/
+int32_t DownmixLib_QueryNumberEffects(uint32_t *pNumEffects);
+int32_t DownmixLib_QueryEffect(uint32_t index,
+ effect_descriptor_t *pDescriptor);
+int32_t DownmixLib_Create(const effect_uuid_t *uuid,
+ int32_t sessionId,
+ int32_t ioId,
+ effect_handle_t *pHandle);
+int32_t DownmixLib_Release(effect_handle_t handle);
+int32_t DownmixLib_GetDescriptor(const effect_uuid_t *uuid,
+ effect_descriptor_t *pDescriptor);
+
+static int Downmix_Process(effect_handle_t self,
+ audio_buffer_t *inBuffer,
+ audio_buffer_t *outBuffer);
+static int Downmix_Command(effect_handle_t self,
+ uint32_t cmdCode,
+ uint32_t cmdSize,
+ void *pCmdData,
+ uint32_t *replySize,
+ void *pReplyData);
+static int Downmix_GetDescriptor(effect_handle_t self,
+ effect_descriptor_t *pDescriptor);
+
+
+/*------------------------------------
+ * internal functions
+ *------------------------------------
+*/
+int Downmix_Init(downmix_module_t *pDwmModule);
+int Downmix_Configure(downmix_module_t *pDwmModule, effect_config_t *pConfig, bool init);
+int Downmix_Reset(downmix_object_t *pDownmixer, bool init);
+int Downmix_setParameter(downmix_object_t *pDownmixer, int32_t param, size_t size, void *pValue);
+int Downmix_getParameter(downmix_object_t *pDownmixer, int32_t param, size_t *pSize, void *pValue);
+
+void Downmix_foldFromQuad(int16_t *pSrc, int16_t*pDst, size_t numFrames, bool accumulate);
+void Downmix_foldFromSurround(int16_t *pSrc, int16_t*pDst, size_t numFrames, bool accumulate);
+void Downmix_foldFrom5Point1(int16_t *pSrc, int16_t*pDst, size_t numFrames, bool accumulate);
+void Downmix_foldFrom7Point1(int16_t *pSrc, int16_t*pDst, size_t numFrames, bool accumulate);
+
+#endif /*ANDROID_EFFECTDOWNMIX_H_*/
diff --git a/media/libeffects/factory/Android.mk b/media/libeffects/factory/Android.mk
index 2f2b974..6e69151 100644
--- a/media/libeffects/factory/Android.mk
+++ b/media/libeffects/factory/Android.mk
@@ -15,6 +15,6 @@
LOCAL_SHARED_LIBRARIES += libdl
LOCAL_C_INCLUDES := \
- system/media/audio_effects/include
+ $(call include-path-for, audio-effects)
include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libeffects/factory/EffectsFactory.c b/media/libeffects/factory/EffectsFactory.c
index 9f6599f..59cd9e3 100644
--- a/media/libeffects/factory/EffectsFactory.c
+++ b/media/libeffects/factory/EffectsFactory.c
@@ -53,8 +53,8 @@
static lib_entry_t *getLibrary(const char *path);
static void resetEffectEnumeration();
static uint32_t updateNumEffects();
-static int findEffect(effect_uuid_t *type,
- effect_uuid_t *uuid,
+static int findEffect(const effect_uuid_t *type,
+ const effect_uuid_t *uuid,
lib_entry_t **lib,
effect_descriptor_t **desc);
static void dumpEffectDescriptor(effect_descriptor_t *desc, char *str, size_t len);
@@ -236,7 +236,7 @@
return ret;
}
-int EffectGetDescriptor(effect_uuid_t *uuid, effect_descriptor_t *pDescriptor)
+int EffectGetDescriptor(const effect_uuid_t *uuid, effect_descriptor_t *pDescriptor)
{
lib_entry_t *l = NULL;
effect_descriptor_t *d = NULL;
@@ -257,7 +257,7 @@
return ret;
}
-int EffectCreate(effect_uuid_t *uuid, int32_t sessionId, int32_t ioId, effect_handle_t *pHandle)
+int EffectCreate(const effect_uuid_t *uuid, int32_t sessionId, int32_t ioId, effect_handle_t *pHandle)
{
list_elem_t *e = gLibraryList;
lib_entry_t *l = NULL;
@@ -372,7 +372,7 @@
return ret;
}
-int EffectIsNullUuid(effect_uuid_t *uuid)
+int EffectIsNullUuid(const effect_uuid_t *uuid)
{
if (memcmp(uuid, EFFECT_UUID_NULL, sizeof(effect_uuid_t))) {
return 0;
@@ -628,8 +628,8 @@
return cnt;
}
-int findEffect(effect_uuid_t *type,
- effect_uuid_t *uuid,
+int findEffect(const effect_uuid_t *type,
+ const effect_uuid_t *uuid,
lib_entry_t **lib,
effect_descriptor_t **desc)
{
diff --git a/media/libeffects/lvm/wrapper/Android.mk b/media/libeffects/lvm/wrapper/Android.mk
index f097dd0..4313424 100644
--- a/media/libeffects/lvm/wrapper/Android.mk
+++ b/media/libeffects/lvm/wrapper/Android.mk
@@ -26,7 +26,7 @@
$(LOCAL_PATH)/Bundle \
$(LOCAL_PATH)/../lib/Common/lib/ \
$(LOCAL_PATH)/../lib/Bundle/lib/ \
- system/media/audio_effects/include
+ $(call include-path-for, audio-effects)
include $(BUILD_SHARED_LIBRARY)
@@ -55,6 +55,6 @@
$(LOCAL_PATH)/Reverb \
$(LOCAL_PATH)/../lib/Common/lib/ \
$(LOCAL_PATH)/../lib/Reverb/lib/ \
- system/media/audio_effects/include
+ $(call include-path-for, audio-effects)
-include $(BUILD_SHARED_LIBRARY)
\ No newline at end of file
+include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
index 62be78c..ca93ce5 100644
--- a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
+++ b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
@@ -24,7 +24,7 @@
#include <stdlib.h>
#include <string.h>
#include <new>
-#include <EffectBundle.h>
+#include "EffectBundle.h"
// effect_handle_t interface implementation for bass boost
@@ -133,7 +133,8 @@
int LvmEffect_enable (EffectContext *pContext);
int LvmEffect_disable (EffectContext *pContext);
void LvmEffect_free (EffectContext *pContext);
-int Effect_configure (EffectContext *pContext, effect_config_t *pConfig);
+int Effect_setConfig (EffectContext *pContext, effect_config_t *pConfig);
+void Effect_getConfig (EffectContext *pContext, effect_config_t *pConfig);
int BassBoost_setParameter (EffectContext *pContext, void *pParam, void *pValue);
int BassBoost_getParameter (EffectContext *pContext,
void *pParam,
@@ -194,7 +195,7 @@
return 0;
} /* end EffectQueryEffect */
-extern "C" int EffectCreate(effect_uuid_t *uuid,
+extern "C" int EffectCreate(const effect_uuid_t *uuid,
int32_t sessionId,
int32_t ioId,
effect_handle_t *pHandle){
@@ -470,7 +471,7 @@
} /* end EffectRelease */
-extern "C" int EffectGetDescriptor(effect_uuid_t *uuid,
+extern "C" int EffectGetDescriptor(const effect_uuid_t *uuid,
effect_descriptor_t *pDescriptor) {
const effect_descriptor_t *desc = NULL;
@@ -936,7 +937,7 @@
} /* end LvmEffect_free */
//----------------------------------------------------------------------------
-// Effect_configure()
+// Effect_setConfig()
//----------------------------------------------------------------------------
// Purpose: Set input and output audio configuration.
//
@@ -949,9 +950,9 @@
//
//----------------------------------------------------------------------------
-int Effect_configure(EffectContext *pContext, effect_config_t *pConfig){
+int Effect_setConfig(EffectContext *pContext, effect_config_t *pConfig){
LVM_Fs_en SampleRate;
- //ALOGV("\tEffect_configure start");
+ //ALOGV("\tEffect_setConfig start");
CHECK_ARG(pContext != NULL);
CHECK_ARG(pConfig != NULL);
@@ -992,7 +993,7 @@
pContext->pBundledContext->SamplesPerSecond = 48000*2; // 2 secs Stereo
break;
default:
- ALOGV("\tEffect_Configure invalid sampling rate %d", pConfig->inputCfg.samplingRate);
+ ALOGV("\tEffect_setConfig invalid sampling rate %d", pConfig->inputCfg.samplingRate);
return -EINVAL;
}
@@ -1001,28 +1002,47 @@
LVM_ControlParams_t ActiveParams;
LVM_ReturnStatus_en LvmStatus = LVM_SUCCESS;
- ALOGV("\tEffect_configure change sampling rate to %d", SampleRate);
+ ALOGV("\tEffect_setConfig change sampling rate to %d", SampleRate);
/* Get the current settings */
LvmStatus = LVM_GetControlParameters(pContext->pBundledContext->hInstance,
&ActiveParams);
- LVM_ERROR_CHECK(LvmStatus, "LVM_GetControlParameters", "Effect_configure")
+ LVM_ERROR_CHECK(LvmStatus, "LVM_GetControlParameters", "Effect_setConfig")
if(LvmStatus != LVM_SUCCESS) return -EINVAL;
LvmStatus = LVM_SetControlParameters(pContext->pBundledContext->hInstance, &ActiveParams);
- LVM_ERROR_CHECK(LvmStatus, "LVM_SetControlParameters", "Effect_configure")
- ALOGV("\tEffect_configure Succesfully called LVM_SetControlParameters\n");
+ LVM_ERROR_CHECK(LvmStatus, "LVM_SetControlParameters", "Effect_setConfig")
+ ALOGV("\tEffect_setConfig Succesfully called LVM_SetControlParameters\n");
pContext->pBundledContext->SampleRate = SampleRate;
}else{
- //ALOGV("\tEffect_configure keep sampling rate at %d", SampleRate);
+ //ALOGV("\tEffect_setConfig keep sampling rate at %d", SampleRate);
}
- //ALOGV("\tEffect_configure End....");
+ //ALOGV("\tEffect_setConfig End....");
return 0;
-} /* end Effect_configure */
+} /* end Effect_setConfig */
+
+//----------------------------------------------------------------------------
+// Effect_getConfig()
+//----------------------------------------------------------------------------
+// Purpose: Get input and output audio configuration.
+//
+// Inputs:
+// pContext: effect engine context
+// pConfig: pointer to effect_config_t structure holding input and output
+// configuration parameters
+//
+// Outputs:
+//
+//----------------------------------------------------------------------------
+
+void Effect_getConfig(EffectContext *pContext, effect_config_t *pConfig)
+{
+ memcpy(pConfig, &pContext->config, sizeof(effect_config_t));
+} /* end Effect_getConfig */
//----------------------------------------------------------------------------
// BassGetStrength()
@@ -2778,23 +2798,34 @@
}
break;
- case EFFECT_CMD_CONFIGURE:
- //ALOGV("\tEffect_command cmdCode Case: EFFECT_CMD_CONFIGURE start");
+ case EFFECT_CMD_SET_CONFIG:
+ //ALOGV("\tEffect_command cmdCode Case: EFFECT_CMD_SET_CONFIG start");
if (pCmdData == NULL||
cmdSize != sizeof(effect_config_t)||
pReplyData == NULL||
*replySize != sizeof(int)){
ALOGV("\tLVM_ERROR : Effect_command cmdCode Case: "
- "EFFECT_CMD_CONFIGURE: ERROR");
+ "EFFECT_CMD_SET_CONFIG: ERROR");
return -EINVAL;
}
- *(int *) pReplyData = android::Effect_configure(pContext, (effect_config_t *) pCmdData);
- //ALOGV("\tEffect_command cmdCode Case: EFFECT_CMD_CONFIGURE end");
+ *(int *) pReplyData = android::Effect_setConfig(pContext, (effect_config_t *) pCmdData);
+ //ALOGV("\tEffect_command cmdCode Case: EFFECT_CMD_SET_CONFIG end");
+ break;
+
+ case EFFECT_CMD_GET_CONFIG:
+ if (pReplyData == NULL ||
+ *replySize != sizeof(effect_config_t)) {
+ ALOGV("\tLVM_ERROR : Effect_command cmdCode Case: "
+ "EFFECT_CMD_GET_CONFIG: ERROR");
+ return -EINVAL;
+ }
+
+ android::Effect_getConfig(pContext, (effect_config_t *)pReplyData);
break;
case EFFECT_CMD_RESET:
//ALOGV("\tEffect_command cmdCode Case: EFFECT_CMD_RESET start");
- android::Effect_configure(pContext, &pContext->config);
+ android::Effect_setConfig(pContext, &pContext->config);
//ALOGV("\tEffect_command cmdCode Case: EFFECT_CMD_RESET end");
break;
@@ -3078,20 +3109,20 @@
if (pContext->pBundledContext->bBassEnabled == LVM_TRUE) {
ALOGV("\tEFFECT_CMD_SET_DEVICE disable LVM_BASS_BOOST %d",
- *(int32_t *)pCmdData);
+ *(int32_t *)pCmdData);
android::LvmEffect_disable(pContext);
}
pContext->pBundledContext->bBassTempDisabled = LVM_TRUE;
} else {
ALOGV("\tEFFECT_CMD_SET_DEVICE device is valid for LVM_BASS_BOOST %d",
- *(int32_t *)pCmdData);
+ *(int32_t *)pCmdData);
// If a device supports bassboost and the effect has been temporarily disabled
// previously then re-enable it
if (pContext->pBundledContext->bBassEnabled == LVM_TRUE) {
ALOGV("\tEFFECT_CMD_SET_DEVICE re-enable LVM_BASS_BOOST %d",
- *(int32_t *)pCmdData);
+ *(int32_t *)pCmdData);
android::LvmEffect_enable(pContext);
}
pContext->pBundledContext->bBassTempDisabled = LVM_FALSE;
diff --git a/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp b/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp
index 1825aab..9599dcc 100755
--- a/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp
+++ b/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp
@@ -24,8 +24,9 @@
#include <stdlib.h>
#include <string.h>
#include <new>
-#include <EffectReverb.h>
-#include <LVREV.h>
+#include "EffectReverb.h"
+// from Reverb/lib
+#include "LVREV.h"
// effect_handle_t interface implementation for reverb
extern "C" const struct effect_interface_s gReverbInterface;
@@ -175,7 +176,8 @@
//--- local function prototypes
int Reverb_init (ReverbContext *pContext);
void Reverb_free (ReverbContext *pContext);
-int Reverb_configure (ReverbContext *pContext, effect_config_t *pConfig);
+int Reverb_setConfig (ReverbContext *pContext, effect_config_t *pConfig);
+void Reverb_getConfig (ReverbContext *pContext, effect_config_t *pConfig);
int Reverb_setParameter (ReverbContext *pContext, void *pParam, void *pValue);
int Reverb_getParameter (ReverbContext *pContext,
void *pParam,
@@ -209,7 +211,7 @@
return 0;
} /* end EffectQueryEffect */
-extern "C" int EffectCreate(effect_uuid_t *uuid,
+extern "C" int EffectCreate(const effect_uuid_t *uuid,
int32_t sessionId,
int32_t ioId,
effect_handle_t *pHandle){
@@ -316,7 +318,7 @@
return 0;
} /* end EffectRelease */
-extern "C" int EffectGetDescriptor(effect_uuid_t *uuid,
+extern "C" int EffectGetDescriptor(const effect_uuid_t *uuid,
effect_descriptor_t *pDescriptor) {
int i;
int length = sizeof(gDescriptors) / sizeof(const effect_descriptor_t *);
@@ -609,7 +611,7 @@
} /* end Reverb_free */
//----------------------------------------------------------------------------
-// Reverb_configure()
+// Reverb_setConfig()
//----------------------------------------------------------------------------
// Purpose: Set input and output audio configuration.
//
@@ -622,9 +624,9 @@
//
//----------------------------------------------------------------------------
-int Reverb_configure(ReverbContext *pContext, effect_config_t *pConfig){
+int Reverb_setConfig(ReverbContext *pContext, effect_config_t *pConfig){
LVM_Fs_en SampleRate;
- //ALOGV("\tReverb_configure start");
+ //ALOGV("\tReverb_setConfig start");
CHECK_ARG(pContext != NULL);
CHECK_ARG(pConfig != NULL);
@@ -642,7 +644,7 @@
return -EINVAL;
}
- //ALOGV("\tReverb_configure calling memcpy");
+ //ALOGV("\tReverb_setConfig calling memcpy");
memcpy(&pContext->config, pConfig, sizeof(effect_config_t));
@@ -666,7 +668,7 @@
SampleRate = LVM_FS_48000;
break;
default:
- ALOGV("\rReverb_Configure invalid sampling rate %d", pConfig->inputCfg.samplingRate);
+ ALOGV("\rReverb_setConfig invalid sampling rate %d", pConfig->inputCfg.samplingRate);
return -EINVAL;
}
@@ -675,28 +677,46 @@
LVREV_ControlParams_st ActiveParams;
LVREV_ReturnStatus_en LvmStatus = LVREV_SUCCESS;
- //ALOGV("\tReverb_configure change sampling rate to %d", SampleRate);
+ //ALOGV("\tReverb_setConfig change sampling rate to %d", SampleRate);
/* Get the current settings */
LvmStatus = LVREV_GetControlParameters(pContext->hInstance,
&ActiveParams);
- LVM_ERROR_CHECK(LvmStatus, "LVREV_GetControlParameters", "Reverb_configure")
+ LVM_ERROR_CHECK(LvmStatus, "LVREV_GetControlParameters", "Reverb_setConfig")
if(LvmStatus != LVREV_SUCCESS) return -EINVAL;
LvmStatus = LVREV_SetControlParameters(pContext->hInstance, &ActiveParams);
- LVM_ERROR_CHECK(LvmStatus, "LVREV_SetControlParameters", "Reverb_configure")
- //ALOGV("\tReverb_configure Succesfully called LVREV_SetControlParameters\n");
+ LVM_ERROR_CHECK(LvmStatus, "LVREV_SetControlParameters", "Reverb_setConfig")
+ //ALOGV("\tReverb_setConfig Succesfully called LVREV_SetControlParameters\n");
}else{
- //ALOGV("\tReverb_configure keep sampling rate at %d", SampleRate);
+ //ALOGV("\tReverb_setConfig keep sampling rate at %d", SampleRate);
}
- //ALOGV("\tReverb_configure End");
+ //ALOGV("\tReverb_setConfig End");
return 0;
-} /* end Reverb_configure */
+} /* end Reverb_setConfig */
+//----------------------------------------------------------------------------
+// Reverb_getConfig()
+//----------------------------------------------------------------------------
+// Purpose: Get input and output audio configuration.
+//
+// Inputs:
+// pContext: effect engine context
+// pConfig: pointer to effect_config_t structure holding input and output
+// configuration parameters
+//
+// Outputs:
+//
+//----------------------------------------------------------------------------
+
+void Reverb_getConfig(ReverbContext *pContext, effect_config_t *pConfig)
+{
+ memcpy(pConfig, &pContext->config, sizeof(effect_config_t));
+} /* end Reverb_getConfig */
//----------------------------------------------------------------------------
// Reverb_init()
@@ -1924,24 +1944,36 @@
*(int *) pReplyData = 0;
break;
- case EFFECT_CMD_CONFIGURE:
+ case EFFECT_CMD_SET_CONFIG:
//ALOGV("\tReverb_command cmdCode Case: "
- // "EFFECT_CMD_CONFIGURE start");
- if (pCmdData == NULL||
- cmdSize != sizeof(effect_config_t)||
- pReplyData == NULL||
- *replySize != sizeof(int)){
+ // "EFFECT_CMD_SET_CONFIG start");
+ if (pCmdData == NULL ||
+ cmdSize != sizeof(effect_config_t) ||
+ pReplyData == NULL ||
+ *replySize != sizeof(int)) {
ALOGV("\tLVM_ERROR : Reverb_command cmdCode Case: "
- "EFFECT_CMD_CONFIGURE: ERROR");
+ "EFFECT_CMD_SET_CONFIG: ERROR");
return -EINVAL;
}
- *(int *) pReplyData = Reverb_configure(pContext, (effect_config_t *) pCmdData);
+ *(int *) pReplyData = android::Reverb_setConfig(pContext,
+ (effect_config_t *) pCmdData);
+ break;
+
+ case EFFECT_CMD_GET_CONFIG:
+ if (pReplyData == NULL ||
+ *replySize != sizeof(effect_config_t)) {
+ ALOGV("\tLVM_ERROR : Reverb_command cmdCode Case: "
+ "EFFECT_CMD_GET_CONFIG: ERROR");
+ return -EINVAL;
+ }
+
+ android::Reverb_getConfig(pContext, (effect_config_t *)pReplyData);
break;
case EFFECT_CMD_RESET:
//ALOGV("\tReverb_command cmdCode Case: "
// "EFFECT_CMD_RESET start");
- Reverb_configure(pContext, &pContext->config);
+ Reverb_setConfig(pContext, &pContext->config);
break;
case EFFECT_CMD_GET_PARAM:{
diff --git a/media/libeffects/preprocessing/Android.mk b/media/libeffects/preprocessing/Android.mk
index 77d40b6..c13b9d4 100755
--- a/media/libeffects/preprocessing/Android.mk
+++ b/media/libeffects/preprocessing/Android.mk
@@ -13,8 +13,8 @@
LOCAL_C_INCLUDES += \
external/webrtc/src \
external/webrtc/src/modules/interface \
- external/webrtc/src/modules/audio_processing/main/interface \
- system/media/audio_effects/include
+ external/webrtc/src/modules/audio_processing/interface \
+ $(call include-path-for, audio-effects)
LOCAL_C_INCLUDES += $(call include-path-for, speex)
diff --git a/media/libeffects/preprocessing/PreProcessing.cpp b/media/libeffects/preprocessing/PreProcessing.cpp
index 6267d1d..1d76f62 100755
--- a/media/libeffects/preprocessing/PreProcessing.cpp
+++ b/media/libeffects/preprocessing/PreProcessing.cpp
@@ -24,8 +24,8 @@
#include <audio_effects/effect_aec.h>
#include <audio_effects/effect_agc.h>
#include <audio_effects/effect_ns.h>
-#include "modules/interface/module_common_types.h"
-#include "modules/audio_processing/main/interface/audio_processing.h"
+#include <module_common_types.h>
+#include <audio_processing.h>
#include "speex/speex_resampler.h"
@@ -220,8 +220,8 @@
// Automatic Gain Control (AGC)
//------------------------------------------------------------------------------
-static const int kAgcDefaultTargetLevel = 0;
-static const int kAgcDefaultCompGain = 90;
+static const int kAgcDefaultTargetLevel = 3;
+static const int kAgcDefaultCompGain = 9;
static const bool kAgcDefaultLimiter = true;
int AgcInit (preproc_effect_t *effect)
@@ -845,6 +845,17 @@
config->inputCfg.samplingRate, config->inputCfg.channels);
int status;
+ // if at least one process is enabled, do not accept configuration changes
+ if (session->enabledMsk) {
+ if (session->samplingRate != config->inputCfg.samplingRate ||
+ session->inChannelCount != inCnl ||
+ session->outChannelCount != outCnl) {
+ return -ENOSYS;
+ } else {
+ return 0;
+ }
+ }
+
// AEC implementation is limited to 16kHz
if (config->inputCfg.samplingRate >= 32000 && !(session->createdMsk & (1 << PREPROC_AEC))) {
session->apmSamplingRate = 32000;
@@ -940,6 +951,18 @@
return 0;
}
+void Session_GetConfig(preproc_session_t *session, effect_config_t *config)
+{
+ memset(config, 0, sizeof(effect_config_t));
+ config->inputCfg.samplingRate = config->outputCfg.samplingRate = session->samplingRate;
+ config->inputCfg.format = config->outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
+ config->inputCfg.channels = audio_channel_in_mask_from_count(session->inChannelCount);
+ // "out" doesn't mean output device, so this is the correct API to convert channel count to mask
+ config->outputCfg.channels = audio_channel_in_mask_from_count(session->outChannelCount);
+ config->inputCfg.mask = config->outputCfg.mask =
+ (EFFECT_CONFIG_SMP_RATE | EFFECT_CONFIG_CHANNELS | EFFECT_CONFIG_FORMAT);
+}
+
int Session_SetReverseConfig(preproc_session_t *session, effect_config_t *config)
{
if (config->inputCfg.samplingRate != config->outputCfg.samplingRate ||
@@ -969,6 +992,17 @@
return 0;
}
+void Session_GetReverseConfig(preproc_session_t *session, effect_config_t *config)
+{
+ memset(config, 0, sizeof(effect_config_t));
+ config->inputCfg.samplingRate = config->outputCfg.samplingRate = session->samplingRate;
+ config->inputCfg.format = config->outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
+ config->inputCfg.channels = config->outputCfg.channels =
+ audio_channel_in_mask_from_count(session->revChannelCount);
+ config->inputCfg.mask = config->outputCfg.mask =
+ (EFFECT_CONFIG_SMP_RATE | EFFECT_CONFIG_CHANNELS | EFFECT_CONFIG_FORMAT);
+}
+
void Session_SetProcEnabled(preproc_session_t *session, uint32_t procId, bool enabled)
{
if (enabled) {
@@ -1048,7 +1082,7 @@
return sInitStatus;
}
-const effect_descriptor_t *PreProc_GetDescriptor(effect_uuid_t *uuid)
+const effect_descriptor_t *PreProc_GetDescriptor(const effect_uuid_t *uuid)
{
size_t i;
for (i = 0; i < PREPROC_NUM_EFFECTS; i++) {
@@ -1250,29 +1284,42 @@
*(int *)pReplyData = 0;
break;
- case EFFECT_CMD_CONFIGURE:
+ case EFFECT_CMD_SET_CONFIG:
if (pCmdData == NULL||
cmdSize != sizeof(effect_config_t)||
pReplyData == NULL||
*replySize != sizeof(int)){
ALOGV("PreProcessingFx_Command cmdCode Case: "
- "EFFECT_CMD_CONFIGURE: ERROR");
+ "EFFECT_CMD_SET_CONFIG: ERROR");
return -EINVAL;
}
*(int *)pReplyData = Session_SetConfig(effect->session, (effect_config_t *)pCmdData);
if (*(int *)pReplyData != 0) {
break;
}
- *(int *)pReplyData = Effect_SetState(effect, PREPROC_EFFECT_STATE_CONFIG);
+ if (effect->state != PREPROC_EFFECT_STATE_ACTIVE) {
+ *(int *)pReplyData = Effect_SetState(effect, PREPROC_EFFECT_STATE_CONFIG);
+ }
break;
- case EFFECT_CMD_CONFIGURE_REVERSE:
- if (pCmdData == NULL||
- cmdSize != sizeof(effect_config_t)||
- pReplyData == NULL||
- *replySize != sizeof(int)){
+ case EFFECT_CMD_GET_CONFIG:
+ if (pReplyData == NULL ||
+ *replySize != sizeof(effect_config_t)) {
+ ALOGV("\tLVM_ERROR : PreProcessingFx_Command cmdCode Case: "
+ "EFFECT_CMD_GET_CONFIG: ERROR");
+ return -EINVAL;
+ }
+
+ Session_GetConfig(effect->session, (effect_config_t *)pReplyData);
+ break;
+
+ case EFFECT_CMD_SET_CONFIG_REVERSE:
+ if (pCmdData == NULL ||
+ cmdSize != sizeof(effect_config_t) ||
+ pReplyData == NULL ||
+ *replySize != sizeof(int)) {
ALOGV("PreProcessingFx_Command cmdCode Case: "
- "EFFECT_CMD_CONFIGURE_REVERSE: ERROR");
+ "EFFECT_CMD_SET_CONFIG_REVERSE: ERROR");
return -EINVAL;
}
*(int *)pReplyData = Session_SetReverseConfig(effect->session,
@@ -1282,6 +1329,16 @@
}
break;
+ case EFFECT_CMD_GET_CONFIG_REVERSE:
+ if (pReplyData == NULL ||
+ *replySize != sizeof(effect_config_t)){
+ ALOGV("PreProcessingFx_Command cmdCode Case: "
+ "EFFECT_CMD_GET_CONFIG_REVERSE: ERROR");
+ return -EINVAL;
+ }
+ Session_GetReverseConfig(effect->session, (effect_config_t *)pCmdData);
+ break;
+
case EFFECT_CMD_RESET:
if (effect->ops->reset) {
effect->ops->reset(effect);
@@ -1523,7 +1580,7 @@
return 0;
}
-int PreProcessingLib_Create(effect_uuid_t *uuid,
+int PreProcessingLib_Create(const effect_uuid_t *uuid,
int32_t sessionId,
int32_t ioId,
effect_handle_t *pInterface)
@@ -1575,7 +1632,7 @@
return Session_ReleaseEffect(fx->session, fx);
}
-int PreProcessingLib_GetDescriptor(effect_uuid_t *uuid,
+int PreProcessingLib_GetDescriptor(const effect_uuid_t *uuid,
effect_descriptor_t *pDescriptor) {
if (pDescriptor == NULL || uuid == NULL){
diff --git a/media/libeffects/testlibs/Android.mk_ b/media/libeffects/testlibs/Android.mk_
index 249ebf4..2954908 100644
--- a/media/libeffects/testlibs/Android.mk_
+++ b/media/libeffects/testlibs/Android.mk_
@@ -23,7 +23,7 @@
endif
LOCAL_C_INCLUDES := \
- system/media/audio_effects/include \
+ $(call include-path-for, audio-effects) \
$(call include-path-for, graphics corecg)
LOCAL_MODULE_TAGS := optional
@@ -60,7 +60,7 @@
LOCAL_C_INCLUDES := \
$(call include-path-for, graphics corecg) \
- system/media/audio_effects/include
+ $(call include-path-for, audio-effects)
LOCAL_MODULE_TAGS := optional
diff --git a/media/libeffects/testlibs/AudioBiquadFilter.cpp b/media/libeffects/testlibs/AudioBiquadFilter.cpp
index 72917a3..16dd1c5 100644
--- a/media/libeffects/testlibs/AudioBiquadFilter.cpp
+++ b/media/libeffects/testlibs/AudioBiquadFilter.cpp
@@ -17,12 +17,10 @@
#include <string.h>
#include <assert.h>
+#include <cutils/compiler.h>
#include "AudioBiquadFilter.h"
-#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
-#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
-
namespace android {
const audio_coef_t AudioBiquadFilter::IDENTITY_COEFS[AudioBiquadFilter::NUM_COEFS] = { AUDIO_COEF_ONE, 0, 0, 0, 0 };
@@ -55,7 +53,7 @@
void AudioBiquadFilter::setCoefs(const audio_coef_t coefs[NUM_COEFS], bool immediate) {
memcpy(mTargetCoefs, coefs, sizeof(mTargetCoefs));
if (mState & STATE_ENABLED_MASK) {
- if (UNLIKELY(immediate)) {
+ if (CC_UNLIKELY(immediate)) {
memcpy(mCoefs, coefs, sizeof(mCoefs));
setState(STATE_NORMAL);
} else {
@@ -70,7 +68,7 @@
}
void AudioBiquadFilter::enable(bool immediate) {
- if (UNLIKELY(immediate)) {
+ if (CC_UNLIKELY(immediate)) {
memcpy(mCoefs, mTargetCoefs, sizeof(mCoefs));
setState(STATE_NORMAL);
} else {
@@ -79,7 +77,7 @@
}
void AudioBiquadFilter::disable(bool immediate) {
- if (UNLIKELY(immediate)) {
+ if (CC_UNLIKELY(immediate)) {
memcpy(mCoefs, IDENTITY_COEFS, sizeof(mCoefs));
setState(STATE_BYPASS);
} else {
@@ -142,7 +140,7 @@
audio_sample_t * out,
int frameCount) {
// The common case is in-place processing, because this is what the EQ does.
- if (UNLIKELY(in != out)) {
+ if (CC_UNLIKELY(in != out)) {
memcpy(out, in, frameCount * mNumChannels * sizeof(audio_sample_t));
}
}
diff --git a/media/libeffects/testlibs/AudioCoefInterpolator.cpp b/media/libeffects/testlibs/AudioCoefInterpolator.cpp
index 039ab9f..6b56922 100644
--- a/media/libeffects/testlibs/AudioCoefInterpolator.cpp
+++ b/media/libeffects/testlibs/AudioCoefInterpolator.cpp
@@ -16,10 +16,10 @@
*/
#include <string.h>
-#include "AudioCoefInterpolator.h"
-#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
-#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
+#include <cutils/compiler.h>
+
+#include "AudioCoefInterpolator.h"
namespace android {
@@ -44,9 +44,9 @@
size_t index = 0;
size_t dim = mNumInDims;
while (dim-- > 0) {
- if (UNLIKELY(intCoord[dim] < 0)) {
+ if (CC_UNLIKELY(intCoord[dim] < 0)) {
fracCoord[dim] = 0;
- } else if (UNLIKELY(intCoord[dim] >= (int)mInDims[dim] - 1)) {
+ } else if (CC_UNLIKELY(intCoord[dim] >= (int)mInDims[dim] - 1)) {
fracCoord[dim] = 0;
index += mInDimOffsets[dim] * (mInDims[dim] - 1);
} else {
@@ -63,7 +63,7 @@
memcpy(out, mTable + index, mNumOutDims * sizeof(audio_coef_t));
} else {
getCoefRecurse(index, fracCoord, out, dim + 1);
- if (LIKELY(fracCoord != 0)) {
+ if (CC_LIKELY(fracCoord != 0)) {
audio_coef_t tempCoef[MAX_OUT_DIMS];
getCoefRecurse(index + mInDimOffsets[dim], fracCoord, tempCoef,
dim + 1);
diff --git a/media/libeffects/testlibs/AudioCommon.h b/media/libeffects/testlibs/AudioCommon.h
index 444f93a..e8080dc 100644
--- a/media/libeffects/testlibs/AudioCommon.h
+++ b/media/libeffects/testlibs/AudioCommon.h
@@ -20,6 +20,7 @@
#include <stdint.h>
#include <stddef.h>
+#include <cutils/compiler.h>
namespace android {
@@ -76,9 +77,9 @@
// Convert a audio_sample_t sample to S15 (with clipping)
inline int16_t audio_sample_t_to_s15_clip(audio_sample_t sample) {
// TODO: optimize for targets supporting this as an atomic operation.
- if (__builtin_expect(sample >= (0x7FFF << 9), 0)) {
+ if (CC_UNLIKELY(sample >= (0x7FFF << 9))) {
return 0x7FFF;
- } else if (__builtin_expect(sample <= -(0x8000 << 9), 0)) {
+ } else if (CC_UNLIKELY(sample <= -(0x8000 << 9))) {
return 0x8000;
} else {
return audio_sample_t_to_s15(sample);
diff --git a/media/libeffects/testlibs/AudioPeakingFilter.cpp b/media/libeffects/testlibs/AudioPeakingFilter.cpp
index 60fefe6..99323ac 100644
--- a/media/libeffects/testlibs/AudioPeakingFilter.cpp
+++ b/media/libeffects/testlibs/AudioPeakingFilter.cpp
@@ -21,9 +21,7 @@
#include <new>
#include <assert.h>
-
-#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
-#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
+#include <cutils/compiler.h>
namespace android {
// Format of the coefficient table:
@@ -66,12 +64,12 @@
void AudioPeakingFilter::setFrequency(uint32_t millihertz) {
mNominalFrequency = millihertz;
- if (UNLIKELY(millihertz > mNiquistFreq / 2)) {
+ if (CC_UNLIKELY(millihertz > mNiquistFreq / 2)) {
millihertz = mNiquistFreq / 2;
}
uint32_t normFreq = static_cast<uint32_t>(
(static_cast<uint64_t>(millihertz) * mFrequencyFactor) >> 10);
- if (LIKELY(normFreq > (1 << 23))) {
+ if (CC_LIKELY(normFreq > (1 << 23))) {
mFrequency = (Effects_log2(normFreq) - ((32-9) << 15)) << (FREQ_PRECISION_BITS - 15);
} else {
mFrequency = 0;
@@ -107,11 +105,11 @@
int32_t halfBW = (((mBandwidth + 1) / 2) << 15) / 1200;
low = static_cast<uint32_t>((static_cast<uint64_t>(mNominalFrequency) * Effects_exp2(-halfBW + (16 << 15))) >> 16);
- if (UNLIKELY(halfBW >= (16 << 15))) {
+ if (CC_UNLIKELY(halfBW >= (16 << 15))) {
high = mNiquistFreq;
} else {
high = static_cast<uint32_t>((static_cast<uint64_t>(mNominalFrequency) * Effects_exp2(halfBW + (16 << 15))) >> 16);
- if (UNLIKELY(high > mNiquistFreq)) {
+ if (CC_UNLIKELY(high > mNiquistFreq)) {
high = mNiquistFreq;
}
}
diff --git a/media/libeffects/testlibs/AudioShelvingFilter.cpp b/media/libeffects/testlibs/AudioShelvingFilter.cpp
index b8650ba..e031287 100644
--- a/media/libeffects/testlibs/AudioShelvingFilter.cpp
+++ b/media/libeffects/testlibs/AudioShelvingFilter.cpp
@@ -21,9 +21,7 @@
#include <new>
#include <assert.h>
-
-#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
-#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
+#include <cutils/compiler.h>
namespace android {
// Format of the coefficient tables:
@@ -71,13 +69,13 @@
void AudioShelvingFilter::setFrequency(uint32_t millihertz) {
mNominalFrequency = millihertz;
- if (UNLIKELY(millihertz > mNiquistFreq / 2)) {
+ if (CC_UNLIKELY(millihertz > mNiquistFreq / 2)) {
millihertz = mNiquistFreq / 2;
}
uint32_t normFreq = static_cast<uint32_t>(
(static_cast<uint64_t>(millihertz) * mFrequencyFactor) >> 10);
uint32_t log2minFreq = (mType == kLowShelf ? (32-10) : (32-2));
- if (LIKELY(normFreq > (1U << log2minFreq))) {
+ if (CC_LIKELY(normFreq > (1U << log2minFreq))) {
mFrequency = (Effects_log2(normFreq) - (log2minFreq << 15)) << (FREQ_PRECISION_BITS - 15);
} else {
mFrequency = 0;
diff --git a/media/libeffects/testlibs/EffectEqualizer.cpp b/media/libeffects/testlibs/EffectEqualizer.cpp
index 43f34de..35a4a61 100644
--- a/media/libeffects/testlibs/EffectEqualizer.cpp
+++ b/media/libeffects/testlibs/EffectEqualizer.cpp
@@ -114,7 +114,7 @@
//--- local function prototypes
int Equalizer_init(EqualizerContext *pContext);
-int Equalizer_configure(EqualizerContext *pContext, effect_config_t *pConfig);
+int Equalizer_setConfig(EqualizerContext *pContext, effect_config_t *pConfig);
int Equalizer_getParameter(AudioEqualizer * pEqualizer, int32_t *pParam, size_t *pValueSize, void *pValue);
int Equalizer_setParameter(AudioEqualizer * pEqualizer, int32_t *pParam, void *pValue);
@@ -140,7 +140,7 @@
return 0;
} /* end EffectQueryNext */
-extern "C" int EffectCreate(effect_uuid_t *uuid,
+extern "C" int EffectCreate(const effect_uuid_t *uuid,
int32_t sessionId,
int32_t ioId,
effect_handle_t *pHandle) {
@@ -195,7 +195,7 @@
return 0;
} /* end EffectRelease */
-extern "C" int EffectGetDescriptor(effect_uuid_t *uuid,
+extern "C" int EffectGetDescriptor(const effect_uuid_t *uuid,
effect_descriptor_t *pDescriptor) {
if (pDescriptor == NULL || uuid == NULL){
@@ -224,7 +224,7 @@
}
//----------------------------------------------------------------------------
-// Equalizer_configure()
+// Equalizer_setConfig()
//----------------------------------------------------------------------------
// Purpose: Set input and output audio configuration.
//
@@ -237,9 +237,9 @@
//
//----------------------------------------------------------------------------
-int Equalizer_configure(EqualizerContext *pContext, effect_config_t *pConfig)
+int Equalizer_setConfig(EqualizerContext *pContext, effect_config_t *pConfig)
{
- ALOGV("Equalizer_configure start");
+ ALOGV("Equalizer_setConfig start");
CHECK_ARG(pContext != NULL);
CHECK_ARG(pConfig != NULL);
@@ -272,7 +272,26 @@
pConfig->outputCfg.accessMode);
return 0;
-} // end Equalizer_configure
+} // end Equalizer_setConfig
+
+//----------------------------------------------------------------------------
+// Equalizer_getConfig()
+//----------------------------------------------------------------------------
+// Purpose: Get input and output audio configuration.
+//
+// Inputs:
+// pContext: effect engine context
+// pConfig: pointer to effect_config_t structure holding input and output
+// configuration parameters
+//
+// Outputs:
+//
+//----------------------------------------------------------------------------
+
+void Equalizer_getConfig(EqualizerContext *pContext, effect_config_t *pConfig)
+{
+ memcpy(pConfig, &pContext->config, sizeof(effect_config_t));
+} // end Equalizer_getConfig
//----------------------------------------------------------------------------
@@ -332,7 +351,7 @@
pContext->pEqualizer->enable(true);
- Equalizer_configure(pContext, &pContext->config);
+ Equalizer_setConfig(pContext, &pContext->config);
return 0;
} // end Equalizer_init
@@ -643,16 +662,22 @@
}
*(int *) pReplyData = Equalizer_init(pContext);
break;
- case EFFECT_CMD_CONFIGURE:
+ case EFFECT_CMD_SET_CONFIG:
if (pCmdData == NULL || cmdSize != sizeof(effect_config_t)
|| pReplyData == NULL || *replySize != sizeof(int)) {
return -EINVAL;
}
- *(int *) pReplyData = Equalizer_configure(pContext,
+ *(int *) pReplyData = Equalizer_setConfig(pContext,
(effect_config_t *) pCmdData);
break;
+ case EFFECT_CMD_GET_CONFIG:
+ if (pReplyData == NULL || *replySize != sizeof(effect_config_t)) {
+ return -EINVAL;
+ }
+ Equalizer_getConfig(pContext, (effect_config_t *) pCmdData);
+ break;
case EFFECT_CMD_RESET:
- Equalizer_configure(pContext, &pContext->config);
+ Equalizer_setConfig(pContext, &pContext->config);
break;
case EFFECT_CMD_GET_PARAM: {
if (pCmdData == NULL || cmdSize < (int)(sizeof(effect_param_t) + sizeof(int32_t)) ||
diff --git a/media/libeffects/testlibs/EffectReverb.c b/media/libeffects/testlibs/EffectReverb.c
index d22868a..8351712 100644
--- a/media/libeffects/testlibs/EffectReverb.c
+++ b/media/libeffects/testlibs/EffectReverb.c
@@ -111,7 +111,7 @@
return 0;
}
-int EffectCreate(effect_uuid_t *uuid,
+int EffectCreate(const effect_uuid_t *uuid,
int32_t sessionId,
int32_t ioId,
effect_handle_t *pHandle) {
@@ -182,7 +182,7 @@
return 0;
}
-int EffectGetDescriptor(effect_uuid_t *uuid,
+int EffectGetDescriptor(const effect_uuid_t *uuid,
effect_descriptor_t *pDescriptor) {
int i;
int length = sizeof(gDescriptors) / sizeof(const effect_descriptor_t *);
@@ -318,14 +318,20 @@
pRvbModule->context.mState = REVERB_STATE_INITIALIZED;
}
break;
- case EFFECT_CMD_CONFIGURE:
+ case EFFECT_CMD_SET_CONFIG:
if (pCmdData == NULL || cmdSize != sizeof(effect_config_t)
|| pReplyData == NULL || *replySize != sizeof(int)) {
return -EINVAL;
}
- *(int *) pReplyData = Reverb_Configure(pRvbModule,
+ *(int *) pReplyData = Reverb_setConfig(pRvbModule,
(effect_config_t *)pCmdData, false);
break;
+ case EFFECT_CMD_GET_CONFIG:
+ if (pReplyData == NULL || *replySize != sizeof(effect_config_t)) {
+ return -EINVAL;
+ }
+ Reverb_getConfig(pRvbModule, (effect_config_t *) pCmdData);
+ break;
case EFFECT_CMD_RESET:
Reverb_Reset(pReverb, false);
break;
@@ -492,7 +498,7 @@
pRvbModule->config.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE;
pRvbModule->config.outputCfg.mask = EFFECT_CONFIG_ALL;
- ret = Reverb_Configure(pRvbModule, &pRvbModule->config, true);
+ ret = Reverb_setConfig(pRvbModule, &pRvbModule->config, true);
if (ret < 0) {
ALOGV("Reverb_Init error %d on module %p", ret, pRvbModule);
}
@@ -501,7 +507,7 @@
}
/*----------------------------------------------------------------------------
- * Reverb_Init()
+ * Reverb_setConfig()
*----------------------------------------------------------------------------
* Purpose:
* Set input and output audio configuration.
@@ -518,7 +524,7 @@
*----------------------------------------------------------------------------
*/
-int Reverb_Configure(reverb_module_t *pRvbModule, effect_config_t *pConfig,
+int Reverb_setConfig(reverb_module_t *pRvbModule, effect_config_t *pConfig,
bool init) {
reverb_object_t *pReverb = &pRvbModule->context;
int bufferSizeInSamples;
@@ -531,12 +537,12 @@
|| pConfig->outputCfg.channels != OUTPUT_CHANNELS
|| pConfig->inputCfg.format != AUDIO_FORMAT_PCM_16_BIT
|| pConfig->outputCfg.format != AUDIO_FORMAT_PCM_16_BIT) {
- ALOGV("Reverb_Configure invalid config");
+ ALOGV("Reverb_setConfig invalid config");
return -EINVAL;
}
if ((pReverb->m_Aux && (pConfig->inputCfg.channels != AUDIO_CHANNEL_OUT_MONO)) ||
(!pReverb->m_Aux && (pConfig->inputCfg.channels != AUDIO_CHANNEL_OUT_STEREO))) {
- ALOGV("Reverb_Configure invalid config");
+ ALOGV("Reverb_setConfig invalid config");
return -EINVAL;
}
@@ -576,7 +582,7 @@
pReverb->m_nCosWT_5KHz = 25997;
break;
default:
- ALOGV("Reverb_Configure invalid sampling rate %d", pReverb->m_nSamplingRate);
+ ALOGV("Reverb_setConfig invalid sampling rate %d", pReverb->m_nSamplingRate);
return -EINVAL;
}
@@ -620,6 +626,28 @@
}
/*----------------------------------------------------------------------------
+ * Reverb_getConfig()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Get input and output audio configuration.
+ *
+ * Inputs:
+ * pRvbModule - pointer to reverb effect module
+ * pConfig - pointer to effect_config_t structure containing input
+ * and output audio parameters configuration
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+ */
+
+void Reverb_getConfig(reverb_module_t *pRvbModule, effect_config_t *pConfig)
+{
+ memcpy(pConfig, &pRvbModule->config, sizeof(effect_config_t));
+}
+
+/*----------------------------------------------------------------------------
* Reverb_Reset()
*----------------------------------------------------------------------------
* Purpose:
@@ -844,7 +872,7 @@
if (param == REVERB_PARAM_ROOM_HF_LEVEL) {
break;
}
- pValue32 = &pProperties->decayTime;
+ pValue32 = (int32_t *)&pProperties->decayTime;
/* FALL THROUGH */
case REVERB_PARAM_DECAY_TIME:
@@ -916,7 +944,7 @@
if (param == REVERB_PARAM_REFLECTIONS_LEVEL) {
break;
}
- pValue32 = &pProperties->reflectionsDelay;
+ pValue32 = (int32_t *)&pProperties->reflectionsDelay;
/* FALL THROUGH */
case REVERB_PARAM_REFLECTIONS_DELAY:
@@ -940,7 +968,7 @@
if (param == REVERB_PARAM_REVERB_LEVEL) {
break;
}
- pValue32 = &pProperties->reverbDelay;
+ pValue32 = (int32_t *)&pProperties->reverbDelay;
/* FALL THROUGH */
case REVERB_PARAM_REVERB_DELAY:
diff --git a/media/libeffects/testlibs/EffectReverb.h b/media/libeffects/testlibs/EffectReverb.h
index 8e2cc31..1fb14a7 100644
--- a/media/libeffects/testlibs/EffectReverb.h
+++ b/media/libeffects/testlibs/EffectReverb.h
@@ -303,12 +303,12 @@
int EffectQueryNumberEffects(uint32_t *pNumEffects);
int EffectQueryEffect(uint32_t index,
effect_descriptor_t *pDescriptor);
-int EffectCreate(effect_uuid_t *effectUID,
+int EffectCreate(const effect_uuid_t *effectUID,
int32_t sessionId,
int32_t ioId,
effect_handle_t *pHandle);
int EffectRelease(effect_handle_t handle);
-int EffectGetDescriptor(effect_uuid_t *uuid,
+int EffectGetDescriptor(const effect_uuid_t *uuid,
effect_descriptor_t *pDescriptor);
static int Reverb_Process(effect_handle_t self,
@@ -329,7 +329,8 @@
*/
int Reverb_Init(reverb_module_t *pRvbModule, int aux, int preset);
-int Reverb_Configure(reverb_module_t *pRvbModule, effect_config_t *pConfig, bool init);
+int Reverb_setConfig(reverb_module_t *pRvbModule, effect_config_t *pConfig, bool init);
+void Reverb_getConfig(reverb_module_t *pRvbModule, effect_config_t *pConfig);
void Reverb_Reset(reverb_object_t *pReverb, bool init);
int Reverb_setParameter (reverb_object_t *pReverb, int32_t param, size_t size, void *pValue);
diff --git a/media/libeffects/visualizer/Android.mk b/media/libeffects/visualizer/Android.mk
index 2160177..76b5110 100644
--- a/media/libeffects/visualizer/Android.mk
+++ b/media/libeffects/visualizer/Android.mk
@@ -17,7 +17,7 @@
LOCAL_C_INCLUDES := \
$(call include-path-for, graphics corecg) \
- system/media/audio_effects/include
+ $(call include-path-for, audio-effects)
include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libeffects/visualizer/EffectVisualizer.cpp b/media/libeffects/visualizer/EffectVisualizer.cpp
index c441710..51c8b68 100644
--- a/media/libeffects/visualizer/EffectVisualizer.cpp
+++ b/media/libeffects/visualizer/EffectVisualizer.cpp
@@ -78,7 +78,7 @@
}
//----------------------------------------------------------------------------
-// Visualizer_configure()
+// Visualizer_setConfig()
//----------------------------------------------------------------------------
// Purpose: Set input and output audio configuration.
//
@@ -91,9 +91,9 @@
//
//----------------------------------------------------------------------------
-int Visualizer_configure(VisualizerContext *pContext, effect_config_t *pConfig)
+int Visualizer_setConfig(VisualizerContext *pContext, effect_config_t *pConfig)
{
- ALOGV("Visualizer_configure start");
+ ALOGV("Visualizer_setConfig start");
if (pConfig->inputCfg.samplingRate != pConfig->outputCfg.samplingRate) return -EINVAL;
if (pConfig->inputCfg.channels != pConfig->outputCfg.channels) return -EINVAL;
@@ -112,6 +112,26 @@
//----------------------------------------------------------------------------
+// Visualizer_getConfig()
+//----------------------------------------------------------------------------
+// Purpose: Get input and output audio configuration.
+//
+// Inputs:
+// pContext: effect engine context
+// pConfig: pointer to effect_config_t structure holding input and output
+// configuration parameters
+//
+// Outputs:
+//
+//----------------------------------------------------------------------------
+
+void Visualizer_getConfig(VisualizerContext *pContext, effect_config_t *pConfig)
+{
+ memcpy(pConfig, &pContext->mConfig, sizeof(effect_config_t));
+}
+
+
+//----------------------------------------------------------------------------
// Visualizer_init()
//----------------------------------------------------------------------------
// Purpose: Initialize engine with default configuration.
@@ -144,7 +164,7 @@
pContext->mCaptureSize = VISUALIZER_CAPTURE_SIZE_MAX;
- Visualizer_configure(pContext, &pContext->mConfig);
+ Visualizer_setConfig(pContext, &pContext->mConfig);
return 0;
}
@@ -170,7 +190,7 @@
return 0;
}
-int VisualizerLib_Create(effect_uuid_t *uuid,
+int VisualizerLib_Create(const effect_uuid_t *uuid,
int32_t sessionId,
int32_t ioId,
effect_handle_t *pHandle) {
@@ -220,7 +240,7 @@
return 0;
}
-int VisualizerLib_GetDescriptor(effect_uuid_t *uuid,
+int VisualizerLib_GetDescriptor(const effect_uuid_t *uuid,
effect_descriptor_t *pDescriptor) {
if (pDescriptor == NULL || uuid == NULL){
@@ -337,14 +357,21 @@
}
*(int *) pReplyData = Visualizer_init(pContext);
break;
- case EFFECT_CMD_CONFIGURE:
+ case EFFECT_CMD_SET_CONFIG:
if (pCmdData == NULL || cmdSize != sizeof(effect_config_t)
|| pReplyData == NULL || *replySize != sizeof(int)) {
return -EINVAL;
}
- *(int *) pReplyData = Visualizer_configure(pContext,
+ *(int *) pReplyData = Visualizer_setConfig(pContext,
(effect_config_t *) pCmdData);
break;
+ case EFFECT_CMD_GET_CONFIG:
+ if (pReplyData == NULL ||
+ *replySize != sizeof(effect_config_t)) {
+ return -EINVAL;
+ }
+ Visualizer_getConfig(pContext, (effect_config_t *)pReplyData);
+ break;
case EFFECT_CMD_RESET:
Visualizer_reset(pContext);
break;
diff --git a/media/libmedia/Android.mk b/media/libmedia/Android.mk
index 7af4a87..21e8f29 100644
--- a/media/libmedia/Android.mk
+++ b/media/libmedia/Android.mk
@@ -43,13 +43,12 @@
IEffectClient.cpp \
AudioEffect.cpp \
Visualizer.cpp \
- MemoryLeakTrackUtil.cpp \
- fixedfft.cpp.arm
+ MemoryLeakTrackUtil.cpp
LOCAL_SHARED_LIBRARIES := \
libui libcutils libutils libbinder libsonivox libicuuc libexpat \
libcamera_client libstagefright_foundation \
- libgui libdl
+ libgui libdl libaudioutils libmedia_native
LOCAL_WHOLE_STATIC_LIBRARY := libmedia_helper
@@ -58,9 +57,10 @@
LOCAL_C_INCLUDES := \
$(JNI_H_INCLUDE) \
$(call include-path-for, graphics corecg) \
- $(TOP)/frameworks/base/include/media/stagefright/openmax \
+ $(TOP)/frameworks/native/include/media/openmax \
external/icu4c/common \
external/expat/lib \
- system/media/audio_effects/include
+ $(call include-path-for, audio-effects) \
+ $(call include-path-for, audio-utils)
include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libmedia/AudioEffect.cpp b/media/libmedia/AudioEffect.cpp
index 6639d06..34451ca 100644
--- a/media/libmedia/AudioEffect.cpp
+++ b/media/libmedia/AudioEffect.cpp
@@ -159,7 +159,7 @@
mCblk->buffer = (uint8_t *)mCblk + bufOffset;
iEffect->asBinder()->linkToDeath(mIEffectClient);
- ALOGV("set() %p OK effect: %s id: %d status %d enabled %d, ", this, mDescriptor.name, mId, mStatus, mEnabled);
+ ALOGV("set() %p OK effect: %s id: %d status %d enabled %d", this, mDescriptor.name, mId, mStatus, mEnabled);
return mStatus;
}
@@ -174,7 +174,7 @@
mIEffect->disconnect();
mIEffect->asBinder()->unlinkToDeath(mIEffectClient);
}
- IPCThreadState::self()->flushCommands();
+ IPCThreadState::self()->flushCommands();
}
mIEffect.clear();
mIEffectClient.clear();
@@ -202,7 +202,7 @@
status_t AudioEffect::setEnabled(bool enabled)
{
if (mStatus != NO_ERROR) {
- return INVALID_OPERATION;
+ return (mStatus == ALREADY_EXISTS) ? (status_t) INVALID_OPERATION : mStatus;
}
status_t status = NO_ERROR;
@@ -231,7 +231,7 @@
{
if (mStatus != NO_ERROR && mStatus != ALREADY_EXISTS) {
ALOGV("command() bad status %d", mStatus);
- return INVALID_OPERATION;
+ return mStatus;
}
if (cmdCode == EFFECT_CMD_ENABLE || cmdCode == EFFECT_CMD_DISABLE) {
@@ -263,7 +263,7 @@
status_t AudioEffect::setParameter(effect_param_t *param)
{
if (mStatus != NO_ERROR) {
- return INVALID_OPERATION;
+ return (mStatus == ALREADY_EXISTS) ? (status_t) INVALID_OPERATION : mStatus;
}
if (param == NULL || param->psize == 0 || param->vsize == 0) {
@@ -281,7 +281,7 @@
status_t AudioEffect::setParameterDeferred(effect_param_t *param)
{
if (mStatus != NO_ERROR) {
- return INVALID_OPERATION;
+ return (mStatus == ALREADY_EXISTS) ? (status_t) INVALID_OPERATION : mStatus;
}
if (param == NULL || param->psize == 0 || param->vsize == 0) {
@@ -307,7 +307,7 @@
status_t AudioEffect::setParameterCommit()
{
if (mStatus != NO_ERROR) {
- return INVALID_OPERATION;
+ return (mStatus == ALREADY_EXISTS) ? (status_t) INVALID_OPERATION : mStatus;
}
Mutex::Autolock _l(mCblk->lock);
@@ -321,7 +321,7 @@
status_t AudioEffect::getParameter(effect_param_t *param)
{
if (mStatus != NO_ERROR && mStatus != ALREADY_EXISTS) {
- return INVALID_OPERATION;
+ return mStatus;
}
if (param == NULL || param->psize == 0 || param->vsize == 0) {
@@ -341,8 +341,8 @@
void AudioEffect::binderDied()
{
ALOGW("IEffect died");
- mStatus = NO_INIT;
- if (mCbf) {
+ mStatus = DEAD_OBJECT;
+ if (mCbf != NULL) {
status_t status = DEAD_OBJECT;
mCbf(EVENT_ERROR, mUserData, &status);
}
@@ -363,7 +363,7 @@
mStatus = ALREADY_EXISTS;
}
}
- if (mCbf) {
+ if (mCbf != NULL) {
mCbf(EVENT_CONTROL_STATUS_CHANGED, mUserData, &controlGranted);
}
}
@@ -373,7 +373,7 @@
ALOGV("enableStatusChanged %p enabled %d mCbf %p", this, enabled, mCbf);
if (mStatus == ALREADY_EXISTS) {
mEnabled = enabled;
- if (mCbf) {
+ if (mCbf != NULL) {
mCbf(EVENT_ENABLE_STATUS_CHANGED, mUserData, &enabled);
}
}
@@ -389,7 +389,7 @@
return;
}
- if (mCbf && cmdCode == EFFECT_CMD_SET_PARAM) {
+ if (mCbf != NULL && cmdCode == EFFECT_CMD_SET_PARAM) {
effect_param_t *cmd = (effect_param_t *)cmdData;
cmd->status = *(int32_t *)replyData;
mCbf(EVENT_PARAMETER_CHANGED, mUserData, cmd);
@@ -412,7 +412,8 @@
return af->queryEffect(index, descriptor);
}
-status_t AudioEffect::getEffectDescriptor(effect_uuid_t *uuid, effect_descriptor_t *descriptor)
+status_t AudioEffect::getEffectDescriptor(const effect_uuid_t *uuid,
+ effect_descriptor_t *descriptor) /*const*/
{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
@@ -479,4 +480,3 @@
}; // namespace android
-
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 34a5eb7..05ade75 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -39,9 +39,7 @@
#include <system/audio.h>
#include <cutils/bitops.h>
-
-#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
-#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
+#include <cutils/compiler.h>
namespace android {
// ---------------------------------------------------------------------------
@@ -50,7 +48,7 @@
status_t AudioRecord::getMinFrameCount(
int* frameCount,
uint32_t sampleRate,
- int format,
+ audio_format_t format,
int channelCount)
{
size_t size = 0;
@@ -80,22 +78,24 @@
// ---------------------------------------------------------------------------
AudioRecord::AudioRecord()
- : mStatus(NO_INIT), mSessionId(0)
+ : mStatus(NO_INIT), mSessionId(0),
+ mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
{
}
AudioRecord::AudioRecord(
- int inputSource,
+ audio_source_t inputSource,
uint32_t sampleRate,
- int format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
- uint32_t flags,
+ record_flags flags,
callback_t cbf,
void* user,
int notificationFrames,
int sessionId)
- : mStatus(NO_INIT), mSessionId(0)
+ : mStatus(NO_INIT), mSessionId(0),
+ mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
{
mStatus = set(inputSource, sampleRate, format, channelMask,
frameCount, flags, cbf, user, notificationFrames, sessionId);
@@ -119,12 +119,12 @@
}
status_t AudioRecord::set(
- int inputSource,
+ audio_source_t inputSource,
uint32_t sampleRate,
- int format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
- uint32_t flags,
+ record_flags flags,
callback_t cbf,
void* user,
int notificationFrames,
@@ -148,7 +148,7 @@
sampleRate = DEFAULT_SAMPLE_RATE;
}
// these below should probably come from the audioFlinger too...
- if (format == 0) {
+ if (format == AUDIO_FORMAT_DEFAULT) {
format = AUDIO_FORMAT_PCM_16_BIT;
}
// validate parameters
@@ -206,11 +206,8 @@
return status;
}
- if (cbf != 0) {
+ if (cbf != NULL) {
mClientRecordThread = new ClientRecordThread(*this, threadCanCallJava);
- if (mClientRecordThread == 0) {
- return NO_INIT;
- }
}
mStatus = NO_ERROR;
@@ -231,7 +228,7 @@
mMarkerReached = false;
mNewPosition = 0;
mUpdatePeriod = 0;
- mInputSource = (uint8_t)inputSource;
+ mInputSource = inputSource;
mFlags = flags;
mInput = input;
AudioSystem::acquireAudioSessionId(mSessionId);
@@ -251,7 +248,7 @@
return mLatency;
}
-int AudioRecord::format() const
+audio_format_t AudioRecord::format() const
{
return mFormat;
}
@@ -266,7 +263,7 @@
return mFrameCount;
}
-int AudioRecord::frameSize() const
+size_t AudioRecord::frameSize() const
{
if (audio_is_linear_pcm(mFormat)) {
return channelCount()*audio_bytes_per_sample(mFormat);
@@ -275,9 +272,9 @@
}
}
-int AudioRecord::inputSource() const
+audio_source_t AudioRecord::inputSource() const
{
- return (int)mInputSource;
+ return mInputSource;
}
// -------------------------------------------------------------------------
@@ -296,22 +293,36 @@
return WOULD_BLOCK;
}
}
- t->mLock.lock();
- }
+ }
AutoMutex lock(mLock);
// acquire a strong reference on the IAudioRecord and IMemory so that they cannot be destroyed
// while we are accessing the cblk
- sp <IAudioRecord> audioRecord = mAudioRecord;
- sp <IMemory> iMem = mCblkMemory;
+ sp<IAudioRecord> audioRecord = mAudioRecord;
+ sp<IMemory> iMem = mCblkMemory;
audio_track_cblk_t* cblk = mCblk;
if (mActive == 0) {
mActive = 1;
+ pid_t tid;
+ if (t != 0) {
+ mReadyToRun = WOULD_BLOCK;
+ t->run("AudioRecord", ANDROID_PRIORITY_AUDIO);
+ tid = t->getTid(); // pid_t is unknown until run()
+ ALOGV("getTid=%d", tid);
+ if (tid == -1) {
+ tid = 0;
+ }
+ // thread blocks in readyToRun()
+ } else {
+ tid = 0; // not gettid()
+ }
+
cblk->lock.lock();
if (!(cblk->flags & CBLK_INVALID_MSK)) {
cblk->lock.unlock();
- ret = mAudioRecord->start();
+ ALOGV("mAudioRecord->start(tid=%d)", tid);
+ ret = mAudioRecord->start(tid);
cblk->lock.lock();
if (ret == DEAD_OBJECT) {
android_atomic_or(CBLK_INVALID_ON, &cblk->flags);
@@ -326,19 +337,22 @@
cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
cblk->waitTimeMs = 0;
if (t != 0) {
- t->run("ClientRecordThread", ANDROID_PRIORITY_AUDIO);
+ // thread unblocks in readyToRun() and returns NO_ERROR
+ mReadyToRun = NO_ERROR;
+ mCondition.signal();
} else {
- setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_AUDIO);
+ mPreviousPriority = getpriority(PRIO_PROCESS, 0);
+ mPreviousSchedulingGroup = androidGetThreadSchedulingGroup(0);
+ androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
}
} else {
mActive = 0;
+ // thread unblocks in readyToRun() and returns NO_INIT
+ mReadyToRun = NO_INIT;
+ mCondition.signal();
}
}
- if (t != 0) {
- t->mLock.unlock();
- }
-
return ret;
}
@@ -348,10 +362,6 @@
ALOGV("stop");
- if (t != 0) {
- t->mLock.lock();
- }
-
AutoMutex lock(mLock);
if (mActive == 1) {
mActive = 0;
@@ -363,14 +373,11 @@
if (t != 0) {
t->requestExit();
} else {
- setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
+ setpriority(PRIO_PROCESS, 0, mPreviousPriority);
+ androidSetThreadSchedulingGroup(0, mPreviousSchedulingGroup);
}
}
- if (t != 0) {
- t->mLock.unlock();
- }
-
return NO_ERROR;
}
@@ -379,7 +386,7 @@
return !mActive;
}
-uint32_t AudioRecord::getSampleRate()
+uint32_t AudioRecord::getSampleRate() const
{
AutoMutex lock(mLock);
return mCblk->sampleRate;
@@ -387,7 +394,7 @@
status_t AudioRecord::setMarkerPosition(uint32_t marker)
{
- if (mCbf == 0) return INVALID_OPERATION;
+ if (mCbf == NULL) return INVALID_OPERATION;
mMarkerPosition = marker;
mMarkerReached = false;
@@ -395,9 +402,9 @@
return NO_ERROR;
}
-status_t AudioRecord::getMarkerPosition(uint32_t *marker)
+status_t AudioRecord::getMarkerPosition(uint32_t *marker) const
{
- if (marker == 0) return BAD_VALUE;
+ if (marker == NULL) return BAD_VALUE;
*marker = mMarkerPosition;
@@ -406,7 +413,7 @@
status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod)
{
- if (mCbf == 0) return INVALID_OPERATION;
+ if (mCbf == NULL) return INVALID_OPERATION;
uint32_t curPosition;
getPosition(&curPosition);
@@ -416,18 +423,18 @@
return NO_ERROR;
}
-status_t AudioRecord::getPositionUpdatePeriod(uint32_t *updatePeriod)
+status_t AudioRecord::getPositionUpdatePeriod(uint32_t *updatePeriod) const
{
- if (updatePeriod == 0) return BAD_VALUE;
+ if (updatePeriod == NULL) return BAD_VALUE;
*updatePeriod = mUpdatePeriod;
return NO_ERROR;
}
-status_t AudioRecord::getPosition(uint32_t *position)
+status_t AudioRecord::getPosition(uint32_t *position) const
{
- if (position == 0) return BAD_VALUE;
+ if (position == NULL) return BAD_VALUE;
AutoMutex lock(mLock);
*position = mCblk->user;
@@ -435,7 +442,7 @@
return NO_ERROR;
}
-unsigned int AudioRecord::getInputFramesLost()
+unsigned int AudioRecord::getInputFramesLost() const
{
if (mActive)
return AudioSystem::getInputFramesLost(mInput);
@@ -448,7 +455,7 @@
// must be called with mLock held
status_t AudioRecord::openRecord_l(
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
@@ -508,11 +515,11 @@
goto start_loop_here;
while (framesReady == 0) {
active = mActive;
- if (UNLIKELY(!active)) {
+ if (CC_UNLIKELY(!active)) {
cblk->lock.unlock();
return NO_MORE_BUFFERS;
}
- if (UNLIKELY(!waitCount)) {
+ if (CC_UNLIKELY(!waitCount)) {
cblk->lock.unlock();
return WOULD_BLOCK;
}
@@ -529,13 +536,13 @@
if (cblk->flags & CBLK_INVALID_MSK) {
goto create_new_record;
}
- if (__builtin_expect(result!=NO_ERROR, false)) {
+ if (CC_UNLIKELY(result != NO_ERROR)) {
cblk->waitTimeMs += waitTimeMs;
if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
ALOGW( "obtainBuffer timed out (is the CPU pegged?) "
"user=%08x, server=%08x", cblk->user, cblk->server);
cblk->lock.unlock();
- result = mAudioRecord->start();
+ result = mAudioRecord->start(0); // callback thread hasn't changed
cblk->lock.lock();
if (result == DEAD_OBJECT) {
android_atomic_or(CBLK_INVALID_ON, &cblk->flags);
@@ -590,7 +597,7 @@
mCblk->stepUser(audioBuffer->frameCount);
}
-audio_io_handle_t AudioRecord::getInput()
+audio_io_handle_t AudioRecord::getInput() const
{
AutoMutex lock(mLock);
return mInput;
@@ -608,7 +615,7 @@
return mInput;
}
-int AudioRecord::getSessionId()
+int AudioRecord::getSessionId() const
{
return mSessionId;
}
@@ -631,8 +638,8 @@
mLock.lock();
// acquire a strong reference on the IAudioRecord and IMemory so that they cannot be destroyed
// while we are accessing the cblk
- sp <IAudioRecord> audioRecord = mAudioRecord;
- sp <IMemory> iMem = mCblkMemory;
+ sp<IAudioRecord> audioRecord = mAudioRecord;
+ sp<IMemory> iMem = mCblkMemory;
mLock.unlock();
do {
@@ -677,8 +684,8 @@
mLock.lock();
// acquire a strong reference on the IAudioRecord and IMemory so that they cannot be destroyed
// while we are accessing the cblk
- sp <IAudioRecord> audioRecord = mAudioRecord;
- sp <IMemory> iMem = mCblkMemory;
+ sp<IAudioRecord> audioRecord = mAudioRecord;
+ sp<IMemory> iMem = mCblkMemory;
audio_track_cblk_t* cblk = mCblk;
mLock.unlock();
@@ -773,7 +780,7 @@
result = openRecord_l(cblk->sampleRate, mFormat, mChannelMask,
mFrameCount, mFlags, getInput_l());
if (result == NO_ERROR) {
- result = mAudioRecord->start();
+ result = mAudioRecord->start(0); // callback thread hasn't changed
}
if (result != NO_ERROR) {
mActive = false;
@@ -799,7 +806,7 @@
}
}
ALOGV("restoreRecord_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x",
- result, mActive, mCblk, cblk, mCblk->flags, cblk->flags);
+ result, mActive, mCblk, cblk, mCblk->flags, cblk->flags);
if (result == NO_ERROR) {
// from now on we switch to the newly created cblk
@@ -824,7 +831,15 @@
return mReceiver.processAudioBuffer(this);
}
+status_t AudioRecord::ClientRecordThread::readyToRun()
+{
+ AutoMutex(mReceiver.mLock);
+ while (mReceiver.mReadyToRun == WOULD_BLOCK) {
+ mReceiver.mCondition.wait(mReceiver.mLock);
+ }
+ return mReceiver.mReadyToRun;
+}
+
// -------------------------------------------------------------------------
}; // namespace android
-
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index f7f129c..33c7d03 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -35,12 +35,13 @@
sp<AudioSystem::AudioFlingerClient> AudioSystem::gAudioFlingerClient;
audio_error_callback AudioSystem::gAudioErrorCallback = NULL;
// Cached values
-DefaultKeyedVector<int, audio_io_handle_t> AudioSystem::gStreamOutputMap(0);
+
+DefaultKeyedVector<audio_stream_type_t, audio_io_handle_t> AudioSystem::gStreamOutputMap(0);
DefaultKeyedVector<audio_io_handle_t, AudioSystem::OutputDescriptor *> AudioSystem::gOutputs(0);
-// Cached values for recording queries
+// Cached values for recording queries, all protected by gLock
uint32_t AudioSystem::gPrevInSamplingRate = 16000;
-int AudioSystem::gPrevInFormat = AUDIO_FORMAT_PCM_16_BIT;
+audio_format_t AudioSystem::gPrevInFormat = AUDIO_FORMAT_PCM_16_BIT;
int AudioSystem::gPrevInChannelCount = 1;
size_t AudioSystem::gInBuffSize = 0;
@@ -49,7 +50,7 @@
const sp<IAudioFlinger>& AudioSystem::get_audio_flinger()
{
Mutex::Autolock _l(gLock);
- if (gAudioFlinger.get() == 0) {
+ if (gAudioFlinger == 0) {
sp<IServiceManager> sm = defaultServiceManager();
sp<IBinder> binder;
do {
@@ -58,14 +59,14 @@
break;
ALOGW("AudioFlinger not published, waiting...");
usleep(500000); // 0.5 s
- } while(true);
+ } while (true);
if (gAudioFlingerClient == NULL) {
gAudioFlingerClient = new AudioFlingerClient();
} else {
if (gAudioErrorCallback) {
gAudioErrorCallback(NO_ERROR);
}
- }
+ }
binder->linkToDeath(gAudioFlingerClient);
gAudioFlinger = interface_cast<IAudioFlinger>(binder);
gAudioFlinger->registerClient(gAudioFlingerClient);
@@ -120,7 +121,8 @@
return NO_ERROR;
}
-status_t AudioSystem::setStreamVolume(int stream, float value, int output)
+status_t AudioSystem::setStreamVolume(audio_stream_type_t stream, float value,
+ audio_io_handle_t output)
{
if (uint32_t(stream) >= AUDIO_STREAM_CNT) return BAD_VALUE;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
@@ -129,7 +131,7 @@
return NO_ERROR;
}
-status_t AudioSystem::setStreamMute(int stream, bool mute)
+status_t AudioSystem::setStreamMute(audio_stream_type_t stream, bool mute)
{
if (uint32_t(stream) >= AUDIO_STREAM_CNT) return BAD_VALUE;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
@@ -138,7 +140,8 @@
return NO_ERROR;
}
-status_t AudioSystem::getStreamVolume(int stream, float* volume, int output)
+status_t AudioSystem::getStreamVolume(audio_stream_type_t stream, float* volume,
+ audio_io_handle_t output)
{
if (uint32_t(stream) >= AUDIO_STREAM_CNT) return BAD_VALUE;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
@@ -147,7 +150,7 @@
return NO_ERROR;
}
-status_t AudioSystem::getStreamMute(int stream, bool* mute)
+status_t AudioSystem::getStreamMute(audio_stream_type_t stream, bool* mute)
{
if (uint32_t(stream) >= AUDIO_STREAM_CNT) return BAD_VALUE;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
@@ -156,9 +159,9 @@
return NO_ERROR;
}
-status_t AudioSystem::setMode(int mode)
+status_t AudioSystem::setMode(audio_mode_t mode)
{
- if (mode >= AUDIO_MODE_CNT) return BAD_VALUE;
+ if (uint32_t(mode) >= AUDIO_MODE_CNT) return BAD_VALUE;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
return af->setMode(mode);
@@ -203,7 +206,12 @@
return volume ? 100 - int(dBConvertInverse * log(volume) + 0.5) : 0;
}
-status_t AudioSystem::getOutputSamplingRate(int* samplingRate, int streamType)
+// DEPRECATED
+status_t AudioSystem::getOutputSamplingRate(int* samplingRate, int streamType) {
+ return getOutputSamplingRate(samplingRate, (audio_stream_type_t)streamType);
+}
+
+status_t AudioSystem::getOutputSamplingRate(int* samplingRate, audio_stream_type_t streamType)
{
OutputDescriptor *outputDesc;
audio_io_handle_t output;
@@ -212,14 +220,14 @@
streamType = AUDIO_STREAM_MUSIC;
}
- output = getOutput((audio_stream_type_t)streamType);
+ output = getOutput(streamType);
if (output == 0) {
return PERMISSION_DENIED;
}
gLock.lock();
outputDesc = AudioSystem::gOutputs.valueFor(output);
- if (outputDesc == 0) {
+ if (outputDesc == NULL) {
ALOGV("getOutputSamplingRate() no output descriptor for output %d in gOutputs", output);
gLock.unlock();
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
@@ -236,7 +244,12 @@
return NO_ERROR;
}
-status_t AudioSystem::getOutputFrameCount(int* frameCount, int streamType)
+// DEPRECATED
+status_t AudioSystem::getOutputFrameCount(int* frameCount, int streamType) {
+ return getOutputFrameCount(frameCount, (audio_stream_type_t)streamType);
+}
+
+status_t AudioSystem::getOutputFrameCount(int* frameCount, audio_stream_type_t streamType)
{
OutputDescriptor *outputDesc;
audio_io_handle_t output;
@@ -245,14 +258,14 @@
streamType = AUDIO_STREAM_MUSIC;
}
- output = getOutput((audio_stream_type_t)streamType);
+ output = getOutput(streamType);
if (output == 0) {
return PERMISSION_DENIED;
}
gLock.lock();
outputDesc = AudioSystem::gOutputs.valueFor(output);
- if (outputDesc == 0) {
+ if (outputDesc == NULL) {
gLock.unlock();
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
@@ -267,7 +280,7 @@
return NO_ERROR;
}
-status_t AudioSystem::getOutputLatency(uint32_t* latency, int streamType)
+status_t AudioSystem::getOutputLatency(uint32_t* latency, audio_stream_type_t streamType)
{
OutputDescriptor *outputDesc;
audio_io_handle_t output;
@@ -276,14 +289,14 @@
streamType = AUDIO_STREAM_MUSIC;
}
- output = getOutput((audio_stream_type_t)streamType);
+ output = getOutput(streamType);
if (output == 0) {
return PERMISSION_DENIED;
}
gLock.lock();
outputDesc = AudioSystem::gOutputs.valueFor(output);
- if (outputDesc == 0) {
+ if (outputDesc == NULL) {
gLock.unlock();
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
@@ -298,25 +311,30 @@
return NO_ERROR;
}
-status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, int format, int channelCount,
+status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount,
size_t* buffSize)
{
+ gLock.lock();
// Do we have a stale gInBufferSize or are we requesting the input buffer size for new values
- if ((gInBuffSize == 0) || (sampleRate != gPrevInSamplingRate) || (format != gPrevInFormat)
+ size_t inBuffSize = gInBuffSize;
+ if ((inBuffSize == 0) || (sampleRate != gPrevInSamplingRate) || (format != gPrevInFormat)
|| (channelCount != gPrevInChannelCount)) {
+ gLock.unlock();
+ const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
+ if (af == 0) {
+ return PERMISSION_DENIED;
+ }
+ inBuffSize = af->getInputBufferSize(sampleRate, format, channelCount);
+ gLock.lock();
// save the request params
gPrevInSamplingRate = sampleRate;
gPrevInFormat = format;
gPrevInChannelCount = channelCount;
- gInBuffSize = 0;
- const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
- if (af == 0) {
- return PERMISSION_DENIED;
- }
- gInBuffSize = af->getInputBufferSize(sampleRate, format, channelCount);
+ gInBuffSize = inBuffSize;
}
- *buffSize = gInBuffSize;
+ gLock.unlock();
+ *buffSize = inBuffSize;
return NO_ERROR;
}
@@ -328,7 +346,7 @@
return af->setVoiceVolume(value);
}
-status_t AudioSystem::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int stream)
+status_t AudioSystem::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, audio_stream_type_t stream)
{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
@@ -337,7 +355,7 @@
stream = AUDIO_STREAM_MUSIC;
}
- return af->getRenderPosition(halFrames, dspFrames, getOutput((audio_stream_type_t)stream));
+ return af->getRenderPosition(halFrames, dspFrames, getOutput(stream));
}
unsigned int AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) {
@@ -386,10 +404,11 @@
ALOGW("AudioFlinger server died!");
}
-void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, int ioHandle, void *param2) {
+void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, audio_io_handle_t ioHandle,
+ const void *param2) {
ALOGV("ioConfigChanged() event %d", event);
- OutputDescriptor *desc;
- uint32_t stream;
+ const OutputDescriptor *desc;
+ audio_stream_type_t stream;
if (ioHandle == 0) return;
@@ -397,8 +416,8 @@
switch (event) {
case STREAM_CONFIG_CHANGED:
- if (param2 == 0) break;
- stream = *(uint32_t *)param2;
+ if (param2 == NULL) break;
+ stream = *(const audio_stream_type_t *)param2;
ALOGV("ioConfigChanged() STREAM_CONFIG_CHANGED stream %d, output %d", stream, ioHandle);
if (gStreamOutputMap.indexOfKey(stream) >= 0) {
gStreamOutputMap.replaceValueFor(stream, ioHandle);
@@ -409,8 +428,8 @@
ALOGV("ioConfigChanged() opening already existing output! %d", ioHandle);
break;
}
- if (param2 == 0) break;
- desc = (OutputDescriptor *)param2;
+ if (param2 == NULL) break;
+ desc = (const OutputDescriptor *)param2;
OutputDescriptor *outputDesc = new OutputDescriptor(*desc);
gOutputs.add(ioHandle, outputDesc);
@@ -438,8 +457,8 @@
ALOGW("ioConfigChanged() modifying unknow output! %d", ioHandle);
break;
}
- if (param2 == 0) break;
- desc = (OutputDescriptor *)param2;
+ if (param2 == NULL) break;
+ desc = (const OutputDescriptor *)param2;
ALOGV("ioConfigChanged() new config for output %d samplingRate %d, format %d channels %d frameCount %d latency %d",
ioHandle, desc->samplingRate, desc->format,
@@ -462,8 +481,8 @@
gAudioErrorCallback = cb;
}
-bool AudioSystem::routedToA2dpOutput(int streamType) {
- switch(streamType) {
+bool AudioSystem::routedToA2dpOutput(audio_stream_type_t streamType) {
+ switch (streamType) {
case AUDIO_STREAM_MUSIC:
case AUDIO_STREAM_VOICE_CALL:
case AUDIO_STREAM_BLUETOOTH_SCO:
@@ -484,7 +503,7 @@
const sp<IAudioPolicyService>& AudioSystem::get_audio_policy_service()
{
gLock.lock();
- if (gAudioPolicyService.get() == 0) {
+ if (gAudioPolicyService == 0) {
sp<IServiceManager> sm = defaultServiceManager();
sp<IBinder> binder;
do {
@@ -493,7 +512,7 @@
break;
ALOGW("AudioPolicyService not published, waiting...");
usleep(500000); // 0.5 s
- } while(true);
+ } while (true);
if (gAudioPolicyServiceClient == NULL) {
gAudioPolicyServiceClient = new AudioPolicyServiceClient();
}
@@ -531,21 +550,15 @@
return aps->getDeviceConnectionState(device, device_address);
}
-status_t AudioSystem::setPhoneState(int state)
+status_t AudioSystem::setPhoneState(audio_mode_t state)
{
+ if (uint32_t(state) >= AUDIO_MODE_CNT) return BAD_VALUE;
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
return aps->setPhoneState(state);
}
-status_t AudioSystem::setRingerMode(uint32_t mode, uint32_t mask)
-{
- const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
- if (aps == 0) return PERMISSION_DENIED;
- return aps->setRingerMode(mode, mask);
-}
-
status_t AudioSystem::setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
@@ -563,7 +576,7 @@
audio_io_handle_t AudioSystem::getOutput(audio_stream_type_t stream,
uint32_t samplingRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channels,
audio_policy_output_flags_t flags)
{
@@ -621,9 +634,9 @@
aps->releaseOutput(output);
}
-audio_io_handle_t AudioSystem::getInput(int inputSource,
+audio_io_handle_t AudioSystem::getInput(audio_source_t inputSource,
uint32_t samplingRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channels,
audio_in_acoustics_t acoustics,
int sessionId)
@@ -663,18 +676,22 @@
return aps->initStreamVolume(stream, indexMin, indexMax);
}
-status_t AudioSystem::setStreamVolumeIndex(audio_stream_type_t stream, int index)
+status_t AudioSystem::setStreamVolumeIndex(audio_stream_type_t stream,
+ int index,
+ audio_devices_t device)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
- return aps->setStreamVolumeIndex(stream, index);
+ return aps->setStreamVolumeIndex(stream, index, device);
}
-status_t AudioSystem::getStreamVolumeIndex(audio_stream_type_t stream, int *index)
+status_t AudioSystem::getStreamVolumeIndex(audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
- return aps->getStreamVolumeIndex(stream, index);
+ return aps->getStreamVolumeIndex(stream, index, device);
}
uint32_t AudioSystem::getStrategyForStream(audio_stream_type_t stream)
@@ -684,10 +701,10 @@
return aps->getStrategyForStream(stream);
}
-uint32_t AudioSystem::getDevicesForStream(audio_stream_type_t stream)
+audio_devices_t AudioSystem::getDevicesForStream(audio_stream_type_t stream)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
- if (aps == 0) return 0;
+ if (aps == 0) return (audio_devices_t)0;
return aps->getDevicesForStream(stream);
}
@@ -723,7 +740,7 @@
return aps->setEffectEnabled(id, enabled);
}
-status_t AudioSystem::isStreamActive(int stream, bool* state, uint32_t inPastMs)
+status_t AudioSystem::isStreamActive(audio_stream_type_t stream, bool* state, uint32_t inPastMs)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
@@ -751,4 +768,3 @@
}
}; // namespace android
-
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 3a8bfa3..048be1d 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -1,4 +1,4 @@
-/* //device/extlibs/pv/android/AudioTrack.cpp
+/*
**
** Copyright 2007, The Android Open Source Project
**
@@ -38,12 +38,12 @@
#include <utils/Atomic.h>
#include <cutils/bitops.h>
+#include <cutils/compiler.h>
#include <system/audio.h>
#include <system/audio_policy.h>
-#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
-#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
+#include <audio_utils/primitives.h>
namespace android {
// ---------------------------------------------------------------------------
@@ -51,7 +51,7 @@
// static
status_t AudioTrack::getMinFrameCount(
int* frameCount,
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate)
{
int afSampleRate;
@@ -72,18 +72,42 @@
if (minBufCount < 2) minBufCount = 2;
*frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :
- afFrameCount * minBufCount * sampleRate / afSampleRate;
+ afFrameCount * minBufCount * sampleRate / afSampleRate;
return NO_ERROR;
}
// ---------------------------------------------------------------------------
AudioTrack::AudioTrack()
- : mStatus(NO_INIT)
+ : mStatus(NO_INIT),
+ mIsTimed(false),
+ mPreviousPriority(ANDROID_PRIORITY_NORMAL),
+ mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
{
}
AudioTrack::AudioTrack(
+ audio_stream_type_t streamType,
+ uint32_t sampleRate,
+ audio_format_t format,
+ int channelMask,
+ int frameCount,
+ audio_policy_output_flags_t flags,
+ callback_t cbf,
+ void* user,
+ int notificationFrames,
+ int sessionId)
+ : mStatus(NO_INIT),
+ mIsTimed(false),
+ mPreviousPriority(ANDROID_PRIORITY_NORMAL),
+ mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
+{
+ mStatus = set(streamType, sampleRate, format, channelMask,
+ frameCount, flags, cbf, user, notificationFrames,
+ 0, false, sessionId);
+}
+
+AudioTrack::AudioTrack(
int streamType,
uint32_t sampleRate,
int format,
@@ -94,25 +118,30 @@
void* user,
int notificationFrames,
int sessionId)
- : mStatus(NO_INIT)
+ : mStatus(NO_INIT),
+ mIsTimed(false),
+ mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
{
- mStatus = set(streamType, sampleRate, format, channelMask,
- frameCount, flags, cbf, user, notificationFrames,
+ mStatus = set((audio_stream_type_t)streamType, sampleRate, (audio_format_t)format, channelMask,
+ frameCount, (audio_policy_output_flags_t)flags, cbf, user, notificationFrames,
0, false, sessionId);
}
AudioTrack::AudioTrack(
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate,
- int format,
+ audio_format_t format,
int channelMask,
const sp<IMemory>& sharedBuffer,
- uint32_t flags,
+ audio_policy_output_flags_t flags,
callback_t cbf,
void* user,
int notificationFrames,
int sessionId)
- : mStatus(NO_INIT)
+ : mStatus(NO_INIT),
+ mIsTimed(false),
+ mPreviousPriority(ANDROID_PRIORITY_NORMAL),
+ mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
{
mStatus = set(streamType, sampleRate, format, channelMask,
0, flags, cbf, user, notificationFrames,
@@ -139,12 +168,12 @@
}
status_t AudioTrack::set(
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate,
- int format,
+ audio_format_t format,
int channelMask,
int frameCount,
- uint32_t flags,
+ audio_policy_output_flags_t flags,
callback_t cbf,
void* user,
int notificationFrames,
@@ -178,7 +207,7 @@
sampleRate = afSampleRate;
}
// these below should probably come from the audioFlinger too...
- if (format == 0) {
+ if (format == AUDIO_FORMAT_DEFAULT) {
format = AUDIO_FORMAT_PCM_16_BIT;
}
if (channelMask == 0) {
@@ -193,7 +222,7 @@
// force direct flag if format is not linear PCM
if (!audio_is_linear_pcm(format)) {
- flags |= AUDIO_POLICY_OUTPUT_FLAG_DIRECT;
+ flags = (audio_policy_output_flags_t) (flags | AUDIO_POLICY_OUTPUT_FLAG_DIRECT);
}
if (!audio_is_output_channel(channelMask)) {
@@ -203,9 +232,9 @@
uint32_t channelCount = popcount(channelMask);
audio_io_handle_t output = AudioSystem::getOutput(
- (audio_stream_type_t)streamType,
- sampleRate,format, channelMask,
- (audio_policy_output_flags_t)flags);
+ streamType,
+ sampleRate, format, channelMask,
+ flags);
if (output == 0) {
ALOGE("Could not get audio output for stream type %d", streamType);
@@ -214,7 +243,7 @@
mVolume[LEFT] = 1.0f;
mVolume[RIGHT] = 1.0f;
- mSendLevel = 0;
+ mSendLevel = 0.0f;
mFrameCount = frameCount;
mNotificationFramesReq = notificationFrames;
mSessionId = sessionId;
@@ -223,7 +252,7 @@
// create the IAudioTrack
status_t status = createTrack_l(streamType,
sampleRate,
- (uint32_t)format,
+ format,
(uint32_t)channelMask,
frameCount,
flags,
@@ -235,23 +264,19 @@
return status;
}
- if (cbf != 0) {
+ if (cbf != NULL) {
mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
- if (mAudioTrackThread == 0) {
- ALOGE("Could not create callback thread");
- return NO_INIT;
- }
}
mStatus = NO_ERROR;
mStreamType = streamType;
- mFormat = (uint32_t)format;
+ mFormat = format;
mChannelMask = (uint32_t)channelMask;
mChannelCount = channelCount;
mSharedBuffer = sharedBuffer;
mMuted = false;
- mActive = 0;
+ mActive = false;
mCbf = cbf;
mUserData = user;
mLoopCount = 0;
@@ -278,12 +303,12 @@
return mLatency;
}
-int AudioTrack::streamType() const
+audio_stream_type_t AudioTrack::streamType() const
{
return mStreamType;
}
-int AudioTrack::format() const
+audio_format_t AudioTrack::format() const
{
return mFormat;
}
@@ -298,7 +323,7 @@
return mCblk->frameCount;
}
-int AudioTrack::frameSize() const
+size_t AudioTrack::frameSize() const
{
if (audio_is_linear_pcm(mFormat)) {
return channelCount()*audio_bytes_per_sample(mFormat);
@@ -327,34 +352,43 @@
return;
}
}
- t->mLock.lock();
- }
+ }
AutoMutex lock(mLock);
// acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
// while we are accessing the cblk
- sp <IAudioTrack> audioTrack = mAudioTrack;
- sp <IMemory> iMem = mCblkMemory;
+ sp<IAudioTrack> audioTrack = mAudioTrack;
+ sp<IMemory> iMem = mCblkMemory;
audio_track_cblk_t* cblk = mCblk;
- if (mActive == 0) {
+ if (!mActive) {
mFlushed = false;
- mActive = 1;
+ mActive = true;
mNewPosition = cblk->server + mUpdatePeriod;
cblk->lock.lock();
cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
cblk->waitTimeMs = 0;
android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags);
+ pid_t tid;
if (t != 0) {
- t->run("AudioTrackThread", ANDROID_PRIORITY_AUDIO);
+ t->run("AudioTrack", ANDROID_PRIORITY_AUDIO);
+ tid = t->getTid(); // pid_t is unknown until run()
+ ALOGV("getTid=%d", tid);
+ if (tid == -1) {
+ tid = 0;
+ }
} else {
- setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_AUDIO);
+ mPreviousPriority = getpriority(PRIO_PROCESS, 0);
+ mPreviousSchedulingGroup = androidGetThreadSchedulingGroup(0);
+ androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
+ tid = 0; // not gettid()
}
ALOGV("start %p before lock cblk %p", this, mCblk);
if (!(cblk->flags & CBLK_INVALID_MSK)) {
cblk->lock.unlock();
- status = mAudioTrack->start();
+ ALOGV("mAudioTrack->start(tid=%d)", tid);
+ status = mAudioTrack->start(tid);
cblk->lock.lock();
if (status == DEAD_OBJECT) {
android_atomic_or(CBLK_INVALID_ON, &cblk->flags);
@@ -366,18 +400,16 @@
cblk->lock.unlock();
if (status != NO_ERROR) {
ALOGV("start() failed");
- mActive = 0;
+ mActive = false;
if (t != 0) {
t->requestExit();
} else {
- setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
+ setpriority(PRIO_PROCESS, 0, mPreviousPriority);
+ androidSetThreadSchedulingGroup(0, mPreviousSchedulingGroup);
}
}
}
- if (t != 0) {
- t->mLock.unlock();
- }
}
void AudioTrack::stop()
@@ -385,13 +417,10 @@
sp<AudioTrackThread> t = mAudioTrackThread;
ALOGV("stop %p", this);
- if (t != 0) {
- t->mLock.lock();
- }
AutoMutex lock(mLock);
- if (mActive == 1) {
- mActive = 0;
+ if (mActive) {
+ mActive = false;
mCblk->cv.signal();
mAudioTrack->stop();
// Cancel loops (If we are in the middle of a loop, playback
@@ -408,18 +437,17 @@
if (t != 0) {
t->requestExit();
} else {
- setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
+ setpriority(PRIO_PROCESS, 0, mPreviousPriority);
+ androidSetThreadSchedulingGroup(0, mPreviousSchedulingGroup);
}
}
- if (t != 0) {
- t->mLock.unlock();
- }
}
bool AudioTrack::stopped() const
{
- return !mActive;
+ AutoMutex lock(mLock);
+ return stopped_l();
}
void AudioTrack::flush()
@@ -451,8 +479,8 @@
{
ALOGV("pause");
AutoMutex lock(mLock);
- if (mActive == 1) {
- mActive = 0;
+ if (mActive) {
+ mActive = false;
mAudioTrack->pause();
}
}
@@ -470,7 +498,7 @@
status_t AudioTrack::setVolume(float left, float right)
{
- if (left > 1.0f || right > 1.0f) {
+ if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) {
return BAD_VALUE;
}
@@ -478,13 +506,12 @@
mVolume[LEFT] = left;
mVolume[RIGHT] = right;
- // write must be atomic
- mCblk->volumeLR = (uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000);
+ mCblk->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000));
return NO_ERROR;
}
-void AudioTrack::getVolume(float* left, float* right)
+void AudioTrack::getVolume(float* left, float* right) const
{
if (left != NULL) {
*left = mVolume[LEFT];
@@ -497,19 +524,19 @@
status_t AudioTrack::setAuxEffectSendLevel(float level)
{
ALOGV("setAuxEffectSendLevel(%f)", level);
- if (level > 1.0f) {
+ if (level < 0.0f || level > 1.0f) {
return BAD_VALUE;
}
AutoMutex lock(mLock);
mSendLevel = level;
- mCblk->sendLevel = uint16_t(level * 0x1000);
+ mCblk->setSendLevel(level);
return NO_ERROR;
}
-void AudioTrack::getAuxEffectSendLevel(float* level)
+void AudioTrack::getAuxEffectSendLevel(float* level) const
{
if (level != NULL) {
*level = mSendLevel;
@@ -520,6 +547,10 @@
{
int afSamplingRate;
+ if (mIsTimed) {
+ return INVALID_OPERATION;
+ }
+
if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) {
return NO_INIT;
}
@@ -531,8 +562,12 @@
return NO_ERROR;
}
-uint32_t AudioTrack::getSampleRate()
+uint32_t AudioTrack::getSampleRate() const
{
+ if (mIsTimed) {
+ return INVALID_OPERATION;
+ }
+
AutoMutex lock(mLock);
return mCblk->sampleRate;
}
@@ -558,6 +593,10 @@
return NO_ERROR;
}
+ if (mIsTimed) {
+ return INVALID_OPERATION;
+ }
+
if (loopStart >= loopEnd ||
loopEnd - loopStart > cblk->frameCount ||
cblk->server > loopStart) {
@@ -579,29 +618,9 @@
return NO_ERROR;
}
-status_t AudioTrack::getLoop(uint32_t *loopStart, uint32_t *loopEnd, int *loopCount)
-{
- AutoMutex lock(mLock);
- if (loopStart != 0) {
- *loopStart = mCblk->loopStart;
- }
- if (loopEnd != 0) {
- *loopEnd = mCblk->loopEnd;
- }
- if (loopCount != 0) {
- if (mCblk->loopCount < 0) {
- *loopCount = -1;
- } else {
- *loopCount = mCblk->loopCount;
- }
- }
-
- return NO_ERROR;
-}
-
status_t AudioTrack::setMarkerPosition(uint32_t marker)
{
- if (mCbf == 0) return INVALID_OPERATION;
+ if (mCbf == NULL) return INVALID_OPERATION;
mMarkerPosition = marker;
mMarkerReached = false;
@@ -609,9 +628,9 @@
return NO_ERROR;
}
-status_t AudioTrack::getMarkerPosition(uint32_t *marker)
+status_t AudioTrack::getMarkerPosition(uint32_t *marker) const
{
- if (marker == 0) return BAD_VALUE;
+ if (marker == NULL) return BAD_VALUE;
*marker = mMarkerPosition;
@@ -620,7 +639,7 @@
status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
{
- if (mCbf == 0) return INVALID_OPERATION;
+ if (mCbf == NULL) return INVALID_OPERATION;
uint32_t curPosition;
getPosition(&curPosition);
@@ -630,9 +649,9 @@
return NO_ERROR;
}
-status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod)
+status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const
{
- if (updatePeriod == 0) return BAD_VALUE;
+ if (updatePeriod == NULL) return BAD_VALUE;
*updatePeriod = mUpdatePeriod;
@@ -641,10 +660,13 @@
status_t AudioTrack::setPosition(uint32_t position)
{
- AutoMutex lock(mLock);
- Mutex::Autolock _l(mCblk->lock);
+ if (mIsTimed) return INVALID_OPERATION;
- if (!stopped()) return INVALID_OPERATION;
+ AutoMutex lock(mLock);
+
+ if (!stopped_l()) return INVALID_OPERATION;
+
+ Mutex::Autolock _l(mCblk->lock);
if (position > mCblk->user) return BAD_VALUE;
@@ -656,7 +678,7 @@
status_t AudioTrack::getPosition(uint32_t *position)
{
- if (position == 0) return BAD_VALUE;
+ if (position == NULL) return BAD_VALUE;
AutoMutex lock(mLock);
*position = mFlushed ? 0 : mCblk->server;
@@ -667,7 +689,7 @@
{
AutoMutex lock(mLock);
- if (!stopped()) return INVALID_OPERATION;
+ if (!stopped_l()) return INVALID_OPERATION;
flush_l();
@@ -685,11 +707,11 @@
// must be called with mLock held
audio_io_handle_t AudioTrack::getOutput_l()
{
- return AudioSystem::getOutput((audio_stream_type_t)mStreamType,
- mCblk->sampleRate, mFormat, mChannelMask, (audio_policy_output_flags_t)mFlags);
+ return AudioSystem::getOutput(mStreamType,
+ mCblk->sampleRate, mFormat, mChannelMask, mFlags);
}
-int AudioTrack::getSessionId()
+int AudioTrack::getSessionId() const
{
return mSessionId;
}
@@ -708,12 +730,12 @@
// must be called with mLock held
status_t AudioTrack::createTrack_l(
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
- uint32_t flags,
+ audio_policy_output_flags_t flags,
const sp<IMemory>& sharedBuffer,
audio_io_handle_t output,
bool enforceFrameCount)
@@ -721,8 +743,8 @@
status_t status;
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
if (audioFlinger == 0) {
- ALOGE("Could not get audioflinger");
- return NO_INIT;
+ ALOGE("Could not get audioflinger");
+ return NO_INIT;
}
int afSampleRate;
@@ -768,7 +790,7 @@
frameCount = minFrameCount;
}
} else {
- // Ensure that buffer alignment matches channelcount
+ // Ensure that buffer alignment matches channelCount
int channelCount = popcount(channelMask);
if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) {
ALOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount);
@@ -787,6 +809,7 @@
((uint16_t)flags) << 16,
sharedBuffer,
output,
+ mIsTimed,
&mSessionId,
&status);
@@ -799,9 +822,7 @@
ALOGE("Could not get control block");
return NO_INIT;
}
- mAudioTrack.clear();
mAudioTrack = track;
- mCblkMemory.clear();
mCblkMemory = cblk;
mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
android_atomic_or(CBLK_DIRECTION_OUT, &mCblk->flags);
@@ -809,12 +830,12 @@
mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
} else {
mCblk->buffers = sharedBuffer->pointer();
- // Force buffer full condition as data is already present in shared memory
+ // Force buffer full condition as data is already present in shared memory
mCblk->stepUser(mCblk->frameCount);
}
- mCblk->volumeLR = (uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000);
- mCblk->sendLevel = uint16_t(mSendLevel * 0x1000);
+ mCblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000));
+ mCblk->setSendLevel(mSendLevel);
mAudioTrack->attachAuxEffect(mAuxEffectId);
mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
mCblk->waitTimeMs = 0;
@@ -826,7 +847,7 @@
status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
{
AutoMutex lock(mLock);
- int active;
+ bool active;
status_t result = NO_ERROR;
audio_track_cblk_t* cblk = mCblk;
uint32_t framesReq = audioBuffer->frameCount;
@@ -848,12 +869,12 @@
goto start_loop_here;
while (framesAvail == 0) {
active = mActive;
- if (UNLIKELY(!active)) {
+ if (CC_UNLIKELY(!active)) {
ALOGV("Not active and NO_MORE_BUFFERS");
cblk->lock.unlock();
return NO_MORE_BUFFERS;
}
- if (UNLIKELY(!waitCount)) {
+ if (CC_UNLIKELY(!waitCount)) {
cblk->lock.unlock();
return WOULD_BLOCK;
}
@@ -862,7 +883,7 @@
result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
cblk->lock.unlock();
mLock.lock();
- if (mActive == 0) {
+ if (!mActive) {
return status_t(STOPPED);
}
cblk->lock.lock();
@@ -871,7 +892,7 @@
if (cblk->flags & CBLK_INVALID_MSK) {
goto create_new_track;
}
- if (__builtin_expect(result!=NO_ERROR, false)) {
+ if (CC_UNLIKELY(result != NO_ERROR)) {
cblk->waitTimeMs += waitTimeMs;
if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
// timing out when a loop has been set and we have already written upto loop end
@@ -881,7 +902,7 @@
"user=%08x, server=%08x", this, cblk->user, cblk->server);
//unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140)
cblk->lock.unlock();
- result = mAudioTrack->start();
+ result = mAudioTrack->start(0); // callback thread hasn't changed
cblk->lock.lock();
if (result == DEAD_OBJECT) {
android_atomic_or(CBLK_INVALID_ON, &cblk->flags);
@@ -913,7 +934,7 @@
if (mActive && (cblk->flags & CBLK_DISABLED_MSK)) {
android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags);
ALOGW("obtainBuffer() track %p disabled, restarting", this);
- mAudioTrack->start();
+ mAudioTrack->start(0); // callback thread hasn't changed
}
cblk->waitTimeMs = 0;
@@ -955,9 +976,11 @@
{
if (mSharedBuffer != 0) return INVALID_OPERATION;
+ if (mIsTimed) return INVALID_OPERATION;
if (ssize_t(userSize) < 0) {
- // sanity-check. user is most-likely passing an error code.
+ // Sanity-check: user is most-likely passing an error code, and it would
+ // make the return value ambiguous (actualSize vs error).
ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)",
buffer, userSize, userSize);
return BAD_VALUE;
@@ -968,20 +991,18 @@
// acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
// while we are accessing the cblk
mLock.lock();
- sp <IAudioTrack> audioTrack = mAudioTrack;
- sp <IMemory> iMem = mCblkMemory;
+ sp<IAudioTrack> audioTrack = mAudioTrack;
+ sp<IMemory> iMem = mCblkMemory;
mLock.unlock();
ssize_t written = 0;
const int8_t *src = (const int8_t *)buffer;
Buffer audioBuffer;
- size_t frameSz = (size_t)frameSize();
+ size_t frameSz = frameSize();
do {
audioBuffer.frameCount = userSize/frameSz;
- // Calling obtainBuffer() with a negative wait count causes
- // an (almost) infinite wait time.
status_t err = obtainBuffer(&audioBuffer, -1);
if (err < 0) {
// out of buffers, return #bytes written
@@ -995,12 +1016,7 @@
if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) {
// Divide capacity by 2 to take expansion into account
toWrite = audioBuffer.size>>1;
- // 8 to 16 bit conversion
- int count = toWrite;
- int16_t *dst = (int16_t *)(audioBuffer.i8);
- while(count--) {
- *dst++ = (int16_t)(*src++^0x80) << 8;
- }
+ memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) src, toWrite);
} else {
toWrite = audioBuffer.size;
memcpy(audioBuffer.i8, src, toWrite);
@@ -1017,6 +1033,59 @@
// -------------------------------------------------------------------------
+TimedAudioTrack::TimedAudioTrack() {
+ mIsTimed = true;
+}
+
+status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer)
+{
+ status_t result = UNKNOWN_ERROR;
+
+ // If the track is not invalid already, try to allocate a buffer. alloc
+ // fails indicating that the server is dead, flag the track as invalid so
+ // we can attempt to restore in in just a bit.
+ if (!(mCblk->flags & CBLK_INVALID_MSK)) {
+ result = mAudioTrack->allocateTimedBuffer(size, buffer);
+ if (result == DEAD_OBJECT) {
+ android_atomic_or(CBLK_INVALID_ON, &mCblk->flags);
+ }
+ }
+
+ // If the track is invalid at this point, attempt to restore it. and try the
+ // allocation one more time.
+ if (mCblk->flags & CBLK_INVALID_MSK) {
+ mCblk->lock.lock();
+ result = restoreTrack_l(mCblk, false);
+ mCblk->lock.unlock();
+
+ if (result == OK)
+ result = mAudioTrack->allocateTimedBuffer(size, buffer);
+ }
+
+ return result;
+}
+
+status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer,
+ int64_t pts)
+{
+ // restart track if it was disabled by audioflinger due to previous underrun
+ if (mActive && (mCblk->flags & CBLK_DISABLED_MSK)) {
+ android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags);
+ ALOGW("queueTimedBuffer() track %p disabled, restarting", this);
+ mAudioTrack->start(0);
+ }
+
+ return mAudioTrack->queueTimedBuffer(buffer, pts);
+}
+
+status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform,
+ TargetTimeline target)
+{
+ return mAudioTrack->setMediaTimeTransform(xform, target);
+}
+
+// -------------------------------------------------------------------------
+
bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
{
Buffer audioBuffer;
@@ -1026,13 +1095,14 @@
mLock.lock();
// acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed
// while we are accessing the cblk
- sp <IAudioTrack> audioTrack = mAudioTrack;
- sp <IMemory> iMem = mCblkMemory;
+ sp<IAudioTrack> audioTrack = mAudioTrack;
+ sp<IMemory> iMem = mCblkMemory;
audio_track_cblk_t* cblk = mCblk;
+ bool active = mActive;
mLock.unlock();
// Manage underrun callback
- if (mActive && (cblk->framesAvailable() == cblk->frameCount)) {
+ if (active && (cblk->framesAvailable() == cblk->frameCount)) {
ALOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags);
if (!(android_atomic_or(CBLK_UNDERRUN_ON, &cblk->flags) & CBLK_UNDERRUN_MSK)) {
mCbf(EVENT_UNDERRUN, mUserData, 0);
@@ -1075,6 +1145,9 @@
frames = mRemainingFrames;
}
+ // See description of waitCount parameter at declaration of obtainBuffer().
+ // The logic below prevents us from being stuck below at obtainBuffer()
+ // not being able to handle timed events (position, markers, loops).
int32_t waitCount = -1;
if (mUpdatePeriod || (!mMarkerReached && mMarkerPosition) || mLoopCount) {
waitCount = 1;
@@ -1084,9 +1157,6 @@
audioBuffer.frameCount = frames;
- // Calling obtainBuffer() with a wait count of 1
- // limits wait time to WAIT_PERIOD_MS. This prevents from being
- // stuck here not being able to handle timed events (position, markers, loops).
status_t err = obtainBuffer(&audioBuffer, waitCount);
if (err < NO_ERROR) {
if (err != TIMED_OUT) {
@@ -1120,19 +1190,14 @@
if (writtenSize > reqSize) writtenSize = reqSize;
if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) {
- // 8 to 16 bit conversion
- const int8_t *src = audioBuffer.i8 + writtenSize-1;
- int count = writtenSize;
- int16_t *dst = audioBuffer.i16 + writtenSize-1;
- while(count--) {
- *dst-- = (int16_t)(*src--^0x80) << 8;
- }
+ // 8 to 16 bit conversion, note that source and destination are the same address
+ memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize);
writtenSize <<= 1;
}
audioBuffer.size = writtenSize;
// NOTE: mCblk->frameSize is not equal to AudioTrack::frameSize() for
- // 8 bit PCM data: in this case, mCblk->frameSize is based on a sampel size of
+ // 8 bit PCM data: in this case, mCblk->frameSize is based on a sample size of
// 16 bit.
audioBuffer.frameCount = writtenSize/mCblk->frameSize;
@@ -1159,7 +1224,7 @@
if (!(android_atomic_or(CBLK_RESTORING_ON, &cblk->flags) & CBLK_RESTORING_MSK)) {
ALOGW("dead IAudioTrack, creating a new one from %s TID %d",
- fromStart ? "start()" : "obtainBuffer()", gettid());
+ fromStart ? "start()" : "obtainBuffer()", gettid());
// signal old cblk condition so that other threads waiting for available buffers stop
// waiting now
@@ -1213,7 +1278,7 @@
}
}
if (mActive) {
- result = mAudioTrack->start();
+ result = mAudioTrack->start(0); // callback thread hasn't changed
ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result);
}
if (fromStart && result == NO_ERROR) {
@@ -1245,7 +1310,7 @@
}
}
ALOGV("restoreTrack_l() status %d mActive %d cblk %p, old cblk %p flags %08x old flags %08x",
- result, mActive, mCblk, cblk, mCblk->flags, cblk->flags);
+ result, mActive, mCblk, cblk, mCblk->flags, cblk->flags);
if (result == NO_ERROR) {
// from now on we switch to the newly created cblk
@@ -1304,15 +1369,15 @@
audio_track_cblk_t::audio_track_cblk_t()
: lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0),
- userBase(0), serverBase(0), buffers(0), frameCount(0),
- loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), volumeLR(0),
- sendLevel(0), flags(0)
+ userBase(0), serverBase(0), buffers(NULL), frameCount(0),
+ loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), mVolumeLR(0x10001000),
+ mSendLevel(0), flags(0)
{
}
uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount)
{
- uint32_t u = this->user;
+ uint32_t u = user;
u += frameCount;
// Ensure that user is never ahead of server for AudioRecord
@@ -1321,16 +1386,16 @@
if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) {
bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
}
- } else if (u > this->server) {
- ALOGW("stepServer occured after track reset");
- u = this->server;
+ } else if (u > server) {
+ ALOGW("stepServer occurred after track reset");
+ u = server;
}
if (u >= userBase + this->frameCount) {
userBase += this->frameCount;
}
- this->user = u;
+ user = u;
// Clear flow control error condition as new data has been written/read to/from buffer.
if (flags & CBLK_UNDERRUN_MSK) {
@@ -1347,7 +1412,7 @@
return false;
}
- uint32_t s = this->server;
+ uint32_t s = server;
s += frameCount;
if (flags & CBLK_DIRECTION_MSK) {
@@ -1360,9 +1425,9 @@
// while the mixer is processing a block: in this case,
// stepServer() is called After the flush() has reset u & s and
// we have s > u
- if (s > this->user) {
- ALOGW("stepServer occured after track reset");
- s = this->user;
+ if (s > user) {
+ ALOGW("stepServer occurred after track reset");
+ s = user;
}
}
@@ -1378,7 +1443,7 @@
serverBase += this->frameCount;
}
- this->server = s;
+ server = s;
if (!(flags & CBLK_INVALID_MSK)) {
cv.signal();
@@ -1389,7 +1454,7 @@
void* audio_track_cblk_t::buffer(uint32_t offset) const
{
- return (int8_t *)this->buffers + (offset - userBase) * this->frameSize;
+ return (int8_t *)buffers + (offset - userBase) * frameSize;
}
uint32_t audio_track_cblk_t::framesAvailable()
@@ -1400,8 +1465,8 @@
uint32_t audio_track_cblk_t::framesAvailable_l()
{
- uint32_t u = this->user;
- uint32_t s = this->server;
+ uint32_t u = user;
+ uint32_t s = server;
if (flags & CBLK_DIRECTION_MSK) {
uint32_t limit = (s < loopStart) ? s : loopStart;
@@ -1413,8 +1478,8 @@
uint32_t audio_track_cblk_t::framesReady()
{
- uint32_t u = this->user;
- uint32_t s = this->server;
+ uint32_t u = user;
+ uint32_t s = server;
if (flags & CBLK_DIRECTION_MSK) {
if (u < loopEnd) {
@@ -1459,4 +1524,3 @@
// -------------------------------------------------------------------------
}; // namespace android
-
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index abd491f..07b12e4 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -82,14 +82,15 @@
virtual sp<IAudioTrack> createTrack(
pid_t pid,
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
- int output,
+ audio_io_handle_t output,
+ bool isTimed,
int *sessionId,
status_t *status)
{
@@ -97,14 +98,15 @@
sp<IAudioTrack> track;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
data.writeInt32(pid);
- data.writeInt32(streamType);
+ data.writeInt32((int32_t) streamType);
data.writeInt32(sampleRate);
data.writeInt32(format);
data.writeInt32(channelMask);
data.writeInt32(frameCount);
data.writeInt32(flags);
data.writeStrongBinder(sharedBuffer->asBinder());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
+ data.writeInt32(isTimed);
int lSessionId = 0;
if (sessionId != NULL) {
lSessionId = *sessionId;
@@ -129,9 +131,9 @@
virtual sp<IAudioRecord> openRecord(
pid_t pid,
- int input,
+ audio_io_handle_t input,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
@@ -142,7 +144,7 @@
sp<IAudioRecord> record;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
data.writeInt32(pid);
- data.writeInt32(input);
+ data.writeInt32((int32_t) input);
data.writeInt32(sampleRate);
data.writeInt32(format);
data.writeInt32(channelMask);
@@ -170,47 +172,47 @@
return record;
}
- virtual uint32_t sampleRate(int output) const
+ virtual uint32_t sampleRate(audio_io_handle_t output) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(SAMPLE_RATE, data, &reply);
return reply.readInt32();
}
- virtual int channelCount(int output) const
+ virtual int channelCount(audio_io_handle_t output) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(CHANNEL_COUNT, data, &reply);
return reply.readInt32();
}
- virtual uint32_t format(int output) const
+ virtual audio_format_t format(audio_io_handle_t output) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(FORMAT, data, &reply);
- return reply.readInt32();
+ return (audio_format_t) reply.readInt32();
}
- virtual size_t frameCount(int output) const
+ virtual size_t frameCount(audio_io_handle_t output) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(FRAME_COUNT, data, &reply);
return reply.readInt32();
}
- virtual uint32_t latency(int output) const
+ virtual uint32_t latency(audio_io_handle_t output) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(LATENCY, data, &reply);
return reply.readInt32();
}
@@ -249,47 +251,48 @@
return reply.readInt32();
}
- virtual status_t setStreamVolume(int stream, float value, int output)
+ virtual status_t setStreamVolume(audio_stream_type_t stream, float value,
+ audio_io_handle_t output)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(stream);
+ data.writeInt32((int32_t) stream);
data.writeFloat(value);
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(SET_STREAM_VOLUME, data, &reply);
return reply.readInt32();
}
- virtual status_t setStreamMute(int stream, bool muted)
+ virtual status_t setStreamMute(audio_stream_type_t stream, bool muted)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(stream);
+ data.writeInt32((int32_t) stream);
data.writeInt32(muted);
remote()->transact(SET_STREAM_MUTE, data, &reply);
return reply.readInt32();
}
- virtual float streamVolume(int stream, int output) const
+ virtual float streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(stream);
- data.writeInt32(output);
+ data.writeInt32((int32_t) stream);
+ data.writeInt32((int32_t) output);
remote()->transact(STREAM_VOLUME, data, &reply);
return reply.readFloat();
}
- virtual bool streamMute(int stream) const
+ virtual bool streamMute(audio_stream_type_t stream) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(stream);
+ data.writeInt32((int32_t) stream);
remote()->transact(STREAM_MUTE, data, &reply);
return reply.readInt32();
}
- virtual status_t setMode(int mode)
+ virtual status_t setMode(audio_mode_t mode)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -315,21 +318,21 @@
return reply.readInt32();
}
- virtual status_t setParameters(int ioHandle, const String8& keyValuePairs)
+ virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(ioHandle);
+ data.writeInt32((int32_t) ioHandle);
data.writeString8(keyValuePairs);
remote()->transact(SET_PARAMETERS, data, &reply);
return reply.readInt32();
}
- virtual String8 getParameters(int ioHandle, const String8& keys)
+ virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(ioHandle);
+ data.writeInt32((int32_t) ioHandle);
data.writeString8(keys);
remote()->transact(GET_PARAMETERS, data, &reply);
return reply.readString8();
@@ -343,7 +346,7 @@
remote()->transact(REGISTER_CLIENT, data, &reply);
}
- virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
+ virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -354,17 +357,17 @@
return reply.readInt32();
}
- virtual int openOutput(uint32_t *pDevices,
+ virtual audio_io_handle_t openOutput(uint32_t *pDevices,
uint32_t *pSamplingRate,
- uint32_t *pFormat,
+ audio_format_t *pFormat,
uint32_t *pChannels,
uint32_t *pLatencyMs,
- uint32_t flags)
+ audio_policy_output_flags_t flags)
{
Parcel data, reply;
uint32_t devices = pDevices ? *pDevices : 0;
uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
- uint32_t format = pFormat ? *pFormat : 0;
+ audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT;
uint32_t channels = pChannels ? *pChannels : 0;
uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
@@ -374,15 +377,15 @@
data.writeInt32(format);
data.writeInt32(channels);
data.writeInt32(latency);
- data.writeInt32(flags);
+ data.writeInt32((int32_t) flags);
remote()->transact(OPEN_OUTPUT, data, &reply);
- int output = reply.readInt32();
- ALOGV("openOutput() returned output, %p", output);
+ audio_io_handle_t output = (audio_io_handle_t) reply.readInt32();
+ ALOGV("openOutput() returned output, %d", output);
devices = reply.readInt32();
if (pDevices) *pDevices = devices;
samplingRate = reply.readInt32();
if (pSamplingRate) *pSamplingRate = samplingRate;
- format = reply.readInt32();
+ format = (audio_format_t) reply.readInt32();
if (pFormat) *pFormat = format;
channels = reply.readInt32();
if (pChannels) *pChannels = channels;
@@ -391,53 +394,54 @@
return output;
}
- virtual int openDuplicateOutput(int output1, int output2)
+ virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
+ audio_io_handle_t output2)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output1);
- data.writeInt32(output2);
+ data.writeInt32((int32_t) output1);
+ data.writeInt32((int32_t) output2);
remote()->transact(OPEN_DUPLICATE_OUTPUT, data, &reply);
- return reply.readInt32();
+ return (audio_io_handle_t) reply.readInt32();
}
- virtual status_t closeOutput(int output)
+ virtual status_t closeOutput(audio_io_handle_t output)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(CLOSE_OUTPUT, data, &reply);
return reply.readInt32();
}
- virtual status_t suspendOutput(int output)
+ virtual status_t suspendOutput(audio_io_handle_t output)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(SUSPEND_OUTPUT, data, &reply);
return reply.readInt32();
}
- virtual status_t restoreOutput(int output)
+ virtual status_t restoreOutput(audio_io_handle_t output)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(RESTORE_OUTPUT, data, &reply);
return reply.readInt32();
}
- virtual int openInput(uint32_t *pDevices,
+ virtual audio_io_handle_t openInput(uint32_t *pDevices,
uint32_t *pSamplingRate,
- uint32_t *pFormat,
+ audio_format_t *pFormat,
uint32_t *pChannels,
- uint32_t acoustics)
+ audio_in_acoustics_t acoustics)
{
Parcel data, reply;
uint32_t devices = pDevices ? *pDevices : 0;
uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
- uint32_t format = pFormat ? *pFormat : 0;
+ audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT;
uint32_t channels = pChannels ? *pChannels : 0;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -445,14 +449,14 @@
data.writeInt32(samplingRate);
data.writeInt32(format);
data.writeInt32(channels);
- data.writeInt32(acoustics);
+ data.writeInt32((int32_t) acoustics);
remote()->transact(OPEN_INPUT, data, &reply);
- int input = reply.readInt32();
+ audio_io_handle_t input = (audio_io_handle_t) reply.readInt32();
devices = reply.readInt32();
if (pDevices) *pDevices = devices;
samplingRate = reply.readInt32();
if (pSamplingRate) *pSamplingRate = samplingRate;
- format = reply.readInt32();
+ format = (audio_format_t) reply.readInt32();
if (pFormat) *pFormat = format;
channels = reply.readInt32();
if (pChannels) *pChannels = channels;
@@ -468,12 +472,12 @@
return reply.readInt32();
}
- virtual status_t setStreamOutput(uint32_t stream, int output)
+ virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(stream);
- data.writeInt32(output);
+ data.writeInt32((int32_t) stream);
+ data.writeInt32((int32_t) output);
remote()->transact(SET_STREAM_OUTPUT, data, &reply);
return reply.readInt32();
}
@@ -487,11 +491,12 @@
return reply.readInt32();
}
- virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output)
+ virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
+ audio_io_handle_t output) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(GET_RENDER_POSITION, data, &reply);
status_t status = reply.readInt32();
if (status == NO_ERROR) {
@@ -507,11 +512,11 @@
return status;
}
- virtual unsigned int getInputFramesLost(int ioHandle)
+ virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(ioHandle);
+ data.writeInt32((int32_t) ioHandle);
remote()->transact(GET_INPUT_FRAMES_LOST, data, &reply);
return reply.readInt32();
}
@@ -544,7 +549,7 @@
remote()->transact(RELEASE_AUDIO_SESSION_ID, data, &reply);
}
- virtual status_t queryNumberEffects(uint32_t *numEffects)
+ virtual status_t queryNumberEffects(uint32_t *numEffects) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -556,13 +561,13 @@
if (status != NO_ERROR) {
return status;
}
- if (numEffects) {
+ if (numEffects != NULL) {
*numEffects = (uint32_t)reply.readInt32();
}
return NO_ERROR;
}
- virtual status_t queryEffect(uint32_t index, effect_descriptor_t *pDescriptor)
+ virtual status_t queryEffect(uint32_t index, effect_descriptor_t *pDescriptor) const
{
if (pDescriptor == NULL) {
return BAD_VALUE;
@@ -582,7 +587,8 @@
return NO_ERROR;
}
- virtual status_t getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *pDescriptor)
+ virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid,
+ effect_descriptor_t *pDescriptor) const
{
if (pUuid == NULL || pDescriptor == NULL) {
return BAD_VALUE;
@@ -606,7 +612,7 @@
effect_descriptor_t *pDesc,
const sp<IEffectClient>& client,
int32_t priority,
- int output,
+ audio_io_handle_t output,
int sessionId,
status_t *status,
int *id,
@@ -616,18 +622,18 @@
sp<IEffect> effect;
if (pDesc == NULL) {
- return effect;
- if (status) {
- *status = BAD_VALUE;
- }
- }
+ return effect;
+ if (status) {
+ *status = BAD_VALUE;
+ }
+ }
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
data.writeInt32(pid);
data.write(pDesc, sizeof(effect_descriptor_t));
data.writeStrongBinder(client->asBinder());
data.writeInt32(priority);
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
data.writeInt32(sessionId);
status_t lStatus = remote()->transact(CREATE_EFFECT, data, &reply);
@@ -640,7 +646,7 @@
*id = tmp;
}
tmp = reply.readInt32();
- if (enabled) {
+ if (enabled != NULL) {
*enabled = tmp;
}
effect = interface_cast<IEffect>(reply.readStrongBinder());
@@ -653,13 +659,14 @@
return effect;
}
- virtual status_t moveEffects(int session, int srcOutput, int dstOutput)
+ virtual status_t moveEffects(int session, audio_io_handle_t srcOutput,
+ audio_io_handle_t dstOutput)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
data.writeInt32(session);
- data.writeInt32(srcOutput);
- data.writeInt32(dstOutput);
+ data.writeInt32((int32_t) srcOutput);
+ data.writeInt32((int32_t) dstOutput);
remote()->transact(MOVE_EFFECTS, data, &reply);
return reply.readInt32();
}
@@ -672,23 +679,24 @@
status_t BnAudioFlinger::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
- switch(code) {
+ switch (code) {
case CREATE_TRACK: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
pid_t pid = data.readInt32();
int streamType = data.readInt32();
uint32_t sampleRate = data.readInt32();
- int format = data.readInt32();
+ audio_format_t format = (audio_format_t) data.readInt32();
int channelCount = data.readInt32();
size_t bufferCount = data.readInt32();
uint32_t flags = data.readInt32();
sp<IMemory> buffer = interface_cast<IMemory>(data.readStrongBinder());
- int output = data.readInt32();
+ audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
+ bool isTimed = data.readInt32();
int sessionId = data.readInt32();
status_t status;
sp<IAudioTrack> track = createTrack(pid,
- streamType, sampleRate, format,
- channelCount, bufferCount, flags, buffer, output, &sessionId, &status);
+ (audio_stream_type_t) streamType, sampleRate, format,
+ channelCount, bufferCount, flags, buffer, output, isTimed, &sessionId, &status);
reply->writeInt32(sessionId);
reply->writeInt32(status);
reply->writeStrongBinder(track->asBinder());
@@ -697,9 +705,9 @@
case OPEN_RECORD: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
pid_t pid = data.readInt32();
- int input = data.readInt32();
+ audio_io_handle_t input = (audio_io_handle_t) data.readInt32();
uint32_t sampleRate = data.readInt32();
- int format = data.readInt32();
+ audio_format_t format = (audio_format_t) data.readInt32();
int channelCount = data.readInt32();
size_t bufferCount = data.readInt32();
uint32_t flags = data.readInt32();
@@ -714,30 +722,30 @@
} break;
case SAMPLE_RATE: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32( sampleRate(data.readInt32()) );
+ reply->writeInt32( sampleRate((audio_io_handle_t) data.readInt32()) );
return NO_ERROR;
} break;
case CHANNEL_COUNT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32( channelCount(data.readInt32()) );
+ reply->writeInt32( channelCount((audio_io_handle_t) data.readInt32()) );
return NO_ERROR;
} break;
case FORMAT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32( format(data.readInt32()) );
+ reply->writeInt32( format((audio_io_handle_t) data.readInt32()) );
return NO_ERROR;
} break;
case FRAME_COUNT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32( frameCount(data.readInt32()) );
+ reply->writeInt32( frameCount((audio_io_handle_t) data.readInt32()) );
return NO_ERROR;
} break;
case LATENCY: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32( latency(data.readInt32()) );
+ reply->writeInt32( latency((audio_io_handle_t) data.readInt32()) );
return NO_ERROR;
} break;
- case SET_MASTER_VOLUME: {
+ case SET_MASTER_VOLUME: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
reply->writeInt32( setMasterVolume(data.readFloat()) );
return NO_ERROR;
@@ -761,32 +769,32 @@
CHECK_INTERFACE(IAudioFlinger, data, reply);
int stream = data.readInt32();
float volume = data.readFloat();
- int output = data.readInt32();
- reply->writeInt32( setStreamVolume(stream, volume, output) );
+ audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
+ reply->writeInt32( setStreamVolume((audio_stream_type_t) stream, volume, output) );
return NO_ERROR;
} break;
case SET_STREAM_MUTE: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
int stream = data.readInt32();
- reply->writeInt32( setStreamMute(stream, data.readInt32()) );
+ reply->writeInt32( setStreamMute((audio_stream_type_t) stream, data.readInt32()) );
return NO_ERROR;
} break;
case STREAM_VOLUME: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
int stream = data.readInt32();
int output = data.readInt32();
- reply->writeFloat( streamVolume(stream, output) );
+ reply->writeFloat( streamVolume((audio_stream_type_t) stream, output) );
return NO_ERROR;
} break;
case STREAM_MUTE: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
int stream = data.readInt32();
- reply->writeInt32( streamMute(stream) );
+ reply->writeInt32( streamMute((audio_stream_type_t) stream) );
return NO_ERROR;
} break;
case SET_MODE: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- int mode = data.readInt32();
+ audio_mode_t mode = (audio_mode_t) data.readInt32();
reply->writeInt32( setMode(mode) );
return NO_ERROR;
} break;
@@ -803,18 +811,18 @@
} break;
case SET_PARAMETERS: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- int ioHandle = data.readInt32();
+ audio_io_handle_t ioHandle = (audio_io_handle_t) data.readInt32();
String8 keyValuePairs(data.readString8());
reply->writeInt32(setParameters(ioHandle, keyValuePairs));
return NO_ERROR;
- } break;
+ } break;
case GET_PARAMETERS: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- int ioHandle = data.readInt32();
+ audio_io_handle_t ioHandle = (audio_io_handle_t) data.readInt32();
String8 keys(data.readString8());
reply->writeString8(getParameters(ioHandle, keys));
return NO_ERROR;
- } break;
+ } break;
case REGISTER_CLIENT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
@@ -825,7 +833,7 @@
case GET_INPUTBUFFERSIZE: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
uint32_t sampleRate = data.readInt32();
- int format = data.readInt32();
+ audio_format_t format = (audio_format_t) data.readInt32();
int channelCount = data.readInt32();
reply->writeInt32( getInputBufferSize(sampleRate, format, channelCount) );
return NO_ERROR;
@@ -834,18 +842,18 @@
CHECK_INTERFACE(IAudioFlinger, data, reply);
uint32_t devices = data.readInt32();
uint32_t samplingRate = data.readInt32();
- uint32_t format = data.readInt32();
+ audio_format_t format = (audio_format_t) data.readInt32();
uint32_t channels = data.readInt32();
uint32_t latency = data.readInt32();
- uint32_t flags = data.readInt32();
- int output = openOutput(&devices,
+ audio_policy_output_flags_t flags = (audio_policy_output_flags_t) data.readInt32();
+ audio_io_handle_t output = openOutput(&devices,
&samplingRate,
&format,
&channels,
&latency,
flags);
ALOGV("OPEN_OUTPUT output, %p", output);
- reply->writeInt32(output);
+ reply->writeInt32((int32_t) output);
reply->writeInt32(devices);
reply->writeInt32(samplingRate);
reply->writeInt32(format);
@@ -855,40 +863,40 @@
} break;
case OPEN_DUPLICATE_OUTPUT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- int output1 = data.readInt32();
- int output2 = data.readInt32();
- reply->writeInt32(openDuplicateOutput(output1, output2));
+ audio_io_handle_t output1 = (audio_io_handle_t) data.readInt32();
+ audio_io_handle_t output2 = (audio_io_handle_t) data.readInt32();
+ reply->writeInt32((int32_t) openDuplicateOutput(output1, output2));
return NO_ERROR;
} break;
case CLOSE_OUTPUT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32(closeOutput(data.readInt32()));
+ reply->writeInt32(closeOutput((audio_io_handle_t) data.readInt32()));
return NO_ERROR;
} break;
case SUSPEND_OUTPUT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32(suspendOutput(data.readInt32()));
+ reply->writeInt32(suspendOutput((audio_io_handle_t) data.readInt32()));
return NO_ERROR;
} break;
case RESTORE_OUTPUT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32(restoreOutput(data.readInt32()));
+ reply->writeInt32(restoreOutput((audio_io_handle_t) data.readInt32()));
return NO_ERROR;
} break;
case OPEN_INPUT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
uint32_t devices = data.readInt32();
uint32_t samplingRate = data.readInt32();
- uint32_t format = data.readInt32();
+ audio_format_t format = (audio_format_t) data.readInt32();
uint32_t channels = data.readInt32();
- uint32_t acoutics = data.readInt32();
+ audio_in_acoustics_t acoustics = (audio_in_acoustics_t) data.readInt32();
- int input = openInput(&devices,
+ audio_io_handle_t input = openInput(&devices,
&samplingRate,
&format,
&channels,
- acoutics);
- reply->writeInt32(input);
+ acoustics);
+ reply->writeInt32((int32_t) input);
reply->writeInt32(devices);
reply->writeInt32(samplingRate);
reply->writeInt32(format);
@@ -897,14 +905,14 @@
} break;
case CLOSE_INPUT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32(closeInput(data.readInt32()));
+ reply->writeInt32(closeInput((audio_io_handle_t) data.readInt32()));
return NO_ERROR;
} break;
case SET_STREAM_OUTPUT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
uint32_t stream = data.readInt32();
- int output = data.readInt32();
- reply->writeInt32(setStreamOutput(stream, output));
+ audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
+ reply->writeInt32(setStreamOutput((audio_stream_type_t) stream, output));
return NO_ERROR;
} break;
case SET_VOICE_VOLUME: {
@@ -915,7 +923,7 @@
} break;
case GET_RENDER_POSITION: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- int output = data.readInt32();
+ audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
uint32_t halFrames;
uint32_t dspFrames;
status_t status = getRenderPosition(&halFrames, &dspFrames, output);
@@ -928,7 +936,7 @@
}
case GET_INPUT_FRAMES_LOST: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- int ioHandle = data.readInt32();
+ audio_io_handle_t ioHandle = (audio_io_handle_t) data.readInt32();
reply->writeInt32(getInputFramesLost(ioHandle));
return NO_ERROR;
} break;
@@ -988,7 +996,7 @@
data.read(&desc, sizeof(effect_descriptor_t));
sp<IEffectClient> client = interface_cast<IEffectClient>(data.readStrongBinder());
int32_t priority = data.readInt32();
- int output = data.readInt32();
+ audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
int sessionId = data.readInt32();
status_t status;
int id;
@@ -1005,8 +1013,8 @@
case MOVE_EFFECTS: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
int session = data.readInt32();
- int srcOutput = data.readInt32();
- int dstOutput = data.readInt32();
+ audio_io_handle_t srcOutput = (audio_io_handle_t) data.readInt32();
+ audio_io_handle_t dstOutput = (audio_io_handle_t) data.readInt32();
reply->writeInt32(moveEffects(session, srcOutput, dstOutput));
return NO_ERROR;
} break;
diff --git a/media/libmedia/IAudioFlingerClient.cpp b/media/libmedia/IAudioFlingerClient.cpp
index 5a3f250..4178b29 100644
--- a/media/libmedia/IAudioFlingerClient.cpp
+++ b/media/libmedia/IAudioFlingerClient.cpp
@@ -39,18 +39,18 @@
{
}
- void ioConfigChanged(int event, int ioHandle, void *param2)
+ void ioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlingerClient::getInterfaceDescriptor());
data.writeInt32(event);
- data.writeInt32(ioHandle);
+ data.writeInt32((int32_t) ioHandle);
if (event == AudioSystem::STREAM_CONFIG_CHANGED) {
- uint32_t stream = *(uint32_t *)param2;
+ uint32_t stream = *(const uint32_t *)param2;
ALOGV("ioConfigChanged stream %d", stream);
data.writeInt32(stream);
} else if (event != AudioSystem::OUTPUT_CLOSED && event != AudioSystem::INPUT_CLOSED) {
- AudioSystem::OutputDescriptor *desc = (AudioSystem::OutputDescriptor *)param2;
+ const AudioSystem::OutputDescriptor *desc = (const AudioSystem::OutputDescriptor *)param2;
data.writeInt32(desc->samplingRate);
data.writeInt32(desc->format);
data.writeInt32(desc->channels);
@@ -68,12 +68,12 @@
status_t BnAudioFlingerClient::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
- switch(code) {
+ switch (code) {
case IO_CONFIG_CHANGED: {
CHECK_INTERFACE(IAudioFlingerClient, data, reply);
int event = data.readInt32();
- int ioHandle = data.readInt32();
- void *param2 = 0;
+ audio_io_handle_t ioHandle = (audio_io_handle_t) data.readInt32();
+ const void *param2 = NULL;
AudioSystem::OutputDescriptor desc;
uint32_t stream;
if (event == AudioSystem::STREAM_CONFIG_CHANGED) {
diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp
index 50b4855..5040bd9 100644
--- a/media/libmedia/IAudioPolicyService.cpp
+++ b/media/libmedia/IAudioPolicyService.cpp
@@ -33,7 +33,7 @@
SET_DEVICE_CONNECTION_STATE = IBinder::FIRST_CALL_TRANSACTION,
GET_DEVICE_CONNECTION_STATE,
SET_PHONE_STATE,
- SET_RINGER_MODE,
+ SET_RINGER_MODE, // reserved, no longer used
SET_FORCE_USE,
GET_FORCE_USE,
GET_OUTPUT,
@@ -91,7 +91,7 @@
return static_cast <audio_policy_dev_state_t>(reply.readInt32());
}
- virtual status_t setPhoneState(int state)
+ virtual status_t setPhoneState(audio_mode_t state)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
@@ -100,16 +100,6 @@
return static_cast <status_t> (reply.readInt32());
}
- virtual status_t setRingerMode(uint32_t mode, uint32_t mask)
- {
- Parcel data, reply;
- data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
- data.writeInt32(mode);
- data.writeInt32(mask);
- remote()->transact(SET_RINGER_MODE, data, &reply);
- return static_cast <status_t> (reply.readInt32());
- }
-
virtual status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config)
{
Parcel data, reply;
@@ -132,7 +122,7 @@
virtual audio_io_handle_t getOutput(
audio_stream_type_t stream,
uint32_t samplingRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channels,
audio_policy_output_flags_t flags)
{
@@ -154,7 +144,7 @@
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
data.writeInt32(output);
- data.writeInt32(stream);
+ data.writeInt32((int32_t) stream);
data.writeInt32(session);
remote()->transact(START_OUTPUT, data, &reply);
return static_cast <status_t> (reply.readInt32());
@@ -167,7 +157,7 @@
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
data.writeInt32(output);
- data.writeInt32(stream);
+ data.writeInt32((int32_t) stream);
data.writeInt32(session);
remote()->transact(STOP_OUTPUT, data, &reply);
return static_cast <status_t> (reply.readInt32());
@@ -182,16 +172,16 @@
}
virtual audio_io_handle_t getInput(
- int inputSource,
+ audio_source_t inputSource,
uint32_t samplingRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channels,
audio_in_acoustics_t acoustics,
int audioSession)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
- data.writeInt32(inputSource);
+ data.writeInt32((int32_t) inputSource);
data.writeInt32(samplingRate);
data.writeInt32(static_cast <uint32_t>(format));
data.writeInt32(channels);
@@ -240,21 +230,28 @@
return static_cast <status_t> (reply.readInt32());
}
- virtual status_t setStreamVolumeIndex(audio_stream_type_t stream, int index)
+ virtual status_t setStreamVolumeIndex(audio_stream_type_t stream,
+ int index,
+ audio_devices_t device)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
data.writeInt32(static_cast <uint32_t>(stream));
data.writeInt32(index);
+ data.writeInt32(static_cast <uint32_t>(device));
remote()->transact(SET_STREAM_VOLUME, data, &reply);
return static_cast <status_t> (reply.readInt32());
}
- virtual status_t getStreamVolumeIndex(audio_stream_type_t stream, int *index)
+ virtual status_t getStreamVolumeIndex(audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
data.writeInt32(static_cast <uint32_t>(stream));
+ data.writeInt32(static_cast <uint32_t>(device));
+
remote()->transact(GET_STREAM_VOLUME, data, &reply);
int lIndex = reply.readInt32();
if (index) *index = lIndex;
@@ -270,13 +267,13 @@
return reply.readInt32();
}
- virtual uint32_t getDevicesForStream(audio_stream_type_t stream)
+ virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
data.writeInt32(static_cast <uint32_t>(stream));
remote()->transact(GET_DEVICES_FOR_STREAM, data, &reply);
- return (uint32_t) reply.readInt32();
+ return (audio_devices_t) reply.readInt32();
}
virtual audio_io_handle_t getOutputForEffect(effect_descriptor_t *desc)
@@ -324,11 +321,11 @@
return static_cast <status_t> (reply.readInt32());
}
- virtual bool isStreamActive(int stream, uint32_t inPastMs) const
+ virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
- data.writeInt32(stream);
+ data.writeInt32((int32_t) stream);
data.writeInt32(inPastMs);
remote()->transact(IS_STREAM_ACTIVE, data, &reply);
return reply.readInt32();
@@ -368,7 +365,7 @@
status_t BnAudioPolicyService::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
- switch(code) {
+ switch (code) {
case SET_DEVICE_CONNECTION_STATE: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
audio_devices_t device =
@@ -394,15 +391,7 @@
case SET_PHONE_STATE: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
- reply->writeInt32(static_cast <uint32_t>(setPhoneState(data.readInt32())));
- return NO_ERROR;
- } break;
-
- case SET_RINGER_MODE: {
- CHECK_INTERFACE(IAudioPolicyService, data, reply);
- uint32_t mode = data.readInt32();
- uint32_t mask = data.readInt32();
- reply->writeInt32(static_cast <uint32_t>(setRingerMode(mode, mask)));
+ reply->writeInt32(static_cast <uint32_t>(setPhoneState((audio_mode_t) data.readInt32())));
return NO_ERROR;
} break;
@@ -427,7 +416,7 @@
audio_stream_type_t stream =
static_cast <audio_stream_type_t>(data.readInt32());
uint32_t samplingRate = data.readInt32();
- uint32_t format = data.readInt32();
+ audio_format_t format = (audio_format_t) data.readInt32();
uint32_t channels = data.readInt32();
audio_policy_output_flags_t flags =
static_cast <audio_policy_output_flags_t>(data.readInt32());
@@ -472,9 +461,9 @@
case GET_INPUT: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
- int inputSource = data.readInt32();
+ audio_source_t inputSource = (audio_source_t) data.readInt32();
uint32_t samplingRate = data.readInt32();
- uint32_t format = data.readInt32();
+ audio_format_t format = (audio_format_t) data.readInt32();
uint32_t channels = data.readInt32();
audio_in_acoustics_t acoustics =
static_cast <audio_in_acoustics_t>(data.readInt32());
@@ -525,7 +514,10 @@
audio_stream_type_t stream =
static_cast <audio_stream_type_t>(data.readInt32());
int index = data.readInt32();
- reply->writeInt32(static_cast <uint32_t>(setStreamVolumeIndex(stream, index)));
+ audio_devices_t device = static_cast <audio_devices_t>(data.readInt32());
+ reply->writeInt32(static_cast <uint32_t>(setStreamVolumeIndex(stream,
+ index,
+ device)));
return NO_ERROR;
} break;
@@ -533,8 +525,9 @@
CHECK_INTERFACE(IAudioPolicyService, data, reply);
audio_stream_type_t stream =
static_cast <audio_stream_type_t>(data.readInt32());
+ audio_devices_t device = static_cast <audio_devices_t>(data.readInt32());
int index;
- status_t status = getStreamVolumeIndex(stream, &index);
+ status_t status = getStreamVolumeIndex(stream, &index, device);
reply->writeInt32(index);
reply->writeInt32(static_cast <uint32_t>(status));
return NO_ERROR;
@@ -598,9 +591,9 @@
case IS_STREAM_ACTIVE: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
- int stream = data.readInt32();
+ audio_stream_type_t stream = (audio_stream_type_t) data.readInt32();
uint32_t inPastMs = (uint32_t)data.readInt32();
- reply->writeInt32( isStreamActive(stream, inPastMs) );
+ reply->writeInt32( isStreamActive((audio_stream_type_t) stream, inPastMs) );
return NO_ERROR;
} break;
diff --git a/media/libmedia/IAudioRecord.cpp b/media/libmedia/IAudioRecord.cpp
index 8c7a960..377b9a8 100644
--- a/media/libmedia/IAudioRecord.cpp
+++ b/media/libmedia/IAudioRecord.cpp
@@ -2,16 +2,16 @@
**
** Copyright 2007, The Android Open Source Project
**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
+** Licensed under the Apache License, Version 2.0 (the "License");
+** you may not use this file except in compliance with the License.
+** You may obtain a copy of the License at
**
-** http://www.apache.org/licenses/LICENSE-2.0
+** http://www.apache.org/licenses/LICENSE-2.0
**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-** See the License for the specific language governing permissions and
+** Unless required by applicable law or agreed to in writing, software
+** distributed under the License is distributed on an "AS IS" BASIS,
+** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+** See the License for the specific language governing permissions and
** limitations under the License.
*/
@@ -41,11 +41,12 @@
: BpInterface<IAudioRecord>(impl)
{
}
-
- virtual status_t start()
+
+ virtual status_t start(pid_t tid)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioRecord::getInterfaceDescriptor());
+ data.writeInt32(tid);
status_t status = remote()->transact(START, data, &reply);
if (status == NO_ERROR) {
status = reply.readInt32();
@@ -54,14 +55,14 @@
}
return status;
}
-
+
virtual void stop()
{
Parcel data, reply;
data.writeInterfaceToken(IAudioRecord::getInterfaceDescriptor());
remote()->transact(STOP, data, &reply);
}
-
+
virtual sp<IMemory> getCblk() const
{
Parcel data, reply;
@@ -72,7 +73,7 @@
cblk = interface_cast<IMemory>(reply.readStrongBinder());
}
return cblk;
- }
+ }
};
IMPLEMENT_META_INTERFACE(AudioRecord, "android.media.IAudioRecord");
@@ -82,15 +83,15 @@
status_t BnAudioRecord::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
- switch(code) {
- case GET_CBLK: {
+ switch (code) {
+ case GET_CBLK: {
CHECK_INTERFACE(IAudioRecord, data, reply);
reply->writeStrongBinder(getCblk()->asBinder());
return NO_ERROR;
} break;
case START: {
CHECK_INTERFACE(IAudioRecord, data, reply);
- reply->writeInt32(start());
+ reply->writeInt32(start(data.readInt32()));
return NO_ERROR;
} break;
case STOP: {
@@ -104,4 +105,3 @@
}
}; // namespace android
-
diff --git a/media/libmedia/IAudioTrack.cpp b/media/libmedia/IAudioTrack.cpp
index 0b372f3..09f31a7 100644
--- a/media/libmedia/IAudioTrack.cpp
+++ b/media/libmedia/IAudioTrack.cpp
@@ -1,17 +1,17 @@
-/* //device/extlibs/pv/android/IAudioTrack.cpp
+/*
**
** Copyright 2007, The Android Open Source Project
**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
+** Licensed under the Apache License, Version 2.0 (the "License");
+** you may not use this file except in compliance with the License.
+** You may obtain a copy of the License at
**
-** http://www.apache.org/licenses/LICENSE-2.0
+** http://www.apache.org/licenses/LICENSE-2.0
**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-** See the License for the specific language governing permissions and
+** Unless required by applicable law or agreed to in writing, software
+** distributed under the License is distributed on an "AS IS" BASIS,
+** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+** See the License for the specific language governing permissions and
** limitations under the License.
*/
@@ -35,7 +35,10 @@
FLUSH,
MUTE,
PAUSE,
- ATTACH_AUX_EFFECT
+ ATTACH_AUX_EFFECT,
+ ALLOCATE_TIMED_BUFFER,
+ QUEUE_TIMED_BUFFER,
+ SET_MEDIA_TIME_TRANSFORM,
};
class BpAudioTrack : public BpInterface<IAudioTrack>
@@ -45,11 +48,24 @@
: BpInterface<IAudioTrack>(impl)
{
}
-
- virtual status_t start()
+
+ virtual sp<IMemory> getCblk() const
+ {
+ Parcel data, reply;
+ sp<IMemory> cblk;
+ data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_CBLK, data, &reply);
+ if (status == NO_ERROR) {
+ cblk = interface_cast<IMemory>(reply.readStrongBinder());
+ }
+ return cblk;
+ }
+
+ virtual status_t start(pid_t tid)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
+ data.writeInt32(tid);
status_t status = remote()->transact(START, data, &reply);
if (status == NO_ERROR) {
status = reply.readInt32();
@@ -58,14 +74,14 @@
}
return status;
}
-
+
virtual void stop()
{
Parcel data, reply;
data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
remote()->transact(STOP, data, &reply);
}
-
+
virtual void flush()
{
Parcel data, reply;
@@ -80,25 +96,13 @@
data.writeInt32(e);
remote()->transact(MUTE, data, &reply);
}
-
+
virtual void pause()
{
Parcel data, reply;
data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
remote()->transact(PAUSE, data, &reply);
}
-
- virtual sp<IMemory> getCblk() const
- {
- Parcel data, reply;
- sp<IMemory> cblk;
- data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
- status_t status = remote()->transact(GET_CBLK, data, &reply);
- if (status == NO_ERROR) {
- cblk = interface_cast<IMemory>(reply.readStrongBinder());
- }
- return cblk;
- }
virtual status_t attachAuxEffect(int effectId)
{
@@ -113,6 +117,52 @@
}
return status;
}
+
+ virtual status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer) {
+ Parcel data, reply;
+ data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
+ data.writeInt32(size);
+ status_t status = remote()->transact(ALLOCATE_TIMED_BUFFER,
+ data, &reply);
+ if (status == NO_ERROR) {
+ status = reply.readInt32();
+ if (status == NO_ERROR) {
+ *buffer = interface_cast<IMemory>(reply.readStrongBinder());
+ }
+ }
+ return status;
+ }
+
+ virtual status_t queueTimedBuffer(const sp<IMemory>& buffer,
+ int64_t pts) {
+ Parcel data, reply;
+ data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
+ data.writeStrongBinder(buffer->asBinder());
+ data.writeInt64(pts);
+ status_t status = remote()->transact(QUEUE_TIMED_BUFFER,
+ data, &reply);
+ if (status == NO_ERROR) {
+ status = reply.readInt32();
+ }
+ return status;
+ }
+
+ virtual status_t setMediaTimeTransform(const LinearTransform& xform,
+ int target) {
+ Parcel data, reply;
+ data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
+ data.writeInt64(xform.a_zero);
+ data.writeInt64(xform.b_zero);
+ data.writeInt32(xform.a_to_b_numer);
+ data.writeInt32(xform.a_to_b_denom);
+ data.writeInt32(target);
+ status_t status = remote()->transact(SET_MEDIA_TIME_TRANSFORM,
+ data, &reply);
+ if (status == NO_ERROR) {
+ status = reply.readInt32();
+ }
+ return status;
+ }
};
IMPLEMENT_META_INTERFACE(AudioTrack, "android.media.IAudioTrack");
@@ -122,15 +172,15 @@
status_t BnAudioTrack::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
- switch(code) {
- case GET_CBLK: {
+ switch (code) {
+ case GET_CBLK: {
CHECK_INTERFACE(IAudioTrack, data, reply);
reply->writeStrongBinder(getCblk()->asBinder());
return NO_ERROR;
} break;
case START: {
CHECK_INTERFACE(IAudioTrack, data, reply);
- reply->writeInt32(start());
+ reply->writeInt32(start(data.readInt32()));
return NO_ERROR;
} break;
case STOP: {
@@ -158,10 +208,38 @@
reply->writeInt32(attachAuxEffect(data.readInt32()));
return NO_ERROR;
} break;
+ case ALLOCATE_TIMED_BUFFER: {
+ CHECK_INTERFACE(IAudioTrack, data, reply);
+ sp<IMemory> buffer;
+ status_t status = allocateTimedBuffer(data.readInt32(), &buffer);
+ reply->writeInt32(status);
+ if (status == NO_ERROR) {
+ reply->writeStrongBinder(buffer->asBinder());
+ }
+ return NO_ERROR;
+ } break;
+ case QUEUE_TIMED_BUFFER: {
+ CHECK_INTERFACE(IAudioTrack, data, reply);
+ sp<IMemory> buffer = interface_cast<IMemory>(
+ data.readStrongBinder());
+ uint64_t pts = data.readInt64();
+ reply->writeInt32(queueTimedBuffer(buffer, pts));
+ return NO_ERROR;
+ } break;
+ case SET_MEDIA_TIME_TRANSFORM: {
+ CHECK_INTERFACE(IAudioTrack, data, reply);
+ LinearTransform xform;
+ xform.a_zero = data.readInt64();
+ xform.b_zero = data.readInt64();
+ xform.a_to_b_numer = data.readInt32();
+ xform.a_to_b_denom = data.readInt32();
+ int target = data.readInt32();
+ reply->writeInt32(setMediaTimeTransform(xform, target));
+ return NO_ERROR;
+ } break;
default:
return BBinder::onTransact(code, data, reply, flags);
}
}
}; // namespace android
-
diff --git a/media/libmedia/IEffect.cpp b/media/libmedia/IEffect.cpp
index d469e28..a303a8f 100644
--- a/media/libmedia/IEffect.cpp
+++ b/media/libmedia/IEffect.cpp
@@ -83,8 +83,15 @@
size = *pReplySize;
}
data.writeInt32(size);
- remote()->transact(COMMAND, data, &reply);
- status_t status = reply.readInt32();
+
+ status_t status = remote()->transact(COMMAND, data, &reply);
+ if (status != NO_ERROR) {
+ if (pReplySize != NULL)
+ *pReplySize = 0;
+ return status;
+ }
+
+ status = reply.readInt32();
size = reply.readInt32();
if (size != 0 && pReplyData != NULL && pReplySize != NULL) {
reply.read(pReplyData, size);
@@ -122,7 +129,7 @@
status_t BnEffect::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
- switch(code) {
+ switch (code) {
case ENABLE: {
ALOGV("ENABLE");
CHECK_INTERFACE(IEffect, data, reply);
@@ -179,10 +186,10 @@
} break;
case GET_CBLK: {
- CHECK_INTERFACE(IEffect, data, reply);
- reply->writeStrongBinder(getCblk()->asBinder());
- return NO_ERROR;
- } break;
+ CHECK_INTERFACE(IEffect, data, reply);
+ reply->writeStrongBinder(getCblk()->asBinder());
+ return NO_ERROR;
+ } break;
default:
return BBinder::onTransact(code, data, reply, flags);
@@ -192,4 +199,3 @@
// ----------------------------------------------------------------------------
}; // namespace android
-
diff --git a/media/libmedia/IEffectClient.cpp b/media/libmedia/IEffectClient.cpp
index 4693b45..aef4371 100644
--- a/media/libmedia/IEffectClient.cpp
+++ b/media/libmedia/IEffectClient.cpp
@@ -94,7 +94,7 @@
status_t BnEffectClient::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
- switch(code) {
+ switch (code) {
case CONTROL_STATUS_CHANGED: {
ALOGV("CONTROL_STATUS_CHANGED");
CHECK_INTERFACE(IEffectClient, data, reply);
@@ -142,4 +142,3 @@
// ----------------------------------------------------------------------------
}; // namespace android
-
diff --git a/media/libmedia/IMediaDeathNotifier.cpp b/media/libmedia/IMediaDeathNotifier.cpp
index 8525482..9199db6 100644
--- a/media/libmedia/IMediaDeathNotifier.cpp
+++ b/media/libmedia/IMediaDeathNotifier.cpp
@@ -36,17 +36,17 @@
{
ALOGV("getMediaPlayerService");
Mutex::Autolock _l(sServiceLock);
- if (sMediaPlayerService.get() == 0) {
+ if (sMediaPlayerService == 0) {
sp<IServiceManager> sm = defaultServiceManager();
sp<IBinder> binder;
do {
binder = sm->getService(String16("media.player"));
if (binder != 0) {
break;
- }
- ALOGW("Media player service not published, waiting...");
- usleep(500000); // 0.5 s
- } while(true);
+ }
+ ALOGW("Media player service not published, waiting...");
+ usleep(500000); // 0.5 s
+ } while (true);
if (sDeathNotifier == NULL) {
sDeathNotifier = new DeathNotifier();
diff --git a/media/libmedia/IMediaPlayer.cpp b/media/libmedia/IMediaPlayer.cpp
index 9c1e6b7..0bb237d 100644
--- a/media/libmedia/IMediaPlayer.cpp
+++ b/media/libmedia/IMediaPlayer.cpp
@@ -15,6 +15,7 @@
** limitations under the License.
*/
+#include <arpa/inet.h>
#include <stdint.h>
#include <sys/types.h>
@@ -23,8 +24,6 @@
#include <media/IMediaPlayer.h>
#include <media/IStreamSource.h>
-#include <surfaceflinger/ISurface.h>
-#include <surfaceflinger/Surface.h>
#include <gui/ISurfaceTexture.h>
#include <utils/String8.h>
@@ -55,6 +54,8 @@
SET_VIDEO_SURFACETEXTURE,
SET_PARAMETER,
GET_PARAMETER,
+ SET_RETRANSMIT_ENDPOINT,
+ SET_NEXT_PLAYER,
};
class BpMediaPlayer: public BpInterface<IMediaPlayer>
@@ -198,11 +199,11 @@
return reply.readInt32();
}
- status_t setAudioStreamType(int type)
+ status_t setAudioStreamType(audio_stream_type_t stream)
{
Parcel data, reply;
data.writeInterfaceToken(IMediaPlayer::getInterfaceDescriptor());
- data.writeInt32(type);
+ data.writeInt32((int32_t) stream);
remote()->transact(SET_AUDIO_STREAM_TYPE, data, &reply);
return reply.readInt32();
}
@@ -291,6 +292,33 @@
return remote()->transact(GET_PARAMETER, data, reply);
}
+ status_t setRetransmitEndpoint(const struct sockaddr_in* endpoint) {
+ Parcel data, reply;
+ status_t err;
+
+ data.writeInterfaceToken(IMediaPlayer::getInterfaceDescriptor());
+ if (NULL != endpoint) {
+ data.writeInt32(sizeof(*endpoint));
+ data.write(endpoint, sizeof(*endpoint));
+ } else {
+ data.writeInt32(0);
+ }
+
+ err = remote()->transact(SET_RETRANSMIT_ENDPOINT, data, &reply);
+ if (OK != err) {
+ return err;
+ }
+ return reply.readInt32();
+ }
+
+ status_t setNextPlayer(const sp<IMediaPlayer>& player) {
+ Parcel data, reply;
+ data.writeInterfaceToken(IMediaPlayer::getInterfaceDescriptor());
+ sp<IBinder> b(player->asBinder());
+ data.writeStrongBinder(b);
+ remote()->transact(SET_NEXT_PLAYER, data, &reply);
+ return reply.readInt32();
+ }
};
IMPLEMENT_META_INTERFACE(MediaPlayer, "android.media.IMediaPlayer");
@@ -300,7 +328,7 @@
status_t BnMediaPlayer::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
- switch(code) {
+ switch (code) {
case DISCONNECT: {
CHECK_INTERFACE(IMediaPlayer, data, reply);
disconnect();
@@ -397,7 +425,7 @@
} break;
case SET_AUDIO_STREAM_TYPE: {
CHECK_INTERFACE(IMediaPlayer, data, reply);
- reply->writeInt32(setAudioStreamType(data.readInt32()));
+ reply->writeInt32(setAudioStreamType((audio_stream_type_t) data.readInt32()));
return NO_ERROR;
} break;
case SET_LOOPING: {
@@ -459,6 +487,24 @@
CHECK_INTERFACE(IMediaPlayer, data, reply);
return getParameter(data.readInt32(), reply);
} break;
+ case SET_RETRANSMIT_ENDPOINT: {
+ CHECK_INTERFACE(IMediaPlayer, data, reply);
+
+ struct sockaddr_in endpoint;
+ int amt = data.readInt32();
+ if (amt == sizeof(endpoint)) {
+ data.read(&endpoint, sizeof(struct sockaddr_in));
+ reply->writeInt32(setRetransmitEndpoint(&endpoint));
+ } else {
+ reply->writeInt32(setRetransmitEndpoint(NULL));
+ }
+ return NO_ERROR;
+ } break;
+ case SET_NEXT_PLAYER: {
+ CHECK_INTERFACE(IMediaPlayer, data, reply);
+ reply->writeInt32(setNextPlayer(interface_cast<IMediaPlayer>(data.readStrongBinder())));
+ return NO_ERROR;
+ } break;
default:
return BBinder::onTransact(code, data, reply, flags);
}
diff --git a/media/libmedia/IMediaPlayerClient.cpp b/media/libmedia/IMediaPlayerClient.cpp
index 1f135c4..a670c96 100644
--- a/media/libmedia/IMediaPlayerClient.cpp
+++ b/media/libmedia/IMediaPlayerClient.cpp
@@ -56,7 +56,7 @@
status_t BnMediaPlayerClient::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
- switch(code) {
+ switch (code) {
case NOTIFY: {
CHECK_INTERFACE(IMediaPlayerClient, data, reply);
int msg = data.readInt32();
diff --git a/media/libmedia/IMediaPlayerService.cpp b/media/libmedia/IMediaPlayerService.cpp
index 8e4dd04..f5fccef 100644
--- a/media/libmedia/IMediaPlayerService.cpp
+++ b/media/libmedia/IMediaPlayerService.cpp
@@ -78,7 +78,7 @@
return interface_cast<IMediaRecorder>(reply.readStrongBinder());
}
- virtual sp<IMemory> decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, int* pFormat)
+ virtual sp<IMemory> decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat)
{
Parcel data, reply;
data.writeInterfaceToken(IMediaPlayerService::getInterfaceDescriptor());
@@ -86,11 +86,11 @@
remote()->transact(DECODE_URL, data, &reply);
*pSampleRate = uint32_t(reply.readInt32());
*pNumChannels = reply.readInt32();
- *pFormat = reply.readInt32();
+ *pFormat = (audio_format_t) reply.readInt32();
return interface_cast<IMemory>(reply.readStrongBinder());
}
- virtual sp<IMemory> decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, int* pFormat)
+ virtual sp<IMemory> decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat)
{
Parcel data, reply;
data.writeInterfaceToken(IMediaPlayerService::getInterfaceDescriptor());
@@ -100,7 +100,7 @@
remote()->transact(DECODE_FD, data, &reply);
*pSampleRate = uint32_t(reply.readInt32());
*pNumChannels = reply.readInt32();
- *pFormat = reply.readInt32();
+ *pFormat = (audio_format_t) reply.readInt32();
return interface_cast<IMemory>(reply.readStrongBinder());
}
@@ -132,7 +132,7 @@
status_t BnMediaPlayerService::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
- switch(code) {
+ switch (code) {
case CREATE: {
CHECK_INTERFACE(IMediaPlayerService, data, reply);
pid_t pid = data.readInt32();
@@ -148,11 +148,11 @@
const char* url = data.readCString();
uint32_t sampleRate;
int numChannels;
- int format;
+ audio_format_t format;
sp<IMemory> player = decode(url, &sampleRate, &numChannels, &format);
reply->writeInt32(sampleRate);
reply->writeInt32(numChannels);
- reply->writeInt32(format);
+ reply->writeInt32((int32_t) format);
reply->writeStrongBinder(player->asBinder());
return NO_ERROR;
} break;
@@ -163,11 +163,11 @@
int64_t length = data.readInt64();
uint32_t sampleRate;
int numChannels;
- int format;
+ audio_format_t format;
sp<IMemory> player = decode(fd, offset, length, &sampleRate, &numChannels, &format);
reply->writeInt32(sampleRate);
reply->writeInt32(numChannels);
- reply->writeInt32(format);
+ reply->writeInt32((int32_t) format);
reply->writeStrongBinder(player->asBinder());
return NO_ERROR;
} break;
diff --git a/media/libmedia/IMediaRecorder.cpp b/media/libmedia/IMediaRecorder.cpp
index 42f55c2..a710fd7 100644
--- a/media/libmedia/IMediaRecorder.cpp
+++ b/media/libmedia/IMediaRecorder.cpp
@@ -19,10 +19,10 @@
#define LOG_TAG "IMediaRecorder"
#include <utils/Log.h>
#include <binder/Parcel.h>
-#include <surfaceflinger/Surface.h>
#include <camera/ICamera.h>
#include <media/IMediaRecorderClient.h>
#include <media/IMediaRecorder.h>
+#include <gui/Surface.h>
#include <gui/ISurfaceTexture.h>
#include <unistd.h>
@@ -289,7 +289,7 @@
status_t BnMediaRecorder::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
- switch(code) {
+ switch (code) {
case RELEASE: {
ALOGV("RELEASE");
CHECK_INTERFACE(IMediaRecorder, data, reply);
diff --git a/media/libmedia/IMediaRecorderClient.cpp b/media/libmedia/IMediaRecorderClient.cpp
index ff235c9..e7907e3 100644
--- a/media/libmedia/IMediaRecorderClient.cpp
+++ b/media/libmedia/IMediaRecorderClient.cpp
@@ -53,7 +53,7 @@
status_t BnMediaRecorderClient::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
- switch(code) {
+ switch (code) {
case NOTIFY: {
CHECK_INTERFACE(IMediaRecorderClient, data, reply);
int msg = data.readInt32();
diff --git a/media/libmedia/IOMX.cpp b/media/libmedia/IOMX.cpp
index d2f5f71..48e427a 100644
--- a/media/libmedia/IOMX.cpp
+++ b/media/libmedia/IOMX.cpp
@@ -22,8 +22,6 @@
#include <binder/Parcel.h>
#include <media/IOMX.h>
#include <media/stagefright/foundation/ADebug.h>
-#include <surfaceflinger/ISurface.h>
-#include <surfaceflinger/Surface.h>
namespace android {
@@ -59,9 +57,10 @@
: BpInterface<IOMX>(impl) {
}
- virtual bool livesLocally(pid_t pid) {
+ virtual bool livesLocally(node_id node, pid_t pid) {
Parcel data, reply;
data.writeInterfaceToken(IOMX::getInterfaceDescriptor());
+ data.writeIntPtr((intptr_t)node);
data.writeInt32(pid);
remote()->transact(LIVES_LOCALLY, data, &reply);
@@ -417,7 +416,9 @@
case LIVES_LOCALLY:
{
CHECK_INTERFACE(IOMX, data, reply);
- reply->writeInt32(livesLocally((pid_t)data.readInt32()));
+ node_id node = (void *)data.readIntPtr();
+ pid_t pid = (pid_t)data.readInt32();
+ reply->writeInt32(livesLocally(node, pid));
return OK;
}
diff --git a/media/libmedia/JetPlayer.cpp b/media/libmedia/JetPlayer.cpp
index fa5b67a..f1f62f7 100644
--- a/media/libmedia/JetPlayer.cpp
+++ b/media/libmedia/JetPlayer.cpp
@@ -74,14 +74,14 @@
// init the EAS library
result = EAS_Init(&mEasData);
- if( result != EAS_SUCCESS) {
+ if (result != EAS_SUCCESS) {
ALOGE("JetPlayer::init(): Error initializing Sonivox EAS library, aborting.");
mState = EAS_STATE_ERROR;
return result;
}
// init the JET library with the default app event controller range
result = JET_Init(mEasData, NULL, sizeof(S_JET_CONFIG));
- if( result != EAS_SUCCESS) {
+ if (result != EAS_SUCCESS) {
ALOGE("JetPlayer::init(): Error initializing JET library, aborting.");
mState = EAS_STATE_ERROR;
return result;
@@ -91,16 +91,17 @@
mAudioTrack = new AudioTrack();
mAudioTrack->set(AUDIO_STREAM_MUSIC, //TODO parametrize this
pLibConfig->sampleRate,
- 1, // format = PCM 16bits per sample,
- (pLibConfig->numChannels == 2) ? AUDIO_CHANNEL_OUT_STEREO : AUDIO_CHANNEL_OUT_MONO,
+ AUDIO_FORMAT_PCM_16_BIT,
+ audio_channel_out_mask_from_count(pLibConfig->numChannels),
mTrackBufferSize,
- 0);
+ AUDIO_POLICY_OUTPUT_FLAG_NONE);
// create render and playback thread
{
Mutex::Autolock l(mMutex);
ALOGV("JetPlayer::init(): trying to start render thread");
- createThreadEtc(renderThread, this, "jetRenderThread", ANDROID_PRIORITY_AUDIO);
+ mThread = new JetPlayerThread(this);
+ mThread->run("jetRenderThread", ANDROID_PRIORITY_AUDIO);
mCondition.wait(mMutex);
}
if (mTid > 0) {
@@ -150,18 +151,12 @@
mAudioBuffer = NULL;
}
mEasData = NULL;
-
+
return EAS_SUCCESS;
}
//-------------------------------------------------------------------------------------------------
-int JetPlayer::renderThread(void* p) {
-
- return ((JetPlayer*)p)->render();
-}
-
-//-------------------------------------------------------------------------------------------------
int JetPlayer::render() {
EAS_RESULT result = EAS_FAILURE;
EAS_I32 count;
@@ -171,12 +166,8 @@
ALOGV("JetPlayer::render(): entering");
// allocate render buffer
- mAudioBuffer =
+ mAudioBuffer =
new EAS_PCM[pLibConfig->mixBufferSize * pLibConfig->numChannels * MIX_NUM_BUFFERS];
- if (!mAudioBuffer) {
- ALOGE("JetPlayer::render(): mAudioBuffer allocate failed");
- goto threadExit;
- }
// signal main thread that we started
{
@@ -186,8 +177,8 @@
mCondition.signal();
}
- while (1) {
-
+ while (1) {
+
mMutex.lock(); // [[[[[[[[ LOCK ---------------------------------------
if (mEasData == NULL) {
@@ -195,20 +186,20 @@
ALOGV("JetPlayer::render(): NULL EAS data, exiting render.");
goto threadExit;
}
-
+
// nothing to render, wait for client thread to wake us up
while (!mRender)
{
ALOGV("JetPlayer::render(): signal wait");
- if (audioStarted) {
- mAudioTrack->pause();
+ if (audioStarted) {
+ mAudioTrack->pause();
// we have to restart the playback once we start rendering again
audioStarted = false;
}
mCondition.wait(mMutex);
ALOGV("JetPlayer::render(): signal rx'd");
}
-
+
// render midi data into the input buffer
int num_output = 0;
EAS_PCM* p = mAudioBuffer;
@@ -219,8 +210,8 @@
}
p += count * pLibConfig->numChannels;
num_output += count * pLibConfig->numChannels * sizeof(EAS_PCM);
-
- // send events that were generated (if any) to the event callback
+
+ // send events that were generated (if any) to the event callback
fireEventsFromJetQueue();
}
@@ -255,14 +246,12 @@
}//while (1)
threadExit:
- if (mAudioTrack) {
+ if (mAudioTrack != NULL) {
mAudioTrack->stop();
mAudioTrack->flush();
}
- if (mAudioBuffer) {
- delete [] mAudioBuffer;
- mAudioBuffer = NULL;
- }
+ delete [] mAudioBuffer;
+ mAudioBuffer = NULL;
mMutex.lock();
mTid = -1;
mCondition.signal();
@@ -276,9 +265,9 @@
// precondition: mMutex locked
void JetPlayer::fireUpdateOnStatusChange()
{
- if( (mJetStatus.currentUserID != mPreviousJetStatus.currentUserID)
+ if ( (mJetStatus.currentUserID != mPreviousJetStatus.currentUserID)
||(mJetStatus.segmentRepeatCount != mPreviousJetStatus.segmentRepeatCount) ) {
- if(mEventCallback) {
+ if (mEventCallback) {
mEventCallback(
JetPlayer::JET_USERID_UPDATE,
mJetStatus.currentUserID,
@@ -289,8 +278,8 @@
mPreviousJetStatus.segmentRepeatCount = mJetStatus.segmentRepeatCount;
}
- if(mJetStatus.numQueuedSegments != mPreviousJetStatus.numQueuedSegments) {
- if(mEventCallback) {
+ if (mJetStatus.numQueuedSegments != mPreviousJetStatus.numQueuedSegments) {
+ if (mEventCallback) {
mEventCallback(
JetPlayer::JET_NUMQUEUEDSEGMENT_UPDATE,
mJetStatus.numQueuedSegments,
@@ -300,8 +289,8 @@
mPreviousJetStatus.numQueuedSegments = mJetStatus.numQueuedSegments;
}
- if(mJetStatus.paused != mPreviousJetStatus.paused) {
- if(mEventCallback) {
+ if (mJetStatus.paused != mPreviousJetStatus.paused) {
+ if (mEventCallback) {
mEventCallback(JetPlayer::JET_PAUSE_UPDATE,
mJetStatus.paused,
-1,
@@ -318,7 +307,7 @@
// precondition: mMutex locked
void JetPlayer::fireEventsFromJetQueue()
{
- if(!mEventCallback) {
+ if (!mEventCallback) {
// no callback, just empty the event queue
while (JET_GetEvent(mEasData, NULL, NULL)) { }
return;
@@ -343,8 +332,8 @@
Mutex::Autolock lock(mMutex);
mEasJetFileLoc = (EAS_FILE_LOCATOR) malloc(sizeof(EAS_FILE));
- memset(mJetFilePath, 0, 256);
- strncpy(mJetFilePath, path, strlen(path));
+ strncpy(mJetFilePath, path, sizeof(mJetFilePath));
+ mJetFilePath[sizeof(mJetFilePath) - 1] = '\0';
mEasJetFileLoc->path = mJetFilePath;
mEasJetFileLoc->fd = 0;
@@ -352,7 +341,7 @@
mEasJetFileLoc->offset = 0;
EAS_RESULT result = JET_OpenFile(mEasData, mEasJetFileLoc);
- if(result != EAS_SUCCESS)
+ if (result != EAS_SUCCESS)
mState = EAS_STATE_ERROR;
else
mState = EAS_STATE_OPEN;
@@ -364,7 +353,7 @@
int JetPlayer::loadFromFD(const int fd, const long long offset, const long long length)
{
ALOGV("JetPlayer::loadFromFD(): fd=%d offset=%lld length=%lld", fd, offset, length);
-
+
Mutex::Autolock lock(mMutex);
mEasJetFileLoc = (EAS_FILE_LOCATOR) malloc(sizeof(EAS_FILE));
@@ -372,9 +361,9 @@
mEasJetFileLoc->offset = offset;
mEasJetFileLoc->length = length;
mEasJetFileLoc->path = NULL;
-
+
EAS_RESULT result = JET_OpenFile(mEasData, mEasJetFileLoc);
- if(result != EAS_SUCCESS)
+ if (result != EAS_SUCCESS)
mState = EAS_STATE_ERROR;
else
mState = EAS_STATE_OPEN;
@@ -403,7 +392,7 @@
JET_Status(mEasData, &mJetStatus);
this->dumpJetStatus(&mJetStatus);
-
+
fireUpdateOnStatusChange();
// wake up render thread
@@ -479,7 +468,7 @@
void JetPlayer::dumpJetStatus(S_JET_STATUS* pJetStatus)
{
- if(pJetStatus!=NULL)
+ if (pJetStatus!=NULL)
ALOGV(">> current JET player status: userID=%d segmentRepeatCount=%d numQueuedSegments=%d paused=%d",
pJetStatus->currentUserID, pJetStatus->segmentRepeatCount,
pJetStatus->numQueuedSegments, pJetStatus->paused);
@@ -489,4 +478,3 @@
} // end namespace android
-
diff --git a/media/libmedia/MediaProfiles.cpp b/media/libmedia/MediaProfiles.cpp
index c905762..c224f06 100644
--- a/media/libmedia/MediaProfiles.cpp
+++ b/media/libmedia/MediaProfiles.cpp
@@ -25,8 +25,8 @@
#include <cutils/properties.h>
#include <expat.h>
#include <media/MediaProfiles.h>
-#include <media/stagefright/MediaDebug.h>
-#include <media/stagefright/openmax/OMX_Video.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <OMX_Video.h>
namespace android {
@@ -349,7 +349,7 @@
{
CHECK(!strcmp("quality", atts[0]));
int quality = atoi(atts[1]);
- ALOGV("%s: cameraId=%d, quality=%d\n", __func__, cameraId, quality);
+ ALOGV("%s: cameraId=%d, quality=%d", __func__, cameraId, quality);
ImageEncodingQualityLevels *levels = findImageEncodingQualityLevels(cameraId);
if (levels == NULL) {
@@ -1099,12 +1099,12 @@
camcorder_quality quality) const
{
ALOGV("getCamcorderProfileParamByName: %s for camera %d, quality %d",
- name, cameraId, quality);
+ name, cameraId, quality);
int index = getCamcorderProfileIndex(cameraId, quality);
if (index == -1) {
ALOGE("The given camcorder profile camera %d quality %d is not found",
- cameraId, quality);
+ cameraId, quality);
return -1;
}
diff --git a/media/libmedia/MediaScanner.cpp b/media/libmedia/MediaScanner.cpp
index 79cab74..28b5aa7 100644
--- a/media/libmedia/MediaScanner.cpp
+++ b/media/libmedia/MediaScanner.cpp
@@ -54,7 +54,7 @@
void MediaScanner::loadSkipList() {
mSkipList = (char *)malloc(PROPERTY_VALUE_MAX * sizeof(char));
if (mSkipList) {
- property_get("testing.mediascanner.skiplist", mSkipList, "");
+ property_get("testing.mediascanner.skiplist", mSkipList, "");
}
if (!mSkipList || (strlen(mSkipList) == 0)) {
free(mSkipList);
@@ -135,15 +135,15 @@
struct dirent* entry;
if (shouldSkipDirectory(path)) {
- ALOGD("Skipping: %s", path);
- return MEDIA_SCAN_RESULT_OK;
+ ALOGD("Skipping: %s", path);
+ return MEDIA_SCAN_RESULT_OK;
}
// Treat all files as non-media in directories that contain a ".nomedia" file
if (pathRemaining >= 8 /* strlen(".nomedia") */ ) {
strcpy(fileSpot, ".nomedia");
if (access(path, F_OK) == 0) {
- ALOGV("found .nomedia, setting noMedia flag\n");
+ ALOGV("found .nomedia, setting noMedia flag");
noMedia = true;
}
diff --git a/media/libmedia/MediaScannerClient.cpp b/media/libmedia/MediaScannerClient.cpp
index 40b8188..e1e3348 100644
--- a/media/libmedia/MediaScannerClient.cpp
+++ b/media/libmedia/MediaScannerClient.cpp
@@ -142,12 +142,12 @@
UConverter *conv = ucnv_open(enc, &status);
if (U_FAILURE(status)) {
- ALOGE("could not create UConverter for %s\n", enc);
+ ALOGE("could not create UConverter for %s", enc);
return;
}
UConverter *utf8Conv = ucnv_open("UTF-8", &status);
if (U_FAILURE(status)) {
- ALOGE("could not create UConverter for UTF-8\n");
+ ALOGE("could not create UConverter for UTF-8");
ucnv_close(conv);
return;
}
@@ -173,6 +173,7 @@
const char* source = mValues->getEntry(i);
int targetLength = len * 3 + 1;
char* buffer = new char[targetLength];
+ // don't normally check for NULL, but in this case targetLength may be large
if (!buffer)
break;
char* target = buffer;
@@ -180,7 +181,7 @@
ucnv_convertEx(utf8Conv, conv, &target, target + targetLength,
&source, (const char *)dest, NULL, NULL, NULL, NULL, TRUE, TRUE, &status);
if (U_FAILURE(status)) {
- ALOGE("ucnv_convertEx failed: %d\n", status);
+ ALOGE("ucnv_convertEx failed: %d", status);
mValues->setEntry(i, "???");
} else {
// zero terminate
@@ -227,4 +228,3 @@
}
} // namespace android
-
diff --git a/media/libmedia/Metadata.cpp b/media/libmedia/Metadata.cpp
index 546a9b0..ef8a9ed 100644
--- a/media/libmedia/Metadata.cpp
+++ b/media/libmedia/Metadata.cpp
@@ -57,7 +57,7 @@
Metadata::Metadata(Parcel *p)
:mData(p),
- mBegin(p->dataPosition()) { }
+ mBegin(p->dataPosition()) { }
Metadata::~Metadata() { }
diff --git a/media/libmedia/ToneGenerator.cpp b/media/libmedia/ToneGenerator.cpp
index 35dfbb8..717d316 100644
--- a/media/libmedia/ToneGenerator.cpp
+++ b/media/libmedia/ToneGenerator.cpp
@@ -268,8 +268,8 @@
repeatCnt: 0,
repeatSegment: 0 }, // TONE_CDMA_CALL_SIGNAL_ISDN_SP_PRI
{ segments: { { duration: 0, waveFreq: { 0 }, 0, 0} },
- repeatCnt: 0,
- repeatSegment: 0 }, // TONE_CDMA_CALL_SIGNAL_ISDN_PAT3
+ repeatCnt: 0,
+ repeatSegment: 0 }, // TONE_CDMA_CALL_SIGNAL_ISDN_PAT3
{ segments: { { duration: 32, waveFreq: { 2091, 0 }, 0, 0 },
{ duration: 64, waveFreq: { 2556, 0 }, 4, 0 },
{ duration: 20, waveFreq: { 2091, 0 }, 0, 0 },
@@ -751,7 +751,7 @@
// Used by ToneGenerator::getToneForRegion() to convert user specified supervisory tone type
// to actual tone for current region.
-const unsigned char ToneGenerator::sToneMappingTable[NUM_REGIONS-1][NUM_SUP_TONES] = {
+const unsigned char /*tone_type*/ ToneGenerator::sToneMappingTable[NUM_REGIONS-1][NUM_SUP_TONES] = {
{ // ANSI
TONE_ANSI_DIAL, // TONE_SUP_DIAL
TONE_ANSI_BUSY, // TONE_SUP_BUSY
@@ -791,16 +791,16 @@
// generators, instantiates output audio track.
//
// Input:
-// streamType: Type of stream used for tone playback (enum AudioTrack::stream_type)
+// streamType: Type of stream used for tone playback
// volume: volume applied to tone (0.0 to 1.0)
//
// Output:
// none
//
////////////////////////////////////////////////////////////////////////////////
-ToneGenerator::ToneGenerator(int streamType, float volume, bool threadCanCallJava) {
+ToneGenerator::ToneGenerator(audio_stream_type_t streamType, float volume, bool threadCanCallJava) {
- ALOGV("ToneGenerator constructor: streamType=%d, volume=%f\n", streamType, volume);
+ ALOGV("ToneGenerator constructor: streamType=%d, volume=%f", streamType, volume);
mState = TONE_IDLE;
@@ -811,9 +811,9 @@
mThreadCanCallJava = threadCanCallJava;
mStreamType = streamType;
mVolume = volume;
- mpAudioTrack = 0;
- mpToneDesc = 0;
- mpNewToneDesc = 0;
+ mpAudioTrack = NULL;
+ mpToneDesc = NULL;
+ mpNewToneDesc = NULL;
// Generate tone by chunks of 20 ms to keep cadencing precision
mProcessSize = (mSamplingRate * 20) / 1000;
@@ -829,9 +829,9 @@
}
if (initAudioTrack()) {
- ALOGV("ToneGenerator INIT OK, time: %d\n", (unsigned int)(systemTime()/1000000));
+ ALOGV("ToneGenerator INIT OK, time: %d", (unsigned int)(systemTime()/1000000));
} else {
- ALOGV("!!!ToneGenerator INIT FAILED!!!\n");
+ ALOGV("!!!ToneGenerator INIT FAILED!!!");
}
}
@@ -853,11 +853,11 @@
//
////////////////////////////////////////////////////////////////////////////////
ToneGenerator::~ToneGenerator() {
- ALOGV("ToneGenerator destructor\n");
+ ALOGV("ToneGenerator destructor");
- if (mpAudioTrack) {
+ if (mpAudioTrack != NULL) {
stopTone();
- ALOGV("Delete Track: %p\n", mpAudioTrack);
+ ALOGV("Delete Track: %p", mpAudioTrack);
delete mpAudioTrack;
}
}
@@ -878,7 +878,7 @@
// none
//
////////////////////////////////////////////////////////////////////////////////
-bool ToneGenerator::startTone(int toneType, int durationMs) {
+bool ToneGenerator::startTone(tone_type toneType, int durationMs) {
bool lResult = false;
status_t lStatus;
@@ -892,7 +892,7 @@
}
}
- ALOGV("startTone\n");
+ ALOGV("startTone");
mLock.lock();
@@ -915,7 +915,7 @@
if (mState == TONE_INIT) {
if (prepareWave()) {
- ALOGV("Immediate start, time %d\n", (unsigned int)(systemTime()/1000000));
+ ALOGV("Immediate start, time %d", (unsigned int)(systemTime()/1000000));
lResult = true;
mState = TONE_STARTING;
mLock.unlock();
@@ -934,7 +934,7 @@
mState = TONE_IDLE;
}
} else {
- ALOGV("Delayed start\n");
+ ALOGV("Delayed start");
mState = TONE_RESTARTING;
lStatus = mWaitCbkCond.waitRelative(mLock, seconds(3));
if (lStatus == NO_ERROR) {
@@ -949,8 +949,8 @@
}
mLock.unlock();
- ALOGV_IF(lResult, "Tone started, time %d\n", (unsigned int)(systemTime()/1000000));
- ALOGW_IF(!lResult, "Tone start failed!!!, time %d\n", (unsigned int)(systemTime()/1000000));
+ ALOGV_IF(lResult, "Tone started, time %d", (unsigned int)(systemTime()/1000000));
+ ALOGW_IF(!lResult, "Tone start failed!!!, time %d", (unsigned int)(systemTime()/1000000));
return lResult;
}
@@ -1012,23 +1012,19 @@
if (mpAudioTrack) {
delete mpAudioTrack;
- mpAudioTrack = 0;
+ mpAudioTrack = NULL;
}
- // Open audio track in mono, PCM 16bit, default sampling rate, default buffer size
+ // Open audio track in mono, PCM 16bit, default sampling rate, default buffer size
mpAudioTrack = new AudioTrack();
- if (mpAudioTrack == 0) {
- ALOGE("AudioTrack allocation failed");
- goto initAudioTrack_exit;
- }
- ALOGV("Create Track: %p\n", mpAudioTrack);
+ ALOGV("Create Track: %p", mpAudioTrack);
mpAudioTrack->set(mStreamType,
0,
AUDIO_FORMAT_PCM_16_BIT,
AUDIO_CHANNEL_OUT_MONO,
0,
- 0,
+ AUDIO_POLICY_OUTPUT_FLAG_NONE,
audioCallback,
this,
0,
@@ -1049,10 +1045,10 @@
initAudioTrack_exit:
// Cleanup
- if (mpAudioTrack) {
- ALOGV("Delete Track I: %p\n", mpAudioTrack);
+ if (mpAudioTrack != NULL) {
+ ALOGV("Delete Track I: %p", mpAudioTrack);
delete mpAudioTrack;
- mpAudioTrack = 0;
+ mpAudioTrack = NULL;
}
return false;
@@ -1145,7 +1141,7 @@
if (lpToneGen->mTotalSmp > lpToneGen->mNextSegSmp) {
// Time to go to next sequence segment
- ALOGV("End Segment, time: %d\n", (unsigned int)(systemTime()/1000000));
+ ALOGV("End Segment, time: %d", (unsigned int)(systemTime()/1000000));
lGenSmp = lReqSmp;
@@ -1160,13 +1156,13 @@
lpWaveGen->getSamples(lpOut, lGenSmp, lWaveCmd);
lFrequency = lpToneDesc->segments[lpToneGen->mCurSegment].waveFreq[++lFreqIdx];
}
- ALOGV("ON->OFF, lGenSmp: %d, lReqSmp: %d\n", lGenSmp, lReqSmp);
+ ALOGV("ON->OFF, lGenSmp: %d, lReqSmp: %d", lGenSmp, lReqSmp);
}
// check if we need to loop and loop for the reqd times
if (lpToneDesc->segments[lpToneGen->mCurSegment].loopCnt) {
if (lpToneGen->mLoopCounter < lpToneDesc->segments[lpToneGen->mCurSegment].loopCnt) {
- ALOGV ("in if loop loopCnt(%d) loopctr(%d), CurSeg(%d) \n",
+ ALOGV ("in if loop loopCnt(%d) loopctr(%d), CurSeg(%d)",
lpToneDesc->segments[lpToneGen->mCurSegment].loopCnt,
lpToneGen->mLoopCounter,
lpToneGen->mCurSegment);
@@ -1176,14 +1172,14 @@
// completed loop. go to next segment
lpToneGen->mLoopCounter = 0;
lpToneGen->mCurSegment++;
- ALOGV ("in else loop loopCnt(%d) loopctr(%d), CurSeg(%d) \n",
+ ALOGV ("in else loop loopCnt(%d) loopctr(%d), CurSeg(%d)",
lpToneDesc->segments[lpToneGen->mCurSegment].loopCnt,
lpToneGen->mLoopCounter,
lpToneGen->mCurSegment);
}
} else {
lpToneGen->mCurSegment++;
- ALOGV ("Goto next seg loopCnt(%d) loopctr(%d), CurSeg(%d) \n",
+ ALOGV ("Goto next seg loopCnt(%d) loopctr(%d), CurSeg(%d)",
lpToneDesc->segments[lpToneGen->mCurSegment].loopCnt,
lpToneGen->mLoopCounter,
lpToneGen->mCurSegment);
@@ -1192,32 +1188,32 @@
// Handle loop if last segment reached
if (lpToneDesc->segments[lpToneGen->mCurSegment].duration == 0) {
- ALOGV("Last Seg: %d\n", lpToneGen->mCurSegment);
+ ALOGV("Last Seg: %d", lpToneGen->mCurSegment);
// Pre increment loop count and restart if total count not reached. Stop sequence otherwise
if (++lpToneGen->mCurCount <= lpToneDesc->repeatCnt) {
- ALOGV("Repeating Count: %d\n", lpToneGen->mCurCount);
+ ALOGV("Repeating Count: %d", lpToneGen->mCurCount);
lpToneGen->mCurSegment = lpToneDesc->repeatSegment;
if (lpToneDesc->segments[lpToneDesc->repeatSegment].waveFreq[0] != 0) {
lWaveCmd = WaveGenerator::WAVEGEN_START;
}
- ALOGV("New segment %d, Next Time: %d\n", lpToneGen->mCurSegment,
+ ALOGV("New segment %d, Next Time: %d", lpToneGen->mCurSegment,
(lpToneGen->mNextSegSmp*1000)/lpToneGen->mSamplingRate);
} else {
lGenSmp = 0;
- ALOGV("End repeat, time: %d\n", (unsigned int)(systemTime()/1000000));
+ ALOGV("End repeat, time: %d", (unsigned int)(systemTime()/1000000));
}
} else {
- ALOGV("New segment %d, Next Time: %d\n", lpToneGen->mCurSegment,
+ ALOGV("New segment %d, Next Time: %d", lpToneGen->mCurSegment,
(lpToneGen->mNextSegSmp*1000)/lpToneGen->mSamplingRate);
if (lpToneDesc->segments[lpToneGen->mCurSegment].waveFreq[0] != 0) {
// If next segment is not silent, OFF -> ON transition : reset wave generator
lWaveCmd = WaveGenerator::WAVEGEN_START;
- ALOGV("OFF->ON, lGenSmp: %d, lReqSmp: %d\n", lGenSmp, lReqSmp);
+ ALOGV("OFF->ON, lGenSmp: %d, lReqSmp: %d", lGenSmp, lReqSmp);
} else {
lGenSmp = 0;
}
@@ -1255,13 +1251,13 @@
switch (lpToneGen->mState) {
case TONE_RESTARTING:
- ALOGV("Cbk restarting track\n");
+ ALOGV("Cbk restarting track");
if (lpToneGen->prepareWave()) {
lpToneGen->mState = TONE_STARTING;
// must reload lpToneDesc as prepareWave() may change mpToneDesc
lpToneDesc = lpToneGen->mpToneDesc;
} else {
- ALOGW("Cbk restarting prepareWave() failed\n");
+ ALOGW("Cbk restarting prepareWave() failed");
lpToneGen->mState = TONE_IDLE;
lpToneGen->mpAudioTrack->stop();
// Force loop exit
@@ -1270,14 +1266,14 @@
lSignal = true;
break;
case TONE_STOPPING:
- ALOGV("Cbk Stopping\n");
+ ALOGV("Cbk Stopping");
lpToneGen->mState = TONE_STOPPED;
// Force loop exit
lNumSmp = 0;
break;
case TONE_STOPPED:
lpToneGen->mState = TONE_INIT;
- ALOGV("Cbk Stopped track\n");
+ ALOGV("Cbk Stopped track");
lpToneGen->mpAudioTrack->stop();
// Force loop exit
lNumSmp = 0;
@@ -1285,12 +1281,12 @@
lSignal = true;
break;
case TONE_STARTING:
- ALOGV("Cbk starting track\n");
+ ALOGV("Cbk starting track");
lpToneGen->mState = TONE_PLAYING;
lSignal = true;
- break;
+ break;
case TONE_PLAYING:
- break;
+ break;
default:
// Force loop exit
lNumSmp = 0;
@@ -1321,7 +1317,7 @@
bool ToneGenerator::prepareWave() {
unsigned int segmentIdx = 0;
- if (!mpNewToneDesc) {
+ if (mpNewToneDesc == NULL) {
return false;
}
@@ -1353,9 +1349,6 @@
new ToneGenerator::WaveGenerator((unsigned short)mSamplingRate,
frequency,
TONEGEN_GAIN/lNumWaves);
- if (lpWaveGen == 0) {
- goto prepareWave_exit;
- }
mWaveGens.add(frequency, lpWaveGen);
}
frequency = mpNewToneDesc->segments[segmentIdx].waveFreq[++freqIdx];
@@ -1375,12 +1368,6 @@
}
return true;
-
-prepareWave_exit:
-
- clearWaveGens();
-
- return false;
}
@@ -1447,13 +1434,13 @@
// none
//
////////////////////////////////////////////////////////////////////////////////
-int ToneGenerator::getToneForRegion(int toneType) {
- int regionTone;
+ToneGenerator::tone_type ToneGenerator::getToneForRegion(tone_type toneType) {
+ tone_type regionTone;
if (mRegion == CEPT || toneType < FIRST_SUP_TONE || toneType > LAST_SUP_TONE) {
regionTone = toneType;
} else {
- regionTone = sToneMappingTable[mRegion][toneType - FIRST_SUP_TONE];
+ regionTone = (tone_type) sToneMappingTable[mRegion][toneType - FIRST_SUP_TONE];
}
ALOGV("getToneForRegion, tone %d, region %d, regionTone %d", toneType, mRegion, regionTone);
@@ -1504,7 +1491,7 @@
d0 = 32767;
mA1_Q14 = (short) d0;
- ALOGV("WaveGenerator init, mA1_Q14: %d, mS2_0: %d, mAmplitude_Q15: %d\n",
+ ALOGV("WaveGenerator init, mA1_Q14: %d, mS2_0: %d, mAmplitude_Q15: %d",
mA1_Q14, mS2_0, mAmplitude_Q15);
}
@@ -1591,4 +1578,3 @@
}
} // end namespace android
-
diff --git a/media/libmedia/Visualizer.cpp b/media/libmedia/Visualizer.cpp
index d08ffa5..bcd6ae4 100644
--- a/media/libmedia/Visualizer.cpp
+++ b/media/libmedia/Visualizer.cpp
@@ -27,8 +27,7 @@
#include <cutils/bitops.h>
#include <media/Visualizer.h>
-
-extern void fixed_fft_real(int n, int32_t *v);
+#include <audio_utils/fixedfft.h>
namespace android {
@@ -54,7 +53,7 @@
status_t Visualizer::setEnabled(bool enabled)
{
- Mutex::Autolock _l(mLock);
+ Mutex::Autolock _l(mCaptureLock);
sp<CaptureThread> t = mCaptureThread;
if (t != 0) {
@@ -67,14 +66,14 @@
}
}
t->mLock.lock();
- }
+ }
status_t status = AudioEffect::setEnabled(enabled);
if (status == NO_ERROR) {
if (t != 0) {
if (enabled) {
- t->run("AudioTrackThread");
+ t->run("Visualizer");
} else {
t->requestExit();
}
@@ -93,7 +92,7 @@
if (rate > CAPTURE_RATE_MAX) {
return BAD_VALUE;
}
- Mutex::Autolock _l(mLock);
+ Mutex::Autolock _l(mCaptureLock);
if (mEnabled) {
return INVALID_OPERATION;
@@ -115,10 +114,6 @@
if (cbk != NULL) {
mCaptureThread = new CaptureThread(*this, rate, ((flags & CAPTURE_CALL_JAVA) != 0));
- if (mCaptureThread == 0) {
- ALOGE("Could not create callback thread");
- return NO_INIT;
- }
}
ALOGV("setCaptureCallBack() rate: %d thread %p flags 0x%08x",
rate, mCaptureThread.get(), mCaptureFlags);
@@ -133,7 +128,7 @@
return BAD_VALUE;
}
- Mutex::Autolock _l(mLock);
+ Mutex::Autolock _l(mCaptureLock);
if (mEnabled) {
return INVALID_OPERATION;
}
@@ -173,7 +168,7 @@
uint32_t replySize = mCaptureSize;
status = command(VISUALIZER_CMD_CAPTURE, 0, NULL, &replySize, waveform);
ALOGV("getWaveForm() command returned %d", status);
- if (replySize == 0) {
+ if ((status == NO_ERROR) && (replySize == 0)) {
status = NOT_ENOUGH_DATA;
}
} else {
@@ -235,7 +230,7 @@
void Visualizer::periodicCapture()
{
- Mutex::Autolock _l(mLock);
+ Mutex::Autolock _l(mCaptureLock);
ALOGV("periodicCapture() %p mCaptureCallBack %p mCaptureFlags 0x%08x",
this, mCaptureCallBack, mCaptureFlags);
if (mCaptureCallBack != NULL &&
@@ -325,4 +320,3 @@
}
}; // namespace android
-
diff --git a/media/libmedia/autodetect.cpp b/media/libmedia/autodetect.cpp
index dfcc6a0..be5c3b2 100644
--- a/media/libmedia/autodetect.cpp
+++ b/media/libmedia/autodetect.cpp
@@ -16,7 +16,7 @@
#include "autodetect.h"
-typedef struct CharRange {
+struct CharRange {
uint16_t first;
uint16_t last;
};
diff --git a/media/libmedia/fixedfft.cpp b/media/libmedia/fixedfft.cpp
deleted file mode 100644
index 2b495e6..0000000
--- a/media/libmedia/fixedfft.cpp
+++ /dev/null
@@ -1,162 +0,0 @@
-/*
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-/* A Fixed point implementation of Fast Fourier Transform (FFT). Complex numbers
- * are represented by 32-bit integers, where higher 16 bits are real part and
- * lower ones are imaginary part. Few compromises are made between efficiency,
- * accuracy, and maintainability. To make it fast, arithmetic shifts are used
- * instead of divisions, and bitwise inverses are used instead of negates. To
- * keep it small, only radix-2 Cooley-Tukey algorithm is implemented, and only
- * half of the twiddle factors are stored. Although there are still ways to make
- * it even faster or smaller, it costs too much on one of the aspects.
- */
-
-#include <stdio.h>
-#include <stdint.h>
-#ifdef __arm__
-#include <machine/cpu-features.h>
-#endif
-
-#define LOG_FFT_SIZE 10
-#define MAX_FFT_SIZE (1 << LOG_FFT_SIZE)
-
-static const int32_t twiddle[MAX_FFT_SIZE / 4] = {
- 0x00008000, 0xff378001, 0xfe6e8002, 0xfda58006, 0xfcdc800a, 0xfc13800f,
- 0xfb4a8016, 0xfa81801e, 0xf9b88027, 0xf8ef8032, 0xf827803e, 0xf75e804b,
- 0xf6958059, 0xf5cd8068, 0xf5058079, 0xf43c808b, 0xf374809e, 0xf2ac80b2,
- 0xf1e480c8, 0xf11c80de, 0xf05580f6, 0xef8d8110, 0xeec6812a, 0xedff8146,
- 0xed388163, 0xec718181, 0xebab81a0, 0xeae481c1, 0xea1e81e2, 0xe9588205,
- 0xe892822a, 0xe7cd824f, 0xe7078276, 0xe642829d, 0xe57d82c6, 0xe4b982f1,
- 0xe3f4831c, 0xe3308349, 0xe26d8377, 0xe1a983a6, 0xe0e683d6, 0xe0238407,
- 0xdf61843a, 0xde9e846e, 0xdddc84a3, 0xdd1b84d9, 0xdc598511, 0xdb998549,
- 0xdad88583, 0xda1885be, 0xd95885fa, 0xd8988637, 0xd7d98676, 0xd71b86b6,
- 0xd65c86f6, 0xd59e8738, 0xd4e1877b, 0xd42487c0, 0xd3678805, 0xd2ab884c,
- 0xd1ef8894, 0xd13488dd, 0xd0798927, 0xcfbe8972, 0xcf0489be, 0xce4b8a0c,
- 0xcd928a5a, 0xccd98aaa, 0xcc218afb, 0xcb698b4d, 0xcab28ba0, 0xc9fc8bf5,
- 0xc9468c4a, 0xc8908ca1, 0xc7db8cf8, 0xc7278d51, 0xc6738dab, 0xc5c08e06,
- 0xc50d8e62, 0xc45b8ebf, 0xc3a98f1d, 0xc2f88f7d, 0xc2488fdd, 0xc198903e,
- 0xc0e990a1, 0xc03a9105, 0xbf8c9169, 0xbedf91cf, 0xbe329236, 0xbd86929e,
- 0xbcda9307, 0xbc2f9371, 0xbb8593dc, 0xbadc9448, 0xba3394b5, 0xb98b9523,
- 0xb8e39592, 0xb83c9603, 0xb7969674, 0xb6f196e6, 0xb64c9759, 0xb5a897ce,
- 0xb5059843, 0xb46298b9, 0xb3c09930, 0xb31f99a9, 0xb27f9a22, 0xb1df9a9c,
- 0xb1409b17, 0xb0a29b94, 0xb0059c11, 0xaf689c8f, 0xaecc9d0e, 0xae319d8e,
- 0xad979e0f, 0xacfd9e91, 0xac659f14, 0xabcd9f98, 0xab36a01c, 0xaaa0a0a2,
- 0xaa0aa129, 0xa976a1b0, 0xa8e2a238, 0xa84fa2c2, 0xa7bda34c, 0xa72ca3d7,
- 0xa69ca463, 0xa60ca4f0, 0xa57ea57e, 0xa4f0a60c, 0xa463a69c, 0xa3d7a72c,
- 0xa34ca7bd, 0xa2c2a84f, 0xa238a8e2, 0xa1b0a976, 0xa129aa0a, 0xa0a2aaa0,
- 0xa01cab36, 0x9f98abcd, 0x9f14ac65, 0x9e91acfd, 0x9e0fad97, 0x9d8eae31,
- 0x9d0eaecc, 0x9c8faf68, 0x9c11b005, 0x9b94b0a2, 0x9b17b140, 0x9a9cb1df,
- 0x9a22b27f, 0x99a9b31f, 0x9930b3c0, 0x98b9b462, 0x9843b505, 0x97ceb5a8,
- 0x9759b64c, 0x96e6b6f1, 0x9674b796, 0x9603b83c, 0x9592b8e3, 0x9523b98b,
- 0x94b5ba33, 0x9448badc, 0x93dcbb85, 0x9371bc2f, 0x9307bcda, 0x929ebd86,
- 0x9236be32, 0x91cfbedf, 0x9169bf8c, 0x9105c03a, 0x90a1c0e9, 0x903ec198,
- 0x8fddc248, 0x8f7dc2f8, 0x8f1dc3a9, 0x8ebfc45b, 0x8e62c50d, 0x8e06c5c0,
- 0x8dabc673, 0x8d51c727, 0x8cf8c7db, 0x8ca1c890, 0x8c4ac946, 0x8bf5c9fc,
- 0x8ba0cab2, 0x8b4dcb69, 0x8afbcc21, 0x8aaaccd9, 0x8a5acd92, 0x8a0cce4b,
- 0x89becf04, 0x8972cfbe, 0x8927d079, 0x88ddd134, 0x8894d1ef, 0x884cd2ab,
- 0x8805d367, 0x87c0d424, 0x877bd4e1, 0x8738d59e, 0x86f6d65c, 0x86b6d71b,
- 0x8676d7d9, 0x8637d898, 0x85fad958, 0x85beda18, 0x8583dad8, 0x8549db99,
- 0x8511dc59, 0x84d9dd1b, 0x84a3dddc, 0x846ede9e, 0x843adf61, 0x8407e023,
- 0x83d6e0e6, 0x83a6e1a9, 0x8377e26d, 0x8349e330, 0x831ce3f4, 0x82f1e4b9,
- 0x82c6e57d, 0x829de642, 0x8276e707, 0x824fe7cd, 0x822ae892, 0x8205e958,
- 0x81e2ea1e, 0x81c1eae4, 0x81a0ebab, 0x8181ec71, 0x8163ed38, 0x8146edff,
- 0x812aeec6, 0x8110ef8d, 0x80f6f055, 0x80def11c, 0x80c8f1e4, 0x80b2f2ac,
- 0x809ef374, 0x808bf43c, 0x8079f505, 0x8068f5cd, 0x8059f695, 0x804bf75e,
- 0x803ef827, 0x8032f8ef, 0x8027f9b8, 0x801efa81, 0x8016fb4a, 0x800ffc13,
- 0x800afcdc, 0x8006fda5, 0x8002fe6e, 0x8001ff37,
-};
-
-/* Returns the multiplication of \conj{a} and {b}. */
-static inline int32_t mult(int32_t a, int32_t b)
-{
-#if __ARM_ARCH__ >= 6
- int32_t t = b;
- __asm__("smuad %0, %0, %1" : "+r" (t) : "r" (a));
- __asm__("smusdx %0, %0, %1" : "+r" (b) : "r" (a));
- __asm__("pkhtb %0, %0, %1, ASR #16" : "+r" (t) : "r" (b));
- return t;
-#else
- return (((a >> 16) * (b >> 16) + (int16_t)a * (int16_t)b) & ~0xFFFF) |
- ((((a >> 16) * (int16_t)b - (int16_t)a * (b >> 16)) >> 16) & 0xFFFF);
-#endif
-}
-
-static inline int32_t half(int32_t a)
-{
-#if __ARM_ARCH__ >= 6
- __asm__("shadd16 %0, %0, %1" : "+r" (a) : "r" (0));
- return a;
-#else
- return ((a >> 1) & ~0x8000) | (a & 0x8000);
-#endif
-}
-
-void fixed_fft(int n, int32_t *v)
-{
- int scale = LOG_FFT_SIZE, i, p, r;
-
- for (r = 0, i = 1; i < n; ++i) {
- for (p = n; !(p & r); p >>= 1, r ^= p);
- if (i < r) {
- int32_t t = v[i];
- v[i] = v[r];
- v[r] = t;
- }
- }
-
- for (p = 1; p < n; p <<= 1) {
- --scale;
-
- for (i = 0; i < n; i += p << 1) {
- int32_t x = half(v[i]);
- int32_t y = half(v[i + p]);
- v[i] = x + y;
- v[i + p] = x - y;
- }
-
- for (r = 1; r < p; ++r) {
- int32_t w = MAX_FFT_SIZE / 4 - (r << scale);
- i = w >> 31;
- w = twiddle[(w ^ i) - i] ^ (i << 16);
- for (i = r; i < n; i += p << 1) {
- int32_t x = half(v[i]);
- int32_t y = mult(w, v[i + p]);
- v[i] = x - y;
- v[i + p] = x + y;
- }
- }
- }
-}
-
-void fixed_fft_real(int n, int32_t *v)
-{
- int scale = LOG_FFT_SIZE, m = n >> 1, i;
-
- fixed_fft(n, v);
- for (i = 1; i <= n; i <<= 1, --scale);
- v[0] = mult(~v[0], 0x80008000);
- v[m] = half(v[m]);
-
- for (i = 1; i < n >> 1; ++i) {
- int32_t x = half(v[i]);
- int32_t z = half(v[n - i]);
- int32_t y = z - (x ^ 0xFFFF);
- x = half(x + (z ^ 0xFFFF));
- y = mult(y, twiddle[i << scale]);
- v[i] = x - y;
- v[n - i] = (x + y) ^ 0xFFFF;
- }
-}
diff --git a/media/libmedia/mediametadataretriever.cpp b/media/libmedia/mediametadataretriever.cpp
index 88e269f..b0241aa 100644
--- a/media/libmedia/mediametadataretriever.cpp
+++ b/media/libmedia/mediametadataretriever.cpp
@@ -35,7 +35,7 @@
const sp<IMediaPlayerService>& MediaMetadataRetriever::getService()
{
Mutex::Autolock lock(sServiceLock);
- if (sService.get() == 0) {
+ if (sService == 0) {
sp<IServiceManager> sm = defaultServiceManager();
sp<IBinder> binder;
do {
@@ -45,7 +45,7 @@
}
ALOGW("MediaPlayerService not published, waiting...");
usleep(500000); // 0.5 s
- } while(true);
+ } while (true);
if (sDeathNotifier == NULL) {
sDeathNotifier = new DeathNotifier();
}
diff --git a/media/libmedia/mediaplayer.cpp b/media/libmedia/mediaplayer.cpp
index 2284927..b52a37d 100644
--- a/media/libmedia/mediaplayer.cpp
+++ b/media/libmedia/mediaplayer.cpp
@@ -1,4 +1,4 @@
-/* mediaplayer.cpp
+/*
**
** Copyright 2006, The Android Open Source Project
**
@@ -30,9 +30,7 @@
#include <gui/SurfaceTextureClient.h>
#include <media/mediaplayer.h>
-#include <media/AudioTrack.h>
-
-#include <surfaceflinger/Surface.h>
+#include <media/AudioSystem.h>
#include <binder/MemoryBase.h>
@@ -63,6 +61,7 @@
mAudioSessionId = AudioSystem::newAudioSessionId();
AudioSystem::acquireAudioSessionId(mAudioSessionId);
mSendLevel = 0;
+ mRetransmitEndpointValid = false;
}
MediaPlayer::~MediaPlayer()
@@ -95,6 +94,7 @@
mCurrentPosition = -1;
mSeekPosition = -1;
mVideoWidth = mVideoHeight = 0;
+ mRetransmitEndpointValid = false;
}
status_t MediaPlayer::setListener(const sp<MediaPlayerListener>& listener)
@@ -146,7 +146,8 @@
const sp<IMediaPlayerService>& service(getMediaPlayerService());
if (service != 0) {
sp<IMediaPlayer> player(service->create(getpid(), this, mAudioSessionId));
- if (NO_ERROR != player->setDataSource(url, headers)) {
+ if ((NO_ERROR != doSetRetransmitEndpoint(player)) ||
+ (NO_ERROR != player->setDataSource(url, headers))) {
player.clear();
}
err = attachNewPlayer(player);
@@ -162,7 +163,8 @@
const sp<IMediaPlayerService>& service(getMediaPlayerService());
if (service != 0) {
sp<IMediaPlayer> player(service->create(getpid(), this, mAudioSessionId));
- if (NO_ERROR != player->setDataSource(fd, offset, length)) {
+ if ((NO_ERROR != doSetRetransmitEndpoint(player)) ||
+ (NO_ERROR != player->setDataSource(fd, offset, length))) {
player.clear();
}
err = attachNewPlayer(player);
@@ -177,7 +179,8 @@
const sp<IMediaPlayerService>& service(getMediaPlayerService());
if (service != 0) {
sp<IMediaPlayer> player(service->create(getpid(), this, mAudioSessionId));
- if (NO_ERROR != player->setDataSource(source)) {
+ if ((NO_ERROR != doSetRetransmitEndpoint(player)) ||
+ (NO_ERROR != player->setDataSource(source))) {
player.clear();
}
err = attachNewPlayer(player);
@@ -192,8 +195,8 @@
(mCurrentState != MEDIA_PLAYER_STATE_ERROR) &&
((mCurrentState & MEDIA_PLAYER_IDLE) != MEDIA_PLAYER_IDLE);
if ((mPlayer != NULL) && hasBeenInitialized) {
- ALOGV("invoke %d", request.dataSize());
- return mPlayer->invoke(request, reply);
+ ALOGV("invoke %d", request.dataSize());
+ return mPlayer->invoke(request, reply);
}
ALOGE("invoke failed: wrong state %X", mCurrentState);
return INVALID_OPERATION;
@@ -471,6 +474,20 @@
return NO_ERROR;
}
+status_t MediaPlayer::doSetRetransmitEndpoint(const sp<IMediaPlayer>& player) {
+ Mutex::Autolock _l(mLock);
+
+ if (player == NULL) {
+ return UNKNOWN_ERROR;
+ }
+
+ if (mRetransmitEndpointValid) {
+ return player->setRetransmitEndpoint(&mRetransmitEndpoint);
+ }
+
+ return OK;
+}
+
status_t MediaPlayer::reset()
{
ALOGV("reset");
@@ -478,7 +495,7 @@
return reset_l();
}
-status_t MediaPlayer::setAudioStreamType(int type)
+status_t MediaPlayer::setAudioStreamType(audio_stream_type_t type)
{
ALOGV("MediaPlayer::setAudioStreamType");
Mutex::Autolock _l(mLock);
@@ -539,9 +556,9 @@
return BAD_VALUE;
}
if (sessionId != mAudioSessionId) {
- AudioSystem::releaseAudioSessionId(mAudioSessionId);
- AudioSystem::acquireAudioSessionId(sessionId);
- mAudioSessionId = sessionId;
+ AudioSystem::releaseAudioSessionId(mAudioSessionId);
+ AudioSystem::acquireAudioSessionId(sessionId);
+ mAudioSessionId = sessionId;
}
return NO_ERROR;
}
@@ -593,12 +610,40 @@
ALOGV("MediaPlayer::getParameter(%d)", key);
Mutex::Autolock _l(mLock);
if (mPlayer != NULL) {
- return mPlayer->getParameter(key, reply);
+ return mPlayer->getParameter(key, reply);
}
ALOGV("getParameter: no active player");
return INVALID_OPERATION;
}
+status_t MediaPlayer::setRetransmitEndpoint(const char* addrString,
+ uint16_t port) {
+ ALOGV("MediaPlayer::setRetransmitEndpoint(%s:%hu)",
+ addrString ? addrString : "(null)", port);
+
+ Mutex::Autolock _l(mLock);
+ if ((mPlayer != NULL) || (mCurrentState != MEDIA_PLAYER_IDLE))
+ return INVALID_OPERATION;
+
+ if (NULL == addrString) {
+ mRetransmitEndpointValid = false;
+ return OK;
+ }
+
+ struct in_addr saddr;
+ if(!inet_aton(addrString, &saddr)) {
+ return BAD_VALUE;
+ }
+
+ memset(&mRetransmitEndpoint, 0, sizeof(&mRetransmitEndpoint));
+ mRetransmitEndpoint.sin_family = AF_INET;
+ mRetransmitEndpoint.sin_addr = saddr;
+ mRetransmitEndpoint.sin_port = htons(port);
+ mRetransmitEndpointValid = true;
+
+ return OK;
+}
+
void MediaPlayer::notify(int msg, int ext1, int ext2, const Parcel *obj)
{
ALOGV("message received msg=%d, ext1=%d, ext2=%d", msg, ext1, ext2);
@@ -613,7 +658,7 @@
// and seekTo within the same process.
// FIXME: Remember, this is a hack, it's not even a hack that is applied
// consistently for all use-cases, this needs to be revisited.
- if (mLockThreadId != getThreadId()) {
+ if (mLockThreadId != getThreadId()) {
mLock.lock();
locked = true;
}
@@ -709,7 +754,7 @@
}
}
-/*static*/ sp<IMemory> MediaPlayer::decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, int* pFormat)
+/*static*/ sp<IMemory> MediaPlayer::decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat)
{
ALOGV("decode(%s)", url);
sp<IMemory> p;
@@ -729,7 +774,7 @@
notify(MEDIA_ERROR, MEDIA_ERROR_SERVER_DIED, 0);
}
-/*static*/ sp<IMemory> MediaPlayer::decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, int* pFormat)
+/*static*/ sp<IMemory> MediaPlayer::decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat)
{
ALOGV("decode(%d, %lld, %lld)", fd, offset, length);
sp<IMemory> p;
@@ -743,4 +788,11 @@
}
+status_t MediaPlayer::setNextMediaPlayer(const sp<MediaPlayer>& next) {
+ if (mPlayer == NULL) {
+ return NO_INIT;
+ }
+ return mPlayer->setNextPlayer(next == NULL ? NULL : next->mPlayer);
+}
+
}; // namespace android
diff --git a/media/libmedia/mediarecorder.cpp b/media/libmedia/mediarecorder.cpp
index 8d947d8..9541015 100644
--- a/media/libmedia/mediarecorder.cpp
+++ b/media/libmedia/mediarecorder.cpp
@@ -18,7 +18,6 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "MediaRecorder"
#include <utils/Log.h>
-#include <surfaceflinger/Surface.h>
#include <media/mediarecorder.h>
#include <binder/IServiceManager.h>
#include <utils/String8.h>
@@ -32,7 +31,7 @@
status_t MediaRecorder::setCamera(const sp<ICamera>& camera, const sp<ICameraRecordingProxy>& proxy)
{
ALOGV("setCamera(%p,%p)", camera.get(), proxy.get());
- if(mMediaRecorder == NULL) {
+ if (mMediaRecorder == NULL) {
ALOGE("media recorder is not initialized yet");
return INVALID_OPERATION;
}
@@ -53,7 +52,7 @@
status_t MediaRecorder::setPreviewSurface(const sp<Surface>& surface)
{
ALOGV("setPreviewSurface(%p)", surface.get());
- if(mMediaRecorder == NULL) {
+ if (mMediaRecorder == NULL) {
ALOGE("media recorder is not initialized yet");
return INVALID_OPERATION;
}
@@ -78,7 +77,7 @@
status_t MediaRecorder::init()
{
ALOGV("init");
- if(mMediaRecorder == NULL) {
+ if (mMediaRecorder == NULL) {
ALOGE("media recorder is not initialized yet");
return INVALID_OPERATION;
}
@@ -108,7 +107,7 @@
status_t MediaRecorder::setVideoSource(int vs)
{
ALOGV("setVideoSource(%d)", vs);
- if(mMediaRecorder == NULL) {
+ if (mMediaRecorder == NULL) {
ALOGE("media recorder is not initialized yet");
return INVALID_OPERATION;
}
@@ -143,7 +142,7 @@
status_t MediaRecorder::setAudioSource(int as)
{
ALOGV("setAudioSource(%d)", as);
- if(mMediaRecorder == NULL) {
+ if (mMediaRecorder == NULL) {
ALOGE("media recorder is not initialized yet");
return INVALID_OPERATION;
}
@@ -176,7 +175,7 @@
status_t MediaRecorder::setOutputFormat(int of)
{
ALOGV("setOutputFormat(%d)", of);
- if(mMediaRecorder == NULL) {
+ if (mMediaRecorder == NULL) {
ALOGE("media recorder is not initialized yet");
return INVALID_OPERATION;
}
@@ -202,7 +201,7 @@
status_t MediaRecorder::setVideoEncoder(int ve)
{
ALOGV("setVideoEncoder(%d)", ve);
- if(mMediaRecorder == NULL) {
+ if (mMediaRecorder == NULL) {
ALOGE("media recorder is not initialized yet");
return INVALID_OPERATION;
}
@@ -232,7 +231,7 @@
status_t MediaRecorder::setAudioEncoder(int ae)
{
ALOGV("setAudioEncoder(%d)", ae);
- if(mMediaRecorder == NULL) {
+ if (mMediaRecorder == NULL) {
ALOGE("media recorder is not initialized yet");
return INVALID_OPERATION;
}
@@ -262,7 +261,7 @@
status_t MediaRecorder::setOutputFile(const char* path)
{
ALOGV("setOutputFile(%s)", path);
- if(mMediaRecorder == NULL) {
+ if (mMediaRecorder == NULL) {
ALOGE("media recorder is not initialized yet");
return INVALID_OPERATION;
}
@@ -288,7 +287,7 @@
status_t MediaRecorder::setOutputFile(int fd, int64_t offset, int64_t length)
{
ALOGV("setOutputFile(%d, %lld, %lld)", fd, offset, length);
- if(mMediaRecorder == NULL) {
+ if (mMediaRecorder == NULL) {
ALOGE("media recorder is not initialized yet");
return INVALID_OPERATION;
}
@@ -325,7 +324,7 @@
status_t MediaRecorder::setVideoSize(int width, int height)
{
ALOGV("setVideoSize(%d, %d)", width, height);
- if(mMediaRecorder == NULL) {
+ if (mMediaRecorder == NULL) {
ALOGE("media recorder is not initialized yet");
return INVALID_OPERATION;
}
@@ -368,7 +367,7 @@
status_t MediaRecorder::setVideoFrameRate(int frames_per_second)
{
ALOGV("setVideoFrameRate(%d)", frames_per_second);
- if(mMediaRecorder == NULL) {
+ if (mMediaRecorder == NULL) {
ALOGE("media recorder is not initialized yet");
return INVALID_OPERATION;
}
@@ -392,7 +391,7 @@
status_t MediaRecorder::setParameters(const String8& params) {
ALOGV("setParameters(%s)", params.string());
- if(mMediaRecorder == NULL) {
+ if (mMediaRecorder == NULL) {
ALOGE("media recorder is not initialized yet");
return INVALID_OPERATION;
}
@@ -420,7 +419,7 @@
status_t MediaRecorder::prepare()
{
ALOGV("prepare");
- if(mMediaRecorder == NULL) {
+ if (mMediaRecorder == NULL) {
ALOGE("media recorder is not initialized yet");
return INVALID_OPERATION;
}
@@ -459,7 +458,7 @@
status_t MediaRecorder::getMaxAmplitude(int* max)
{
ALOGV("getMaxAmplitude");
- if(mMediaRecorder == NULL) {
+ if (mMediaRecorder == NULL) {
ALOGE("media recorder is not initialized yet");
return INVALID_OPERATION;
}
@@ -537,7 +536,7 @@
doCleanUp();
status_t ret = UNKNOWN_ERROR;
- switch(mCurrentState) {
+ switch (mCurrentState) {
case MEDIA_RECORDER_IDLE:
ret = OK;
break;
@@ -548,7 +547,7 @@
case MEDIA_RECORDER_ERROR: {
ret = doReset();
if (OK != ret) {
- return ret; // No need to continue
+ return ret; // No need to continue
}
} // Intentional fall through
case MEDIA_RECORDER_INITIALIZED:
diff --git a/media/libmediaplayerservice/Android.mk b/media/libmediaplayerservice/Android.mk
index a3e2517..ba5c776 100644
--- a/media/libmediaplayerservice/Android.mk
+++ b/media/libmediaplayerservice/Android.mk
@@ -23,25 +23,27 @@
libvorbisidec \
libsonivox \
libmedia \
+ libmedia_native \
libcamera_client \
libandroid_runtime \
libstagefright \
libstagefright_omx \
libstagefright_foundation \
libgui \
- libdl
+ libdl \
+ libaah_rtp
LOCAL_STATIC_LIBRARIES := \
libstagefright_nuplayer \
libstagefright_rtsp \
-LOCAL_C_INCLUDES := \
+LOCAL_C_INCLUDES := \
$(JNI_H_INCLUDE) \
$(call include-path-for, graphics corecg) \
- $(TOP)/frameworks/base/include/media/stagefright/openmax \
$(TOP)/frameworks/base/media/libstagefright/include \
$(TOP)/frameworks/base/media/libstagefright/rtsp \
- $(TOP)/external/tremolo/Tremolo \
+ $(TOP)/frameworks/native/include/media/openmax \
+ $(TOP)/external/tremolo/Tremolo
LOCAL_MODULE:= libmediaplayerservice
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index 72678b9..8f62ee4 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -70,6 +70,11 @@
#include <OMX.h>
+namespace android {
+sp<MediaPlayerBase> createAAH_TXPlayer();
+sp<MediaPlayerBase> createAAH_RXPlayer();
+}
+
namespace {
using android::media::Metadata;
using android::status_t;
@@ -487,6 +492,7 @@
mStatus = NO_INIT;
mAudioSessionId = audioSessionId;
mUID = uid;
+ mRetransmitEndpointValid = false;
#if CALLBACK_ANTAGONIZER
ALOGD("create Antagonizer");
@@ -593,6 +599,10 @@
return NU_PLAYER;
}
+ if (!strncasecmp("aahRX://", url, 8)) {
+ return AAH_RX_PLAYER;
+ }
+
// use MidiFile for MIDI extensions
int lenURL = strlen(url);
for (int i = 0; i < NELEM(FILE_EXTS); ++i) {
@@ -608,6 +618,44 @@
return getDefaultPlayerType();
}
+player_type MediaPlayerService::Client::getPlayerType(int fd,
+ int64_t offset,
+ int64_t length)
+{
+ // Until re-transmit functionality is added to the existing core android
+ // players, we use the special AAH TX player whenever we were configured
+ // for retransmission.
+ if (mRetransmitEndpointValid) {
+ return AAH_TX_PLAYER;
+ }
+
+ return android::getPlayerType(fd, offset, length);
+}
+
+player_type MediaPlayerService::Client::getPlayerType(const char* url)
+{
+ // Until re-transmit functionality is added to the existing core android
+ // players, we use the special AAH TX player whenever we were configured
+ // for retransmission.
+ if (mRetransmitEndpointValid) {
+ return AAH_TX_PLAYER;
+ }
+
+ return android::getPlayerType(url);
+}
+
+player_type MediaPlayerService::Client::getPlayerType(
+ const sp<IStreamSource> &source) {
+ // Until re-transmit functionality is added to the existing core android
+ // players, we use the special AAH TX player whenever we were configured
+ // for retransmission.
+ if (mRetransmitEndpointValid) {
+ return AAH_TX_PLAYER;
+ }
+
+ return NU_PLAYER;
+}
+
static sp<MediaPlayerBase> createPlayer(player_type playerType, void* cookie,
notify_callback_f notifyFunc)
{
@@ -629,6 +677,14 @@
ALOGV("Create Test Player stub");
p = new TestPlayerStub();
break;
+ case AAH_RX_PLAYER:
+ ALOGV(" create A@H RX Player");
+ p = createAAH_RXPlayer();
+ break;
+ case AAH_TX_PLAYER:
+ ALOGV(" create A@H TX Player");
+ p = createAAH_TXPlayer();
+ break;
default:
ALOGE("Unknown player type: %d", playerType);
return NULL;
@@ -665,6 +721,49 @@
return p;
}
+sp<MediaPlayerBase> MediaPlayerService::Client::setDataSource_pre(
+ player_type playerType)
+{
+ ALOGV("player type = %d", playerType);
+
+ // create the right type of player
+ sp<MediaPlayerBase> p = createPlayer(playerType);
+ if (p == NULL) {
+ return p;
+ }
+
+ if (!p->hardwareOutput()) {
+ mAudioOutput = new AudioOutput(mAudioSessionId);
+ static_cast<MediaPlayerInterface*>(p.get())->setAudioSink(mAudioOutput);
+ }
+
+ return p;
+}
+
+void MediaPlayerService::Client::setDataSource_post(
+ const sp<MediaPlayerBase>& p,
+ status_t status)
+{
+ ALOGV(" setDataSource");
+ mStatus = status;
+ if (mStatus != OK) {
+ ALOGE(" error: %d", mStatus);
+ return;
+ }
+
+ // Set the re-transmission endpoint if one was chosen.
+ if (mRetransmitEndpointValid) {
+ mStatus = p->setRetransmitEndpoint(&mRetransmitEndpoint);
+ if (mStatus != NO_ERROR) {
+ ALOGE("setRetransmitEndpoint error: %d", mStatus);
+ }
+ }
+
+ if (mStatus == OK) {
+ mPlayer = p;
+ }
+}
+
status_t MediaPlayerService::Client::setDataSource(
const char *url, const KeyedVector<String8, String8> *headers)
{
@@ -696,25 +795,12 @@
return mStatus;
} else {
player_type playerType = getPlayerType(url);
- ALOGV("player type = %d", playerType);
-
- // create the right type of player
- sp<MediaPlayerBase> p = createPlayer(playerType);
- if (p == NULL) return NO_INIT;
-
- if (!p->hardwareOutput()) {
- mAudioOutput = new AudioOutput(mAudioSessionId);
- static_cast<MediaPlayerInterface*>(p.get())->setAudioSink(mAudioOutput);
+ sp<MediaPlayerBase> p = setDataSource_pre(playerType);
+ if (p == NULL) {
+ return NO_INIT;
}
- // now set data source
- ALOGV(" setDataSource");
- mStatus = p->setDataSource(url, headers);
- if (mStatus == NO_ERROR) {
- mPlayer = p;
- } else {
- ALOGE(" error: %d", mStatus);
- }
+ setDataSource_post(p, p->setDataSource(url, headers));
return mStatus;
}
}
@@ -745,46 +831,34 @@
ALOGV("calculated length = %lld", length);
}
+ // Until re-transmit functionality is added to the existing core android
+ // players, we use the special AAH TX player whenever we were configured for
+ // retransmission.
player_type playerType = getPlayerType(fd, offset, length);
- ALOGV("player type = %d", playerType);
-
- // create the right type of player
- sp<MediaPlayerBase> p = createPlayer(playerType);
- if (p == NULL) return NO_INIT;
-
- if (!p->hardwareOutput()) {
- mAudioOutput = new AudioOutput(mAudioSessionId);
- static_cast<MediaPlayerInterface*>(p.get())->setAudioSink(mAudioOutput);
+ sp<MediaPlayerBase> p = setDataSource_pre(playerType);
+ if (p == NULL) {
+ return NO_INIT;
}
// now set data source
- mStatus = p->setDataSource(fd, offset, length);
- if (mStatus == NO_ERROR) mPlayer = p;
-
+ setDataSource_post(p, p->setDataSource(fd, offset, length));
return mStatus;
}
status_t MediaPlayerService::Client::setDataSource(
const sp<IStreamSource> &source) {
// create the right type of player
- sp<MediaPlayerBase> p = createPlayer(NU_PLAYER);
-
+ // Until re-transmit functionality is added to the existing core android
+ // players, we use the special AAH TX player whenever we were configured for
+ // retransmission.
+ player_type playerType = getPlayerType(source);
+ sp<MediaPlayerBase> p = setDataSource_pre(playerType);
if (p == NULL) {
return NO_INIT;
}
- if (!p->hardwareOutput()) {
- mAudioOutput = new AudioOutput(mAudioSessionId);
- static_cast<MediaPlayerInterface*>(p.get())->setAudioSink(mAudioOutput);
- }
-
// now set data source
- mStatus = p->setDataSource(source);
-
- if (mStatus == OK) {
- mPlayer = p;
- }
-
+ setDataSource_post(p, p->setDataSource(source));
return mStatus;
}
@@ -994,6 +1068,20 @@
return ret;
}
+status_t MediaPlayerService::Client::setNextPlayer(const sp<IMediaPlayer>& player) {
+ ALOGV("setNextPlayer");
+ Mutex::Autolock l(mLock);
+ sp<Client> c = static_cast<Client*>(player.get());
+ mNextClient = c;
+ if (mAudioOutput != NULL && c != NULL) {
+ mAudioOutput->setNextOutput(c->mAudioOutput);
+ } else {
+ ALOGE("no current audio output");
+ }
+ return OK;
+}
+
+
status_t MediaPlayerService::Client::seekTo(int msec)
{
ALOGV("[%d] seekTo(%d)", mConnId, msec);
@@ -1005,12 +1093,13 @@
status_t MediaPlayerService::Client::reset()
{
ALOGV("[%d] reset", mConnId);
+ mRetransmitEndpointValid = false;
sp<MediaPlayerBase> p = getPlayer();
if (p == 0) return UNKNOWN_ERROR;
return p->reset();
}
-status_t MediaPlayerService::Client::setAudioStreamType(int type)
+status_t MediaPlayerService::Client::setAudioStreamType(audio_stream_type_t type)
{
ALOGV("[%d] setAudioStreamType(%d)", mConnId, type);
// TODO: for hardware output, call player instead
@@ -1031,9 +1120,21 @@
status_t MediaPlayerService::Client::setVolume(float leftVolume, float rightVolume)
{
ALOGV("[%d] setVolume(%f, %f)", mConnId, leftVolume, rightVolume);
- // TODO: for hardware output, call player instead
- Mutex::Autolock l(mLock);
- if (mAudioOutput != 0) mAudioOutput->setVolume(leftVolume, rightVolume);
+
+ // for hardware output, call player instead
+ sp<MediaPlayerBase> p = getPlayer();
+ {
+ Mutex::Autolock l(mLock);
+ if (p != 0 && p->hardwareOutput()) {
+ MediaPlayerHWInterface* hwp =
+ reinterpret_cast<MediaPlayerHWInterface*>(p.get());
+ return hwp->setVolume(leftVolume, rightVolume);
+ } else {
+ if (mAudioOutput != 0) mAudioOutput->setVolume(leftVolume, rightVolume);
+ return NO_ERROR;
+ }
+ }
+
return NO_ERROR;
}
@@ -1067,11 +1168,50 @@
return p->getParameter(key, reply);
}
+status_t MediaPlayerService::Client::setRetransmitEndpoint(
+ const struct sockaddr_in* endpoint) {
+
+ if (NULL != endpoint) {
+ uint32_t a = ntohl(endpoint->sin_addr.s_addr);
+ uint16_t p = ntohs(endpoint->sin_port);
+ ALOGV("[%d] setRetransmitEndpoint(%u.%u.%u.%u:%hu)", mConnId,
+ (a >> 24), (a >> 16) & 0xFF, (a >> 8) & 0xFF, (a & 0xFF), p);
+ } else {
+ ALOGV("[%d] setRetransmitEndpoint = <none>", mConnId);
+ }
+
+ sp<MediaPlayerBase> p = getPlayer();
+
+ // Right now, the only valid time to set a retransmit endpoint is before
+ // player selection has been made (since the presence or absence of a
+ // retransmit endpoint is going to determine which player is selected during
+ // setDataSource).
+ if (p != 0) return INVALID_OPERATION;
+
+ if (NULL != endpoint) {
+ mRetransmitEndpoint = *endpoint;
+ mRetransmitEndpointValid = true;
+ } else {
+ mRetransmitEndpointValid = false;
+ }
+
+ return NO_ERROR;
+}
+
void MediaPlayerService::Client::notify(
void* cookie, int msg, int ext1, int ext2, const Parcel *obj)
{
Client* client = static_cast<Client*>(cookie);
+ {
+ Mutex::Autolock l(client->mLock);
+ if (msg == MEDIA_PLAYBACK_COMPLETE && client->mNextClient != NULL) {
+ client->mAudioOutput->switchToNextOutput();
+ client->mNextClient->start();
+ client->mNextClient->mClient->notify(MEDIA_INFO, MEDIA_INFO_STARTED_AS_NEXT, 0, obj);
+ }
+ }
+
if (MEDIA_INFO == msg &&
MEDIA_INFO_METADATA_UPDATE == ext1) {
const media::Metadata::Type metadata_type = ext2;
@@ -1149,7 +1289,7 @@
static size_t kDefaultHeapSize = 1024 * 1024; // 1MB
-sp<IMemory> MediaPlayerService::decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, int* pFormat)
+sp<IMemory> MediaPlayerService::decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat)
{
ALOGV("decode(%s)", url);
sp<MemoryBase> mem;
@@ -1197,7 +1337,7 @@
mem = new MemoryBase(cache->getHeap(), 0, cache->size());
*pSampleRate = cache->sampleRate();
*pNumChannels = cache->channelCount();
- *pFormat = (int)cache->format();
+ *pFormat = cache->format();
ALOGV("return memory @ %p, sampleRate=%u, channelCount = %d, format = %d", mem->pointer(), *pSampleRate, *pNumChannels, *pFormat);
Exit:
@@ -1205,7 +1345,7 @@
return mem;
}
-sp<IMemory> MediaPlayerService::decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, int* pFormat)
+sp<IMemory> MediaPlayerService::decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat)
{
ALOGV("decode(%d, %lld, %lld)", fd, offset, length);
sp<MemoryBase> mem;
@@ -1259,12 +1399,16 @@
MediaPlayerService::AudioOutput::AudioOutput(int sessionId)
: mCallback(NULL),
mCallbackCookie(NULL),
+ mCallbackData(NULL),
mSessionId(sessionId) {
ALOGV("AudioOutput(%d)", sessionId);
mTrack = 0;
+ mRecycledTrack = 0;
mStreamType = AUDIO_STREAM_MUSIC;
mLeftVolume = 1.0;
mRightVolume = 1.0;
+ mPlaybackRatePermille = 1000;
+ mSampleRateHz = 0;
mMsecsPerFrame = 0;
mAuxEffectId = 0;
mSendLevel = 0.0;
@@ -1274,6 +1418,8 @@
MediaPlayerService::AudioOutput::~AudioOutput()
{
close();
+ delete mRecycledTrack;
+ delete mCallbackData;
}
void MediaPlayerService::AudioOutput::setMinBufferCount()
@@ -1339,7 +1485,8 @@
}
status_t MediaPlayerService::AudioOutput::open(
- uint32_t sampleRate, int channelCount, int format, int bufferCount,
+ uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
+ audio_format_t format, int bufferCount,
AudioCallback cb, void *cookie)
{
mCallback = cb;
@@ -1351,8 +1498,8 @@
bufferCount = mMinBufferCount;
}
- ALOGV("open(%u, %d, %d, %d, %d)", sampleRate, channelCount, format, bufferCount,mSessionId);
- if (mTrack) close();
+ ALOGV("open(%u, %d, 0x%x, %d, %d, %d)", sampleRate, channelCount, channelMask,
+ format, bufferCount, mSessionId);
int afSampleRate;
int afFrameCount;
int frameCount;
@@ -1366,17 +1513,64 @@
frameCount = (sampleRate*afFrameCount*bufferCount)/afSampleRate;
+ if (channelMask == CHANNEL_MASK_USE_CHANNEL_ORDER) {
+ channelMask = audio_channel_out_mask_from_count(channelCount);
+ if (0 == channelMask) {
+ ALOGE("open() error, can\'t derive mask for %d audio channels", channelCount);
+ return NO_INIT;
+ }
+ }
+ if (mRecycledTrack) {
+ // check if the existing track can be reused as-is, or if a new track needs to be created.
+
+ bool reuse = true;
+ if ((mCallbackData == NULL && mCallback != NULL) ||
+ (mCallbackData != NULL && mCallback == NULL)) {
+ // recycled track uses callbacks but the caller wants to use writes, or vice versa
+ ALOGV("can't chain callback and write");
+ reuse = false;
+ } else if ((mRecycledTrack->getSampleRate() != sampleRate) ||
+ (mRecycledTrack->channelCount() != channelCount) ||
+ (mRecycledTrack->frameCount() != frameCount)) {
+ ALOGV("samplerate, channelcount or framecount differ");
+ reuse = false;
+ }
+ if (reuse) {
+ ALOGV("chaining to next output");
+ close();
+ mTrack = mRecycledTrack;
+ mRecycledTrack = NULL;
+ if (mCallbackData != NULL) {
+ mCallbackData->setOutput(this);
+ }
+ return OK;
+ }
+
+ // if we're not going to reuse the track, unblock and flush it
+ if (mCallbackData != NULL) {
+ mCallbackData->setOutput(NULL);
+ mCallbackData->endTrackSwitch();
+ }
+ mRecycledTrack->flush();
+ delete mRecycledTrack;
+ mRecycledTrack = NULL;
+ delete mCallbackData;
+ mCallbackData = NULL;
+ close();
+ }
+
AudioTrack *t;
if (mCallback != NULL) {
+ mCallbackData = new CallbackData(this);
t = new AudioTrack(
mStreamType,
sampleRate,
format,
- (channelCount == 2) ? AUDIO_CHANNEL_OUT_STEREO : AUDIO_CHANNEL_OUT_MONO,
+ channelMask,
frameCount,
- 0 /* flags */,
+ AUDIO_POLICY_OUTPUT_FLAG_NONE,
CallbackWrapper,
- this,
+ mCallbackData,
0,
mSessionId);
} else {
@@ -1384,9 +1578,9 @@
mStreamType,
sampleRate,
format,
- (channelCount == 2) ? AUDIO_CHANNEL_OUT_STEREO : AUDIO_CHANNEL_OUT_MONO,
+ channelMask,
frameCount,
- 0,
+ AUDIO_POLICY_OUTPUT_FLAG_NONE,
NULL,
NULL,
0,
@@ -1402,9 +1596,14 @@
ALOGV("setVolume");
t->setVolume(mLeftVolume, mRightVolume);
- mMsecsPerFrame = 1.e3 / (float) sampleRate;
+ mSampleRateHz = sampleRate;
+ mMsecsPerFrame = mPlaybackRatePermille / (float) sampleRate;
mTrack = t;
+ status_t res = t->setSampleRate(mPlaybackRatePermille * mSampleRateHz / 1000);
+ if (res != NO_ERROR) {
+ return res;
+ }
t->setAuxEffectSendLevel(mSendLevel);
return t->attachAuxEffect(mAuxEffectId);;
}
@@ -1412,6 +1611,9 @@
void MediaPlayerService::AudioOutput::start()
{
ALOGV("start");
+ if (mCallbackData != NULL) {
+ mCallbackData->endTrackSwitch();
+ }
if (mTrack) {
mTrack->setVolume(mLeftVolume, mRightVolume);
mTrack->setAuxEffectSendLevel(mSendLevel);
@@ -1419,8 +1621,27 @@
}
}
+void MediaPlayerService::AudioOutput::setNextOutput(const sp<AudioOutput>& nextOutput) {
+ mNextOutput = nextOutput;
+}
+void MediaPlayerService::AudioOutput::switchToNextOutput() {
+ ALOGV("switchToNextOutput");
+ if (mNextOutput != NULL) {
+ if (mCallbackData != NULL) {
+ mCallbackData->beginTrackSwitch();
+ }
+ delete mNextOutput->mCallbackData;
+ mNextOutput->mCallbackData = mCallbackData;
+ mCallbackData = NULL;
+ mNextOutput->mRecycledTrack = mTrack;
+ mTrack = NULL;
+ mNextOutput->mSampleRateHz = mSampleRateHz;
+ mNextOutput->mMsecsPerFrame = mMsecsPerFrame;
+ }
+}
+
ssize_t MediaPlayerService::AudioOutput::write(const void* buffer, size_t size)
{
LOG_FATAL_IF(mCallback != NULL, "Don't call write if supplying a callback.");
@@ -1468,6 +1689,22 @@
}
}
+status_t MediaPlayerService::AudioOutput::setPlaybackRatePermille(int32_t ratePermille)
+{
+ ALOGV("setPlaybackRatePermille(%d)", ratePermille);
+ status_t res = NO_ERROR;
+ if (mTrack) {
+ res = mTrack->setSampleRate(ratePermille * mSampleRateHz / 1000);
+ } else {
+ res = NO_INIT;
+ }
+ mPlaybackRatePermille = ratePermille;
+ if (mSampleRateHz != 0) {
+ mMsecsPerFrame = mPlaybackRatePermille / (float) mSampleRateHz;
+ }
+ return res;
+}
+
status_t MediaPlayerService::AudioOutput::setAuxEffectSendLevel(float level)
{
ALOGV("setAuxEffectSendLevel(%f)", level);
@@ -1496,13 +1733,22 @@
return;
}
- AudioOutput *me = (AudioOutput *)cookie;
+ CallbackData *data = (CallbackData*)cookie;
+ data->lock();
+ AudioOutput *me = data->getOutput();
AudioTrack::Buffer *buffer = (AudioTrack::Buffer *)info;
+ if (me == NULL) {
+ // no output set, likely because the track was scheduled to be reused
+ // by another player, but the format turned out to be incompatible.
+ data->unlock();
+ buffer->size = 0;
+ return;
+ }
size_t actualSize = (*me->mCallback)(
me, buffer->raw, buffer->size, me->mCallbackCookie);
- if (actualSize == 0 && buffer->size > 0) {
+ if (actualSize == 0 && buffer->size > 0 && me->mNextOutput == NULL) {
// We've reached EOS but the audio track is not stopped yet,
// keep playing silence.
@@ -1511,6 +1757,7 @@
}
buffer->size = actualSize;
+ data->unlock();
}
int MediaPlayerService::AudioOutput::getSessionId()
@@ -1610,17 +1857,18 @@
////////////////////////////////////////////////////////////////////////////////
status_t MediaPlayerService::AudioCache::open(
- uint32_t sampleRate, int channelCount, int format, int bufferCount,
+ uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
+ audio_format_t format, int bufferCount,
AudioCallback cb, void *cookie)
{
- ALOGV("open(%u, %d, %d, %d)", sampleRate, channelCount, format, bufferCount);
+ ALOGV("open(%u, %d, 0x%x, %d, %d)", sampleRate, channelCount, channelMask, format, bufferCount);
if (mHeap->getHeapID() < 0) {
return NO_INIT;
}
mSampleRate = sampleRate;
mChannelCount = (uint16_t)channelCount;
- mFormat = (uint16_t)format;
+ mFormat = format;
mMsecsPerFrame = 1.e3 / (float) sampleRate;
if (cb != NULL) {
diff --git a/media/libmediaplayerservice/MediaPlayerService.h b/media/libmediaplayerservice/MediaPlayerService.h
index b04fddb..d4e0eb1 100644
--- a/media/libmediaplayerservice/MediaPlayerService.h
+++ b/media/libmediaplayerservice/MediaPlayerService.h
@@ -18,6 +18,8 @@
#ifndef ANDROID_MEDIAPLAYERSERVICE_H
#define ANDROID_MEDIAPLAYERSERVICE_H
+#include <arpa/inet.h>
+
#include <utils/Log.h>
#include <utils/threads.h>
#include <utils/List.h>
@@ -29,11 +31,13 @@
#include <media/IMediaPlayerService.h>
#include <media/MediaPlayerInterface.h>
#include <media/Metadata.h>
+#include <media/stagefright/foundation/ABase.h>
#include <system/audio.h>
namespace android {
+class AudioTrack;
class IMediaRecorder;
class IMediaMetadataRetriever;
class IOMX;
@@ -66,7 +70,9 @@
class AudioOutput : public MediaPlayerBase::AudioSink
{
- public:
+ class CallbackData;
+
+ public:
AudioOutput(int sessionId);
virtual ~AudioOutput();
@@ -82,8 +88,8 @@
virtual int getSessionId();
virtual status_t open(
- uint32_t sampleRate, int channelCount,
- int format, int bufferCount,
+ uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
+ audio_format_t format, int bufferCount,
AudioCallback cb, void *cookie);
virtual void start();
@@ -92,25 +98,35 @@
virtual void flush();
virtual void pause();
virtual void close();
- void setAudioStreamType(int streamType) { mStreamType = streamType; }
+ void setAudioStreamType(audio_stream_type_t streamType) { mStreamType = streamType; }
void setVolume(float left, float right);
+ virtual status_t setPlaybackRatePermille(int32_t ratePermille);
status_t setAuxEffectSendLevel(float level);
status_t attachAuxEffect(int effectId);
virtual status_t dump(int fd, const Vector<String16>& args) const;
static bool isOnEmulator();
static int getMinBufferCount();
+ void setNextOutput(const sp<AudioOutput>& nextOutput);
+ void switchToNextOutput();
+ virtual bool needsTrailingPadding() { return mNextOutput == NULL; }
+
private:
static void setMinBufferCount();
static void CallbackWrapper(
int event, void *me, void *info);
AudioTrack* mTrack;
+ AudioTrack* mRecycledTrack;
+ sp<AudioOutput> mNextOutput;
AudioCallback mCallback;
void * mCallbackCookie;
- int mStreamType;
+ CallbackData * mCallbackData;
+ audio_stream_type_t mStreamType;
float mLeftVolume;
float mRightVolume;
+ int32_t mPlaybackRatePermille;
+ uint32_t mSampleRateHz; // sample rate of the content, as set in open()
float mMsecsPerFrame;
int mSessionId;
float mSendLevel;
@@ -118,7 +134,38 @@
static bool mIsOnEmulator;
static int mMinBufferCount; // 12 for emulator; otherwise 4
- };
+ // CallbackData is what is passed to the AudioTrack as the "user" data.
+ // We need to be able to target this to a different Output on the fly,
+ // so we can't use the Output itself for this.
+ class CallbackData {
+ public:
+ CallbackData(AudioOutput *cookie) {
+ mData = cookie;
+ mSwitching = false;
+ }
+ AudioOutput * getOutput() { return mData;}
+ void setOutput(AudioOutput* newcookie) { mData = newcookie; }
+ // lock/unlock are used by the callback before accessing the payload of this object
+ void lock() { mLock.lock(); }
+ void unlock() { mLock.unlock(); }
+ // beginTrackSwitch/endTrackSwitch are used when this object is being handed over
+ // to the next sink.
+ void beginTrackSwitch() { mLock.lock(); mSwitching = true; }
+ void endTrackSwitch() {
+ if (mSwitching) {
+ mLock.unlock();
+ }
+ mSwitching = false;
+ }
+ private:
+ AudioOutput * mData;
+ mutable Mutex mLock;
+ bool mSwitching;
+ DISALLOW_EVIL_CONSTRUCTORS(CallbackData);
+ };
+
+ }; // AudioOutput
+
class AudioCache : public MediaPlayerBase::AudioSink
{
@@ -138,8 +185,8 @@
virtual int getSessionId();
virtual status_t open(
- uint32_t sampleRate, int channelCount, int format,
- int bufferCount = 1,
+ uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
+ audio_format_t format, int bufferCount = 1,
AudioCallback cb = NULL, void *cookie = NULL);
virtual void start();
@@ -148,10 +195,11 @@
virtual void flush() {}
virtual void pause() {}
virtual void close() {}
- void setAudioStreamType(int streamType) {}
+ void setAudioStreamType(audio_stream_type_t streamType) {}
void setVolume(float left, float right) {}
+ virtual status_t setPlaybackRatePermille(int32_t ratePermille) { return INVALID_OPERATION; }
uint32_t sampleRate() const { return mSampleRate; }
- uint32_t format() const { return (uint32_t)mFormat; }
+ audio_format_t format() const { return mFormat; }
size_t size() const { return mSize; }
status_t wait();
@@ -169,7 +217,7 @@
sp<MemoryHeapBase> mHeap;
float mMsecsPerFrame;
uint16_t mChannelCount;
- uint16_t mFormat;
+ audio_format_t mFormat;
ssize_t mFrameCount;
uint32_t mSampleRate;
uint32_t mSize;
@@ -177,7 +225,7 @@
bool mCommandComplete;
sp<Thread> mCallbackThread;
- };
+ }; // AudioCache
public:
static void instantiate();
@@ -189,8 +237,8 @@
virtual sp<IMediaPlayer> create(pid_t pid, const sp<IMediaPlayerClient>& client, int audioSessionId);
- virtual sp<IMemory> decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, int* pFormat);
- virtual sp<IMemory> decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, int* pFormat);
+ virtual sp<IMemory> decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat);
+ virtual sp<IMemory> decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat);
virtual sp<IOMX> getOMX();
virtual status_t dump(int fd, const Vector<String16>& args);
@@ -258,7 +306,7 @@
virtual status_t getCurrentPosition(int* msec);
virtual status_t getDuration(int* msec);
virtual status_t reset();
- virtual status_t setAudioStreamType(int type);
+ virtual status_t setAudioStreamType(audio_stream_type_t type);
virtual status_t setLooping(int loop);
virtual status_t setVolume(float leftVolume, float rightVolume);
virtual status_t invoke(const Parcel& request, Parcel *reply);
@@ -270,6 +318,8 @@
virtual status_t attachAuxEffect(int effectId);
virtual status_t setParameter(int key, const Parcel &request);
virtual status_t getParameter(int key, Parcel *reply);
+ virtual status_t setRetransmitEndpoint(const struct sockaddr_in* endpoint);
+ virtual status_t setNextPlayer(const sp<IMediaPlayer>& player);
sp<MediaPlayerBase> createPlayer(player_type playerType);
@@ -281,6 +331,14 @@
virtual status_t setDataSource(const sp<IStreamSource> &source);
+ sp<MediaPlayerBase> setDataSource_pre(player_type playerType);
+ void setDataSource_post(const sp<MediaPlayerBase>& p,
+ status_t status);
+
+ player_type getPlayerType(int fd, int64_t offset, int64_t length);
+ player_type getPlayerType(const char* url);
+ player_type getPlayerType(const sp<IStreamSource> &source);
+
static void notify(void* cookie, int msg,
int ext1, int ext2, const Parcel *obj);
@@ -332,6 +390,9 @@
uid_t mUID;
sp<ANativeWindow> mConnectedWindow;
sp<IBinder> mConnectedWindowBinder;
+ struct sockaddr_in mRetransmitEndpoint;
+ bool mRetransmitEndpointValid;
+ sp<Client> mNextClient;
// Metadata filters.
media::Metadata::Filter mMetadataAllow; // protected by mLock
@@ -346,7 +407,7 @@
#if CALLBACK_ANTAGONIZER
Antagonizer* mAntagonizer;
#endif
- };
+ }; // Client
// ----------------------------------------------------------------------------
diff --git a/media/libmediaplayerservice/MediaRecorderClient.cpp b/media/libmediaplayerservice/MediaRecorderClient.cpp
index d219fc2..beda945 100644
--- a/media/libmediaplayerservice/MediaRecorderClient.cpp
+++ b/media/libmediaplayerservice/MediaRecorderClient.cpp
@@ -33,8 +33,6 @@
#include <utils/String16.h>
-#include <media/AudioTrack.h>
-
#include <system/audio.h>
#include "MediaRecorderClient.h"
diff --git a/media/libmediaplayerservice/MidiFile.cpp b/media/libmediaplayerservice/MidiFile.cpp
index d89b5f4..8db5b9b 100644
--- a/media/libmediaplayerservice/MidiFile.cpp
+++ b/media/libmediaplayerservice/MidiFile.cpp
@@ -86,7 +86,8 @@
// create playback thread
{
Mutex::Autolock l(mMutex);
- createThreadEtc(renderThread, this, "midithread", ANDROID_PRIORITY_AUDIO);
+ mThread = new MidiFileThread(this);
+ mThread->run("midithread", ANDROID_PRIORITY_AUDIO);
mCondition.wait(mMutex);
ALOGV("thread started");
}
@@ -420,18 +421,14 @@
}
status_t MidiFile::createOutputTrack() {
- if (mAudioSink->open(pLibConfig->sampleRate, pLibConfig->numChannels, AUDIO_FORMAT_PCM_16_BIT, 2) != NO_ERROR) {
+ if (mAudioSink->open(pLibConfig->sampleRate, pLibConfig->numChannels,
+ CHANNEL_MASK_USE_CHANNEL_ORDER, AUDIO_FORMAT_PCM_16_BIT, 2) != NO_ERROR) {
ALOGE("mAudioSink open failed");
return ERROR_OPEN_FAILED;
}
return NO_ERROR;
}
-int MidiFile::renderThread(void* p) {
-
- return ((MidiFile*)p)->render();
-}
-
int MidiFile::render() {
EAS_RESULT result = EAS_FAILURE;
EAS_I32 count;
diff --git a/media/libmediaplayerservice/MidiFile.h b/media/libmediaplayerservice/MidiFile.h
index 3469389..f6f8f7b 100644
--- a/media/libmediaplayerservice/MidiFile.h
+++ b/media/libmediaplayerservice/MidiFile.h
@@ -19,11 +19,11 @@
#define ANDROID_MIDIFILE_H
#include <media/MediaPlayerInterface.h>
-#include <media/AudioTrack.h>
#include <libsonivox/eas.h>
namespace android {
+// Note that the name MidiFile is misleading; this actually represents a MIDI file player
class MidiFile : public MediaPlayerInterface {
public:
MidiFile();
@@ -65,7 +65,6 @@
private:
status_t createOutputTrack();
status_t reset_nosync();
- static int renderThread(void*);
int render();
void updateState(){ EAS_State(mEasData, mEasHandle, &mState); }
@@ -78,12 +77,35 @@
EAS_I32 mDuration;
EAS_STATE mState;
EAS_FILE mFileLocator;
- int mStreamType;
+ audio_stream_type_t mStreamType;
bool mLoop;
volatile bool mExit;
bool mPaused;
volatile bool mRender;
pid_t mTid;
+
+ class MidiFileThread : public Thread {
+ public:
+ MidiFileThread(MidiFile *midiPlayer) : mMidiFile(midiPlayer) {
+ }
+
+ protected:
+ virtual ~MidiFileThread() {}
+
+ private:
+ MidiFile *mMidiFile;
+
+ bool threadLoop() {
+ int result;
+ result = mMidiFile->render();
+ return false;
+ }
+
+ MidiFileThread(const MidiFileThread &);
+ MidiFileThread &operator=(const MidiFileThread &);
+ };
+
+ sp<MidiFileThread> mThread;
};
}; // namespace android
diff --git a/media/libmediaplayerservice/StagefrightPlayer.cpp b/media/libmediaplayerservice/StagefrightPlayer.cpp
index 6d7771a..619c149 100644
--- a/media/libmediaplayerservice/StagefrightPlayer.cpp
+++ b/media/libmediaplayerservice/StagefrightPlayer.cpp
@@ -166,7 +166,8 @@
}
status_t StagefrightPlayer::invoke(const Parcel &request, Parcel *reply) {
- return INVALID_OPERATION;
+ ALOGV("invoke()");
+ return mPlayer->invoke(request, reply);
}
void StagefrightPlayer::setAudioSink(const sp<AudioSink> &audioSink) {
@@ -176,7 +177,7 @@
}
status_t StagefrightPlayer::setParameter(int key, const Parcel &request) {
- ALOGV("setParameter");
+ ALOGV("setParameter(key=%d)", key);
return mPlayer->setParameter(key, request);
}
diff --git a/media/libmediaplayerservice/StagefrightRecorder.cpp b/media/libmediaplayerservice/StagefrightRecorder.cpp
index 4632016..ca79657 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.cpp
+++ b/media/libmediaplayerservice/StagefrightRecorder.cpp
@@ -24,6 +24,7 @@
#include <binder/IServiceManager.h>
#include <media/IMediaPlayerService.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/AudioSource.h>
#include <media/stagefright/AMRWriter.h>
#include <media/stagefright/AACWriter.h>
@@ -31,7 +32,6 @@
#include <media/stagefright/CameraSourceTimeLapse.h>
#include <media/stagefright/MPEG2TSWriter.h>
#include <media/stagefright/MPEG4Writer.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/OMXClient.h>
@@ -40,7 +40,7 @@
#include <media/MediaProfiles.h>
#include <camera/ICamera.h>
#include <camera/CameraParameters.h>
-#include <surfaceflinger/Surface.h>
+#include <gui/Surface.h>
#include <utils/Errors.h>
#include <sys/types.h>
@@ -241,8 +241,8 @@
status_t StagefrightRecorder::setOutputFile(int fd, int64_t offset, int64_t length) {
ALOGV("setOutputFile: %d, %lld, %lld", fd, offset, length);
// These don't make any sense, do they?
- CHECK_EQ(offset, 0);
- CHECK_EQ(length, 0);
+ CHECK_EQ(offset, 0ll);
+ CHECK_EQ(length, 0ll);
if (fd < 0) {
ALOGE("Invalid file descriptor: %d", fd);
@@ -734,7 +734,7 @@
}
status_t StagefrightRecorder::start() {
- CHECK(mOutputFd >= 0);
+ CHECK_GE(mOutputFd, 0);
if (mWriter != NULL) {
ALOGE("File writer is not avaialble");
@@ -837,7 +837,7 @@
}
OMXClient client;
- CHECK_EQ(client.connect(), OK);
+ CHECK_EQ(client.connect(), (status_t)OK);
sp<MediaSource> audioEncoder =
OMXCodec::Create(client.interface(), encMeta,
@@ -850,9 +850,9 @@
status_t StagefrightRecorder::startAACRecording() {
// FIXME:
// Add support for OUTPUT_FORMAT_AAC_ADIF
- CHECK(mOutputFormat == OUTPUT_FORMAT_AAC_ADTS);
+ CHECK_EQ(mOutputFormat, OUTPUT_FORMAT_AAC_ADTS);
- CHECK(mAudioEncoder == AUDIO_ENCODER_AAC);
+ CHECK_EQ(mAudioEncoder, AUDIO_ENCODER_AAC);
CHECK(mAudioSource != AUDIO_SOURCE_CNT);
mWriter = new AACWriter(mOutputFd);
@@ -1291,6 +1291,12 @@
videoSize.width = mVideoWidth;
videoSize.height = mVideoHeight;
if (mCaptureTimeLapse) {
+ if (mTimeBetweenTimeLapseFrameCaptureUs < 0) {
+ ALOGE("Invalid mTimeBetweenTimeLapseFrameCaptureUs value: %lld",
+ mTimeBetweenTimeLapseFrameCaptureUs);
+ return BAD_VALUE;
+ }
+
mCameraSourceTimeLapse = CameraSourceTimeLapse::CreateFromCamera(
mCamera, mCameraProxy, mCameraId,
videoSize, mFrameRate, mPreviewSurface,
@@ -1386,7 +1392,7 @@
}
OMXClient client;
- CHECK_EQ(client.connect(), OK);
+ CHECK_EQ(client.connect(), (status_t)OK);
uint32_t encoder_flags = 0;
if (mIsMetaDataStoredInVideoBuffers) {
diff --git a/media/libmediaplayerservice/nuplayer/Android.mk b/media/libmediaplayerservice/nuplayer/Android.mk
index 33e2f93..9b485d7 100644
--- a/media/libmediaplayerservice/nuplayer/Android.mk
+++ b/media/libmediaplayerservice/nuplayer/Android.mk
@@ -12,11 +12,11 @@
StreamingSource.cpp \
LOCAL_C_INCLUDES := \
- $(TOP)/frameworks/base/include/media/stagefright/openmax \
+ $(TOP)/frameworks/base/media/libstagefright/httplive \
$(TOP)/frameworks/base/media/libstagefright/include \
- $(TOP)/frameworks/base/media/libstagefright/mpeg2ts \
- $(TOP)/frameworks/base/media/libstagefright/httplive \
- $(TOP)/frameworks/base/media/libstagefright/rtsp \
+ $(TOP)/frameworks/base/media/libstagefright/mpeg2ts \
+ $(TOP)/frameworks/base/media/libstagefright/rtsp \
+ $(TOP)/frameworks/native/include/media/openmax
LOCAL_MODULE:= libstagefright_nuplayer
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
index a00aaa5..526120a 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
@@ -38,7 +38,6 @@
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
-#include <surfaceflinger/Surface.h>
#include <gui/ISurfaceTexture.h>
#include "avc_utils.h"
@@ -337,6 +336,7 @@
CHECK_EQ(mAudioSink->open(
sampleRate,
numChannels,
+ CHANNEL_MASK_USE_CHANNEL_ORDER,
AUDIO_FORMAT_PCM_16_BIT,
8 /* bufferCount */),
(status_t)OK);
@@ -387,10 +387,10 @@
audio ? "audio" : "video");
mRenderer->queueEOS(audio, UNKNOWN_ERROR);
- } else {
- CHECK_EQ((int)what, (int)ACodec::kWhatDrainThisBuffer);
-
+ } else if (what == ACodec::kWhatDrainThisBuffer) {
renderBuffer(audio, codecRequest);
+ } else {
+ ALOGV("Unhandled codec notification %d.", what);
}
break;
@@ -480,7 +480,7 @@
// completed.
ALOGV("postponing reset mFlushingAudio=%d, mFlushingVideo=%d",
- mFlushingAudio, mFlushingVideo);
+ mFlushingAudio, mFlushingVideo);
mResetPostponed = true;
break;
@@ -690,7 +690,7 @@
bool timeChange = (type & ATSParser::DISCONTINUITY_TIME) != 0;
ALOGI("%s discontinuity (formatChange=%d, time=%d)",
- audio ? "audio" : "video", formatChange, timeChange);
+ audio ? "audio" : "video", formatChange, timeChange);
if (audio) {
mSkipRenderingAudioUntilMediaTimeUs = -1;
@@ -768,7 +768,7 @@
mediaTimeUs / 1E6);
#endif
- reply->setObject("buffer", accessUnit);
+ reply->setBuffer("buffer", accessUnit);
reply->post();
return OK;
@@ -793,10 +793,8 @@
return;
}
- sp<RefBase> obj;
- CHECK(msg->findObject("buffer", &obj));
-
- sp<ABuffer> buffer = static_cast<ABuffer *>(obj.get());
+ sp<ABuffer> buffer;
+ CHECK(msg->findBuffer("buffer", &buffer));
int64_t &skipUntilMediaTimeUs =
audio
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.h b/media/libmediaplayerservice/nuplayer/NuPlayer.h
index ffc710e..6be14be 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.h
@@ -21,8 +21,6 @@
#include <media/MediaPlayerInterface.h>
#include <media/stagefright/foundation/AHandler.h>
#include <media/stagefright/NativeWindowWrapper.h>
-#include <gui/SurfaceTextureClient.h>
-#include <surfaceflinger/Surface.h>
namespace android {
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
index 56c2773..460fc98 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
@@ -29,8 +29,6 @@
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/Utils.h>
-#include <surfaceflinger/Surface.h>
-#include <gui/ISurfaceTexture.h>
namespace android {
@@ -214,8 +212,6 @@
buffer->meta()->setInt32("csd", true);
mCSD.push(buffer);
-
- msg->setObject("csd", buffer);
} else if (meta->findData(kKeyESDS, &type, &data, &size)) {
ESDS esds((const char *)data, size);
CHECK_EQ(esds.InitCheck(), (status_t)OK);
@@ -242,9 +238,8 @@
CHECK(msg->findMessage("reply", &reply));
#if 0
- sp<RefBase> obj;
- CHECK(msg->findObject("buffer", &obj));
- sp<ABuffer> outBuffer = static_cast<ABuffer *>(obj.get());
+ sp<ABuffer> outBuffer;
+ CHECK(msg->findBuffer("buffer", &outBuffer));
#else
sp<ABuffer> outBuffer;
#endif
@@ -253,7 +248,7 @@
outBuffer = mCSD.editItemAt(mCSDIndex++);
outBuffer->meta()->setInt64("timeUs", 0);
- reply->setObject("buffer", outBuffer);
+ reply->setBuffer("buffer", outBuffer);
reply->post();
return;
}
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
index 074cb4f..5738ecb 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
@@ -60,7 +60,7 @@
const sp<AMessage> ¬ifyConsumed) {
sp<AMessage> msg = new AMessage(kWhatQueueBuffer, id());
msg->setInt32("audio", static_cast<int32_t>(audio));
- msg->setObject("buffer", buffer);
+ msg->setBuffer("buffer", buffer);
msg->setMessage("notifyConsumed", notifyConsumed);
msg->post();
}
@@ -376,7 +376,7 @@
bool tooLate = (mVideoLateByUs > 40000);
if (tooLate) {
- ALOGV("video late by %lld us (%.2f secs)", lateByUs, lateByUs / 1E6);
+ ALOGV("video late by %lld us (%.2f secs)", mVideoLateByUs, mVideoLateByUs / 1E6);
} else {
ALOGV("rendering video at media time %.2f secs", mediaTimeUs / 1E6);
}
@@ -411,9 +411,8 @@
return;
}
- sp<RefBase> obj;
- CHECK(msg->findObject("buffer", &obj));
- sp<ABuffer> buffer = static_cast<ABuffer *>(obj.get());
+ sp<ABuffer> buffer;
+ CHECK(msg->findBuffer("buffer", &buffer));
sp<AMessage> notifyConsumed;
CHECK(msg->findMessage("notifyConsumed", ¬ifyConsumed));
@@ -629,7 +628,7 @@
}
ALOGV("now paused audio queue has %d entries, video has %d entries",
- mAudioQueue.size(), mVideoQueue.size());
+ mAudioQueue.size(), mVideoQueue.size());
mPaused = true;
}
diff --git a/media/libmediaplayerservice/nuplayer/RTSPSource.cpp b/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
index 6eb0d07..4c65b65 100644
--- a/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
@@ -218,10 +218,8 @@
CHECK(msg->findSize("trackIndex", &trackIndex));
CHECK_LT(trackIndex, mTracks.size());
- sp<RefBase> obj;
- CHECK(msg->findObject("accessUnit", &obj));
-
- sp<ABuffer> accessUnit = static_cast<ABuffer *>(obj.get());
+ sp<ABuffer> accessUnit;
+ CHECK(msg->findBuffer("accessUnit", &accessUnit));
int32_t damaged;
if (accessUnit->meta()->findInt32("damaged", &damaged)
diff --git a/media/libstagefright/AACExtractor.cpp b/media/libstagefright/AACExtractor.cpp
index 52b1200..4d1072f 100644
--- a/media/libstagefright/AACExtractor.cpp
+++ b/media/libstagefright/AACExtractor.cpp
@@ -22,9 +22,10 @@
#include "include/avc_utils.h"
#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaSource.h>
@@ -131,18 +132,28 @@
return frameSize;
}
-AACExtractor::AACExtractor(const sp<DataSource> &source)
+AACExtractor::AACExtractor(
+ const sp<DataSource> &source, const sp<AMessage> &_meta)
: mDataSource(source),
mInitCheck(NO_INIT),
mFrameDurationUs(0) {
- String8 mimeType;
- float confidence;
- if (!SniffAAC(mDataSource, &mimeType, &confidence, NULL)) {
- return;
+ sp<AMessage> meta = _meta;
+
+ if (meta == NULL) {
+ String8 mimeType;
+ float confidence;
+ sp<AMessage> _meta;
+
+ if (!SniffAAC(mDataSource, &mimeType, &confidence, &meta)) {
+ return;
+ }
}
+ int64_t offset;
+ CHECK(meta->findInt64("offset", &offset));
+
uint8_t profile, sf_index, channel, header[2];
- if (mDataSource->readAt(2, &header, 2) < 2) {
+ if (mDataSource->readAt(offset + 2, &header, 2) < 2) {
return;
}
@@ -156,7 +167,6 @@
mMeta = MakeAACCodecSpecificData(profile, sf_index, channel);
- off64_t offset = 0;
off64_t streamSize, numFrames = 0;
size_t frameSize = 0;
int64_t duration = 0;
@@ -245,7 +255,12 @@
status_t AACSource::start(MetaData *params) {
CHECK(!mStarted);
- mOffset = 0;
+ if (mOffsetVector.empty()) {
+ mOffset = 0;
+ } else {
+ mOffset = mOffsetVector.itemAt(0);
+ }
+
mCurrentTimeUs = 0;
mGroup = new MediaBufferGroup;
mGroup->add_buffer(new MediaBuffer(kMaxFrameSize));
@@ -318,10 +333,39 @@
bool SniffAAC(
const sp<DataSource> &source, String8 *mimeType, float *confidence,
- sp<AMessage> *) {
+ sp<AMessage> *meta) {
+ off64_t pos = 0;
+
+ for (;;) {
+ uint8_t id3header[10];
+ if (source->readAt(pos, id3header, sizeof(id3header))
+ < (ssize_t)sizeof(id3header)) {
+ return false;
+ }
+
+ if (memcmp("ID3", id3header, 3)) {
+ break;
+ }
+
+ // Skip the ID3v2 header.
+
+ size_t len =
+ ((id3header[6] & 0x7f) << 21)
+ | ((id3header[7] & 0x7f) << 14)
+ | ((id3header[8] & 0x7f) << 7)
+ | (id3header[9] & 0x7f);
+
+ len += 10;
+
+ pos += len;
+
+ ALOGV("skipped ID3 tag, new starting offset is %lld (0x%016llx)",
+ pos, pos);
+ }
+
uint8_t header[2];
- if (source->readAt(0, &header, 2) != 2) {
+ if (source->readAt(pos, &header, 2) != 2) {
return false;
}
@@ -329,6 +373,10 @@
if ((header[0] == 0xff) && ((header[1] & 0xf6) == 0xf0)) {
*mimeType = MEDIA_MIMETYPE_AUDIO_AAC_ADTS;
*confidence = 0.2;
+
+ *meta = new AMessage;
+ (*meta)->setInt64("offset", pos);
+
return true;
}
diff --git a/media/libstagefright/AACWriter.cpp b/media/libstagefright/AACWriter.cpp
index 1673ccd..9cdb463 100644
--- a/media/libstagefright/AACWriter.cpp
+++ b/media/libstagefright/AACWriter.cpp
@@ -60,7 +60,7 @@
AACWriter::~AACWriter() {
if (mStarted) {
- stop();
+ reset();
}
if (mFd != -1) {
@@ -152,7 +152,7 @@
return OK;
}
-status_t AACWriter::stop() {
+status_t AACWriter::reset() {
if (!mStarted) {
return OK;
}
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index ca44ea3..e5ad4b7 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -26,14 +26,12 @@
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/MediaCodecList.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/NativeWindowWrapper.h>
#include <media/stagefright/OMXClient.h>
#include <media/stagefright/OMXCodec.h>
-#include <surfaceflinger/Surface.h>
-#include <gui/SurfaceTextureClient.h>
-
#include <OMX_Component.h>
namespace android {
@@ -168,15 +166,36 @@
protected:
virtual bool onMessageReceived(const sp<AMessage> &msg);
+ virtual void stateEntered();
private:
void onSetup(const sp<AMessage> &msg);
+ bool onAllocateComponent(const sp<AMessage> &msg);
DISALLOW_EVIL_CONSTRUCTORS(UninitializedState);
};
////////////////////////////////////////////////////////////////////////////////
+struct ACodec::LoadedState : public ACodec::BaseState {
+ LoadedState(ACodec *codec);
+
+protected:
+ virtual bool onMessageReceived(const sp<AMessage> &msg);
+ virtual void stateEntered();
+
+private:
+ friend struct ACodec::UninitializedState;
+
+ bool onConfigureComponent(const sp<AMessage> &msg);
+ void onStart();
+ void onShutdown(bool keepComponentAllocated);
+
+ DISALLOW_EVIL_CONSTRUCTORS(LoadedState);
+};
+
+////////////////////////////////////////////////////////////////////////////////
+
struct ACodec::LoadedToIdleState : public ACodec::BaseState {
LoadedToIdleState(ACodec *codec);
@@ -265,6 +284,8 @@
private:
void changeStateIfWeOwnAllBuffers();
+ bool mComponentNowIdle;
+
DISALLOW_EVIL_CONSTRUCTORS(ExecutingToIdleState);
};
@@ -308,9 +329,13 @@
////////////////////////////////////////////////////////////////////////////////
ACodec::ACodec()
- : mNode(NULL),
- mSentFormat(false) {
+ : mQuirks(0),
+ mNode(NULL),
+ mSentFormat(false),
+ mIsEncoder(false),
+ mShutdownInProgress(false) {
mUninitializedState = new UninitializedState(this);
+ mLoadedState = new LoadedState(this);
mLoadedToIdleState = new LoadedToIdleState(this);
mIdleToExecutingState = new IdleToExecutingState(this);
mExecutingState = new ExecutingState(this);
@@ -341,6 +366,22 @@
msg->post();
}
+void ACodec::initiateAllocateComponent(const sp<AMessage> &msg) {
+ msg->setWhat(kWhatAllocateComponent);
+ msg->setTarget(id());
+ msg->post();
+}
+
+void ACodec::initiateConfigureComponent(const sp<AMessage> &msg) {
+ msg->setWhat(kWhatConfigureComponent);
+ msg->setTarget(id());
+ msg->post();
+}
+
+void ACodec::initiateStart() {
+ (new AMessage(kWhatStart, id()))->post();
+}
+
void ACodec::signalFlush() {
ALOGV("[%s] signalFlush", mComponentName.c_str());
(new AMessage(kWhatFlush, id()))->post();
@@ -350,8 +391,10 @@
(new AMessage(kWhatResume, id()))->post();
}
-void ACodec::initiateShutdown() {
- (new AMessage(kWhatShutdown, id()))->post();
+void ACodec::initiateShutdown(bool keepComponentAllocated) {
+ sp<AMessage> msg = new AMessage(kWhatShutdown, id());
+ msg->setInt32("keepComponentAllocated", keepComponentAllocated);
+ msg->post();
}
status_t ACodec::allocateBuffersOnPort(OMX_U32 portIndex) {
@@ -360,62 +403,71 @@
CHECK(mDealer[portIndex] == NULL);
CHECK(mBuffers[portIndex].isEmpty());
+ status_t err;
if (mNativeWindow != NULL && portIndex == kPortIndexOutput) {
- return allocateOutputBuffersFromNativeWindow();
+ err = allocateOutputBuffersFromNativeWindow();
+ } else {
+ OMX_PARAM_PORTDEFINITIONTYPE def;
+ InitOMXParams(&def);
+ def.nPortIndex = portIndex;
+
+ err = mOMX->getParameter(
+ mNode, OMX_IndexParamPortDefinition, &def, sizeof(def));
+
+ if (err == OK) {
+ ALOGV("[%s] Allocating %lu buffers of size %lu on %s port",
+ mComponentName.c_str(),
+ def.nBufferCountActual, def.nBufferSize,
+ portIndex == kPortIndexInput ? "input" : "output");
+
+ size_t totalSize = def.nBufferCountActual * def.nBufferSize;
+ mDealer[portIndex] = new MemoryDealer(totalSize, "ACodec");
+
+ for (OMX_U32 i = 0; i < def.nBufferCountActual; ++i) {
+ sp<IMemory> mem = mDealer[portIndex]->allocate(def.nBufferSize);
+ CHECK(mem.get() != NULL);
+
+ IOMX::buffer_id buffer;
+
+ uint32_t requiresAllocateBufferBit =
+ (portIndex == kPortIndexInput)
+ ? OMXCodec::kRequiresAllocateBufferOnInputPorts
+ : OMXCodec::kRequiresAllocateBufferOnOutputPorts;
+
+ if (mQuirks & requiresAllocateBufferBit) {
+ err = mOMX->allocateBufferWithBackup(
+ mNode, portIndex, mem, &buffer);
+ } else {
+ err = mOMX->useBuffer(mNode, portIndex, mem, &buffer);
+ }
+
+ BufferInfo info;
+ info.mBufferID = buffer;
+ info.mStatus = BufferInfo::OWNED_BY_US;
+ info.mData = new ABuffer(mem->pointer(), def.nBufferSize);
+ mBuffers[portIndex].push(info);
+ }
+ }
}
- OMX_PARAM_PORTDEFINITIONTYPE def;
- InitOMXParams(&def);
- def.nPortIndex = portIndex;
-
- status_t err = mOMX->getParameter(
- mNode, OMX_IndexParamPortDefinition, &def, sizeof(def));
-
if (err != OK) {
return err;
}
- ALOGV("[%s] Allocating %lu buffers of size %lu on %s port",
- mComponentName.c_str(),
- def.nBufferCountActual, def.nBufferSize,
- portIndex == kPortIndexInput ? "input" : "output");
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", ACodec::kWhatBuffersAllocated);
- size_t totalSize = def.nBufferCountActual * def.nBufferSize;
- mDealer[portIndex] = new MemoryDealer(totalSize, "OMXCodec");
+ notify->setInt32("portIndex", portIndex);
+ for (size_t i = 0; i < mBuffers[portIndex].size(); ++i) {
+ AString name = StringPrintf("buffer-id_%d", i);
+ notify->setPointer(name.c_str(), mBuffers[portIndex][i].mBufferID);
- for (OMX_U32 i = 0; i < def.nBufferCountActual; ++i) {
- sp<IMemory> mem = mDealer[portIndex]->allocate(def.nBufferSize);
- CHECK(mem.get() != NULL);
-
- IOMX::buffer_id buffer;
-
- if (!strcasecmp(
- mComponentName.c_str(), "OMX.TI.DUCATI1.VIDEO.DECODER")) {
- if (portIndex == kPortIndexInput && i == 0) {
- // Only log this warning once per allocation round.
-
- ALOGW("OMX.TI.DUCATI1.VIDEO.DECODER requires the use of "
- "OMX_AllocateBuffer instead of the preferred "
- "OMX_UseBuffer. Vendor must fix this.");
- }
-
- err = mOMX->allocateBufferWithBackup(
- mNode, portIndex, mem, &buffer);
- } else {
- err = mOMX->useBuffer(mNode, portIndex, mem, &buffer);
- }
-
- if (err != OK) {
- return err;
- }
-
- BufferInfo info;
- info.mBufferID = buffer;
- info.mStatus = BufferInfo::OWNED_BY_US;
- info.mData = new ABuffer(mem->pointer(), def.nBufferSize);
- mBuffers[portIndex].push(info);
+ name = StringPrintf("data_%d", i);
+ notify->setBuffer(name.c_str(), mBuffers[portIndex][i].mData);
}
+ notify->post();
+
return OK;
}
@@ -671,7 +723,7 @@
return NULL;
}
-void ACodec::setComponentRole(
+status_t ACodec::setComponentRole(
bool isEncoder, const char *mime) {
struct MimeToRole {
const char *mime;
@@ -694,12 +746,18 @@
"audio_decoder.aac", "audio_encoder.aac" },
{ MEDIA_MIMETYPE_AUDIO_VORBIS,
"audio_decoder.vorbis", "audio_encoder.vorbis" },
+ { MEDIA_MIMETYPE_AUDIO_G711_MLAW,
+ "audio_decoder.g711mlaw", "audio_encoder.g711mlaw" },
+ { MEDIA_MIMETYPE_AUDIO_G711_ALAW,
+ "audio_decoder.g711alaw", "audio_encoder.g711alaw" },
{ MEDIA_MIMETYPE_VIDEO_AVC,
"video_decoder.avc", "video_encoder.avc" },
{ MEDIA_MIMETYPE_VIDEO_MPEG4,
"video_decoder.mpeg4", "video_encoder.mpeg4" },
{ MEDIA_MIMETYPE_VIDEO_H263,
"video_decoder.h263", "video_encoder.h263" },
+ { MEDIA_MIMETYPE_VIDEO_VPX,
+ "video_decoder.vpx", "video_encoder.vpx" },
};
static const size_t kNumMimeToRole =
@@ -713,7 +771,7 @@
}
if (i == kNumMimeToRole) {
- return;
+ return ERROR_UNSUPPORTED;
}
const char *role =
@@ -736,50 +794,79 @@
if (err != OK) {
ALOGW("[%s] Failed to set standard component role '%s'.",
mComponentName.c_str(), role);
+
+ return err;
}
}
+
+ return OK;
}
-void ACodec::configureCodec(
+status_t ACodec::configureCodec(
const char *mime, const sp<AMessage> &msg) {
- setComponentRole(false /* isEncoder */, mime);
+ int32_t encoder;
+ if (!msg->findInt32("encoder", &encoder)) {
+ encoder = false;
+ }
+
+ mIsEncoder = encoder;
+
+ status_t err = setComponentRole(encoder /* isEncoder */, mime);
+
+ if (err != OK) {
+ return err;
+ }
+
+ int32_t bitRate = 0;
+ if (encoder && !msg->findInt32("bitrate", &bitRate)) {
+ return INVALID_OPERATION;
+ }
if (!strncasecmp(mime, "video/", 6)) {
- int32_t width, height;
- CHECK(msg->findInt32("width", &width));
- CHECK(msg->findInt32("height", &height));
-
- CHECK_EQ(setupVideoDecoder(mime, width, height),
- (status_t)OK);
+ if (encoder) {
+ err = setupVideoEncoder(mime, msg);
+ } else {
+ int32_t width, height;
+ if (!msg->findInt32("width", &width)
+ || !msg->findInt32("height", &height)) {
+ err = INVALID_OPERATION;
+ } else {
+ err = setupVideoDecoder(mime, width, height);
+ }
+ }
} else if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_AAC)) {
int32_t numChannels, sampleRate;
- CHECK(msg->findInt32("channel-count", &numChannels));
- CHECK(msg->findInt32("sample-rate", &sampleRate));
-
- CHECK_EQ(setupAACDecoder(numChannels, sampleRate), (status_t)OK);
+ if (!msg->findInt32("channel-count", &numChannels)
+ || !msg->findInt32("sample-rate", &sampleRate)) {
+ err = INVALID_OPERATION;
+ } else {
+ err = setupAACCodec(encoder, numChannels, sampleRate, bitRate);
+ }
} else if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_AMR_NB)) {
- CHECK_EQ(setupAMRDecoder(false /* isWAMR */), (status_t)OK);
+ err = setupAMRCodec(encoder, false /* isWAMR */, bitRate);
} else if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_AMR_WB)) {
- CHECK_EQ(setupAMRDecoder(true /* isWAMR */), (status_t)OK);
+ err = setupAMRCodec(encoder, true /* isWAMR */, bitRate);
} else if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_G711_ALAW)
|| !strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_G711_MLAW)) {
// These are PCM-like formats with a fixed sample rate but
// a variable number of channels.
int32_t numChannels;
- CHECK(msg->findInt32("channel-count", &numChannels));
-
- CHECK_EQ(setupG711Decoder(numChannels), (status_t)OK);
+ if (!msg->findInt32("channel-count", &numChannels)) {
+ err = INVALID_OPERATION;
+ } else {
+ err = setupG711Codec(encoder, numChannels);
+ }
}
int32_t maxInputSize;
if (msg->findInt32("max-input-size", &maxInputSize)) {
- CHECK_EQ(setMinBufferSize(kPortIndexInput, (size_t)maxInputSize),
- (status_t)OK);
+ err = setMinBufferSize(kPortIndexInput, (size_t)maxInputSize);
} else if (!strcmp("OMX.Nvidia.aac.decoder", mComponentName.c_str())) {
- CHECK_EQ(setMinBufferSize(kPortIndexInput, 8192), // XXX
- (status_t)OK);
+ err = setMinBufferSize(kPortIndexInput, 8192); // XXX
}
+
+ return err;
}
status_t ACodec::setMinBufferSize(OMX_U32 portIndex, size_t size) {
@@ -819,12 +906,113 @@
return OK;
}
-status_t ACodec::setupAACDecoder(int32_t numChannels, int32_t sampleRate) {
+status_t ACodec::selectAudioPortFormat(
+ OMX_U32 portIndex, OMX_AUDIO_CODINGTYPE desiredFormat) {
+ OMX_AUDIO_PARAM_PORTFORMATTYPE format;
+ InitOMXParams(&format);
+
+ format.nPortIndex = portIndex;
+ for (OMX_U32 index = 0;; ++index) {
+ format.nIndex = index;
+
+ status_t err = mOMX->getParameter(
+ mNode, OMX_IndexParamAudioPortFormat,
+ &format, sizeof(format));
+
+ if (err != OK) {
+ return err;
+ }
+
+ if (format.eEncoding == desiredFormat) {
+ break;
+ }
+ }
+
+ return mOMX->setParameter(
+ mNode, OMX_IndexParamAudioPortFormat, &format, sizeof(format));
+}
+
+status_t ACodec::setupAACCodec(
+ bool encoder,
+ int32_t numChannels, int32_t sampleRate, int32_t bitRate) {
+ status_t err = setupRawAudioFormat(
+ encoder ? kPortIndexInput : kPortIndexOutput,
+ sampleRate,
+ numChannels);
+
+ if (err != OK) {
+ return err;
+ }
+
+ if (encoder) {
+ err = selectAudioPortFormat(kPortIndexOutput, OMX_AUDIO_CodingAAC);
+
+ if (err != OK) {
+ return err;
+ }
+
+ OMX_PARAM_PORTDEFINITIONTYPE def;
+ InitOMXParams(&def);
+ def.nPortIndex = kPortIndexOutput;
+
+ err = mOMX->getParameter(
+ mNode, OMX_IndexParamPortDefinition, &def, sizeof(def));
+
+ if (err != OK) {
+ return err;
+ }
+
+ def.format.audio.bFlagErrorConcealment = OMX_TRUE;
+ def.format.audio.eEncoding = OMX_AUDIO_CodingAAC;
+
+ err = mOMX->setParameter(
+ mNode, OMX_IndexParamPortDefinition, &def, sizeof(def));
+
+ if (err != OK) {
+ return err;
+ }
+
+ OMX_AUDIO_PARAM_AACPROFILETYPE profile;
+ InitOMXParams(&profile);
+ profile.nPortIndex = kPortIndexOutput;
+
+ err = mOMX->getParameter(
+ mNode, OMX_IndexParamAudioAac, &profile, sizeof(profile));
+
+ if (err != OK) {
+ return err;
+ }
+
+ profile.nChannels = numChannels;
+
+ profile.eChannelMode =
+ (numChannels == 1)
+ ? OMX_AUDIO_ChannelModeMono: OMX_AUDIO_ChannelModeStereo;
+
+ profile.nSampleRate = sampleRate;
+ profile.nBitRate = bitRate;
+ profile.nAudioBandWidth = 0;
+ profile.nFrameLength = 0;
+ profile.nAACtools = OMX_AUDIO_AACToolAll;
+ profile.nAACERtools = OMX_AUDIO_AACERNone;
+ profile.eAACProfile = OMX_AUDIO_AACObjectLC;
+ profile.eAACStreamFormat = OMX_AUDIO_AACStreamFormatMP4FF;
+
+ err = mOMX->setParameter(
+ mNode, OMX_IndexParamAudioAac, &profile, sizeof(profile));
+
+ if (err != OK) {
+ return err;
+ }
+
+ return err;
+ }
+
OMX_AUDIO_PARAM_AACPROFILETYPE profile;
InitOMXParams(&profile);
profile.nPortIndex = kPortIndexInput;
- status_t err = mOMX->getParameter(
+ err = mOMX->getParameter(
mNode, OMX_IndexParamAudioAac, &profile, sizeof(profile));
if (err != OK) {
@@ -835,16 +1023,59 @@
profile.nSampleRate = sampleRate;
profile.eAACStreamFormat = OMX_AUDIO_AACStreamFormatMP4ADTS;
- err = mOMX->setParameter(
+ return mOMX->setParameter(
mNode, OMX_IndexParamAudioAac, &profile, sizeof(profile));
-
- return err;
}
-status_t ACodec::setupAMRDecoder(bool isWAMR) {
+static OMX_AUDIO_AMRBANDMODETYPE pickModeFromBitRate(
+ bool isAMRWB, int32_t bps) {
+ if (isAMRWB) {
+ if (bps <= 6600) {
+ return OMX_AUDIO_AMRBandModeWB0;
+ } else if (bps <= 8850) {
+ return OMX_AUDIO_AMRBandModeWB1;
+ } else if (bps <= 12650) {
+ return OMX_AUDIO_AMRBandModeWB2;
+ } else if (bps <= 14250) {
+ return OMX_AUDIO_AMRBandModeWB3;
+ } else if (bps <= 15850) {
+ return OMX_AUDIO_AMRBandModeWB4;
+ } else if (bps <= 18250) {
+ return OMX_AUDIO_AMRBandModeWB5;
+ } else if (bps <= 19850) {
+ return OMX_AUDIO_AMRBandModeWB6;
+ } else if (bps <= 23050) {
+ return OMX_AUDIO_AMRBandModeWB7;
+ }
+
+ // 23850 bps
+ return OMX_AUDIO_AMRBandModeWB8;
+ } else { // AMRNB
+ if (bps <= 4750) {
+ return OMX_AUDIO_AMRBandModeNB0;
+ } else if (bps <= 5150) {
+ return OMX_AUDIO_AMRBandModeNB1;
+ } else if (bps <= 5900) {
+ return OMX_AUDIO_AMRBandModeNB2;
+ } else if (bps <= 6700) {
+ return OMX_AUDIO_AMRBandModeNB3;
+ } else if (bps <= 7400) {
+ return OMX_AUDIO_AMRBandModeNB4;
+ } else if (bps <= 7950) {
+ return OMX_AUDIO_AMRBandModeNB5;
+ } else if (bps <= 10200) {
+ return OMX_AUDIO_AMRBandModeNB6;
+ }
+
+ // 12200 bps
+ return OMX_AUDIO_AMRBandModeNB7;
+ }
+}
+
+status_t ACodec::setupAMRCodec(bool encoder, bool isWAMR, int32_t bitrate) {
OMX_AUDIO_PARAM_AMRTYPE def;
InitOMXParams(&def);
- def.nPortIndex = kPortIndexInput;
+ def.nPortIndex = encoder ? kPortIndexOutput : kPortIndexInput;
status_t err =
mOMX->getParameter(mNode, OMX_IndexParamAudioAmr, &def, sizeof(def));
@@ -854,14 +1085,24 @@
}
def.eAMRFrameFormat = OMX_AUDIO_AMRFrameFormatFSF;
+ def.eAMRBandMode = pickModeFromBitRate(isWAMR, bitrate);
- def.eAMRBandMode =
- isWAMR ? OMX_AUDIO_AMRBandModeWB0 : OMX_AUDIO_AMRBandModeNB0;
+ err = mOMX->setParameter(
+ mNode, OMX_IndexParamAudioAmr, &def, sizeof(def));
- return mOMX->setParameter(mNode, OMX_IndexParamAudioAmr, &def, sizeof(def));
+ if (err != OK) {
+ return err;
+ }
+
+ return setupRawAudioFormat(
+ encoder ? kPortIndexInput : kPortIndexOutput,
+ isWAMR ? 16000 : 8000 /* sampleRate */,
+ 1 /* numChannels */);
}
-status_t ACodec::setupG711Decoder(int32_t numChannels) {
+status_t ACodec::setupG711Codec(bool encoder, int32_t numChannels) {
+ CHECK(!encoder); // XXX TODO
+
return setupRawAudioFormat(
kPortIndexInput, 8000 /* sampleRate */, numChannels);
}
@@ -1001,22 +1242,36 @@
&format, sizeof(format));
}
-status_t ACodec::setupVideoDecoder(
- const char *mime, int32_t width, int32_t height) {
- OMX_VIDEO_CODINGTYPE compressionFormat = OMX_VIDEO_CodingUnused;
+static status_t GetVideoCodingTypeFromMime(
+ const char *mime, OMX_VIDEO_CODINGTYPE *codingType) {
if (!strcasecmp(MEDIA_MIMETYPE_VIDEO_AVC, mime)) {
- compressionFormat = OMX_VIDEO_CodingAVC;
+ *codingType = OMX_VIDEO_CodingAVC;
} else if (!strcasecmp(MEDIA_MIMETYPE_VIDEO_MPEG4, mime)) {
- compressionFormat = OMX_VIDEO_CodingMPEG4;
+ *codingType = OMX_VIDEO_CodingMPEG4;
} else if (!strcasecmp(MEDIA_MIMETYPE_VIDEO_H263, mime)) {
- compressionFormat = OMX_VIDEO_CodingH263;
+ *codingType = OMX_VIDEO_CodingH263;
} else if (!strcasecmp(MEDIA_MIMETYPE_VIDEO_MPEG2, mime)) {
- compressionFormat = OMX_VIDEO_CodingMPEG2;
+ *codingType = OMX_VIDEO_CodingMPEG2;
+ } else if (!strcasecmp(MEDIA_MIMETYPE_VIDEO_VPX, mime)) {
+ *codingType = OMX_VIDEO_CodingVPX;
} else {
- TRESPASS();
+ *codingType = OMX_VIDEO_CodingUnused;
+ return ERROR_UNSUPPORTED;
}
- status_t err = setVideoPortFormatType(
+ return OK;
+}
+
+status_t ACodec::setupVideoDecoder(
+ const char *mime, int32_t width, int32_t height) {
+ OMX_VIDEO_CODINGTYPE compressionFormat;
+ status_t err = GetVideoCodingTypeFromMime(mime, &compressionFormat);
+
+ if (err != OK) {
+ return err;
+ }
+
+ err = setVideoPortFormatType(
kPortIndexInput, compressionFormat, OMX_COLOR_FormatUnused);
if (err != OK) {
@@ -1046,6 +1301,489 @@
return OK;
}
+status_t ACodec::setupVideoEncoder(const char *mime, const sp<AMessage> &msg) {
+ int32_t tmp;
+ if (!msg->findInt32("color-format", &tmp)) {
+ return INVALID_OPERATION;
+ }
+
+ OMX_COLOR_FORMATTYPE colorFormat =
+ static_cast<OMX_COLOR_FORMATTYPE>(tmp);
+
+ status_t err = setVideoPortFormatType(
+ kPortIndexInput, OMX_VIDEO_CodingUnused, colorFormat);
+
+ if (err != OK) {
+ ALOGE("[%s] does not support color format %d",
+ mComponentName.c_str(), colorFormat);
+
+ return err;
+ }
+
+ /* Input port configuration */
+
+ OMX_PARAM_PORTDEFINITIONTYPE def;
+ InitOMXParams(&def);
+
+ OMX_VIDEO_PORTDEFINITIONTYPE *video_def = &def.format.video;
+
+ def.nPortIndex = kPortIndexInput;
+
+ err = mOMX->getParameter(
+ mNode, OMX_IndexParamPortDefinition, &def, sizeof(def));
+
+ if (err != OK) {
+ return err;
+ }
+
+ int32_t width, height, bitrate;
+ if (!msg->findInt32("width", &width)
+ || !msg->findInt32("height", &height)
+ || !msg->findInt32("bitrate", &bitrate)) {
+ return INVALID_OPERATION;
+ }
+
+ video_def->nFrameWidth = width;
+ video_def->nFrameHeight = height;
+
+ int32_t stride;
+ if (!msg->findInt32("stride", &stride)) {
+ stride = width;
+ }
+
+ video_def->nStride = stride;
+
+ int32_t sliceHeight;
+ if (!msg->findInt32("slice-height", &sliceHeight)) {
+ sliceHeight = height;
+ }
+
+ video_def->nSliceHeight = sliceHeight;
+
+ def.nBufferSize = (video_def->nStride * video_def->nSliceHeight * 3) / 2;
+
+ float frameRate;
+ if (!msg->findFloat("frame-rate", &frameRate)) {
+ int32_t tmp;
+ if (!msg->findInt32("frame-rate", &tmp)) {
+ return INVALID_OPERATION;
+ }
+ frameRate = (float)tmp;
+ }
+
+ video_def->xFramerate = (OMX_U32)(frameRate * 65536.0f);
+ video_def->eCompressionFormat = OMX_VIDEO_CodingUnused;
+ video_def->eColorFormat = colorFormat;
+
+ err = mOMX->setParameter(
+ mNode, OMX_IndexParamPortDefinition, &def, sizeof(def));
+
+ if (err != OK) {
+ ALOGE("[%s] failed to set input port definition parameters.",
+ mComponentName.c_str());
+
+ return err;
+ }
+
+ /* Output port configuration */
+
+ OMX_VIDEO_CODINGTYPE compressionFormat;
+ err = GetVideoCodingTypeFromMime(mime, &compressionFormat);
+
+ if (err != OK) {
+ return err;
+ }
+
+ err = setVideoPortFormatType(
+ kPortIndexOutput, compressionFormat, OMX_COLOR_FormatUnused);
+
+ if (err != OK) {
+ ALOGE("[%s] does not support compression format %d",
+ mComponentName.c_str(), compressionFormat);
+
+ return err;
+ }
+
+ def.nPortIndex = kPortIndexOutput;
+
+ err = mOMX->getParameter(
+ mNode, OMX_IndexParamPortDefinition, &def, sizeof(def));
+
+ if (err != OK) {
+ return err;
+ }
+
+ video_def->nFrameWidth = width;
+ video_def->nFrameHeight = height;
+ video_def->xFramerate = 0;
+ video_def->nBitrate = bitrate;
+ video_def->eCompressionFormat = compressionFormat;
+ video_def->eColorFormat = OMX_COLOR_FormatUnused;
+
+ err = mOMX->setParameter(
+ mNode, OMX_IndexParamPortDefinition, &def, sizeof(def));
+
+ if (err != OK) {
+ ALOGE("[%s] failed to set output port definition parameters.",
+ mComponentName.c_str());
+
+ return err;
+ }
+
+ switch (compressionFormat) {
+ case OMX_VIDEO_CodingMPEG4:
+ err = setupMPEG4EncoderParameters(msg);
+ break;
+
+ case OMX_VIDEO_CodingH263:
+ err = setupH263EncoderParameters(msg);
+ break;
+
+ case OMX_VIDEO_CodingAVC:
+ err = setupAVCEncoderParameters(msg);
+ break;
+
+ default:
+ break;
+ }
+
+ ALOGI("setupVideoEncoder succeeded");
+
+ return err;
+}
+
+static OMX_U32 setPFramesSpacing(int32_t iFramesInterval, int32_t frameRate) {
+ if (iFramesInterval < 0) {
+ return 0xFFFFFFFF;
+ } else if (iFramesInterval == 0) {
+ return 0;
+ }
+ OMX_U32 ret = frameRate * iFramesInterval;
+ CHECK(ret > 1);
+ return ret;
+}
+
+status_t ACodec::setupMPEG4EncoderParameters(const sp<AMessage> &msg) {
+ int32_t bitrate, iFrameInterval;
+ if (!msg->findInt32("bitrate", &bitrate)
+ || !msg->findInt32("i-frame-interval", &iFrameInterval)) {
+ return INVALID_OPERATION;
+ }
+
+ float frameRate;
+ if (!msg->findFloat("frame-rate", &frameRate)) {
+ int32_t tmp;
+ if (!msg->findInt32("frame-rate", &tmp)) {
+ return INVALID_OPERATION;
+ }
+ frameRate = (float)tmp;
+ }
+
+ OMX_VIDEO_PARAM_MPEG4TYPE mpeg4type;
+ InitOMXParams(&mpeg4type);
+ mpeg4type.nPortIndex = kPortIndexOutput;
+
+ status_t err = mOMX->getParameter(
+ mNode, OMX_IndexParamVideoMpeg4, &mpeg4type, sizeof(mpeg4type));
+
+ if (err != OK) {
+ return err;
+ }
+
+ mpeg4type.nSliceHeaderSpacing = 0;
+ mpeg4type.bSVH = OMX_FALSE;
+ mpeg4type.bGov = OMX_FALSE;
+
+ mpeg4type.nAllowedPictureTypes =
+ OMX_VIDEO_PictureTypeI | OMX_VIDEO_PictureTypeP;
+
+ mpeg4type.nPFrames = setPFramesSpacing(iFrameInterval, frameRate);
+ if (mpeg4type.nPFrames == 0) {
+ mpeg4type.nAllowedPictureTypes = OMX_VIDEO_PictureTypeI;
+ }
+ mpeg4type.nBFrames = 0;
+ mpeg4type.nIDCVLCThreshold = 0;
+ mpeg4type.bACPred = OMX_TRUE;
+ mpeg4type.nMaxPacketSize = 256;
+ mpeg4type.nTimeIncRes = 1000;
+ mpeg4type.nHeaderExtension = 0;
+ mpeg4type.bReversibleVLC = OMX_FALSE;
+
+ int32_t profile;
+ if (msg->findInt32("profile", &profile)) {
+ int32_t level;
+ if (!msg->findInt32("level", &level)) {
+ return INVALID_OPERATION;
+ }
+
+ err = verifySupportForProfileAndLevel(profile, level);
+
+ if (err != OK) {
+ return err;
+ }
+
+ mpeg4type.eProfile = static_cast<OMX_VIDEO_MPEG4PROFILETYPE>(profile);
+ mpeg4type.eLevel = static_cast<OMX_VIDEO_MPEG4LEVELTYPE>(level);
+ }
+
+ err = mOMX->setParameter(
+ mNode, OMX_IndexParamVideoMpeg4, &mpeg4type, sizeof(mpeg4type));
+
+ if (err != OK) {
+ return err;
+ }
+
+ err = configureBitrate(bitrate);
+
+ if (err != OK) {
+ return err;
+ }
+
+ return setupErrorCorrectionParameters();
+}
+
+status_t ACodec::setupH263EncoderParameters(const sp<AMessage> &msg) {
+ int32_t bitrate, iFrameInterval;
+ if (!msg->findInt32("bitrate", &bitrate)
+ || !msg->findInt32("i-frame-interval", &iFrameInterval)) {
+ return INVALID_OPERATION;
+ }
+
+ float frameRate;
+ if (!msg->findFloat("frame-rate", &frameRate)) {
+ int32_t tmp;
+ if (!msg->findInt32("frame-rate", &tmp)) {
+ return INVALID_OPERATION;
+ }
+ frameRate = (float)tmp;
+ }
+
+ OMX_VIDEO_PARAM_H263TYPE h263type;
+ InitOMXParams(&h263type);
+ h263type.nPortIndex = kPortIndexOutput;
+
+ status_t err = mOMX->getParameter(
+ mNode, OMX_IndexParamVideoH263, &h263type, sizeof(h263type));
+
+ if (err != OK) {
+ return err;
+ }
+
+ h263type.nAllowedPictureTypes =
+ OMX_VIDEO_PictureTypeI | OMX_VIDEO_PictureTypeP;
+
+ h263type.nPFrames = setPFramesSpacing(iFrameInterval, frameRate);
+ if (h263type.nPFrames == 0) {
+ h263type.nAllowedPictureTypes = OMX_VIDEO_PictureTypeI;
+ }
+ h263type.nBFrames = 0;
+
+ int32_t profile;
+ if (msg->findInt32("profile", &profile)) {
+ int32_t level;
+ if (!msg->findInt32("level", &level)) {
+ return INVALID_OPERATION;
+ }
+
+ err = verifySupportForProfileAndLevel(profile, level);
+
+ if (err != OK) {
+ return err;
+ }
+
+ h263type.eProfile = static_cast<OMX_VIDEO_H263PROFILETYPE>(profile);
+ h263type.eLevel = static_cast<OMX_VIDEO_H263LEVELTYPE>(level);
+ }
+
+ h263type.bPLUSPTYPEAllowed = OMX_FALSE;
+ h263type.bForceRoundingTypeToZero = OMX_FALSE;
+ h263type.nPictureHeaderRepetition = 0;
+ h263type.nGOBHeaderInterval = 0;
+
+ err = mOMX->setParameter(
+ mNode, OMX_IndexParamVideoH263, &h263type, sizeof(h263type));
+
+ if (err != OK) {
+ return err;
+ }
+
+ err = configureBitrate(bitrate);
+
+ if (err != OK) {
+ return err;
+ }
+
+ return setupErrorCorrectionParameters();
+}
+
+status_t ACodec::setupAVCEncoderParameters(const sp<AMessage> &msg) {
+ int32_t bitrate, iFrameInterval;
+ if (!msg->findInt32("bitrate", &bitrate)
+ || !msg->findInt32("i-frame-interval", &iFrameInterval)) {
+ return INVALID_OPERATION;
+ }
+
+ float frameRate;
+ if (!msg->findFloat("frame-rate", &frameRate)) {
+ int32_t tmp;
+ if (!msg->findInt32("frame-rate", &tmp)) {
+ return INVALID_OPERATION;
+ }
+ frameRate = (float)tmp;
+ }
+
+ OMX_VIDEO_PARAM_AVCTYPE h264type;
+ InitOMXParams(&h264type);
+ h264type.nPortIndex = kPortIndexOutput;
+
+ status_t err = mOMX->getParameter(
+ mNode, OMX_IndexParamVideoAvc, &h264type, sizeof(h264type));
+
+ if (err != OK) {
+ return err;
+ }
+
+ h264type.nAllowedPictureTypes =
+ OMX_VIDEO_PictureTypeI | OMX_VIDEO_PictureTypeP;
+
+ int32_t profile;
+ if (msg->findInt32("profile", &profile)) {
+ int32_t level;
+ if (!msg->findInt32("level", &level)) {
+ return INVALID_OPERATION;
+ }
+
+ err = verifySupportForProfileAndLevel(profile, level);
+
+ if (err != OK) {
+ return err;
+ }
+
+ h264type.eProfile = static_cast<OMX_VIDEO_AVCPROFILETYPE>(profile);
+ h264type.eLevel = static_cast<OMX_VIDEO_AVCLEVELTYPE>(level);
+ }
+
+ // XXX
+ if (!strncmp(mComponentName.c_str(), "OMX.TI.DUCATI1", 14)) {
+ h264type.eProfile = OMX_VIDEO_AVCProfileBaseline;
+ }
+
+ if (h264type.eProfile == OMX_VIDEO_AVCProfileBaseline) {
+ h264type.nSliceHeaderSpacing = 0;
+ h264type.bUseHadamard = OMX_TRUE;
+ h264type.nRefFrames = 1;
+ h264type.nBFrames = 0;
+ h264type.nPFrames = setPFramesSpacing(iFrameInterval, frameRate);
+ if (h264type.nPFrames == 0) {
+ h264type.nAllowedPictureTypes = OMX_VIDEO_PictureTypeI;
+ }
+ h264type.nRefIdx10ActiveMinus1 = 0;
+ h264type.nRefIdx11ActiveMinus1 = 0;
+ h264type.bEntropyCodingCABAC = OMX_FALSE;
+ h264type.bWeightedPPrediction = OMX_FALSE;
+ h264type.bconstIpred = OMX_FALSE;
+ h264type.bDirect8x8Inference = OMX_FALSE;
+ h264type.bDirectSpatialTemporal = OMX_FALSE;
+ h264type.nCabacInitIdc = 0;
+ }
+
+ if (h264type.nBFrames != 0) {
+ h264type.nAllowedPictureTypes |= OMX_VIDEO_PictureTypeB;
+ }
+
+ h264type.bEnableUEP = OMX_FALSE;
+ h264type.bEnableFMO = OMX_FALSE;
+ h264type.bEnableASO = OMX_FALSE;
+ h264type.bEnableRS = OMX_FALSE;
+ h264type.bFrameMBsOnly = OMX_TRUE;
+ h264type.bMBAFF = OMX_FALSE;
+ h264type.eLoopFilterMode = OMX_VIDEO_AVCLoopFilterEnable;
+
+ if (!strcasecmp("OMX.Nvidia.h264.encoder", mComponentName.c_str())) {
+ h264type.eLevel = OMX_VIDEO_AVCLevelMax;
+ }
+
+ err = mOMX->setParameter(
+ mNode, OMX_IndexParamVideoAvc, &h264type, sizeof(h264type));
+
+ if (err != OK) {
+ return err;
+ }
+
+ return configureBitrate(bitrate);
+}
+
+status_t ACodec::verifySupportForProfileAndLevel(
+ int32_t profile, int32_t level) {
+ OMX_VIDEO_PARAM_PROFILELEVELTYPE params;
+ InitOMXParams(¶ms);
+ params.nPortIndex = kPortIndexOutput;
+
+ for (params.nProfileIndex = 0;; ++params.nProfileIndex) {
+ status_t err = mOMX->getParameter(
+ mNode,
+ OMX_IndexParamVideoProfileLevelQuerySupported,
+ ¶ms,
+ sizeof(params));
+
+ if (err != OK) {
+ return err;
+ }
+
+ int32_t supportedProfile = static_cast<int32_t>(params.eProfile);
+ int32_t supportedLevel = static_cast<int32_t>(params.eLevel);
+
+ if (profile == supportedProfile && level <= supportedLevel) {
+ return OK;
+ }
+ }
+}
+
+status_t ACodec::configureBitrate(int32_t bitrate) {
+ OMX_VIDEO_PARAM_BITRATETYPE bitrateType;
+ InitOMXParams(&bitrateType);
+ bitrateType.nPortIndex = kPortIndexOutput;
+
+ status_t err = mOMX->getParameter(
+ mNode, OMX_IndexParamVideoBitrate,
+ &bitrateType, sizeof(bitrateType));
+
+ if (err != OK) {
+ return err;
+ }
+
+ bitrateType.eControlRate = OMX_Video_ControlRateVariable;
+ bitrateType.nTargetBitrate = bitrate;
+
+ return mOMX->setParameter(
+ mNode, OMX_IndexParamVideoBitrate,
+ &bitrateType, sizeof(bitrateType));
+}
+
+status_t ACodec::setupErrorCorrectionParameters() {
+ OMX_VIDEO_PARAM_ERRORCORRECTIONTYPE errorCorrectionType;
+ InitOMXParams(&errorCorrectionType);
+ errorCorrectionType.nPortIndex = kPortIndexOutput;
+
+ status_t err = mOMX->getParameter(
+ mNode, OMX_IndexParamVideoErrorCorrection,
+ &errorCorrectionType, sizeof(errorCorrectionType));
+
+ if (err != OK) {
+ return OK; // Optional feature. Ignore this failure
+ }
+
+ errorCorrectionType.bEnableHEC = OMX_FALSE;
+ errorCorrectionType.bEnableResync = OMX_TRUE;
+ errorCorrectionType.nResynchMarkerSpacing = 256;
+ errorCorrectionType.bEnableDataPartitioning = OMX_FALSE;
+ errorCorrectionType.bEnableRVLC = OMX_FALSE;
+
+ return mOMX->setParameter(
+ mNode, OMX_IndexParamVideoErrorCorrection,
+ &errorCorrectionType, sizeof(errorCorrectionType));
+}
+
status_t ACodec::setVideoFormatOnPort(
OMX_U32 portIndex,
int32_t width, int32_t height, OMX_VIDEO_CODINGTYPE compressionFormat) {
@@ -1166,6 +1904,9 @@
notify->setString("mime", MEDIA_MIMETYPE_VIDEO_RAW);
notify->setInt32("width", videoDef->nFrameWidth);
notify->setInt32("height", videoDef->nFrameHeight);
+ notify->setInt32("stride", videoDef->nStride);
+ notify->setInt32("slice-height", videoDef->nSliceHeight);
+ notify->setInt32("color-format", videoDef->eColorFormat);
OMX_CONFIG_RECTTYPE rect;
InitOMXParams(&rect);
@@ -1241,10 +1982,11 @@
mSentFormat = true;
}
-void ACodec::signalError(OMX_ERRORTYPE error) {
+void ACodec::signalError(OMX_ERRORTYPE error, status_t internalError) {
sp<AMessage> notify = mNotify->dup();
notify->setInt32("what", ACodec::kWhatError);
notify->setInt32("omx-error", error);
+ notify->setInt32("err", internalError);
notify->post();
}
@@ -1417,7 +2159,7 @@
notify->setPointer("buffer-id", info->mBufferID);
info->mData->meta()->clear();
- notify->setObject("buffer", info->mData);
+ notify->setBuffer("buffer", info->mData);
sp<AMessage> reply = new AMessage(kWhatInputBufferFilled, mCodec->id());
reply->setPointer("buffer-id", info->mBufferID);
@@ -1433,18 +2175,26 @@
IOMX::buffer_id bufferID;
CHECK(msg->findPointer("buffer-id", &bufferID));
- sp<RefBase> obj;
+ sp<ABuffer> buffer;
int32_t err = OK;
- if (!msg->findObject("buffer", &obj)) {
+ bool eos = false;
+
+ if (!msg->findBuffer("buffer", &buffer)) {
CHECK(msg->findInt32("err", &err));
ALOGV("[%s] saw error %d instead of an input buffer",
mCodec->mComponentName.c_str(), err);
- obj.clear();
+ buffer.clear();
+
+ eos = true;
}
- sp<ABuffer> buffer = static_cast<ABuffer *>(obj.get());
+ int32_t tmp;
+ if (buffer != NULL && buffer->meta()->findInt32("eos", &tmp) && tmp) {
+ eos = true;
+ err = ERROR_END_OF_STREAM;
+ }
BufferInfo *info = mCodec->findBufferByID(kPortIndexInput, bufferID);
CHECK_EQ((int)info->mStatus, (int)BufferInfo::OWNED_BY_UPSTREAM);
@@ -1456,7 +2206,7 @@
switch (mode) {
case KEEP_BUFFERS:
{
- if (buffer == NULL) {
+ if (eos) {
if (!mCodec->mPortEOS[kPortIndexInput]) {
mCodec->mPortEOS[kPortIndexInput] = true;
mCodec->mInputEOSResult = err;
@@ -1467,9 +2217,7 @@
case RESUBMIT_BUFFERS:
{
- if (buffer != NULL) {
- CHECK(!mCodec->mPortEOS[kPortIndexInput]);
-
+ if (buffer != NULL && !mCodec->mPortEOS[kPortIndexInput]) {
int64_t timeUs;
CHECK(buffer->meta()->findInt64("timeUs", &timeUs));
@@ -1480,6 +2228,10 @@
flags |= OMX_BUFFERFLAG_CODECCONFIG;
}
+ if (eos) {
+ flags |= OMX_BUFFERFLAG_EOS;
+ }
+
if (buffer != info->mData) {
if (0 && !(flags & OMX_BUFFERFLAG_CODECCONFIG)) {
ALOGV("[%s] Needs to copy input data.",
@@ -1493,6 +2245,9 @@
if (flags & OMX_BUFFERFLAG_CODECCONFIG) {
ALOGV("[%s] calling emptyBuffer %p w/ codec specific data",
mCodec->mComponentName.c_str(), bufferID);
+ } else if (flags & OMX_BUFFERFLAG_EOS) {
+ ALOGV("[%s] calling emptyBuffer %p w/ EOS",
+ mCodec->mComponentName.c_str(), bufferID);
} else {
ALOGV("[%s] calling emptyBuffer %p w/ time %lld us",
mCodec->mComponentName.c_str(), bufferID, timeUs);
@@ -1509,7 +2264,15 @@
info->mStatus = BufferInfo::OWNED_BY_COMPONENT;
- getMoreInputDataIfPossible();
+ if (!eos) {
+ getMoreInputDataIfPossible();
+ } else {
+ ALOGV("[%s] Signalled EOS on the input port",
+ mCodec->mComponentName.c_str());
+
+ mCodec->mPortEOS[kPortIndexInput] = true;
+ mCodec->mInputEOSResult = err;
+ }
} else if (!mCodec->mPortEOS[kPortIndexInput]) {
if (err != ERROR_END_OF_STREAM) {
ALOGV("[%s] Signalling EOS on the input port "
@@ -1582,8 +2345,8 @@
int64_t timeUs,
void *platformPrivate,
void *dataPtr) {
- ALOGV("[%s] onOMXFillBufferDone %p time %lld us",
- mCodec->mComponentName.c_str(), bufferID, timeUs);
+ ALOGV("[%s] onOMXFillBufferDone %p time %lld us, flags = 0x%08lx",
+ mCodec->mComponentName.c_str(), bufferID, timeUs, flags);
ssize_t index;
BufferInfo *info =
@@ -1601,46 +2364,48 @@
case RESUBMIT_BUFFERS:
{
- if (rangeLength == 0) {
- if (!(flags & OMX_BUFFERFLAG_EOS)) {
- ALOGV("[%s] calling fillBuffer %p",
- mCodec->mComponentName.c_str(), info->mBufferID);
+ if (rangeLength == 0 && !(flags & OMX_BUFFERFLAG_EOS)) {
+ ALOGV("[%s] calling fillBuffer %p",
+ mCodec->mComponentName.c_str(), info->mBufferID);
- CHECK_EQ(mCodec->mOMX->fillBuffer(
- mCodec->mNode, info->mBufferID),
- (status_t)OK);
+ CHECK_EQ(mCodec->mOMX->fillBuffer(
+ mCodec->mNode, info->mBufferID),
+ (status_t)OK);
- info->mStatus = BufferInfo::OWNED_BY_COMPONENT;
- }
- } else {
- if (!mCodec->mSentFormat) {
- mCodec->sendFormatChange();
- }
-
- if (mCodec->mNativeWindow == NULL) {
- info->mData->setRange(rangeOffset, rangeLength);
- }
-
- info->mData->meta()->setInt64("timeUs", timeUs);
-
- sp<AMessage> notify = mCodec->mNotify->dup();
- notify->setInt32("what", ACodec::kWhatDrainThisBuffer);
- notify->setPointer("buffer-id", info->mBufferID);
- notify->setObject("buffer", info->mData);
-
- sp<AMessage> reply =
- new AMessage(kWhatOutputBufferDrained, mCodec->id());
-
- reply->setPointer("buffer-id", info->mBufferID);
-
- notify->setMessage("reply", reply);
-
- notify->post();
-
- info->mStatus = BufferInfo::OWNED_BY_DOWNSTREAM;
+ info->mStatus = BufferInfo::OWNED_BY_COMPONENT;
+ break;
}
+ if (!mCodec->mIsEncoder && !mCodec->mSentFormat) {
+ mCodec->sendFormatChange();
+ }
+
+ if (mCodec->mNativeWindow == NULL) {
+ info->mData->setRange(rangeOffset, rangeLength);
+ }
+
+ info->mData->meta()->setInt64("timeUs", timeUs);
+
+ sp<AMessage> notify = mCodec->mNotify->dup();
+ notify->setInt32("what", ACodec::kWhatDrainThisBuffer);
+ notify->setPointer("buffer-id", info->mBufferID);
+ notify->setBuffer("buffer", info->mData);
+ notify->setInt32("flags", flags);
+
+ sp<AMessage> reply =
+ new AMessage(kWhatOutputBufferDrained, mCodec->id());
+
+ reply->setPointer("buffer-id", info->mBufferID);
+
+ notify->setMessage("reply", reply);
+
+ notify->post();
+
+ info->mStatus = BufferInfo::OWNED_BY_DOWNSTREAM;
+
if (flags & OMX_BUFFERFLAG_EOS) {
+ ALOGV("[%s] saw output EOS", mCodec->mComponentName.c_str());
+
sp<AMessage> notify = mCodec->mNotify->dup();
notify->setInt32("what", ACodec::kWhatEOS);
notify->setInt32("err", mCodec->mInputEOSResult);
@@ -1678,12 +2443,13 @@
&& msg->findInt32("render", &render) && render != 0) {
// The client wants this buffer to be rendered.
- if (mCodec->mNativeWindow->queueBuffer(
+ status_t err;
+ if ((err = mCodec->mNativeWindow->queueBuffer(
mCodec->mNativeWindow.get(),
- info->mGraphicBuffer.get()) == OK) {
+ info->mGraphicBuffer.get())) == OK) {
info->mStatus = BufferInfo::OWNED_BY_NATIVE_WINDOW;
} else {
- mCodec->signalError();
+ mCodec->signalError(OMX_ErrorUndefined, err);
info->mStatus = BufferInfo::OWNED_BY_US;
}
} else {
@@ -1746,6 +2512,10 @@
: BaseState(codec) {
}
+void ACodec::UninitializedState::stateEntered() {
+ ALOGV("Now uninitialized");
+}
+
bool ACodec::UninitializedState::onMessageReceived(const sp<AMessage> &msg) {
bool handled = false;
@@ -1758,8 +2528,20 @@
break;
}
+ case ACodec::kWhatAllocateComponent:
+ {
+ onAllocateComponent(msg);
+ handled = true;
+ break;
+ }
+
case ACodec::kWhatShutdown:
{
+ int32_t keepComponentAllocated;
+ CHECK(msg->findInt32(
+ "keepComponentAllocated", &keepComponentAllocated));
+ CHECK(!keepComponentAllocated);
+
sp<AMessage> notify = mCodec->mNotify->dup();
notify->setInt32("what", ACodec::kWhatShutdownCompleted);
notify->post();
@@ -1787,30 +2569,60 @@
void ACodec::UninitializedState::onSetup(
const sp<AMessage> &msg) {
+ if (onAllocateComponent(msg)
+ && mCodec->mLoadedState->onConfigureComponent(msg)) {
+ mCodec->mLoadedState->onStart();
+ }
+}
+
+bool ACodec::UninitializedState::onAllocateComponent(const sp<AMessage> &msg) {
+ ALOGV("onAllocateComponent");
+
+ CHECK(mCodec->mNode == NULL);
+
OMXClient client;
CHECK_EQ(client.connect(), (status_t)OK);
sp<IOMX> omx = client.interface();
- AString mime;
- CHECK(msg->findString("mime", &mime));
-
Vector<String8> matchingCodecs;
- OMXCodec::findMatchingCodecs(
- mime.c_str(),
- false, // createEncoder
- NULL, // matchComponentName
- 0, // flags
- &matchingCodecs);
+ Vector<uint32_t> matchingCodecQuirks;
+
+ AString mime;
+
+ AString componentName;
+ uint32_t quirks;
+ if (msg->findString("componentName", &componentName)) {
+ matchingCodecs.push_back(String8(componentName.c_str()));
+
+ if (!OMXCodec::findCodecQuirks(componentName.c_str(), &quirks)) {
+ quirks = 0;
+ }
+ matchingCodecQuirks.push_back(quirks);
+ } else {
+ CHECK(msg->findString("mime", &mime));
+
+ int32_t encoder;
+ if (!msg->findInt32("encoder", &encoder)) {
+ encoder = false;
+ }
+
+ OMXCodec::findMatchingCodecs(
+ mime.c_str(),
+ encoder, // createEncoder
+ NULL, // matchComponentName
+ 0, // flags
+ &matchingCodecs,
+ &matchingCodecQuirks);
+ }
sp<CodecObserver> observer = new CodecObserver;
IOMX::node_id node = NULL;
- AString componentName;
-
for (size_t matchIndex = 0; matchIndex < matchingCodecs.size();
++matchIndex) {
componentName = matchingCodecs.itemAt(matchIndex).string();
+ quirks = matchingCodecQuirks.itemAt(matchIndex);
pid_t tid = androidGetTid();
int prevPriority = androidGetThreadPriority(tid);
@@ -1826,16 +2638,22 @@
}
if (node == NULL) {
- ALOGE("Unable to instantiate a decoder for type '%s'.", mime.c_str());
+ if (!mime.empty()) {
+ ALOGE("Unable to instantiate a decoder for type '%s'.",
+ mime.c_str());
+ } else {
+ ALOGE("Unable to instantiate decoder '%s'.", componentName.c_str());
+ }
mCodec->signalError(OMX_ErrorComponentNotFound);
- return;
+ return false;
}
sp<AMessage> notify = new AMessage(kWhatOMXMessage, mCodec->id());
observer->setNotificationMessage(notify);
mCodec->mComponentName = componentName;
+ mCodec->mQuirks = quirks;
mCodec->mOMX = omx;
mCodec->mNode = node;
@@ -1844,20 +2662,145 @@
mCodec->mInputEOSResult = OK;
- mCodec->configureCodec(mime.c_str(), msg);
+ {
+ sp<AMessage> notify = mCodec->mNotify->dup();
+ notify->setInt32("what", ACodec::kWhatComponentAllocated);
+ notify->setString("componentName", mCodec->mComponentName.c_str());
+ notify->post();
+ }
+
+ mCodec->changeState(mCodec->mLoadedState);
+
+ return true;
+}
+
+////////////////////////////////////////////////////////////////////////////////
+
+ACodec::LoadedState::LoadedState(ACodec *codec)
+ : BaseState(codec) {
+}
+
+void ACodec::LoadedState::stateEntered() {
+ ALOGV("[%s] Now Loaded", mCodec->mComponentName.c_str());
+
+ if (mCodec->mShutdownInProgress) {
+ bool keepComponentAllocated = mCodec->mKeepComponentAllocated;
+
+ mCodec->mShutdownInProgress = false;
+ mCodec->mKeepComponentAllocated = false;
+
+ onShutdown(keepComponentAllocated);
+ }
+}
+
+void ACodec::LoadedState::onShutdown(bool keepComponentAllocated) {
+ if (!keepComponentAllocated) {
+ CHECK_EQ(mCodec->mOMX->freeNode(mCodec->mNode), (status_t)OK);
+
+ mCodec->mNativeWindow.clear();
+ mCodec->mNode = NULL;
+ mCodec->mOMX.clear();
+ mCodec->mQuirks = 0;
+ mCodec->mComponentName.clear();
+
+ mCodec->changeState(mCodec->mUninitializedState);
+ }
+
+ sp<AMessage> notify = mCodec->mNotify->dup();
+ notify->setInt32("what", ACodec::kWhatShutdownCompleted);
+ notify->post();
+}
+
+bool ACodec::LoadedState::onMessageReceived(const sp<AMessage> &msg) {
+ bool handled = false;
+
+ switch (msg->what()) {
+ case ACodec::kWhatConfigureComponent:
+ {
+ onConfigureComponent(msg);
+ handled = true;
+ break;
+ }
+
+ case ACodec::kWhatStart:
+ {
+ onStart();
+ handled = true;
+ break;
+ }
+
+ case ACodec::kWhatShutdown:
+ {
+ int32_t keepComponentAllocated;
+ CHECK(msg->findInt32(
+ "keepComponentAllocated", &keepComponentAllocated));
+
+ onShutdown(keepComponentAllocated);
+
+ handled = true;
+ break;
+ }
+
+ case ACodec::kWhatFlush:
+ {
+ sp<AMessage> notify = mCodec->mNotify->dup();
+ notify->setInt32("what", ACodec::kWhatFlushCompleted);
+ notify->post();
+
+ handled = true;
+ break;
+ }
+
+ default:
+ return BaseState::onMessageReceived(msg);
+ }
+
+ return handled;
+}
+
+bool ACodec::LoadedState::onConfigureComponent(
+ const sp<AMessage> &msg) {
+ ALOGV("onConfigureComponent");
+
+ CHECK(mCodec->mNode != NULL);
+
+ AString mime;
+ CHECK(msg->findString("mime", &mime));
+
+ status_t err = mCodec->configureCodec(mime.c_str(), msg);
+
+ if (err != OK) {
+ ALOGE("[%s] configureCodec returning error %d",
+ mCodec->mComponentName.c_str(), err);
+
+ mCodec->signalError(OMX_ErrorUndefined, err);
+ return false;
+ }
sp<RefBase> obj;
if (msg->findObject("native-window", &obj)
- && strncmp("OMX.google.", componentName.c_str(), 11)) {
+ && strncmp("OMX.google.", mCodec->mComponentName.c_str(), 11)) {
sp<NativeWindowWrapper> nativeWindow(
static_cast<NativeWindowWrapper *>(obj.get()));
CHECK(nativeWindow != NULL);
mCodec->mNativeWindow = nativeWindow->getNativeWindow();
}
-
CHECK_EQ((status_t)OK, mCodec->initNativeWindow());
- CHECK_EQ(omx->sendCommand(node, OMX_CommandStateSet, OMX_StateIdle),
+ {
+ sp<AMessage> notify = mCodec->mNotify->dup();
+ notify->setInt32("what", ACodec::kWhatComponentConfigured);
+ notify->post();
+ }
+
+ return true;
+}
+
+void ACodec::LoadedState::onStart() {
+ ALOGV("onStart");
+
+ CHECK_EQ(mCodec->mOMX->sendCommand(
+ mCodec->mNode, OMX_CommandStateSet, OMX_StateIdle),
(status_t)OK);
mCodec->changeState(mCodec->mLoadedToIdleState);
@@ -1878,7 +2821,7 @@
"(error 0x%08x)",
err);
- mCodec->signalError();
+ mCodec->signalError(OMX_ErrorUndefined, err);
}
}
@@ -2042,6 +2985,13 @@
switch (msg->what()) {
case kWhatShutdown:
{
+ int32_t keepComponentAllocated;
+ CHECK(msg->findInt32(
+ "keepComponentAllocated", &keepComponentAllocated));
+
+ mCodec->mShutdownInProgress = true;
+ mCodec->mKeepComponentAllocated = keepComponentAllocated;
+
mActive = false;
CHECK_EQ(mCodec->mOMX->sendCommand(
@@ -2202,7 +3152,7 @@
"port reconfiguration (error 0x%08x)",
err);
- mCodec->signalError();
+ mCodec->signalError(OMX_ErrorUndefined, err);
// This is technically not correct, since we were unable
// to allocate output buffers and therefore the output port
@@ -2240,7 +3190,8 @@
////////////////////////////////////////////////////////////////////////////////
ACodec::ExecutingToIdleState::ExecutingToIdleState(ACodec *codec)
- : BaseState(codec) {
+ : BaseState(codec),
+ mComponentNowIdle(false) {
}
bool ACodec::ExecutingToIdleState::onMessageReceived(const sp<AMessage> &msg) {
@@ -2274,6 +3225,7 @@
void ACodec::ExecutingToIdleState::stateEntered() {
ALOGV("[%s] Now Executing->Idle", mCodec->mComponentName.c_str());
+ mComponentNowIdle = false;
mCodec->mSentFormat = false;
}
@@ -2285,6 +3237,8 @@
CHECK_EQ(data1, (OMX_U32)OMX_CommandStateSet);
CHECK_EQ(data2, (OMX_U32)OMX_StateIdle);
+ mComponentNowIdle = true;
+
changeStateIfWeOwnAllBuffers();
return true;
@@ -2303,7 +3257,7 @@
}
void ACodec::ExecutingToIdleState::changeStateIfWeOwnAllBuffers() {
- if (mCodec->allYourBuffersAreBelongToUs()) {
+ if (mComponentNowIdle && mCodec->allYourBuffersAreBelongToUs()) {
CHECK_EQ(mCodec->mOMX->sendCommand(
mCodec->mNode, OMX_CommandStateSet, OMX_StateLoaded),
(status_t)OK);
@@ -2375,20 +3329,7 @@
CHECK_EQ(data1, (OMX_U32)OMX_CommandStateSet);
CHECK_EQ(data2, (OMX_U32)OMX_StateLoaded);
- ALOGV("[%s] Now Loaded", mCodec->mComponentName.c_str());
-
- CHECK_EQ(mCodec->mOMX->freeNode(mCodec->mNode), (status_t)OK);
-
- mCodec->mNativeWindow.clear();
- mCodec->mNode = NULL;
- mCodec->mOMX.clear();
- mCodec->mComponentName.clear();
-
- mCodec->changeState(mCodec->mUninitializedState);
-
- sp<AMessage> notify = mCodec->mNotify->dup();
- notify->setInt32("what", ACodec::kWhatShutdownCompleted);
- notify->post();
+ mCodec->changeState(mCodec->mLoadedState);
return true;
}
diff --git a/media/libstagefright/AMRExtractor.cpp b/media/libstagefright/AMRExtractor.cpp
index 5a28347..03dcbf9 100644
--- a/media/libstagefright/AMRExtractor.cpp
+++ b/media/libstagefright/AMRExtractor.cpp
@@ -20,9 +20,9 @@
#include "include/AMRExtractor.h"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaSource.h>
diff --git a/media/libstagefright/AMRWriter.cpp b/media/libstagefright/AMRWriter.cpp
index 6c4e307..ca85640 100644
--- a/media/libstagefright/AMRWriter.cpp
+++ b/media/libstagefright/AMRWriter.cpp
@@ -14,9 +14,9 @@
* limitations under the License.
*/
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/AMRWriter.h>
#include <media/stagefright/MediaBuffer.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaSource.h>
@@ -52,7 +52,7 @@
AMRWriter::~AMRWriter() {
if (mStarted) {
- stop();
+ reset();
}
if (mFd != -1) {
@@ -152,7 +152,7 @@
return OK;
}
-status_t AMRWriter::stop() {
+status_t AMRWriter::reset() {
if (!mStarted) {
return OK;
}
diff --git a/media/libstagefright/AVIExtractor.cpp b/media/libstagefright/AVIExtractor.cpp
index a3187b7..5a6211e 100644
--- a/media/libstagefright/AVIExtractor.cpp
+++ b/media/libstagefright/AVIExtractor.cpp
@@ -577,6 +577,7 @@
case FOURCC('a', 'v', 'c', '1'):
case FOURCC('d', 'a', 'v', 'c'):
case FOURCC('x', '2', '6', '4'):
+ case FOURCC('H', '2', '6', '4'):
case FOURCC('v', 's', 's', 'h'):
return MEDIA_MIMETYPE_VIDEO_AVC;
diff --git a/media/libstagefright/Android.mk b/media/libstagefright/Android.mk
index 690deac..77714f3 100644
--- a/media/libstagefright/Android.mk
+++ b/media/libstagefright/Android.mk
@@ -9,6 +9,7 @@
AACWriter.cpp \
AMRExtractor.cpp \
AMRWriter.cpp \
+ AVIExtractor.cpp \
AudioPlayer.cpp \
AudioSource.cpp \
AwesomePlayer.cpp \
@@ -28,12 +29,14 @@
MPEG4Writer.cpp \
MediaBuffer.cpp \
MediaBufferGroup.cpp \
+ MediaCodec.cpp \
+ MediaCodecList.cpp \
MediaDefs.cpp \
MediaExtractor.cpp \
MediaSource.cpp \
- MediaSourceSplitter.cpp \
MetaData.cpp \
NuCachedSource2.cpp \
+ NuMediaExtractor.cpp \
OMXClient.cpp \
OMXCodec.cpp \
OggExtractor.cpp \
@@ -54,31 +57,41 @@
LOCAL_C_INCLUDES:= \
$(JNI_H_INCLUDE) \
- $(TOP)/frameworks/base/include/media/stagefright/openmax \
+ $(TOP)/frameworks/base/include/media/stagefright/timedtext \
+ $(TOP)/frameworks/native/include/media/hardware \
+ $(TOP)/frameworks/native/include/media/openmax \
+ $(TOP)/external/expat/lib \
$(TOP)/external/flac/include \
$(TOP)/external/tremolo \
$(TOP)/external/openssl/include \
LOCAL_SHARED_LIBRARIES := \
- libbinder \
- libmedia \
- libutils \
- libcutils \
- libui \
- libsonivox \
- libvorbisidec \
- libstagefright_yuv \
+ libbinder \
libcamera_client \
- libdrmframework \
- libcrypto \
- libssl \
- libgui \
+ libchromium_net \
+ libcrypto \
+ libcutils \
+ libdl \
+ libdrmframework \
+ libexpat \
+ libgui \
+ libicui18n \
+ libicuuc \
+ liblog \
+ libmedia \
+ libmedia_native \
+ libsonivox \
+ libssl \
+ libstagefright_omx \
+ libstagefright_yuv \
+ libui \
+ libutils \
+ libvorbisidec \
+ libz \
LOCAL_STATIC_LIBRARIES := \
libstagefright_color_conversion \
libstagefright_aacenc \
- libstagefright_amrnbenc \
- libstagefright_amrwbenc \
libstagefright_avcenc \
libstagefright_m4vh263enc \
libstagefright_matroska \
@@ -88,59 +101,15 @@
libstagefright_httplive \
libstagefright_id3 \
libFLAC \
+ libstagefright_chromium_http \
-################################################################################
-
-# The following was shamelessly copied from external/webkit/Android.mk and
-# currently must follow the same logic to determine how webkit was built and
-# if it's safe to link against libchromium.net
-
-# V8 also requires an ARMv7 CPU, and since we must use jsc, we cannot
-# use the Chrome http stack either.
-ifneq ($(strip $(ARCH_ARM_HAVE_ARMV7A)),true)
- USE_ALT_HTTP := true
-endif
-
-# See if the user has specified a stack they want to use
-HTTP_STACK = $(HTTP)
-# We default to the Chrome HTTP stack.
-DEFAULT_HTTP = chrome
-ALT_HTTP = android
-
-ifneq ($(HTTP_STACK),chrome)
- ifneq ($(HTTP_STACK),android)
- # No HTTP stack is specified, pickup the one we want as default.
- ifeq ($(USE_ALT_HTTP),true)
- HTTP_STACK = $(ALT_HTTP)
- else
- HTTP_STACK = $(DEFAULT_HTTP)
- endif
- endif
-endif
-
-ifeq ($(HTTP_STACK),chrome)
-
-LOCAL_SHARED_LIBRARIES += \
- liblog \
- libicuuc \
- libicui18n \
- libz \
- libdl \
-
-LOCAL_STATIC_LIBRARIES += \
- libstagefright_chromium_http
-
-LOCAL_SHARED_LIBRARIES += libstlport libchromium_net
+LOCAL_SHARED_LIBRARIES += libstlport
include external/stlport/libstlport.mk
+# TODO: Chromium is always available, so this flag can be removed.
LOCAL_CPPFLAGS += -DCHROMIUM_AVAILABLE=1
-endif # ifeq ($(HTTP_STACK),chrome)
-
-################################################################################
-
LOCAL_SHARED_LIBRARIES += \
- libstagefright_amrnb_common \
libstagefright_enc_common \
libstagefright_avc_common \
libstagefright_foundation \
diff --git a/media/libstagefright/AudioPlayer.cpp b/media/libstagefright/AudioPlayer.cpp
index 13c624d..0f816e7 100644
--- a/media/libstagefright/AudioPlayer.cpp
+++ b/media/libstagefright/AudioPlayer.cpp
@@ -110,13 +110,18 @@
success = format->findInt32(kKeySampleRate, &mSampleRate);
CHECK(success);
- int32_t numChannels;
+ int32_t numChannels, channelMask;
success = format->findInt32(kKeyChannelCount, &numChannels);
CHECK(success);
+ if(!format->findInt32(kKeyChannelMask, &channelMask)) {
+ ALOGW("source format didn't specify channel mask, using channel order");
+ channelMask = CHANNEL_MASK_USE_CHANNEL_ORDER;
+ }
+
if (mAudioSink.get() != NULL) {
status_t err = mAudioSink->open(
- mSampleRate, numChannels, AUDIO_FORMAT_PCM_16_BIT,
+ mSampleRate, numChannels, channelMask, AUDIO_FORMAT_PCM_16_BIT,
DEFAULT_AUDIOSINK_BUFFERCOUNT,
&AudioPlayer::AudioSinkCallback, this);
if (err != OK) {
@@ -137,12 +142,16 @@
mAudioSink->start();
} else {
+ // playing to an AudioTrack, set up mask if necessary
+ audio_channel_mask_t audioMask = channelMask == CHANNEL_MASK_USE_CHANNEL_ORDER ?
+ audio_channel_out_mask_from_count(numChannels) : channelMask;
+ if (0 == audioMask) {
+ return BAD_VALUE;
+ }
+
mAudioTrack = new AudioTrack(
- AUDIO_STREAM_MUSIC, mSampleRate, AUDIO_FORMAT_PCM_16_BIT,
- (numChannels == 2)
- ? AUDIO_CHANNEL_OUT_STEREO
- : AUDIO_CHANNEL_OUT_MONO,
- 0, 0, &AudioCallback, this, 0);
+ AUDIO_STREAM_MUSIC, mSampleRate, AUDIO_FORMAT_PCM_16_BIT, audioMask,
+ 0, AUDIO_POLICY_OUTPUT_FLAG_NONE, &AudioCallback, this, 0);
if ((err = mAudioTrack->initCheck()) != OK) {
delete mAudioTrack;
@@ -268,6 +277,16 @@
return mReachedEOS;
}
+status_t AudioPlayer::setPlaybackRatePermille(int32_t ratePermille) {
+ if (mAudioSink.get() != NULL) {
+ return mAudioSink->setPlaybackRatePermille(ratePermille);
+ } else if (mAudioTrack != NULL){
+ return mAudioTrack->setSampleRate(ratePermille * mSampleRate / 1000);
+ } else {
+ return NO_INIT;
+ }
+}
+
// static
size_t AudioPlayer::AudioSinkCallback(
MediaPlayerBase::AudioSink *audioSink,
@@ -400,7 +419,11 @@
timeToCompletionUs, timeToCompletionUs / 1E6);
postEOS = true;
- postEOSDelayUs = timeToCompletionUs + mLatencyUs;
+ if (mAudioSink->needsTrailingPadding()) {
+ postEOSDelayUs = timeToCompletionUs + mLatencyUs;
+ } else {
+ postEOSDelayUs = 0;
+ }
}
mReachedEOS = true;
@@ -418,8 +441,11 @@
kKeyTime, &mPositionTimeMediaUs));
mPositionTimeRealUs =
- ((mNumFramesPlayed + size_done / mFrameSize) * 1000000)
+ -mLatencyUs + ((mNumFramesPlayed + size_done / mFrameSize) * 1000000)
/ mSampleRate;
+ if (mPositionTimeRealUs < 0) {
+ mPositionTimeRealUs = 0;
+ }
ALOGV("buffer->size() = %d, "
"mPositionTimeMediaUs=%.2f mPositionTimeRealUs=%.2f",
@@ -474,7 +500,9 @@
int64_t AudioPlayer::getRealTimeUsLocked() const {
CHECK(mStarted);
CHECK_NE(mSampleRate, 0);
- return -mLatencyUs + (mNumFramesPlayed * 1000000) / mSampleRate;
+ int64_t t = -mLatencyUs + (mNumFramesPlayed * 1000000) / mSampleRate;
+ if (t < 0) return 0;
+ return t;
}
int64_t AudioPlayer::getMediaTimeUs() {
diff --git a/media/libstagefright/AudioSource.cpp b/media/libstagefright/AudioSource.cpp
index 2172cc0..0f1d841 100644
--- a/media/libstagefright/AudioSource.cpp
+++ b/media/libstagefright/AudioSource.cpp
@@ -47,21 +47,22 @@
}
AudioSource::AudioSource(
- int inputSource, uint32_t sampleRate, uint32_t channels)
+ audio_source_t inputSource, uint32_t sampleRate, uint32_t channelCount)
: mStarted(false),
mSampleRate(sampleRate),
mPrevSampleTimeUs(0),
mNumFramesReceived(0),
mNumClientOwnedBuffers(0) {
- ALOGV("sampleRate: %d, channels: %d", sampleRate, channels);
- CHECK(channels == 1 || channels == 2);
- uint32_t flags = AudioRecord::RECORD_AGC_ENABLE |
+ ALOGV("sampleRate: %d, channelCount: %d", sampleRate, channelCount);
+ CHECK(channelCount == 1 || channelCount == 2);
+ AudioRecord::record_flags flags = (AudioRecord::record_flags)
+ (AudioRecord::RECORD_AGC_ENABLE |
AudioRecord::RECORD_NS_ENABLE |
- AudioRecord::RECORD_IIR_ENABLE;
+ AudioRecord::RECORD_IIR_ENABLE);
mRecord = new AudioRecord(
inputSource, sampleRate, AUDIO_FORMAT_PCM_16_BIT,
- channels > 1? AUDIO_CHANNEL_IN_STEREO: AUDIO_CHANNEL_IN_MONO,
+ audio_channel_in_mask_from_count(channelCount),
4 * kMaxBufferSize / sizeof(int16_t), /* Enable ping-pong buffers */
flags,
AudioRecordCallbackFunction,
@@ -72,7 +73,7 @@
AudioSource::~AudioSource() {
if (mStarted) {
- stop();
+ reset();
}
delete mRecord;
@@ -130,7 +131,7 @@
}
}
-status_t AudioSource::stop() {
+status_t AudioSource::reset() {
Mutex::Autolock autoLock(mLock);
if (!mStarted) {
return UNKNOWN_ERROR;
@@ -282,8 +283,6 @@
mPrevSampleTimeUs = mStartTimeUs;
}
- int64_t timestampUs = mPrevSampleTimeUs;
-
size_t numLostBytes = 0;
if (mNumFramesReceived > 0) { // Ignore earlier frame lost
// getInputFramesLost() returns the number of lost frames.
@@ -293,37 +292,58 @@
CHECK_EQ(numLostBytes & 1, 0u);
CHECK_EQ(audioBuffer.size & 1, 0u);
- size_t bufferSize = numLostBytes + audioBuffer.size;
- MediaBuffer *buffer = new MediaBuffer(bufferSize);
if (numLostBytes > 0) {
- memset(buffer->data(), 0, numLostBytes);
- memcpy((uint8_t *) buffer->data() + numLostBytes,
- audioBuffer.i16, audioBuffer.size);
- } else {
- if (audioBuffer.size == 0) {
- ALOGW("Nothing is available from AudioRecord callback buffer");
- buffer->release();
- return OK;
- }
- memcpy((uint8_t *) buffer->data(),
- audioBuffer.i16, audioBuffer.size);
+ // Loss of audio frames should happen rarely; thus the LOGW should
+ // not cause a logging spam
+ ALOGW("Lost audio record data: %d bytes", numLostBytes);
}
+ while (numLostBytes > 0) {
+ size_t bufferSize = numLostBytes;
+ if (numLostBytes > kMaxBufferSize) {
+ numLostBytes -= kMaxBufferSize;
+ bufferSize = kMaxBufferSize;
+ } else {
+ numLostBytes = 0;
+ }
+ MediaBuffer *lostAudioBuffer = new MediaBuffer(bufferSize);
+ memset(lostAudioBuffer->data(), 0, bufferSize);
+ lostAudioBuffer->set_range(0, bufferSize);
+ queueInputBuffer_l(lostAudioBuffer, timeUs);
+ }
+
+ if (audioBuffer.size == 0) {
+ ALOGW("Nothing is available from AudioRecord callback buffer");
+ return OK;
+ }
+
+ const size_t bufferSize = audioBuffer.size;
+ MediaBuffer *buffer = new MediaBuffer(bufferSize);
+ memcpy((uint8_t *) buffer->data(),
+ audioBuffer.i16, audioBuffer.size);
buffer->set_range(0, bufferSize);
- timestampUs += ((1000000LL * (bufferSize >> 1)) +
- (mSampleRate >> 1)) / mSampleRate;
+ queueInputBuffer_l(buffer, timeUs);
+ return OK;
+}
+
+void AudioSource::queueInputBuffer_l(MediaBuffer *buffer, int64_t timeUs) {
+ const size_t bufferSize = buffer->range_length();
+ const size_t frameSize = mRecord->frameSize();
+ const int64_t timestampUs =
+ mPrevSampleTimeUs +
+ ((1000000LL * (bufferSize / frameSize)) +
+ (mSampleRate >> 1)) / mSampleRate;
if (mNumFramesReceived == 0) {
buffer->meta_data()->setInt64(kKeyAnchorTime, mStartTimeUs);
}
+
buffer->meta_data()->setInt64(kKeyTime, mPrevSampleTimeUs);
buffer->meta_data()->setInt64(kKeyDriftTime, timeUs - mInitialReadTimeUs);
mPrevSampleTimeUs = timestampUs;
- mNumFramesReceived += buffer->range_length() / sizeof(int16_t);
+ mNumFramesReceived += bufferSize / frameSize;
mBuffersReceived.push_back(buffer);
mFrameAvailableCondition.signal();
-
- return OK;
}
void AudioSource::trackMaxAmplitude(int16_t *data, int nSamples) {
diff --git a/media/libstagefright/AwesomePlayer.cpp b/media/libstagefright/AwesomePlayer.cpp
index d0cb7ff..f96a4df 100644
--- a/media/libstagefright/AwesomePlayer.cpp
+++ b/media/libstagefright/AwesomePlayer.cpp
@@ -30,13 +30,12 @@
#include "include/MPEG2TSExtractor.h"
#include "include/WVMExtractor.h"
-#include "timedtext/TimedTextPlayer.h"
-
#include <binder/IPCThreadState.h>
#include <binder/IServiceManager.h>
#include <media/IMediaPlayerService.h>
#include <media/stagefright/foundation/hexdump.h>
#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/timedtext/TimedTextDriver.h>
#include <media/stagefright/AudioPlayer.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/FileSource.h>
@@ -47,10 +46,8 @@
#include <media/stagefright/MetaData.h>
#include <media/stagefright/OMXCodec.h>
-#include <surfaceflinger/Surface.h>
#include <gui/ISurfaceTexture.h>
#include <gui/SurfaceTextureClient.h>
-#include <surfaceflinger/ISurfaceComposer.h>
#include <media/stagefright/foundation/AMessage.h>
@@ -192,7 +189,7 @@
mVideoBuffer(NULL),
mDecryptHandle(NULL),
mLastVideoTimeUs(-1),
- mTextPlayer(NULL) {
+ mTextDriver(NULL) {
CHECK_EQ(mClient.connect(), (status_t)OK);
DataSource::RegisterDefaultSniffers();
@@ -335,6 +332,14 @@
return UNKNOWN_ERROR;
}
+ if (extractor->getDrmFlag()) {
+ checkDrmStatus(dataSource);
+ }
+
+ return setDataSource_l(extractor);
+}
+
+void AwesomePlayer::checkDrmStatus(const sp<DataSource>& dataSource) {
dataSource->getDrmInfo(mDecryptHandle, &mDrmManagerClient);
if (mDecryptHandle != NULL) {
CHECK(mDrmManagerClient);
@@ -342,8 +347,6 @@
notifyListener_l(MEDIA_ERROR, MEDIA_ERROR_UNKNOWN, ERROR_DRM_NO_LICENSE);
}
}
-
- return setDataSource_l(extractor);
}
status_t AwesomePlayer::setDataSource_l(const sp<MediaExtractor> &extractor) {
@@ -524,9 +527,9 @@
delete mAudioPlayer;
mAudioPlayer = NULL;
- if (mTextPlayer != NULL) {
- delete mTextPlayer;
- mTextPlayer = NULL;
+ if (mTextDriver != NULL) {
+ delete mTextDriver;
+ mTextDriver = NULL;
}
mVideoRenderer.clear();
@@ -1111,8 +1114,8 @@
modifyFlags(AUDIO_RUNNING, CLEAR);
}
- if (mFlags & TEXTPLAYER_STARTED) {
- mTextPlayer->pause();
+ if (mFlags & TEXTPLAYER_INITIALIZED) {
+ mTextDriver->pause();
modifyFlags(TEXT_RUNNING, CLEAR);
}
@@ -1265,32 +1268,6 @@
return OK;
}
-status_t AwesomePlayer::setTimedTextTrackIndex(int32_t index) {
- if (mTextPlayer != NULL) {
- if (index >= 0) { // to turn on a text track
- status_t err = mTextPlayer->setTimedTextTrackIndex(index);
- if (err != OK) {
- return err;
- }
-
- modifyFlags(TEXT_RUNNING, SET);
- modifyFlags(TEXTPLAYER_STARTED, SET);
- return OK;
- } else { // to turn off the text track display
- if (mFlags & TEXT_RUNNING) {
- modifyFlags(TEXT_RUNNING, CLEAR);
- }
- if (mFlags & TEXTPLAYER_STARTED) {
- modifyFlags(TEXTPLAYER_STARTED, CLEAR);
- }
-
- return mTextPlayer->setTimedTextTrackIndex(index);
- }
- } else {
- return INVALID_OPERATION;
- }
-}
-
status_t AwesomePlayer::seekTo_l(int64_t timeUs) {
if (mFlags & CACHE_UNDERRUN) {
modifyFlags(CACHE_UNDERRUN, CLEAR);
@@ -1312,8 +1289,8 @@
seekAudioIfNecessary_l();
- if (mFlags & TEXTPLAYER_STARTED) {
- mTextPlayer->seekTo(mSeekTimeUs);
+ if (mFlags & TEXTPLAYER_INITIALIZED) {
+ mTextDriver->seekToAsync(mSeekTimeUs);
}
if (!(mFlags & PLAYING)) {
@@ -1354,15 +1331,15 @@
mAudioTrack = source;
}
-void AwesomePlayer::addTextSource(sp<MediaSource> source) {
+void AwesomePlayer::addTextSource(const sp<MediaSource>& source) {
Mutex::Autolock autoLock(mTimedTextLock);
CHECK(source != NULL);
- if (mTextPlayer == NULL) {
- mTextPlayer = new TimedTextPlayer(this, mListener, &mQueue);
+ if (mTextDriver == NULL) {
+ mTextDriver = new TimedTextDriver(mListener);
}
- mTextPlayer->addTextSource(source);
+ mTextDriver->addInBandTextSource(source);
}
status_t AwesomePlayer::initAudioDecoder() {
@@ -1603,7 +1580,7 @@
mSeekTimeUs,
mSeeking == SEEK_VIDEO_ONLY
? MediaSource::ReadOptions::SEEK_NEXT_SYNC
- : MediaSource::ReadOptions::SEEK_CLOSEST_SYNC);
+ : MediaSource::ReadOptions::SEEK_CLOSEST);
}
for (;;) {
status_t err = mVideoSource->read(&mVideoBuffer, &options);
@@ -1688,8 +1665,8 @@
}
}
- if ((mFlags & TEXTPLAYER_STARTED) && !(mFlags & (TEXT_RUNNING | SEEK_PREVIEW))) {
- mTextPlayer->resume();
+ if ((mFlags & TEXTPLAYER_INITIALIZED) && !(mFlags & (TEXT_RUNNING | SEEK_PREVIEW))) {
+ mTextDriver->start();
modifyFlags(TEXT_RUNNING, SET);
}
@@ -2095,7 +2072,7 @@
String8 mimeType;
float confidence;
sp<AMessage> dummy;
- bool success = SniffDRM(dataSource, &mimeType, &confidence, &dummy);
+ bool success = SniffWVM(dataSource, &mimeType, &confidence, &dummy);
if (!success
|| strcasecmp(
@@ -2105,6 +2082,8 @@
mWVMExtractor = new WVMExtractor(dataSource);
mWVMExtractor->setAdaptiveStreamingMode(true);
+ if (mUIDValid)
+ mWVMExtractor->setUID(mUID);
extractor = mWVMExtractor;
} else {
extractor = MediaExtractor::Create(
@@ -2115,13 +2094,8 @@
}
}
- dataSource->getDrmInfo(mDecryptHandle, &mDrmManagerClient);
-
- if (mDecryptHandle != NULL) {
- CHECK(mDrmManagerClient);
- if (RightsStatus::RIGHTS_VALID != mDecryptHandle->status) {
- notifyListener_l(MEDIA_ERROR, MEDIA_ERROR_UNKNOWN, ERROR_DRM_NO_LICENSE);
- }
+ if (extractor->getDrmFlag()) {
+ checkDrmStatus(dataSource);
}
status_t err = setDataSource_l(extractor);
@@ -2232,24 +2206,18 @@
status_t AwesomePlayer::setParameter(int key, const Parcel &request) {
switch (key) {
- case KEY_PARAMETER_TIMED_TEXT_TRACK_INDEX:
- {
- Mutex::Autolock autoLock(mTimedTextLock);
- return setTimedTextTrackIndex(request.readInt32());
- }
- case KEY_PARAMETER_TIMED_TEXT_ADD_OUT_OF_BAND_SOURCE:
- {
- Mutex::Autolock autoLock(mTimedTextLock);
- if (mTextPlayer == NULL) {
- mTextPlayer = new TimedTextPlayer(this, mListener, &mQueue);
- }
-
- return mTextPlayer->setParameter(key, request);
- }
case KEY_PARAMETER_CACHE_STAT_COLLECT_FREQ_MS:
{
return setCacheStatCollectFreq(request);
}
+ case KEY_PARAMETER_PLAYBACK_RATE_PERMILLE:
+ {
+ if (mAudioPlayer != NULL) {
+ return mAudioPlayer->setPlaybackRatePermille(request.readInt32());
+ } else {
+ return NO_INIT;
+ }
+ }
default:
{
return ERROR_UNSUPPORTED;
@@ -2286,6 +2254,90 @@
}
}
+status_t AwesomePlayer::invoke(const Parcel &request, Parcel *reply) {
+ if (NULL == reply) {
+ return android::BAD_VALUE;
+ }
+ int32_t methodId;
+ status_t ret = request.readInt32(&methodId);
+ if (ret != android::OK) {
+ return ret;
+ }
+ switch(methodId) {
+ case INVOKE_ID_GET_TRACK_INFO:
+ {
+ Mutex::Autolock autoLock(mTimedTextLock);
+ if (mTextDriver == NULL) {
+ return INVALID_OPERATION;
+ }
+ mTextDriver->getTrackInfo(reply);
+ return OK;
+ }
+ case INVOKE_ID_ADD_EXTERNAL_SOURCE:
+ {
+ Mutex::Autolock autoLock(mTimedTextLock);
+ if (mTextDriver == NULL) {
+ mTextDriver = new TimedTextDriver(mListener);
+ }
+ // String values written in Parcel are UTF-16 values.
+ String8 uri(request.readString16());
+ String8 mimeType(request.readString16());
+ return mTextDriver->addOutOfBandTextSource(uri, mimeType);
+ }
+ case INVOKE_ID_ADD_EXTERNAL_SOURCE_FD:
+ {
+ Mutex::Autolock autoLock(mTimedTextLock);
+ if (mTextDriver == NULL) {
+ mTextDriver = new TimedTextDriver(mListener);
+ }
+ int fd = request.readFileDescriptor();
+ off64_t offset = request.readInt64();
+ size_t length = request.readInt64();
+ String8 mimeType(request.readString16());
+ return mTextDriver->addOutOfBandTextSource(
+ fd, offset, length, mimeType);
+ }
+ case INVOKE_ID_SELECT_TRACK:
+ {
+ Mutex::Autolock autoLock(mTimedTextLock);
+ if (mTextDriver == NULL) {
+ return INVALID_OPERATION;
+ }
+
+ status_t err = mTextDriver->selectTrack(
+ request.readInt32());
+ if (err == OK) {
+ modifyFlags(TEXTPLAYER_INITIALIZED, SET);
+ if (mFlags & PLAYING && !(mFlags & TEXT_RUNNING)) {
+ mTextDriver->start();
+ modifyFlags(TEXT_RUNNING, SET);
+ }
+ }
+ return err;
+ }
+ case INVOKE_ID_UNSELECT_TRACK:
+ {
+ Mutex::Autolock autoLock(mTimedTextLock);
+ if (mTextDriver == NULL) {
+ return INVALID_OPERATION;
+ }
+ status_t err = mTextDriver->unselectTrack(
+ request.readInt32());
+ if (err == OK) {
+ modifyFlags(TEXTPLAYER_INITIALIZED, CLEAR);
+ modifyFlags(TEXT_RUNNING, CLEAR);
+ }
+ return err;
+ }
+ default:
+ {
+ return ERROR_UNSUPPORTED;
+ }
+ }
+ // It will not reach here.
+ return OK;
+}
+
bool AwesomePlayer::isStreamingHTTP() const {
return mCachedSource != NULL || mWVMExtractor != NULL;
}
diff --git a/media/libstagefright/CameraSource.cpp b/media/libstagefright/CameraSource.cpp
index 1850c9c..fd3f892 100755
--- a/media/libstagefright/CameraSource.cpp
+++ b/media/libstagefright/CameraSource.cpp
@@ -20,14 +20,14 @@
#include <OMX_Component.h>
#include <binder/IPCThreadState.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/CameraSource.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
#include <camera/Camera.h>
#include <camera/CameraParameters.h>
-#include <surfaceflinger/Surface.h>
+#include <gui/Surface.h>
#include <utils/String8.h>
#include <cutils/properties.h>
@@ -114,7 +114,7 @@
ALOGE("Uknown color format (%s), please add it to "
"CameraSource::getColorFormat", colorFormat);
- CHECK_EQ(0, "Unknown color format");
+ CHECK(!"Unknown color format");
}
CameraSource *CameraSource::Create() {
@@ -182,7 +182,7 @@
int32_t cameraId) {
if (camera == 0) {
- mCamera = Camera::connect(cameraId);
+ mCamera = Camera::connect(cameraId, false, false);
if (mCamera == 0) return -EBUSY;
mCameraFlags &= ~FLAGS_HOT_CAMERA;
} else {
@@ -515,9 +515,13 @@
return err;
}
- // This CHECK is good, since we just passed the lock/unlock
- // check earlier by calling mCamera->setParameters().
- CHECK_EQ(OK, mCamera->setPreviewDisplay(mSurface));
+ // Set the preview display. Skip this if mSurface is null because
+ // applications may already set a surface to the camera.
+ if (mSurface != NULL) {
+ // This CHECK is good, since we just passed the lock/unlock
+ // check earlier by calling mCamera->setParameters().
+ CHECK_EQ((status_t)OK, mCamera->setPreviewDisplay(mSurface));
+ }
// By default, do not store metadata in video buffers
mIsMetaDataStoredInVideoBuffers = false;
@@ -548,7 +552,7 @@
CameraSource::~CameraSource() {
if (mStarted) {
- stop();
+ reset();
} else if (mInitCheck == OK) {
// Camera is initialized but because start() is never called,
// the lock on Camera is never released(). This makes sure
@@ -566,7 +570,8 @@
if (mCameraFlags & FLAGS_HOT_CAMERA) {
mCamera->unlock();
mCamera.clear();
- CHECK_EQ(OK, mCameraRecordingProxy->startRecording(new ProxyListener(this)));
+ CHECK_EQ((status_t)OK,
+ mCameraRecordingProxy->startRecording(new ProxyListener(this)));
} else {
mCamera->setListener(new CameraSourceListener(this));
mCamera->startRecording();
@@ -632,8 +637,8 @@
mCameraFlags = 0;
}
-status_t CameraSource::stop() {
- ALOGD("stop: E");
+status_t CameraSource::reset() {
+ ALOGD("reset: E");
Mutex::Autolock autoLock(mLock);
mStarted = false;
mFrameAvailableCondition.signal();
@@ -670,7 +675,7 @@
}
CHECK_EQ(mNumFramesReceived, mNumFramesEncoded + mNumFramesDropped);
- ALOGD("stop: X");
+ ALOGD("reset: X");
return OK;
}
@@ -718,7 +723,7 @@
return;
}
}
- CHECK_EQ(0, "signalBufferReturned: bogus buffer");
+ CHECK(!"signalBufferReturned: bogus buffer");
}
status_t CameraSource::read(
diff --git a/media/libstagefright/CameraSourceTimeLapse.cpp b/media/libstagefright/CameraSourceTimeLapse.cpp
index 263ab50..26ce7ae 100644
--- a/media/libstagefright/CameraSourceTimeLapse.cpp
+++ b/media/libstagefright/CameraSourceTimeLapse.cpp
@@ -20,9 +20,9 @@
#include <binder/IPCThreadState.h>
#include <binder/MemoryBase.h>
#include <binder/MemoryHeapBase.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/CameraSource.h>
#include <media/stagefright/CameraSourceTimeLapse.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MetaData.h>
#include <camera/Camera.h>
#include <camera/CameraParameters.h>
@@ -87,6 +87,10 @@
}
CameraSourceTimeLapse::~CameraSourceTimeLapse() {
+ if (mLastReadBufferCopy) {
+ mLastReadBufferCopy->release();
+ mLastReadBufferCopy = NULL;
+ }
}
void CameraSourceTimeLapse::startQuickReadReturns() {
@@ -204,15 +208,6 @@
}
}
-void CameraSourceTimeLapse::stopCameraRecording() {
- ALOGV("stopCameraRecording");
- CameraSource::stopCameraRecording();
- if (mLastReadBufferCopy) {
- mLastReadBufferCopy->release();
- mLastReadBufferCopy = NULL;
- }
-}
-
sp<IMemory> CameraSourceTimeLapse::createIMemoryCopy(
const sp<IMemory> &source_data) {
diff --git a/media/libstagefright/DRMExtractor.cpp b/media/libstagefright/DRMExtractor.cpp
index 9452ab1..524c3aa 100644
--- a/media/libstagefright/DRMExtractor.cpp
+++ b/media/libstagefright/DRMExtractor.cpp
@@ -23,6 +23,7 @@
#include <arpa/inet.h>
#include <utils/String8.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/Utils.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/MediaSource.h>
@@ -30,7 +31,6 @@
#include <media/stagefright/MetaData.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaBuffer.h>
-#include <media/stagefright/MediaDebug.h>
#include <drm/drm_framework_common.h>
#include <utils/Errors.h>
@@ -282,13 +282,13 @@
if (decryptHandle != NULL) {
if (decryptHandle->decryptApiType == DecryptApiType::CONTAINER_BASED) {
*mimeType = String8("drm+container_based+") + decryptHandle->mimeType;
+ *confidence = 10.0f;
} else if (decryptHandle->decryptApiType == DecryptApiType::ELEMENTARY_STREAM_BASED) {
*mimeType = String8("drm+es_based+") + decryptHandle->mimeType;
- } else if (decryptHandle->decryptApiType == DecryptApiType::WV_BASED) {
- *mimeType = MEDIA_MIMETYPE_CONTAINER_WVM;
- ALOGW("SniffWVM: found match\n");
+ *confidence = 10.0f;
+ } else {
+ return false;
}
- *confidence = 10.0f;
return true;
}
diff --git a/media/libstagefright/DataSource.cpp b/media/libstagefright/DataSource.cpp
index 43539bb..d0a7880 100644
--- a/media/libstagefright/DataSource.cpp
+++ b/media/libstagefright/DataSource.cpp
@@ -15,6 +15,12 @@
*/
#include "include/AMRExtractor.h"
+#include "include/AVIExtractor.h"
+
+#if CHROMIUM_AVAILABLE
+#include "include/DataUriSource.h"
+#endif
+
#include "include/MP3Extractor.h"
#include "include/MPEG4Extractor.h"
#include "include/WAVExtractor.h"
@@ -26,6 +32,7 @@
#include "include/DRMExtractor.h"
#include "include/FLACExtractor.h"
#include "include/AACExtractor.h"
+#include "include/WVMExtractor.h"
#include "matroska/MatroskaExtractor.h"
@@ -112,7 +119,9 @@
RegisterSniffer(SniffMPEG2TS);
RegisterSniffer(SniffMP3);
RegisterSniffer(SniffAAC);
+ RegisterSniffer(SniffAVI);
RegisterSniffer(SniffMPEG2PS);
+ RegisterSniffer(SniffWVM);
char value[PROPERTY_VALUE_MAX];
if (property_get("drm.service.enabled", value, NULL)
@@ -134,6 +143,10 @@
return NULL;
}
source = new NuCachedSource2(httpSource);
+# if CHROMIUM_AVAILABLE
+ } else if (!strncasecmp("data:", uri, 5)) {
+ source = new DataUriSource(uri);
+#endif
} else {
// Assume it's a filename.
source = new FileSource(uri);
diff --git a/media/libstagefright/FileSource.cpp b/media/libstagefright/FileSource.cpp
index 73cb48c..73c8d03 100644
--- a/media/libstagefright/FileSource.cpp
+++ b/media/libstagefright/FileSource.cpp
@@ -14,8 +14,8 @@
* limitations under the License.
*/
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/FileSource.h>
-#include <media/stagefright/MediaDebug.h>
#include <sys/types.h>
#include <unistd.h>
#include <sys/types.h>
@@ -127,7 +127,7 @@
return OK;
}
-sp<DecryptHandle> FileSource::DrmInitialization() {
+sp<DecryptHandle> FileSource::DrmInitialization(const char *mime) {
if (mDrmManagerClient == NULL) {
mDrmManagerClient = new DrmManagerClient();
}
@@ -138,7 +138,7 @@
if (mDecryptHandle == NULL) {
mDecryptHandle = mDrmManagerClient->openDecryptSession(
- mFd, mOffset, mLength);
+ mFd, mOffset, mLength, mime);
}
if (mDecryptHandle == NULL) {
diff --git a/media/libstagefright/JPEGSource.cpp b/media/libstagefright/JPEGSource.cpp
index e818115..bafa4b2 100644
--- a/media/libstagefright/JPEGSource.cpp
+++ b/media/libstagefright/JPEGSource.cpp
@@ -18,10 +18,10 @@
#define LOG_TAG "JPEGSource"
#include <utils/Log.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/JPEGSource.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
@@ -59,7 +59,7 @@
mWidth(0),
mHeight(0),
mOffset(0) {
- CHECK_EQ(parseJPEG(), OK);
+ CHECK_EQ(parseJPEG(), (status_t)OK);
CHECK(mSource->getSize(&mSize) == OK);
}
diff --git a/media/libstagefright/MP3Extractor.cpp b/media/libstagefright/MP3Extractor.cpp
index 2215c07..6abaf23 100644
--- a/media/libstagefright/MP3Extractor.cpp
+++ b/media/libstagefright/MP3Extractor.cpp
@@ -311,10 +311,25 @@
mMeta->setInt32(kKeyBitRate, bitrate * 1000);
mMeta->setInt32(kKeyChannelCount, num_channels);
- mSeeker = XINGSeeker::CreateFromSource(mDataSource, mFirstFramePos);
+ sp<XINGSeeker> seeker = XINGSeeker::CreateFromSource(mDataSource, mFirstFramePos);
- if (mSeeker == NULL) {
+ if (seeker == NULL) {
mSeeker = VBRISeeker::CreateFromSource(mDataSource, post_id3_pos);
+ } else {
+ mSeeker = seeker;
+ int encd = seeker->getEncoderDelay();
+ int encp = seeker->getEncoderPadding();
+ if (encd != 0 || encp != 0) {
+ mMeta->setInt32(kKeyEncoderDelay, encd);
+ mMeta->setInt32(kKeyEncoderPadding, encp);
+ }
+ }
+
+ if (mSeeker != NULL) {
+ // While it is safe to send the XING/VBRI frame to the decoder, this will
+ // result in an extra 1152 samples being output. The real first frame to
+ // decode is after the XING/VBRI frame, so skip there.
+ mFirstFramePos += frame_size;
}
int64_t durationUs;
@@ -333,6 +348,37 @@
}
mInitCheck = OK;
+
+ // get iTunes-style gapless info if present
+ ID3 id3(mDataSource);
+ if (id3.isValid()) {
+ ID3::Iterator *com = new ID3::Iterator(id3, "COM");
+ if (com->done()) {
+ delete com;
+ com = new ID3::Iterator(id3, "COMM");
+ }
+ while(!com->done()) {
+ String8 commentdesc;
+ String8 commentvalue;
+ com->getString(&commentdesc, &commentvalue);
+ const char * desc = commentdesc.string();
+ const char * value = commentvalue.string();
+
+ // first 3 characters are the language, which we don't care about
+ if(strlen(desc) > 3 && strcmp(desc + 3, "iTunSMPB") == 0) {
+
+ int32_t delay, padding;
+ if (sscanf(value, " %*x %x %x %*x", &delay, &padding) == 2) {
+ mMeta->setInt32(kKeyEncoderDelay, delay);
+ mMeta->setInt32(kKeyEncoderPadding, padding);
+ }
+ break;
+ }
+ com->next();
+ }
+ delete com;
+ com = NULL;
+ }
}
size_t MP3Extractor::countTracks() {
diff --git a/media/libstagefright/MPEG2TSWriter.cpp b/media/libstagefright/MPEG2TSWriter.cpp
index 36009ab..f702376 100644
--- a/media/libstagefright/MPEG2TSWriter.cpp
+++ b/media/libstagefright/MPEG2TSWriter.cpp
@@ -244,7 +244,7 @@
sp<AMessage> notify = mNotify->dup();
notify->setInt32("what", kNotifyBuffer);
- notify->setObject("buffer", out);
+ notify->setBuffer("buffer", out);
notify->setInt32("oob", true);
notify->post();
}
@@ -270,7 +270,7 @@
copy->meta()->setInt32("isSync", true);
}
- notify->setObject("buffer", copy);
+ notify->setBuffer("buffer", copy);
notify->post();
}
@@ -351,7 +351,7 @@
sp<AMessage> notify = mNotify->dup();
notify->setInt32("what", kNotifyBuffer);
- notify->setObject("buffer", mAACBuffer);
+ notify->setBuffer("buffer", mAACBuffer);
notify->post();
mAACBuffer.clear();
@@ -513,7 +513,7 @@
MPEG2TSWriter::~MPEG2TSWriter() {
if (mStarted) {
- stop();
+ reset();
}
mLooper->unregisterHandler(mReflector->id());
@@ -564,7 +564,7 @@
return OK;
}
-status_t MPEG2TSWriter::stop() {
+status_t MPEG2TSWriter::reset() {
CHECK(mStarted);
for (size_t i = 0; i < mSources.size(); ++i) {
@@ -614,10 +614,8 @@
++mNumSourcesDone;
} else if (what == SourceInfo::kNotifyBuffer) {
- sp<RefBase> obj;
- CHECK(msg->findObject("buffer", &obj));
-
- sp<ABuffer> buffer = static_cast<ABuffer *>(obj.get());
+ sp<ABuffer> buffer;
+ CHECK(msg->findBuffer("buffer", &buffer));
int32_t oob;
if (msg->findInt32("oob", &oob) && oob) {
diff --git a/media/libstagefright/MPEG4Extractor.cpp b/media/libstagefright/MPEG4Extractor.cpp
index 22bdd95..6c95d4e 100644
--- a/media/libstagefright/MPEG4Extractor.cpp
+++ b/media/libstagefright/MPEG4Extractor.cpp
@@ -20,7 +20,6 @@
#include "include/MPEG4Extractor.h"
#include "include/SampleTable.h"
#include "include/ESDS.h"
-#include "timedtext/TimedTextPlayer.h"
#include <arpa/inet.h>
@@ -1372,8 +1371,9 @@
uint32_t type = ntohl(buffer);
// For the 3GPP file format, the handler-type within the 'hdlr' box
- // shall be 'text'
- if (type == FOURCC('t', 'e', 'x', 't')) {
+ // shall be 'text'. We also want to support 'sbtl' handler type
+ // for a practical reason as various MPEG4 containers use it.
+ if (type == FOURCC('t', 'e', 'x', 't') || type == FOURCC('s', 'b', 't', 'l')) {
mLastTrack->meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_TEXT_3GPP);
}
@@ -2429,4 +2429,3 @@
}
} // namespace android
-
diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp
index 06dd875..7ebbe1d 100755
--- a/media/libstagefright/MPEG4Writer.cpp
+++ b/media/libstagefright/MPEG4Writer.cpp
@@ -23,10 +23,10 @@
#include <pthread.h>
#include <sys/prctl.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MPEG4Writer.h>
#include <media/stagefright/MediaBuffer.h>
#include <media/stagefright/MetaData.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaSource.h>
@@ -70,6 +70,10 @@
status_t dump(int fd, const Vector<String16>& args) const;
private:
+ enum {
+ kMaxCttsOffsetTimeUs = 1000000LL, // 1 second
+ };
+
MPEG4Writer *mOwner;
sp<MetaData> mMeta;
sp<MediaSource> mSource;
@@ -137,11 +141,12 @@
: sampleCount(count), sampleDuration(timescaledDur) {}
uint32_t sampleCount;
- int32_t sampleDuration; // time scale based
+ uint32_t sampleDuration; // time scale based
};
- bool mHasNegativeCttsDeltaDuration;
size_t mNumCttsTableEntries;
List<CttsTableEntry> mCttsTableEntries;
+ int64_t mMinCttsOffsetTimeUs;
+ int64_t mMaxCttsOffsetTimeUs;
// Sequence parameter set or picture parameter set
struct AVCParamSet {
@@ -172,6 +177,8 @@
// Update the audio track's drift information.
void updateDriftTime(const sp<MetaData>& meta);
+ int32_t getStartTimeOffsetScaledTime() const;
+
static void *ThreadWrapper(void *me);
status_t threadEntry();
@@ -282,7 +289,7 @@
}
MPEG4Writer::~MPEG4Writer() {
- stop();
+ reset();
while (!mTracks.empty()) {
List<Track *>::iterator it = mTracks.begin();
@@ -471,7 +478,7 @@
!param->findInt32(kKeyTimeScale, &mTimeScale)) {
mTimeScale = 1000;
}
- CHECK(mTimeScale > 0);
+ CHECK_GT(mTimeScale, 0);
ALOGV("movie time scale: %d", mTimeScale);
mStreamableFile = true;
@@ -490,7 +497,7 @@
}
mEstimatedMoovBoxSize = estimateMoovBoxSize(bitRate);
}
- CHECK(mEstimatedMoovBoxSize >= 8);
+ CHECK_GE(mEstimatedMoovBoxSize, 8);
lseek64(mFd, mFreeBoxOffset, SEEK_SET);
writeInt32(mEstimatedMoovBoxSize);
write("free", 4);
@@ -616,7 +623,7 @@
mStarted = false;
}
-status_t MPEG4Writer::stop() {
+status_t MPEG4Writer::reset() {
if (mInitCheck != OK) {
return OK;
} else {
@@ -684,7 +691,7 @@
mWriteMoovBoxToMemory = false;
if (mStreamableFile) {
- CHECK(mMoovBoxBufferOffset + 8 <= mEstimatedMoovBoxSize);
+ CHECK_LE(mMoovBoxBufferOffset + 8, mEstimatedMoovBoxSize);
// Moov box
lseek64(mFd, mFreeBoxOffset, SEEK_SET);
@@ -856,7 +863,7 @@
mOffset += length + 4;
} else {
- CHECK(length < 65536);
+ CHECK_LT(length, 65536);
uint8_t x = length >> 8;
::write(mFd, &x, 1);
@@ -1085,7 +1092,7 @@
void MPEG4Writer::setStartTimestampUs(int64_t timeUs) {
ALOGI("setStartTimestampUs: %lld", timeUs);
- CHECK(timeUs >= 0);
+ CHECK_GE(timeUs, 0ll);
Mutex::Autolock autoLock(mLock);
if (mStartTimestampUs < 0 || mStartTimestampUs > timeUs) {
mStartTimestampUs = timeUs;
@@ -1186,9 +1193,6 @@
if (mIsAudio) {
return;
}
- if (duration < 0 && !mHasNegativeCttsDeltaDuration) {
- mHasNegativeCttsDeltaDuration = true;
- }
CttsTableEntry cttsEntry(sampleCount, duration);
mCttsTableEntries.push_back(cttsEntry);
++mNumCttsTableEntries;
@@ -1218,7 +1222,7 @@
mTimeScale = timeScale;
}
- CHECK(mTimeScale > 0);
+ CHECK_GT(mTimeScale, 0);
}
void MPEG4Writer::Track::getCodecSpecificDataFromInputFormatIfPossible() {
@@ -1299,7 +1303,7 @@
}
}
- CHECK("Received a chunk for a unknown track" == 0);
+ CHECK(!"Received a chunk for a unknown track");
}
void MPEG4Writer::writeChunkToFile(Chunk* chunk) {
@@ -1509,7 +1513,6 @@
mMdatSizeBytes = 0;
mMaxChunkDurationUs = 0;
- mHasNegativeCttsDeltaDuration = false;
pthread_create(&mThread, &attr, ThreadWrapper, this);
pthread_attr_destroy(&attr);
@@ -1833,29 +1836,18 @@
int32_t nChunks = 0;
int32_t nZeroLengthFrames = 0;
int64_t lastTimestampUs = 0; // Previous sample time stamp
- int64_t lastCttsTimeUs = 0; // Previous sample time stamp
int64_t lastDurationUs = 0; // Between the previous two samples
int64_t currDurationTicks = 0; // Timescale based ticks
int64_t lastDurationTicks = 0; // Timescale based ticks
int32_t sampleCount = 1; // Sample count in the current stts table entry
- int64_t currCttsDurTicks = 0; // Timescale based ticks
- int64_t lastCttsDurTicks = 0; // Timescale based ticks
- int32_t cttsSampleCount = 1; // Sample count in the current ctts table entry
- uint32_t previousSampleSize = 0; // Size of the previous sample
+ uint32_t previousSampleSize = 0; // Size of the previous sample
int64_t previousPausedDurationUs = 0;
int64_t timestampUs = 0;
- int64_t cttsDeltaTimeUs = 0;
- bool hasBFrames = false;
+ int64_t cttsOffsetTimeUs = 0;
+ int64_t currCttsOffsetTimeTicks = 0; // Timescale based ticks
+ int64_t lastCttsOffsetTimeTicks = -1; // Timescale based ticks
+ int32_t cttsSampleCount = 0; // Sample count in the current ctts table entry
-#if 1
- // XXX: Samsung's video encoder's output buffer timestamp
- // is not correct. see bug 4724339
- char value[PROPERTY_VALUE_MAX];
- if (property_get("rw.media.record.hasb", value, NULL) &&
- (!strcasecmp(value, "true") || !strcasecmp(value, "1"))) {
- hasBFrames = true;
- }
-#endif
if (mIsAudio) {
prctl(PR_SET_NAME, (unsigned long)"AudioTrackEncoding", 0, 0, 0);
} else {
@@ -1897,7 +1889,7 @@
(const uint8_t *)buffer->data()
+ buffer->range_offset(),
buffer->range_length());
- CHECK_EQ(OK, err);
+ CHECK_EQ((status_t)OK, err);
} else if (mIsMPEG4) {
mCodecSpecificDataSize = buffer->range_length();
mCodecSpecificData = malloc(mCodecSpecificDataSize);
@@ -1963,32 +1955,64 @@
if (mResumed) {
int64_t durExcludingEarlierPausesUs = timestampUs - previousPausedDurationUs;
- CHECK(durExcludingEarlierPausesUs >= 0);
+ CHECK_GE(durExcludingEarlierPausesUs, 0ll);
int64_t pausedDurationUs = durExcludingEarlierPausesUs - mTrackDurationUs;
- CHECK(pausedDurationUs >= lastDurationUs);
+ CHECK_GE(pausedDurationUs, lastDurationUs);
previousPausedDurationUs += pausedDurationUs - lastDurationUs;
mResumed = false;
}
timestampUs -= previousPausedDurationUs;
- CHECK(timestampUs >= 0);
- if (!mIsAudio && hasBFrames) {
+ CHECK_GE(timestampUs, 0ll);
+ if (!mIsAudio) {
/*
* Composition time: timestampUs
* Decoding time: decodingTimeUs
- * Composition time delta = composition time - decoding time
- *
- * We save picture decoding time stamp delta in stts table entries,
- * and composition time delta duration in ctts table entries.
+ * Composition time offset = composition time - decoding time
*/
int64_t decodingTimeUs;
CHECK(meta_data->findInt64(kKeyDecodingTime, &decodingTimeUs));
decodingTimeUs -= previousPausedDurationUs;
- int64_t timeUs = decodingTimeUs;
- cttsDeltaTimeUs = timestampUs - decodingTimeUs;
+ cttsOffsetTimeUs =
+ timestampUs + kMaxCttsOffsetTimeUs - decodingTimeUs;
+ CHECK_GE(cttsOffsetTimeUs, 0ll);
timestampUs = decodingTimeUs;
- ALOGV("decoding time: %lld and ctts delta time: %lld",
- timestampUs, cttsDeltaTimeUs);
+ ALOGV("decoding time: %lld and ctts offset time: %lld",
+ timestampUs, cttsOffsetTimeUs);
+
+ // Update ctts box table if necessary
+ currCttsOffsetTimeTicks =
+ (cttsOffsetTimeUs * mTimeScale + 500000LL) / 1000000LL;
+ CHECK_LE(currCttsOffsetTimeTicks, 0x0FFFFFFFFLL);
+ if (mNumSamples == 0) {
+ // Force the first ctts table entry to have one single entry
+ // so that we can do adjustment for the initial track start
+ // time offset easily in writeCttsBox().
+ lastCttsOffsetTimeTicks = currCttsOffsetTimeTicks;
+ addOneCttsTableEntry(1, currCttsOffsetTimeTicks);
+ cttsSampleCount = 0; // No sample in ctts box is pending
+ } else {
+ if (currCttsOffsetTimeTicks != lastCttsOffsetTimeTicks) {
+ addOneCttsTableEntry(cttsSampleCount, lastCttsOffsetTimeTicks);
+ lastCttsOffsetTimeTicks = currCttsOffsetTimeTicks;
+ cttsSampleCount = 1; // One sample in ctts box is pending
+ } else {
+ ++cttsSampleCount;
+ }
+ }
+
+ // Update ctts time offset range
+ if (mNumSamples == 0) {
+ mMinCttsOffsetTimeUs = currCttsOffsetTimeTicks;
+ mMaxCttsOffsetTimeUs = currCttsOffsetTimeTicks;
+ } else {
+ if (currCttsOffsetTimeTicks > mMaxCttsOffsetTimeUs) {
+ mMaxCttsOffsetTimeUs = currCttsOffsetTimeTicks;
+ } else if (currCttsOffsetTimeTicks < mMinCttsOffsetTimeUs) {
+ mMinCttsOffsetTimeUs = currCttsOffsetTimeTicks;
+ }
+ }
+
}
if (mIsRealTimeRecording) {
@@ -1997,7 +2021,7 @@
}
}
- CHECK(timestampUs >= 0);
+ CHECK_GE(timestampUs, 0ll);
ALOGV("%s media time stamp: %lld and previous paused duration %lld",
mIsAudio? "Audio": "Video", timestampUs, previousPausedDurationUs);
if (timestampUs > mTrackDurationUs) {
@@ -2012,6 +2036,7 @@
currDurationTicks =
((timestampUs * mTimeScale + 500000LL) / 1000000LL -
(lastTimestampUs * mTimeScale + 500000LL) / 1000000LL);
+ CHECK_GE(currDurationTicks, 0ll);
mSampleSizes.push_back(sampleSize);
++mNumSamples;
@@ -2020,25 +2045,12 @@
// Force the first sample to have its own stts entry so that
// we can adjust its value later to maintain the A/V sync.
if (mNumSamples == 3 || currDurationTicks != lastDurationTicks) {
- ALOGV("%s lastDurationUs: %lld us, currDurationTicks: %lld us",
- mIsAudio? "Audio": "Video", lastDurationUs, currDurationTicks);
addOneSttsTableEntry(sampleCount, lastDurationTicks);
sampleCount = 1;
} else {
++sampleCount;
}
- if (!mIsAudio) {
- currCttsDurTicks =
- ((cttsDeltaTimeUs * mTimeScale + 500000LL) / 1000000LL -
- (lastCttsTimeUs * mTimeScale + 500000LL) / 1000000LL);
- if (currCttsDurTicks != lastCttsDurTicks) {
- addOneCttsTableEntry(cttsSampleCount, lastCttsDurTicks);
- cttsSampleCount = 1;
- } else {
- ++cttsSampleCount;
- }
- }
}
if (mSamplesHaveSameSize) {
if (mNumSamples >= 2 && previousSampleSize != sampleSize) {
@@ -2052,11 +2064,6 @@
lastDurationTicks = currDurationTicks;
lastTimestampUs = timestampUs;
- if (!mIsAudio) {
- lastCttsDurTicks = currCttsDurTicks;
- lastCttsTimeUs = cttsDeltaTimeUs;
- }
-
if (isSync != 0) {
addOneStssTableEntry(mNumSamples);
}
@@ -2125,10 +2132,8 @@
if (mNumSamples == 1) {
lastDurationUs = 0; // A single sample's duration
lastDurationTicks = 0;
- lastCttsDurTicks = 0;
} else {
++sampleCount; // Count for the last sample
- ++cttsSampleCount;
}
if (mNumSamples <= 2) {
@@ -2140,7 +2145,14 @@
addOneSttsTableEntry(sampleCount, lastDurationTicks);
}
- addOneCttsTableEntry(cttsSampleCount, lastCttsDurTicks);
+ // The last ctts box may not have been written yet, and this
+ // is to make sure that we write out the last ctts box.
+ if (currCttsOffsetTimeTicks == lastCttsOffsetTimeTicks) {
+ if (cttsSampleCount > 0) {
+ addOneCttsTableEntry(cttsSampleCount, lastCttsOffsetTimeTicks);
+ }
+ }
+
mTrackDurationUs += lastDurationUs;
mReachedEOS = true;
@@ -2406,7 +2418,7 @@
mOwner->writeInt16(0x18); // depth
mOwner->writeInt16(-1); // predefined
- CHECK(23 + mCodecSpecificDataSize < 128);
+ CHECK_LT(23 + mCodecSpecificDataSize, 128);
if (!strcasecmp(MEDIA_MIMETYPE_VIDEO_MPEG4, mime)) {
writeMp4vEsdsBox();
@@ -2465,10 +2477,10 @@
void MPEG4Writer::Track::writeMp4aEsdsBox() {
mOwner->beginBox("esds");
CHECK(mCodecSpecificData);
- CHECK(mCodecSpecificDataSize > 0);
+ CHECK_GT(mCodecSpecificDataSize, 0);
// Make sure all sizes encode to a single byte.
- CHECK(mCodecSpecificDataSize + 23 < 128);
+ CHECK_LT(mCodecSpecificDataSize + 23, 128);
mOwner->writeInt32(0); // version=0, flags=0
mOwner->writeInt8(0x03); // ES_DescrTag
@@ -2502,7 +2514,7 @@
void MPEG4Writer::Track::writeMp4vEsdsBox() {
CHECK(mCodecSpecificData);
- CHECK(mCodecSpecificDataSize > 0);
+ CHECK_GT(mCodecSpecificDataSize, 0);
mOwner->beginBox("esds");
mOwner->writeInt32(0); // version=0, flags=0
@@ -2662,7 +2674,7 @@
void MPEG4Writer::Track::writeAvccBox() {
CHECK(mCodecSpecificData);
- CHECK(mCodecSpecificDataSize >= 5);
+ CHECK_GE(mCodecSpecificDataSize, 5);
// Patch avcc's lengthSize field to match the number
// of bytes we use to indicate the size of a nal unit.
@@ -2690,23 +2702,26 @@
mOwner->endBox(); // pasp
}
+int32_t MPEG4Writer::Track::getStartTimeOffsetScaledTime() const {
+ int64_t trackStartTimeOffsetUs = 0;
+ int64_t moovStartTimeUs = mOwner->getStartTimestampUs();
+ if (mStartTimestampUs != moovStartTimeUs) {
+ CHECK_GT(mStartTimestampUs, moovStartTimeUs);
+ trackStartTimeOffsetUs = mStartTimestampUs - moovStartTimeUs;
+ }
+ return (trackStartTimeOffsetUs * mTimeScale + 500000LL) / 1000000LL;
+}
+
void MPEG4Writer::Track::writeSttsBox() {
mOwner->beginBox("stts");
mOwner->writeInt32(0); // version=0, flags=0
mOwner->writeInt32(mNumSttsTableEntries);
// Compensate for small start time difference from different media tracks
- int64_t trackStartTimeOffsetUs = 0;
- int64_t moovStartTimeUs = mOwner->getStartTimestampUs();
- if (mStartTimestampUs != moovStartTimeUs) {
- CHECK(mStartTimestampUs > moovStartTimeUs);
- trackStartTimeOffsetUs = mStartTimestampUs - moovStartTimeUs;
- }
List<SttsTableEntry>::iterator it = mSttsTableEntries.begin();
CHECK(it != mSttsTableEntries.end() && it->sampleCount == 1);
mOwner->writeInt32(it->sampleCount);
- int32_t dur = (trackStartTimeOffsetUs * mTimeScale + 500000LL) / 1000000LL;
- mOwner->writeInt32(dur + it->sampleDuration);
+ mOwner->writeInt32(getStartTimeOffsetScaledTime() + it->sampleDuration);
int64_t totalCount = 1;
while (++it != mSttsTableEntries.end()) {
@@ -2714,7 +2729,7 @@
mOwner->writeInt32(it->sampleDuration);
totalCount += it->sampleCount;
}
- CHECK(totalCount == mNumSamples);
+ CHECK_EQ(totalCount, mNumSamples);
mOwner->endBox(); // stts
}
@@ -2723,6 +2738,11 @@
return;
}
+ // There is no B frame at all
+ if (mMinCttsOffsetTimeUs == mMaxCttsOffsetTimeUs) {
+ return;
+ }
+
// Do not write ctts box when there is no need to have it.
if ((mNumCttsTableEntries == 1 &&
mCttsTableEntries.begin()->sampleDuration == 0) ||
@@ -2730,24 +2750,29 @@
return;
}
- ALOGV("ctts box has %d entries", mNumCttsTableEntries);
+ ALOGD("ctts box has %d entries with range [%lld, %lld]",
+ mNumCttsTableEntries, mMinCttsOffsetTimeUs, mMaxCttsOffsetTimeUs);
mOwner->beginBox("ctts");
- if (mHasNegativeCttsDeltaDuration) {
- mOwner->writeInt32(0x00010000); // version=1, flags=0
- } else {
- mOwner->writeInt32(0); // version=0, flags=0
- }
+ // Version 1 allows to use negative offset time value, but
+ // we are sticking to version 0 for now.
+ mOwner->writeInt32(0); // version=0, flags=0
mOwner->writeInt32(mNumCttsTableEntries);
- int64_t totalCount = 0;
- for (List<CttsTableEntry>::iterator it = mCttsTableEntries.begin();
- it != mCttsTableEntries.end(); ++it) {
+ // Compensate for small start time difference from different media tracks
+ List<CttsTableEntry>::iterator it = mCttsTableEntries.begin();
+ CHECK(it != mCttsTableEntries.end() && it->sampleCount == 1);
+ mOwner->writeInt32(it->sampleCount);
+ mOwner->writeInt32(getStartTimeOffsetScaledTime() +
+ it->sampleDuration - mMinCttsOffsetTimeUs);
+
+ int64_t totalCount = 1;
+ while (++it != mCttsTableEntries.end()) {
mOwner->writeInt32(it->sampleCount);
- mOwner->writeInt32(it->sampleDuration);
+ mOwner->writeInt32(it->sampleDuration - mMinCttsOffsetTimeUs);
totalCount += it->sampleCount;
}
- CHECK(totalCount == mNumSamples);
+ CHECK_EQ(totalCount, mNumSamples);
mOwner->endBox(); // ctts
}
diff --git a/media/libstagefright/MediaBuffer.cpp b/media/libstagefright/MediaBuffer.cpp
index 96271e4..11b80bf 100644
--- a/media/libstagefright/MediaBuffer.cpp
+++ b/media/libstagefright/MediaBuffer.cpp
@@ -22,8 +22,8 @@
#include <stdlib.h>
#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaBuffer.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MetaData.h>
#include <ui/GraphicBuffer.h>
@@ -157,7 +157,7 @@
}
MediaBuffer::~MediaBuffer() {
- CHECK_EQ(mObserver, NULL);
+ CHECK(mObserver == NULL);
if (mOwnsData && mData != NULL) {
free(mData);
@@ -188,7 +188,7 @@
}
MediaBuffer *MediaBuffer::clone() {
- CHECK_EQ(mGraphicBuffer, NULL);
+ CHECK(mGraphicBuffer == NULL);
MediaBuffer *buffer = new MediaBuffer(mData, mSize);
buffer->set_range(mRangeOffset, mRangeLength);
diff --git a/media/libstagefright/MediaBufferGroup.cpp b/media/libstagefright/MediaBufferGroup.cpp
index c8d05f4..80aae51 100644
--- a/media/libstagefright/MediaBufferGroup.cpp
+++ b/media/libstagefright/MediaBufferGroup.cpp
@@ -17,9 +17,9 @@
#define LOG_TAG "MediaBufferGroup"
#include <utils/Log.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaBuffer.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
namespace android {
diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
new file mode 100644
index 0000000..a9e7f36
--- /dev/null
+++ b/media/libstagefright/MediaCodec.cpp
@@ -0,0 +1,1217 @@
+/*
+ * Copyright 2012, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "MediaCodec"
+#include <utils/Log.h>
+
+#include <media/stagefright/MediaCodec.h>
+
+#include "include/SoftwareRenderer.h"
+
+#include <gui/SurfaceTextureClient.h>
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/ACodec.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/NativeWindowWrapper.h>
+
+namespace android {
+
+// static
+sp<MediaCodec> MediaCodec::CreateByType(
+ const sp<ALooper> &looper, const char *mime, bool encoder) {
+ sp<MediaCodec> codec = new MediaCodec(looper);
+ if (codec->init(mime, true /* nameIsType */, encoder) != OK) {
+ return NULL;
+ }
+
+ return codec;
+}
+
+// static
+sp<MediaCodec> MediaCodec::CreateByComponentName(
+ const sp<ALooper> &looper, const char *name) {
+ sp<MediaCodec> codec = new MediaCodec(looper);
+ if (codec->init(name, false /* nameIsType */, false /* encoder */) != OK) {
+ return NULL;
+ }
+
+ return codec;
+}
+
+MediaCodec::MediaCodec(const sp<ALooper> &looper)
+ : mState(UNINITIALIZED),
+ mLooper(looper),
+ mCodec(new ACodec),
+ mFlags(0),
+ mSoftRenderer(NULL),
+ mDequeueInputTimeoutGeneration(0),
+ mDequeueInputReplyID(0),
+ mDequeueOutputTimeoutGeneration(0),
+ mDequeueOutputReplyID(0) {
+}
+
+MediaCodec::~MediaCodec() {
+ CHECK_EQ(mState, UNINITIALIZED);
+}
+
+// static
+status_t MediaCodec::PostAndAwaitResponse(
+ const sp<AMessage> &msg, sp<AMessage> *response) {
+ status_t err = msg->postAndAwaitResponse(response);
+
+ if (err != OK) {
+ return err;
+ }
+
+ if (!(*response)->findInt32("err", &err)) {
+ err = OK;
+ }
+
+ return err;
+}
+
+status_t MediaCodec::init(const char *name, bool nameIsType, bool encoder) {
+ // Current video decoders do not return from OMX_FillThisBuffer
+ // quickly, violating the OpenMAX specs, until that is remedied
+ // we need to invest in an extra looper to free the main event
+ // queue.
+ bool needDedicatedLooper = false;
+ if (nameIsType && !strncasecmp(name, "video/", 6)) {
+ needDedicatedLooper = true;
+ } else if (!nameIsType && !strncmp(name, "OMX.TI.DUCATI1.VIDEO.", 21)) {
+ needDedicatedLooper = true;
+ }
+
+ if (needDedicatedLooper) {
+ if (mCodecLooper == NULL) {
+ mCodecLooper = new ALooper;
+ mCodecLooper->setName("CodecLooper");
+ mCodecLooper->start(false, false, ANDROID_PRIORITY_AUDIO);
+ }
+
+ mCodecLooper->registerHandler(mCodec);
+ } else {
+ mLooper->registerHandler(mCodec);
+ }
+
+ mLooper->registerHandler(this);
+
+ mCodec->setNotificationMessage(new AMessage(kWhatCodecNotify, id()));
+
+ sp<AMessage> msg = new AMessage(kWhatInit, id());
+ msg->setString("name", name);
+ msg->setInt32("nameIsType", nameIsType);
+
+ if (nameIsType) {
+ msg->setInt32("encoder", encoder);
+ }
+
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t MediaCodec::configure(
+ const sp<AMessage> &format,
+ const sp<SurfaceTextureClient> &nativeWindow,
+ uint32_t flags) {
+ sp<AMessage> msg = new AMessage(kWhatConfigure, id());
+
+ msg->setMessage("format", format);
+ msg->setInt32("flags", flags);
+
+ if (nativeWindow != NULL) {
+ if (!(mFlags & kFlagIsSoftwareCodec)) {
+ msg->setObject(
+ "native-window",
+ new NativeWindowWrapper(nativeWindow));
+ } else {
+ mNativeWindow = nativeWindow;
+ }
+ }
+
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t MediaCodec::start() {
+ sp<AMessage> msg = new AMessage(kWhatStart, id());
+
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t MediaCodec::stop() {
+ sp<AMessage> msg = new AMessage(kWhatStop, id());
+
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t MediaCodec::release() {
+ sp<AMessage> msg = new AMessage(kWhatRelease, id());
+
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t MediaCodec::queueInputBuffer(
+ size_t index,
+ size_t offset,
+ size_t size,
+ int64_t presentationTimeUs,
+ uint32_t flags) {
+ sp<AMessage> msg = new AMessage(kWhatQueueInputBuffer, id());
+ msg->setSize("index", index);
+ msg->setSize("offset", offset);
+ msg->setSize("size", size);
+ msg->setInt64("timeUs", presentationTimeUs);
+ msg->setInt32("flags", flags);
+
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t MediaCodec::dequeueInputBuffer(size_t *index, int64_t timeoutUs) {
+ sp<AMessage> msg = new AMessage(kWhatDequeueInputBuffer, id());
+ msg->setInt64("timeoutUs", timeoutUs);
+
+ sp<AMessage> response;
+ status_t err;
+ if ((err = PostAndAwaitResponse(msg, &response)) != OK) {
+ return err;
+ }
+
+ CHECK(response->findSize("index", index));
+
+ return OK;
+}
+
+status_t MediaCodec::dequeueOutputBuffer(
+ size_t *index,
+ size_t *offset,
+ size_t *size,
+ int64_t *presentationTimeUs,
+ uint32_t *flags,
+ int64_t timeoutUs) {
+ sp<AMessage> msg = new AMessage(kWhatDequeueOutputBuffer, id());
+ msg->setInt64("timeoutUs", timeoutUs);
+
+ sp<AMessage> response;
+ status_t err;
+ if ((err = PostAndAwaitResponse(msg, &response)) != OK) {
+ return err;
+ }
+
+ CHECK(response->findSize("index", index));
+ CHECK(response->findSize("offset", offset));
+ CHECK(response->findSize("size", size));
+ CHECK(response->findInt64("timeUs", presentationTimeUs));
+ CHECK(response->findInt32("flags", (int32_t *)flags));
+
+ return OK;
+}
+
+status_t MediaCodec::renderOutputBufferAndRelease(size_t index) {
+ sp<AMessage> msg = new AMessage(kWhatReleaseOutputBuffer, id());
+ msg->setSize("index", index);
+ msg->setInt32("render", true);
+
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t MediaCodec::releaseOutputBuffer(size_t index) {
+ sp<AMessage> msg = new AMessage(kWhatReleaseOutputBuffer, id());
+ msg->setSize("index", index);
+
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t MediaCodec::getOutputFormat(sp<AMessage> *format) const {
+ sp<AMessage> msg = new AMessage(kWhatGetOutputFormat, id());
+
+ sp<AMessage> response;
+ status_t err;
+ if ((err = PostAndAwaitResponse(msg, &response)) != OK) {
+ return err;
+ }
+
+ CHECK(response->findMessage("format", format));
+
+ return OK;
+}
+
+status_t MediaCodec::getInputBuffers(Vector<sp<ABuffer> > *buffers) const {
+ sp<AMessage> msg = new AMessage(kWhatGetBuffers, id());
+ msg->setInt32("portIndex", kPortIndexInput);
+ msg->setPointer("buffers", buffers);
+
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t MediaCodec::getOutputBuffers(Vector<sp<ABuffer> > *buffers) const {
+ sp<AMessage> msg = new AMessage(kWhatGetBuffers, id());
+ msg->setInt32("portIndex", kPortIndexOutput);
+ msg->setPointer("buffers", buffers);
+
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t MediaCodec::flush() {
+ sp<AMessage> msg = new AMessage(kWhatFlush, id());
+
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+////////////////////////////////////////////////////////////////////////////////
+
+void MediaCodec::cancelPendingDequeueOperations() {
+ if (mFlags & kFlagDequeueInputPending) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", INVALID_OPERATION);
+ response->postReply(mDequeueInputReplyID);
+
+ ++mDequeueInputTimeoutGeneration;
+ mDequeueInputReplyID = 0;
+ mFlags &= ~kFlagDequeueInputPending;
+ }
+
+ if (mFlags & kFlagDequeueOutputPending) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", INVALID_OPERATION);
+ response->postReply(mDequeueOutputReplyID);
+
+ ++mDequeueOutputTimeoutGeneration;
+ mDequeueOutputReplyID = 0;
+ mFlags &= ~kFlagDequeueOutputPending;
+ }
+}
+
+bool MediaCodec::handleDequeueInputBuffer(uint32_t replyID, bool newRequest) {
+ if (mState != STARTED
+ || (mFlags & kFlagStickyError)
+ || (newRequest && (mFlags & kFlagDequeueInputPending))) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", INVALID_OPERATION);
+
+ response->postReply(replyID);
+
+ return true;
+ }
+
+ ssize_t index = dequeuePortBuffer(kPortIndexInput);
+
+ if (index < 0) {
+ CHECK_EQ(index, -EAGAIN);
+ return false;
+ }
+
+ sp<AMessage> response = new AMessage;
+ response->setSize("index", index);
+ response->postReply(replyID);
+
+ return true;
+}
+
+bool MediaCodec::handleDequeueOutputBuffer(uint32_t replyID, bool newRequest) {
+ sp<AMessage> response = new AMessage;
+
+ if (mState != STARTED
+ || (mFlags & kFlagStickyError)
+ || (newRequest && (mFlags & kFlagDequeueOutputPending))) {
+ response->setInt32("err", INVALID_OPERATION);
+ } else if (mFlags & kFlagOutputBuffersChanged) {
+ response->setInt32("err", INFO_OUTPUT_BUFFERS_CHANGED);
+ mFlags &= ~kFlagOutputBuffersChanged;
+ } else if (mFlags & kFlagOutputFormatChanged) {
+ response->setInt32("err", INFO_FORMAT_CHANGED);
+ mFlags &= ~kFlagOutputFormatChanged;
+ } else {
+ ssize_t index = dequeuePortBuffer(kPortIndexOutput);
+
+ if (index < 0) {
+ CHECK_EQ(index, -EAGAIN);
+ return false;
+ }
+
+ const sp<ABuffer> &buffer =
+ mPortBuffers[kPortIndexOutput].itemAt(index).mData;
+
+ response->setSize("index", index);
+ response->setSize("offset", buffer->offset());
+ response->setSize("size", buffer->size());
+
+ int64_t timeUs;
+ CHECK(buffer->meta()->findInt64("timeUs", &timeUs));
+
+ response->setInt64("timeUs", timeUs);
+
+ int32_t omxFlags;
+ CHECK(buffer->meta()->findInt32("omxFlags", &omxFlags));
+
+ uint32_t flags = 0;
+ if (omxFlags & OMX_BUFFERFLAG_SYNCFRAME) {
+ flags |= BUFFER_FLAG_SYNCFRAME;
+ }
+ if (omxFlags & OMX_BUFFERFLAG_CODECCONFIG) {
+ flags |= BUFFER_FLAG_CODECCONFIG;
+ }
+ if (omxFlags & OMX_BUFFERFLAG_EOS) {
+ flags |= BUFFER_FLAG_EOS;
+ }
+
+ response->setInt32("flags", flags);
+ }
+
+ response->postReply(replyID);
+
+ return true;
+}
+
+void MediaCodec::onMessageReceived(const sp<AMessage> &msg) {
+ switch (msg->what()) {
+ case kWhatCodecNotify:
+ {
+ int32_t what;
+ CHECK(msg->findInt32("what", &what));
+
+ switch (what) {
+ case ACodec::kWhatError:
+ {
+ int32_t omxError, internalError;
+ CHECK(msg->findInt32("omx-error", &omxError));
+ CHECK(msg->findInt32("err", &internalError));
+
+ ALOGE("Codec reported an error. "
+ "(omx error 0x%08x, internalError %d)",
+ omxError, internalError);
+
+ bool sendErrorReponse = true;
+
+ switch (mState) {
+ case INITIALIZING:
+ {
+ setState(UNINITIALIZED);
+ break;
+ }
+
+ case CONFIGURING:
+ {
+ setState(INITIALIZED);
+ break;
+ }
+
+ case STARTING:
+ {
+ setState(CONFIGURED);
+ break;
+ }
+
+ case STOPPING:
+ case RELEASING:
+ {
+ // Ignore the error, assuming we'll still get
+ // the shutdown complete notification.
+
+ sendErrorReponse = false;
+ break;
+ }
+
+ case FLUSHING:
+ {
+ setState(STARTED);
+ break;
+ }
+
+ case STARTED:
+ {
+ sendErrorReponse = false;
+
+ mFlags |= kFlagStickyError;
+
+ cancelPendingDequeueOperations();
+ break;
+ }
+
+ default:
+ {
+ sendErrorReponse = false;
+
+ mFlags |= kFlagStickyError;
+ break;
+ }
+ }
+
+ if (sendErrorReponse) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", UNKNOWN_ERROR);
+
+ response->postReply(mReplyID);
+ }
+ break;
+ }
+
+ case ACodec::kWhatComponentAllocated:
+ {
+ CHECK_EQ(mState, INITIALIZING);
+ setState(INITIALIZED);
+
+ AString componentName;
+ CHECK(msg->findString("componentName", &componentName));
+
+ if (componentName.startsWith("OMX.google.")) {
+ mFlags |= kFlagIsSoftwareCodec;
+ } else {
+ mFlags &= ~kFlagIsSoftwareCodec;
+ }
+
+ (new AMessage)->postReply(mReplyID);
+ break;
+ }
+
+ case ACodec::kWhatComponentConfigured:
+ {
+ CHECK_EQ(mState, CONFIGURING);
+ setState(CONFIGURED);
+
+ (new AMessage)->postReply(mReplyID);
+ break;
+ }
+
+ case ACodec::kWhatBuffersAllocated:
+ {
+ int32_t portIndex;
+ CHECK(msg->findInt32("portIndex", &portIndex));
+
+ ALOGV("%s buffers allocated",
+ portIndex == kPortIndexInput ? "input" : "output");
+
+ CHECK(portIndex == kPortIndexInput
+ || portIndex == kPortIndexOutput);
+
+ mPortBuffers[portIndex].clear();
+
+ Vector<BufferInfo> *buffers = &mPortBuffers[portIndex];
+ for (size_t i = 0;; ++i) {
+ AString name = StringPrintf("buffer-id_%d", i);
+
+ void *bufferID;
+ if (!msg->findPointer(name.c_str(), &bufferID)) {
+ break;
+ }
+
+ name = StringPrintf("data_%d", i);
+
+ BufferInfo info;
+ info.mBufferID = bufferID;
+ info.mOwnedByClient = false;
+ CHECK(msg->findBuffer(name.c_str(), &info.mData));
+
+ buffers->push_back(info);
+ }
+
+ if (portIndex == kPortIndexOutput) {
+ if (mState == STARTING) {
+ // We're always allocating output buffers after
+ // allocating input buffers, so this is a good
+ // indication that now all buffers are allocated.
+ setState(STARTED);
+ (new AMessage)->postReply(mReplyID);
+ } else {
+ mFlags |= kFlagOutputBuffersChanged;
+ }
+ }
+ break;
+ }
+
+ case ACodec::kWhatOutputFormatChanged:
+ {
+ ALOGV("codec output format changed");
+
+ if ((mFlags & kFlagIsSoftwareCodec)
+ && mNativeWindow != NULL) {
+ AString mime;
+ CHECK(msg->findString("mime", &mime));
+
+ if (!strncasecmp("video/", mime.c_str(), 6)) {
+ delete mSoftRenderer;
+ mSoftRenderer = NULL;
+
+ int32_t width, height;
+ CHECK(msg->findInt32("width", &width));
+ CHECK(msg->findInt32("height", &height));
+
+ int32_t colorFormat;
+ CHECK(msg->findInt32(
+ "color-format", &colorFormat));
+
+ sp<MetaData> meta = new MetaData;
+ meta->setInt32(kKeyWidth, width);
+ meta->setInt32(kKeyHeight, height);
+ meta->setInt32(kKeyColorFormat, colorFormat);
+
+ mSoftRenderer =
+ new SoftwareRenderer(mNativeWindow, meta);
+ }
+ }
+
+ mOutputFormat = msg;
+ mFlags |= kFlagOutputFormatChanged;
+ break;
+ }
+
+ case ACodec::kWhatFillThisBuffer:
+ {
+ /* size_t index = */updateBuffers(kPortIndexInput, msg);
+
+ if (mState == FLUSHING
+ || mState == STOPPING
+ || mState == RELEASING) {
+ returnBuffersToCodecOnPort(kPortIndexInput);
+ break;
+ }
+
+ if (mFlags & kFlagDequeueInputPending) {
+ CHECK(handleDequeueInputBuffer(mDequeueInputReplyID));
+
+ ++mDequeueInputTimeoutGeneration;
+ mFlags &= ~kFlagDequeueInputPending;
+ mDequeueInputReplyID = 0;
+ }
+ break;
+ }
+
+ case ACodec::kWhatDrainThisBuffer:
+ {
+ /* size_t index = */updateBuffers(kPortIndexOutput, msg);
+
+ if (mState == FLUSHING
+ || mState == STOPPING
+ || mState == RELEASING) {
+ returnBuffersToCodecOnPort(kPortIndexOutput);
+ break;
+ }
+
+ sp<ABuffer> buffer;
+ CHECK(msg->findBuffer("buffer", &buffer));
+
+ int32_t omxFlags;
+ CHECK(msg->findInt32("flags", &omxFlags));
+
+ buffer->meta()->setInt32("omxFlags", omxFlags);
+
+ if (mFlags & kFlagDequeueOutputPending) {
+ CHECK(handleDequeueOutputBuffer(mDequeueOutputReplyID));
+
+ ++mDequeueOutputTimeoutGeneration;
+ mFlags &= ~kFlagDequeueOutputPending;
+ mDequeueOutputReplyID = 0;
+ }
+ break;
+ }
+
+ case ACodec::kWhatEOS:
+ {
+ // We already notify the client of this by using the
+ // corresponding flag in "onOutputBufferReady".
+ break;
+ }
+
+ case ACodec::kWhatShutdownCompleted:
+ {
+ if (mState == STOPPING) {
+ setState(INITIALIZED);
+ } else {
+ CHECK_EQ(mState, RELEASING);
+ setState(UNINITIALIZED);
+ }
+
+ (new AMessage)->postReply(mReplyID);
+ break;
+ }
+
+ case ACodec::kWhatFlushCompleted:
+ {
+ CHECK_EQ(mState, FLUSHING);
+ setState(STARTED);
+
+ mCodec->signalResume();
+
+ (new AMessage)->postReply(mReplyID);
+ break;
+ }
+
+ default:
+ TRESPASS();
+ }
+ break;
+ }
+
+ case kWhatInit:
+ {
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ if (mState != UNINITIALIZED) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", INVALID_OPERATION);
+
+ response->postReply(replyID);
+ break;
+ }
+
+ mReplyID = replyID;
+ setState(INITIALIZING);
+
+ AString name;
+ CHECK(msg->findString("name", &name));
+
+ int32_t nameIsType;
+ int32_t encoder = false;
+ CHECK(msg->findInt32("nameIsType", &nameIsType));
+ if (nameIsType) {
+ CHECK(msg->findInt32("encoder", &encoder));
+ }
+
+ sp<AMessage> format = new AMessage;
+
+ if (nameIsType) {
+ format->setString("mime", name.c_str());
+ format->setInt32("encoder", encoder);
+ } else {
+ format->setString("componentName", name.c_str());
+ }
+
+ mCodec->initiateAllocateComponent(format);
+ break;
+ }
+
+ case kWhatConfigure:
+ {
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ if (mState != INITIALIZED) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", INVALID_OPERATION);
+
+ response->postReply(replyID);
+ break;
+ }
+
+ mReplyID = replyID;
+ setState(CONFIGURING);
+
+ sp<RefBase> obj;
+ if (!msg->findObject("native-window", &obj)) {
+ obj.clear();
+ }
+
+ sp<AMessage> format;
+ CHECK(msg->findMessage("format", &format));
+
+ if (obj != NULL) {
+ format->setObject("native-window", obj);
+ }
+
+ uint32_t flags;
+ CHECK(msg->findInt32("flags", (int32_t *)&flags));
+
+ if (flags & CONFIGURE_FLAG_ENCODE) {
+ format->setInt32("encoder", true);
+ }
+
+ mCodec->initiateConfigureComponent(format);
+ break;
+ }
+
+ case kWhatStart:
+ {
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ if (mState != CONFIGURED) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", INVALID_OPERATION);
+
+ response->postReply(replyID);
+ break;
+ }
+
+ mReplyID = replyID;
+ setState(STARTING);
+
+ mCodec->initiateStart();
+ break;
+ }
+
+ case kWhatStop:
+ {
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ if (mState != INITIALIZED
+ && mState != CONFIGURED && mState != STARTED) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", INVALID_OPERATION);
+
+ response->postReply(replyID);
+ break;
+ }
+
+ mReplyID = replyID;
+ setState(STOPPING);
+
+ mCodec->initiateShutdown(true /* keepComponentAllocated */);
+ returnBuffersToCodec();
+ break;
+ }
+
+ case kWhatRelease:
+ {
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ if (mState != INITIALIZED
+ && mState != CONFIGURED && mState != STARTED) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", INVALID_OPERATION);
+
+ response->postReply(replyID);
+ break;
+ }
+
+ mReplyID = replyID;
+ setState(RELEASING);
+
+ mCodec->initiateShutdown();
+ returnBuffersToCodec();
+ break;
+ }
+
+ case kWhatDequeueInputBuffer:
+ {
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ if (handleDequeueInputBuffer(replyID, true /* new request */)) {
+ break;
+ }
+
+ int64_t timeoutUs;
+ CHECK(msg->findInt64("timeoutUs", &timeoutUs));
+
+ if (timeoutUs == 0ll) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", -EAGAIN);
+ response->postReply(replyID);
+ break;
+ }
+
+ mFlags |= kFlagDequeueInputPending;
+ mDequeueInputReplyID = replyID;
+
+ if (timeoutUs > 0ll) {
+ sp<AMessage> timeoutMsg =
+ new AMessage(kWhatDequeueInputTimedOut, id());
+ timeoutMsg->setInt32(
+ "generation", ++mDequeueInputTimeoutGeneration);
+ timeoutMsg->post(timeoutUs);
+ }
+ break;
+ }
+
+ case kWhatDequeueInputTimedOut:
+ {
+ int32_t generation;
+ CHECK(msg->findInt32("generation", &generation));
+
+ if (generation != mDequeueInputTimeoutGeneration) {
+ // Obsolete
+ break;
+ }
+
+ CHECK(mFlags & kFlagDequeueInputPending);
+
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", -EAGAIN);
+ response->postReply(mDequeueInputReplyID);
+
+ mFlags &= ~kFlagDequeueInputPending;
+ mDequeueInputReplyID = 0;
+ break;
+ }
+
+ case kWhatQueueInputBuffer:
+ {
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ if (mState != STARTED || (mFlags & kFlagStickyError)) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", INVALID_OPERATION);
+
+ response->postReply(replyID);
+ break;
+ }
+
+ status_t err = onQueueInputBuffer(msg);
+
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", err);
+ response->postReply(replyID);
+ break;
+ }
+
+ case kWhatDequeueOutputBuffer:
+ {
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ if (handleDequeueOutputBuffer(replyID, true /* new request */)) {
+ break;
+ }
+
+ int64_t timeoutUs;
+ CHECK(msg->findInt64("timeoutUs", &timeoutUs));
+
+ if (timeoutUs == 0ll) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", -EAGAIN);
+ response->postReply(replyID);
+ break;
+ }
+
+ mFlags |= kFlagDequeueOutputPending;
+ mDequeueOutputReplyID = replyID;
+
+ if (timeoutUs > 0ll) {
+ sp<AMessage> timeoutMsg =
+ new AMessage(kWhatDequeueOutputTimedOut, id());
+ timeoutMsg->setInt32(
+ "generation", ++mDequeueOutputTimeoutGeneration);
+ timeoutMsg->post(timeoutUs);
+ }
+ break;
+ }
+
+ case kWhatDequeueOutputTimedOut:
+ {
+ int32_t generation;
+ CHECK(msg->findInt32("generation", &generation));
+
+ if (generation != mDequeueOutputTimeoutGeneration) {
+ // Obsolete
+ break;
+ }
+
+ CHECK(mFlags & kFlagDequeueOutputPending);
+
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", -EAGAIN);
+ response->postReply(mDequeueOutputReplyID);
+
+ mFlags &= ~kFlagDequeueOutputPending;
+ mDequeueOutputReplyID = 0;
+ break;
+ }
+
+ case kWhatReleaseOutputBuffer:
+ {
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ if (mState != STARTED || (mFlags & kFlagStickyError)) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", INVALID_OPERATION);
+
+ response->postReply(replyID);
+ break;
+ }
+
+ status_t err = onReleaseOutputBuffer(msg);
+
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", err);
+ response->postReply(replyID);
+ break;
+ }
+
+ case kWhatGetBuffers:
+ {
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ if (mState != STARTED || (mFlags & kFlagStickyError)) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", INVALID_OPERATION);
+
+ response->postReply(replyID);
+ break;
+ }
+
+ int32_t portIndex;
+ CHECK(msg->findInt32("portIndex", &portIndex));
+
+ Vector<sp<ABuffer> > *dstBuffers;
+ CHECK(msg->findPointer("buffers", (void **)&dstBuffers));
+
+ dstBuffers->clear();
+ const Vector<BufferInfo> &srcBuffers = mPortBuffers[portIndex];
+
+ for (size_t i = 0; i < srcBuffers.size(); ++i) {
+ const BufferInfo &info = srcBuffers.itemAt(i);
+
+ dstBuffers->push_back(info.mData);
+ }
+
+ (new AMessage)->postReply(replyID);
+ break;
+ }
+
+ case kWhatFlush:
+ {
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ if (mState != STARTED || (mFlags & kFlagStickyError)) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", INVALID_OPERATION);
+
+ response->postReply(replyID);
+ break;
+ }
+
+ mReplyID = replyID;
+ setState(FLUSHING);
+
+ mCodec->signalFlush();
+ returnBuffersToCodec();
+ break;
+ }
+
+ case kWhatGetOutputFormat:
+ {
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ if ((mState != STARTED && mState != FLUSHING)
+ || (mFlags & kFlagStickyError)) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", INVALID_OPERATION);
+
+ response->postReply(replyID);
+ break;
+ }
+
+ sp<AMessage> response = new AMessage;
+ response->setMessage("format", mOutputFormat);
+ response->postReply(replyID);
+ break;
+ }
+
+ default:
+ TRESPASS();
+ }
+}
+
+void MediaCodec::setState(State newState) {
+ if (newState == UNINITIALIZED) {
+ delete mSoftRenderer;
+ mSoftRenderer = NULL;
+
+ mNativeWindow.clear();
+
+ mOutputFormat.clear();
+ mFlags &= ~kFlagOutputFormatChanged;
+ mFlags &= ~kFlagOutputBuffersChanged;
+ mFlags &= ~kFlagStickyError;
+ }
+
+ mState = newState;
+
+ cancelPendingDequeueOperations();
+}
+
+void MediaCodec::returnBuffersToCodec() {
+ returnBuffersToCodecOnPort(kPortIndexInput);
+ returnBuffersToCodecOnPort(kPortIndexOutput);
+}
+
+void MediaCodec::returnBuffersToCodecOnPort(int32_t portIndex) {
+ CHECK(portIndex == kPortIndexInput || portIndex == kPortIndexOutput);
+
+ Vector<BufferInfo> *buffers = &mPortBuffers[portIndex];
+
+ for (size_t i = 0; i < buffers->size(); ++i) {
+ BufferInfo *info = &buffers->editItemAt(i);
+
+ if (info->mNotify != NULL) {
+ sp<AMessage> msg = info->mNotify;
+ info->mNotify = NULL;
+ info->mOwnedByClient = false;
+
+ if (portIndex == kPortIndexInput) {
+ msg->setInt32("err", ERROR_END_OF_STREAM);
+ }
+ msg->post();
+ }
+ }
+
+ mAvailPortBuffers[portIndex].clear();
+}
+
+size_t MediaCodec::updateBuffers(
+ int32_t portIndex, const sp<AMessage> &msg) {
+ CHECK(portIndex == kPortIndexInput || portIndex == kPortIndexOutput);
+
+ void *bufferID;
+ CHECK(msg->findPointer("buffer-id", &bufferID));
+
+ Vector<BufferInfo> *buffers = &mPortBuffers[portIndex];
+
+ for (size_t i = 0; i < buffers->size(); ++i) {
+ BufferInfo *info = &buffers->editItemAt(i);
+
+ if (info->mBufferID == bufferID) {
+ CHECK(info->mNotify == NULL);
+ CHECK(msg->findMessage("reply", &info->mNotify));
+
+ mAvailPortBuffers[portIndex].push_back(i);
+
+ return i;
+ }
+ }
+
+ TRESPASS();
+
+ return 0;
+}
+
+status_t MediaCodec::onQueueInputBuffer(const sp<AMessage> &msg) {
+ size_t index;
+ size_t offset;
+ size_t size;
+ int64_t timeUs;
+ uint32_t flags;
+ CHECK(msg->findSize("index", &index));
+ CHECK(msg->findSize("offset", &offset));
+ CHECK(msg->findSize("size", &size));
+ CHECK(msg->findInt64("timeUs", &timeUs));
+ CHECK(msg->findInt32("flags", (int32_t *)&flags));
+
+ if (index >= mPortBuffers[kPortIndexInput].size()) {
+ return -ERANGE;
+ }
+
+ BufferInfo *info = &mPortBuffers[kPortIndexInput].editItemAt(index);
+
+ if (info->mNotify == NULL || !info->mOwnedByClient) {
+ return -EACCES;
+ }
+
+ if (offset + size > info->mData->capacity()) {
+ return -EINVAL;
+ }
+
+ sp<AMessage> reply = info->mNotify;
+ info->mNotify = NULL;
+ info->mOwnedByClient = false;
+
+ info->mData->setRange(offset, size);
+ info->mData->meta()->setInt64("timeUs", timeUs);
+
+ if (flags & BUFFER_FLAG_EOS) {
+ info->mData->meta()->setInt32("eos", true);
+ }
+
+ if (flags & BUFFER_FLAG_CODECCONFIG) {
+ info->mData->meta()->setInt32("csd", true);
+ }
+
+ reply->setBuffer("buffer", info->mData);
+ reply->post();
+
+ return OK;
+}
+
+status_t MediaCodec::onReleaseOutputBuffer(const sp<AMessage> &msg) {
+ size_t index;
+ CHECK(msg->findSize("index", &index));
+
+ int32_t render;
+ if (!msg->findInt32("render", &render)) {
+ render = 0;
+ }
+
+ if (mState != STARTED) {
+ return -EINVAL;
+ }
+
+ if (index >= mPortBuffers[kPortIndexOutput].size()) {
+ return -ERANGE;
+ }
+
+ BufferInfo *info = &mPortBuffers[kPortIndexOutput].editItemAt(index);
+
+ if (info->mNotify == NULL || !info->mOwnedByClient) {
+ return -EACCES;
+ }
+
+ if (render) {
+ info->mNotify->setInt32("render", true);
+
+ if (mSoftRenderer != NULL) {
+ mSoftRenderer->render(
+ info->mData->data(), info->mData->size(), NULL);
+ }
+ }
+
+ info->mNotify->post();
+ info->mNotify = NULL;
+ info->mOwnedByClient = false;
+
+ return OK;
+}
+
+ssize_t MediaCodec::dequeuePortBuffer(int32_t portIndex) {
+ CHECK(portIndex == kPortIndexInput || portIndex == kPortIndexOutput);
+
+ List<size_t> *availBuffers = &mAvailPortBuffers[portIndex];
+
+ if (availBuffers->empty()) {
+ return -EAGAIN;
+ }
+
+ size_t index = *availBuffers->begin();
+ availBuffers->erase(availBuffers->begin());
+
+ BufferInfo *info = &mPortBuffers[portIndex].editItemAt(index);
+ CHECK(!info->mOwnedByClient);
+ info->mOwnedByClient = true;
+
+ return index;
+}
+
+} // namespace android
diff --git a/media/libstagefright/MediaCodecList.cpp b/media/libstagefright/MediaCodecList.cpp
new file mode 100644
index 0000000..6b64e21
--- /dev/null
+++ b/media/libstagefright/MediaCodecList.cpp
@@ -0,0 +1,475 @@
+/*
+ * Copyright 2012, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "MediaCodecList"
+#include <utils/Log.h>
+
+#include <media/stagefright/MediaCodecList.h>
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/MediaErrors.h>
+#include <utils/threads.h>
+
+#include <expat.h>
+
+namespace android {
+
+static Mutex sInitMutex;
+
+// static
+MediaCodecList *MediaCodecList::sCodecList;
+
+// static
+const MediaCodecList *MediaCodecList::getInstance() {
+ Mutex::Autolock autoLock(sInitMutex);
+
+ if (sCodecList == NULL) {
+ sCodecList = new MediaCodecList;
+ }
+
+ return sCodecList->initCheck() == OK ? sCodecList : NULL;
+}
+
+MediaCodecList::MediaCodecList()
+ : mInitCheck(NO_INIT) {
+ FILE *file = fopen("/etc/media_codecs.xml", "r");
+
+ if (file == NULL) {
+ ALOGW("unable to open media codecs configuration xml file.");
+ return;
+ }
+
+ parseXMLFile(file);
+
+ if (mInitCheck == OK) {
+ // These are currently still used by the video editing suite.
+
+ addMediaCodec(true /* encoder */, "AACEncoder", "audio/mp4a-latm");
+ addMediaCodec(true /* encoder */, "AVCEncoder", "video/avc");
+
+ addMediaCodec(true /* encoder */, "M4vH263Encoder");
+ addType("video/3gpp");
+ addType("video/mp4v-es");
+ }
+
+#if 0
+ for (size_t i = 0; i < mCodecInfos.size(); ++i) {
+ const CodecInfo &info = mCodecInfos.itemAt(i);
+
+ AString line = info.mName;
+ line.append(" supports ");
+ for (size_t j = 0; j < mTypes.size(); ++j) {
+ uint32_t value = mTypes.valueAt(j);
+
+ if (info.mTypes & (1ul << value)) {
+ line.append(mTypes.keyAt(j));
+ line.append(" ");
+ }
+ }
+
+ ALOGI("%s", line.c_str());
+ }
+#endif
+
+ fclose(file);
+ file = NULL;
+}
+
+MediaCodecList::~MediaCodecList() {
+}
+
+status_t MediaCodecList::initCheck() const {
+ return mInitCheck;
+}
+
+void MediaCodecList::parseXMLFile(FILE *file) {
+ mInitCheck = OK;
+ mCurrentSection = SECTION_TOPLEVEL;
+ mDepth = 0;
+
+ XML_Parser parser = ::XML_ParserCreate(NULL);
+ CHECK(parser != NULL);
+
+ ::XML_SetUserData(parser, this);
+ ::XML_SetElementHandler(
+ parser, StartElementHandlerWrapper, EndElementHandlerWrapper);
+
+ const int BUFF_SIZE = 512;
+ while (mInitCheck == OK) {
+ void *buff = ::XML_GetBuffer(parser, BUFF_SIZE);
+ if (buff == NULL) {
+ ALOGE("failed to in call to XML_GetBuffer()");
+ mInitCheck = UNKNOWN_ERROR;
+ break;
+ }
+
+ int bytes_read = ::fread(buff, 1, BUFF_SIZE, file);
+ if (bytes_read < 0) {
+ ALOGE("failed in call to read");
+ mInitCheck = ERROR_IO;
+ break;
+ }
+
+ if (::XML_ParseBuffer(parser, bytes_read, bytes_read == 0)
+ != XML_STATUS_OK) {
+ mInitCheck = ERROR_MALFORMED;
+ break;
+ }
+
+ if (bytes_read == 0) {
+ break;
+ }
+ }
+
+ ::XML_ParserFree(parser);
+
+ if (mInitCheck == OK) {
+ for (size_t i = mCodecInfos.size(); i-- > 0;) {
+ CodecInfo *info = &mCodecInfos.editItemAt(i);
+
+ if (info->mTypes == 0) {
+ // No types supported by this component???
+
+ ALOGW("Component %s does not support any type of media?",
+ info->mName.c_str());
+
+ mCodecInfos.removeAt(i);
+ }
+ }
+ }
+
+ if (mInitCheck != OK) {
+ mCodecInfos.clear();
+ mCodecQuirks.clear();
+ }
+}
+
+// static
+void MediaCodecList::StartElementHandlerWrapper(
+ void *me, const char *name, const char **attrs) {
+ static_cast<MediaCodecList *>(me)->startElementHandler(name, attrs);
+}
+
+// static
+void MediaCodecList::EndElementHandlerWrapper(void *me, const char *name) {
+ static_cast<MediaCodecList *>(me)->endElementHandler(name);
+}
+
+void MediaCodecList::startElementHandler(
+ const char *name, const char **attrs) {
+ if (mInitCheck != OK) {
+ return;
+ }
+
+ switch (mCurrentSection) {
+ case SECTION_TOPLEVEL:
+ {
+ if (!strcmp(name, "Decoders")) {
+ mCurrentSection = SECTION_DECODERS;
+ } else if (!strcmp(name, "Encoders")) {
+ mCurrentSection = SECTION_ENCODERS;
+ }
+ break;
+ }
+
+ case SECTION_DECODERS:
+ {
+ if (!strcmp(name, "MediaCodec")) {
+ mInitCheck =
+ addMediaCodecFromAttributes(false /* encoder */, attrs);
+
+ mCurrentSection = SECTION_DECODER;
+ }
+ break;
+ }
+
+ case SECTION_ENCODERS:
+ {
+ if (!strcmp(name, "MediaCodec")) {
+ mInitCheck =
+ addMediaCodecFromAttributes(true /* encoder */, attrs);
+
+ mCurrentSection = SECTION_ENCODER;
+ }
+ break;
+ }
+
+ case SECTION_DECODER:
+ case SECTION_ENCODER:
+ {
+ if (!strcmp(name, "Quirk")) {
+ mInitCheck = addQuirk(attrs);
+ } else if (!strcmp(name, "Type")) {
+ mInitCheck = addTypeFromAttributes(attrs);
+ }
+ break;
+ }
+
+ default:
+ break;
+ }
+
+ ++mDepth;
+}
+
+void MediaCodecList::endElementHandler(const char *name) {
+ if (mInitCheck != OK) {
+ return;
+ }
+
+ switch (mCurrentSection) {
+ case SECTION_DECODERS:
+ {
+ if (!strcmp(name, "Decoders")) {
+ mCurrentSection = SECTION_TOPLEVEL;
+ }
+ break;
+ }
+
+ case SECTION_ENCODERS:
+ {
+ if (!strcmp(name, "Encoders")) {
+ mCurrentSection = SECTION_TOPLEVEL;
+ }
+ break;
+ }
+
+ case SECTION_DECODER:
+ {
+ if (!strcmp(name, "MediaCodec")) {
+ mCurrentSection = SECTION_DECODERS;
+ }
+ break;
+ }
+
+ case SECTION_ENCODER:
+ {
+ if (!strcmp(name, "MediaCodec")) {
+ mCurrentSection = SECTION_ENCODERS;
+ }
+ break;
+ }
+
+ default:
+ break;
+ }
+
+ --mDepth;
+}
+
+status_t MediaCodecList::addMediaCodecFromAttributes(
+ bool encoder, const char **attrs) {
+ const char *name = NULL;
+ const char *type = NULL;
+
+ size_t i = 0;
+ while (attrs[i] != NULL) {
+ if (!strcmp(attrs[i], "name")) {
+ if (attrs[i + 1] == NULL) {
+ return -EINVAL;
+ }
+ name = attrs[i + 1];
+ ++i;
+ } else if (!strcmp(attrs[i], "type")) {
+ if (attrs[i + 1] == NULL) {
+ return -EINVAL;
+ }
+ type = attrs[i + 1];
+ ++i;
+ } else {
+ return -EINVAL;
+ }
+
+ ++i;
+ }
+
+ if (name == NULL) {
+ return -EINVAL;
+ }
+
+ addMediaCodec(encoder, name, type);
+
+ return OK;
+}
+
+void MediaCodecList::addMediaCodec(
+ bool encoder, const char *name, const char *type) {
+ mCodecInfos.push();
+ CodecInfo *info = &mCodecInfos.editItemAt(mCodecInfos.size() - 1);
+ info->mName = name;
+ info->mIsEncoder = encoder;
+ info->mTypes = 0;
+ info->mQuirks = 0;
+
+ if (type != NULL) {
+ addType(type);
+ }
+}
+
+status_t MediaCodecList::addQuirk(const char **attrs) {
+ const char *name = NULL;
+
+ size_t i = 0;
+ while (attrs[i] != NULL) {
+ if (!strcmp(attrs[i], "name")) {
+ if (attrs[i + 1] == NULL) {
+ return -EINVAL;
+ }
+ name = attrs[i + 1];
+ ++i;
+ } else {
+ return -EINVAL;
+ }
+
+ ++i;
+ }
+
+ if (name == NULL) {
+ return -EINVAL;
+ }
+
+ uint32_t bit;
+ ssize_t index = mCodecQuirks.indexOfKey(name);
+ if (index < 0) {
+ bit = mCodecQuirks.size();
+
+ if (bit == 32) {
+ ALOGW("Too many distinct quirk names in configuration.");
+ return OK;
+ }
+
+ mCodecQuirks.add(name, bit);
+ } else {
+ bit = mCodecQuirks.valueAt(index);
+ }
+
+ CodecInfo *info = &mCodecInfos.editItemAt(mCodecInfos.size() - 1);
+ info->mQuirks |= 1ul << bit;
+
+ return OK;
+}
+
+status_t MediaCodecList::addTypeFromAttributes(const char **attrs) {
+ const char *name = NULL;
+
+ size_t i = 0;
+ while (attrs[i] != NULL) {
+ if (!strcmp(attrs[i], "name")) {
+ if (attrs[i + 1] == NULL) {
+ return -EINVAL;
+ }
+ name = attrs[i + 1];
+ ++i;
+ } else {
+ return -EINVAL;
+ }
+
+ ++i;
+ }
+
+ if (name == NULL) {
+ return -EINVAL;
+ }
+
+ addType(name);
+
+ return OK;
+}
+
+void MediaCodecList::addType(const char *name) {
+ uint32_t bit;
+ ssize_t index = mTypes.indexOfKey(name);
+ if (index < 0) {
+ bit = mTypes.size();
+
+ if (bit == 32) {
+ ALOGW("Too many distinct type names in configuration.");
+ return;
+ }
+
+ mTypes.add(name, bit);
+ } else {
+ bit = mTypes.valueAt(index);
+ }
+
+ CodecInfo *info = &mCodecInfos.editItemAt(mCodecInfos.size() - 1);
+ info->mTypes |= 1ul << bit;
+}
+
+ssize_t MediaCodecList::findCodecByType(
+ const char *type, bool encoder, size_t startIndex) const {
+ ssize_t typeIndex = mTypes.indexOfKey(type);
+
+ if (typeIndex < 0) {
+ return -ENOENT;
+ }
+
+ uint32_t typeMask = 1ul << mTypes.valueAt(typeIndex);
+
+ while (startIndex < mCodecInfos.size()) {
+ const CodecInfo &info = mCodecInfos.itemAt(startIndex);
+
+ if (info.mIsEncoder == encoder && (info.mTypes & typeMask)) {
+ return startIndex;
+ }
+
+ ++startIndex;
+ }
+
+ return -ENOENT;
+}
+
+ssize_t MediaCodecList::findCodecByName(const char *name) const {
+ for (size_t i = 0; i < mCodecInfos.size(); ++i) {
+ const CodecInfo &info = mCodecInfos.itemAt(i);
+
+ if (info.mName == name) {
+ return i;
+ }
+ }
+
+ return -ENOENT;
+}
+
+const char *MediaCodecList::getCodecName(size_t index) const {
+ if (index >= mCodecInfos.size()) {
+ return NULL;
+ }
+
+ const CodecInfo &info = mCodecInfos.itemAt(index);
+ return info.mName.c_str();
+}
+
+bool MediaCodecList::codecHasQuirk(
+ size_t index, const char *quirkName) const {
+ if (index >= mCodecInfos.size()) {
+ return NULL;
+ }
+
+ const CodecInfo &info = mCodecInfos.itemAt(index);
+
+ if (info.mQuirks != 0) {
+ ssize_t index = mCodecQuirks.indexOfKey(quirkName);
+ if (index >= 0 && info.mQuirks & (1ul << mCodecQuirks.valueAt(index))) {
+ return true;
+ }
+ }
+
+ return false;
+}
+
+} // namespace android
diff --git a/media/libstagefright/MediaDefs.cpp b/media/libstagefright/MediaDefs.cpp
index 444e823..2549de6 100644
--- a/media/libstagefright/MediaDefs.cpp
+++ b/media/libstagefright/MediaDefs.cpp
@@ -52,5 +52,6 @@
const char *MEDIA_MIMETYPE_CONTAINER_WVM = "video/wvm";
const char *MEDIA_MIMETYPE_TEXT_3GPP = "text/3gpp-tt";
+const char *MEDIA_MIMETYPE_TEXT_SUBRIP = "application/x-subrip";
} // namespace android
diff --git a/media/libstagefright/MediaExtractor.cpp b/media/libstagefright/MediaExtractor.cpp
index 7b17d65..2171492 100644
--- a/media/libstagefright/MediaExtractor.cpp
+++ b/media/libstagefright/MediaExtractor.cpp
@@ -19,6 +19,7 @@
#include <utils/Log.h>
#include "include/AMRExtractor.h"
+#include "include/AVIExtractor.h"
#include "include/MP3Extractor.h"
#include "include/MPEG4Extractor.h"
#include "include/WAVExtractor.h"
@@ -109,10 +110,12 @@
ret = new MatroskaExtractor(source);
} else if (!strcasecmp(mime, MEDIA_MIMETYPE_CONTAINER_MPEG2TS)) {
ret = new MPEG2TSExtractor(source);
+ } else if (!strcasecmp(mime, MEDIA_MIMETYPE_CONTAINER_AVI)) {
+ ret = new AVIExtractor(source);
} else if (!strcasecmp(mime, MEDIA_MIMETYPE_CONTAINER_WVM)) {
ret = new WVMExtractor(source);
} else if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_AAC_ADTS)) {
- ret = new AACExtractor(source);
+ ret = new AACExtractor(source, meta);
} else if (!strcasecmp(mime, MEDIA_MIMETYPE_CONTAINER_MPEG2PS)) {
ret = new MPEG2PSExtractor(source);
}
diff --git a/media/libstagefright/MediaSourceSplitter.cpp b/media/libstagefright/MediaSourceSplitter.cpp
deleted file mode 100644
index 8af0694..0000000
--- a/media/libstagefright/MediaSourceSplitter.cpp
+++ /dev/null
@@ -1,234 +0,0 @@
-/*
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "MediaSourceSplitter"
-#include <utils/Log.h>
-
-#include <media/stagefright/MediaSourceSplitter.h>
-#include <media/stagefright/MediaDebug.h>
-#include <media/stagefright/MediaBuffer.h>
-#include <media/stagefright/MetaData.h>
-
-namespace android {
-
-MediaSourceSplitter::MediaSourceSplitter(sp<MediaSource> mediaSource) {
- mNumberOfClients = 0;
- mSource = mediaSource;
- mSourceStarted = false;
-
- mNumberOfClientsStarted = 0;
- mNumberOfCurrentReads = 0;
- mCurrentReadBit = 0;
- mLastReadCompleted = true;
-}
-
-MediaSourceSplitter::~MediaSourceSplitter() {
-}
-
-sp<MediaSource> MediaSourceSplitter::createClient() {
- Mutex::Autolock autoLock(mLock);
-
- sp<MediaSource> client = new Client(this, mNumberOfClients++);
- mClientsStarted.push(false);
- mClientsDesiredReadBit.push(0);
- return client;
-}
-
-status_t MediaSourceSplitter::start(int clientId, MetaData *params) {
- Mutex::Autolock autoLock(mLock);
-
- ALOGV("start client (%d)", clientId);
- if (mClientsStarted[clientId]) {
- return OK;
- }
-
- mNumberOfClientsStarted++;
-
- if (!mSourceStarted) {
- ALOGV("Starting real source from client (%d)", clientId);
- status_t err = mSource->start(params);
-
- if (err == OK) {
- mSourceStarted = true;
- mClientsStarted.editItemAt(clientId) = true;
- mClientsDesiredReadBit.editItemAt(clientId) = !mCurrentReadBit;
- }
-
- return err;
- } else {
- mClientsStarted.editItemAt(clientId) = true;
- if (mLastReadCompleted) {
- // Last read was completed. So join in the threads for the next read.
- mClientsDesiredReadBit.editItemAt(clientId) = !mCurrentReadBit;
- } else {
- // Last read is ongoing. So join in the threads for the current read.
- mClientsDesiredReadBit.editItemAt(clientId) = mCurrentReadBit;
- }
- return OK;
- }
-}
-
-status_t MediaSourceSplitter::stop(int clientId) {
- Mutex::Autolock autoLock(mLock);
-
- ALOGV("stop client (%d)", clientId);
- CHECK(clientId >= 0 && clientId < mNumberOfClients);
- CHECK(mClientsStarted[clientId]);
-
- if (--mNumberOfClientsStarted == 0) {
- ALOGV("Stopping real source from client (%d)", clientId);
- status_t err = mSource->stop();
- mSourceStarted = false;
- mClientsStarted.editItemAt(clientId) = false;
- return err;
- } else {
- mClientsStarted.editItemAt(clientId) = false;
- if (!mLastReadCompleted && (mClientsDesiredReadBit[clientId] == mCurrentReadBit)) {
- // !mLastReadCompleted implies that buffer has been read from source, but all
- // clients haven't read it.
- // mClientsDesiredReadBit[clientId] == mCurrentReadBit implies that this
- // client would have wanted to read from this buffer. (i.e. it has not yet
- // called read() for the current read buffer.)
- // Since other threads may be waiting for all the clients' reads to complete,
- // signal that this read has been aborted.
- signalReadComplete_lock(true);
- }
- return OK;
- }
-}
-
-sp<MetaData> MediaSourceSplitter::getFormat(int clientId) {
- Mutex::Autolock autoLock(mLock);
-
- ALOGV("getFormat client (%d)", clientId);
- return mSource->getFormat();
-}
-
-status_t MediaSourceSplitter::read(int clientId,
- MediaBuffer **buffer, const MediaSource::ReadOptions *options) {
- Mutex::Autolock autoLock(mLock);
-
- CHECK(clientId >= 0 && clientId < mNumberOfClients);
-
- ALOGV("read client (%d)", clientId);
- *buffer = NULL;
-
- if (!mClientsStarted[clientId]) {
- return OK;
- }
-
- if (mCurrentReadBit != mClientsDesiredReadBit[clientId]) {
- // Desired buffer has not been read from source yet.
-
- // If the current client is the special client with clientId = 0
- // then read from source, else wait until the client 0 has finished
- // reading from source.
- if (clientId == 0) {
- // Wait for all client's last read to complete first so as to not
- // corrupt the buffer at mLastReadMediaBuffer.
- waitForAllClientsLastRead_lock(clientId);
-
- readFromSource_lock(options);
- *buffer = mLastReadMediaBuffer;
- } else {
- waitForReadFromSource_lock(clientId);
-
- *buffer = mLastReadMediaBuffer;
- (*buffer)->add_ref();
- }
- CHECK(mCurrentReadBit == mClientsDesiredReadBit[clientId]);
- } else {
- // Desired buffer has already been read from source. Use the cached data.
- CHECK(clientId != 0);
-
- *buffer = mLastReadMediaBuffer;
- (*buffer)->add_ref();
- }
-
- mClientsDesiredReadBit.editItemAt(clientId) = !mClientsDesiredReadBit[clientId];
- signalReadComplete_lock(false);
-
- return mLastReadStatus;
-}
-
-void MediaSourceSplitter::readFromSource_lock(const MediaSource::ReadOptions *options) {
- mLastReadStatus = mSource->read(&mLastReadMediaBuffer , options);
-
- mCurrentReadBit = !mCurrentReadBit;
- mLastReadCompleted = false;
- mReadFromSourceCondition.broadcast();
-}
-
-void MediaSourceSplitter::waitForReadFromSource_lock(int32_t clientId) {
- mReadFromSourceCondition.wait(mLock);
-}
-
-void MediaSourceSplitter::waitForAllClientsLastRead_lock(int32_t clientId) {
- if (mLastReadCompleted) {
- return;
- }
- mAllReadsCompleteCondition.wait(mLock);
- CHECK(mLastReadCompleted);
-}
-
-void MediaSourceSplitter::signalReadComplete_lock(bool readAborted) {
- if (!readAborted) {
- mNumberOfCurrentReads++;
- }
-
- if (mNumberOfCurrentReads == mNumberOfClientsStarted) {
- mLastReadCompleted = true;
- mNumberOfCurrentReads = 0;
- mAllReadsCompleteCondition.broadcast();
- }
-}
-
-status_t MediaSourceSplitter::pause(int clientId) {
- return ERROR_UNSUPPORTED;
-}
-
-// Client
-
-MediaSourceSplitter::Client::Client(
- sp<MediaSourceSplitter> splitter,
- int32_t clientId) {
- mSplitter = splitter;
- mClientId = clientId;
-}
-
-status_t MediaSourceSplitter::Client::start(MetaData *params) {
- return mSplitter->start(mClientId, params);
-}
-
-status_t MediaSourceSplitter::Client::stop() {
- return mSplitter->stop(mClientId);
-}
-
-sp<MetaData> MediaSourceSplitter::Client::getFormat() {
- return mSplitter->getFormat(mClientId);
-}
-
-status_t MediaSourceSplitter::Client::read(
- MediaBuffer **buffer, const ReadOptions *options) {
- return mSplitter->read(mClientId, buffer, options);
-}
-
-status_t MediaSourceSplitter::Client::pause() {
- return mSplitter->pause(mClientId);
-}
-
-} // namespace android
diff --git a/media/libstagefright/MetaData.cpp b/media/libstagefright/MetaData.cpp
index 884f3b4..66dec90 100644
--- a/media/libstagefright/MetaData.cpp
+++ b/media/libstagefright/MetaData.cpp
@@ -17,7 +17,7 @@
#include <stdlib.h>
#include <string.h>
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MetaData.h>
namespace android {
diff --git a/media/libstagefright/NuCachedSource2.cpp b/media/libstagefright/NuCachedSource2.cpp
index 249c298..0957426 100644
--- a/media/libstagefright/NuCachedSource2.cpp
+++ b/media/libstagefright/NuCachedSource2.cpp
@@ -370,6 +370,7 @@
&& (mSource->flags() & DataSource::kIsHTTPBasedSource)) {
ALOGV("Disconnecting at high watermark");
static_cast<HTTPBase *>(mSource.get())->disconnect();
+ mFinalStatus = -EAGAIN;
}
}
} else {
@@ -549,7 +550,7 @@
size_t delta = offset - mCacheOffset;
- if (mFinalStatus != OK) {
+ if (mFinalStatus != OK && mNumRetriesLeft == 0) {
if (delta >= mCache->totalSize()) {
return mFinalStatus;
}
@@ -591,7 +592,7 @@
size_t totalSize = mCache->totalSize();
CHECK_EQ(mCache->releaseFromStart(totalSize), totalSize);
- mFinalStatus = OK;
+ mNumRetriesLeft = kMaxNumRetries;
mFetching = true;
return OK;
@@ -603,8 +604,8 @@
restartPrefetcherIfNecessary_l(true /* ignore low water threshold */);
}
-sp<DecryptHandle> NuCachedSource2::DrmInitialization() {
- return mSource->DrmInitialization();
+sp<DecryptHandle> NuCachedSource2::DrmInitialization(const char* mime) {
+ return mSource->DrmInitialization(mime);
}
void NuCachedSource2::getDrmInfo(sp<DecryptHandle> &handle, DrmManagerClient **client) {
diff --git a/media/libstagefright/NuMediaExtractor.cpp b/media/libstagefright/NuMediaExtractor.cpp
new file mode 100644
index 0000000..afd4763
--- /dev/null
+++ b/media/libstagefright/NuMediaExtractor.cpp
@@ -0,0 +1,433 @@
+/*
+ * Copyright 2012, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "NuMediaExtractor"
+#include <utils/Log.h>
+
+#include <media/stagefright/NuMediaExtractor.h>
+
+#include "include/ESDS.h"
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/DataSource.h>
+#include <media/stagefright/MediaBuffer.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaExtractor.h>
+#include <media/stagefright/MediaSource.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/Utils.h>
+
+namespace android {
+
+NuMediaExtractor::NuMediaExtractor() {
+}
+
+NuMediaExtractor::~NuMediaExtractor() {
+ releaseTrackSamples();
+
+ for (size_t i = 0; i < mSelectedTracks.size(); ++i) {
+ TrackInfo *info = &mSelectedTracks.editItemAt(i);
+
+ CHECK_EQ((status_t)OK, info->mSource->stop());
+ }
+
+ mSelectedTracks.clear();
+}
+
+status_t NuMediaExtractor::setDataSource(const char *path) {
+ sp<DataSource> dataSource = DataSource::CreateFromURI(path);
+
+ if (dataSource == NULL) {
+ return -ENOENT;
+ }
+
+ mImpl = MediaExtractor::Create(dataSource);
+
+ if (mImpl == NULL) {
+ return ERROR_UNSUPPORTED;
+ }
+
+ return OK;
+}
+
+size_t NuMediaExtractor::countTracks() const {
+ return mImpl == NULL ? 0 : mImpl->countTracks();
+}
+
+status_t NuMediaExtractor::getTrackFormat(
+ size_t index, sp<AMessage> *format) const {
+ *format = NULL;
+
+ if (mImpl == NULL) {
+ return -EINVAL;
+ }
+
+ if (index >= mImpl->countTracks()) {
+ return -ERANGE;
+ }
+
+ sp<MetaData> meta = mImpl->getTrackMetaData(index);
+
+ const char *mime;
+ CHECK(meta->findCString(kKeyMIMEType, &mime));
+
+ sp<AMessage> msg = new AMessage;
+ msg->setString("mime", mime);
+
+ if (!strncasecmp("video/", mime, 6)) {
+ int32_t width, height;
+ CHECK(meta->findInt32(kKeyWidth, &width));
+ CHECK(meta->findInt32(kKeyHeight, &height));
+
+ msg->setInt32("width", width);
+ msg->setInt32("height", height);
+ } else {
+ CHECK(!strncasecmp("audio/", mime, 6));
+
+ int32_t numChannels, sampleRate;
+ CHECK(meta->findInt32(kKeyChannelCount, &numChannels));
+ CHECK(meta->findInt32(kKeySampleRate, &sampleRate));
+
+ msg->setInt32("channel-count", numChannels);
+ msg->setInt32("sample-rate", sampleRate);
+ }
+
+ int32_t maxInputSize;
+ if (meta->findInt32(kKeyMaxInputSize, &maxInputSize)) {
+ msg->setInt32("max-input-size", maxInputSize);
+ }
+
+ uint32_t type;
+ const void *data;
+ size_t size;
+ if (meta->findData(kKeyAVCC, &type, &data, &size)) {
+ // Parse the AVCDecoderConfigurationRecord
+
+ const uint8_t *ptr = (const uint8_t *)data;
+
+ CHECK(size >= 7);
+ CHECK_EQ((unsigned)ptr[0], 1u); // configurationVersion == 1
+ uint8_t profile = ptr[1];
+ uint8_t level = ptr[3];
+
+ // There is decodable content out there that fails the following
+ // assertion, let's be lenient for now...
+ // CHECK((ptr[4] >> 2) == 0x3f); // reserved
+
+ size_t lengthSize = 1 + (ptr[4] & 3);
+
+ // commented out check below as H264_QVGA_500_NO_AUDIO.3gp
+ // violates it...
+ // CHECK((ptr[5] >> 5) == 7); // reserved
+
+ size_t numSeqParameterSets = ptr[5] & 31;
+
+ ptr += 6;
+ size -= 6;
+
+ sp<ABuffer> buffer = new ABuffer(1024);
+ buffer->setRange(0, 0);
+
+ for (size_t i = 0; i < numSeqParameterSets; ++i) {
+ CHECK(size >= 2);
+ size_t length = U16_AT(ptr);
+
+ ptr += 2;
+ size -= 2;
+
+ CHECK(size >= length);
+
+ memcpy(buffer->data() + buffer->size(), "\x00\x00\x00\x01", 4);
+ memcpy(buffer->data() + buffer->size() + 4, ptr, length);
+ buffer->setRange(0, buffer->size() + 4 + length);
+
+ ptr += length;
+ size -= length;
+ }
+
+ buffer->meta()->setInt32("csd", true);
+ buffer->meta()->setInt64("timeUs", 0);
+
+ msg->setBuffer("csd-0", buffer);
+
+ buffer = new ABuffer(1024);
+ buffer->setRange(0, 0);
+
+ CHECK(size >= 1);
+ size_t numPictureParameterSets = *ptr;
+ ++ptr;
+ --size;
+
+ for (size_t i = 0; i < numPictureParameterSets; ++i) {
+ CHECK(size >= 2);
+ size_t length = U16_AT(ptr);
+
+ ptr += 2;
+ size -= 2;
+
+ CHECK(size >= length);
+
+ memcpy(buffer->data() + buffer->size(), "\x00\x00\x00\x01", 4);
+ memcpy(buffer->data() + buffer->size() + 4, ptr, length);
+ buffer->setRange(0, buffer->size() + 4 + length);
+
+ ptr += length;
+ size -= length;
+ }
+
+ buffer->meta()->setInt32("csd", true);
+ buffer->meta()->setInt64("timeUs", 0);
+ msg->setBuffer("csd-1", buffer);
+ } else if (meta->findData(kKeyESDS, &type, &data, &size)) {
+ ESDS esds((const char *)data, size);
+ CHECK_EQ(esds.InitCheck(), (status_t)OK);
+
+ const void *codec_specific_data;
+ size_t codec_specific_data_size;
+ esds.getCodecSpecificInfo(
+ &codec_specific_data, &codec_specific_data_size);
+
+ sp<ABuffer> buffer = new ABuffer(codec_specific_data_size);
+
+ memcpy(buffer->data(), codec_specific_data,
+ codec_specific_data_size);
+
+ buffer->meta()->setInt32("csd", true);
+ buffer->meta()->setInt64("timeUs", 0);
+ msg->setBuffer("csd-0", buffer);
+ } else if (meta->findData(kKeyVorbisInfo, &type, &data, &size)) {
+ sp<ABuffer> buffer = new ABuffer(size);
+ memcpy(buffer->data(), data, size);
+
+ buffer->meta()->setInt32("csd", true);
+ buffer->meta()->setInt64("timeUs", 0);
+ msg->setBuffer("csd-0", buffer);
+
+ if (!meta->findData(kKeyVorbisBooks, &type, &data, &size)) {
+ return -EINVAL;
+ }
+
+ buffer = new ABuffer(size);
+ memcpy(buffer->data(), data, size);
+
+ buffer->meta()->setInt32("csd", true);
+ buffer->meta()->setInt64("timeUs", 0);
+ msg->setBuffer("csd-1", buffer);
+ }
+
+ *format = msg;
+
+ return OK;
+}
+
+status_t NuMediaExtractor::selectTrack(size_t index) {
+ if (mImpl == NULL) {
+ return -EINVAL;
+ }
+
+ if (index >= mImpl->countTracks()) {
+ return -ERANGE;
+ }
+
+ for (size_t i = 0; i < mSelectedTracks.size(); ++i) {
+ TrackInfo *info = &mSelectedTracks.editItemAt(i);
+
+ if (info->mTrackIndex == index) {
+ // This track has already been selected.
+ return OK;
+ }
+ }
+
+ sp<MediaSource> source = mImpl->getTrack(index);
+
+ CHECK_EQ((status_t)OK, source->start());
+
+ mSelectedTracks.push();
+ TrackInfo *info = &mSelectedTracks.editItemAt(mSelectedTracks.size() - 1);
+
+ info->mSource = source;
+ info->mTrackIndex = index;
+ info->mFinalResult = OK;
+ info->mSample = NULL;
+ info->mSampleTimeUs = -1ll;
+ info->mFlags = 0;
+
+ const char *mime;
+ CHECK(source->getFormat()->findCString(kKeyMIMEType, &mime));
+
+ if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_VORBIS)) {
+ info->mFlags |= kIsVorbis;
+ }
+
+ return OK;
+}
+
+void NuMediaExtractor::releaseTrackSamples() {
+ for (size_t i = 0; i < mSelectedTracks.size(); ++i) {
+ TrackInfo *info = &mSelectedTracks.editItemAt(i);
+
+ if (info->mSample != NULL) {
+ info->mSample->release();
+ info->mSample = NULL;
+
+ info->mSampleTimeUs = -1ll;
+ }
+ }
+}
+
+ssize_t NuMediaExtractor::fetchTrackSamples(int64_t seekTimeUs) {
+ TrackInfo *minInfo = NULL;
+ ssize_t minIndex = -1;
+
+ for (size_t i = 0; i < mSelectedTracks.size(); ++i) {
+ TrackInfo *info = &mSelectedTracks.editItemAt(i);
+
+ if (seekTimeUs >= 0ll) {
+ info->mFinalResult = OK;
+
+ if (info->mSample != NULL) {
+ info->mSample->release();
+ info->mSample = NULL;
+ info->mSampleTimeUs = -1ll;
+ }
+ } else if (info->mFinalResult != OK) {
+ continue;
+ }
+
+ if (info->mSample == NULL) {
+ MediaSource::ReadOptions options;
+ if (seekTimeUs >= 0ll) {
+ options.setSeekTo(seekTimeUs);
+ }
+ status_t err = info->mSource->read(&info->mSample, &options);
+
+ if (err != OK) {
+ CHECK(info->mSample == NULL);
+
+ info->mFinalResult = err;
+ info->mSampleTimeUs = -1ll;
+ continue;
+ } else {
+ CHECK(info->mSample != NULL);
+ CHECK(info->mSample->meta_data()->findInt64(
+ kKeyTime, &info->mSampleTimeUs));
+ }
+ }
+
+ if (minInfo == NULL || info->mSampleTimeUs < minInfo->mSampleTimeUs) {
+ minInfo = info;
+ minIndex = i;
+ }
+ }
+
+ return minIndex;
+}
+
+status_t NuMediaExtractor::seekTo(int64_t timeUs) {
+ return fetchTrackSamples(timeUs);
+}
+
+status_t NuMediaExtractor::advance() {
+ ssize_t minIndex = fetchTrackSamples();
+
+ if (minIndex < 0) {
+ return ERROR_END_OF_STREAM;
+ }
+
+ TrackInfo *info = &mSelectedTracks.editItemAt(minIndex);
+
+ info->mSample->release();
+ info->mSample = NULL;
+ info->mSampleTimeUs = -1ll;
+
+ return OK;
+}
+
+status_t NuMediaExtractor::readSampleData(const sp<ABuffer> &buffer) {
+ ssize_t minIndex = fetchTrackSamples();
+
+ if (minIndex < 0) {
+ return ERROR_END_OF_STREAM;
+ }
+
+ TrackInfo *info = &mSelectedTracks.editItemAt(minIndex);
+
+ size_t sampleSize = info->mSample->range_length();
+
+ if (info->mFlags & kIsVorbis) {
+ // Each sample's data is suffixed by the number of page samples
+ // or -1 if not available.
+ sampleSize += sizeof(int32_t);
+ }
+
+ if (buffer->capacity() < sampleSize) {
+ return -ENOMEM;
+ }
+
+ const uint8_t *src =
+ (const uint8_t *)info->mSample->data()
+ + info->mSample->range_offset();
+
+ memcpy((uint8_t *)buffer->data(), src, info->mSample->range_length());
+
+ if (info->mFlags & kIsVorbis) {
+ int32_t numPageSamples;
+ if (!info->mSample->meta_data()->findInt32(
+ kKeyValidSamples, &numPageSamples)) {
+ numPageSamples = -1;
+ }
+
+ memcpy((uint8_t *)buffer->data() + info->mSample->range_length(),
+ &numPageSamples,
+ sizeof(numPageSamples));
+ }
+
+ buffer->setRange(0, sampleSize);
+
+ return OK;
+}
+
+status_t NuMediaExtractor::getSampleTrackIndex(size_t *trackIndex) {
+ ssize_t minIndex = fetchTrackSamples();
+
+ if (minIndex < 0) {
+ return ERROR_END_OF_STREAM;
+ }
+
+ TrackInfo *info = &mSelectedTracks.editItemAt(minIndex);
+ *trackIndex = info->mTrackIndex;
+
+ return OK;
+}
+
+status_t NuMediaExtractor::getSampleTime(int64_t *sampleTimeUs) {
+ ssize_t minIndex = fetchTrackSamples();
+
+ if (minIndex < 0) {
+ return ERROR_END_OF_STREAM;
+ }
+
+ TrackInfo *info = &mSelectedTracks.editItemAt(minIndex);
+ *sampleTimeUs = info->mSampleTimeUs;
+
+ return OK;
+}
+
+} // namespace android
diff --git a/media/libstagefright/OMXClient.cpp b/media/libstagefright/OMXClient.cpp
index 9de873e..7cdb793 100644
--- a/media/libstagefright/OMXClient.cpp
+++ b/media/libstagefright/OMXClient.cpp
@@ -20,11 +20,299 @@
#include <binder/IServiceManager.h>
#include <media/IMediaPlayerService.h>
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/OMXClient.h>
+#include <utils/KeyedVector.h>
+
+#include "include/OMX.h"
namespace android {
+struct MuxOMX : public IOMX {
+ MuxOMX(const sp<IOMX> &remoteOMX);
+ virtual ~MuxOMX();
+
+ virtual IBinder *onAsBinder() { return NULL; }
+
+ virtual bool livesLocally(node_id node, pid_t pid);
+
+ virtual status_t listNodes(List<ComponentInfo> *list);
+
+ virtual status_t allocateNode(
+ const char *name, const sp<IOMXObserver> &observer,
+ node_id *node);
+
+ virtual status_t freeNode(node_id node);
+
+ virtual status_t sendCommand(
+ node_id node, OMX_COMMANDTYPE cmd, OMX_S32 param);
+
+ virtual status_t getParameter(
+ node_id node, OMX_INDEXTYPE index,
+ void *params, size_t size);
+
+ virtual status_t setParameter(
+ node_id node, OMX_INDEXTYPE index,
+ const void *params, size_t size);
+
+ virtual status_t getConfig(
+ node_id node, OMX_INDEXTYPE index,
+ void *params, size_t size);
+
+ virtual status_t setConfig(
+ node_id node, OMX_INDEXTYPE index,
+ const void *params, size_t size);
+
+ virtual status_t getState(
+ node_id node, OMX_STATETYPE* state);
+
+ virtual status_t storeMetaDataInBuffers(
+ node_id node, OMX_U32 port_index, OMX_BOOL enable);
+
+ virtual status_t enableGraphicBuffers(
+ node_id node, OMX_U32 port_index, OMX_BOOL enable);
+
+ virtual status_t getGraphicBufferUsage(
+ node_id node, OMX_U32 port_index, OMX_U32* usage);
+
+ virtual status_t useBuffer(
+ node_id node, OMX_U32 port_index, const sp<IMemory> ¶ms,
+ buffer_id *buffer);
+
+ virtual status_t useGraphicBuffer(
+ node_id node, OMX_U32 port_index,
+ const sp<GraphicBuffer> &graphicBuffer, buffer_id *buffer);
+
+ virtual status_t allocateBuffer(
+ node_id node, OMX_U32 port_index, size_t size,
+ buffer_id *buffer, void **buffer_data);
+
+ virtual status_t allocateBufferWithBackup(
+ node_id node, OMX_U32 port_index, const sp<IMemory> ¶ms,
+ buffer_id *buffer);
+
+ virtual status_t freeBuffer(
+ node_id node, OMX_U32 port_index, buffer_id buffer);
+
+ virtual status_t fillBuffer(node_id node, buffer_id buffer);
+
+ virtual status_t emptyBuffer(
+ node_id node,
+ buffer_id buffer,
+ OMX_U32 range_offset, OMX_U32 range_length,
+ OMX_U32 flags, OMX_TICKS timestamp);
+
+ virtual status_t getExtensionIndex(
+ node_id node,
+ const char *parameter_name,
+ OMX_INDEXTYPE *index);
+
+private:
+ mutable Mutex mLock;
+
+ sp<IOMX> mRemoteOMX;
+ sp<IOMX> mLocalOMX;
+
+ KeyedVector<node_id, bool> mIsLocalNode;
+
+ bool isLocalNode(node_id node) const;
+ bool isLocalNode_l(node_id node) const;
+ const sp<IOMX> &getOMX(node_id node) const;
+ const sp<IOMX> &getOMX_l(node_id node) const;
+
+ static bool IsSoftwareComponent(const char *name);
+
+ DISALLOW_EVIL_CONSTRUCTORS(MuxOMX);
+};
+
+MuxOMX::MuxOMX(const sp<IOMX> &remoteOMX)
+ : mRemoteOMX(remoteOMX) {
+}
+
+MuxOMX::~MuxOMX() {
+}
+
+bool MuxOMX::isLocalNode(node_id node) const {
+ Mutex::Autolock autoLock(mLock);
+
+ return isLocalNode_l(node);
+}
+
+bool MuxOMX::isLocalNode_l(node_id node) const {
+ return mIsLocalNode.indexOfKey(node) >= 0;
+}
+
+// static
+bool MuxOMX::IsSoftwareComponent(const char *name) {
+ return !strncasecmp(name, "OMX.google.", 11);
+}
+
+const sp<IOMX> &MuxOMX::getOMX(node_id node) const {
+ return isLocalNode(node) ? mLocalOMX : mRemoteOMX;
+}
+
+const sp<IOMX> &MuxOMX::getOMX_l(node_id node) const {
+ return isLocalNode_l(node) ? mLocalOMX : mRemoteOMX;
+}
+
+bool MuxOMX::livesLocally(node_id node, pid_t pid) {
+ return getOMX(node)->livesLocally(node, pid);
+}
+
+status_t MuxOMX::listNodes(List<ComponentInfo> *list) {
+ Mutex::Autolock autoLock(mLock);
+
+ if (mLocalOMX == NULL) {
+ mLocalOMX = new OMX;
+ }
+
+ return mLocalOMX->listNodes(list);
+}
+
+status_t MuxOMX::allocateNode(
+ const char *name, const sp<IOMXObserver> &observer,
+ node_id *node) {
+ Mutex::Autolock autoLock(mLock);
+
+ sp<IOMX> omx;
+
+ if (IsSoftwareComponent(name)) {
+ if (mLocalOMX == NULL) {
+ mLocalOMX = new OMX;
+ }
+ omx = mLocalOMX;
+ } else {
+ omx = mRemoteOMX;
+ }
+
+ status_t err = omx->allocateNode(name, observer, node);
+
+ if (err != OK) {
+ return err;
+ }
+
+ if (omx == mLocalOMX) {
+ mIsLocalNode.add(*node, true);
+ }
+
+ return OK;
+}
+
+status_t MuxOMX::freeNode(node_id node) {
+ Mutex::Autolock autoLock(mLock);
+
+ status_t err = getOMX_l(node)->freeNode(node);
+
+ if (err != OK) {
+ return err;
+ }
+
+ mIsLocalNode.removeItem(node);
+
+ return OK;
+}
+
+status_t MuxOMX::sendCommand(
+ node_id node, OMX_COMMANDTYPE cmd, OMX_S32 param) {
+ return getOMX(node)->sendCommand(node, cmd, param);
+}
+
+status_t MuxOMX::getParameter(
+ node_id node, OMX_INDEXTYPE index,
+ void *params, size_t size) {
+ return getOMX(node)->getParameter(node, index, params, size);
+}
+
+status_t MuxOMX::setParameter(
+ node_id node, OMX_INDEXTYPE index,
+ const void *params, size_t size) {
+ return getOMX(node)->setParameter(node, index, params, size);
+}
+
+status_t MuxOMX::getConfig(
+ node_id node, OMX_INDEXTYPE index,
+ void *params, size_t size) {
+ return getOMX(node)->getConfig(node, index, params, size);
+}
+
+status_t MuxOMX::setConfig(
+ node_id node, OMX_INDEXTYPE index,
+ const void *params, size_t size) {
+ return getOMX(node)->setConfig(node, index, params, size);
+}
+
+status_t MuxOMX::getState(
+ node_id node, OMX_STATETYPE* state) {
+ return getOMX(node)->getState(node, state);
+}
+
+status_t MuxOMX::storeMetaDataInBuffers(
+ node_id node, OMX_U32 port_index, OMX_BOOL enable) {
+ return getOMX(node)->storeMetaDataInBuffers(node, port_index, enable);
+}
+
+status_t MuxOMX::enableGraphicBuffers(
+ node_id node, OMX_U32 port_index, OMX_BOOL enable) {
+ return getOMX(node)->enableGraphicBuffers(node, port_index, enable);
+}
+
+status_t MuxOMX::getGraphicBufferUsage(
+ node_id node, OMX_U32 port_index, OMX_U32* usage) {
+ return getOMX(node)->getGraphicBufferUsage(node, port_index, usage);
+}
+
+status_t MuxOMX::useBuffer(
+ node_id node, OMX_U32 port_index, const sp<IMemory> ¶ms,
+ buffer_id *buffer) {
+ return getOMX(node)->useBuffer(node, port_index, params, buffer);
+}
+
+status_t MuxOMX::useGraphicBuffer(
+ node_id node, OMX_U32 port_index,
+ const sp<GraphicBuffer> &graphicBuffer, buffer_id *buffer) {
+ return getOMX(node)->useGraphicBuffer(
+ node, port_index, graphicBuffer, buffer);
+}
+
+status_t MuxOMX::allocateBuffer(
+ node_id node, OMX_U32 port_index, size_t size,
+ buffer_id *buffer, void **buffer_data) {
+ return getOMX(node)->allocateBuffer(
+ node, port_index, size, buffer, buffer_data);
+}
+
+status_t MuxOMX::allocateBufferWithBackup(
+ node_id node, OMX_U32 port_index, const sp<IMemory> ¶ms,
+ buffer_id *buffer) {
+ return getOMX(node)->allocateBufferWithBackup(
+ node, port_index, params, buffer);
+}
+
+status_t MuxOMX::freeBuffer(
+ node_id node, OMX_U32 port_index, buffer_id buffer) {
+ return getOMX(node)->freeBuffer(node, port_index, buffer);
+}
+
+status_t MuxOMX::fillBuffer(node_id node, buffer_id buffer) {
+ return getOMX(node)->fillBuffer(node, buffer);
+}
+
+status_t MuxOMX::emptyBuffer(
+ node_id node,
+ buffer_id buffer,
+ OMX_U32 range_offset, OMX_U32 range_length,
+ OMX_U32 flags, OMX_TICKS timestamp) {
+ return getOMX(node)->emptyBuffer(
+ node, buffer, range_offset, range_length, flags, timestamp);
+}
+
+status_t MuxOMX::getExtensionIndex(
+ node_id node,
+ const char *parameter_name,
+ OMX_INDEXTYPE *index) {
+ return getOMX(node)->getExtensionIndex(node, parameter_name, index);
+}
+
OMXClient::OMXClient() {
}
@@ -38,10 +326,19 @@
mOMX = service->getOMX();
CHECK(mOMX.get() != NULL);
+ if (!mOMX->livesLocally(NULL /* node */, getpid())) {
+ ALOGI("Using client-side OMX mux.");
+ mOMX = new MuxOMX(mOMX);
+ }
+
return OK;
}
void OMXClient::disconnect() {
+ if (mOMX.get() != NULL) {
+ mOMX.clear();
+ mOMX = NULL;
+ }
}
} // namespace android
diff --git a/media/libstagefright/OMXCodec.cpp b/media/libstagefright/OMXCodec.cpp
index 60d9bb7..d5e6bec 100755
--- a/media/libstagefright/OMXCodec.cpp
+++ b/media/libstagefright/OMXCodec.cpp
@@ -19,8 +19,6 @@
#include <utils/Log.h>
#include "include/AACEncoder.h"
-#include "include/AMRNBEncoder.h"
-#include "include/AMRWBEncoder.h"
#include "include/AVCEncoder.h"
#include "include/M4vH263Encoder.h"
@@ -29,12 +27,13 @@
#include <binder/IServiceManager.h>
#include <binder/MemoryDealer.h>
#include <binder/ProcessState.h>
+#include <HardwareAPI.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/IMediaPlayerService.h>
-#include <media/stagefright/HardwareAPI.h>
#include <media/stagefright/MediaBuffer.h>
#include <media/stagefright/MediaBufferGroup.h>
#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MediaCodecList.h>
#include <media/stagefright/MediaExtractor.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/OMXCodec.h>
@@ -59,11 +58,6 @@
// component in question is buggy or not.
const static uint32_t kMaxColorFormatSupported = 1000;
-struct CodecInfo {
- const char *mime;
- const char *codec;
-};
-
#define FACTORY_CREATE_ENCODER(name) \
static sp<MediaSource> Make##name(const sp<MediaSource> &source, const sp<MetaData> &meta) { \
return new name(source, meta); \
@@ -71,8 +65,6 @@
#define FACTORY_REF(name) { #name, Make##name },
-FACTORY_CREATE_ENCODER(AMRNBEncoder)
-FACTORY_CREATE_ENCODER(AMRWBEncoder)
FACTORY_CREATE_ENCODER(AACEncoder)
FACTORY_CREATE_ENCODER(AVCEncoder)
FACTORY_CREATE_ENCODER(M4vH263Encoder)
@@ -86,8 +78,6 @@
};
static const FactoryInfo kFactoryInfo[] = {
- FACTORY_REF(AMRNBEncoder)
- FACTORY_REF(AMRWBEncoder)
FACTORY_REF(AACEncoder)
FACTORY_REF(AVCEncoder)
FACTORY_REF(M4vH263Encoder)
@@ -102,82 +92,8 @@
return NULL;
}
+#undef FACTORY_CREATE_ENCODER
#undef FACTORY_REF
-#undef FACTORY_CREATE
-
-static const CodecInfo kDecoderInfo[] = {
- { MEDIA_MIMETYPE_IMAGE_JPEG, "OMX.TI.JPEG.decode" },
-// { MEDIA_MIMETYPE_AUDIO_MPEG, "OMX.TI.MP3.decode" },
- { MEDIA_MIMETYPE_AUDIO_MPEG, "OMX.google.mp3.decoder" },
- { MEDIA_MIMETYPE_AUDIO_MPEG_LAYER_II, "OMX.Nvidia.mp2.decoder" },
-// { MEDIA_MIMETYPE_AUDIO_AMR_NB, "OMX.TI.AMR.decode" },
-// { MEDIA_MIMETYPE_AUDIO_AMR_NB, "OMX.Nvidia.amr.decoder" },
- { MEDIA_MIMETYPE_AUDIO_AMR_NB, "OMX.google.amrnb.decoder" },
-// { MEDIA_MIMETYPE_AUDIO_AMR_NB, "OMX.Nvidia.amrwb.decoder" },
- { MEDIA_MIMETYPE_AUDIO_AMR_WB, "OMX.TI.WBAMR.decode" },
- { MEDIA_MIMETYPE_AUDIO_AMR_WB, "OMX.google.amrwb.decoder" },
-// { MEDIA_MIMETYPE_AUDIO_AAC, "OMX.Nvidia.aac.decoder" },
- { MEDIA_MIMETYPE_AUDIO_AAC, "OMX.TI.AAC.decode" },
- { MEDIA_MIMETYPE_AUDIO_AAC, "OMX.google.aac.decoder" },
- { MEDIA_MIMETYPE_AUDIO_G711_ALAW, "OMX.google.g711.alaw.decoder" },
- { MEDIA_MIMETYPE_AUDIO_G711_MLAW, "OMX.google.g711.mlaw.decoder" },
- { MEDIA_MIMETYPE_VIDEO_MPEG4, "OMX.TI.DUCATI1.VIDEO.DECODER" },
- { MEDIA_MIMETYPE_VIDEO_MPEG4, "OMX.Nvidia.mp4.decode" },
- { MEDIA_MIMETYPE_VIDEO_MPEG4, "OMX.qcom.7x30.video.decoder.mpeg4" },
- { MEDIA_MIMETYPE_VIDEO_MPEG4, "OMX.qcom.video.decoder.mpeg4" },
- { MEDIA_MIMETYPE_VIDEO_MPEG4, "OMX.TI.Video.Decoder" },
- { MEDIA_MIMETYPE_VIDEO_MPEG4, "OMX.SEC.MPEG4.Decoder" },
- { MEDIA_MIMETYPE_VIDEO_MPEG4, "OMX.google.mpeg4.decoder" },
- { MEDIA_MIMETYPE_VIDEO_H263, "OMX.TI.DUCATI1.VIDEO.DECODER" },
- { MEDIA_MIMETYPE_VIDEO_H263, "OMX.Nvidia.h263.decode" },
- { MEDIA_MIMETYPE_VIDEO_H263, "OMX.qcom.7x30.video.decoder.h263" },
- { MEDIA_MIMETYPE_VIDEO_H263, "OMX.qcom.video.decoder.h263" },
- { MEDIA_MIMETYPE_VIDEO_H263, "OMX.SEC.H263.Decoder" },
- { MEDIA_MIMETYPE_VIDEO_H263, "OMX.google.h263.decoder" },
- { MEDIA_MIMETYPE_VIDEO_AVC, "OMX.TI.DUCATI1.VIDEO.DECODER" },
- { MEDIA_MIMETYPE_VIDEO_AVC, "OMX.Nvidia.h264.decode" },
- { MEDIA_MIMETYPE_VIDEO_AVC, "OMX.qcom.7x30.video.decoder.avc" },
- { MEDIA_MIMETYPE_VIDEO_AVC, "OMX.qcom.video.decoder.avc" },
- { MEDIA_MIMETYPE_VIDEO_AVC, "OMX.TI.Video.Decoder" },
- { MEDIA_MIMETYPE_VIDEO_AVC, "OMX.SEC.AVC.Decoder" },
- { MEDIA_MIMETYPE_VIDEO_AVC, "OMX.google.h264.decoder" },
- { MEDIA_MIMETYPE_VIDEO_AVC, "OMX.google.avc.decoder" },
- { MEDIA_MIMETYPE_AUDIO_VORBIS, "OMX.google.vorbis.decoder" },
- { MEDIA_MIMETYPE_VIDEO_VPX, "OMX.google.vpx.decoder" },
- { MEDIA_MIMETYPE_VIDEO_MPEG2, "OMX.Nvidia.mpeg2v.decode" },
-};
-
-static const CodecInfo kEncoderInfo[] = {
- { MEDIA_MIMETYPE_AUDIO_AMR_NB, "OMX.TI.AMR.encode" },
- { MEDIA_MIMETYPE_AUDIO_AMR_NB, "AMRNBEncoder" },
- { MEDIA_MIMETYPE_AUDIO_AMR_WB, "OMX.TI.WBAMR.encode" },
- { MEDIA_MIMETYPE_AUDIO_AMR_WB, "AMRWBEncoder" },
- { MEDIA_MIMETYPE_AUDIO_AAC, "OMX.TI.AAC.encode" },
- { MEDIA_MIMETYPE_AUDIO_AAC, "AACEncoder" },
- { MEDIA_MIMETYPE_VIDEO_MPEG4, "OMX.TI.DUCATI1.VIDEO.MPEG4E" },
- { MEDIA_MIMETYPE_VIDEO_MPEG4, "OMX.qcom.7x30.video.encoder.mpeg4" },
- { MEDIA_MIMETYPE_VIDEO_MPEG4, "OMX.qcom.video.encoder.mpeg4" },
- { MEDIA_MIMETYPE_VIDEO_MPEG4, "OMX.TI.Video.encoder" },
- { MEDIA_MIMETYPE_VIDEO_MPEG4, "OMX.Nvidia.mp4.encoder" },
- { MEDIA_MIMETYPE_VIDEO_MPEG4, "OMX.SEC.MPEG4.Encoder" },
- { MEDIA_MIMETYPE_VIDEO_MPEG4, "M4vH263Encoder" },
- { MEDIA_MIMETYPE_VIDEO_H263, "OMX.TI.DUCATI1.VIDEO.MPEG4E" },
- { MEDIA_MIMETYPE_VIDEO_H263, "OMX.qcom.7x30.video.encoder.h263" },
- { MEDIA_MIMETYPE_VIDEO_H263, "OMX.qcom.video.encoder.h263" },
- { MEDIA_MIMETYPE_VIDEO_H263, "OMX.TI.Video.encoder" },
- { MEDIA_MIMETYPE_VIDEO_H263, "OMX.Nvidia.h263.encoder" },
- { MEDIA_MIMETYPE_VIDEO_H263, "OMX.SEC.H263.Encoder" },
- { MEDIA_MIMETYPE_VIDEO_H263, "M4vH263Encoder" },
- { MEDIA_MIMETYPE_VIDEO_AVC, "OMX.TI.DUCATI1.VIDEO.H264E" },
- { MEDIA_MIMETYPE_VIDEO_AVC, "OMX.qcom.7x30.video.encoder.avc" },
- { MEDIA_MIMETYPE_VIDEO_AVC, "OMX.qcom.video.encoder.avc" },
- { MEDIA_MIMETYPE_VIDEO_AVC, "OMX.TI.Video.encoder" },
- { MEDIA_MIMETYPE_VIDEO_AVC, "OMX.Nvidia.h264.encoder" },
- { MEDIA_MIMETYPE_VIDEO_AVC, "OMX.SEC.AVC.Encoder" },
- { MEDIA_MIMETYPE_VIDEO_AVC, "AVCEncoder" },
-};
-
-#undef OPTIONAL
#define CODEC_LOGI(x, ...) ALOGI("[%s] "x, mComponentName, ##__VA_ARGS__)
#define CODEC_LOGV(x, ...) ALOGV("[%s] "x, mComponentName, ##__VA_ARGS__)
@@ -212,22 +128,6 @@
OMXCodecObserver &operator=(const OMXCodecObserver &);
};
-static const char *GetCodec(const CodecInfo *info, size_t numInfos,
- const char *mime, int index) {
- CHECK(index >= 0);
- for(size_t i = 0; i < numInfos; ++i) {
- if (!strcasecmp(mime, info[i].mime)) {
- if (index == 0) {
- return info[i].codec;
- }
-
- --index;
- }
- }
-
- return NULL;
-}
-
template<class T>
static void InitOMXParams(T *params) {
params->nSize = sizeof(T);
@@ -283,119 +183,36 @@
}
// static
-uint32_t OMXCodec::getComponentQuirks(
- const char *componentName, bool isEncoder) {
- uint32_t quirks = 0;
-
- if (!strcmp(componentName, "OMX.Nvidia.amr.decoder") ||
- !strcmp(componentName, "OMX.Nvidia.amrwb.decoder") ||
- !strcmp(componentName, "OMX.Nvidia.aac.decoder") ||
- !strcmp(componentName, "OMX.Nvidia.mp3.decoder")) {
- quirks |= kDecoderLiesAboutNumberOfChannels;
- }
-
- if (!strcmp(componentName, "OMX.TI.MP3.decode")) {
- quirks |= kNeedsFlushBeforeDisable;
- quirks |= kDecoderLiesAboutNumberOfChannels;
- }
- if (!strcmp(componentName, "OMX.TI.AAC.decode")) {
- quirks |= kNeedsFlushBeforeDisable;
- quirks |= kRequiresFlushCompleteEmulation;
- quirks |= kSupportsMultipleFramesPerInputBuffer;
- }
- if (!strncmp(componentName, "OMX.qcom.video.encoder.", 23)) {
- quirks |= kRequiresLoadedToIdleAfterAllocation;
- quirks |= kRequiresAllocateBufferOnInputPorts;
- quirks |= kRequiresAllocateBufferOnOutputPorts;
- if (!strncmp(componentName, "OMX.qcom.video.encoder.avc", 26)) {
-
- // The AVC encoder advertises the size of output buffers
- // based on the input video resolution and assumes
- // the worst/least compression ratio is 0.5. It is found that
- // sometimes, the output buffer size is larger than
- // size advertised by the encoder.
- quirks |= kRequiresLargerEncoderOutputBuffer;
- }
- }
- if (!strncmp(componentName, "OMX.qcom.7x30.video.encoder.", 28)) {
- }
- if (!strncmp(componentName, "OMX.qcom.video.decoder.", 23)) {
- quirks |= kRequiresAllocateBufferOnOutputPorts;
- quirks |= kDefersOutputBufferAllocation;
- }
- if (!strncmp(componentName, "OMX.qcom.7x30.video.decoder.", 28)) {
- quirks |= kRequiresAllocateBufferOnInputPorts;
- quirks |= kRequiresAllocateBufferOnOutputPorts;
- quirks |= kDefersOutputBufferAllocation;
- }
-
- if (!strcmp(componentName, "OMX.TI.DUCATI1.VIDEO.DECODER")) {
- quirks |= kRequiresAllocateBufferOnInputPorts;
- quirks |= kRequiresAllocateBufferOnOutputPorts;
- }
-
- // FIXME:
- // Remove the quirks after the work is done.
- else if (!strcmp(componentName, "OMX.TI.DUCATI1.VIDEO.MPEG4E") ||
- !strcmp(componentName, "OMX.TI.DUCATI1.VIDEO.H264E")) {
-
- quirks |= kRequiresAllocateBufferOnInputPorts;
- quirks |= kRequiresAllocateBufferOnOutputPorts;
- }
- else if (!strncmp(componentName, "OMX.TI.", 7)) {
- // Apparently I must not use OMX_UseBuffer on either input or
- // output ports on any of the TI components or quote:
- // "(I) may have unexpected problem (sic) which can be timing related
- // and hard to reproduce."
-
- quirks |= kRequiresAllocateBufferOnInputPorts;
- quirks |= kRequiresAllocateBufferOnOutputPorts;
- if (!strncmp(componentName, "OMX.TI.Video.encoder", 20)) {
- quirks |= kAvoidMemcopyInputRecordingFrames;
- }
- }
-
- if (!strcmp(componentName, "OMX.TI.Video.Decoder")) {
- quirks |= kInputBufferSizesAreBogus;
- }
-
- if (!strncmp(componentName, "OMX.SEC.", 8) && !isEncoder) {
- // These output buffers contain no video data, just some
- // opaque information that allows the overlay to display their
- // contents.
- quirks |= kOutputBuffersAreUnreadable;
- }
-
- return quirks;
-}
-
-// static
void OMXCodec::findMatchingCodecs(
const char *mime,
bool createEncoder, const char *matchComponentName,
uint32_t flags,
- Vector<String8> *matchingCodecs) {
+ Vector<String8> *matchingCodecs,
+ Vector<uint32_t> *matchingCodecQuirks) {
matchingCodecs->clear();
- for (int index = 0;; ++index) {
- const char *componentName;
+ if (matchingCodecQuirks) {
+ matchingCodecQuirks->clear();
+ }
- if (createEncoder) {
- componentName = GetCodec(
- kEncoderInfo,
- sizeof(kEncoderInfo) / sizeof(kEncoderInfo[0]),
- mime, index);
- } else {
- componentName = GetCodec(
- kDecoderInfo,
- sizeof(kDecoderInfo) / sizeof(kDecoderInfo[0]),
- mime, index);
- }
+ const MediaCodecList *list = MediaCodecList::getInstance();
+ if (list == NULL) {
+ return;
+ }
- if (!componentName) {
+ size_t index = 0;
+ for (;;) {
+ ssize_t matchIndex =
+ list->findCodecByType(mime, createEncoder, index);
+
+ if (matchIndex < 0) {
break;
}
+ index = matchIndex + 1;
+
+ const char *componentName = list->getCodecName(matchIndex);
+
// If a specific codec is requested, skip the non-matching ones.
if (matchComponentName && strcmp(componentName, matchComponentName)) {
continue;
@@ -410,6 +227,10 @@
(!(flags & (kSoftwareCodecsOnly | kHardwareCodecsOnly)))) {
matchingCodecs->push(String8(componentName));
+
+ if (matchingCodecQuirks) {
+ matchingCodecQuirks->push(getComponentQuirks(list, matchIndex));
+ }
}
}
@@ -419,6 +240,45 @@
}
// static
+uint32_t OMXCodec::getComponentQuirks(
+ const MediaCodecList *list, size_t index) {
+ uint32_t quirks = 0;
+ if (list->codecHasQuirk(
+ index, "requires-allocate-on-input-ports")) {
+ quirks |= kRequiresAllocateBufferOnInputPorts;
+ }
+ if (list->codecHasQuirk(
+ index, "requires-allocate-on-output-ports")) {
+ quirks |= kRequiresAllocateBufferOnOutputPorts;
+ }
+ if (list->codecHasQuirk(
+ index, "output-buffers-are-unreadable")) {
+ quirks |= kOutputBuffersAreUnreadable;
+ }
+
+ return quirks;
+}
+
+// static
+bool OMXCodec::findCodecQuirks(const char *componentName, uint32_t *quirks) {
+ const MediaCodecList *list = MediaCodecList::getInstance();
+
+ if (list == NULL) {
+ return false;
+ }
+
+ ssize_t index = list->findCodecByName(componentName);
+
+ if (index < 0) {
+ return false;
+ }
+
+ *quirks = getComponentQuirks(list, index);
+
+ return true;
+}
+
+// static
sp<MediaSource> OMXCodec::Create(
const sp<IOMX> &omx,
const sp<MetaData> &meta, bool createEncoder,
@@ -440,8 +300,10 @@
CHECK(success);
Vector<String8> matchingCodecs;
+ Vector<uint32_t> matchingCodecQuirks;
findMatchingCodecs(
- mime, createEncoder, matchComponentName, flags, &matchingCodecs);
+ mime, createEncoder, matchComponentName, flags,
+ &matchingCodecs, &matchingCodecQuirks);
if (matchingCodecs.isEmpty()) {
return NULL;
@@ -452,6 +314,7 @@
for (size_t i = 0; i < matchingCodecs.size(); ++i) {
const char *componentNameBase = matchingCodecs[i].string();
+ uint32_t quirks = matchingCodecQuirks[i];
const char *componentName = componentNameBase;
AString tmp;
@@ -475,8 +338,6 @@
ALOGV("Attempting to allocate OMX node '%s'", componentName);
- uint32_t quirks = getComponentQuirks(componentNameBase, createEncoder);
-
if (!createEncoder
&& (quirks & kOutputBuffersAreUnreadable)
&& (flags & kClientNeedsFramebuffer)) {
@@ -632,16 +493,6 @@
CODEC_LOGI(
"AVC profile = %u (%s), level = %u",
profile, AVCProfileToString(profile), level);
-
- if (!strcmp(mComponentName, "OMX.TI.Video.Decoder")
- && (profile != kAVCProfileBaseline || level > 30)) {
- // This stream exceeds the decoder's capabilities. The decoder
- // does not handle this gracefully and would clobber the heap
- // and wreak havoc instead...
-
- ALOGE("Profile and/or level exceed the decoder's capabilities.");
- return ERROR_UNSUPPORTED;
- }
} else if (meta->findData(kKeyVorbisInfo, &type, &data, &size)) {
addCodecSpecificData(data, size);
@@ -697,40 +548,11 @@
}
}
- if (!strcasecmp(mMIME, MEDIA_MIMETYPE_IMAGE_JPEG)
- && !strcmp(mComponentName, "OMX.TI.JPEG.decode")) {
- OMX_COLOR_FORMATTYPE format =
- OMX_COLOR_Format32bitARGB8888;
- // OMX_COLOR_FormatYUV420PackedPlanar;
- // OMX_COLOR_FormatCbYCrY;
- // OMX_COLOR_FormatYUV411Planar;
-
- int32_t width, height;
- bool success = meta->findInt32(kKeyWidth, &width);
- success = success && meta->findInt32(kKeyHeight, &height);
-
- int32_t compressedSize;
- success = success && meta->findInt32(
- kKeyMaxInputSize, &compressedSize);
-
- CHECK(success);
- CHECK(compressedSize > 0);
-
- setImageOutputFormat(format, width, height);
- setJPEGInputFormat(width, height, (OMX_U32)compressedSize);
- }
-
int32_t maxInputSize;
if (meta->findInt32(kKeyMaxInputSize, &maxInputSize)) {
setMinBufferSize(kPortIndexInput, (OMX_U32)maxInputSize);
}
- if (!strcmp(mComponentName, "OMX.TI.AMR.encode")
- || !strcmp(mComponentName, "OMX.TI.WBAMR.encode")
- || !strcmp(mComponentName, "OMX.TI.AAC.encode")) {
- setMinBufferSize(kPortIndexOutput, 8192); // XXX
- }
-
initOutputFormat(meta);
if ((mFlags & kClientNeedsFramebuffer)
@@ -834,21 +656,6 @@
index, format.eCompressionFormat, format.eColorFormat);
#endif
- if (!strcmp("OMX.TI.Video.encoder", mComponentName)) {
- if (portIndex == kPortIndexInput
- && colorFormat == format.eColorFormat) {
- // eCompressionFormat does not seem right.
- found = true;
- break;
- }
- if (portIndex == kPortIndexOutput
- && compressionFormat == format.eCompressionFormat) {
- // eColorFormat does not seem right.
- found = true;
- break;
- }
- }
-
if (format.eCompressionFormat == compressionFormat
&& format.eColorFormat == colorFormat) {
found = true;
@@ -911,13 +718,8 @@
int32_t targetColorFormat;
if (meta->findInt32(kKeyColorFormat, &targetColorFormat)) {
*colorFormat = (OMX_COLOR_FORMATTYPE) targetColorFormat;
- } else {
- if (!strcasecmp("OMX.TI.Video.encoder", mComponentName)) {
- *colorFormat = OMX_COLOR_FormatYCbYCr;
- }
}
-
// Check whether the target color format is supported.
return isColorFormatSupported(*colorFormat, kPortIndexInput);
}
@@ -1482,11 +1284,12 @@
const sp<MediaSource> &source,
const sp<ANativeWindow> &nativeWindow)
: mOMX(omx),
- mOMXLivesLocally(omx->livesLocally(getpid())),
+ mOMXLivesLocally(omx->livesLocally(node, getpid())),
mNode(node),
mQuirks(quirks),
mFlags(flags),
mIsEncoder(isEncoder),
+ mIsVideo(!strncasecmp("video/", mime, 6)),
mMIME(strdup(mime)),
mComponentName(strdup(componentName)),
mSource(source),
@@ -1529,8 +1332,6 @@
"audio_decoder.mp1", "audio_encoder.mp1" },
{ MEDIA_MIMETYPE_AUDIO_MPEG_LAYER_II,
"audio_decoder.mp2", "audio_encoder.mp2" },
- { MEDIA_MIMETYPE_AUDIO_MPEG,
- "audio_decoder.mp3", "audio_encoder.mp3" },
{ MEDIA_MIMETYPE_AUDIO_AMR_NB,
"audio_decoder.amrnb", "audio_encoder.amrnb" },
{ MEDIA_MIMETYPE_AUDIO_AMR_WB,
@@ -1539,12 +1340,18 @@
"audio_decoder.aac", "audio_encoder.aac" },
{ MEDIA_MIMETYPE_AUDIO_VORBIS,
"audio_decoder.vorbis", "audio_encoder.vorbis" },
+ { MEDIA_MIMETYPE_AUDIO_G711_MLAW,
+ "audio_decoder.g711mlaw", "audio_encoder.g711mlaw" },
+ { MEDIA_MIMETYPE_AUDIO_G711_ALAW,
+ "audio_decoder.g711alaw", "audio_encoder.g711alaw" },
{ MEDIA_MIMETYPE_VIDEO_AVC,
"video_decoder.avc", "video_encoder.avc" },
{ MEDIA_MIMETYPE_VIDEO_MPEG4,
"video_decoder.mpeg4", "video_encoder.mpeg4" },
{ MEDIA_MIMETYPE_VIDEO_H263,
"video_decoder.h263", "video_encoder.h263" },
+ { MEDIA_MIMETYPE_VIDEO_VPX,
+ "video_decoder.vpx", "video_encoder.vpx" },
};
static const size_t kNumMimeToRole =
@@ -2191,8 +1998,8 @@
}
}
-int64_t OMXCodec::retrieveDecodingTimeUs(bool isCodecSpecific) {
- CHECK(mIsEncoder);
+int64_t OMXCodec::getDecodingTimeUs() {
+ CHECK(mIsEncoder && mIsVideo);
if (mDecodingTimeList.empty()) {
CHECK(mSignalledEOS || mNoMoreOutputData);
@@ -2203,12 +2010,7 @@
List<int64_t>::iterator it = mDecodingTimeList.begin();
int64_t timeUs = *it;
-
- // If the output buffer is codec specific configuration,
- // do not remove the decoding time from the list.
- if (!isCodecSpecific) {
- mDecodingTimeList.erase(it);
- }
+ mDecodingTimeList.erase(it);
return timeUs;
}
@@ -2387,8 +2189,8 @@
mNoMoreOutputData = true;
}
- if (mIsEncoder) {
- int64_t decodingTimeUs = retrieveDecodingTimeUs(isCodecSpecific);
+ if (mIsEncoder && mIsVideo) {
+ int64_t decodingTimeUs = isCodecSpecific? 0: getDecodingTimeUs();
buffer->meta_data()->setInt64(kKeyDecodingTime, decodingTimeUs);
}
@@ -3249,7 +3051,7 @@
int64_t lastBufferTimeUs;
CHECK(srcBuffer->meta_data()->findInt64(kKeyTime, &lastBufferTimeUs));
CHECK(lastBufferTimeUs >= 0);
- if (mIsEncoder) {
+ if (mIsEncoder && mIsVideo) {
mDecodingTimeList.push_back(lastBufferTimeUs);
}
@@ -3333,13 +3135,6 @@
info->mStatus = OWNED_BY_COMPONENT;
- // This component does not ever signal the EOS flag on output buffers,
- // Thanks for nothing.
- if (mSignalledEOS && !strcmp(mComponentName, "OMX.TI.Video.encoder")) {
- mNoMoreOutputData = true;
- mBufferFilled.signal();
- }
-
return true;
}
@@ -3565,6 +3360,7 @@
//////////////// output port ////////////////////
// format
OMX_AUDIO_PARAM_PORTFORMATTYPE format;
+ InitOMXParams(&format);
format.nPortIndex = kPortIndexOutput;
format.nIndex = 0;
status_t err = OMX_ErrorNone;
diff --git a/media/libstagefright/OggExtractor.cpp b/media/libstagefright/OggExtractor.cpp
index 73efc27..5e79e78 100644
--- a/media/libstagefright/OggExtractor.cpp
+++ b/media/libstagefright/OggExtractor.cpp
@@ -21,10 +21,10 @@
#include "include/OggExtractor.h"
#include <cutils/properties.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/MediaBuffer.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaSource.h>
diff --git a/media/libstagefright/SampleIterator.cpp b/media/libstagefright/SampleIterator.cpp
index 81ec5c1..eae721b 100644
--- a/media/libstagefright/SampleIterator.cpp
+++ b/media/libstagefright/SampleIterator.cpp
@@ -22,8 +22,8 @@
#include <arpa/inet.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/DataSource.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/Utils.h>
#include "include/SampleTable.h"
diff --git a/media/libstagefright/SampleTable.cpp b/media/libstagefright/SampleTable.cpp
index 8d80d63..d9858d7 100644
--- a/media/libstagefright/SampleTable.cpp
+++ b/media/libstagefright/SampleTable.cpp
@@ -618,26 +618,31 @@
}
uint32_t left = 0;
- while (left < mNumSyncSamples) {
- uint32_t x = mSyncSamples[left];
+ uint32_t right = mNumSyncSamples;
+ while (left < right) {
+ uint32_t center = left + (right - left) / 2;
+ uint32_t x = mSyncSamples[center];
- if (x >= start_sample_index) {
+ if (start_sample_index < x) {
+ right = center;
+ } else if (start_sample_index > x) {
+ left = center + 1;
+ } else {
+ left = center;
break;
}
-
- ++left;
}
-
if (left == mNumSyncSamples) {
if (flags == kFlagAfter) {
ALOGE("tried to find a sync frame after the last one: %d", left);
return ERROR_OUT_OF_RANGE;
}
+ left = left - 1;
}
- if (left > 0) {
- --left;
- }
+ // Now ssi[left] is the sync sample index just before (or at)
+ // start_sample_index.
+ // Also start_sample_index < ssi[left + 1], if left + 1 < mNumSyncSamples.
uint32_t x = mSyncSamples[left];
@@ -682,7 +687,11 @@
x = mSyncSamples[left - 1];
- CHECK(x <= start_sample_index);
+ if (x > start_sample_index) {
+ // The table of sync sample indices was not sorted
+ // properly.
+ return ERROR_MALFORMED;
+ }
}
break;
}
@@ -696,7 +705,11 @@
x = mSyncSamples[left + 1];
- CHECK(x >= start_sample_index);
+ if (x < start_sample_index) {
+ // The table of sync sample indices was not sorted
+ // properly.
+ return ERROR_MALFORMED;
+ }
}
break;
diff --git a/media/libstagefright/StagefrightMetadataRetriever.cpp b/media/libstagefright/StagefrightMetadataRetriever.cpp
index 43bfd9e..35f9c1f 100644
--- a/media/libstagefright/StagefrightMetadataRetriever.cpp
+++ b/media/libstagefright/StagefrightMetadataRetriever.cpp
@@ -20,10 +20,10 @@
#include "include/StagefrightMetadataRetriever.h"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/ColorConverter.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/FileSource.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaExtractor.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/OMXCodec.h>
@@ -37,7 +37,7 @@
ALOGV("StagefrightMetadataRetriever()");
DataSource::RegisterDefaultSniffers();
- CHECK_EQ(mClient.connect(), OK);
+ CHECK_EQ(mClient.connect(), (status_t)OK);
}
StagefrightMetadataRetriever::~StagefrightMetadataRetriever() {
@@ -169,7 +169,7 @@
|| (buffer != NULL && buffer->range_length() == 0));
if (err != OK) {
- CHECK_EQ(buffer, NULL);
+ CHECK(buffer == NULL);
ALOGV("decoding frame failed.");
decoder->stop();
diff --git a/media/libstagefright/SurfaceMediaSource.cpp b/media/libstagefright/SurfaceMediaSource.cpp
index 48df058..7481e2e 100644
--- a/media/libstagefright/SurfaceMediaSource.cpp
+++ b/media/libstagefright/SurfaceMediaSource.cpp
@@ -16,22 +16,23 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "SurfaceMediaSource"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/SurfaceMediaSource.h>
-#include <ui/GraphicBuffer.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/MediaDefs.h>
-#include <media/stagefright/MediaDebug.h>
-#include <media/stagefright/openmax/OMX_IVCommon.h>
-#include <media/stagefright/MetadataBufferType.h>
+#include <OMX_IVCommon.h>
+#include <MetadataBufferType.h>
-#include <surfaceflinger/ISurfaceComposer.h>
-#include <surfaceflinger/SurfaceComposerClient.h>
-#include <surfaceflinger/IGraphicBufferAlloc.h>
+#include <ui/GraphicBuffer.h>
+#include <gui/ISurfaceComposer.h>
+#include <gui/IGraphicBufferAlloc.h>
#include <OMX_Component.h>
#include <utils/Log.h>
#include <utils/String8.h>
+#include <private/gui/ComposerService.h>
+
namespace android {
SurfaceMediaSource::SurfaceMediaSource(uint32_t bufW, uint32_t bufH) :
@@ -53,12 +54,15 @@
ALOGV("SurfaceMediaSource::SurfaceMediaSource");
sp<ISurfaceComposer> composer(ComposerService::getComposerService());
mGraphicBufferAlloc = composer->createGraphicBufferAlloc();
+ if (mGraphicBufferAlloc == 0) {
+ ALOGE("createGraphicBufferAlloc() failed in SurfaceMediaSource()");
+ }
}
SurfaceMediaSource::~SurfaceMediaSource() {
ALOGV("SurfaceMediaSource::~SurfaceMediaSource");
if (!mStopped) {
- stop();
+ reset();
}
}
@@ -714,9 +718,9 @@
}
-status_t SurfaceMediaSource::stop()
+status_t SurfaceMediaSource::reset()
{
- ALOGV("Stop");
+ ALOGV("Reset");
Mutex::Autolock lock(mMutex);
// TODO: Add waiting on mFrameCompletedCondition here?
@@ -853,7 +857,7 @@
}
if (!foundBuffer) {
- CHECK_EQ(0, "signalBufferReturned: bogus buffer");
+ CHECK(!"signalBufferReturned: bogus buffer");
}
}
diff --git a/media/libstagefright/ThrottledSource.cpp b/media/libstagefright/ThrottledSource.cpp
index 88e07b0..b1fcafd 100644
--- a/media/libstagefright/ThrottledSource.cpp
+++ b/media/libstagefright/ThrottledSource.cpp
@@ -16,7 +16,7 @@
#include "include/ThrottledSource.h"
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
namespace android {
diff --git a/media/libstagefright/TimedEventQueue.cpp b/media/libstagefright/TimedEventQueue.cpp
index 12c9c36..6d345bb 100644
--- a/media/libstagefright/TimedEventQueue.cpp
+++ b/media/libstagefright/TimedEventQueue.cpp
@@ -26,12 +26,10 @@
#include "include/TimedEventQueue.h"
-#include <cutils/sched_policy.h>
-
#include <sys/prctl.h>
#include <sys/time.h>
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
#ifdef ANDROID_SIMULATOR
#include <jni.h>
diff --git a/media/libstagefright/VBRISeeker.cpp b/media/libstagefright/VBRISeeker.cpp
index 6ac5a83..bcba874 100644
--- a/media/libstagefright/VBRISeeker.cpp
+++ b/media/libstagefright/VBRISeeker.cpp
@@ -92,7 +92,7 @@
}
sp<VBRISeeker> seeker = new VBRISeeker;
- seeker->mBasePos = post_id3_pos;
+ seeker->mBasePos = post_id3_pos + frameSize;
seeker->mDurationUs = durationUs;
off64_t offset = post_id3_pos;
diff --git a/media/libstagefright/VideoSourceDownSampler.cpp b/media/libstagefright/VideoSourceDownSampler.cpp
index 1b66990..90a42c9 100644
--- a/media/libstagefright/VideoSourceDownSampler.cpp
+++ b/media/libstagefright/VideoSourceDownSampler.cpp
@@ -17,9 +17,9 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "VideoSourceDownSampler"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/VideoSourceDownSampler.h>
#include <media/stagefright/MediaBuffer.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/YUVImage.h>
#include <media/stagefright/YUVCanvas.h>
diff --git a/media/libstagefright/WAVExtractor.cpp b/media/libstagefright/WAVExtractor.cpp
index 0bcaf08..c35a77a 100644
--- a/media/libstagefright/WAVExtractor.cpp
+++ b/media/libstagefright/WAVExtractor.cpp
@@ -20,23 +20,30 @@
#include "include/WAVExtractor.h"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaSource.h>
#include <media/stagefright/MetaData.h>
#include <utils/String8.h>
+#include <cutils/bitops.h>
+
+#define CHANNEL_MASK_USE_CHANNEL_ORDER 0
namespace android {
enum {
- WAVE_FORMAT_PCM = 1,
- WAVE_FORMAT_ALAW = 6,
- WAVE_FORMAT_MULAW = 7,
+ WAVE_FORMAT_PCM = 0x0001,
+ WAVE_FORMAT_ALAW = 0x0006,
+ WAVE_FORMAT_MULAW = 0x0007,
+ WAVE_FORMAT_EXTENSIBLE = 0xFFFE
};
+static const char* WAVEEXT_SUBFORMAT = "\x00\x00\x00\x00\x10\x00\x80\x00\x00\xAA\x00\x38\x9B\x71";
+
+
static uint32_t U32_LE_AT(const uint8_t *ptr) {
return ptr[3] << 24 | ptr[2] << 16 | ptr[1] << 8 | ptr[0];
}
@@ -84,7 +91,8 @@
WAVExtractor::WAVExtractor(const sp<DataSource> &source)
: mDataSource(source),
- mValidFormat(false) {
+ mValidFormat(false),
+ mChannelMask(CHANNEL_MASK_USE_CHANNEL_ORDER) {
mInitCheck = init();
}
@@ -161,21 +169,37 @@
return NO_INIT;
}
- uint8_t formatSpec[16];
- if (mDataSource->readAt(offset, formatSpec, 16) < 16) {
+ uint8_t formatSpec[40];
+ if (mDataSource->readAt(offset, formatSpec, 2) < 2) {
return NO_INIT;
}
mWaveFormat = U16_LE_AT(formatSpec);
if (mWaveFormat != WAVE_FORMAT_PCM
&& mWaveFormat != WAVE_FORMAT_ALAW
- && mWaveFormat != WAVE_FORMAT_MULAW) {
+ && mWaveFormat != WAVE_FORMAT_MULAW
+ && mWaveFormat != WAVE_FORMAT_EXTENSIBLE) {
return ERROR_UNSUPPORTED;
}
+ uint8_t fmtSize = 16;
+ if (mWaveFormat == WAVE_FORMAT_EXTENSIBLE) {
+ fmtSize = 40;
+ }
+ if (mDataSource->readAt(offset, formatSpec, fmtSize) < fmtSize) {
+ return NO_INIT;
+ }
+
mNumChannels = U16_LE_AT(&formatSpec[2]);
- if (mNumChannels != 1 && mNumChannels != 2) {
- return ERROR_UNSUPPORTED;
+ if (mWaveFormat != WAVE_FORMAT_EXTENSIBLE) {
+ if (mNumChannels != 1 && mNumChannels != 2) {
+ ALOGW("More than 2 channels (%d) in non-WAVE_EXT, unknown channel mask",
+ mNumChannels);
+ }
+ } else {
+ if (mNumChannels < 1 && mNumChannels > 8) {
+ return ERROR_UNSUPPORTED;
+ }
}
mSampleRate = U32_LE_AT(&formatSpec[4]);
@@ -186,7 +210,8 @@
mBitsPerSample = U16_LE_AT(&formatSpec[14]);
- if (mWaveFormat == WAVE_FORMAT_PCM) {
+ if (mWaveFormat == WAVE_FORMAT_PCM
+ || mWaveFormat == WAVE_FORMAT_EXTENSIBLE) {
if (mBitsPerSample != 8 && mBitsPerSample != 16
&& mBitsPerSample != 24) {
return ERROR_UNSUPPORTED;
@@ -199,6 +224,42 @@
}
}
+ if (mWaveFormat == WAVE_FORMAT_EXTENSIBLE) {
+ uint16_t validBitsPerSample = U16_LE_AT(&formatSpec[18]);
+ if (validBitsPerSample != mBitsPerSample) {
+ ALOGE("validBits(%d) != bitsPerSample(%d) are not supported",
+ validBitsPerSample, mBitsPerSample);
+ return ERROR_UNSUPPORTED;
+ }
+
+ mChannelMask = U32_LE_AT(&formatSpec[20]);
+ ALOGV("numChannels=%d channelMask=0x%x", mNumChannels, mChannelMask);
+ if ((mChannelMask >> 18) != 0) {
+ ALOGE("invalid channel mask 0x%x", mChannelMask);
+ return ERROR_MALFORMED;
+ }
+
+ if ((mChannelMask != CHANNEL_MASK_USE_CHANNEL_ORDER)
+ && (popcount(mChannelMask) != mNumChannels)) {
+ ALOGE("invalid number of channels (%d) in channel mask (0x%x)",
+ popcount(mChannelMask), mChannelMask);
+ return ERROR_MALFORMED;
+ }
+
+ // In a WAVE_EXT header, the first two bytes of the GUID stored at byte 24 contain
+ // the sample format, using the same definitions as a regular WAV header
+ mWaveFormat = U16_LE_AT(&formatSpec[24]);
+ if (mWaveFormat != WAVE_FORMAT_PCM
+ && mWaveFormat != WAVE_FORMAT_ALAW
+ && mWaveFormat != WAVE_FORMAT_MULAW) {
+ return ERROR_UNSUPPORTED;
+ }
+ if (memcmp(&formatSpec[26], WAVEEXT_SUBFORMAT, 14)) {
+ ALOGE("unsupported GUID");
+ return ERROR_UNSUPPORTED;
+ }
+ }
+
mValidFormat = true;
} else if (!memcmp(chunkHeader, "data", 4)) {
if (mValidFormat) {
@@ -217,13 +278,14 @@
kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_G711_ALAW);
break;
default:
- CHECK_EQ(mWaveFormat, WAVE_FORMAT_MULAW);
+ CHECK_EQ(mWaveFormat, (uint16_t)WAVE_FORMAT_MULAW);
mTrackMeta->setCString(
kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_G711_MLAW);
break;
}
mTrackMeta->setInt32(kKeyChannelCount, mNumChannels);
+ mTrackMeta->setInt32(kKeyChannelMask, mChannelMask);
mTrackMeta->setInt32(kKeySampleRate, mSampleRate);
size_t bytesPerSample = mBitsPerSample >> 3;
@@ -362,7 +424,7 @@
// Convert 8-bit unsigned samples to 16-bit signed.
MediaBuffer *tmp;
- CHECK_EQ(mGroup->acquire_buffer(&tmp), OK);
+ CHECK_EQ(mGroup->acquire_buffer(&tmp), (status_t)OK);
// The new buffer holds the sample number of samples, but each
// one is 2 bytes wide.
diff --git a/media/libstagefright/WVMExtractor.cpp b/media/libstagefright/WVMExtractor.cpp
index 2092cb6..dac8106 100644
--- a/media/libstagefright/WVMExtractor.cpp
+++ b/media/libstagefright/WVMExtractor.cpp
@@ -21,6 +21,7 @@
#include <arpa/inet.h>
#include <utils/String8.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/Utils.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/MediaSource.h>
@@ -28,7 +29,6 @@
#include <media/stagefright/MetaData.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaBuffer.h>
-#include <media/stagefright/MediaDebug.h>
#include <dlfcn.h>
#include <utils/Errors.h>
@@ -45,17 +45,12 @@
static Mutex gWVMutex;
WVMExtractor::WVMExtractor(const sp<DataSource> &source)
- : mDataSource(source) {
- {
- Mutex::Autolock autoLock(gWVMutex);
- if (gVendorLibHandle == NULL) {
- gVendorLibHandle = dlopen("libwvm.so", RTLD_NOW);
- }
+ : mDataSource(source)
+{
+ Mutex::Autolock autoLock(gWVMutex);
- if (gVendorLibHandle == NULL) {
- ALOGE("Failed to open libwvm.so");
- return;
- }
+ if (!getVendorLibHandle()) {
+ return;
}
typedef WVMLoadableExtractor *(*GetInstanceFunc)(sp<DataSource>);
@@ -64,13 +59,28 @@
"_ZN7android11GetInstanceENS_2spINS_10DataSourceEEE");
if (getInstanceFunc) {
+ CHECK(source->DrmInitialization(MEDIA_MIMETYPE_CONTAINER_WVM) != NULL);
mImpl = (*getInstanceFunc)(source);
CHECK(mImpl != NULL);
+ setDrmFlag(true);
} else {
ALOGE("Failed to locate GetInstance in libwvm.so");
}
}
+bool WVMExtractor::getVendorLibHandle()
+{
+ if (gVendorLibHandle == NULL) {
+ gVendorLibHandle = dlopen("libwvm.so", RTLD_NOW);
+ }
+
+ if (gVendorLibHandle == NULL) {
+ ALOGE("Failed to open libwvm.so");
+ }
+
+ return gVendorLibHandle != NULL;
+}
+
WVMExtractor::~WVMExtractor() {
}
@@ -113,5 +123,39 @@
}
}
+void WVMExtractor::setUID(uid_t uid) {
+ if (mImpl != NULL) {
+ mImpl->setUID(uid);
+ }
+}
+
+bool SniffWVM(
+ const sp<DataSource> &source, String8 *mimeType, float *confidence,
+ sp<AMessage> *) {
+
+ Mutex::Autolock autoLock(gWVMutex);
+
+ if (!WVMExtractor::getVendorLibHandle()) {
+ return false;
+ }
+
+ typedef WVMLoadableExtractor *(*SnifferFunc)(const sp<DataSource>&);
+ SnifferFunc snifferFunc =
+ (SnifferFunc) dlsym(gVendorLibHandle,
+ "_ZN7android15IsWidevineMediaERKNS_2spINS_10DataSourceEEE");
+
+ if (snifferFunc) {
+ if ((*snifferFunc)(source)) {
+ *mimeType = MEDIA_MIMETYPE_CONTAINER_WVM;
+ *confidence = 10.0f;
+ return true;
+ }
+ } else {
+ ALOGE("IsWidevineMedia not found in libwvm.so");
+ }
+
+ return false;
+}
+
} //namespace android
diff --git a/media/libstagefright/XINGSeeker.cpp b/media/libstagefright/XINGSeeker.cpp
index 2091381..9c91134 100644
--- a/media/libstagefright/XINGSeeker.cpp
+++ b/media/libstagefright/XINGSeeker.cpp
@@ -14,39 +14,22 @@
* limitations under the License.
*/
+#define LOG_TAG "XINGSEEKER"
+#include <utils/Log.h>
+
#include "include/XINGSeeker.h"
+#include "include/avc_utils.h"
#include <media/stagefright/DataSource.h>
#include <media/stagefright/Utils.h>
namespace android {
-static bool parse_xing_header(
- const sp<DataSource> &source, off64_t first_frame_pos,
- int32_t *frame_number = NULL, int32_t *byte_number = NULL,
- unsigned char *table_of_contents = NULL,
- int32_t *quality_indicator = NULL, int64_t *duration = NULL);
-
-// static
-sp<XINGSeeker> XINGSeeker::CreateFromSource(
- const sp<DataSource> &source, off64_t first_frame_pos) {
- sp<XINGSeeker> seeker = new XINGSeeker;
-
- seeker->mFirstFramePos = first_frame_pos;
-
- if (!parse_xing_header(
- source, first_frame_pos,
- NULL, &seeker->mSizeBytes, seeker->mTableOfContents,
- NULL, &seeker->mDurationUs)) {
- return NULL;
- }
-
- return seeker;
-}
-
XINGSeeker::XINGSeeker()
: mDurationUs(-1),
- mSizeBytes(0) {
+ mSizeBytes(0),
+ mEncoderDelay(0),
+ mEncoderPadding(0) {
}
bool XINGSeeker::getDuration(int64_t *durationUs) {
@@ -60,7 +43,7 @@
}
bool XINGSeeker::getOffsetForTime(int64_t *timeUs, off64_t *pos) {
- if (mSizeBytes == 0 || mTableOfContents[0] <= 0 || mDurationUs < 0) {
+ if (mSizeBytes == 0 || !mTOCValid || mDurationUs < 0) {
return false;
}
@@ -76,10 +59,10 @@
if ( a == 0 ) {
fa = 0.0f;
} else {
- fa = (float)mTableOfContents[a-1];
+ fa = (float)mTOC[a-1];
}
if ( a < 99 ) {
- fb = (float)mTableOfContents[a];
+ fb = (float)mTOC[a];
} else {
fb = 256.0f;
}
@@ -91,59 +74,50 @@
return true;
}
-static bool parse_xing_header(
- const sp<DataSource> &source, off64_t first_frame_pos,
- int32_t *frame_number, int32_t *byte_number,
- unsigned char *table_of_contents, int32_t *quality_indicator,
- int64_t *duration) {
- if (frame_number) {
- *frame_number = 0;
- }
- if (byte_number) {
- *byte_number = 0;
- }
- if (table_of_contents) {
- table_of_contents[0] = 0;
- }
- if (quality_indicator) {
- *quality_indicator = 0;
- }
- if (duration) {
- *duration = 0;
- }
+// static
+sp<XINGSeeker> XINGSeeker::CreateFromSource(
+ const sp<DataSource> &source, off64_t first_frame_pos) {
+ sp<XINGSeeker> seeker = new XINGSeeker;
+
+ seeker->mFirstFramePos = first_frame_pos;
+
+ seeker->mSizeBytes = 0;
+ seeker->mTOCValid = false;
+ seeker->mDurationUs = 0;
uint8_t buffer[4];
int offset = first_frame_pos;
if (source->readAt(offset, &buffer, 4) < 4) { // get header
- return false;
+ return NULL;
}
offset += 4;
- uint8_t id, layer, sr_index, mode;
- layer = (buffer[1] >> 1) & 3;
- id = (buffer[1] >> 3) & 3;
- sr_index = (buffer[2] >> 2) & 3;
- mode = (buffer[3] >> 6) & 3;
- if (layer == 0) {
- return false;
+ int header = U32_AT(buffer);;
+ size_t xingframesize = 0;
+ int sampling_rate = 0;
+ int num_channels;
+ int samples_per_frame = 0;
+ if (!GetMPEGAudioFrameSize(header, &xingframesize, &sampling_rate, &num_channels,
+ NULL, &samples_per_frame)) {
+ return NULL;
}
- if (id == 1) {
- return false;
- }
- if (sr_index == 3) {
- return false;
- }
+ seeker->mFirstFramePos += xingframesize;
+
+ uint8_t version = (buffer[1] >> 3) & 3;
+
// determine offset of XING header
- if(id&1) { // mpeg1
- if (mode != 3) offset += 32;
+ if(version & 1) { // mpeg1
+ if (num_channels != 1) offset += 32;
else offset += 17;
- } else { // mpeg2
- if (mode != 3) offset += 17;
+ } else { // mpeg 2 or 2.5
+ if (num_channels != 1) offset += 17;
else offset += 9;
}
+ int xingbase = offset;
+
if (source->readAt(offset, &buffer, 4) < 4) { // XING header ID
- return false;
+ return NULL;
}
offset += 4;
// Check XING ID
@@ -151,70 +125,71 @@
|| (buffer[2] != 'n') || (buffer[3] != 'g')) {
if ((buffer[0] != 'I') || (buffer[1] != 'n')
|| (buffer[2] != 'f') || (buffer[3] != 'o')) {
- return false;
+ return NULL;
}
}
if (source->readAt(offset, &buffer, 4) < 4) { // flags
- return false;
+ return NULL;
}
offset += 4;
uint32_t flags = U32_AT(buffer);
if (flags & 0x0001) { // Frames field is present
if (source->readAt(offset, buffer, 4) < 4) {
- return false;
+ return NULL;
}
- if (frame_number) {
- *frame_number = U32_AT(buffer);
- }
- int32_t frame = U32_AT(buffer);
- // Samples per Frame: 1. index = MPEG Version ID, 2. index = Layer
- const int samplesPerFrames[2][3] =
- {
- { 384, 1152, 576 }, // MPEG 2, 2.5: layer1, layer2, layer3
- { 384, 1152, 1152 }, // MPEG 1: layer1, layer2, layer3
- };
- // sampling rates in hertz: 1. index = MPEG Version ID, 2. index = sampling rate index
- const int samplingRates[4][3] =
- {
- { 11025, 12000, 8000, }, // MPEG 2.5
- { 0, 0, 0, }, // reserved
- { 22050, 24000, 16000, }, // MPEG 2
- { 44100, 48000, 32000, } // MPEG 1
- };
- if (duration) {
- *duration = (int64_t)frame * samplesPerFrames[id&1][3-layer] * 1000000LL
- / samplingRates[id][sr_index];
- }
+ int32_t frames = U32_AT(buffer);
+ seeker->mDurationUs = (int64_t)frames * samples_per_frame * 1000000LL / sampling_rate;
offset += 4;
}
if (flags & 0x0002) { // Bytes field is present
- if (byte_number) {
- if (source->readAt(offset, buffer, 4) < 4) {
- return false;
- }
- *byte_number = U32_AT(buffer);
+ if (source->readAt(offset, buffer, 4) < 4) {
+ return NULL;
}
+ seeker->mSizeBytes = U32_AT(buffer);
offset += 4;
}
if (flags & 0x0004) { // TOC field is present
- if (table_of_contents) {
- if (source->readAt(offset + 1, table_of_contents, 99) < 99) {
- return false;
- }
+ if (source->readAt(offset + 1, seeker->mTOC, 99) < 99) {
+ return NULL;
}
+ seeker->mTOCValid = true;
offset += 100;
}
+
+#if 0
if (flags & 0x0008) { // Quality indicator field is present
- if (quality_indicator) {
- if (source->readAt(offset, buffer, 4) < 4) {
- return false;
- }
- *quality_indicator = U32_AT(buffer);
+ if (source->readAt(offset, buffer, 4) < 4) {
+ return NULL;
}
+ // do something with the quality indicator
+ offset += 4;
}
- return true;
+
+ if (source->readAt(xingbase + 0xaf - 0x24, &buffer, 1) < 1) { // encoding flags
+ return false;
+ }
+
+ ALOGV("nogap preceding: %s, nogap continued in next: %s",
+ (buffer[0] & 0x80) ? "true" : "false",
+ (buffer[0] & 0x40) ? "true" : "false");
+#endif
+
+ if (source->readAt(xingbase + 0xb1 - 0x24, &buffer, 3) == 3) {
+ seeker->mEncoderDelay = (buffer[0] << 4) + (buffer[1] >> 4);
+ seeker->mEncoderPadding = ((buffer[1] & 0xf) << 8) + buffer[2];
+ }
+
+ return seeker;
+}
+
+int32_t XINGSeeker::getEncoderDelay() {
+ return mEncoderDelay;
+}
+
+int32_t XINGSeeker::getEncoderPadding() {
+ return mEncoderPadding;
}
} // namespace android
diff --git a/media/libstagefright/chromium_http/Android.mk b/media/libstagefright/chromium_http/Android.mk
index 6573e3c..e37b4a8 100644
--- a/media/libstagefright/chromium_http/Android.mk
+++ b/media/libstagefright/chromium_http/Android.mk
@@ -3,13 +3,14 @@
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
- ChromiumHTTPDataSource.cpp \
- support.cpp \
+ DataUriSource.cpp \
+ ChromiumHTTPDataSource.cpp \
+ support.cpp
LOCAL_C_INCLUDES:= \
$(JNI_H_INCLUDE) \
frameworks/base/media/libstagefright \
- $(TOP)/frameworks/base/include/media/stagefright/openmax \
+ $(TOP)/frameworks/native/include/media/openmax \
external/chromium \
external/chromium/android
diff --git a/media/libstagefright/chromium_http/ChromiumHTTPDataSource.cpp b/media/libstagefright/chromium_http/ChromiumHTTPDataSource.cpp
index 180460b..76f7946 100644
--- a/media/libstagefright/chromium_http/ChromiumHTTPDataSource.cpp
+++ b/media/libstagefright/chromium_http/ChromiumHTTPDataSource.cpp
@@ -259,7 +259,7 @@
mCondition.broadcast();
}
-sp<DecryptHandle> ChromiumHTTPDataSource::DrmInitialization() {
+sp<DecryptHandle> ChromiumHTTPDataSource::DrmInitialization(const char* mime) {
Mutex::Autolock autoLock(mLock);
if (mDrmManagerClient == NULL) {
@@ -275,7 +275,7 @@
* original one
*/
mDecryptHandle = mDrmManagerClient->openDecryptSession(
- String8(mURI.c_str()));
+ String8(mURI.c_str()), mime);
}
if (mDecryptHandle == NULL) {
diff --git a/media/libstagefright/chromium_http/DataUriSource.cpp b/media/libstagefright/chromium_http/DataUriSource.cpp
new file mode 100644
index 0000000..ecf3fa1
--- /dev/null
+++ b/media/libstagefright/chromium_http/DataUriSource.cpp
@@ -0,0 +1,68 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <include/DataUriSource.h>
+
+#include <net/base/data_url.h>
+#include <googleurl/src/gurl.h>
+
+
+namespace android {
+
+DataUriSource::DataUriSource(const char *uri) :
+ mDataUri(uri),
+ mInited(NO_INIT) {
+
+ // Copy1: const char *uri -> String8 mDataUri.
+ std::string mimeTypeStr, unusedCharsetStr, dataStr;
+ // Copy2: String8 mDataUri -> std::string
+ const bool ret = net::DataURL::Parse(
+ GURL(std::string(mDataUri.string())),
+ &mimeTypeStr, &unusedCharsetStr, &dataStr);
+ // Copy3: std::string dataStr -> AString mData
+ mData.setTo(dataStr.data(), dataStr.length());
+ mInited = ret ? OK : UNKNOWN_ERROR;
+
+ // The chromium data url implementation defaults to using "text/plain"
+ // if no mime type is specified. We prefer to leave this unspecified
+ // instead, since the mime type is sniffed in most cases.
+ if (mimeTypeStr != "text/plain") {
+ mMimeType = mimeTypeStr.c_str();
+ }
+}
+
+ssize_t DataUriSource::readAt(off64_t offset, void *out, size_t size) {
+ if (mInited != OK) {
+ return mInited;
+ }
+
+ const off64_t length = mData.size();
+ if (offset >= length) {
+ return UNKNOWN_ERROR;
+ }
+
+ const char *dataBuf = mData.c_str();
+ const size_t bytesToCopy =
+ offset + size >= length ? (length - offset) : size;
+
+ if (bytesToCopy > 0) {
+ memcpy(out, dataBuf + offset, bytesToCopy);
+ }
+
+ return bytesToCopy;
+}
+
+} // namespace android
diff --git a/media/libstagefright/codecs/aacdec/Android.mk b/media/libstagefright/codecs/aacdec/Android.mk
index 20c7bc0..fd6de79 100644
--- a/media/libstagefright/codecs/aacdec/Android.mk
+++ b/media/libstagefright/codecs/aacdec/Android.mk
@@ -164,7 +164,7 @@
LOCAL_C_INCLUDES := \
frameworks/base/media/libstagefright/include \
- frameworks/base/include/media/stagefright/openmax \
+ frameworks/native/include/media/openmax
LOCAL_CFLAGS := -DOSCL_IMPORT_REF=
diff --git a/media/libstagefright/codecs/aacdec/SoftAAC.cpp b/media/libstagefright/codecs/aacdec/SoftAAC.cpp
index da9d280..ea6c360 100644
--- a/media/libstagefright/codecs/aacdec/SoftAAC.cpp
+++ b/media/libstagefright/codecs/aacdec/SoftAAC.cpp
@@ -218,6 +218,18 @@
return OMX_ErrorNone;
}
+ case OMX_IndexParamAudioPcm:
+ {
+ const OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
+ (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
+
+ if (pcmParams->nPortIndex != 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
default:
return SimpleSoftOMXComponent::internalSetParameter(index, params);
}
diff --git a/media/libstagefright/codecs/aacenc/AACEncoder.cpp b/media/libstagefright/codecs/aacenc/AACEncoder.cpp
index 2b8633d..8b5007e 100644
--- a/media/libstagefright/codecs/aacenc/AACEncoder.cpp
+++ b/media/libstagefright/codecs/aacenc/AACEncoder.cpp
@@ -22,8 +22,8 @@
#include "voAAC.h"
#include "cmnMemory.h"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
@@ -114,8 +114,8 @@
ALOGV("setAudioSpecificConfigData: %d hz, %d bps, and %d channels",
mSampleRate, mBitRate, mChannels);
- int32_t index;
- CHECK_EQ(OK, getSampleRateTableIndex(mSampleRate, index));
+ int32_t index = 0;
+ CHECK_EQ((status_t)OK, getSampleRateTableIndex(mSampleRate, index));
if (mChannels > 2 || mChannels <= 0) {
ALOGE("Unsupported number of channels(%d)", mChannels);
return UNKNOWN_ERROR;
@@ -142,7 +142,7 @@
mBufferGroup = new MediaBufferGroup;
mBufferGroup->add_buffer(new MediaBuffer(2048));
- CHECK_EQ(OK, initCheck());
+ CHECK_EQ((status_t)OK, initCheck());
mNumInputSamples = 0;
mAnchorTimeUs = 0;
@@ -183,7 +183,7 @@
mSource->stop();
if (mEncoderHandle) {
- CHECK_EQ(VO_ERR_NONE, mApiHandle->Uninit(mEncoderHandle));
+ CHECK_EQ((VO_U32)VO_ERR_NONE, mApiHandle->Uninit(mEncoderHandle));
mEncoderHandle = NULL;
}
delete mApiHandle;
@@ -223,7 +223,7 @@
CHECK(options == NULL || !options->getSeekTo(&seekTimeUs, &mode));
MediaBuffer *buffer;
- CHECK_EQ(mBufferGroup->acquire_buffer(&buffer), OK);
+ CHECK_EQ(mBufferGroup->acquire_buffer(&buffer), (status_t)OK);
uint8_t *outPtr = (uint8_t *)buffer->data();
bool readFromSource = false;
int64_t wallClockTimeUs = -1;
@@ -255,7 +255,7 @@
}
size_t align = mInputBuffer->range_length() % sizeof(int16_t);
- CHECK_EQ(align, 0);
+ CHECK_EQ(align, (size_t)0);
int64_t timeUs;
if (mInputBuffer->meta_data()->findInt64(kKeyDriftTime, &timeUs)) {
diff --git a/media/libstagefright/codecs/aacenc/Android.mk b/media/libstagefright/codecs/aacenc/Android.mk
index c2579c7..b47cb1e 100644
--- a/media/libstagefright/codecs/aacenc/Android.mk
+++ b/media/libstagefright/codecs/aacenc/Android.mk
@@ -85,3 +85,29 @@
endif
include $(BUILD_STATIC_LIBRARY)
+
+################################################################################
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES := \
+ SoftAACEncoder.cpp
+
+LOCAL_C_INCLUDES := \
+ frameworks/base/media/libstagefright/include \
+ frameworks/base/media/libstagefright/codecs/common/include \
+ frameworks/native/include/media/openmax
+
+LOCAL_CFLAGS := -DOSCL_IMPORT_REF=
+
+LOCAL_STATIC_LIBRARIES := \
+ libstagefright_aacenc
+
+LOCAL_SHARED_LIBRARIES := \
+ libstagefright_omx libstagefright_foundation libutils \
+ libstagefright_enc_common
+
+LOCAL_MODULE := libstagefright_soft_aacenc
+LOCAL_MODULE_TAGS := optional
+
+include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libstagefright/codecs/aacenc/SoftAACEncoder.cpp b/media/libstagefright/codecs/aacenc/SoftAACEncoder.cpp
new file mode 100644
index 0000000..c6724c2
--- /dev/null
+++ b/media/libstagefright/codecs/aacenc/SoftAACEncoder.cpp
@@ -0,0 +1,560 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "SoftAACEncoder"
+#include <utils/Log.h>
+
+#include "SoftAACEncoder.h"
+
+#include "voAAC.h"
+#include "cmnMemory.h"
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/hexdump.h>
+
+namespace android {
+
+template<class T>
+static void InitOMXParams(T *params) {
+ params->nSize = sizeof(T);
+ params->nVersion.s.nVersionMajor = 1;
+ params->nVersion.s.nVersionMinor = 0;
+ params->nVersion.s.nRevision = 0;
+ params->nVersion.s.nStep = 0;
+}
+
+SoftAACEncoder::SoftAACEncoder(
+ const char *name,
+ const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData,
+ OMX_COMPONENTTYPE **component)
+ : SimpleSoftOMXComponent(name, callbacks, appData, component),
+ mEncoderHandle(NULL),
+ mApiHandle(NULL),
+ mMemOperator(NULL),
+ mNumChannels(1),
+ mSampleRate(44100),
+ mBitRate(0),
+ mSentCodecSpecificData(false),
+ mInputSize(0),
+ mInputFrame(NULL),
+ mInputTimeUs(-1ll),
+ mSawInputEOS(false),
+ mSignalledError(false) {
+ initPorts();
+ CHECK_EQ(initEncoder(), (status_t)OK);
+
+ setAudioParams();
+}
+
+SoftAACEncoder::~SoftAACEncoder() {
+ delete[] mInputFrame;
+ mInputFrame = NULL;
+
+ if (mEncoderHandle) {
+ CHECK_EQ(VO_ERR_NONE, mApiHandle->Uninit(mEncoderHandle));
+ mEncoderHandle = NULL;
+ }
+
+ delete mApiHandle;
+ mApiHandle = NULL;
+
+ delete mMemOperator;
+ mMemOperator = NULL;
+}
+
+void SoftAACEncoder::initPorts() {
+ OMX_PARAM_PORTDEFINITIONTYPE def;
+ InitOMXParams(&def);
+
+ def.nPortIndex = 0;
+ def.eDir = OMX_DirInput;
+ def.nBufferCountMin = kNumBuffers;
+ def.nBufferCountActual = def.nBufferCountMin;
+ def.nBufferSize = kNumSamplesPerFrame * sizeof(int16_t) * 2;
+ def.bEnabled = OMX_TRUE;
+ def.bPopulated = OMX_FALSE;
+ def.eDomain = OMX_PortDomainAudio;
+ def.bBuffersContiguous = OMX_FALSE;
+ def.nBufferAlignment = 1;
+
+ def.format.audio.cMIMEType = const_cast<char *>("audio/raw");
+ def.format.audio.pNativeRender = NULL;
+ def.format.audio.bFlagErrorConcealment = OMX_FALSE;
+ def.format.audio.eEncoding = OMX_AUDIO_CodingPCM;
+
+ addPort(def);
+
+ def.nPortIndex = 1;
+ def.eDir = OMX_DirOutput;
+ def.nBufferCountMin = kNumBuffers;
+ def.nBufferCountActual = def.nBufferCountMin;
+ def.nBufferSize = 8192;
+ def.bEnabled = OMX_TRUE;
+ def.bPopulated = OMX_FALSE;
+ def.eDomain = OMX_PortDomainAudio;
+ def.bBuffersContiguous = OMX_FALSE;
+ def.nBufferAlignment = 2;
+
+ def.format.audio.cMIMEType = const_cast<char *>("audio/aac");
+ def.format.audio.pNativeRender = NULL;
+ def.format.audio.bFlagErrorConcealment = OMX_FALSE;
+ def.format.audio.eEncoding = OMX_AUDIO_CodingAAC;
+
+ addPort(def);
+}
+
+status_t SoftAACEncoder::initEncoder() {
+ mApiHandle = new VO_AUDIO_CODECAPI;
+
+ if (VO_ERR_NONE != voGetAACEncAPI(mApiHandle)) {
+ ALOGE("Failed to get api handle");
+ return UNKNOWN_ERROR;
+ }
+
+ mMemOperator = new VO_MEM_OPERATOR;
+ mMemOperator->Alloc = cmnMemAlloc;
+ mMemOperator->Copy = cmnMemCopy;
+ mMemOperator->Free = cmnMemFree;
+ mMemOperator->Set = cmnMemSet;
+ mMemOperator->Check = cmnMemCheck;
+
+ VO_CODEC_INIT_USERDATA userData;
+ memset(&userData, 0, sizeof(userData));
+ userData.memflag = VO_IMF_USERMEMOPERATOR;
+ userData.memData = (VO_PTR) mMemOperator;
+ if (VO_ERR_NONE !=
+ mApiHandle->Init(&mEncoderHandle, VO_AUDIO_CodingAAC, &userData)) {
+ ALOGE("Failed to init AAC encoder");
+ return UNKNOWN_ERROR;
+ }
+
+ return OK;
+}
+
+OMX_ERRORTYPE SoftAACEncoder::internalGetParameter(
+ OMX_INDEXTYPE index, OMX_PTR params) {
+ switch (index) {
+ case OMX_IndexParamAudioPortFormat:
+ {
+ OMX_AUDIO_PARAM_PORTFORMATTYPE *formatParams =
+ (OMX_AUDIO_PARAM_PORTFORMATTYPE *)params;
+
+ if (formatParams->nPortIndex > 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (formatParams->nIndex > 0) {
+ return OMX_ErrorNoMore;
+ }
+
+ formatParams->eEncoding =
+ (formatParams->nPortIndex == 0)
+ ? OMX_AUDIO_CodingPCM : OMX_AUDIO_CodingAAC;
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioAac:
+ {
+ OMX_AUDIO_PARAM_AACPROFILETYPE *aacParams =
+ (OMX_AUDIO_PARAM_AACPROFILETYPE *)params;
+
+ if (aacParams->nPortIndex != 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ aacParams->nBitRate = mBitRate;
+ aacParams->nAudioBandWidth = 0;
+ aacParams->nAACtools = 0;
+ aacParams->nAACERtools = 0;
+ aacParams->eAACProfile = OMX_AUDIO_AACObjectMain;
+ aacParams->eAACStreamFormat = OMX_AUDIO_AACStreamFormatMP4FF;
+ aacParams->eChannelMode = OMX_AUDIO_ChannelModeStereo;
+
+ aacParams->nChannels = mNumChannels;
+ aacParams->nSampleRate = mSampleRate;
+ aacParams->nFrameLength = 0;
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioPcm:
+ {
+ OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
+ (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
+
+ if (pcmParams->nPortIndex != 0) {
+ return OMX_ErrorUndefined;
+ }
+
+ pcmParams->eNumData = OMX_NumericalDataSigned;
+ pcmParams->eEndian = OMX_EndianBig;
+ pcmParams->bInterleaved = OMX_TRUE;
+ pcmParams->nBitPerSample = 16;
+ pcmParams->ePCMMode = OMX_AUDIO_PCMModeLinear;
+ pcmParams->eChannelMapping[0] = OMX_AUDIO_ChannelLF;
+ pcmParams->eChannelMapping[1] = OMX_AUDIO_ChannelRF;
+
+ pcmParams->nChannels = mNumChannels;
+ pcmParams->nSamplingRate = mSampleRate;
+
+ return OMX_ErrorNone;
+ }
+
+ default:
+ return SimpleSoftOMXComponent::internalGetParameter(index, params);
+ }
+}
+
+OMX_ERRORTYPE SoftAACEncoder::internalSetParameter(
+ OMX_INDEXTYPE index, const OMX_PTR params) {
+ switch (index) {
+ case OMX_IndexParamStandardComponentRole:
+ {
+ const OMX_PARAM_COMPONENTROLETYPE *roleParams =
+ (const OMX_PARAM_COMPONENTROLETYPE *)params;
+
+ if (strncmp((const char *)roleParams->cRole,
+ "audio_encoder.aac",
+ OMX_MAX_STRINGNAME_SIZE - 1)) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioPortFormat:
+ {
+ const OMX_AUDIO_PARAM_PORTFORMATTYPE *formatParams =
+ (const OMX_AUDIO_PARAM_PORTFORMATTYPE *)params;
+
+ if (formatParams->nPortIndex > 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (formatParams->nIndex > 0) {
+ return OMX_ErrorNoMore;
+ }
+
+ if ((formatParams->nPortIndex == 0
+ && formatParams->eEncoding != OMX_AUDIO_CodingPCM)
+ || (formatParams->nPortIndex == 1
+ && formatParams->eEncoding != OMX_AUDIO_CodingAAC)) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioAac:
+ {
+ OMX_AUDIO_PARAM_AACPROFILETYPE *aacParams =
+ (OMX_AUDIO_PARAM_AACPROFILETYPE *)params;
+
+ if (aacParams->nPortIndex != 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ mBitRate = aacParams->nBitRate;
+ mNumChannels = aacParams->nChannels;
+ mSampleRate = aacParams->nSampleRate;
+
+ if (setAudioParams() != OK) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioPcm:
+ {
+ OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
+ (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
+
+ if (pcmParams->nPortIndex != 0) {
+ return OMX_ErrorUndefined;
+ }
+
+ mNumChannels = pcmParams->nChannels;
+ mSampleRate = pcmParams->nSamplingRate;
+
+ if (setAudioParams() != OK) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+
+ default:
+ return SimpleSoftOMXComponent::internalSetParameter(index, params);
+ }
+}
+
+status_t SoftAACEncoder::setAudioParams() {
+ // We call this whenever sample rate, number of channels or bitrate change
+ // in reponse to setParameter calls.
+
+ ALOGV("setAudioParams: %lu Hz, %lu channels, %lu bps",
+ mSampleRate, mNumChannels, mBitRate);
+
+ status_t err = setAudioSpecificConfigData();
+
+ if (err != OK) {
+ return err;
+ }
+
+ AACENC_PARAM params;
+ memset(¶ms, 0, sizeof(params));
+ params.sampleRate = mSampleRate;
+ params.bitRate = mBitRate;
+ params.nChannels = mNumChannels;
+ params.adtsUsed = 0; // We add adts header in the file writer if needed.
+ if (VO_ERR_NONE != mApiHandle->SetParam(
+ mEncoderHandle, VO_PID_AAC_ENCPARAM, ¶ms)) {
+ ALOGE("Failed to set AAC encoder parameters");
+ return UNKNOWN_ERROR;
+ }
+
+ return OK;
+}
+
+static status_t getSampleRateTableIndex(int32_t sampleRate, int32_t &index) {
+ static const int32_t kSampleRateTable[] = {
+ 96000, 88200, 64000, 48000, 44100, 32000,
+ 24000, 22050, 16000, 12000, 11025, 8000
+ };
+ const int32_t tableSize =
+ sizeof(kSampleRateTable) / sizeof(kSampleRateTable[0]);
+
+ for (int32_t i = 0; i < tableSize; ++i) {
+ if (sampleRate == kSampleRateTable[i]) {
+ index = i;
+ return OK;
+ }
+ }
+
+ return UNKNOWN_ERROR;
+}
+
+status_t SoftAACEncoder::setAudioSpecificConfigData() {
+ // The AAC encoder's audio specific config really only encodes
+ // number of channels and the sample rate (mapped to an index into
+ // a fixed sample rate table).
+
+ int32_t index;
+ status_t err = getSampleRateTableIndex(mSampleRate, index);
+ if (err != OK) {
+ ALOGE("Unsupported sample rate (%lu Hz)", mSampleRate);
+ return err;
+ }
+
+ if (mNumChannels > 2 || mNumChannels <= 0) {
+ ALOGE("Unsupported number of channels(%lu)", mNumChannels);
+ return UNKNOWN_ERROR;
+ }
+
+ // OMX_AUDIO_AACObjectLC
+ mAudioSpecificConfigData[0] = ((0x02 << 3) | (index >> 1));
+ mAudioSpecificConfigData[1] = ((index & 0x01) << 7) | (mNumChannels << 3);
+
+ return OK;
+}
+
+void SoftAACEncoder::onQueueFilled(OMX_U32 portIndex) {
+ if (mSignalledError) {
+ return;
+ }
+
+ List<BufferInfo *> &inQueue = getPortQueue(0);
+ List<BufferInfo *> &outQueue = getPortQueue(1);
+
+ if (!mSentCodecSpecificData) {
+ // The very first thing we want to output is the codec specific
+ // data. It does not require any input data but we will need an
+ // output buffer to store it in.
+
+ if (outQueue.empty()) {
+ return;
+ }
+
+ BufferInfo *outInfo = *outQueue.begin();
+ OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+ outHeader->nFilledLen = sizeof(mAudioSpecificConfigData);
+ outHeader->nFlags = OMX_BUFFERFLAG_CODECCONFIG;
+
+ uint8_t *out = outHeader->pBuffer + outHeader->nOffset;
+ memcpy(out, mAudioSpecificConfigData, sizeof(mAudioSpecificConfigData));
+
+#if 0
+ ALOGI("sending codec specific data.");
+ hexdump(out, sizeof(mAudioSpecificConfigData));
+#endif
+
+ outQueue.erase(outQueue.begin());
+ outInfo->mOwnedByUs = false;
+ notifyFillBufferDone(outHeader);
+
+ mSentCodecSpecificData = true;
+ }
+
+ size_t numBytesPerInputFrame =
+ mNumChannels * kNumSamplesPerFrame * sizeof(int16_t);
+
+ for (;;) {
+ // We do the following until we run out of buffers.
+
+ while (mInputSize < numBytesPerInputFrame) {
+ // As long as there's still input data to be read we
+ // will drain "kNumSamplesPerFrame * mNumChannels" samples
+ // into the "mInputFrame" buffer and then encode those
+ // as a unit into an output buffer.
+
+ if (mSawInputEOS || inQueue.empty()) {
+ return;
+ }
+
+ BufferInfo *inInfo = *inQueue.begin();
+ OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+
+ const void *inData = inHeader->pBuffer + inHeader->nOffset;
+
+ size_t copy = numBytesPerInputFrame - mInputSize;
+ if (copy > inHeader->nFilledLen) {
+ copy = inHeader->nFilledLen;
+ }
+
+ if (mInputFrame == NULL) {
+ mInputFrame = new int16_t[kNumSamplesPerFrame * mNumChannels];
+ }
+
+ if (mInputSize == 0) {
+ mInputTimeUs = inHeader->nTimeStamp;
+ }
+
+ memcpy((uint8_t *)mInputFrame + mInputSize, inData, copy);
+ mInputSize += copy;
+
+ inHeader->nOffset += copy;
+ inHeader->nFilledLen -= copy;
+
+ // "Time" on the input buffer has in effect advanced by the
+ // number of audio frames we just advanced nOffset by.
+ inHeader->nTimeStamp +=
+ (copy * 1000000ll / mSampleRate)
+ / (mNumChannels * sizeof(int16_t));
+
+ if (inHeader->nFilledLen == 0) {
+ if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
+ ALOGV("saw input EOS");
+ mSawInputEOS = true;
+
+ // Pad any remaining data with zeroes.
+ memset((uint8_t *)mInputFrame + mInputSize,
+ 0,
+ numBytesPerInputFrame - mInputSize);
+
+ mInputSize = numBytesPerInputFrame;
+ }
+
+ inQueue.erase(inQueue.begin());
+ inInfo->mOwnedByUs = false;
+ notifyEmptyBufferDone(inHeader);
+
+ inData = NULL;
+ inHeader = NULL;
+ inInfo = NULL;
+ }
+ }
+
+ // At this point we have all the input data necessary to encode
+ // a single frame, all we need is an output buffer to store the result
+ // in.
+
+ if (outQueue.empty()) {
+ return;
+ }
+
+ BufferInfo *outInfo = *outQueue.begin();
+ OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+
+ VO_CODECBUFFER inputData;
+ memset(&inputData, 0, sizeof(inputData));
+ inputData.Buffer = (unsigned char *)mInputFrame;
+ inputData.Length = numBytesPerInputFrame;
+ CHECK(VO_ERR_NONE ==
+ mApiHandle->SetInputData(mEncoderHandle, &inputData));
+
+ VO_CODECBUFFER outputData;
+ memset(&outputData, 0, sizeof(outputData));
+ VO_AUDIO_OUTPUTINFO outputInfo;
+ memset(&outputInfo, 0, sizeof(outputInfo));
+
+ uint8_t *outPtr = (uint8_t *)outHeader->pBuffer + outHeader->nOffset;
+ size_t outAvailable = outHeader->nAllocLen - outHeader->nOffset;
+
+ VO_U32 ret = VO_ERR_NONE;
+ size_t nOutputBytes = 0;
+ do {
+ outputData.Buffer = outPtr;
+ outputData.Length = outAvailable - nOutputBytes;
+ ret = mApiHandle->GetOutputData(
+ mEncoderHandle, &outputData, &outputInfo);
+ if (ret == VO_ERR_NONE) {
+ outPtr += outputData.Length;
+ nOutputBytes += outputData.Length;
+ }
+ } while (ret != VO_ERR_INPUT_BUFFER_SMALL);
+
+ outHeader->nFilledLen = nOutputBytes;
+
+ outHeader->nFlags = OMX_BUFFERFLAG_ENDOFFRAME;
+
+ if (mSawInputEOS) {
+ // We also tag this output buffer with EOS if it corresponds
+ // to the final input buffer.
+ outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+ }
+
+ outHeader->nTimeStamp = mInputTimeUs;
+
+#if 0
+ ALOGI("sending %d bytes of data (time = %lld us, flags = 0x%08lx)",
+ nOutputBytes, mInputTimeUs, outHeader->nFlags);
+
+ hexdump(outHeader->pBuffer + outHeader->nOffset, outHeader->nFilledLen);
+#endif
+
+ outQueue.erase(outQueue.begin());
+ outInfo->mOwnedByUs = false;
+ notifyFillBufferDone(outHeader);
+
+ outHeader = NULL;
+ outInfo = NULL;
+
+ mInputSize = 0;
+ }
+}
+
+} // namespace android
+
+android::SoftOMXComponent *createSoftOMXComponent(
+ const char *name, const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData, OMX_COMPONENTTYPE **component) {
+ return new android::SoftAACEncoder(name, callbacks, appData, component);
+}
diff --git a/media/libstagefright/codecs/aacenc/SoftAACEncoder.h b/media/libstagefright/codecs/aacenc/SoftAACEncoder.h
new file mode 100644
index 0000000..d148eb7
--- /dev/null
+++ b/media/libstagefright/codecs/aacenc/SoftAACEncoder.h
@@ -0,0 +1,82 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef SOFT_AAC_ENCODER_H_
+
+#define SOFT_AAC_ENCODER_H_
+
+#include "SimpleSoftOMXComponent.h"
+
+struct VO_AUDIO_CODECAPI;
+struct VO_MEM_OPERATOR;
+
+namespace android {
+
+struct SoftAACEncoder : public SimpleSoftOMXComponent {
+ SoftAACEncoder(
+ const char *name,
+ const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData,
+ OMX_COMPONENTTYPE **component);
+
+protected:
+ virtual ~SoftAACEncoder();
+
+ virtual OMX_ERRORTYPE internalGetParameter(
+ OMX_INDEXTYPE index, OMX_PTR params);
+
+ virtual OMX_ERRORTYPE internalSetParameter(
+ OMX_INDEXTYPE index, const OMX_PTR params);
+
+ virtual void onQueueFilled(OMX_U32 portIndex);
+
+private:
+ enum {
+ kNumBuffers = 4,
+ kNumSamplesPerFrame = 1024,
+ };
+
+ void *mEncoderHandle;
+ VO_AUDIO_CODECAPI *mApiHandle;
+ VO_MEM_OPERATOR *mMemOperator;
+
+ OMX_U32 mNumChannels;
+ OMX_U32 mSampleRate;
+ OMX_U32 mBitRate;
+
+ bool mSentCodecSpecificData;
+ size_t mInputSize;
+ int16_t *mInputFrame;
+ int64_t mInputTimeUs;
+
+ bool mSawInputEOS;
+
+ uint8_t mAudioSpecificConfigData[2];
+
+ bool mSignalledError;
+
+ void initPorts();
+ status_t initEncoder();
+
+ status_t setAudioSpecificConfigData();
+ status_t setAudioParams();
+
+ DISALLOW_EVIL_CONSTRUCTORS(SoftAACEncoder);
+};
+
+} // namespace android
+
+#endif // SOFT_AAC_ENCODER_H_
diff --git a/media/libstagefright/codecs/aacenc/src/asm/ARMV7/PrePostMDCT_v7.s b/media/libstagefright/codecs/aacenc/src/asm/ARMV7/PrePostMDCT_v7.s
index b2bc9d9..7f6b881 100644
--- a/media/libstagefright/codecs/aacenc/src/asm/ARMV7/PrePostMDCT_v7.s
+++ b/media/libstagefright/codecs/aacenc/src/asm/ARMV7/PrePostMDCT_v7.s
@@ -23,9 +23,13 @@
.section .text
.global PreMDCT
+ .fnstart
PreMDCT:
stmdb sp!, {r4 - r11, lr}
+ .save {r4 - r11, lr}
+ fstmfdd sp!, {d8 - d15}
+ .vsave {d8 - d15}
add r9, r0, r1, lsl #2
sub r3, r9, #32
@@ -74,14 +78,20 @@
bne PreMDCT_LOOP
PreMDCT_END:
+ fldmfdd sp!, {d8 - d15}
ldmia sp!, {r4 - r11, pc}
@ENDP @ |PreMDCT|
+ .fnend
.section .text
.global PostMDCT
+ .fnstart
PostMDCT:
stmdb sp!, {r4 - r11, lr}
+ .save {r4 - r11, lr}
+ fstmfdd sp!, {d8 - d15}
+ .vsave {d8 - d15}
add r9, r0, r1, lsl #2
sub r3, r9, #32
@@ -129,7 +139,8 @@
bne PostMDCT_LOOP
PostMDCT_END:
+ fldmfdd sp!, {d8 - d15}
ldmia sp!, {r4 - r11, pc}
@ENDP @ |PostMDCT|
- .end
+ .fnend
diff --git a/media/libstagefright/codecs/aacenc/src/asm/ARMV7/R4R8First_v7.s b/media/libstagefright/codecs/aacenc/src/asm/ARMV7/R4R8First_v7.s
index 3033156..03fa6a9 100644
--- a/media/libstagefright/codecs/aacenc/src/asm/ARMV7/R4R8First_v7.s
+++ b/media/libstagefright/codecs/aacenc/src/asm/ARMV7/R4R8First_v7.s
@@ -23,9 +23,13 @@
.section .text
.global Radix8First
+ .fnstart
Radix8First:
stmdb sp!, {r4 - r11, lr}
+ .save {r4 - r11, lr}
+ fstmfdd sp!, {d8 - d15}
+ .vsave {d8 - d15}
ldr r3, SQRT1_2
cmp r1, #0
@@ -103,17 +107,23 @@
bne Radix8First_LOOP
Radix8First_END:
+ fldmfdd sp!, {d8 - d15}
ldmia sp!, {r4 - r11, pc}
SQRT1_2:
.word 0x2d413ccd
@ENDP @ |Radix8First|
+ .fnend
.section .text
.global Radix4First
+ .fnstart
Radix4First:
stmdb sp!, {r4 - r11, lr}
+ .save {r4 - r11, lr}
+ fstmfdd sp!, {d8 - d15}
+ .vsave {d8 - d15}
cmp r1, #0
beq Radix4First_END
@@ -140,7 +150,8 @@
bne Radix4First_LOOP
Radix4First_END:
+ fldmfdd sp!, {d8 - d15}
ldmia sp!, {r4 - r11, pc}
@ENDP @ |Radix4First|
- .end
+ .fnend
diff --git a/media/libstagefright/codecs/aacenc/src/asm/ARMV7/Radix4FFT_v7.s b/media/libstagefright/codecs/aacenc/src/asm/ARMV7/Radix4FFT_v7.s
index f874825..431bc30 100644
--- a/media/libstagefright/codecs/aacenc/src/asm/ARMV7/Radix4FFT_v7.s
+++ b/media/libstagefright/codecs/aacenc/src/asm/ARMV7/Radix4FFT_v7.s
@@ -23,9 +23,13 @@
.section .text
.global Radix4FFT
+ .fnstart
Radix4FFT:
stmdb sp!, {r4 - r11, lr}
+ .save {r4 - r11, lr}
+ fstmfdd sp!, {d8 - d15}
+ .vsave {d8 - d15}
mov r1, r1, asr #2
cmp r1, #0
@@ -137,7 +141,8 @@
bne Radix4FFT_LOOP1
Radix4FFT_END:
+ fldmfdd sp!, {d8 - d15}
ldmia sp!, {r4 - r11, pc}
@ENDP @ |Radix4FFT|
- .end
+ .fnend
diff --git a/media/libstagefright/codecs/amrnb/dec/Android.mk b/media/libstagefright/codecs/amrnb/dec/Android.mk
index 23a22ef..b81306d 100644
--- a/media/libstagefright/codecs/amrnb/dec/Android.mk
+++ b/media/libstagefright/codecs/amrnb/dec/Android.mk
@@ -61,12 +61,12 @@
LOCAL_C_INCLUDES := \
frameworks/base/media/libstagefright/include \
- frameworks/base/include/media/stagefright/openmax \
+ frameworks/base/media/libstagefright/codecs/amrwb/src \
+ frameworks/native/include/media/openmax \
$(LOCAL_PATH)/src \
$(LOCAL_PATH)/include \
$(LOCAL_PATH)/../common/include \
- $(LOCAL_PATH)/../common \
- frameworks/base/media/libstagefright/codecs/amrwb/src \
+ $(LOCAL_PATH)/../common
LOCAL_CFLAGS := -DOSCL_IMPORT_REF=
diff --git a/media/libstagefright/codecs/amrnb/dec/SoftAMR.cpp b/media/libstagefright/codecs/amrnb/dec/SoftAMR.cpp
index 7602f2d..796caa4 100644
--- a/media/libstagefright/codecs/amrnb/dec/SoftAMR.cpp
+++ b/media/libstagefright/codecs/amrnb/dec/SoftAMR.cpp
@@ -236,6 +236,18 @@
return OMX_ErrorNone;
}
+ case OMX_IndexParamAudioPcm:
+ {
+ const OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
+ (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
+
+ if (pcmParams->nPortIndex != 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
default:
return SimpleSoftOMXComponent::internalSetParameter(index, params);
}
diff --git a/media/libstagefright/codecs/amrnb/enc/AMRNBEncoder.cpp b/media/libstagefright/codecs/amrnb/enc/AMRNBEncoder.cpp
deleted file mode 100644
index 3afbc4f..0000000
--- a/media/libstagefright/codecs/amrnb/enc/AMRNBEncoder.cpp
+++ /dev/null
@@ -1,249 +0,0 @@
-/*
- * Copyright (C) 2009 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include "AMRNBEncoder.h"
-
-#include "gsmamr_enc.h"
-
-#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
-#include <media/stagefright/MediaDefs.h>
-#include <media/stagefright/MediaErrors.h>
-#include <media/stagefright/MetaData.h>
-
-namespace android {
-
-static const int32_t kNumSamplesPerFrame = 160;
-static const int32_t kSampleRate = 8000;
-
-AMRNBEncoder::AMRNBEncoder(const sp<MediaSource> &source, const sp<MetaData> &meta)
- : mSource(source),
- mMeta(meta),
- mStarted(false),
- mBufferGroup(NULL),
- mEncState(NULL),
- mSidState(NULL),
- mAnchorTimeUs(0),
- mNumFramesOutput(0),
- mInputBuffer(NULL),
- mMode(MR475),
- mNumInputSamples(0) {
-}
-
-AMRNBEncoder::~AMRNBEncoder() {
- if (mStarted) {
- stop();
- }
-}
-
-static Mode PickModeFromBitrate(int32_t bps) {
- if (bps <= 4750) {
- return MR475;
- } else if (bps <= 5150) {
- return MR515;
- } else if (bps <= 5900) {
- return MR59;
- } else if (bps <= 6700) {
- return MR67;
- } else if (bps <= 7400) {
- return MR74;
- } else if (bps <= 7950) {
- return MR795;
- } else if (bps <= 10200) {
- return MR102;
- } else {
- return MR122;
- }
-}
-
-status_t AMRNBEncoder::start(MetaData *params) {
- if (mStarted) {
- ALOGW("Call start() when encoder already started");
- return OK;
- }
-
- mBufferGroup = new MediaBufferGroup;
- mBufferGroup->add_buffer(new MediaBuffer(32));
-
- CHECK_EQ(AMREncodeInit(
- &mEncState, &mSidState, false /* dtx_enable */),
- 0);
-
- status_t err = mSource->start(params);
- if (err != OK) {
- ALOGE("AudioSource is not available");
- return err;
- }
-
- mAnchorTimeUs = 0;
- mNumFramesOutput = 0;
- mStarted = true;
- mNumInputSamples = 0;
-
- int32_t bitrate;
- if (params && params->findInt32(kKeyBitRate, &bitrate)) {
- mMode = PickModeFromBitrate(bitrate);
- } else {
- mMode = MR475;
- }
-
- return OK;
-}
-
-status_t AMRNBEncoder::stop() {
- if (!mStarted) {
- ALOGW("Call stop() when encoder has not started.");
- return OK;
- }
-
- if (mInputBuffer) {
- mInputBuffer->release();
- mInputBuffer = NULL;
- }
-
- delete mBufferGroup;
- mBufferGroup = NULL;
-
- mSource->stop();
-
- AMREncodeExit(&mEncState, &mSidState);
- mEncState = mSidState = NULL;
-
- mStarted = false;
-
- return OK;
-}
-
-sp<MetaData> AMRNBEncoder::getFormat() {
- sp<MetaData> srcFormat = mSource->getFormat();
-
- mMeta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_AMR_NB);
-
- int64_t durationUs;
- if (srcFormat->findInt64(kKeyDuration, &durationUs)) {
- mMeta->setInt64(kKeyDuration, durationUs);
- }
-
- mMeta->setCString(kKeyDecoderComponent, "AMRNBEncoder");
-
- return mMeta;
-}
-
-status_t AMRNBEncoder::read(
- MediaBuffer **out, const ReadOptions *options) {
- status_t err;
-
- *out = NULL;
-
- int64_t seekTimeUs;
- ReadOptions::SeekMode mode;
- CHECK(options == NULL || !options->getSeekTo(&seekTimeUs, &mode));
- bool readFromSource = false;
- int64_t wallClockTimeUs = -1;
-
- while (mNumInputSamples < kNumSamplesPerFrame) {
- if (mInputBuffer == NULL) {
- err = mSource->read(&mInputBuffer, options);
-
- if (err != OK) {
- if (mNumInputSamples == 0) {
- return ERROR_END_OF_STREAM;
- }
- memset(&mInputFrame[mNumInputSamples],
- 0,
- sizeof(int16_t)
- * (kNumSamplesPerFrame - mNumInputSamples));
- mNumInputSamples = kNumSamplesPerFrame;
- break;
- }
-
- size_t align = mInputBuffer->range_length() % sizeof(int16_t);
- CHECK_EQ(align, 0);
- readFromSource = true;
-
- int64_t timeUs;
- if (mInputBuffer->meta_data()->findInt64(kKeyDriftTime, &timeUs)) {
- wallClockTimeUs = timeUs;
- }
- if (mInputBuffer->meta_data()->findInt64(kKeyAnchorTime, &timeUs)) {
- mAnchorTimeUs = timeUs;
- }
- } else {
- readFromSource = false;
- }
-
- size_t copy =
- (kNumSamplesPerFrame - mNumInputSamples) * sizeof(int16_t);
-
- if (copy > mInputBuffer->range_length()) {
- copy = mInputBuffer->range_length();
- }
-
- memcpy(&mInputFrame[mNumInputSamples],
- (const uint8_t *)mInputBuffer->data()
- + mInputBuffer->range_offset(),
- copy);
-
- mNumInputSamples += copy / sizeof(int16_t);
-
- mInputBuffer->set_range(
- mInputBuffer->range_offset() + copy,
- mInputBuffer->range_length() - copy);
-
- if (mInputBuffer->range_length() == 0) {
- mInputBuffer->release();
- mInputBuffer = NULL;
- }
- }
-
- MediaBuffer *buffer;
- CHECK_EQ(mBufferGroup->acquire_buffer(&buffer), OK);
-
- uint8_t *outPtr = (uint8_t *)buffer->data();
-
- Frame_Type_3GPP frameType;
- int res = AMREncode(
- mEncState, mSidState, (Mode)mMode,
- mInputFrame, outPtr, &frameType, AMR_TX_WMF);
-
- CHECK(res >= 0);
- CHECK((size_t)res < buffer->size());
-
- // Convert header byte from WMF to IETF format.
- outPtr[0] = ((outPtr[0] << 3) | 4) & 0x7c;
-
- buffer->set_range(0, res);
-
- // Each frame of 160 samples is 20ms long.
- int64_t mediaTimeUs = mNumFramesOutput * 20000LL;
- buffer->meta_data()->setInt64(
- kKeyTime, mAnchorTimeUs + mediaTimeUs);
-
- if (readFromSource && wallClockTimeUs != -1) {
- buffer->meta_data()->setInt64(kKeyDriftTime,
- mediaTimeUs - wallClockTimeUs);
- }
-
- ++mNumFramesOutput;
-
- *out = buffer;
-
- mNumInputSamples = 0;
-
- return OK;
-}
-
-} // namespace android
diff --git a/media/libstagefright/codecs/amrnb/enc/Android.mk b/media/libstagefright/codecs/amrnb/enc/Android.mk
index b6aed81..28246ae 100644
--- a/media/libstagefright/codecs/amrnb/enc/Android.mk
+++ b/media/libstagefright/codecs/amrnb/enc/Android.mk
@@ -2,7 +2,6 @@
include $(CLEAR_VARS)
LOCAL_SRC_FILES := \
- AMRNBEncoder.cpp \
src/amrencode.cpp \
src/autocorr.cpp \
src/c1035pf.cpp \
@@ -74,3 +73,30 @@
LOCAL_MODULE := libstagefright_amrnbenc
include $(BUILD_STATIC_LIBRARY)
+
+################################################################################
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES := \
+ SoftAMRNBEncoder.cpp
+
+LOCAL_C_INCLUDES := \
+ frameworks/base/media/libstagefright/include \
+ frameworks/native/include/media/openmax \
+ $(LOCAL_PATH)/src \
+ $(LOCAL_PATH)/include \
+ $(LOCAL_PATH)/../common/include \
+ $(LOCAL_PATH)/../common
+
+LOCAL_STATIC_LIBRARIES := \
+ libstagefright_amrnbenc
+
+LOCAL_SHARED_LIBRARIES := \
+ libstagefright_omx libstagefright_foundation libutils \
+ libstagefright_amrnb_common
+
+LOCAL_MODULE := libstagefright_soft_amrnbenc
+LOCAL_MODULE_TAGS := optional
+
+include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libstagefright/codecs/amrnb/enc/SoftAMRNBEncoder.cpp b/media/libstagefright/codecs/amrnb/enc/SoftAMRNBEncoder.cpp
new file mode 100644
index 0000000..07f8b4f
--- /dev/null
+++ b/media/libstagefright/codecs/amrnb/enc/SoftAMRNBEncoder.cpp
@@ -0,0 +1,404 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "SoftAMRNBEncoder"
+#include <utils/Log.h>
+
+#include "SoftAMRNBEncoder.h"
+
+#include "gsmamr_enc.h"
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/hexdump.h>
+
+namespace android {
+
+static const int32_t kSampleRate = 8000;
+
+template<class T>
+static void InitOMXParams(T *params) {
+ params->nSize = sizeof(T);
+ params->nVersion.s.nVersionMajor = 1;
+ params->nVersion.s.nVersionMinor = 0;
+ params->nVersion.s.nRevision = 0;
+ params->nVersion.s.nStep = 0;
+}
+
+SoftAMRNBEncoder::SoftAMRNBEncoder(
+ const char *name,
+ const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData,
+ OMX_COMPONENTTYPE **component)
+ : SimpleSoftOMXComponent(name, callbacks, appData, component),
+ mEncState(NULL),
+ mSidState(NULL),
+ mBitRate(0),
+ mMode(MR475),
+ mInputSize(0),
+ mInputTimeUs(-1ll),
+ mSawInputEOS(false),
+ mSignalledError(false) {
+ initPorts();
+ CHECK_EQ(initEncoder(), (status_t)OK);
+}
+
+SoftAMRNBEncoder::~SoftAMRNBEncoder() {
+ if (mEncState != NULL) {
+ AMREncodeExit(&mEncState, &mSidState);
+ mEncState = mSidState = NULL;
+ }
+}
+
+void SoftAMRNBEncoder::initPorts() {
+ OMX_PARAM_PORTDEFINITIONTYPE def;
+ InitOMXParams(&def);
+
+ def.nPortIndex = 0;
+ def.eDir = OMX_DirInput;
+ def.nBufferCountMin = kNumBuffers;
+ def.nBufferCountActual = def.nBufferCountMin;
+ def.nBufferSize = kNumSamplesPerFrame * sizeof(int16_t);
+ def.bEnabled = OMX_TRUE;
+ def.bPopulated = OMX_FALSE;
+ def.eDomain = OMX_PortDomainAudio;
+ def.bBuffersContiguous = OMX_FALSE;
+ def.nBufferAlignment = 1;
+
+ def.format.audio.cMIMEType = const_cast<char *>("audio/raw");
+ def.format.audio.pNativeRender = NULL;
+ def.format.audio.bFlagErrorConcealment = OMX_FALSE;
+ def.format.audio.eEncoding = OMX_AUDIO_CodingPCM;
+
+ addPort(def);
+
+ def.nPortIndex = 1;
+ def.eDir = OMX_DirOutput;
+ def.nBufferCountMin = kNumBuffers;
+ def.nBufferCountActual = def.nBufferCountMin;
+ def.nBufferSize = 8192;
+ def.bEnabled = OMX_TRUE;
+ def.bPopulated = OMX_FALSE;
+ def.eDomain = OMX_PortDomainAudio;
+ def.bBuffersContiguous = OMX_FALSE;
+ def.nBufferAlignment = 2;
+
+ def.format.audio.cMIMEType = const_cast<char *>("audio/3gpp");
+ def.format.audio.pNativeRender = NULL;
+ def.format.audio.bFlagErrorConcealment = OMX_FALSE;
+ def.format.audio.eEncoding = OMX_AUDIO_CodingAMR;
+
+ addPort(def);
+}
+
+status_t SoftAMRNBEncoder::initEncoder() {
+ if (AMREncodeInit(&mEncState, &mSidState, false /* dtx_enable */) != 0) {
+ return UNKNOWN_ERROR;
+ }
+
+ return OK;
+}
+
+OMX_ERRORTYPE SoftAMRNBEncoder::internalGetParameter(
+ OMX_INDEXTYPE index, OMX_PTR params) {
+ switch (index) {
+ case OMX_IndexParamAudioPortFormat:
+ {
+ OMX_AUDIO_PARAM_PORTFORMATTYPE *formatParams =
+ (OMX_AUDIO_PARAM_PORTFORMATTYPE *)params;
+
+ if (formatParams->nPortIndex > 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (formatParams->nIndex > 0) {
+ return OMX_ErrorNoMore;
+ }
+
+ formatParams->eEncoding =
+ (formatParams->nPortIndex == 0)
+ ? OMX_AUDIO_CodingPCM : OMX_AUDIO_CodingAMR;
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioAmr:
+ {
+ OMX_AUDIO_PARAM_AMRTYPE *amrParams =
+ (OMX_AUDIO_PARAM_AMRTYPE *)params;
+
+ if (amrParams->nPortIndex != 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ amrParams->nChannels = 1;
+ amrParams->nBitRate = mBitRate;
+ amrParams->eAMRBandMode = (OMX_AUDIO_AMRBANDMODETYPE)(mMode + 1);
+ amrParams->eAMRDTXMode = OMX_AUDIO_AMRDTXModeOff;
+ amrParams->eAMRFrameFormat = OMX_AUDIO_AMRFrameFormatFSF;
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioPcm:
+ {
+ OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
+ (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
+
+ if (pcmParams->nPortIndex != 0) {
+ return OMX_ErrorUndefined;
+ }
+
+ pcmParams->eNumData = OMX_NumericalDataSigned;
+ pcmParams->eEndian = OMX_EndianBig;
+ pcmParams->bInterleaved = OMX_TRUE;
+ pcmParams->nBitPerSample = 16;
+ pcmParams->ePCMMode = OMX_AUDIO_PCMModeLinear;
+ pcmParams->eChannelMapping[0] = OMX_AUDIO_ChannelCF;
+
+ pcmParams->nChannels = 1;
+ pcmParams->nSamplingRate = kSampleRate;
+
+ return OMX_ErrorNone;
+ }
+
+ default:
+ return SimpleSoftOMXComponent::internalGetParameter(index, params);
+ }
+}
+
+OMX_ERRORTYPE SoftAMRNBEncoder::internalSetParameter(
+ OMX_INDEXTYPE index, const OMX_PTR params) {
+ switch (index) {
+ case OMX_IndexParamStandardComponentRole:
+ {
+ const OMX_PARAM_COMPONENTROLETYPE *roleParams =
+ (const OMX_PARAM_COMPONENTROLETYPE *)params;
+
+ if (strncmp((const char *)roleParams->cRole,
+ "audio_encoder.amrnb",
+ OMX_MAX_STRINGNAME_SIZE - 1)) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioPortFormat:
+ {
+ const OMX_AUDIO_PARAM_PORTFORMATTYPE *formatParams =
+ (const OMX_AUDIO_PARAM_PORTFORMATTYPE *)params;
+
+ if (formatParams->nPortIndex > 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (formatParams->nIndex > 0) {
+ return OMX_ErrorNoMore;
+ }
+
+ if ((formatParams->nPortIndex == 0
+ && formatParams->eEncoding != OMX_AUDIO_CodingPCM)
+ || (formatParams->nPortIndex == 1
+ && formatParams->eEncoding != OMX_AUDIO_CodingAMR)) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioAmr:
+ {
+ OMX_AUDIO_PARAM_AMRTYPE *amrParams =
+ (OMX_AUDIO_PARAM_AMRTYPE *)params;
+
+ if (amrParams->nPortIndex != 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (amrParams->nChannels != 1
+ || amrParams->eAMRDTXMode != OMX_AUDIO_AMRDTXModeOff
+ || amrParams->eAMRFrameFormat
+ != OMX_AUDIO_AMRFrameFormatFSF
+ || amrParams->eAMRBandMode < OMX_AUDIO_AMRBandModeNB0
+ || amrParams->eAMRBandMode > OMX_AUDIO_AMRBandModeNB7) {
+ return OMX_ErrorUndefined;
+ }
+
+ mBitRate = amrParams->nBitRate;
+ mMode = amrParams->eAMRBandMode - 1;
+
+ amrParams->eAMRDTXMode = OMX_AUDIO_AMRDTXModeOff;
+ amrParams->eAMRFrameFormat = OMX_AUDIO_AMRFrameFormatFSF;
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioPcm:
+ {
+ OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
+ (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
+
+ if (pcmParams->nPortIndex != 0) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (pcmParams->nChannels != 1
+ || pcmParams->nSamplingRate != kSampleRate) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+
+ default:
+ return SimpleSoftOMXComponent::internalSetParameter(index, params);
+ }
+}
+
+void SoftAMRNBEncoder::onQueueFilled(OMX_U32 portIndex) {
+ if (mSignalledError) {
+ return;
+ }
+
+ List<BufferInfo *> &inQueue = getPortQueue(0);
+ List<BufferInfo *> &outQueue = getPortQueue(1);
+
+ size_t numBytesPerInputFrame = kNumSamplesPerFrame * sizeof(int16_t);
+
+ for (;;) {
+ // We do the following until we run out of buffers.
+
+ while (mInputSize < numBytesPerInputFrame) {
+ // As long as there's still input data to be read we
+ // will drain "kNumSamplesPerFrame" samples
+ // into the "mInputFrame" buffer and then encode those
+ // as a unit into an output buffer.
+
+ if (mSawInputEOS || inQueue.empty()) {
+ return;
+ }
+
+ BufferInfo *inInfo = *inQueue.begin();
+ OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+
+ const void *inData = inHeader->pBuffer + inHeader->nOffset;
+
+ size_t copy = numBytesPerInputFrame - mInputSize;
+ if (copy > inHeader->nFilledLen) {
+ copy = inHeader->nFilledLen;
+ }
+
+ if (mInputSize == 0) {
+ mInputTimeUs = inHeader->nTimeStamp;
+ }
+
+ memcpy((uint8_t *)mInputFrame + mInputSize, inData, copy);
+ mInputSize += copy;
+
+ inHeader->nOffset += copy;
+ inHeader->nFilledLen -= copy;
+
+ // "Time" on the input buffer has in effect advanced by the
+ // number of audio frames we just advanced nOffset by.
+ inHeader->nTimeStamp +=
+ (copy * 1000000ll / kSampleRate) / sizeof(int16_t);
+
+ if (inHeader->nFilledLen == 0) {
+ if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
+ ALOGV("saw input EOS");
+ mSawInputEOS = true;
+
+ // Pad any remaining data with zeroes.
+ memset((uint8_t *)mInputFrame + mInputSize,
+ 0,
+ numBytesPerInputFrame - mInputSize);
+
+ mInputSize = numBytesPerInputFrame;
+ }
+
+ inQueue.erase(inQueue.begin());
+ inInfo->mOwnedByUs = false;
+ notifyEmptyBufferDone(inHeader);
+
+ inData = NULL;
+ inHeader = NULL;
+ inInfo = NULL;
+ }
+ }
+
+ // At this point we have all the input data necessary to encode
+ // a single frame, all we need is an output buffer to store the result
+ // in.
+
+ if (outQueue.empty()) {
+ return;
+ }
+
+ BufferInfo *outInfo = *outQueue.begin();
+ OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+
+ uint8_t *outPtr = outHeader->pBuffer + outHeader->nOffset;
+ size_t outAvailable = outHeader->nAllocLen - outHeader->nOffset;
+
+ Frame_Type_3GPP frameType;
+ int res = AMREncode(
+ mEncState, mSidState, (Mode)mMode,
+ mInputFrame, outPtr, &frameType, AMR_TX_WMF);
+
+ CHECK_GE(res, 0);
+ CHECK_LE((size_t)res, outAvailable);
+
+ // Convert header byte from WMF to IETF format.
+ outPtr[0] = ((outPtr[0] << 3) | 4) & 0x7c;
+
+ outHeader->nFilledLen = res;
+ outHeader->nFlags = OMX_BUFFERFLAG_ENDOFFRAME;
+
+ if (mSawInputEOS) {
+ // We also tag this output buffer with EOS if it corresponds
+ // to the final input buffer.
+ outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+ }
+
+ outHeader->nTimeStamp = mInputTimeUs;
+
+#if 0
+ ALOGI("sending %d bytes of data (time = %lld us, flags = 0x%08lx)",
+ nOutputBytes, mInputTimeUs, outHeader->nFlags);
+
+ hexdump(outHeader->pBuffer + outHeader->nOffset, outHeader->nFilledLen);
+#endif
+
+ outQueue.erase(outQueue.begin());
+ outInfo->mOwnedByUs = false;
+ notifyFillBufferDone(outHeader);
+
+ outHeader = NULL;
+ outInfo = NULL;
+
+ mInputSize = 0;
+ }
+}
+
+} // namespace android
+
+android::SoftOMXComponent *createSoftOMXComponent(
+ const char *name, const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData, OMX_COMPONENTTYPE **component) {
+ return new android::SoftAMRNBEncoder(name, callbacks, appData, component);
+}
diff --git a/media/libstagefright/codecs/amrnb/enc/SoftAMRNBEncoder.h b/media/libstagefright/codecs/amrnb/enc/SoftAMRNBEncoder.h
new file mode 100644
index 0000000..50178c4
--- /dev/null
+++ b/media/libstagefright/codecs/amrnb/enc/SoftAMRNBEncoder.h
@@ -0,0 +1,72 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef SOFT_AMRNB_ENCODER_H_
+
+#define SOFT_AMRNB_ENCODER_H_
+
+#include "SimpleSoftOMXComponent.h"
+
+namespace android {
+
+struct SoftAMRNBEncoder : public SimpleSoftOMXComponent {
+ SoftAMRNBEncoder(
+ const char *name,
+ const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData,
+ OMX_COMPONENTTYPE **component);
+
+protected:
+ virtual ~SoftAMRNBEncoder();
+
+ virtual OMX_ERRORTYPE internalGetParameter(
+ OMX_INDEXTYPE index, OMX_PTR params);
+
+ virtual OMX_ERRORTYPE internalSetParameter(
+ OMX_INDEXTYPE index, const OMX_PTR params);
+
+ virtual void onQueueFilled(OMX_U32 portIndex);
+
+private:
+ enum {
+ kNumBuffers = 4,
+ kNumSamplesPerFrame = 160,
+ };
+
+ void *mEncState;
+ void *mSidState;
+
+ OMX_U32 mBitRate;
+ int mMode;
+
+ size_t mInputSize;
+ int16_t mInputFrame[kNumSamplesPerFrame];
+ int64_t mInputTimeUs;
+
+ bool mSawInputEOS;
+ bool mSignalledError;
+
+ void initPorts();
+ status_t initEncoder();
+
+ status_t setAudioParams();
+
+ DISALLOW_EVIL_CONSTRUCTORS(SoftAMRNBEncoder);
+};
+
+} // namespace android
+
+#endif // SOFT_AMRNB_ENCODER_H_
diff --git a/media/libstagefright/codecs/amrwbenc/AMRWBEncoder.cpp b/media/libstagefright/codecs/amrwbenc/AMRWBEncoder.cpp
deleted file mode 100644
index 60b1163..0000000
--- a/media/libstagefright/codecs/amrwbenc/AMRWBEncoder.cpp
+++ /dev/null
@@ -1,301 +0,0 @@
-/*
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "AMRWBEncoder"
-#include <utils/Log.h>
-
-#include "AMRWBEncoder.h"
-#include "voAMRWB.h"
-#include "cmnMemory.h"
-
-#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
-#include <media/stagefright/MediaDefs.h>
-#include <media/stagefright/MediaErrors.h>
-#include <media/stagefright/MetaData.h>
-
-namespace android {
-
-static const int32_t kNumSamplesPerFrame = 320;
-static const int32_t kBitsPerSample = 16;
-static const int32_t kInputBufferSize = (kBitsPerSample / 8) * kNumSamplesPerFrame;
-static const int32_t kSampleRate = 16000;
-static const int32_t kNumChannels = 1;
-
-AMRWBEncoder::AMRWBEncoder(const sp<MediaSource> &source, const sp<MetaData> &meta)
- : mSource(source),
- mMeta(meta),
- mStarted(false),
- mBufferGroup(NULL),
- mInputBuffer(NULL),
- mEncoderHandle(NULL),
- mApiHandle(NULL),
- mMemOperator(NULL),
- mAnchorTimeUs(0),
- mNumFramesOutput(0),
- mNumInputSamples(0) {
-}
-
-static VOAMRWBMODE pickModeFromBitRate(int32_t bps) {
- CHECK(bps >= 0);
- if (bps <= 6600) {
- return VOAMRWB_MD66;
- } else if (bps <= 8850) {
- return VOAMRWB_MD885;
- } else if (bps <= 12650) {
- return VOAMRWB_MD1265;
- } else if (bps <= 14250) {
- return VOAMRWB_MD1425;
- } else if (bps <= 15850) {
- return VOAMRWB_MD1585;
- } else if (bps <= 18250) {
- return VOAMRWB_MD1825;
- } else if (bps <= 19850) {
- return VOAMRWB_MD1985;
- } else if (bps <= 23050) {
- return VOAMRWB_MD2305;
- }
- return VOAMRWB_MD2385;
-}
-
-status_t AMRWBEncoder::initCheck() {
- CHECK(mApiHandle == NULL && mEncoderHandle == NULL);
- CHECK(mMeta->findInt32(kKeyBitRate, &mBitRate));
-
- mApiHandle = new VO_AUDIO_CODECAPI;
- CHECK(mApiHandle);
-
- if (VO_ERR_NONE != voGetAMRWBEncAPI(mApiHandle)) {
- ALOGE("Failed to get api handle");
- return UNKNOWN_ERROR;
- }
-
- mMemOperator = new VO_MEM_OPERATOR;
- CHECK(mMemOperator != NULL);
- mMemOperator->Alloc = cmnMemAlloc;
- mMemOperator->Copy = cmnMemCopy;
- mMemOperator->Free = cmnMemFree;
- mMemOperator->Set = cmnMemSet;
- mMemOperator->Check = cmnMemCheck;
-
- VO_CODEC_INIT_USERDATA userData;
- memset(&userData, 0, sizeof(userData));
- userData.memflag = VO_IMF_USERMEMOPERATOR;
- userData.memData = (VO_PTR) mMemOperator;
- if (VO_ERR_NONE != mApiHandle->Init(&mEncoderHandle, VO_AUDIO_CodingAMRWB, &userData)) {
- ALOGE("Failed to init AMRWB encoder");
- return UNKNOWN_ERROR;
- }
-
- // Configure AMRWB encoder$
- VOAMRWBMODE mode = pickModeFromBitRate(mBitRate);
- if (VO_ERR_NONE != mApiHandle->SetParam(mEncoderHandle, VO_PID_AMRWB_MODE, &mode)) {
- ALOGE("Failed to set AMRWB encoder mode to %d", mode);
- return UNKNOWN_ERROR;
- }
-
- VOAMRWBFRAMETYPE type = VOAMRWB_RFC3267;
- if (VO_ERR_NONE != mApiHandle->SetParam(mEncoderHandle, VO_PID_AMRWB_FRAMETYPE, &type)) {
- ALOGE("Failed to set AMRWB encoder frame type to %d", type);
- return UNKNOWN_ERROR;
- }
-
- return OK;
-}
-
-AMRWBEncoder::~AMRWBEncoder() {
- if (mStarted) {
- stop();
- }
-}
-
-status_t AMRWBEncoder::start(MetaData *params) {
- if (mStarted) {
- ALOGW("Call start() when encoder already started");
- return OK;
- }
-
- mBufferGroup = new MediaBufferGroup;
-
- // The largest buffer size is header + 477 bits
- mBufferGroup->add_buffer(new MediaBuffer(1024));
-
- CHECK_EQ(OK, initCheck());
-
- mNumFramesOutput = 0;
-
- status_t err = mSource->start(params);
- if (err != OK) {
- ALOGE("AudioSource is not available");
- return err;
- }
- mStarted = true;
-
- return OK;
-}
-
-status_t AMRWBEncoder::stop() {
- if (!mStarted) {
- ALOGW("Call stop() when encoder has not started");
- return OK;
- }
-
- if (mInputBuffer) {
- mInputBuffer->release();
- mInputBuffer = NULL;
- }
-
- delete mBufferGroup;
- mBufferGroup = NULL;
-
-
- CHECK_EQ(VO_ERR_NONE, mApiHandle->Uninit(mEncoderHandle));
- mEncoderHandle = NULL;
-
- delete mApiHandle;
- mApiHandle = NULL;
-
- delete mMemOperator;
- mMemOperator;
-
- mStarted = false;
-
- mSource->stop();
- return OK;
-}
-
-sp<MetaData> AMRWBEncoder::getFormat() {
- sp<MetaData> srcFormat = mSource->getFormat();
-
- mMeta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_AMR_WB);
-
- int64_t durationUs;
- if (srcFormat->findInt64(kKeyDuration, &durationUs)) {
- mMeta->setInt64(kKeyDuration, durationUs);
- }
-
- mMeta->setCString(kKeyDecoderComponent, "AMRWBEncoder");
-
- return mMeta;
-}
-
-status_t AMRWBEncoder::read(
- MediaBuffer **out, const ReadOptions *options) {
- status_t err;
-
- *out = NULL;
-
- int64_t seekTimeUs;
- ReadOptions::SeekMode mode;
- CHECK(options == NULL || !options->getSeekTo(&seekTimeUs, &mode));
- bool readFromSource = false;
- int64_t wallClockTimeUs = -1;
-
- while (mNumInputSamples < kNumSamplesPerFrame) {
- if (mInputBuffer == NULL) {
- err = mSource->read(&mInputBuffer, options);
-
- if (err != OK) {
- if (mNumInputSamples == 0) {
- return ERROR_END_OF_STREAM;
- }
- memset(&mInputFrame[mNumInputSamples],
- 0,
- sizeof(int16_t)
- * (kNumSamplesPerFrame - mNumInputSamples));
- mNumInputSamples = 0;
- break;
- }
-
- size_t align = mInputBuffer->range_length() % sizeof(int16_t);
- CHECK_EQ(align, 0);
-
- int64_t timeUs;
- if (mInputBuffer->meta_data()->findInt64(kKeyDriftTime, &timeUs)) {
- wallClockTimeUs = timeUs;
- }
- if (mInputBuffer->meta_data()->findInt64(kKeyAnchorTime, &timeUs)) {
- mAnchorTimeUs = timeUs;
- }
- readFromSource = true;
- } else {
- readFromSource = false;
- }
-
- size_t copy =
- (kNumSamplesPerFrame - mNumInputSamples) * sizeof(int16_t);
-
- if (copy > mInputBuffer->range_length()) {
- copy = mInputBuffer->range_length();
- }
-
- memcpy(&mInputFrame[mNumInputSamples],
- (const uint8_t *)mInputBuffer->data()
- + mInputBuffer->range_offset(),
- copy);
-
- mInputBuffer->set_range(
- mInputBuffer->range_offset() + copy,
- mInputBuffer->range_length() - copy);
-
- if (mInputBuffer->range_length() == 0) {
- mInputBuffer->release();
- mInputBuffer = NULL;
- }
-
- mNumInputSamples += copy / sizeof(int16_t);
- if (mNumInputSamples >= kNumSamplesPerFrame) {
- mNumInputSamples %= kNumSamplesPerFrame;
- break; // Get a whole input frame 640 bytes
- }
- }
-
- VO_CODECBUFFER inputData;
- memset(&inputData, 0, sizeof(inputData));
- inputData.Buffer = (unsigned char*) mInputFrame;
- inputData.Length = kInputBufferSize;
- CHECK(VO_ERR_NONE == mApiHandle->SetInputData(mEncoderHandle,&inputData));
-
- MediaBuffer *buffer;
- CHECK_EQ(mBufferGroup->acquire_buffer(&buffer), OK);
- uint8_t *outPtr = (uint8_t *)buffer->data();
-
- VO_CODECBUFFER outputData;
- memset(&outputData, 0, sizeof(outputData));
- VO_AUDIO_OUTPUTINFO outputInfo;
- memset(&outputInfo, 0, sizeof(outputInfo));
-
- VO_U32 ret = VO_ERR_NONE;
- outputData.Buffer = outPtr;
- outputData.Length = buffer->size();
- ret = mApiHandle->GetOutputData(mEncoderHandle, &outputData, &outputInfo);
- CHECK(ret == VO_ERR_NONE || ret == VO_ERR_INPUT_BUFFER_SMALL);
-
- buffer->set_range(0, outputData.Length);
- ++mNumFramesOutput;
-
- int64_t mediaTimeUs = mNumFramesOutput * 20000LL;
- buffer->meta_data()->setInt64(kKeyTime, mAnchorTimeUs + mediaTimeUs);
- if (readFromSource && wallClockTimeUs != -1) {
- buffer->meta_data()->setInt64(kKeyDriftTime, mediaTimeUs - wallClockTimeUs);
- }
-
- *out = buffer;
- return OK;
-}
-
-} // namespace android
diff --git a/media/libstagefright/codecs/amrwbenc/Android.mk b/media/libstagefright/codecs/amrwbenc/Android.mk
index ae43870..d3c3041 100644
--- a/media/libstagefright/codecs/amrwbenc/Android.mk
+++ b/media/libstagefright/codecs/amrwbenc/Android.mk
@@ -5,7 +5,6 @@
LOCAL_SRC_FILES := \
- AMRWBEncoder.cpp \
src/autocorr.c \
src/az_isp.c \
src/bits.c \
@@ -117,4 +116,26 @@
include $(BUILD_STATIC_LIBRARY)
+################################################################################
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES := \
+ SoftAMRWBEncoder.cpp
+
+LOCAL_C_INCLUDES := \
+ frameworks/base/media/libstagefright/include \
+ frameworks/base/media/libstagefright/codecs/common/include \
+ frameworks/native/include/media/openmax
+
+LOCAL_STATIC_LIBRARIES := \
+ libstagefright_amrwbenc
+
+LOCAL_SHARED_LIBRARIES := \
+ libstagefright_omx libstagefright_foundation libutils \
+ libstagefright_enc_common
+
+LOCAL_MODULE := libstagefright_soft_amrwbenc
+LOCAL_MODULE_TAGS := optional
+
+include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libstagefright/codecs/amrwbenc/SoftAMRWBEncoder.cpp b/media/libstagefright/codecs/amrwbenc/SoftAMRWBEncoder.cpp
new file mode 100644
index 0000000..9ccb49c
--- /dev/null
+++ b/media/libstagefright/codecs/amrwbenc/SoftAMRWBEncoder.cpp
@@ -0,0 +1,459 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "SoftAMRWBEncoder"
+#include <utils/Log.h>
+
+#include "SoftAMRWBEncoder.h"
+
+#include "cmnMemory.h"
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/hexdump.h>
+
+namespace android {
+
+static const int32_t kSampleRate = 16000;
+
+template<class T>
+static void InitOMXParams(T *params) {
+ params->nSize = sizeof(T);
+ params->nVersion.s.nVersionMajor = 1;
+ params->nVersion.s.nVersionMinor = 0;
+ params->nVersion.s.nRevision = 0;
+ params->nVersion.s.nStep = 0;
+}
+
+SoftAMRWBEncoder::SoftAMRWBEncoder(
+ const char *name,
+ const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData,
+ OMX_COMPONENTTYPE **component)
+ : SimpleSoftOMXComponent(name, callbacks, appData, component),
+ mEncoderHandle(NULL),
+ mApiHandle(NULL),
+ mMemOperator(NULL),
+ mBitRate(0),
+ mMode(VOAMRWB_MD66),
+ mInputSize(0),
+ mInputTimeUs(-1ll),
+ mSawInputEOS(false),
+ mSignalledError(false) {
+ initPorts();
+ CHECK_EQ(initEncoder(), (status_t)OK);
+}
+
+SoftAMRWBEncoder::~SoftAMRWBEncoder() {
+ if (mEncoderHandle != NULL) {
+ CHECK_EQ(VO_ERR_NONE, mApiHandle->Uninit(mEncoderHandle));
+ mEncoderHandle = NULL;
+ }
+
+ delete mApiHandle;
+ mApiHandle = NULL;
+
+ delete mMemOperator;
+ mMemOperator = NULL;
+}
+
+void SoftAMRWBEncoder::initPorts() {
+ OMX_PARAM_PORTDEFINITIONTYPE def;
+ InitOMXParams(&def);
+
+ def.nPortIndex = 0;
+ def.eDir = OMX_DirInput;
+ def.nBufferCountMin = kNumBuffers;
+ def.nBufferCountActual = def.nBufferCountMin;
+ def.nBufferSize = kNumSamplesPerFrame * sizeof(int16_t);
+ def.bEnabled = OMX_TRUE;
+ def.bPopulated = OMX_FALSE;
+ def.eDomain = OMX_PortDomainAudio;
+ def.bBuffersContiguous = OMX_FALSE;
+ def.nBufferAlignment = 1;
+
+ def.format.audio.cMIMEType = const_cast<char *>("audio/raw");
+ def.format.audio.pNativeRender = NULL;
+ def.format.audio.bFlagErrorConcealment = OMX_FALSE;
+ def.format.audio.eEncoding = OMX_AUDIO_CodingPCM;
+
+ addPort(def);
+
+ def.nPortIndex = 1;
+ def.eDir = OMX_DirOutput;
+ def.nBufferCountMin = kNumBuffers;
+ def.nBufferCountActual = def.nBufferCountMin;
+ def.nBufferSize = 8192;
+ def.bEnabled = OMX_TRUE;
+ def.bPopulated = OMX_FALSE;
+ def.eDomain = OMX_PortDomainAudio;
+ def.bBuffersContiguous = OMX_FALSE;
+ def.nBufferAlignment = 2;
+
+ def.format.audio.cMIMEType = const_cast<char *>("audio/amr-wb");
+ def.format.audio.pNativeRender = NULL;
+ def.format.audio.bFlagErrorConcealment = OMX_FALSE;
+ def.format.audio.eEncoding = OMX_AUDIO_CodingAMR;
+
+ addPort(def);
+}
+
+status_t SoftAMRWBEncoder::initEncoder() {
+ mApiHandle = new VO_AUDIO_CODECAPI;
+
+ if (VO_ERR_NONE != voGetAMRWBEncAPI(mApiHandle)) {
+ ALOGE("Failed to get api handle");
+ return UNKNOWN_ERROR;
+ }
+
+ mMemOperator = new VO_MEM_OPERATOR;
+ mMemOperator->Alloc = cmnMemAlloc;
+ mMemOperator->Copy = cmnMemCopy;
+ mMemOperator->Free = cmnMemFree;
+ mMemOperator->Set = cmnMemSet;
+ mMemOperator->Check = cmnMemCheck;
+
+ VO_CODEC_INIT_USERDATA userData;
+ memset(&userData, 0, sizeof(userData));
+ userData.memflag = VO_IMF_USERMEMOPERATOR;
+ userData.memData = (VO_PTR) mMemOperator;
+
+ if (VO_ERR_NONE != mApiHandle->Init(
+ &mEncoderHandle, VO_AUDIO_CodingAMRWB, &userData)) {
+ ALOGE("Failed to init AMRWB encoder");
+ return UNKNOWN_ERROR;
+ }
+
+ VOAMRWBFRAMETYPE type = VOAMRWB_RFC3267;
+ if (VO_ERR_NONE != mApiHandle->SetParam(
+ mEncoderHandle, VO_PID_AMRWB_FRAMETYPE, &type)) {
+ ALOGE("Failed to set AMRWB encoder frame type to %d", type);
+ return UNKNOWN_ERROR;
+ }
+
+ return OK;
+}
+
+OMX_ERRORTYPE SoftAMRWBEncoder::internalGetParameter(
+ OMX_INDEXTYPE index, OMX_PTR params) {
+ switch (index) {
+ case OMX_IndexParamAudioPortFormat:
+ {
+ OMX_AUDIO_PARAM_PORTFORMATTYPE *formatParams =
+ (OMX_AUDIO_PARAM_PORTFORMATTYPE *)params;
+
+ if (formatParams->nPortIndex > 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (formatParams->nIndex > 0) {
+ return OMX_ErrorNoMore;
+ }
+
+ formatParams->eEncoding =
+ (formatParams->nPortIndex == 0)
+ ? OMX_AUDIO_CodingPCM : OMX_AUDIO_CodingAMR;
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioAmr:
+ {
+ OMX_AUDIO_PARAM_AMRTYPE *amrParams =
+ (OMX_AUDIO_PARAM_AMRTYPE *)params;
+
+ if (amrParams->nPortIndex != 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ amrParams->nChannels = 1;
+ amrParams->nBitRate = mBitRate;
+
+ amrParams->eAMRBandMode =
+ (OMX_AUDIO_AMRBANDMODETYPE)(mMode + OMX_AUDIO_AMRBandModeWB0);
+
+ amrParams->eAMRDTXMode = OMX_AUDIO_AMRDTXModeOff;
+ amrParams->eAMRFrameFormat = OMX_AUDIO_AMRFrameFormatFSF;
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioPcm:
+ {
+ OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
+ (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
+
+ if (pcmParams->nPortIndex != 0) {
+ return OMX_ErrorUndefined;
+ }
+
+ pcmParams->eNumData = OMX_NumericalDataSigned;
+ pcmParams->eEndian = OMX_EndianBig;
+ pcmParams->bInterleaved = OMX_TRUE;
+ pcmParams->nBitPerSample = 16;
+ pcmParams->ePCMMode = OMX_AUDIO_PCMModeLinear;
+ pcmParams->eChannelMapping[0] = OMX_AUDIO_ChannelCF;
+
+ pcmParams->nChannels = 1;
+ pcmParams->nSamplingRate = kSampleRate;
+
+ return OMX_ErrorNone;
+ }
+
+ default:
+ return SimpleSoftOMXComponent::internalGetParameter(index, params);
+ }
+}
+
+OMX_ERRORTYPE SoftAMRWBEncoder::internalSetParameter(
+ OMX_INDEXTYPE index, const OMX_PTR params) {
+ switch (index) {
+ case OMX_IndexParamStandardComponentRole:
+ {
+ const OMX_PARAM_COMPONENTROLETYPE *roleParams =
+ (const OMX_PARAM_COMPONENTROLETYPE *)params;
+
+ if (strncmp((const char *)roleParams->cRole,
+ "audio_encoder.amrwb",
+ OMX_MAX_STRINGNAME_SIZE - 1)) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioPortFormat:
+ {
+ const OMX_AUDIO_PARAM_PORTFORMATTYPE *formatParams =
+ (const OMX_AUDIO_PARAM_PORTFORMATTYPE *)params;
+
+ if (formatParams->nPortIndex > 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (formatParams->nIndex > 0) {
+ return OMX_ErrorNoMore;
+ }
+
+ if ((formatParams->nPortIndex == 0
+ && formatParams->eEncoding != OMX_AUDIO_CodingPCM)
+ || (formatParams->nPortIndex == 1
+ && formatParams->eEncoding != OMX_AUDIO_CodingAMR)) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioAmr:
+ {
+ OMX_AUDIO_PARAM_AMRTYPE *amrParams =
+ (OMX_AUDIO_PARAM_AMRTYPE *)params;
+
+ if (amrParams->nPortIndex != 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (amrParams->nChannels != 1
+ || amrParams->eAMRDTXMode != OMX_AUDIO_AMRDTXModeOff
+ || amrParams->eAMRFrameFormat
+ != OMX_AUDIO_AMRFrameFormatFSF
+ || amrParams->eAMRBandMode < OMX_AUDIO_AMRBandModeWB0
+ || amrParams->eAMRBandMode > OMX_AUDIO_AMRBandModeWB8) {
+ return OMX_ErrorUndefined;
+ }
+
+ mBitRate = amrParams->nBitRate;
+
+ mMode = (VOAMRWBMODE)(
+ amrParams->eAMRBandMode - OMX_AUDIO_AMRBandModeWB0);
+
+ amrParams->eAMRDTXMode = OMX_AUDIO_AMRDTXModeOff;
+ amrParams->eAMRFrameFormat = OMX_AUDIO_AMRFrameFormatFSF;
+
+ if (VO_ERR_NONE !=
+ mApiHandle->SetParam(
+ mEncoderHandle, VO_PID_AMRWB_MODE, &mMode)) {
+ ALOGE("Failed to set AMRWB encoder mode to %d", mMode);
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioPcm:
+ {
+ OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
+ (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
+
+ if (pcmParams->nPortIndex != 0) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (pcmParams->nChannels != 1
+ || pcmParams->nSamplingRate != (OMX_U32)kSampleRate) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+
+ default:
+ return SimpleSoftOMXComponent::internalSetParameter(index, params);
+ }
+}
+
+void SoftAMRWBEncoder::onQueueFilled(OMX_U32 portIndex) {
+ if (mSignalledError) {
+ return;
+ }
+
+ List<BufferInfo *> &inQueue = getPortQueue(0);
+ List<BufferInfo *> &outQueue = getPortQueue(1);
+
+ size_t numBytesPerInputFrame = kNumSamplesPerFrame * sizeof(int16_t);
+
+ for (;;) {
+ // We do the following until we run out of buffers.
+
+ while (mInputSize < numBytesPerInputFrame) {
+ // As long as there's still input data to be read we
+ // will drain "kNumSamplesPerFrame" samples
+ // into the "mInputFrame" buffer and then encode those
+ // as a unit into an output buffer.
+
+ if (mSawInputEOS || inQueue.empty()) {
+ return;
+ }
+
+ BufferInfo *inInfo = *inQueue.begin();
+ OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+
+ const void *inData = inHeader->pBuffer + inHeader->nOffset;
+
+ size_t copy = numBytesPerInputFrame - mInputSize;
+ if (copy > inHeader->nFilledLen) {
+ copy = inHeader->nFilledLen;
+ }
+
+ if (mInputSize == 0) {
+ mInputTimeUs = inHeader->nTimeStamp;
+ }
+
+ memcpy((uint8_t *)mInputFrame + mInputSize, inData, copy);
+ mInputSize += copy;
+
+ inHeader->nOffset += copy;
+ inHeader->nFilledLen -= copy;
+
+ // "Time" on the input buffer has in effect advanced by the
+ // number of audio frames we just advanced nOffset by.
+ inHeader->nTimeStamp +=
+ (copy * 1000000ll / kSampleRate) / sizeof(int16_t);
+
+ if (inHeader->nFilledLen == 0) {
+ if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
+ ALOGV("saw input EOS");
+ mSawInputEOS = true;
+
+ // Pad any remaining data with zeroes.
+ memset((uint8_t *)mInputFrame + mInputSize,
+ 0,
+ numBytesPerInputFrame - mInputSize);
+
+ mInputSize = numBytesPerInputFrame;
+ }
+
+ inQueue.erase(inQueue.begin());
+ inInfo->mOwnedByUs = false;
+ notifyEmptyBufferDone(inHeader);
+
+ inData = NULL;
+ inHeader = NULL;
+ inInfo = NULL;
+ }
+ }
+
+ // At this point we have all the input data necessary to encode
+ // a single frame, all we need is an output buffer to store the result
+ // in.
+
+ if (outQueue.empty()) {
+ return;
+ }
+
+ BufferInfo *outInfo = *outQueue.begin();
+ OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+
+ uint8_t *outPtr = outHeader->pBuffer + outHeader->nOffset;
+ size_t outAvailable = outHeader->nAllocLen - outHeader->nOffset;
+
+ VO_CODECBUFFER inputData;
+ memset(&inputData, 0, sizeof(inputData));
+ inputData.Buffer = (unsigned char *) mInputFrame;
+ inputData.Length = mInputSize;
+
+ CHECK_EQ(VO_ERR_NONE,
+ mApiHandle->SetInputData(mEncoderHandle, &inputData));
+
+ VO_CODECBUFFER outputData;
+ memset(&outputData, 0, sizeof(outputData));
+ VO_AUDIO_OUTPUTINFO outputInfo;
+ memset(&outputInfo, 0, sizeof(outputInfo));
+
+ outputData.Buffer = outPtr;
+ outputData.Length = outAvailable;
+ VO_U32 ret = mApiHandle->GetOutputData(
+ mEncoderHandle, &outputData, &outputInfo);
+ CHECK(ret == VO_ERR_NONE || ret == VO_ERR_INPUT_BUFFER_SMALL);
+
+ outHeader->nFilledLen = outputData.Length;
+ outHeader->nFlags = OMX_BUFFERFLAG_ENDOFFRAME;
+
+ if (mSawInputEOS) {
+ // We also tag this output buffer with EOS if it corresponds
+ // to the final input buffer.
+ outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+ }
+
+ outHeader->nTimeStamp = mInputTimeUs;
+
+#if 0
+ ALOGI("sending %ld bytes of data (time = %lld us, flags = 0x%08lx)",
+ outHeader->nFilledLen, mInputTimeUs, outHeader->nFlags);
+
+ hexdump(outHeader->pBuffer + outHeader->nOffset, outHeader->nFilledLen);
+#endif
+
+ outQueue.erase(outQueue.begin());
+ outInfo->mOwnedByUs = false;
+ notifyFillBufferDone(outHeader);
+
+ outHeader = NULL;
+ outInfo = NULL;
+
+ mInputSize = 0;
+ }
+}
+
+} // namespace android
+
+android::SoftOMXComponent *createSoftOMXComponent(
+ const char *name, const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData, OMX_COMPONENTTYPE **component) {
+ return new android::SoftAMRWBEncoder(name, callbacks, appData, component);
+}
diff --git a/media/libstagefright/codecs/amrwbenc/SoftAMRWBEncoder.h b/media/libstagefright/codecs/amrwbenc/SoftAMRWBEncoder.h
new file mode 100644
index 0000000..d0c1dab
--- /dev/null
+++ b/media/libstagefright/codecs/amrwbenc/SoftAMRWBEncoder.h
@@ -0,0 +1,76 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef SOFT_AMRWB_ENCODER_H_
+
+#define SOFT_AMRWB_ENCODER_H_
+
+#include "SimpleSoftOMXComponent.h"
+
+#include "voAMRWB.h"
+
+struct VO_AUDIO_CODECAPI;
+struct VO_MEM_OPERATOR;
+
+namespace android {
+
+struct SoftAMRWBEncoder : public SimpleSoftOMXComponent {
+ SoftAMRWBEncoder(
+ const char *name,
+ const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData,
+ OMX_COMPONENTTYPE **component);
+
+protected:
+ virtual ~SoftAMRWBEncoder();
+
+ virtual OMX_ERRORTYPE internalGetParameter(
+ OMX_INDEXTYPE index, OMX_PTR params);
+
+ virtual OMX_ERRORTYPE internalSetParameter(
+ OMX_INDEXTYPE index, const OMX_PTR params);
+
+ virtual void onQueueFilled(OMX_U32 portIndex);
+
+private:
+ enum {
+ kNumBuffers = 4,
+ kNumSamplesPerFrame = 320,
+ };
+
+ void *mEncoderHandle;
+ VO_AUDIO_CODECAPI *mApiHandle;
+ VO_MEM_OPERATOR *mMemOperator;
+
+ OMX_U32 mBitRate;
+ VOAMRWBMODE mMode;
+
+ size_t mInputSize;
+ int16_t mInputFrame[kNumSamplesPerFrame];
+ int64_t mInputTimeUs;
+
+ bool mSawInputEOS;
+ bool mSignalledError;
+
+ void initPorts();
+ status_t initEncoder();
+
+ DISALLOW_EVIL_CONSTRUCTORS(SoftAMRWBEncoder);
+};
+
+} // namespace android
+
+#endif // SOFT_AMRWB_ENCODER_H_
diff --git a/media/libstagefright/codecs/avc/enc/AVCEncoder.cpp b/media/libstagefright/codecs/avc/enc/AVCEncoder.cpp
index e202a2b..7533f07 100644
--- a/media/libstagefright/codecs/avc/enc/AVCEncoder.cpp
+++ b/media/libstagefright/codecs/avc/enc/AVCEncoder.cpp
@@ -24,8 +24,8 @@
#include "avcenc_int.h"
#include "OMX_Video.h"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
@@ -417,7 +417,7 @@
*out = NULL;
MediaBuffer *outputBuffer;
- CHECK_EQ(OK, mGroup->acquire_buffer(&outputBuffer));
+ CHECK_EQ((status_t)OK, mGroup->acquire_buffer(&outputBuffer));
uint8_t *outPtr = (uint8_t *) outputBuffer->data();
uint32_t dataLength = outputBuffer->size();
@@ -557,9 +557,9 @@
encoderStatus = PVAVCEncodeNAL(mHandle, outPtr, &dataLength, &type);
if (encoderStatus == AVCENC_SUCCESS) {
outputBuffer->meta_data()->setInt32(kKeyIsSyncFrame, mIsIDRFrame);
- CHECK_EQ(NULL, PVAVCEncGetOverrunBuffer(mHandle));
+ CHECK(NULL == PVAVCEncGetOverrunBuffer(mHandle));
} else if (encoderStatus == AVCENC_PICTURE_READY) {
- CHECK_EQ(NULL, PVAVCEncGetOverrunBuffer(mHandle));
+ CHECK(NULL == PVAVCEncGetOverrunBuffer(mHandle));
if (mIsIDRFrame) {
outputBuffer->meta_data()->setInt32(kKeyIsSyncFrame, mIsIDRFrame);
mIsIDRFrame = 0;
diff --git a/media/libstagefright/codecs/avc/enc/Android.mk b/media/libstagefright/codecs/avc/enc/Android.mk
index eb2e320..54cc9b1 100644
--- a/media/libstagefright/codecs/avc/enc/Android.mk
+++ b/media/libstagefright/codecs/avc/enc/Android.mk
@@ -25,8 +25,8 @@
LOCAL_C_INCLUDES := \
$(LOCAL_PATH)/src \
$(LOCAL_PATH)/../common/include \
- $(TOP)/frameworks/base/include/media/stagefright/openmax \
- $(TOP)/frameworks/base/media/libstagefright/include
+ $(TOP)/frameworks/base/media/libstagefright/include \
+ $(TOP)/frameworks/native/include/media/openmax
LOCAL_CFLAGS := \
-D__arm__ \
diff --git a/media/libstagefright/codecs/g711/dec/Android.mk b/media/libstagefright/codecs/g711/dec/Android.mk
index 6692533..42706a5 100644
--- a/media/libstagefright/codecs/g711/dec/Android.mk
+++ b/media/libstagefright/codecs/g711/dec/Android.mk
@@ -6,7 +6,7 @@
LOCAL_C_INCLUDES := \
frameworks/base/media/libstagefright/include \
- frameworks/base/include/media/stagefright/openmax \
+ frameworks/native/include/media/openmax
LOCAL_SHARED_LIBRARIES := \
libstagefright libstagefright_omx libstagefright_foundation libutils
diff --git a/media/libstagefright/codecs/g711/dec/SoftG711.cpp b/media/libstagefright/codecs/g711/dec/SoftG711.cpp
index 32ef003..bcdd3c7 100644
--- a/media/libstagefright/codecs/g711/dec/SoftG711.cpp
+++ b/media/libstagefright/codecs/g711/dec/SoftG711.cpp
@@ -140,7 +140,7 @@
OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
(OMX_AUDIO_PARAM_PCMMODETYPE *)params;
- if (pcmParams->nPortIndex != 0) {
+ if (pcmParams->nPortIndex != 0 && pcmParams->nPortIndex != 1) {
return OMX_ErrorUndefined;
}
@@ -148,7 +148,9 @@
return OMX_ErrorUndefined;
}
- mNumChannels = pcmParams->nChannels;
+ if(pcmParams->nPortIndex == 0) {
+ mNumChannels = pcmParams->nChannels;
+ }
return OMX_ErrorNone;
}
diff --git a/media/libstagefright/codecs/m4v_h263/dec/Android.mk b/media/libstagefright/codecs/m4v_h263/dec/Android.mk
index 2ffa5f2..8c245d1 100644
--- a/media/libstagefright/codecs/m4v_h263/dec/Android.mk
+++ b/media/libstagefright/codecs/m4v_h263/dec/Android.mk
@@ -42,7 +42,7 @@
$(LOCAL_PATH)/src \
$(LOCAL_PATH)/include \
$(TOP)/frameworks/base/media/libstagefright/include \
- $(TOP)/frameworks/base/include/media/stagefright/openmax
+ $(TOP)/frameworks/native/include/media/openmax
LOCAL_CFLAGS := -DOSCL_EXPORT_REF= -DOSCL_IMPORT_REF=
@@ -59,7 +59,7 @@
$(LOCAL_PATH)/src \
$(LOCAL_PATH)/include \
frameworks/base/media/libstagefright/include \
- frameworks/base/include/media/stagefright/openmax \
+ frameworks/native/include/media/openmax
LOCAL_CFLAGS := -DOSCL_EXPORT_REF= -DOSCL_IMPORT_REF=
diff --git a/media/libstagefright/codecs/m4v_h263/enc/Android.mk b/media/libstagefright/codecs/m4v_h263/enc/Android.mk
index 43318e9..2b7c938 100644
--- a/media/libstagefright/codecs/m4v_h263/enc/Android.mk
+++ b/media/libstagefright/codecs/m4v_h263/enc/Android.mk
@@ -31,7 +31,7 @@
LOCAL_C_INCLUDES := \
$(LOCAL_PATH)/src \
$(LOCAL_PATH)/include \
- $(TOP)/frameworks/base/include/media/stagefright/openmax \
- $(TOP)/frameworks/base/media/libstagefright/include
+ $(TOP)/frameworks/base/media/libstagefright/include \
+ $(TOP)/frameworks/native/include/media/openmax
include $(BUILD_STATIC_LIBRARY)
diff --git a/media/libstagefright/codecs/m4v_h263/enc/M4vH263Encoder.cpp b/media/libstagefright/codecs/m4v_h263/enc/M4vH263Encoder.cpp
index d538603..20b0f8d 100644
--- a/media/libstagefright/codecs/m4v_h263/enc/M4vH263Encoder.cpp
+++ b/media/libstagefright/codecs/m4v_h263/enc/M4vH263Encoder.cpp
@@ -23,8 +23,8 @@
#include "mp4enc_api.h"
#include "OMX_Video.h"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
@@ -379,7 +379,7 @@
*out = NULL;
MediaBuffer *outputBuffer;
- CHECK_EQ(OK, mGroup->acquire_buffer(&outputBuffer));
+ CHECK_EQ((status_t)OK, mGroup->acquire_buffer(&outputBuffer));
uint8_t *outPtr = (uint8_t *) outputBuffer->data();
int32_t dataLength = outputBuffer->size();
@@ -467,7 +467,7 @@
mInputBuffer = NULL;
return UNKNOWN_ERROR;
}
- CHECK_EQ(NULL, PVGetOverrunBuffer(mHandle));
+ CHECK(NULL == PVGetOverrunBuffer(mHandle));
if (hintTrack.CodeType == 0) { // I-frame serves as sync frame
outputBuffer->meta_data()->setInt32(kKeyIsSyncFrame, 1);
}
diff --git a/media/libstagefright/codecs/mp3dec/Android.mk b/media/libstagefright/codecs/mp3dec/Android.mk
index a08c9f0..ed51aa5 100644
--- a/media/libstagefright/codecs/mp3dec/Android.mk
+++ b/media/libstagefright/codecs/mp3dec/Android.mk
@@ -65,7 +65,7 @@
LOCAL_C_INCLUDES := \
frameworks/base/media/libstagefright/include \
- frameworks/base/include/media/stagefright/openmax \
+ frameworks/native/include/media/openmax \
$(LOCAL_PATH)/src \
$(LOCAL_PATH)/include
diff --git a/media/libstagefright/codecs/on2/dec/Android.mk b/media/libstagefright/codecs/on2/dec/Android.mk
index 32bbd6b..2997228 100644
--- a/media/libstagefright/codecs/on2/dec/Android.mk
+++ b/media/libstagefright/codecs/on2/dec/Android.mk
@@ -9,7 +9,7 @@
$(TOP)/external/libvpx/vpx_codec \
$(TOP)/external/libvpx/vpx_ports \
frameworks/base/media/libstagefright/include \
- frameworks/base/include/media/stagefright/openmax \
+ frameworks/native/include/media/openmax \
LOCAL_STATIC_LIBRARIES := \
libvpx
diff --git a/media/libstagefright/codecs/on2/h264dec/Android.mk b/media/libstagefright/codecs/on2/h264dec/Android.mk
index 5b3c876..43c20f0 100644
--- a/media/libstagefright/codecs/on2/h264dec/Android.mk
+++ b/media/libstagefright/codecs/on2/h264dec/Android.mk
@@ -35,7 +35,7 @@
LOCAL_C_INCLUDES := $(LOCAL_PATH)/./inc \
frameworks/base/media/libstagefright/include \
- frameworks/base/include/media/stagefright/openmax \
+ frameworks/native/include/media/openmax \
MY_ASM := \
./source/arm_neon_asm_gcc/h264bsdWriteMacroblock.S \
diff --git a/media/libstagefright/codecs/vorbis/dec/Android.mk b/media/libstagefright/codecs/vorbis/dec/Android.mk
index f33f3ac..fca70b7 100644
--- a/media/libstagefright/codecs/vorbis/dec/Android.mk
+++ b/media/libstagefright/codecs/vorbis/dec/Android.mk
@@ -7,7 +7,7 @@
LOCAL_C_INCLUDES := \
external/tremolo \
frameworks/base/media/libstagefright/include \
- frameworks/base/include/media/stagefright/openmax \
+ frameworks/native/include/media/openmax \
LOCAL_SHARED_LIBRARIES := \
libvorbisidec libstagefright libstagefright_omx \
diff --git a/media/libstagefright/colorconversion/Android.mk b/media/libstagefright/colorconversion/Android.mk
index 62ba40f..59a64ba 100644
--- a/media/libstagefright/colorconversion/Android.mk
+++ b/media/libstagefright/colorconversion/Android.mk
@@ -6,7 +6,7 @@
SoftwareRenderer.cpp
LOCAL_C_INCLUDES := \
- $(TOP)/frameworks/base/include/media/stagefright/openmax \
+ $(TOP)/frameworks/native/include/media/openmax \
$(TOP)/hardware/msm7k
LOCAL_MODULE:= libstagefright_color_conversion
diff --git a/media/libstagefright/colorconversion/ColorConverter.cpp b/media/libstagefright/colorconversion/ColorConverter.cpp
index 5cc3f78..597167f 100644
--- a/media/libstagefright/colorconversion/ColorConverter.cpp
+++ b/media/libstagefright/colorconversion/ColorConverter.cpp
@@ -18,8 +18,8 @@
#define LOG_TAG "ColorConverter"
#include <utils/Log.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/ColorConverter.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaErrors.h>
namespace android {
@@ -144,8 +144,8 @@
return ERROR_UNSUPPORTED;
}
- uint32_t *dst_ptr = (uint32_t *)dst.mBits
- + (dst.mCropTop * dst.mWidth + dst.mCropLeft) / 2;
+ uint16_t *dst_ptr = (uint16_t *)dst.mBits
+ + dst.mCropTop * dst.mWidth + dst.mCropLeft;
const uint8_t *src_ptr = (const uint8_t *)src.mBits
+ (src.mCropTop * dst.mWidth + src.mCropLeft) * 2;
@@ -182,11 +182,15 @@
| ((kAdjustedClip[g2] >> 2) << 5)
| (kAdjustedClip[b2] >> 3);
- dst_ptr[x / 2] = (rgb2 << 16) | rgb1;
+ if (x + 1 < src.cropWidth()) {
+ *(uint32_t *)(&dst_ptr[x]) = (rgb2 << 16) | rgb1;
+ } else {
+ dst_ptr[x] = rgb1;
+ }
}
src_ptr += src.mWidth * 2;
- dst_ptr += dst.mWidth / 2;
+ dst_ptr += dst.mWidth;
}
return OK;
@@ -290,15 +294,14 @@
const BitmapParams &src, const BitmapParams &dst) {
uint8_t *kAdjustedClip = initClip();
- if (!((dst.mWidth & 3) == 0
- && (src.mCropLeft & 1) == 0
+ if (!((src.mCropLeft & 1) == 0
&& src.cropWidth() == dst.cropWidth()
&& src.cropHeight() == dst.cropHeight())) {
return ERROR_UNSUPPORTED;
}
- uint32_t *dst_ptr = (uint32_t *)dst.mBits
- + (dst.mCropTop * dst.mWidth + dst.mCropLeft) / 2;
+ uint16_t *dst_ptr = (uint16_t *)dst.mBits
+ + dst.mCropTop * dst.mWidth + dst.mCropLeft;
const uint8_t *src_y =
(const uint8_t *)src.mBits + src.mCropTop * src.mWidth + src.mCropLeft;
@@ -340,7 +343,11 @@
| ((kAdjustedClip[g2] >> 2) << 5)
| (kAdjustedClip[r2] >> 3);
- dst_ptr[x / 2] = (rgb2 << 16) | rgb1;
+ if (x + 1 < src.cropWidth()) {
+ *(uint32_t *)(&dst_ptr[x]) = (rgb2 << 16) | rgb1;
+ } else {
+ dst_ptr[x] = rgb1;
+ }
}
src_y += src.mWidth;
@@ -349,7 +356,7 @@
src_u += src.mWidth;
}
- dst_ptr += dst.mWidth / 2;
+ dst_ptr += dst.mWidth;
}
return OK;
@@ -361,15 +368,14 @@
uint8_t *kAdjustedClip = initClip();
- if (!((dst.mWidth & 3) == 0
- && (src.mCropLeft & 1) == 0
+ if (!((src.mCropLeft & 1) == 0
&& src.cropWidth() == dst.cropWidth()
&& src.cropHeight() == dst.cropHeight())) {
return ERROR_UNSUPPORTED;
}
- uint32_t *dst_ptr = (uint32_t *)dst.mBits
- + (dst.mCropTop * dst.mWidth + dst.mCropLeft) / 2;
+ uint16_t *dst_ptr = (uint16_t *)dst.mBits
+ + dst.mCropTop * dst.mWidth + dst.mCropLeft;
const uint8_t *src_y =
(const uint8_t *)src.mBits + src.mCropTop * src.mWidth + src.mCropLeft;
@@ -411,7 +417,11 @@
| ((kAdjustedClip[g2] >> 2) << 5)
| (kAdjustedClip[r2] >> 3);
- dst_ptr[x / 2] = (rgb2 << 16) | rgb1;
+ if (x + 1 < src.cropWidth()) {
+ *(uint32_t *)(&dst_ptr[x]) = (rgb2 << 16) | rgb1;
+ } else {
+ dst_ptr[x] = rgb1;
+ }
}
src_y += src.mWidth;
@@ -420,7 +430,7 @@
src_u += src.mWidth;
}
- dst_ptr += dst.mWidth / 2;
+ dst_ptr += dst.mWidth;
}
return OK;
@@ -430,15 +440,14 @@
const BitmapParams &src, const BitmapParams &dst) {
uint8_t *kAdjustedClip = initClip();
- if (!((dst.mWidth & 3) == 0
- && (src.mCropLeft & 1) == 0
+ if (!((src.mCropLeft & 1) == 0
&& src.cropWidth() == dst.cropWidth()
&& src.cropHeight() == dst.cropHeight())) {
return ERROR_UNSUPPORTED;
}
- uint32_t *dst_ptr = (uint32_t *)dst.mBits
- + (dst.mCropTop * dst.mWidth + dst.mCropLeft) / 2;
+ uint16_t *dst_ptr = (uint16_t *)dst.mBits
+ + dst.mCropTop * dst.mWidth + dst.mCropLeft;
const uint8_t *src_y = (const uint8_t *)src.mBits;
@@ -478,7 +487,11 @@
| ((kAdjustedClip[g2] >> 2) << 5)
| (kAdjustedClip[b2] >> 3);
- dst_ptr[x / 2] = (rgb2 << 16) | rgb1;
+ if (x + 1 < src.cropWidth()) {
+ *(uint32_t *)(&dst_ptr[x]) = (rgb2 << 16) | rgb1;
+ } else {
+ dst_ptr[x] = rgb1;
+ }
}
src_y += src.mWidth;
@@ -487,7 +500,7 @@
src_u += src.mWidth;
}
- dst_ptr += dst.mWidth / 2;
+ dst_ptr += dst.mWidth;
}
return OK;
diff --git a/media/libstagefright/colorconversion/SoftwareRenderer.cpp b/media/libstagefright/colorconversion/SoftwareRenderer.cpp
index e892f92..059d6b9 100644
--- a/media/libstagefright/colorconversion/SoftwareRenderer.cpp
+++ b/media/libstagefright/colorconversion/SoftwareRenderer.cpp
@@ -19,12 +19,9 @@
#include "../include/SoftwareRenderer.h"
-#include <binder/MemoryHeapBase.h>
-#include <binder/MemoryHeapPmem.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MetaData.h>
-#include <surfaceflinger/Surface.h>
-#include <ui/android_native_buffer.h>
+#include <system/window.h>
#include <ui/GraphicBufferMapper.h>
#include <gui/ISurfaceTexture.h>
diff --git a/media/libstagefright/foundation/AMessage.cpp b/media/libstagefright/foundation/AMessage.cpp
index 0a6776e..8b01ac6 100644
--- a/media/libstagefright/foundation/AMessage.cpp
+++ b/media/libstagefright/foundation/AMessage.cpp
@@ -19,6 +19,7 @@
#include <ctype.h>
#include "AAtomizer.h"
+#include "ABuffer.h"
#include "ADebug.h"
#include "ALooperRoster.h"
#include "AString.h"
@@ -73,6 +74,7 @@
case kTypeObject:
case kTypeMessage:
+ case kTypeBuffer:
{
if (item->u.refValue != NULL) {
item->u.refValue->decStrong(this);
@@ -157,14 +159,23 @@
item->u.stringValue = new AString(s, len < 0 ? strlen(s) : len);
}
-void AMessage::setObject(const char *name, const sp<RefBase> &obj) {
+void AMessage::setObjectInternal(
+ const char *name, const sp<RefBase> &obj, Type type) {
Item *item = allocateItem(name);
- item->mType = kTypeObject;
+ item->mType = type;
if (obj != NULL) { obj->incStrong(this); }
item->u.refValue = obj.get();
}
+void AMessage::setObject(const char *name, const sp<RefBase> &obj) {
+ setObjectInternal(name, obj, kTypeObject);
+}
+
+void AMessage::setBuffer(const char *name, const sp<ABuffer> &buffer) {
+ setObjectInternal(name, sp<RefBase>(buffer), kTypeBuffer);
+}
+
void AMessage::setMessage(const char *name, const sp<AMessage> &obj) {
Item *item = allocateItem(name);
item->mType = kTypeMessage;
@@ -203,6 +214,15 @@
return false;
}
+bool AMessage::findBuffer(const char *name, sp<ABuffer> *buf) const {
+ const Item *item = findItem(name, kTypeBuffer);
+ if (item) {
+ *buf = (ABuffer *)(item->u.refValue);
+ return true;
+ }
+ return false;
+}
+
bool AMessage::findMessage(const char *name, sp<AMessage> *obj) const {
const Item *item = findItem(name, kTypeMessage);
if (item) {
@@ -273,6 +293,7 @@
}
case kTypeObject:
+ case kTypeBuffer:
{
to->u.refValue = from->u.refValue;
to->u.refValue->incStrong(msg.get());
@@ -377,6 +398,10 @@
tmp = StringPrintf(
"RefBase *%s = %p", item.mName, item.u.refValue);
break;
+ case kTypeBuffer:
+ tmp = StringPrintf(
+ "ABuffer *%s = %p", item.mName, item.u.refValue);
+ break;
case kTypeMessage:
tmp = StringPrintf(
"AMessage %s = %s",
@@ -542,4 +567,20 @@
}
}
+size_t AMessage::countEntries() const {
+ return mNumItems;
+}
+
+const char *AMessage::getEntryNameAt(size_t index, Type *type) const {
+ if (index >= mNumItems) {
+ *type = kTypeInt32;
+
+ return NULL;
+ }
+
+ *type = mItems[index].mType;
+
+ return mItems[index].mName;
+}
+
} // namespace android
diff --git a/media/libstagefright/httplive/Android.mk b/media/libstagefright/httplive/Android.mk
index 9225e41..a5990c3 100644
--- a/media/libstagefright/httplive/Android.mk
+++ b/media/libstagefright/httplive/Android.mk
@@ -9,9 +9,9 @@
LOCAL_C_INCLUDES:= \
$(JNI_H_INCLUDE) \
- $(TOP)/frameworks/base/include/media/stagefright/openmax \
- $(TOP)/frameworks/base/media/libstagefright \
- $(TOP)/external/openssl/include
+ $(TOP)/frameworks/base/media/libstagefright \
+ $(TOP)/frameworks/native/include/media/openmax \
+ $(TOP)/external/openssl/include
LOCAL_MODULE:= libstagefright_httplive
diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp
index 0df66f1..0cddd2e 100644
--- a/media/libstagefright/httplive/LiveSession.cpp
+++ b/media/libstagefright/httplive/LiveSession.cpp
@@ -215,7 +215,9 @@
mDisconnectPending = false;
}
-status_t LiveSession::fetchFile(const char *url, sp<ABuffer> *out) {
+status_t LiveSession::fetchFile(
+ const char *url, sp<ABuffer> *out,
+ int64_t range_offset, int64_t range_length) {
*out = NULL;
sp<DataSource> source;
@@ -234,8 +236,18 @@
}
}
- status_t err = mHTTPDataSource->connect(
- url, mExtraHeaders.isEmpty() ? NULL : &mExtraHeaders);
+ KeyedVector<String8, String8> headers = mExtraHeaders;
+ if (range_offset > 0 || range_length >= 0) {
+ headers.add(
+ String8("Range"),
+ String8(
+ StringPrintf(
+ "bytes=%lld-%s",
+ range_offset,
+ range_length < 0
+ ? "" : StringPrintf("%lld", range_offset + range_length - 1).c_str()).c_str()));
+ }
+ status_t err = mHTTPDataSource->connect(url, &headers);
if (err != OK) {
return err;
@@ -270,9 +282,21 @@
buffer = copy;
}
+ size_t maxBytesToRead = bufferRemaining;
+ if (range_length >= 0) {
+ int64_t bytesLeftInRange = range_length - buffer->size();
+ if (bytesLeftInRange < maxBytesToRead) {
+ maxBytesToRead = bytesLeftInRange;
+
+ if (bytesLeftInRange == 0) {
+ break;
+ }
+ }
+ }
+
ssize_t n = source->readAt(
buffer->size(), buffer->data() + buffer->size(),
- bufferRemaining);
+ maxBytesToRead);
if (n < 0) {
return n;
@@ -659,8 +683,15 @@
explicitDiscontinuity = true;
}
+ int64_t range_offset, range_length;
+ if (!itemMeta->findInt64("range-offset", &range_offset)
+ || !itemMeta->findInt64("range-length", &range_length)) {
+ range_offset = 0;
+ range_length = -1;
+ }
+
sp<ABuffer> buffer;
- status_t err = fetchFile(uri.c_str(), &buffer);
+ status_t err = fetchFile(uri.c_str(), &buffer, range_offset, range_length);
if (err != OK) {
ALOGE("failed to fetch .ts segment at url '%s'", uri.c_str());
mDataSource->queueEOS(err);
diff --git a/media/libstagefright/httplive/M3UParser.cpp b/media/libstagefright/httplive/M3UParser.cpp
index 5e30488..7d3cf05 100644
--- a/media/libstagefright/httplive/M3UParser.cpp
+++ b/media/libstagefright/httplive/M3UParser.cpp
@@ -152,6 +152,7 @@
const char *data = (const char *)_data;
size_t offset = 0;
+ uint64_t segmentRangeOffset = 0;
while (offset < size) {
size_t offsetLF = offset;
while (offsetLF < size && data[offsetLF] != '\n') {
@@ -218,6 +219,24 @@
}
mIsVariantPlaylist = true;
err = parseStreamInf(line, &itemMeta);
+ } else if (line.startsWith("#EXT-X-BYTERANGE")) {
+ if (mIsVariantPlaylist) {
+ return ERROR_MALFORMED;
+ }
+
+ uint64_t length, offset;
+ err = parseByteRange(line, segmentRangeOffset, &length, &offset);
+
+ if (err == OK) {
+ if (itemMeta == NULL) {
+ itemMeta = new AMessage;
+ }
+
+ itemMeta->setInt64("range-offset", offset);
+ itemMeta->setInt64("range-length", length);
+
+ segmentRangeOffset = offset + length;
+ }
}
if (err != OK) {
@@ -447,6 +466,52 @@
}
// static
+status_t M3UParser::parseByteRange(
+ const AString &line, uint64_t curOffset,
+ uint64_t *length, uint64_t *offset) {
+ ssize_t colonPos = line.find(":");
+
+ if (colonPos < 0) {
+ return ERROR_MALFORMED;
+ }
+
+ ssize_t atPos = line.find("@", colonPos + 1);
+
+ AString lenStr;
+ if (atPos < 0) {
+ lenStr = AString(line, colonPos + 1, line.size() - colonPos - 1);
+ } else {
+ lenStr = AString(line, colonPos + 1, atPos - colonPos - 1);
+ }
+
+ lenStr.trim();
+
+ const char *s = lenStr.c_str();
+ char *end;
+ *length = strtoull(s, &end, 10);
+
+ if (s == end || *end != '\0') {
+ return ERROR_MALFORMED;
+ }
+
+ if (atPos >= 0) {
+ AString offStr = AString(line, atPos + 1, line.size() - atPos - 1);
+ offStr.trim();
+
+ const char *s = offStr.c_str();
+ *offset = strtoull(s, &end, 10);
+
+ if (s == end || *end != '\0') {
+ return ERROR_MALFORMED;
+ }
+ } else {
+ *offset = curOffset;
+ }
+
+ return OK;
+}
+
+// static
status_t M3UParser::ParseInt32(const char *s, int32_t *x) {
char *end;
long lval = strtol(s, &end, 10);
diff --git a/media/libstagefright/id3/Android.mk b/media/libstagefright/id3/Android.mk
index 23c8e44..ff35d4a 100644
--- a/media/libstagefright/id3/Android.mk
+++ b/media/libstagefright/id3/Android.mk
@@ -16,7 +16,7 @@
testid3.cpp
LOCAL_SHARED_LIBRARIES := \
- libstagefright libutils libbinder
+ libstagefright libutils libbinder libstagefright_foundation
LOCAL_STATIC_LIBRARIES := \
libstagefright_id3
diff --git a/media/libstagefright/id3/ID3.cpp b/media/libstagefright/id3/ID3.cpp
index 6dde9d8..ca14054 100644
--- a/media/libstagefright/id3/ID3.cpp
+++ b/media/libstagefright/id3/ID3.cpp
@@ -20,8 +20,8 @@
#include "../include/ID3.h"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/DataSource.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/Utils.h>
#include <utils/String8.h>
#include <byteswap.h>
@@ -463,40 +463,65 @@
tmp = NULL;
}
-void ID3::Iterator::getString(String8 *id) const {
+// the 2nd argument is used to get the data following the \0 in a comment field
+void ID3::Iterator::getString(String8 *id, String8 *comment) const {
+ getstring(id, false);
+ if (comment != NULL) {
+ getstring(comment, true);
+ }
+}
+
+// comment fields (COM/COMM) contain an initial short descriptor, followed by \0,
+// followed by more data. The data following the \0 can be retrieved by setting
+// "otherdata" to true.
+void ID3::Iterator::getstring(String8 *id, bool otherdata) const {
id->setTo("");
- if (mFrameData == NULL) {
+ const uint8_t *frameData = mFrameData;
+ if (frameData == NULL) {
return;
}
+ uint8_t encoding = *frameData;
+
if (mParent.mVersion == ID3_V1 || mParent.mVersion == ID3_V1_1) {
if (mOffset == 126 || mOffset == 127) {
// Special treatment for the track number and genre.
char tmp[16];
- sprintf(tmp, "%d", (int)*mFrameData);
+ sprintf(tmp, "%d", (int)*frameData);
id->setTo(tmp);
return;
}
- convertISO8859ToString8(mFrameData, mFrameSize, id);
+ convertISO8859ToString8(frameData, mFrameSize, id);
return;
}
size_t n = mFrameSize - getHeaderLength() - 1;
+ if (otherdata) {
+ // skip past the encoding, language, and the 0 separator
+ frameData += 4;
+ int32_t i = n - 4;
+ while(--i >= 0 && *++frameData != 0) ;
+ int skipped = (frameData - mFrameData);
+ if (skipped >= n) {
+ return;
+ }
+ n -= skipped;
+ }
- if (*mFrameData == 0x00) {
+ if (encoding == 0x00) {
// ISO 8859-1
- convertISO8859ToString8(mFrameData + 1, n, id);
- } else if (*mFrameData == 0x03) {
+ convertISO8859ToString8(frameData + 1, n, id);
+ } else if (encoding == 0x03) {
// UTF-8
- id->setTo((const char *)(mFrameData + 1), n);
- } else if (*mFrameData == 0x02) {
+ id->setTo((const char *)(frameData + 1), n);
+ } else if (encoding == 0x02) {
// UTF-16 BE, no byte order mark.
// API wants number of characters, not number of bytes...
int len = n / 2;
- const char16_t *framedata = (const char16_t *) (mFrameData + 1);
+ const char16_t *framedata = (const char16_t *) (frameData + 1);
char16_t *framedatacopy = NULL;
#if BYTE_ORDER == LITTLE_ENDIAN
framedatacopy = new char16_t[len];
@@ -513,7 +538,7 @@
// UCS-2
// API wants number of characters, not number of bytes...
int len = n / 2;
- const char16_t *framedata = (const char16_t *) (mFrameData + 1);
+ const char16_t *framedata = (const char16_t *) (frameData + 1);
char16_t *framedatacopy = NULL;
if (*framedata == 0xfffe) {
// endianness marker doesn't match host endianness, convert
diff --git a/media/libstagefright/id3/testid3.cpp b/media/libstagefright/id3/testid3.cpp
index 0741045..bc4572c 100644
--- a/media/libstagefright/id3/testid3.cpp
+++ b/media/libstagefright/id3/testid3.cpp
@@ -23,7 +23,7 @@
#include <binder/ProcessState.h>
#include <media/stagefright/FileSource.h>
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
#define MAXPATHLEN 256
@@ -70,7 +70,7 @@
void scanFile(const char *path) {
sp<FileSource> file = new FileSource(path);
- CHECK_EQ(file->initCheck(), OK);
+ CHECK_EQ(file->initCheck(), (status_t)OK);
ID3 tag(file);
if (!tag.isValid()) {
diff --git a/media/libstagefright/include/AACDecoder.h b/media/libstagefright/include/AACDecoder.h
deleted file mode 100644
index 886a3b7..0000000
--- a/media/libstagefright/include/AACDecoder.h
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- * Copyright (C) 2009 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef AAC_DECODER_H_
-
-#define AAC_DECODER_H_
-
-#include <media/stagefright/MediaSource.h>
-
-struct tPVMP4AudioDecoderExternal;
-
-namespace android {
-
-struct MediaBufferGroup;
-struct MetaData;
-
-struct AACDecoder : public MediaSource {
- AACDecoder(const sp<MediaSource> &source);
-
- virtual status_t start(MetaData *params);
- virtual status_t stop();
-
- virtual sp<MetaData> getFormat();
-
- virtual status_t read(
- MediaBuffer **buffer, const ReadOptions *options);
-
-protected:
- virtual ~AACDecoder();
-
-private:
- sp<MetaData> mMeta;
- sp<MediaSource> mSource;
- bool mStarted;
-
- MediaBufferGroup *mBufferGroup;
-
- tPVMP4AudioDecoderExternal *mConfig;
- void *mDecoderBuf;
- int64_t mAnchorTimeUs;
- int64_t mNumSamplesOutput;
- status_t mInitCheck;
- int64_t mNumDecodedBuffers;
- int32_t mUpsamplingFactor;
-
- MediaBuffer *mInputBuffer;
-
- status_t initCheck();
- AACDecoder(const AACDecoder &);
- AACDecoder &operator=(const AACDecoder &);
-};
-
-} // namespace android
-
-#endif // AAC_DECODER_H_
diff --git a/media/libstagefright/include/AACExtractor.h b/media/libstagefright/include/AACExtractor.h
index 8e5657b..e98ca82 100644
--- a/media/libstagefright/include/AACExtractor.h
+++ b/media/libstagefright/include/AACExtractor.h
@@ -29,7 +29,7 @@
class AACExtractor : public MediaExtractor {
public:
- AACExtractor(const sp<DataSource> &source);
+ AACExtractor(const sp<DataSource> &source, const sp<AMessage> &meta);
virtual size_t countTracks();
virtual sp<MediaSource> getTrack(size_t index);
diff --git a/media/libstagefright/include/AMRNBDecoder.h b/media/libstagefright/include/AMRNBDecoder.h
deleted file mode 100644
index cf24eda..0000000
--- a/media/libstagefright/include/AMRNBDecoder.h
+++ /dev/null
@@ -1,59 +0,0 @@
-/*
- * Copyright (C) 2009 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef AMR_NB_DECODER_H_
-
-#define AMR_NB_DECODER_H_
-
-#include <media/stagefright/MediaSource.h>
-
-namespace android {
-
-struct MediaBufferGroup;
-
-struct AMRNBDecoder : public MediaSource {
- AMRNBDecoder(const sp<MediaSource> &source);
-
- virtual status_t start(MetaData *params);
- virtual status_t stop();
-
- virtual sp<MetaData> getFormat();
-
- virtual status_t read(
- MediaBuffer **buffer, const ReadOptions *options);
-
-protected:
- virtual ~AMRNBDecoder();
-
-private:
- sp<MediaSource> mSource;
- bool mStarted;
-
- MediaBufferGroup *mBufferGroup;
-
- void *mState;
- int64_t mAnchorTimeUs;
- int64_t mNumSamplesOutput;
-
- MediaBuffer *mInputBuffer;
-
- AMRNBDecoder(const AMRNBDecoder &);
- AMRNBDecoder &operator=(const AMRNBDecoder &);
-};
-
-} // namespace android
-
-#endif // AMR_NB_DECODER_H_
diff --git a/media/libstagefright/include/AMRNBEncoder.h b/media/libstagefright/include/AMRNBEncoder.h
deleted file mode 100644
index 71160e6..0000000
--- a/media/libstagefright/include/AMRNBEncoder.h
+++ /dev/null
@@ -1,66 +0,0 @@
-/*
- * Copyright (C) 2009 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef AMR_NB_ENCODER_H_
-
-#define AMR_NB_ENCODER_H_
-
-#include <media/stagefright/MediaSource.h>
-#include <media/stagefright/MetaData.h>
-
-namespace android {
-
-struct MediaBufferGroup;
-
-struct AMRNBEncoder : public MediaSource {
- AMRNBEncoder(const sp<MediaSource> &source, const sp<MetaData> &meta);
-
- virtual status_t start(MetaData *params);
- virtual status_t stop();
-
- virtual sp<MetaData> getFormat();
-
- virtual status_t read(
- MediaBuffer **buffer, const ReadOptions *options);
-
-protected:
- virtual ~AMRNBEncoder();
-
-private:
- sp<MediaSource> mSource;
- sp<MetaData> mMeta;
- bool mStarted;
-
- MediaBufferGroup *mBufferGroup;
-
- void *mEncState;
- void *mSidState;
- int64_t mAnchorTimeUs;
- int64_t mNumFramesOutput;
-
- MediaBuffer *mInputBuffer;
- int mMode;
-
- int16_t mInputFrame[160];
- int32_t mNumInputSamples;
-
- AMRNBEncoder(const AMRNBEncoder &);
- AMRNBEncoder &operator=(const AMRNBEncoder &);
-};
-
-} // namespace android
-
-#endif // AMR_NB_ENCODER_H_
diff --git a/media/libstagefright/include/AMRWBDecoder.h b/media/libstagefright/include/AMRWBDecoder.h
deleted file mode 100644
index 927c51c..0000000
--- a/media/libstagefright/include/AMRWBDecoder.h
+++ /dev/null
@@ -1,62 +0,0 @@
-/*
- * Copyright (C) 2009 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef AMR_WB_DECODER_H_
-
-#define AMR_WB_DECODER_H_
-
-#include <media/stagefright/MediaSource.h>
-
-namespace android {
-
-struct MediaBufferGroup;
-
-struct AMRWBDecoder : public MediaSource {
- AMRWBDecoder(const sp<MediaSource> &source);
-
- virtual status_t start(MetaData *params);
- virtual status_t stop();
-
- virtual sp<MetaData> getFormat();
-
- virtual status_t read(
- MediaBuffer **buffer, const ReadOptions *options);
-
-protected:
- virtual ~AMRWBDecoder();
-
-private:
- sp<MediaSource> mSource;
- bool mStarted;
-
- MediaBufferGroup *mBufferGroup;
-
- void *mState;
- void *mDecoderBuf;
- int16_t *mDecoderCookie;
- int64_t mAnchorTimeUs;
- int64_t mNumSamplesOutput;
- int16_t mInputSampleBuffer[477];
-
- MediaBuffer *mInputBuffer;
-
- AMRWBDecoder(const AMRWBDecoder &);
- AMRWBDecoder &operator=(const AMRWBDecoder &);
-};
-
-} // namespace android
-
-#endif // AMR_WB_DECODER_H_
diff --git a/media/libstagefright/include/AMRWBEncoder.h b/media/libstagefright/include/AMRWBEncoder.h
deleted file mode 100644
index f2d155f..0000000
--- a/media/libstagefright/include/AMRWBEncoder.h
+++ /dev/null
@@ -1,72 +0,0 @@
-/*
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef AMR_WB_ENCODER_H
-#define AMR_WB_ENCODER_H
-
-#include <media/stagefright/MediaSource.h>
-#include <media/stagefright/MetaData.h>
-
-struct VO_AUDIO_CODECAPI;
-struct VO_MEM_OPERATOR;
-
-namespace android {
-
-struct MediaBufferGroup;
-
-class AMRWBEncoder: public MediaSource {
- public:
- AMRWBEncoder(const sp<MediaSource> &source, const sp<MetaData> &meta);
-
- virtual status_t start(MetaData *params);
- virtual status_t stop();
- virtual sp<MetaData> getFormat();
- virtual status_t read(
- MediaBuffer **buffer, const ReadOptions *options);
-
-
- protected:
- virtual ~AMRWBEncoder();
-
- private:
- sp<MediaSource> mSource;
- sp<MetaData> mMeta;
- bool mStarted;
- MediaBufferGroup *mBufferGroup;
- MediaBuffer *mInputBuffer;
- status_t mInitCheck;
- int32_t mBitRate;
- void *mEncoderHandle;
- VO_AUDIO_CODECAPI *mApiHandle;
- VO_MEM_OPERATOR *mMemOperator;
-
- int64_t mAnchorTimeUs;
- int64_t mNumFramesOutput;
-
- int16_t mInputFrame[320];
- int32_t mNumInputSamples;
-
- status_t initCheck();
-
- AMRWBEncoder& operator=(const AMRWBEncoder &rhs);
- AMRWBEncoder(const AMRWBEncoder& copy);
-
-};
-
-}
-
-#endif //#ifndef AMR_WB_ENCODER_H
-
diff --git a/media/libstagefright/include/AVCDecoder.h b/media/libstagefright/include/AVCDecoder.h
deleted file mode 100644
index eb3b142..0000000
--- a/media/libstagefright/include/AVCDecoder.h
+++ /dev/null
@@ -1,93 +0,0 @@
-/*
- * Copyright (C) 2009 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef AVC_DECODER_H_
-
-#define AVC_DECODER_H_
-
-#include <media/stagefright/MediaBuffer.h>
-#include <media/stagefright/MediaSource.h>
-#include <utils/Vector.h>
-
-struct tagAVCHandle;
-
-namespace android {
-
-struct AVCDecoder : public MediaSource,
- public MediaBufferObserver {
- AVCDecoder(const sp<MediaSource> &source);
-
- virtual status_t start(MetaData *params);
- virtual status_t stop();
-
- virtual sp<MetaData> getFormat();
-
- virtual status_t read(
- MediaBuffer **buffer, const ReadOptions *options);
-
- virtual void signalBufferReturned(MediaBuffer *buffer);
-
-protected:
- virtual ~AVCDecoder();
-
-private:
- sp<MediaSource> mSource;
- bool mStarted;
-
- sp<MetaData> mFormat;
-
- Vector<MediaBuffer *> mCodecSpecificData;
-
- tagAVCHandle *mHandle;
- Vector<MediaBuffer *> mFrames;
- MediaBuffer *mInputBuffer;
-
- int64_t mAnchorTimeUs;
- int64_t mNumSamplesOutput;
- int64_t mPendingSeekTimeUs;
- MediaSource::ReadOptions::SeekMode mPendingSeekMode;
-
- int64_t mTargetTimeUs;
-
- bool mSPSSeen;
- bool mPPSSeen;
-
- void addCodecSpecificData(const uint8_t *data, size_t size);
-
- static int32_t ActivateSPSWrapper(
- void *userData, unsigned int sizeInMbs, unsigned int numBuffers);
-
- static int32_t BindFrameWrapper(
- void *userData, int32_t index, uint8_t **yuv);
-
- static void UnbindFrame(void *userData, int32_t index);
-
- int32_t activateSPS(
- unsigned int sizeInMbs, unsigned int numBuffers);
-
- int32_t bindFrame(int32_t index, uint8_t **yuv);
-
- void releaseFrames();
-
- MediaBuffer *drainOutputBuffer();
-
- AVCDecoder(const AVCDecoder &);
- AVCDecoder &operator=(const AVCDecoder &);
-};
-
-} // namespace android
-
-#endif // AVC_DECODER_H_
diff --git a/media/libstagefright/include/AwesomePlayer.h b/media/libstagefright/include/AwesomePlayer.h
index 0985f47..06e9468 100644
--- a/media/libstagefright/include/AwesomePlayer.h
+++ b/media/libstagefright/include/AwesomePlayer.h
@@ -41,7 +41,7 @@
class DrmManagerClinet;
class DecryptHandle;
-class TimedTextPlayer;
+class TimedTextDriver;
struct WVMExtractor;
struct AwesomeRenderer : public RefBase {
@@ -90,6 +90,7 @@
status_t setParameter(int key, const Parcel &request);
status_t getParameter(int key, Parcel *reply);
+ status_t invoke(const Parcel &request, Parcel *reply);
status_t setCacheStatCollectFreq(const Parcel &request);
status_t seekTo(int64_t timeUs);
@@ -100,8 +101,6 @@
void postAudioEOS(int64_t delayUs = 0ll);
void postAudioSeekComplete();
- status_t setTimedTextTrackIndex(int32_t index);
-
status_t dump(int fd, const Vector<String16> &args) const;
private:
@@ -136,7 +135,7 @@
INCOGNITO = 0x8000,
TEXT_RUNNING = 0x10000,
- TEXTPLAYER_STARTED = 0x20000,
+ TEXTPLAYER_INITIALIZED = 0x20000,
SLOW_DECODER_HACK = 0x40000,
};
@@ -232,7 +231,7 @@
sp<DecryptHandle> mDecryptHandle;
int64_t mLastVideoTimeUs;
- TimedTextPlayer *mTextPlayer;
+ TimedTextDriver *mTextDriver;
mutable Mutex mTimedTextLock;
sp<WVMExtractor> mWVMExtractor;
@@ -258,7 +257,7 @@
void setVideoSource(sp<MediaSource> source);
status_t initVideoDecoder(uint32_t flags = 0);
- void addTextSource(sp<MediaSource> source);
+ void addTextSource(const sp<MediaSource>& source);
void onStreamDone();
@@ -290,6 +289,7 @@
bool isStreamingHTTP() const;
void sendCacheStats();
+ void checkDrmStatus(const sp<DataSource>& dataSource);
enum FlagMode {
SET,
@@ -325,4 +325,3 @@
} // namespace android
#endif // AWESOME_PLAYER_H_
-
diff --git a/media/libstagefright/include/ChromiumHTTPDataSource.h b/media/libstagefright/include/ChromiumHTTPDataSource.h
index 18f8913..82e08fd 100644
--- a/media/libstagefright/include/ChromiumHTTPDataSource.h
+++ b/media/libstagefright/include/ChromiumHTTPDataSource.h
@@ -43,7 +43,7 @@
virtual status_t getSize(off64_t *size);
virtual uint32_t flags();
- virtual sp<DecryptHandle> DrmInitialization();
+ virtual sp<DecryptHandle> DrmInitialization(const char *mime);
virtual void getDrmInfo(sp<DecryptHandle> &handle, DrmManagerClient **client);
diff --git a/media/libstagefright/include/DataUriSource.h b/media/libstagefright/include/DataUriSource.h
new file mode 100644
index 0000000..d223c06
--- /dev/null
+++ b/media/libstagefright/include/DataUriSource.h
@@ -0,0 +1,76 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef DATA_URI_SOURCE_H_
+
+#define DATA_URI_SOURCE_H_
+
+#include <stdio.h>
+
+#include <media/stagefright/DataSource.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/foundation/AString.h>
+
+namespace android {
+
+class DataUriSource : public DataSource {
+public:
+ DataUriSource(const char *uri);
+
+ virtual status_t initCheck() const {
+ return mInited;
+ }
+
+ virtual ssize_t readAt(off64_t offset, void *data, size_t size);
+
+ virtual status_t getSize(off64_t *size) {
+ if (mInited != OK) {
+ return mInited;
+ }
+
+ *size = mData.size();
+ return OK;
+ }
+
+ virtual String8 getUri() {
+ return mDataUri;
+ }
+
+ virtual String8 getMIMEType() const {
+ return mMimeType;
+ }
+
+protected:
+ virtual ~DataUriSource() {
+ // Nothing to delete.
+ }
+
+private:
+ const String8 mDataUri;
+
+ String8 mMimeType;
+ // Use AString because individual bytes may not be valid UTF8 chars.
+ AString mData;
+ status_t mInited;
+
+ // Disallow copy and assign.
+ DataUriSource(const DataUriSource &);
+ DataUriSource &operator=(const DataUriSource &);
+};
+
+} // namespace android
+
+#endif // DATA_URI_SOURCE_H_
diff --git a/media/libstagefright/include/G711Decoder.h b/media/libstagefright/include/G711Decoder.h
deleted file mode 100644
index 8b5143a..0000000
--- a/media/libstagefright/include/G711Decoder.h
+++ /dev/null
@@ -1,57 +0,0 @@
-/*
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef G711_DECODER_H_
-
-#define G711_DECODER_H_
-
-#include <media/stagefright/MediaSource.h>
-
-namespace android {
-
-struct MediaBufferGroup;
-
-struct G711Decoder : public MediaSource {
- G711Decoder(const sp<MediaSource> &source);
-
- virtual status_t start(MetaData *params);
- virtual status_t stop();
-
- virtual sp<MetaData> getFormat();
-
- virtual status_t read(
- MediaBuffer **buffer, const ReadOptions *options);
-
-protected:
- virtual ~G711Decoder();
-
-private:
- sp<MediaSource> mSource;
- bool mStarted;
- bool mIsMLaw;
-
- MediaBufferGroup *mBufferGroup;
-
- static void DecodeALaw(int16_t *out, const uint8_t *in, size_t inSize);
- static void DecodeMLaw(int16_t *out, const uint8_t *in, size_t inSize);
-
- G711Decoder(const G711Decoder &);
- G711Decoder &operator=(const G711Decoder &);
-};
-
-} // namespace android
-
-#endif // G711_DECODER_H_
diff --git a/media/libstagefright/include/ID3.h b/media/libstagefright/include/ID3.h
index 98c82a4..8714008 100644
--- a/media/libstagefright/include/ID3.h
+++ b/media/libstagefright/include/ID3.h
@@ -50,7 +50,7 @@
bool done() const;
void getID(String8 *id) const;
- void getString(String8 *s) const;
+ void getString(String8 *s, String8 *ss = NULL) const;
const uint8_t *getData(size_t *length) const;
void next();
@@ -65,6 +65,7 @@
void findFrame();
size_t getHeaderLength() const;
+ void getstring(String8 *s, bool secondhalf) const;
Iterator(const Iterator &);
Iterator &operator=(const Iterator &);
diff --git a/media/libstagefright/include/LiveSession.h b/media/libstagefright/include/LiveSession.h
index 116ed0e..3a11612 100644
--- a/media/libstagefright/include/LiveSession.h
+++ b/media/libstagefright/include/LiveSession.h
@@ -120,7 +120,10 @@
void onMonitorQueue();
void onSeek(const sp<AMessage> &msg);
- status_t fetchFile(const char *url, sp<ABuffer> *out);
+ status_t fetchFile(
+ const char *url, sp<ABuffer> *out,
+ int64_t range_offset = 0, int64_t range_length = -1);
+
sp<M3UParser> fetchPlaylist(const char *url, bool *unchanged);
size_t getBandwidthIndex();
diff --git a/media/libstagefright/include/M3UParser.h b/media/libstagefright/include/M3UParser.h
index 478582d..e30d6fd 100644
--- a/media/libstagefright/include/M3UParser.h
+++ b/media/libstagefright/include/M3UParser.h
@@ -72,6 +72,10 @@
static status_t parseCipherInfo(
const AString &line, sp<AMessage> *meta, const AString &baseURI);
+ static status_t parseByteRange(
+ const AString &line, uint64_t curOffset,
+ uint64_t *length, uint64_t *offset);
+
static status_t ParseInt32(const char *s, int32_t *x);
static status_t ParseDouble(const char *s, double *x);
diff --git a/media/libstagefright/include/M4vH263Decoder.h b/media/libstagefright/include/M4vH263Decoder.h
deleted file mode 100644
index 7d73e30..0000000
--- a/media/libstagefright/include/M4vH263Decoder.h
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- * Copyright (C) 2009 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef M4V_H263_DECODER_H_
-
-#define M4V_H263_DECODER_H_
-
-#include <media/stagefright/MediaBuffer.h>
-#include <media/stagefright/MediaSource.h>
-
-struct tagvideoDecControls;
-
-namespace android {
-
-struct M4vH263Decoder : public MediaSource,
- public MediaBufferObserver {
- M4vH263Decoder(const sp<MediaSource> &source);
-
- virtual status_t start(MetaData *params);
- virtual status_t stop();
-
- virtual sp<MetaData> getFormat();
-
- virtual status_t read(
- MediaBuffer **buffer, const ReadOptions *options);
-
- virtual void signalBufferReturned(MediaBuffer *buffer);
-
-protected:
- virtual ~M4vH263Decoder();
-
-private:
- sp<MediaSource> mSource;
- bool mStarted;
- int32_t mWidth, mHeight;
-
- sp<MetaData> mFormat;
-
- tagvideoDecControls *mHandle;
- MediaBuffer *mFrames[2];
- MediaBuffer *mInputBuffer;
-
- int64_t mNumSamplesOutput;
- int64_t mTargetTimeUs;
-
- void allocateFrames(int32_t width, int32_t height);
- void releaseFrames();
-
- M4vH263Decoder(const M4vH263Decoder &);
- M4vH263Decoder &operator=(const M4vH263Decoder &);
-};
-
-} // namespace android
-
-#endif // M4V_H263_DECODER_H_
diff --git a/media/libstagefright/include/MP3Decoder.h b/media/libstagefright/include/MP3Decoder.h
deleted file mode 100644
index 4086fb6..0000000
--- a/media/libstagefright/include/MP3Decoder.h
+++ /dev/null
@@ -1,67 +0,0 @@
-/*
- * Copyright (C) 2009 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef MP3_DECODER_H_
-
-#define MP3_DECODER_H_
-
-#include <media/stagefright/MediaSource.h>
-
-struct tPVMP3DecoderExternal;
-
-namespace android {
-
-struct MediaBufferGroup;
-
-struct MP3Decoder : public MediaSource {
- MP3Decoder(const sp<MediaSource> &source);
-
- virtual status_t start(MetaData *params);
- virtual status_t stop();
-
- virtual sp<MetaData> getFormat();
-
- virtual status_t read(
- MediaBuffer **buffer, const ReadOptions *options);
-
-protected:
- virtual ~MP3Decoder();
-
-private:
- sp<MediaSource> mSource;
- sp<MetaData> mMeta;
- int32_t mNumChannels;
-
- bool mStarted;
-
- MediaBufferGroup *mBufferGroup;
-
- tPVMP3DecoderExternal *mConfig;
- void *mDecoderBuf;
- int64_t mAnchorTimeUs;
- int64_t mNumFramesOutput;
-
- MediaBuffer *mInputBuffer;
-
- void init();
-
- MP3Decoder(const MP3Decoder &);
- MP3Decoder &operator=(const MP3Decoder &);
-};
-
-} // namespace android
-
-#endif // MP3_DECODER_H_
diff --git a/media/libstagefright/include/NuCachedSource2.h b/media/libstagefright/include/NuCachedSource2.h
index 7a03e7e..c27a29b 100644
--- a/media/libstagefright/include/NuCachedSource2.h
+++ b/media/libstagefright/include/NuCachedSource2.h
@@ -40,7 +40,7 @@
virtual status_t getSize(off64_t *size);
virtual uint32_t flags();
- virtual sp<DecryptHandle> DrmInitialization();
+ virtual sp<DecryptHandle> DrmInitialization(const char* mime);
virtual void getDrmInfo(sp<DecryptHandle> &handle, DrmManagerClient **client);
virtual String8 getUri();
diff --git a/media/libstagefright/include/OMX.h b/media/libstagefright/include/OMX.h
index 53e764f..2c87b34 100644
--- a/media/libstagefright/include/OMX.h
+++ b/media/libstagefright/include/OMX.h
@@ -31,7 +31,7 @@
public:
OMX();
- virtual bool livesLocally(pid_t pid);
+ virtual bool livesLocally(node_id node, pid_t pid);
virtual status_t listNodes(List<ComponentInfo> *list);
diff --git a/media/libstagefright/include/SoftwareRenderer.h b/media/libstagefright/include/SoftwareRenderer.h
index 8f2ea95..7ab0042 100644
--- a/media/libstagefright/include/SoftwareRenderer.h
+++ b/media/libstagefright/include/SoftwareRenderer.h
@@ -20,7 +20,7 @@
#include <media/stagefright/ColorConverter.h>
#include <utils/RefBase.h>
-#include <ui/android_native_buffer.h>
+#include <system/window.h>
namespace android {
diff --git a/media/libstagefright/include/ThrottledSource.h b/media/libstagefright/include/ThrottledSource.h
index 8928a4a..7fe7c06 100644
--- a/media/libstagefright/include/ThrottledSource.h
+++ b/media/libstagefright/include/ThrottledSource.h
@@ -35,6 +35,11 @@
virtual status_t getSize(off64_t *size);
virtual uint32_t flags();
+ virtual String8 getMIMEType() const {
+ return mSource->getMIMEType();
+ }
+
+
private:
Mutex mLock;
diff --git a/media/libstagefright/include/VPXDecoder.h b/media/libstagefright/include/VPXDecoder.h
deleted file mode 100644
index 3b8362d..0000000
--- a/media/libstagefright/include/VPXDecoder.h
+++ /dev/null
@@ -1,62 +0,0 @@
-/*
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef VPX_DECODER_H_
-
-#define VPX_DECODER_H_
-
-#include <media/stagefright/MediaSource.h>
-#include <utils/Vector.h>
-
-namespace android {
-
-struct MediaBufferGroup;
-
-struct VPXDecoder : public MediaSource {
- VPXDecoder(const sp<MediaSource> &source);
-
- virtual status_t start(MetaData *params);
- virtual status_t stop();
-
- virtual sp<MetaData> getFormat();
-
- virtual status_t read(
- MediaBuffer **buffer, const ReadOptions *options);
-
-protected:
- virtual ~VPXDecoder();
-
-private:
- sp<MediaSource> mSource;
- bool mStarted;
- int32_t mWidth, mHeight;
- size_t mBufferSize;
-
- void *mCtx;
- MediaBufferGroup *mBufferGroup;
-
- int64_t mTargetTimeUs;
-
- sp<MetaData> mFormat;
-
- VPXDecoder(const VPXDecoder &);
- VPXDecoder &operator=(const VPXDecoder &);
-};
-
-} // namespace android
-
-#endif // VPX_DECODER_H_
-
diff --git a/media/libstagefright/include/VorbisDecoder.h b/media/libstagefright/include/VorbisDecoder.h
deleted file mode 100644
index 13e8b77..0000000
--- a/media/libstagefright/include/VorbisDecoder.h
+++ /dev/null
@@ -1,72 +0,0 @@
-/*
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef VORBIS_DECODER_H_
-
-#define VORBIS_DECODER_H_
-
-#include <media/stagefright/MediaSource.h>
-
-struct vorbis_dsp_state;
-struct vorbis_info;
-
-namespace android {
-
-struct MediaBufferGroup;
-
-struct VorbisDecoder : public MediaSource {
- VorbisDecoder(const sp<MediaSource> &source);
-
- virtual status_t start(MetaData *params);
- virtual status_t stop();
-
- virtual sp<MetaData> getFormat();
-
- virtual status_t read(
- MediaBuffer **buffer, const ReadOptions *options);
-
-protected:
- virtual ~VorbisDecoder();
-
-private:
- enum {
- kMaxNumSamplesPerBuffer = 8192 * 2
- };
-
- sp<MediaSource> mSource;
- bool mStarted;
-
- MediaBufferGroup *mBufferGroup;
-
- int32_t mNumChannels;
- int32_t mSampleRate;
- int64_t mAnchorTimeUs;
- int64_t mNumFramesOutput;
- int32_t mNumFramesLeftOnPage;
-
- vorbis_dsp_state *mState;
- vorbis_info *mVi;
-
- int decodePacket(MediaBuffer *packet, MediaBuffer *out);
-
- VorbisDecoder(const VorbisDecoder &);
- VorbisDecoder &operator=(const VorbisDecoder &);
-};
-
-} // namespace android
-
-#endif // VORBIS_DECODER_H_
-
diff --git a/media/libstagefright/include/WAVExtractor.h b/media/libstagefright/include/WAVExtractor.h
index ce1f33a..c567ccd 100644
--- a/media/libstagefright/include/WAVExtractor.h
+++ b/media/libstagefright/include/WAVExtractor.h
@@ -47,6 +47,7 @@
bool mValidFormat;
uint16_t mWaveFormat;
uint16_t mNumChannels;
+ uint32_t mChannelMask;
uint32_t mSampleRate;
uint16_t mBitsPerSample;
off64_t mDataOffset;
diff --git a/media/libstagefright/include/WVMExtractor.h b/media/libstagefright/include/WVMExtractor.h
index deecd25..3c3ca89 100644
--- a/media/libstagefright/include/WVMExtractor.h
+++ b/media/libstagefright/include/WVMExtractor.h
@@ -23,6 +23,8 @@
namespace android {
+struct AMessage;
+class String8;
class DataSource;
class WVMLoadableExtractor : public MediaExtractor {
@@ -32,6 +34,7 @@
virtual int64_t getCachedDurationUs(status_t *finalStatus) = 0;
virtual void setAdaptiveStreamingMode(bool adaptive) = 0;
+ virtual void setUID(uid_t uid) = 0;
};
class WVMExtractor : public MediaExtractor {
@@ -58,6 +61,10 @@
// is used.
void setAdaptiveStreamingMode(bool adaptive);
+ void setUID(uid_t uid);
+
+ static bool getVendorLibHandle();
+
protected:
virtual ~WVMExtractor();
@@ -69,6 +76,10 @@
WVMExtractor &operator=(const WVMExtractor &);
};
+bool SniffWVM(
+ const sp<DataSource> &source, String8 *mimeType, float *confidence,
+ sp<AMessage> *);
+
} // namespace android
#endif // DRM_EXTRACTOR_H_
diff --git a/media/libstagefright/include/XINGSeeker.h b/media/libstagefright/include/XINGSeeker.h
index ec5bd9b..c408576 100644
--- a/media/libstagefright/include/XINGSeeker.h
+++ b/media/libstagefright/include/XINGSeeker.h
@@ -31,13 +31,19 @@
virtual bool getDuration(int64_t *durationUs);
virtual bool getOffsetForTime(int64_t *timeUs, off64_t *pos);
+ virtual int32_t getEncoderDelay();
+ virtual int32_t getEncoderPadding();
+
private:
int64_t mFirstFramePos;
int64_t mDurationUs;
int32_t mSizeBytes;
+ int32_t mEncoderDelay;
+ int32_t mEncoderPadding;
// TOC entries in XING header. Skip the first one since it's always 0.
- unsigned char mTableOfContents[99];
+ unsigned char mTOC[99];
+ bool mTOCValid;
XINGSeeker();
diff --git a/media/libstagefright/matroska/Android.mk b/media/libstagefright/matroska/Android.mk
index 1f1c68b..e67da4c 100644
--- a/media/libstagefright/matroska/Android.mk
+++ b/media/libstagefright/matroska/Android.mk
@@ -7,7 +7,7 @@
LOCAL_C_INCLUDES:= \
$(JNI_H_INCLUDE) \
$(TOP)/external/libvpx/mkvparser \
- $(TOP)/frameworks/base/include/media/stagefright/openmax \
+ $(TOP)/frameworks/native/include/media/openmax \
LOCAL_CFLAGS += -Wno-multichar
diff --git a/media/libstagefright/matroska/MatroskaExtractor.cpp b/media/libstagefright/matroska/MatroskaExtractor.cpp
index 4fbf47e..a0db719 100644
--- a/media/libstagefright/matroska/MatroskaExtractor.cpp
+++ b/media/libstagefright/matroska/MatroskaExtractor.cpp
@@ -93,7 +93,10 @@
void advance();
void reset();
- void seek(int64_t seekTimeUs, bool seekToKeyFrame);
+
+ void seek(
+ int64_t seekTimeUs, bool seekToKeyFrame,
+ int64_t *actualFrameTimeUs);
const mkvparser::Block *block() const;
int64_t blockTimeUs() const;
@@ -303,22 +306,52 @@
} while (!eos() && block()->GetTrackNumber() != mTrackNum);
}
-void BlockIterator::seek(int64_t seekTimeUs, bool seekToKeyFrame) {
+void BlockIterator::seek(
+ int64_t seekTimeUs, bool seekToKeyFrame,
+ int64_t *actualFrameTimeUs) {
Mutex::Autolock autoLock(mExtractor->mLock);
- mCluster = mExtractor->mSegment->FindCluster(seekTimeUs * 1000ll);
+ *actualFrameTimeUs = -1ll;
+
+ int64_t seekTimeNs = seekTimeUs * 1000ll;
+
+ mCluster = mExtractor->mSegment->FindCluster(seekTimeNs);
mBlockEntry = NULL;
mBlockEntryIndex = 0;
- do {
- advance_l();
- }
- while (!eos() && block()->GetTrackNumber() != mTrackNum);
+ long prevKeyFrameBlockEntryIndex = -1;
- if (seekToKeyFrame) {
- while (!eos() && !mBlockEntry->GetBlock()->IsKey()) {
- advance_l();
+ for (;;) {
+ advance_l();
+
+ if (eos()) {
+ break;
}
+
+ if (block()->GetTrackNumber() != mTrackNum) {
+ continue;
+ }
+
+ if (block()->IsKey()) {
+ prevKeyFrameBlockEntryIndex = mBlockEntryIndex - 1;
+ }
+
+ int64_t timeNs = block()->GetTime(mCluster);
+
+ if (timeNs >= seekTimeNs) {
+ *actualFrameTimeUs = (timeNs + 500ll) / 1000ll;
+ break;
+ }
+ }
+
+ if (eos()) {
+ return;
+ }
+
+ if (seekToKeyFrame && !block()->IsKey()) {
+ CHECK_GE(prevKeyFrameBlockEntryIndex, 0);
+ mBlockEntryIndex = prevKeyFrameBlockEntryIndex;
+ advance_l();
}
}
@@ -397,6 +430,8 @@
MediaBuffer **out, const ReadOptions *options) {
*out = NULL;
+ int64_t targetSampleTimeUs = -1ll;
+
int64_t seekTimeUs;
ReadOptions::SeekMode mode;
if (options && options->getSeekTo(&seekTimeUs, &mode)
@@ -406,10 +441,14 @@
// Apparently keyframe indication in audio tracks is unreliable,
// fortunately in all our currently supported audio encodings every
// frame is effectively a keyframe.
- mBlockIter.seek(seekTimeUs, !mIsAudio);
+ int64_t actualFrameTimeUs;
+ mBlockIter.seek(seekTimeUs, !mIsAudio, &actualFrameTimeUs);
+
+ if (mode == ReadOptions::SEEK_CLOSEST) {
+ targetSampleTimeUs = actualFrameTimeUs;
+ }
}
-again:
while (mPendingFrames.empty()) {
status_t err = readBlock();
@@ -424,6 +463,11 @@
mPendingFrames.erase(mPendingFrames.begin());
if (mType != AVC) {
+ if (targetSampleTimeUs >= 0ll) {
+ frame->meta_data()->setInt64(
+ kKeyTargetTime, targetSampleTimeUs);
+ }
+
*out = frame;
return OK;
@@ -506,6 +550,11 @@
frame->release();
frame = NULL;
+ if (targetSampleTimeUs >= 0ll) {
+ buffer->meta_data()->setInt64(
+ kKeyTargetTime, targetSampleTimeUs);
+ }
+
*out = buffer;
return OK;
@@ -610,36 +659,41 @@
return mIsLiveStreaming;
}
-static void addESDSFromAudioSpecificInfo(
- const sp<MetaData> &meta, const void *asi, size_t asiSize) {
+static void addESDSFromCodecPrivate(
+ const sp<MetaData> &meta,
+ bool isAudio, const void *priv, size_t privSize) {
static const uint8_t kStaticESDS[] = {
0x03, 22,
0x00, 0x00, // ES_ID
0x00, // streamDependenceFlag, URL_Flag, OCRstreamFlag
0x04, 17,
- 0x40, // Audio ISO/IEC 14496-3
+ 0x40, // ObjectTypeIndication
0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00,
0x05,
- // AudioSpecificInfo (with size prefix) follows
+ // CodecSpecificInfo (with size prefix) follows
};
// Make sure all sizes can be coded in a single byte.
- CHECK(asiSize + 22 - 2 < 128);
- size_t esdsSize = sizeof(kStaticESDS) + asiSize + 1;
+ CHECK(privSize + 22 - 2 < 128);
+ size_t esdsSize = sizeof(kStaticESDS) + privSize + 1;
uint8_t *esds = new uint8_t[esdsSize];
memcpy(esds, kStaticESDS, sizeof(kStaticESDS));
uint8_t *ptr = esds + sizeof(kStaticESDS);
- *ptr++ = asiSize;
- memcpy(ptr, asi, asiSize);
+ *ptr++ = privSize;
+ memcpy(ptr, priv, privSize);
// Increment by codecPrivateSize less 2 bytes that are accounted for
// already in lengths of 22/17
- esds[1] += asiSize - 2;
- esds[6] += asiSize - 2;
+ esds[1] += privSize - 2;
+ esds[6] += privSize - 2;
+
+ // Set ObjectTypeIndication.
+ esds[7] = isAudio ? 0x40 // Audio ISO/IEC 14496-3
+ : 0x20; // Visual ISO/IEC 14496-2
meta->setData(kKeyESDS, 0, esds, esdsSize);
@@ -707,9 +761,21 @@
if (!strcmp("V_MPEG4/ISO/AVC", codecID)) {
meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_AVC);
meta->setData(kKeyAVCC, 0, codecPrivate, codecPrivateSize);
+ } else if (!strcmp("V_MPEG4/ISO/ASP", codecID)) {
+ if (codecPrivateSize > 0) {
+ meta->setCString(
+ kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_MPEG4);
+ addESDSFromCodecPrivate(
+ meta, false, codecPrivate, codecPrivateSize);
+ } else {
+ ALOGW("%s is detected, but does not have configuration.",
+ codecID);
+ continue;
+ }
} else if (!strcmp("V_VP8", codecID)) {
meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_VPX);
} else {
+ ALOGW("%s is not supported.", codecID);
continue;
}
@@ -727,13 +793,16 @@
meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_AAC);
CHECK(codecPrivateSize >= 2);
- addESDSFromAudioSpecificInfo(
- meta, codecPrivate, codecPrivateSize);
+ addESDSFromCodecPrivate(
+ meta, true, codecPrivate, codecPrivateSize);
} else if (!strcmp("A_VORBIS", codecID)) {
meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_VORBIS);
addVorbisCodecInfo(meta, codecPrivate, codecPrivateSize);
+ } else if (!strcmp("A_MPEG/L3", codecID)) {
+ meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_MPEG);
} else {
+ ALOGW("%s is not supported.", codecID);
continue;
}
diff --git a/media/libstagefright/mpeg2ts/Android.mk b/media/libstagefright/mpeg2ts/Android.mk
index 578c669..ac4c2a1 100644
--- a/media/libstagefright/mpeg2ts/Android.mk
+++ b/media/libstagefright/mpeg2ts/Android.mk
@@ -11,8 +11,8 @@
LOCAL_C_INCLUDES:= \
$(JNI_H_INCLUDE) \
- $(TOP)/frameworks/base/include/media/stagefright/openmax \
- $(TOP)/frameworks/base/media/libstagefright
+ $(TOP)/frameworks/base/media/libstagefright \
+ $(TOP)/frameworks/native/include/media/openmax
LOCAL_MODULE:= libstagefright_mpeg2ts
diff --git a/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp b/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp
index 03033f5..e1589b4 100644
--- a/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp
+++ b/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp
@@ -22,8 +22,8 @@
#include "include/LiveSession.h"
#include "include/NuCachedSource2.h"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/DataSource.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaSource.h>
diff --git a/media/libstagefright/omx/Android.mk b/media/libstagefright/omx/Android.mk
index d844f3d..083c7ef 100644
--- a/media/libstagefright/omx/Android.mk
+++ b/media/libstagefright/omx/Android.mk
@@ -5,7 +5,6 @@
LOCAL_SRC_FILES:= \
OMX.cpp \
- OMXComponentBase.cpp \
OMXMaster.cpp \
OMXNodeInstance.cpp \
SimpleSoftOMXComponent.cpp \
@@ -14,7 +13,8 @@
LOCAL_C_INCLUDES += \
frameworks/base/media/libstagefright \
- $(TOP)/frameworks/base/include/media/stagefright/openmax
+ $(TOP)/frameworks/native/include/media/hardware \
+ $(TOP)/frameworks/native/include/media/openmax
LOCAL_SHARED_LIBRARIES := \
libbinder \
diff --git a/media/libstagefright/omx/OMX.cpp b/media/libstagefright/omx/OMX.cpp
index 694b12d..f11fcd2 100644
--- a/media/libstagefright/omx/OMX.cpp
+++ b/media/libstagefright/omx/OMX.cpp
@@ -25,7 +25,7 @@
#include "../include/OMXNodeInstance.h"
#include <binder/IMemory.h>
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <utils/threads.h>
#include "OMXMaster.h"
@@ -102,7 +102,7 @@
if (status != WOULD_BLOCK) {
// Other than join to self, the only other error return codes are
// whatever readyToRun() returns, and we don't override that
- CHECK_EQ(status, NO_ERROR);
+ CHECK_EQ(status, (status_t)NO_ERROR);
}
}
@@ -185,7 +185,7 @@
instance->onObserverDied(mMaster);
}
-bool OMX::livesLocally(pid_t pid) {
+bool OMX::livesLocally(node_id node, pid_t pid) {
return pid == getpid();
}
diff --git a/media/libstagefright/omx/OMXComponentBase.cpp b/media/libstagefright/omx/OMXComponentBase.cpp
deleted file mode 100644
index 35227a0..0000000
--- a/media/libstagefright/omx/OMXComponentBase.cpp
+++ /dev/null
@@ -1,201 +0,0 @@
-/*
- * Copyright (C) 2009 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include "OMXComponentBase.h"
-
-#include <stdlib.h>
-
-#include <media/stagefright/MediaDebug.h>
-
-namespace android {
-
-OMXComponentBase::OMXComponentBase(
- const OMX_CALLBACKTYPE *callbacks,
- OMX_PTR appData)
- : mCallbacks(callbacks),
- mAppData(appData),
- mComponentHandle(NULL) {
-}
-
-OMXComponentBase::~OMXComponentBase() {}
-
-void OMXComponentBase::setComponentHandle(OMX_COMPONENTTYPE *handle) {
- CHECK_EQ(mComponentHandle, NULL);
- mComponentHandle = handle;
-}
-
-void OMXComponentBase::postEvent(
- OMX_EVENTTYPE event, OMX_U32 param1, OMX_U32 param2) {
- (*mCallbacks->EventHandler)(
- mComponentHandle, mAppData, event, param1, param2, NULL);
-}
-
-void OMXComponentBase::postFillBufferDone(OMX_BUFFERHEADERTYPE *bufHdr) {
- (*mCallbacks->FillBufferDone)(mComponentHandle, mAppData, bufHdr);
-}
-
-void OMXComponentBase::postEmptyBufferDone(OMX_BUFFERHEADERTYPE *bufHdr) {
- (*mCallbacks->EmptyBufferDone)(mComponentHandle, mAppData, bufHdr);
-}
-
-static OMXComponentBase *getBase(OMX_HANDLETYPE hComponent) {
- return (OMXComponentBase *)
- ((OMX_COMPONENTTYPE *)hComponent)->pComponentPrivate;
-}
-
-static OMX_ERRORTYPE SendCommandWrapper(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_IN OMX_COMMANDTYPE Cmd,
- OMX_IN OMX_U32 nParam1,
- OMX_IN OMX_PTR pCmdData) {
- return getBase(hComponent)->sendCommand(Cmd, nParam1, pCmdData);
-}
-
-static OMX_ERRORTYPE GetParameterWrapper(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_IN OMX_INDEXTYPE nParamIndex,
- OMX_INOUT OMX_PTR pComponentParameterStructure) {
- return getBase(hComponent)->getParameter(
- nParamIndex, pComponentParameterStructure);
-}
-
-static OMX_ERRORTYPE SetParameterWrapper(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_IN OMX_INDEXTYPE nIndex,
- OMX_IN OMX_PTR pComponentParameterStructure) {
- return getBase(hComponent)->getParameter(
- nIndex, pComponentParameterStructure);
-}
-
-static OMX_ERRORTYPE GetConfigWrapper(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_IN OMX_INDEXTYPE nIndex,
- OMX_INOUT OMX_PTR pComponentConfigStructure) {
- return getBase(hComponent)->getConfig(nIndex, pComponentConfigStructure);
-}
-
-static OMX_ERRORTYPE SetConfigWrapper(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_IN OMX_INDEXTYPE nIndex,
- OMX_IN OMX_PTR pComponentConfigStructure) {
- return getBase(hComponent)->setConfig(nIndex, pComponentConfigStructure);
-}
-
-static OMX_ERRORTYPE GetExtensionIndexWrapper(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_IN OMX_STRING cParameterName,
- OMX_OUT OMX_INDEXTYPE* pIndexType) {
- return getBase(hComponent)->getExtensionIndex(cParameterName, pIndexType);
-}
-
-static OMX_ERRORTYPE GetStateWrapper(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_OUT OMX_STATETYPE* pState) {
- return getBase(hComponent)->getState(pState);
-}
-
-static OMX_ERRORTYPE UseBufferWrapper(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_INOUT OMX_BUFFERHEADERTYPE** ppBufferHdr,
- OMX_IN OMX_U32 nPortIndex,
- OMX_IN OMX_PTR pAppPrivate,
- OMX_IN OMX_U32 nSizeBytes,
- OMX_IN OMX_U8* pBuffer) {
- return getBase(hComponent)->useBuffer(
- ppBufferHdr, nPortIndex, pAppPrivate, nSizeBytes, pBuffer);
-}
-
-static OMX_ERRORTYPE AllocateBufferWrapper(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_INOUT OMX_BUFFERHEADERTYPE** ppBuffer,
- OMX_IN OMX_U32 nPortIndex,
- OMX_IN OMX_PTR pAppPrivate,
- OMX_IN OMX_U32 nSizeBytes) {
- return getBase(hComponent)->allocateBuffer(
- ppBuffer, nPortIndex, pAppPrivate, nSizeBytes);
-}
-
-static OMX_ERRORTYPE FreeBufferWrapper(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_IN OMX_U32 nPortIndex,
- OMX_IN OMX_BUFFERHEADERTYPE* pBuffer) {
- return getBase(hComponent)->freeBuffer(nPortIndex, pBuffer);
-}
-
-static OMX_ERRORTYPE EmptyThisBufferWrapper(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_IN OMX_BUFFERHEADERTYPE* pBuffer) {
- return getBase(hComponent)->emptyThisBuffer(pBuffer);
-}
-
-static OMX_ERRORTYPE FillThisBufferWrapper(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_IN OMX_BUFFERHEADERTYPE* pBuffer) {
- return getBase(hComponent)->fillThisBuffer(pBuffer);
-}
-
-static OMX_ERRORTYPE ComponentDeInitWrapper(
- OMX_IN OMX_HANDLETYPE hComponent) {
- delete getBase(hComponent);
- delete (OMX_COMPONENTTYPE *)hComponent;
-
- return OMX_ErrorNone;
-}
-
-static OMX_ERRORTYPE ComponentRoleEnumWrapper(
- OMX_IN OMX_HANDLETYPE hComponent,
- OMX_OUT OMX_U8 *cRole,
- OMX_IN OMX_U32 nIndex) {
- return getBase(hComponent)->enumerateRoles(cRole, nIndex);
-}
-
-// static
-OMX_COMPONENTTYPE *OMXComponentBase::MakeComponent(OMXComponentBase *base) {
- OMX_COMPONENTTYPE *result = new OMX_COMPONENTTYPE;
-
- result->nSize = sizeof(OMX_COMPONENTTYPE);
- result->nVersion.s.nVersionMajor = 1;
- result->nVersion.s.nVersionMinor = 0;
- result->nVersion.s.nRevision = 0;
- result->nVersion.s.nStep = 0;
- result->pComponentPrivate = base;
- result->pApplicationPrivate = NULL;
-
- result->GetComponentVersion = NULL;
- result->SendCommand = SendCommandWrapper;
- result->GetParameter = GetParameterWrapper;
- result->SetParameter = SetParameterWrapper;
- result->GetConfig = GetConfigWrapper;
- result->SetConfig = SetConfigWrapper;
- result->GetExtensionIndex = GetExtensionIndexWrapper;
- result->GetState = GetStateWrapper;
- result->ComponentTunnelRequest = NULL;
- result->UseBuffer = UseBufferWrapper;
- result->AllocateBuffer = AllocateBufferWrapper;
- result->FreeBuffer = FreeBufferWrapper;
- result->EmptyThisBuffer = EmptyThisBufferWrapper;
- result->FillThisBuffer = FillThisBufferWrapper;
- result->SetCallbacks = NULL;
- result->ComponentDeInit = ComponentDeInitWrapper;
- result->UseEGLImage = NULL;
- result->ComponentRoleEnum = ComponentRoleEnumWrapper;
-
- base->setComponentHandle(result);
-
- return result;
-}
-
-} // namespace android
diff --git a/media/libstagefright/omx/OMXComponentBase.h b/media/libstagefright/omx/OMXComponentBase.h
deleted file mode 100644
index fd0df0b..0000000
--- a/media/libstagefright/omx/OMXComponentBase.h
+++ /dev/null
@@ -1,96 +0,0 @@
-/*
- * Copyright (C) 2009 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef OMX_COMPONENT_BASE_H_
-
-#define OMX_COMPONENT_BASE_H_
-
-#include <OMX_Component.h>
-
-namespace android {
-
-struct OMXComponentBase {
- OMXComponentBase(
- const OMX_CALLBACKTYPE *callbacks,
- OMX_PTR appData);
-
- virtual ~OMXComponentBase();
-
- virtual OMX_ERRORTYPE sendCommand(
- OMX_COMMANDTYPE cmd, OMX_U32 param, OMX_PTR cmdData) = 0;
-
- virtual OMX_ERRORTYPE getParameter(
- OMX_INDEXTYPE index, OMX_PTR params) = 0;
-
- virtual OMX_ERRORTYPE setParameter(
- OMX_INDEXTYPE index, const OMX_PTR params) = 0;
-
- virtual OMX_ERRORTYPE getConfig(
- OMX_INDEXTYPE index, OMX_PTR config) = 0;
-
- virtual OMX_ERRORTYPE setConfig(
- OMX_INDEXTYPE index, const OMX_PTR config) = 0;
-
- virtual OMX_ERRORTYPE getExtensionIndex(
- const OMX_STRING name, OMX_INDEXTYPE *index) = 0;
-
- virtual OMX_ERRORTYPE useBuffer(
- OMX_BUFFERHEADERTYPE **bufHdr,
- OMX_U32 portIndex,
- OMX_PTR appPrivate,
- OMX_U32 size,
- OMX_U8 *buffer) = 0;
-
- virtual OMX_ERRORTYPE allocateBuffer(
- OMX_BUFFERHEADERTYPE **bufHdr,
- OMX_U32 portIndex,
- OMX_PTR appPrivate,
- OMX_U32 size) = 0;
-
- virtual OMX_ERRORTYPE freeBuffer(
- OMX_U32 portIndex,
- OMX_BUFFERHEADERTYPE *buffer) = 0;
-
- virtual OMX_ERRORTYPE emptyThisBuffer(OMX_BUFFERHEADERTYPE *buffer) = 0;
- virtual OMX_ERRORTYPE fillThisBuffer(OMX_BUFFERHEADERTYPE *buffer) = 0;
-
- virtual OMX_ERRORTYPE enumerateRoles(OMX_U8 *role, OMX_U32 index) = 0;
-
- virtual OMX_ERRORTYPE getState(OMX_STATETYPE *state) = 0;
-
- // Wraps a given OMXComponentBase instance into an OMX_COMPONENTTYPE
- // as required by OpenMAX APIs.
- static OMX_COMPONENTTYPE *MakeComponent(OMXComponentBase *base);
-
-protected:
- void postEvent(OMX_EVENTTYPE event, OMX_U32 param1, OMX_U32 param2);
- void postFillBufferDone(OMX_BUFFERHEADERTYPE *bufHdr);
- void postEmptyBufferDone(OMX_BUFFERHEADERTYPE *bufHdr);
-
-private:
- void setComponentHandle(OMX_COMPONENTTYPE *handle);
-
- const OMX_CALLBACKTYPE *mCallbacks;
- OMX_PTR mAppData;
- OMX_COMPONENTTYPE *mComponentHandle;
-
- OMXComponentBase(const OMXComponentBase &);
- OMXComponentBase &operator=(const OMXComponentBase &);
-};
-
-} // namespace android
-
-#endif // OMX_COMPONENT_BASE_H_
diff --git a/media/libstagefright/omx/OMXMaster.cpp b/media/libstagefright/omx/OMXMaster.cpp
index d698939..6b6d0ab 100644
--- a/media/libstagefright/omx/OMXMaster.cpp
+++ b/media/libstagefright/omx/OMXMaster.cpp
@@ -24,7 +24,7 @@
#include <dlfcn.h>
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
namespace android {
diff --git a/media/libstagefright/omx/OMXMaster.h b/media/libstagefright/omx/OMXMaster.h
index feee1f9..6069741 100644
--- a/media/libstagefright/omx/OMXMaster.h
+++ b/media/libstagefright/omx/OMXMaster.h
@@ -18,7 +18,7 @@
#define OMX_MASTER_H_
-#include <media/stagefright/OMXPluginBase.h>
+#include <OMXPluginBase.h>
#include <utils/threads.h>
#include <utils/KeyedVector.h>
diff --git a/media/libstagefright/omx/OMXNodeInstance.cpp b/media/libstagefright/omx/OMXNodeInstance.cpp
index 8938e33..bff3def 100644
--- a/media/libstagefright/omx/OMXNodeInstance.cpp
+++ b/media/libstagefright/omx/OMXNodeInstance.cpp
@@ -24,8 +24,8 @@
#include <OMX_Component.h>
#include <binder/IMemory.h>
-#include <media/stagefright/HardwareAPI.h>
-#include <media/stagefright/MediaDebug.h>
+#include <HardwareAPI.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaErrors.h>
namespace android {
@@ -91,11 +91,11 @@
}
OMXNodeInstance::~OMXNodeInstance() {
- CHECK_EQ(mHandle, NULL);
+ CHECK(mHandle == NULL);
}
void OMXNodeInstance::setHandle(OMX::node_id node_id, OMX_HANDLETYPE handle) {
- CHECK_EQ(mHandle, NULL);
+ CHECK(mHandle == NULL);
mNodeID = node_id;
mHandle = handle;
}
diff --git a/media/libstagefright/omx/SimpleSoftOMXComponent.cpp b/media/libstagefright/omx/SimpleSoftOMXComponent.cpp
index 0914f32..c79e01f 100644
--- a/media/libstagefright/omx/SimpleSoftOMXComponent.cpp
+++ b/media/libstagefright/omx/SimpleSoftOMXComponent.cpp
@@ -333,8 +333,9 @@
void SimpleSoftOMXComponent::onMessageReceived(const sp<AMessage> &msg) {
Mutex::Autolock autoLock(mLock);
-
- switch (msg->what()) {
+ uint32_t msgType = msg->what();
+ ALOGV("msgType = %d", msgType);
+ switch (msgType) {
case kWhatSendCommand:
{
int32_t cmd, param;
@@ -354,27 +355,27 @@
CHECK(mState == OMX_StateExecuting && mTargetState == mState);
bool found = false;
- for (size_t i = 0; i < mPorts.size(); ++i) {
- PortInfo *port = &mPorts.editItemAt(i);
+ size_t portIndex = (kWhatEmptyThisBuffer == msgType)?
+ header->nInputPortIndex: header->nOutputPortIndex;
+ PortInfo *port = &mPorts.editItemAt(portIndex);
- for (size_t j = 0; j < port->mBuffers.size(); ++j) {
- BufferInfo *buffer = &port->mBuffers.editItemAt(j);
+ for (size_t j = 0; j < port->mBuffers.size(); ++j) {
+ BufferInfo *buffer = &port->mBuffers.editItemAt(j);
- if (buffer->mHeader == header) {
- CHECK(!buffer->mOwnedByUs);
+ if (buffer->mHeader == header) {
+ CHECK(!buffer->mOwnedByUs);
- buffer->mOwnedByUs = true;
+ buffer->mOwnedByUs = true;
- CHECK((msg->what() == kWhatEmptyThisBuffer
- && port->mDef.eDir == OMX_DirInput)
- || (port->mDef.eDir == OMX_DirOutput));
+ CHECK((msgType == kWhatEmptyThisBuffer
+ && port->mDef.eDir == OMX_DirInput)
+ || (port->mDef.eDir == OMX_DirOutput));
- port->mQueue.push_back(buffer);
- onQueueFilled(i);
+ port->mQueue.push_back(buffer);
+ onQueueFilled(portIndex);
- found = true;
- break;
- }
+ found = true;
+ break;
}
}
diff --git a/media/libstagefright/omx/SoftOMXPlugin.cpp b/media/libstagefright/omx/SoftOMXPlugin.cpp
index da3ae42..99ffe7d 100644
--- a/media/libstagefright/omx/SoftOMXPlugin.cpp
+++ b/media/libstagefright/omx/SoftOMXPlugin.cpp
@@ -35,8 +35,11 @@
} kComponents[] = {
{ "OMX.google.aac.decoder", "aacdec", "audio_decoder.aac" },
+ { "OMX.google.aac.encoder", "aacenc", "audio_encoder.aac" },
{ "OMX.google.amrnb.decoder", "amrdec", "audio_decoder.amrnb" },
+ { "OMX.google.amrnb.encoder", "amrnbenc", "audio_encoder.amrnb" },
{ "OMX.google.amrwb.decoder", "amrdec", "audio_decoder.amrwb" },
+ { "OMX.google.amrwb.encoder", "amrwbenc", "audio_encoder.amrwb" },
{ "OMX.google.h264.decoder", "h264dec", "video_decoder.avc" },
{ "OMX.google.g711.alaw.decoder", "g711dec", "audio_decoder.g711alaw" },
{ "OMX.google.g711.mlaw.decoder", "g711dec", "audio_decoder.g711mlaw" },
diff --git a/media/libstagefright/omx/SoftOMXPlugin.h b/media/libstagefright/omx/SoftOMXPlugin.h
index f93c323..c89cd87 100644
--- a/media/libstagefright/omx/SoftOMXPlugin.h
+++ b/media/libstagefright/omx/SoftOMXPlugin.h
@@ -19,7 +19,7 @@
#define SOFT_OMX_PLUGIN_H_
#include <media/stagefright/foundation/ABase.h>
-#include <media/stagefright/OMXPluginBase.h>
+#include <OMXPluginBase.h>
namespace android {
diff --git a/media/libstagefright/omx/tests/Android.mk b/media/libstagefright/omx/tests/Android.mk
index bf69428..07d47a8 100644
--- a/media/libstagefright/omx/tests/Android.mk
+++ b/media/libstagefright/omx/tests/Android.mk
@@ -5,13 +5,15 @@
OMXHarness.cpp \
LOCAL_SHARED_LIBRARIES := \
- libstagefright libbinder libmedia libutils
+ libstagefright libbinder libmedia libutils libstagefright_foundation
-LOCAL_C_INCLUDES:= \
+LOCAL_C_INCLUDES := \
$(JNI_H_INCLUDE) \
frameworks/base/media/libstagefright \
- $(TOP)/frameworks/base/include/media/stagefright/openmax
+ $(TOP)/frameworks/native/include/media/openmax
-LOCAL_MODULE:= omx_tests
+LOCAL_MODULE := omx_tests
+
+LOCAL_MODULE_TAGS := tests
include $(BUILD_EXECUTABLE)
diff --git a/media/libstagefright/omx/tests/OMXHarness.cpp b/media/libstagefright/omx/tests/OMXHarness.cpp
index 8faf544..fab1771 100644
--- a/media/libstagefright/omx/tests/OMXHarness.cpp
+++ b/media/libstagefright/omx/tests/OMXHarness.cpp
@@ -26,9 +26,9 @@
#include <binder/IServiceManager.h>
#include <binder/MemoryDealer.h>
#include <media/IMediaPlayerService.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/MediaBuffer.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaExtractor.h>
@@ -155,7 +155,7 @@
if (err == TIMED_OUT) {
return err;
}
- CHECK_EQ(err, OK);
+ CHECK_EQ(err, (status_t)OK);
}
}
@@ -317,7 +317,7 @@
EXPECT_SUCCESS(err, "allocatePortBuffers(input)");
err = dequeueMessageForNode(node, &msg, DEFAULT_TIMEOUT);
- CHECK_EQ(err, TIMED_OUT);
+ CHECK_EQ(err, (status_t)TIMED_OUT);
Vector<Buffer> outputBuffers;
err = allocatePortBuffers(dealer, node, 1, &outputBuffers);
@@ -412,7 +412,7 @@
// Make sure node doesn't just transition to loaded before we are done
// freeing all input and output buffers.
err = dequeueMessageForNode(node, &msg, DEFAULT_TIMEOUT);
- CHECK_EQ(err, TIMED_OUT);
+ CHECK_EQ(err, (status_t)TIMED_OUT);
for (size_t i = 0; i < inputBuffers.size(); ++i) {
err = mOMX->freeBuffer(node, 0, inputBuffers[i].mID);
@@ -420,7 +420,7 @@
}
err = dequeueMessageForNode(node, &msg, DEFAULT_TIMEOUT);
- CHECK_EQ(err, TIMED_OUT);
+ CHECK_EQ(err, (status_t)TIMED_OUT);
for (size_t i = 0; i < outputBuffers.size(); ++i) {
err = mOMX->freeBuffer(node, 1, outputBuffers[i].mID);
@@ -584,7 +584,7 @@
return UNKNOWN_ERROR;
}
- CHECK_EQ(seekSource->start(), OK);
+ CHECK_EQ(seekSource->start(), (status_t)OK);
sp<MediaSource> codec = OMXCodec::Create(
mOMX, source->getFormat(), false /* createEncoder */,
@@ -592,7 +592,7 @@
CHECK(codec != NULL);
- CHECK_EQ(codec->start(), OK);
+ CHECK_EQ(codec->start(), (status_t)OK);
int64_t durationUs;
CHECK(source->getFormat()->findInt64(kKeyDuration, &durationUs));
@@ -638,7 +638,7 @@
requestedSeekTimeUs, MediaSource::ReadOptions::SEEK_NEXT_SYNC);
if (seekSource->read(&buffer, &options) != OK) {
- CHECK_EQ(buffer, NULL);
+ CHECK(buffer == NULL);
actualSeekTimeUs = -1;
} else {
CHECK(buffer != NULL);
@@ -659,7 +659,7 @@
err = codec->read(&buffer, &options);
options.clearSeekTo();
if (err == INFO_FORMAT_CHANGED) {
- CHECK_EQ(buffer, NULL);
+ CHECK(buffer == NULL);
continue;
}
if (err == OK) {
@@ -670,7 +670,7 @@
continue;
}
} else {
- CHECK_EQ(buffer, NULL);
+ CHECK(buffer == NULL);
}
break;
@@ -679,7 +679,7 @@
if (requestedSeekTimeUs < 0) {
// Linear read.
if (err != OK) {
- CHECK_EQ(buffer, NULL);
+ CHECK(buffer == NULL);
} else {
CHECK(buffer != NULL);
buffer->release();
@@ -694,8 +694,8 @@
"We attempted to seek beyond EOS and expected "
"ERROR_END_OF_STREAM to be returned, but instead "
"we found some other error.");
- CHECK_EQ(err, ERROR_END_OF_STREAM);
- CHECK_EQ(buffer, NULL);
+ CHECK_EQ(err, (status_t)ERROR_END_OF_STREAM);
+ CHECK(buffer == NULL);
} else {
EXPECT(err == OK,
"Expected a valid buffer to be returned from "
@@ -715,7 +715,7 @@
buffer->release();
buffer = NULL;
- CHECK_EQ(codec->stop(), OK);
+ CHECK_EQ(codec->stop(), (status_t)OK);
return UNKNOWN_ERROR;
}
@@ -725,7 +725,7 @@
}
}
- CHECK_EQ(codec->stop(), OK);
+ CHECK_EQ(codec->stop(), (status_t)OK);
return OK;
}
@@ -841,7 +841,7 @@
srand(seed);
sp<Harness> h = new Harness;
- CHECK_EQ(h->initCheck(), OK);
+ CHECK_EQ(h->initCheck(), (status_t)OK);
if (argc == 0) {
h->testAll();
diff --git a/media/libstagefright/rtsp/AAMRAssembler.cpp b/media/libstagefright/rtsp/AAMRAssembler.cpp
index 9d72b1f..fb8abc5 100644
--- a/media/libstagefright/rtsp/AAMRAssembler.cpp
+++ b/media/libstagefright/rtsp/AAMRAssembler.cpp
@@ -211,7 +211,7 @@
}
sp<AMessage> msg = mNotifyMsg->dup();
- msg->setObject("access-unit", accessUnit);
+ msg->setBuffer("access-unit", accessUnit);
msg->post();
queue->erase(queue->begin());
diff --git a/media/libstagefright/rtsp/AAVCAssembler.cpp b/media/libstagefright/rtsp/AAVCAssembler.cpp
index ed8b1df..7ea132e 100644
--- a/media/libstagefright/rtsp/AAVCAssembler.cpp
+++ b/media/libstagefright/rtsp/AAVCAssembler.cpp
@@ -345,7 +345,7 @@
mAccessUnitDamaged = false;
sp<AMessage> msg = mNotifyMsg->dup();
- msg->setObject("access-unit", accessUnit);
+ msg->setBuffer("access-unit", accessUnit);
msg->post();
}
diff --git a/media/libstagefright/rtsp/AH263Assembler.cpp b/media/libstagefright/rtsp/AH263Assembler.cpp
index 498295c..ded70fa 100644
--- a/media/libstagefright/rtsp/AH263Assembler.cpp
+++ b/media/libstagefright/rtsp/AH263Assembler.cpp
@@ -166,7 +166,7 @@
mAccessUnitDamaged = false;
sp<AMessage> msg = mNotifyMsg->dup();
- msg->setObject("access-unit", accessUnit);
+ msg->setBuffer("access-unit", accessUnit);
msg->post();
}
diff --git a/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp b/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
index b0c7007..24c2f30 100644
--- a/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
+++ b/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
@@ -571,7 +571,7 @@
mAccessUnitDamaged = false;
sp<AMessage> msg = mNotifyMsg->dup();
- msg->setObject("access-unit", accessUnit);
+ msg->setBuffer("access-unit", accessUnit);
msg->post();
}
diff --git a/media/libstagefright/rtsp/AMPEG4ElementaryAssembler.cpp b/media/libstagefright/rtsp/AMPEG4ElementaryAssembler.cpp
index 2f2e2c2..687d72b 100644
--- a/media/libstagefright/rtsp/AMPEG4ElementaryAssembler.cpp
+++ b/media/libstagefright/rtsp/AMPEG4ElementaryAssembler.cpp
@@ -368,7 +368,7 @@
mAccessUnitDamaged = false;
sp<AMessage> msg = mNotifyMsg->dup();
- msg->setObject("access-unit", accessUnit);
+ msg->setBuffer("access-unit", accessUnit);
msg->post();
}
diff --git a/media/libstagefright/rtsp/ARTPConnection.cpp b/media/libstagefright/rtsp/ARTPConnection.cpp
index 8c9dd8d..44988a3 100644
--- a/media/libstagefright/rtsp/ARTPConnection.cpp
+++ b/media/libstagefright/rtsp/ARTPConnection.cpp
@@ -639,7 +639,7 @@
void ARTPConnection::injectPacket(int index, const sp<ABuffer> &buffer) {
sp<AMessage> msg = new AMessage(kWhatInjectPacket, id());
msg->setInt32("index", index);
- msg->setObject("buffer", buffer);
+ msg->setBuffer("buffer", buffer);
msg->post();
}
@@ -647,10 +647,8 @@
int32_t index;
CHECK(msg->findInt32("index", &index));
- sp<RefBase> obj;
- CHECK(msg->findObject("buffer", &obj));
-
- sp<ABuffer> buffer = static_cast<ABuffer *>(obj.get());
+ sp<ABuffer> buffer;
+ CHECK(msg->findBuffer("buffer", &buffer));
List<StreamInfo>::iterator it = mStreams.begin();
while (it != mStreams.end()
diff --git a/media/libstagefright/rtsp/ARTPSession.cpp b/media/libstagefright/rtsp/ARTPSession.cpp
index 7a05b88..ba4e33c 100644
--- a/media/libstagefright/rtsp/ARTPSession.cpp
+++ b/media/libstagefright/rtsp/ARTPSession.cpp
@@ -145,10 +145,8 @@
break;
}
- sp<RefBase> obj;
- CHECK(msg->findObject("access-unit", &obj));
-
- sp<ABuffer> accessUnit = static_cast<ABuffer *>(obj.get());
+ sp<ABuffer> accessUnit;
+ CHECK(msg->findBuffer("access-unit", &accessUnit));
uint64_t ntpTime;
CHECK(accessUnit->meta()->findInt64(
diff --git a/media/libstagefright/rtsp/ARTSPConnection.cpp b/media/libstagefright/rtsp/ARTSPConnection.cpp
index 80a010e..539a888 100644
--- a/media/libstagefright/rtsp/ARTSPConnection.cpp
+++ b/media/libstagefright/rtsp/ARTSPConnection.cpp
@@ -612,7 +612,7 @@
if (mObserveBinaryMessage != NULL) {
sp<AMessage> notify = mObserveBinaryMessage->dup();
- notify->setObject("buffer", buffer);
+ notify->setBuffer("buffer", buffer);
notify->post();
} else {
ALOGW("received binary data, but no one cares.");
diff --git a/media/libstagefright/rtsp/ARawAudioAssembler.cpp b/media/libstagefright/rtsp/ARawAudioAssembler.cpp
index 98bee82..0da5dd2 100644
--- a/media/libstagefright/rtsp/ARawAudioAssembler.cpp
+++ b/media/libstagefright/rtsp/ARawAudioAssembler.cpp
@@ -94,7 +94,7 @@
}
sp<AMessage> msg = mNotifyMsg->dup();
- msg->setObject("access-unit", buffer);
+ msg->setBuffer("access-unit", buffer);
msg->post();
queue->erase(queue->begin());
diff --git a/media/libstagefright/rtsp/Android.mk b/media/libstagefright/rtsp/Android.mk
index 8230347..b3bc37c 100644
--- a/media/libstagefright/rtsp/Android.mk
+++ b/media/libstagefright/rtsp/Android.mk
@@ -19,9 +19,9 @@
LOCAL_C_INCLUDES:= \
$(JNI_H_INCLUDE) \
- $(TOP)/frameworks/base/include/media/stagefright/openmax \
- $(TOP)/frameworks/base/media/libstagefright/include \
- $(TOP)/external/openssl/include
+ $(TOP)/frameworks/base/media/libstagefright/include \
+ $(TOP)/frameworks/native/include/media/openmax \
+ $(TOP)/external/openssl/include
LOCAL_MODULE:= libstagefright_rtsp
@@ -47,7 +47,7 @@
LOCAL_C_INCLUDES:= \
$(JNI_H_INCLUDE) \
frameworks/base/media/libstagefright \
- $(TOP)/frameworks/base/include/media/stagefright/openmax
+ $(TOP)/frameworks/native/include/media/openmax
LOCAL_CFLAGS += -Wno-multichar
diff --git a/media/libstagefright/rtsp/MyHandler.h b/media/libstagefright/rtsp/MyHandler.h
index 2391c5c..deee30f 100644
--- a/media/libstagefright/rtsp/MyHandler.h
+++ b/media/libstagefright/rtsp/MyHandler.h
@@ -122,6 +122,7 @@
mSetupTracksSuccessful(false),
mSeekPending(false),
mFirstAccessUnit(true),
+ mAllTracksHaveTime(false),
mNTPAnchorUs(-1),
mMediaAnchorUs(-1),
mLastMediaTimeUs(0),
@@ -723,6 +724,7 @@
mSetupTracksSuccessful = false;
mSeekPending = false;
mFirstAccessUnit = true;
+ mAllTracksHaveTime = false;
mNTPAnchorUs = -1;
mMediaAnchorUs = -1;
mNumAccessUnitsReceived = 0;
@@ -855,10 +857,8 @@
return;
}
- sp<RefBase> obj;
- CHECK(msg->findObject("access-unit", &obj));
-
- sp<ABuffer> accessUnit = static_cast<ABuffer *>(obj.get());
+ sp<ABuffer> accessUnit;
+ CHECK(msg->findBuffer("access-unit", &accessUnit));
uint32_t seqNum = (uint32_t)accessUnit->int32Data();
@@ -930,6 +930,7 @@
info->mNTPAnchorUs = -1;
}
+ mAllTracksHaveTime = false;
mNTPAnchorUs = -1;
int64_t timeUs;
@@ -1002,9 +1003,8 @@
case 'biny':
{
- sp<RefBase> obj;
- CHECK(msg->findObject("buffer", &obj));
- sp<ABuffer> buffer = static_cast<ABuffer *>(obj.get());
+ sp<ABuffer> buffer;
+ CHECK(msg->findBuffer("buffer", &buffer));
int32_t index;
CHECK(buffer->meta()->findInt32("index", &index));
@@ -1037,6 +1037,14 @@
ALOGW("Never received any data, disconnecting.");
(new AMessage('abor', id()))->post();
}
+ } else {
+ if (!mAllTracksHaveTime) {
+ ALOGW("We received some RTCP packets, but time "
+ "could not be established on all tracks, now "
+ "using fake timestamps");
+
+ fakeTimestamps();
+ }
}
break;
}
@@ -1211,6 +1219,7 @@
bool mSeekPending;
bool mFirstAccessUnit;
+ bool mAllTracksHaveTime;
int64_t mNTPAnchorUs;
int64_t mMediaAnchorUs;
int64_t mLastMediaTimeUs;
@@ -1357,6 +1366,7 @@
}
void fakeTimestamps() {
+ mNTPAnchorUs = -1ll;
for (size_t i = 0; i < mTracks.size(); ++i) {
onTimeUpdate(i, 0, 0ll);
}
@@ -1377,6 +1387,21 @@
mNTPAnchorUs = ntpTimeUs;
mMediaAnchorUs = mLastMediaTimeUs;
}
+
+ if (!mAllTracksHaveTime) {
+ bool allTracksHaveTime = true;
+ for (size_t i = 0; i < mTracks.size(); ++i) {
+ TrackInfo *track = &mTracks.editItemAt(i);
+ if (track->mNTPAnchorUs < 0) {
+ allTracksHaveTime = false;
+ break;
+ }
+ }
+ if (allTracksHaveTime) {
+ mAllTracksHaveTime = true;
+ ALOGI("Time now established for all tracks.");
+ }
+ }
}
void onAccessUnitComplete(
@@ -1403,7 +1428,7 @@
TrackInfo *track = &mTracks.editItemAt(trackIndex);
- if (mNTPAnchorUs < 0 || mMediaAnchorUs < 0 || track->mNTPAnchorUs < 0) {
+ if (!mAllTracksHaveTime) {
ALOGV("storing accessUnit, no time established yet");
track->mPackets.push_back(accessUnit);
return;
@@ -1460,7 +1485,7 @@
sp<AMessage> msg = mNotify->dup();
msg->setInt32("what", kWhatAccessUnit);
msg->setSize("trackIndex", trackIndex);
- msg->setObject("accessUnit", accessUnit);
+ msg->setBuffer("accessUnit", accessUnit);
msg->post();
}
diff --git a/media/libstagefright/tests/Android.mk b/media/libstagefright/tests/Android.mk
index 357feb1..656a630 100644
--- a/media/libstagefright/tests/Android.mk
+++ b/media/libstagefright/tests/Android.mk
@@ -38,7 +38,7 @@
external/stlport/stlport \
frameworks/base/media/libstagefright \
frameworks/base/media/libstagefright/include \
- $(TOP)/frameworks/base/include/media/stagefright/openmax \
+ $(TOP)/frameworks/native/include/media/openmax \
include $(BUILD_EXECUTABLE)
diff --git a/media/libstagefright/tests/SurfaceMediaSource_test.cpp b/media/libstagefright/tests/SurfaceMediaSource_test.cpp
index 76b507f..fe77cf7 100644
--- a/media/libstagefright/tests/SurfaceMediaSource_test.cpp
+++ b/media/libstagefright/tests/SurfaceMediaSource_test.cpp
@@ -26,16 +26,16 @@
#include <media/stagefright/SurfaceMediaSource.h>
#include <media/mediarecorder.h>
-#include <gui/SurfaceTextureClient.h>
#include <ui/GraphicBuffer.h>
-#include <surfaceflinger/ISurfaceComposer.h>
-#include <surfaceflinger/Surface.h>
-#include <surfaceflinger/SurfaceComposerClient.h>
+#include <gui/SurfaceTextureClient.h>
+#include <gui/ISurfaceComposer.h>
+#include <gui/Surface.h>
+#include <gui/SurfaceComposerClient.h>
#include <binder/ProcessState.h>
#include <ui/FramebufferNativeWindow.h>
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaBufferGroup.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
@@ -107,7 +107,7 @@
window.get(), NULL);
} else {
ALOGV("No actual display. Choosing EGLSurface based on SurfaceMediaSource");
- sp<SurfaceMediaSource> sms = new SurfaceMediaSource(
+ sp<ISurfaceTexture> sms = new SurfaceMediaSource(
getSurfaceWidth(), getSurfaceHeight());
sp<SurfaceTextureClient> stc = new SurfaceTextureClient(sms);
sp<ANativeWindow> window = stc;
@@ -360,7 +360,8 @@
android::ProcessState::self()->startThreadPool();
mSMS = new SurfaceMediaSource(mYuvTexWidth, mYuvTexHeight);
mSMS->setSynchronousMode(true);
- mSTC = new SurfaceTextureClient(mSMS);
+ // Manual cast is required to avoid constructor ambiguity
+ mSTC = new SurfaceTextureClient(static_cast<sp<ISurfaceTexture> >( mSMS));
mANW = mSTC;
}
@@ -395,7 +396,7 @@
ALOGV("SMS-GLTest::SetUp()");
android::ProcessState::self()->startThreadPool();
mSMS = new SurfaceMediaSource(mYuvTexWidth, mYuvTexHeight);
- mSTC = new SurfaceTextureClient(mSMS);
+ mSTC = new SurfaceTextureClient(static_cast<sp<ISurfaceTexture> >( mSMS));
mANW = mSTC;
// Doing the setup related to the GL Side
@@ -475,7 +476,7 @@
mr->setVideoFrameRate(fps);
mr->prepare();
ALOGV("Starting MediaRecorder...");
- CHECK_EQ(OK, mr->start());
+ CHECK_EQ((status_t)OK, mr->start());
return mr;
}
@@ -757,7 +758,7 @@
ASSERT_EQ(NO_ERROR, native_window_api_disconnect(mANW.get(), NATIVE_WINDOW_API_CPU));
ALOGV("Stopping MediaRecorder...");
- CHECK_EQ(OK, mr->stop());
+ CHECK_EQ((status_t)OK, mr->stop());
mr.clear();
close(fd);
}
@@ -773,7 +774,7 @@
ALOGV("Verify creating a surface w/ right config + dummy writer*********");
mSMS = new SurfaceMediaSource(mYuvTexWidth, mYuvTexHeight);
- mSTC = new SurfaceTextureClient(mSMS);
+ mSTC = new SurfaceTextureClient(static_cast<sp<ISurfaceTexture> >( mSMS));
mANW = mSTC;
DummyRecorder writer(mSMS);
@@ -886,7 +887,7 @@
mEglSurface = EGL_NO_SURFACE;
ALOGV("Stopping MediaRecorder...");
- CHECK_EQ(OK, mr->stop());
+ CHECK_EQ((status_t)OK, mr->stop());
mr.clear();
close(fd);
}
@@ -929,7 +930,7 @@
mEglSurface = EGL_NO_SURFACE;
ALOGV("Stopping MediaRecorder...");
- CHECK_EQ(OK, mr->stop());
+ CHECK_EQ((status_t)OK, mr->stop());
mr.clear();
close(fd);
}
diff --git a/media/libstagefright/timedtext/Android.mk b/media/libstagefright/timedtext/Android.mk
index 59d0e15..d2d5f7b 100644
--- a/media/libstagefright/timedtext/Android.mk
+++ b/media/libstagefright/timedtext/Android.mk
@@ -3,14 +3,17 @@
LOCAL_SRC_FILES:= \
TextDescriptions.cpp \
- TimedTextParser.cpp \
+ TimedTextDriver.cpp \
+ TimedText3GPPSource.cpp \
+ TimedTextSource.cpp \
+ TimedTextSRTSource.cpp \
TimedTextPlayer.cpp
LOCAL_CFLAGS += -Wno-multichar
LOCAL_C_INCLUDES:= \
$(JNI_H_INCLUDE) \
- $(TOP)/frameworks/base/media/libstagefright \
- $(TOP)/frameworks/base/include/media/stagefright/openmax
+ $(TOP)/frameworks/base/include/media/stagefright/timedtext \
+ $(TOP)/frameworks/base/media/libstagefright
LOCAL_MODULE:= libstagefright_timedtext
diff --git a/media/libstagefright/timedtext/TimedText3GPPSource.cpp b/media/libstagefright/timedtext/TimedText3GPPSource.cpp
new file mode 100644
index 0000000..4854121
--- /dev/null
+++ b/media/libstagefright/timedtext/TimedText3GPPSource.cpp
@@ -0,0 +1,119 @@
+ /*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "TimedText3GPPSource"
+#include <utils/Log.h>
+
+#include <binder/Parcel.h>
+#include <media/stagefright/foundation/ADebug.h> // CHECK_XX macro
+#include <media/stagefright/MediaBuffer.h>
+#include <media/stagefright/MediaDefs.h> // for MEDIA_MIMETYPE_xxx
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaSource.h>
+#include <media/stagefright/MetaData.h>
+
+#include "TimedText3GPPSource.h"
+#include "TextDescriptions.h"
+
+namespace android {
+
+TimedText3GPPSource::TimedText3GPPSource(const sp<MediaSource>& mediaSource)
+ : mSource(mediaSource) {
+}
+
+TimedText3GPPSource::~TimedText3GPPSource() {
+}
+
+status_t TimedText3GPPSource::read(
+ int64_t *startTimeUs, int64_t *endTimeUs, Parcel *parcel,
+ const MediaSource::ReadOptions *options) {
+ MediaBuffer *textBuffer = NULL;
+ status_t err = mSource->read(&textBuffer, options);
+ if (err != OK) {
+ return err;
+ }
+ CHECK(textBuffer != NULL);
+ textBuffer->meta_data()->findInt64(kKeyTime, startTimeUs);
+ CHECK_GE(*startTimeUs, 0);
+ extractAndAppendLocalDescriptions(*startTimeUs, textBuffer, parcel);
+ textBuffer->release();
+ // endTimeUs is a dummy parameter for 3gpp timed text format.
+ // Set a negative value to it to mark it is unavailable.
+ *endTimeUs = -1;
+ return OK;
+}
+
+// Each text sample consists of a string of text, optionally with sample
+// modifier description. The modifier description could specify a new
+// text style for the string of text. These descriptions are present only
+// if they are needed. This method is used to extract the modifier
+// description and append it at the end of the text.
+status_t TimedText3GPPSource::extractAndAppendLocalDescriptions(
+ int64_t timeUs, const MediaBuffer *textBuffer, Parcel *parcel) {
+ const void *data;
+ size_t size = 0;
+ int32_t flag = TextDescriptions::LOCAL_DESCRIPTIONS;
+
+ const char *mime;
+ CHECK(mSource->getFormat()->findCString(kKeyMIMEType, &mime));
+ CHECK(strcasecmp(mime, MEDIA_MIMETYPE_TEXT_3GPP) == 0);
+
+ data = textBuffer->data();
+ size = textBuffer->size();
+
+ if (size > 0) {
+ parcel->freeData();
+ flag |= TextDescriptions::IN_BAND_TEXT_3GPP;
+ return TextDescriptions::getParcelOfDescriptions(
+ (const uint8_t *)data, size, flag, timeUs / 1000, parcel);
+ }
+ return OK;
+}
+
+// To extract and send the global text descriptions for all the text samples
+// in the text track or text file.
+// TODO: send error message to application via notifyListener()...?
+status_t TimedText3GPPSource::extractGlobalDescriptions(Parcel *parcel) {
+ const void *data;
+ size_t size = 0;
+ int32_t flag = TextDescriptions::GLOBAL_DESCRIPTIONS;
+
+ const char *mime;
+ CHECK(mSource->getFormat()->findCString(kKeyMIMEType, &mime));
+ CHECK(strcasecmp(mime, MEDIA_MIMETYPE_TEXT_3GPP) == 0);
+
+ uint32_t type;
+ // get the 'tx3g' box content. This box contains the text descriptions
+ // used to render the text track
+ if (!mSource->getFormat()->findData(
+ kKeyTextFormatData, &type, &data, &size)) {
+ return ERROR_MALFORMED;
+ }
+
+ if (size > 0) {
+ flag |= TextDescriptions::IN_BAND_TEXT_3GPP;
+ return TextDescriptions::getParcelOfDescriptions(
+ (const uint8_t *)data, size, flag, 0, parcel);
+ }
+ return OK;
+}
+
+sp<MetaData> TimedText3GPPSource::getFormat() {
+ return mSource->getFormat();
+}
+
+} // namespace android
diff --git a/media/libstagefright/timedtext/TimedText3GPPSource.h b/media/libstagefright/timedtext/TimedText3GPPSource.h
new file mode 100644
index 0000000..4170940
--- /dev/null
+++ b/media/libstagefright/timedtext/TimedText3GPPSource.h
@@ -0,0 +1,57 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef TIMED_TEXT_3GPP_SOURCE_H_
+#define TIMED_TEXT_3GPP_SOURCE_H_
+
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaSource.h>
+
+#include "TimedTextSource.h"
+
+namespace android {
+
+class MediaBuffer;
+class Parcel;
+
+class TimedText3GPPSource : public TimedTextSource {
+public:
+ TimedText3GPPSource(const sp<MediaSource>& mediaSource);
+ virtual status_t start() { return mSource->start(); }
+ virtual status_t stop() { return mSource->stop(); }
+ virtual status_t read(
+ int64_t *startTimeUs,
+ int64_t *endTimeUs,
+ Parcel *parcel,
+ const MediaSource::ReadOptions *options = NULL);
+ virtual status_t extractGlobalDescriptions(Parcel *parcel);
+ virtual sp<MetaData> getFormat();
+
+protected:
+ virtual ~TimedText3GPPSource();
+
+private:
+ sp<MediaSource> mSource;
+
+ status_t extractAndAppendLocalDescriptions(
+ int64_t timeUs, const MediaBuffer *textBuffer, Parcel *parcel);
+
+ DISALLOW_EVIL_CONSTRUCTORS(TimedText3GPPSource);
+};
+
+} // namespace android
+
+#endif // TIMED_TEXT_3GPP_SOURCE_H_
diff --git a/media/libstagefright/timedtext/TimedTextDriver.cpp b/media/libstagefright/timedtext/TimedTextDriver.cpp
new file mode 100644
index 0000000..8ee15f8
--- /dev/null
+++ b/media/libstagefright/timedtext/TimedTextDriver.cpp
@@ -0,0 +1,222 @@
+ /*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "TimedTextDriver"
+#include <utils/Log.h>
+
+#include <binder/IPCThreadState.h>
+
+#include <media/mediaplayer.h>
+#include <media/MediaPlayerInterface.h>
+#include <media/stagefright/DataSource.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaSource.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/Utils.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/ALooper.h>
+#include <media/stagefright/timedtext/TimedTextDriver.h>
+
+#include "TextDescriptions.h"
+#include "TimedTextPlayer.h"
+#include "TimedTextSource.h"
+
+namespace android {
+
+TimedTextDriver::TimedTextDriver(
+ const wp<MediaPlayerBase> &listener)
+ : mLooper(new ALooper),
+ mListener(listener),
+ mState(UNINITIALIZED) {
+ mLooper->setName("TimedTextDriver");
+ mLooper->start();
+ mPlayer = new TimedTextPlayer(listener);
+ mLooper->registerHandler(mPlayer);
+}
+
+TimedTextDriver::~TimedTextDriver() {
+ mTextSourceVector.clear();
+ mLooper->stop();
+}
+
+status_t TimedTextDriver::selectTrack_l(int32_t index) {
+ if (index >= (int)(mTextSourceVector.size())) {
+ return BAD_VALUE;
+ }
+
+ sp<TimedTextSource> source;
+ source = mTextSourceVector.itemAt(index);
+ mPlayer->setDataSource(source);
+ if (mState == UNINITIALIZED) {
+ mState = PAUSED;
+ }
+ mCurrentTrackIndex = index;
+ return OK;
+}
+
+status_t TimedTextDriver::start() {
+ Mutex::Autolock autoLock(mLock);
+ switch (mState) {
+ case UNINITIALIZED:
+ return INVALID_OPERATION;
+ case PLAYING:
+ return OK;
+ case PAUSED:
+ mPlayer->start();
+ break;
+ default:
+ TRESPASS();
+ }
+ mState = PLAYING;
+ return OK;
+}
+
+// TODO: Test if pause() works properly.
+// Scenario 1: start - pause - resume
+// Scenario 2: start - seek
+// Scenario 3: start - pause - seek - resume
+status_t TimedTextDriver::pause() {
+ Mutex::Autolock autoLock(mLock);
+ switch (mState) {
+ case UNINITIALIZED:
+ return INVALID_OPERATION;
+ case PLAYING:
+ mPlayer->pause();
+ break;
+ case PAUSED:
+ return OK;
+ default:
+ TRESPASS();
+ }
+ mState = PAUSED;
+ return OK;
+}
+
+status_t TimedTextDriver::selectTrack(int32_t index) {
+ status_t ret = OK;
+ Mutex::Autolock autoLock(mLock);
+ switch (mState) {
+ case UNINITIALIZED:
+ case PAUSED:
+ ret = selectTrack_l(index);
+ break;
+ case PLAYING:
+ mPlayer->pause();
+ ret = selectTrack_l(index);
+ if (ret != OK) {
+ break;
+ }
+ mPlayer->start();
+ break;
+ defaut:
+ TRESPASS();
+ }
+ return ret;
+}
+
+status_t TimedTextDriver::unselectTrack(int32_t index) {
+ if (mCurrentTrackIndex != index) {
+ return INVALID_OPERATION;
+ }
+ status_t err = pause();
+ if (err != OK) {
+ return err;
+ }
+ Mutex::Autolock autoLock(mLock);
+ mState = UNINITIALIZED;
+ return OK;
+}
+
+status_t TimedTextDriver::seekToAsync(int64_t timeUs) {
+ mPlayer->seekToAsync(timeUs);
+ return OK;
+}
+
+status_t TimedTextDriver::addInBandTextSource(
+ const sp<MediaSource>& mediaSource) {
+ sp<TimedTextSource> source =
+ TimedTextSource::CreateTimedTextSource(mediaSource);
+ if (source == NULL) {
+ return ERROR_UNSUPPORTED;
+ }
+ Mutex::Autolock autoLock(mLock);
+ mTextSourceVector.add(source);
+ return OK;
+}
+
+status_t TimedTextDriver::addOutOfBandTextSource(
+ const char *uri, const char *mimeType) {
+ // TODO: Define "TimedTextSource::CreateFromURI(uri)"
+ // and move below lines there..?
+
+ // To support local subtitle file only for now
+ if (strncasecmp("file://", uri, 7)) {
+ return ERROR_UNSUPPORTED;
+ }
+ sp<DataSource> dataSource =
+ DataSource::CreateFromURI(uri);
+ if (dataSource == NULL) {
+ return ERROR_UNSUPPORTED;
+ }
+
+ sp<TimedTextSource> source;
+ if (strcasecmp(mimeType, MEDIA_MIMETYPE_TEXT_SUBRIP) == 0) {
+ source = TimedTextSource::CreateTimedTextSource(
+ dataSource, TimedTextSource::OUT_OF_BAND_FILE_SRT);
+ }
+
+ if (source == NULL) {
+ return ERROR_UNSUPPORTED;
+ }
+
+ Mutex::Autolock autoLock(mLock);
+ mTextSourceVector.add(source);
+ return OK;
+}
+
+status_t TimedTextDriver::addOutOfBandTextSource(
+ int fd, off64_t offset, size_t length, const char *mimeType) {
+ // Not supported yet. This requires DataSource::sniff to detect various text
+ // formats such as srt/smi/ttml.
+ return ERROR_UNSUPPORTED;
+}
+
+void TimedTextDriver::getTrackInfo(Parcel *parcel) {
+ Mutex::Autolock autoLock(mLock);
+ Vector<sp<TimedTextSource> >::const_iterator iter;
+ parcel->writeInt32(mTextSourceVector.size());
+ for (iter = mTextSourceVector.begin();
+ iter != mTextSourceVector.end(); ++iter) {
+ sp<MetaData> meta = (*iter)->getFormat();
+ if (meta != NULL) {
+ // There are two fields.
+ parcel->writeInt32(2);
+
+ // track type.
+ parcel->writeInt32(MEDIA_TRACK_TYPE_TIMEDTEXT);
+
+ const char *lang = "und";
+ meta->findCString(kKeyMediaLanguage, &lang);
+ parcel->writeString16(String16(lang));
+ } else {
+ parcel->writeInt32(0);
+ }
+ }
+}
+
+} // namespace android
diff --git a/media/libstagefright/timedtext/TimedTextParser.cpp b/media/libstagefright/timedtext/TimedTextParser.cpp
deleted file mode 100644
index 0bada16..0000000
--- a/media/libstagefright/timedtext/TimedTextParser.cpp
+++ /dev/null
@@ -1,257 +0,0 @@
-/*
- * Copyright (C) 2011 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include "TimedTextParser.h"
-#include <media/stagefright/DataSource.h>
-
-namespace android {
-
-TimedTextParser::TimedTextParser()
- : mDataSource(NULL),
- mOffset(0),
- mIndex(0) {
-}
-
-TimedTextParser::~TimedTextParser() {
- reset();
-}
-
-status_t TimedTextParser::init(
- const sp<DataSource> &dataSource, FileType fileType) {
- mDataSource = dataSource;
- mFileType = fileType;
-
- status_t err;
- if ((err = scanFile()) != OK) {
- reset();
- return err;
- }
-
- return OK;
-}
-
-void TimedTextParser::reset() {
- mDataSource.clear();
- mTextVector.clear();
- mOffset = 0;
- mIndex = 0;
-}
-
-// scan the text file to get start/stop time and the
-// offset of each piece of text content
-status_t TimedTextParser::scanFile() {
- if (mFileType != OUT_OF_BAND_FILE_SRT) {
- return ERROR_UNSUPPORTED;
- }
-
- off64_t offset = 0;
- int64_t startTimeUs;
- bool endOfFile = false;
-
- while (!endOfFile) {
- TextInfo info;
- status_t err = getNextInSrtFileFormat(&offset, &startTimeUs, &info);
-
- if (err != OK) {
- if (err == ERROR_END_OF_STREAM) {
- endOfFile = true;
- } else {
- return err;
- }
- } else {
- mTextVector.add(startTimeUs, info);
- }
- }
-
- if (mTextVector.isEmpty()) {
- return ERROR_MALFORMED;
- }
- return OK;
-}
-
-// read one line started from *offset and store it into data.
-status_t TimedTextParser::readNextLine(off64_t *offset, AString *data) {
- char character;
-
- data->clear();
-
- while (true) {
- ssize_t err;
- if ((err = mDataSource->readAt(*offset, &character, 1)) < 1) {
- if (err == 0) {
- return ERROR_END_OF_STREAM;
- }
- return ERROR_IO;
- }
-
- (*offset) ++;
-
- // a line could end with CR, LF or CR + LF
- if (character == 10) {
- break;
- } else if (character == 13) {
- if ((err = mDataSource->readAt(*offset, &character, 1)) < 1) {
- if (err == 0) { // end of the stream
- return OK;
- }
- return ERROR_IO;
- }
-
- (*offset) ++;
-
- if (character != 10) {
- (*offset) --;
- }
- break;
- }
-
- data->append(character);
- }
-
- return OK;
-}
-
-/* SRT format:
- * Subtitle number
- * Start time --> End time
- * Text of subtitle (one or more lines)
- * Blank line
- *
- * .srt file example:
- * 1
- * 00:00:20,000 --> 00:00:24,400
- * Altocumulus clouds occur between six thousand
- *
- * 2
- * 00:00:24,600 --> 00:00:27,800
- * and twenty thousand feet above ground level.
- */
-status_t TimedTextParser::getNextInSrtFileFormat(
- off64_t *offset, int64_t *startTimeUs, TextInfo *info) {
- AString data;
- status_t err;
- if ((err = readNextLine(offset, &data)) != OK) {
- return err;
- }
-
- // to skip the first line
- if ((err = readNextLine(offset, &data)) != OK) {
- return err;
- }
-
- int hour1, hour2, min1, min2, sec1, sec2, msec1, msec2;
- // the start time format is: hours:minutes:seconds,milliseconds
- // 00:00:24,600 --> 00:00:27,800
- if (sscanf(data.c_str(), "%02d:%02d:%02d,%03d --> %02d:%02d:%02d,%03d",
- &hour1, &min1, &sec1, &msec1, &hour2, &min2, &sec2, &msec2) != 8) {
- return ERROR_MALFORMED;
- }
-
- *startTimeUs = ((hour1 * 3600 + min1 * 60 + sec1) * 1000 + msec1) * 1000ll;
- info->endTimeUs = ((hour2 * 3600 + min2 * 60 + sec2) * 1000 + msec2) * 1000ll;
- if (info->endTimeUs <= *startTimeUs) {
- return ERROR_MALFORMED;
- }
-
- info->offset = *offset;
-
- bool needMoreData = true;
- while (needMoreData) {
- if ((err = readNextLine(offset, &data)) != OK) {
- if (err == ERROR_END_OF_STREAM) {
- needMoreData = false;
- } else {
- return err;
- }
- }
-
- if (needMoreData) {
- data.trim();
- if (data.empty()) {
- // it's an empty line used to separate two subtitles
- needMoreData = false;
- }
- }
- }
-
- info->textLen = *offset - info->offset;
-
- return OK;
-}
-
-status_t TimedTextParser::getText(
- AString *text, int64_t *startTimeUs, int64_t *endTimeUs,
- const MediaSource::ReadOptions *options) {
- Mutex::Autolock autoLock(mLock);
-
- text->clear();
-
- int64_t seekTimeUs;
- MediaSource::ReadOptions::SeekMode mode;
- if (options && options->getSeekTo(&seekTimeUs, &mode)) {
- int64_t lastEndTimeUs = mTextVector.valueAt(mTextVector.size() - 1).endTimeUs;
- int64_t firstStartTimeUs = mTextVector.keyAt(0);
-
- if (seekTimeUs < 0 || seekTimeUs > lastEndTimeUs) {
- return ERROR_OUT_OF_RANGE;
- } else if (seekTimeUs < firstStartTimeUs) {
- mIndex = 0;
- } else {
- // binary search
- ssize_t low = 0;
- ssize_t high = mTextVector.size() - 1;
- ssize_t mid = 0;
- int64_t currTimeUs;
-
- while (low <= high) {
- mid = low + (high - low)/2;
- currTimeUs = mTextVector.keyAt(mid);
- const int diff = currTimeUs - seekTimeUs;
-
- if (diff == 0) {
- break;
- } else if (diff < 0) {
- low = mid + 1;
- } else {
- if ((high == mid + 1)
- && (seekTimeUs < mTextVector.keyAt(high))) {
- break;
- }
- high = mid - 1;
- }
- }
-
- mIndex = mid;
- }
- }
-
- TextInfo info = mTextVector.valueAt(mIndex);
- *startTimeUs = mTextVector.keyAt(mIndex);
- *endTimeUs = info.endTimeUs;
- mIndex ++;
-
- char *str = new char[info.textLen];
- if (mDataSource->readAt(info.offset, str, info.textLen) < info.textLen) {
- delete[] str;
- return ERROR_IO;
- }
-
- text->append(str, info.textLen);
- delete[] str;
- return OK;
-}
-
-} // namespace android
diff --git a/media/libstagefright/timedtext/TimedTextParser.h b/media/libstagefright/timedtext/TimedTextParser.h
deleted file mode 100644
index 44774c2..0000000
--- a/media/libstagefright/timedtext/TimedTextParser.h
+++ /dev/null
@@ -1,75 +0,0 @@
-/*
- * Copyright (C) 2011 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef TIMED_TEXT_PARSER_H_
-
-#define TIMED_TEXT_PARSER_H_
-
-#include <media/MediaPlayerInterface.h>
-#include <media/stagefright/foundation/ABase.h>
-#include <media/stagefright/foundation/AString.h>
-#include <media/stagefright/MediaSource.h>
-
-namespace android {
-
-class DataSource;
-
-class TimedTextParser : public RefBase {
-public:
- TimedTextParser();
- virtual ~TimedTextParser();
-
- enum FileType {
- OUT_OF_BAND_FILE_SRT = 1,
- };
-
- status_t getText(AString *text, int64_t *startTimeUs, int64_t *endTimeUs,
- const MediaSource::ReadOptions *options = NULL);
- status_t init(const sp<DataSource> &dataSource, FileType fileType);
- void reset();
-
-private:
- Mutex mLock;
-
- sp<DataSource> mDataSource;
- off64_t mOffset;
-
- struct TextInfo {
- int64_t endTimeUs;
- // the offset of the text in the original file
- off64_t offset;
- int textLen;
- };
-
- int mIndex;
- FileType mFileType;
-
- // the key indicated the start time of the text
- KeyedVector<int64_t, TextInfo> mTextVector;
-
- status_t getNextInSrtFileFormat(
- off64_t *offset, int64_t *startTimeUs, TextInfo *info);
- status_t readNextLine(off64_t *offset, AString *data);
-
- status_t scanFile();
-
- DISALLOW_EVIL_CONSTRUCTORS(TimedTextParser);
-};
-
-} // namespace android
-
-#endif // TIMED_TEXT_PARSER_H_
-
diff --git a/media/libstagefright/timedtext/TimedTextPlayer.cpp b/media/libstagefright/timedtext/TimedTextPlayer.cpp
index 3014b0b..917c62a 100644
--- a/media/libstagefright/timedtext/TimedTextPlayer.cpp
+++ b/media/libstagefright/timedtext/TimedTextPlayer.cpp
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2011 The Android Open Source Project
+ * Copyright (C) 2012 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
@@ -18,399 +18,195 @@
#define LOG_TAG "TimedTextPlayer"
#include <utils/Log.h>
-#include <binder/IPCThreadState.h>
-
-#include <media/stagefright/MediaDebug.h>
-#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/timedtext/TimedTextDriver.h>
#include <media/stagefright/MediaErrors.h>
-#include <media/stagefright/MediaSource.h>
-#include <media/stagefright/MetaData.h>
-#include <media/stagefright/MediaBuffer.h>
-#include <media/stagefright/FileSource.h>
-#include <media/stagefright/Utils.h>
+#include <media/MediaPlayerInterface.h>
-#include "include/AwesomePlayer.h"
#include "TimedTextPlayer.h"
-#include "TimedTextParser.h"
-#include "TextDescriptions.h"
+
+#include "TimedTextSource.h"
namespace android {
-struct TimedTextEvent : public TimedEventQueue::Event {
- TimedTextEvent(
- TimedTextPlayer *player,
- void (TimedTextPlayer::*method)())
- : mPlayer(player),
- mMethod(method) {
- }
+static const int64_t kAdjustmentProcessingTimeUs = 100000ll;
+static const int64_t kWaitTimeUsToRetryRead = 100000ll;
-protected:
- virtual ~TimedTextEvent() {}
-
- virtual void fire(TimedEventQueue *queue, int64_t /* now_us */) {
- (mPlayer->*mMethod)();
- }
-
-private:
- TimedTextPlayer *mPlayer;
- void (TimedTextPlayer::*mMethod)();
-
- TimedTextEvent(const TimedTextEvent &);
- TimedTextEvent &operator=(const TimedTextEvent &);
-};
-
-TimedTextPlayer::TimedTextPlayer(
- AwesomePlayer *observer,
- const wp<MediaPlayerBase> &listener,
- TimedEventQueue *queue)
- : mSource(NULL),
- mOutOfBandSource(NULL),
- mSeekTimeUs(0),
- mStarted(false),
- mTextEventPending(false),
- mQueue(queue),
- mListener(listener),
- mObserver(observer),
- mTextBuffer(NULL),
- mTextParser(NULL),
- mTextType(kNoText) {
- mTextEvent = new TimedTextEvent(this, &TimedTextPlayer::onTextEvent);
+TimedTextPlayer::TimedTextPlayer(const wp<MediaPlayerBase> &listener)
+ : mListener(listener),
+ mSource(NULL),
+ mSendSubtitleGeneration(0) {
}
TimedTextPlayer::~TimedTextPlayer() {
- if (mStarted) {
- reset();
+ if (mSource != NULL) {
+ mSource->stop();
+ mSource.clear();
+ mSource = NULL;
}
-
- mTextTrackVector.clear();
- mTextOutOfBandVector.clear();
}
-status_t TimedTextPlayer::start(uint8_t index) {
- CHECK(!mStarted);
-
- if (index >=
- mTextTrackVector.size() + mTextOutOfBandVector.size()) {
- ALOGE("Incorrect text track index: %d", index);
- return BAD_VALUE;
- }
-
- status_t err;
- if (index < mTextTrackVector.size()) { // start an in-band text
- mSource = mTextTrackVector.itemAt(index);
-
- err = mSource->start();
-
- if (err != OK) {
- return err;
- }
- mTextType = kInBandText;
- } else { // start an out-of-band text
- OutOfBandText text =
- mTextOutOfBandVector.itemAt(index - mTextTrackVector.size());
-
- mOutOfBandSource = text.source;
- TimedTextParser::FileType fileType = text.type;
-
- if (mTextParser == NULL) {
- mTextParser = new TimedTextParser();
- }
-
- if ((err = mTextParser->init(mOutOfBandSource, fileType)) != OK) {
- return err;
- }
- mTextType = kOutOfBandText;
- }
-
- // send sample description format
- if ((err = extractAndSendGlobalDescriptions()) != OK) {
- return err;
- }
-
- int64_t positionUs;
- mObserver->getPosition(&positionUs);
- seekTo(positionUs);
-
- postTextEvent();
-
- mStarted = true;
-
- return OK;
+void TimedTextPlayer::start() {
+ sp<AMessage> msg = new AMessage(kWhatSeek, id());
+ msg->setInt64("seekTimeUs", -1);
+ msg->post();
}
void TimedTextPlayer::pause() {
- CHECK(mStarted);
-
- cancelTextEvent();
+ (new AMessage(kWhatPause, id()))->post();
}
-void TimedTextPlayer::resume() {
- CHECK(mStarted);
-
- postTextEvent();
+void TimedTextPlayer::seekToAsync(int64_t timeUs) {
+ sp<AMessage> msg = new AMessage(kWhatSeek, id());
+ msg->setInt64("seekTimeUs", timeUs);
+ msg->post();
}
-void TimedTextPlayer::reset() {
- CHECK(mStarted);
+void TimedTextPlayer::setDataSource(sp<TimedTextSource> source) {
+ sp<AMessage> msg = new AMessage(kWhatSetSource, id());
+ msg->setObject("source", source);
+ msg->post();
+}
- // send an empty text to clear the screen
- notifyListener(MEDIA_TIMED_TEXT);
-
- cancelTextEvent();
-
- mSeeking = false;
- mStarted = false;
-
- if (mTextType == kInBandText) {
- if (mTextBuffer != NULL) {
- mTextBuffer->release();
- mTextBuffer = NULL;
+void TimedTextPlayer::onMessageReceived(const sp<AMessage> &msg) {
+ switch (msg->what()) {
+ case kWhatPause: {
+ mSendSubtitleGeneration++;
+ break;
}
-
- if (mSource != NULL) {
- mSource->stop();
- mSource.clear();
- mSource = NULL;
+ case kWhatSeek: {
+ int64_t seekTimeUs = 0;
+ msg->findInt64("seekTimeUs", &seekTimeUs);
+ if (seekTimeUs < 0) {
+ sp<MediaPlayerBase> listener = mListener.promote();
+ if (listener != NULL) {
+ int32_t positionMs = 0;
+ listener->getCurrentPosition(&positionMs);
+ seekTimeUs = positionMs * 1000ll;
+ }
+ }
+ doSeekAndRead(seekTimeUs);
+ break;
}
- } else {
- if (mTextParser != NULL) {
- mTextParser.clear();
- mTextParser = NULL;
+ case kWhatSendSubtitle: {
+ int32_t generation;
+ CHECK(msg->findInt32("generation", &generation));
+ if (generation != mSendSubtitleGeneration) {
+ // Drop obsolete msg.
+ break;
+ }
+ sp<RefBase> obj;
+ msg->findObject("subtitle", &obj);
+ if (obj != NULL) {
+ sp<ParcelEvent> parcelEvent;
+ parcelEvent = static_cast<ParcelEvent*>(obj.get());
+ notifyListener(&(parcelEvent->parcel));
+ } else {
+ notifyListener();
+ }
+ doRead();
+ break;
}
- if (mOutOfBandSource != NULL) {
- mOutOfBandSource.clear();
- mOutOfBandSource = NULL;
+ case kWhatSetSource: {
+ sp<RefBase> obj;
+ msg->findObject("source", &obj);
+ if (obj == NULL) break;
+ if (mSource != NULL) {
+ mSource->stop();
+ }
+ mSource = static_cast<TimedTextSource*>(obj.get());
+ status_t err = mSource->start();
+ if (err != OK) {
+ notifyError(err);
+ break;
+ }
+ Parcel parcel;
+ err = mSource->extractGlobalDescriptions(&parcel);
+ if (err != OK) {
+ notifyError(err);
+ break;
+ }
+ notifyListener(&parcel);
+ break;
}
}
}
-status_t TimedTextPlayer::seekTo(int64_t time_us) {
- Mutex::Autolock autoLock(mLock);
-
- mSeeking = true;
- mSeekTimeUs = time_us;
-
- postTextEvent();
-
- return OK;
-}
-
-status_t TimedTextPlayer::setTimedTextTrackIndex(int32_t index) {
- if (index >=
- (int)(mTextTrackVector.size() + mTextOutOfBandVector.size())) {
- return BAD_VALUE;
- }
-
- if (mStarted) {
- reset();
- }
-
- if (index >= 0) {
- return start(index);
- }
- return OK;
-}
-
-void TimedTextPlayer::onTextEvent() {
- Mutex::Autolock autoLock(mLock);
-
- if (!mTextEventPending) {
- return;
- }
- mTextEventPending = false;
-
- if (mData.dataSize() > 0) {
- notifyListener(MEDIA_TIMED_TEXT, &mData);
- mData.freeData();
- }
-
+void TimedTextPlayer::doSeekAndRead(int64_t seekTimeUs) {
MediaSource::ReadOptions options;
- if (mSeeking) {
- options.setSeekTo(mSeekTimeUs,
- MediaSource::ReadOptions::SEEK_PREVIOUS_SYNC);
- mSeeking = false;
-
- notifyListener(MEDIA_TIMED_TEXT); //empty text to clear the screen
- }
-
- int64_t positionUs, timeUs;
- mObserver->getPosition(&positionUs);
-
- if (mTextType == kInBandText) {
- if (mSource->read(&mTextBuffer, &options) != OK) {
- return;
- }
-
- mTextBuffer->meta_data()->findInt64(kKeyTime, &timeUs);
- } else {
- int64_t endTimeUs;
- if (mTextParser->getText(
- &mText, &timeUs, &endTimeUs, &options) != OK) {
- return;
- }
- }
-
- if (timeUs > 0) {
- extractAndAppendLocalDescriptions(timeUs);
- }
-
- if (mTextType == kInBandText) {
- if (mTextBuffer != NULL) {
- mTextBuffer->release();
- mTextBuffer = NULL;
- }
- } else {
- mText.clear();
- }
-
- //send the text now
- if (timeUs <= positionUs + 100000ll) {
- postTextEvent();
- } else {
- postTextEvent(timeUs - positionUs - 100000ll);
- }
+ options.setSeekTo(seekTimeUs, MediaSource::ReadOptions::SEEK_PREVIOUS_SYNC);
+ doRead(&options);
}
-void TimedTextPlayer::postTextEvent(int64_t delayUs) {
- if (mTextEventPending) {
+void TimedTextPlayer::doRead(MediaSource::ReadOptions* options) {
+ int64_t startTimeUs = 0;
+ int64_t endTimeUs = 0;
+ sp<ParcelEvent> parcelEvent = new ParcelEvent();
+ status_t err = mSource->read(&startTimeUs, &endTimeUs,
+ &(parcelEvent->parcel), options);
+ if (err == WOULD_BLOCK) {
+ postTextEventDelayUs(NULL, kWaitTimeUsToRetryRead);
+ return;
+ } else if (err != OK) {
+ notifyError(err);
return;
}
- mTextEventPending = true;
- mQueue->postEventWithDelay(mTextEvent, delayUs < 0 ? 10000 : delayUs);
+ postTextEvent(parcelEvent, startTimeUs);
+ if (endTimeUs > 0) {
+ CHECK_GE(endTimeUs, startTimeUs);
+ // send an empty timed text to clear the subtitle when it reaches to the
+ // end time.
+ postTextEvent(NULL, endTimeUs);
+ }
}
-void TimedTextPlayer::cancelTextEvent() {
- mQueue->cancelEvent(mTextEvent->eventID());
- mTextEventPending = false;
-}
+void TimedTextPlayer::postTextEvent(const sp<ParcelEvent>& parcel, int64_t timeUs) {
+ sp<MediaPlayerBase> listener = mListener.promote();
+ if (listener != NULL) {
+ int64_t positionUs, delayUs;
+ int32_t positionMs = 0;
+ listener->getCurrentPosition(&positionMs);
+ positionUs = positionMs * 1000ll;
-void TimedTextPlayer::addTextSource(sp<MediaSource> source) {
- Mutex::Autolock autoLock(mLock);
- mTextTrackVector.add(source);
-}
-
-status_t TimedTextPlayer::setParameter(int key, const Parcel &request) {
- Mutex::Autolock autoLock(mLock);
-
- if (key == KEY_PARAMETER_TIMED_TEXT_ADD_OUT_OF_BAND_SOURCE) {
- const String16 uri16 = request.readString16();
- String8 uri = String8(uri16);
- KeyedVector<String8, String8> headers;
-
- // To support local subtitle file only for now
- if (strncasecmp("file://", uri.string(), 7)) {
- return INVALID_OPERATION;
- }
- sp<DataSource> dataSource =
- DataSource::CreateFromURI(uri, &headers);
- status_t err = dataSource->initCheck();
-
- if (err != OK) {
- return err;
- }
-
- OutOfBandText text;
- text.source = dataSource;
- if (uri.getPathExtension() == String8(".srt")) {
- text.type = TimedTextParser::OUT_OF_BAND_FILE_SRT;
+ if (timeUs <= positionUs + kAdjustmentProcessingTimeUs) {
+ delayUs = 0;
} else {
- return ERROR_UNSUPPORTED;
+ delayUs = timeUs - positionUs - kAdjustmentProcessingTimeUs;
}
-
- mTextOutOfBandVector.add(text);
-
- return OK;
+ postTextEventDelayUs(parcel, delayUs);
}
- return INVALID_OPERATION;
}
-void TimedTextPlayer::notifyListener(int msg, const Parcel *parcel) {
- if (mListener != NULL) {
- sp<MediaPlayerBase> listener = mListener.promote();
+void TimedTextPlayer::postTextEventDelayUs(const sp<ParcelEvent>& parcel, int64_t delayUs) {
+ sp<MediaPlayerBase> listener = mListener.promote();
+ if (listener != NULL) {
+ sp<AMessage> msg = new AMessage(kWhatSendSubtitle, id());
+ msg->setInt32("generation", mSendSubtitleGeneration);
+ if (parcel != NULL) {
+ msg->setObject("subtitle", parcel);
+ }
+ msg->post(delayUs);
+ }
+}
- if (listener != NULL) {
- if (parcel && (parcel->dataSize() > 0)) {
- listener->sendEvent(msg, 0, 0, parcel);
- } else { // send an empty timed text to clear the screen
- listener->sendEvent(msg);
- }
+void TimedTextPlayer::notifyError(int error) {
+ sp<MediaPlayerBase> listener = mListener.promote();
+ if (listener != NULL) {
+ listener->sendEvent(MEDIA_INFO, MEDIA_INFO_TIMED_TEXT_ERROR, error);
+ }
+}
+
+void TimedTextPlayer::notifyListener(const Parcel *parcel) {
+ sp<MediaPlayerBase> listener = mListener.promote();
+ if (listener != NULL) {
+ if (parcel != NULL && (parcel->dataSize() > 0)) {
+ listener->sendEvent(MEDIA_TIMED_TEXT, 0, 0, parcel);
+ } else { // send an empty timed text to clear the screen
+ listener->sendEvent(MEDIA_TIMED_TEXT);
}
}
}
-// Each text sample consists of a string of text, optionally with sample
-// modifier description. The modifier description could specify a new
-// text style for the string of text. These descriptions are present only
-// if they are needed. This method is used to extract the modifier
-// description and append it at the end of the text.
-status_t TimedTextPlayer::extractAndAppendLocalDescriptions(int64_t timeUs) {
- const void *data;
- size_t size = 0;
- int32_t flag = TextDescriptions::LOCAL_DESCRIPTIONS;
-
- if (mTextType == kInBandText) {
- const char *mime;
- CHECK(mSource->getFormat()->findCString(kKeyMIMEType, &mime));
-
- if (!strcasecmp(mime, MEDIA_MIMETYPE_TEXT_3GPP)) {
- flag |= TextDescriptions::IN_BAND_TEXT_3GPP;
- data = mTextBuffer->data();
- size = mTextBuffer->size();
- } else {
- // support 3GPP only for now
- return ERROR_UNSUPPORTED;
- }
- } else {
- data = mText.c_str();
- size = mText.size();
- flag |= TextDescriptions::OUT_OF_BAND_TEXT_SRT;
- }
-
- if ((size > 0) && (flag != TextDescriptions::LOCAL_DESCRIPTIONS)) {
- mData.freeData();
- return TextDescriptions::getParcelOfDescriptions(
- (const uint8_t *)data, size, flag, timeUs / 1000, &mData);
- }
-
- return OK;
-}
-
-// To extract and send the global text descriptions for all the text samples
-// in the text track or text file.
-status_t TimedTextPlayer::extractAndSendGlobalDescriptions() {
- const void *data;
- size_t size = 0;
- int32_t flag = TextDescriptions::GLOBAL_DESCRIPTIONS;
-
- if (mTextType == kInBandText) {
- const char *mime;
- CHECK(mSource->getFormat()->findCString(kKeyMIMEType, &mime));
-
- // support 3GPP only for now
- if (!strcasecmp(mime, MEDIA_MIMETYPE_TEXT_3GPP)) {
- uint32_t type;
- // get the 'tx3g' box content. This box contains the text descriptions
- // used to render the text track
- if (!mSource->getFormat()->findData(
- kKeyTextFormatData, &type, &data, &size)) {
- return ERROR_MALFORMED;
- }
-
- flag |= TextDescriptions::IN_BAND_TEXT_3GPP;
- }
- }
-
- if ((size > 0) && (flag != TextDescriptions::GLOBAL_DESCRIPTIONS)) {
- Parcel parcel;
- if (TextDescriptions::getParcelOfDescriptions(
- (const uint8_t *)data, size, flag, 0, &parcel) == OK) {
- if (parcel.dataSize() > 0) {
- notifyListener(MEDIA_TIMED_TEXT, &parcel);
- }
- }
- }
-
- return OK;
-}
-}
+} // namespace android
diff --git a/media/libstagefright/timedtext/TimedTextPlayer.h b/media/libstagefright/timedtext/TimedTextPlayer.h
index a744db5..47aff03 100644
--- a/media/libstagefright/timedtext/TimedTextPlayer.h
+++ b/media/libstagefright/timedtext/TimedTextPlayer.h
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2011 The Android Open Source Project
+ * Copyright (C) 2012 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
@@ -15,98 +15,61 @@
*/
#ifndef TIMEDTEXT_PLAYER_H_
-
#define TIMEDTEXT_PLAYER_H_
-#include <media/MediaPlayerInterface.h>
+#include <binder/Parcel.h>
#include <media/stagefright/foundation/ABase.h>
-#include <media/stagefright/foundation/AString.h>
+#include <media/stagefright/foundation/AHandler.h>
+#include <media/stagefright/MediaSource.h>
+#include <utils/RefBase.h>
-#include "include/TimedEventQueue.h"
-#include "TimedTextParser.h"
+#include "TimedTextSource.h"
namespace android {
-class MediaSource;
-class AwesomePlayer;
-class MediaBuffer;
+class AMessage;
+class MediaPlayerBase;
+class TimedTextDriver;
+class TimedTextSource;
-class TimedTextPlayer {
+class TimedTextPlayer : public AHandler {
public:
- TimedTextPlayer(AwesomePlayer *observer,
- const wp<MediaPlayerBase> &listener,
- TimedEventQueue *queue);
+ TimedTextPlayer(const wp<MediaPlayerBase> &listener);
virtual ~TimedTextPlayer();
- // index: the index of the text track which will
- // be turned on
- status_t start(uint8_t index);
-
+ void start();
void pause();
+ void seekToAsync(int64_t timeUs);
+ void setDataSource(sp<TimedTextSource> source);
- void resume();
-
- status_t seekTo(int64_t time_us);
-
- void addTextSource(sp<MediaSource> source);
-
- status_t setTimedTextTrackIndex(int32_t index);
- status_t setParameter(int key, const Parcel &request);
+protected:
+ virtual void onMessageReceived(const sp<AMessage> &msg);
private:
- enum TextType {
- kNoText = 0,
- kInBandText = 1,
- kOutOfBandText = 2,
+ enum {
+ kWhatPause = 'paus',
+ kWhatSeek = 'seek',
+ kWhatSendSubtitle = 'send',
+ kWhatSetSource = 'ssrc',
};
- Mutex mLock;
-
- sp<MediaSource> mSource;
- sp<DataSource> mOutOfBandSource;
-
- bool mSeeking;
- int64_t mSeekTimeUs;
-
- bool mStarted;
-
- sp<TimedEventQueue::Event> mTextEvent;
- bool mTextEventPending;
-
- TimedEventQueue *mQueue;
+ // To add Parcel into an AMessage as an object, it should be 'RefBase'.
+ struct ParcelEvent : public RefBase {
+ Parcel parcel;
+ };
wp<MediaPlayerBase> mListener;
- AwesomePlayer *mObserver;
+ sp<TimedTextSource> mSource;
+ int32_t mSendSubtitleGeneration;
- MediaBuffer *mTextBuffer;
- Parcel mData;
-
- // for in-band timed text
- Vector<sp<MediaSource> > mTextTrackVector;
-
- // for out-of-band timed text
- struct OutOfBandText {
- TimedTextParser::FileType type;
- sp<DataSource> source;
- };
- Vector<OutOfBandText > mTextOutOfBandVector;
-
- sp<TimedTextParser> mTextParser;
- AString mText;
-
- TextType mTextType;
-
- void reset();
-
+ void doSeekAndRead(int64_t seekTimeUs);
+ void doRead(MediaSource::ReadOptions* options = NULL);
void onTextEvent();
- void postTextEvent(int64_t delayUs = -1);
- void cancelTextEvent();
-
- void notifyListener(int msg, const Parcel *parcel = NULL);
-
- status_t extractAndAppendLocalDescriptions(int64_t timeUs);
- status_t extractAndSendGlobalDescriptions();
+ void postTextEvent(const sp<ParcelEvent>& parcel = NULL, int64_t timeUs = -1);
+ void postTextEventDelayUs(const sp<ParcelEvent>& parcel = NULL, int64_t delayUs = -1);
+ void notifyError(int error = 0);
+ void notifyListener(const Parcel *parcel = NULL);
DISALLOW_EVIL_CONSTRUCTORS(TimedTextPlayer);
};
diff --git a/media/libstagefright/timedtext/TimedTextSRTSource.cpp b/media/libstagefright/timedtext/TimedTextSRTSource.cpp
new file mode 100644
index 0000000..7b1f7f6
--- /dev/null
+++ b/media/libstagefright/timedtext/TimedTextSRTSource.cpp
@@ -0,0 +1,286 @@
+ /*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "TimedTextSRTSource"
+#include <utils/Log.h>
+
+#include <binder/Parcel.h>
+#include <media/stagefright/foundation/ADebug.h> // for CHECK_xx
+#include <media/stagefright/foundation/AString.h>
+#include <media/stagefright/DataSource.h>
+#include <media/stagefright/MediaDefs.h> // for MEDIA_MIMETYPE_xxx
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaSource.h>
+#include <media/stagefright/MetaData.h>
+
+#include "TimedTextSRTSource.h"
+#include "TextDescriptions.h"
+
+namespace android {
+
+TimedTextSRTSource::TimedTextSRTSource(const sp<DataSource>& dataSource)
+ : mSource(dataSource),
+ mMetaData(new MetaData),
+ mIndex(0) {
+}
+
+TimedTextSRTSource::~TimedTextSRTSource() {
+}
+
+status_t TimedTextSRTSource::start() {
+ status_t err = scanFile();
+ if (err != OK) {
+ reset();
+ }
+ // TODO: Need to detect the language, because SRT doesn't give language
+ // information explicitly.
+ mMetaData->setCString(kKeyMediaLanguage, "");
+ return err;
+}
+
+void TimedTextSRTSource::reset() {
+ mMetaData->clear();
+ mTextVector.clear();
+ mIndex = 0;
+}
+
+status_t TimedTextSRTSource::stop() {
+ reset();
+ return OK;
+}
+
+status_t TimedTextSRTSource::read(
+ int64_t *startTimeUs,
+ int64_t *endTimeUs,
+ Parcel *parcel,
+ const MediaSource::ReadOptions *options) {
+ AString text;
+ status_t err = getText(options, &text, startTimeUs, endTimeUs);
+ if (err != OK) {
+ return err;
+ }
+
+ CHECK_GE(*startTimeUs, 0);
+ extractAndAppendLocalDescriptions(*startTimeUs, text, parcel);
+ return OK;
+}
+
+status_t TimedTextSRTSource::scanFile() {
+ off64_t offset = 0;
+ int64_t startTimeUs;
+ bool endOfFile = false;
+
+ while (!endOfFile) {
+ TextInfo info;
+ status_t err = getNextSubtitleInfo(&offset, &startTimeUs, &info);
+ switch (err) {
+ case OK:
+ mTextVector.add(startTimeUs, info);
+ break;
+ case ERROR_END_OF_STREAM:
+ endOfFile = true;
+ break;
+ default:
+ return err;
+ }
+ }
+ if (mTextVector.isEmpty()) {
+ return ERROR_MALFORMED;
+ }
+ return OK;
+}
+
+/* SRT format:
+ * Subtitle number
+ * Start time --> End time
+ * Text of subtitle (one or more lines)
+ * Blank lines
+ *
+ * .srt file example:
+ * 1
+ * 00:00:20,000 --> 00:00:24,400
+ * Altocumulus clouds occr between six thousand
+ *
+ * 2
+ * 00:00:24,600 --> 00:00:27,800
+ * and twenty thousand feet above ground level.
+ */
+status_t TimedTextSRTSource::getNextSubtitleInfo(
+ off64_t *offset, int64_t *startTimeUs, TextInfo *info) {
+ AString data;
+ status_t err;
+
+ // To skip blank lines.
+ do {
+ if ((err = readNextLine(offset, &data)) != OK) {
+ return err;
+ }
+ data.trim();
+ } while (data.empty());
+
+ // Just ignore the first non-blank line which is subtitle sequence number.
+ if ((err = readNextLine(offset, &data)) != OK) {
+ return err;
+ }
+ int hour1, hour2, min1, min2, sec1, sec2, msec1, msec2;
+ // the start time format is: hours:minutes:seconds,milliseconds
+ // 00:00:24,600 --> 00:00:27,800
+ if (sscanf(data.c_str(), "%02d:%02d:%02d,%03d --> %02d:%02d:%02d,%03d",
+ &hour1, &min1, &sec1, &msec1, &hour2, &min2, &sec2, &msec2) != 8) {
+ return ERROR_MALFORMED;
+ }
+
+ *startTimeUs = ((hour1 * 3600 + min1 * 60 + sec1) * 1000 + msec1) * 1000ll;
+ info->endTimeUs = ((hour2 * 3600 + min2 * 60 + sec2) * 1000 + msec2) * 1000ll;
+ if (info->endTimeUs <= *startTimeUs) {
+ return ERROR_MALFORMED;
+ }
+
+ info->offset = *offset;
+ bool needMoreData = true;
+ while (needMoreData) {
+ if ((err = readNextLine(offset, &data)) != OK) {
+ if (err == ERROR_END_OF_STREAM) {
+ needMoreData = false;
+ } else {
+ return err;
+ }
+ }
+
+ if (needMoreData) {
+ data.trim();
+ if (data.empty()) {
+ // it's an empty line used to separate two subtitles
+ needMoreData = false;
+ }
+ }
+ }
+ info->textLen = *offset - info->offset;
+ return OK;
+}
+
+status_t TimedTextSRTSource::readNextLine(off64_t *offset, AString *data) {
+ data->clear();
+ while (true) {
+ ssize_t readSize;
+ char character;
+ if ((readSize = mSource->readAt(*offset, &character, 1)) < 1) {
+ if (readSize == 0) {
+ return ERROR_END_OF_STREAM;
+ }
+ return ERROR_IO;
+ }
+
+ (*offset)++;
+
+ // a line could end with CR, LF or CR + LF
+ if (character == 10) {
+ break;
+ } else if (character == 13) {
+ if ((readSize = mSource->readAt(*offset, &character, 1)) < 1) {
+ if (readSize == 0) { // end of the stream
+ return OK;
+ }
+ return ERROR_IO;
+ }
+
+ (*offset)++;
+ if (character != 10) {
+ (*offset)--;
+ }
+ break;
+ }
+ data->append(character);
+ }
+ return OK;
+}
+
+status_t TimedTextSRTSource::getText(
+ const MediaSource::ReadOptions *options,
+ AString *text, int64_t *startTimeUs, int64_t *endTimeUs) {
+ text->clear();
+ int64_t seekTimeUs;
+ MediaSource::ReadOptions::SeekMode mode;
+ if (options != NULL && options->getSeekTo(&seekTimeUs, &mode)) {
+ int64_t lastEndTimeUs =
+ mTextVector.valueAt(mTextVector.size() - 1).endTimeUs;
+ int64_t firstStartTimeUs = mTextVector.keyAt(0);
+ if (seekTimeUs < 0 || seekTimeUs > lastEndTimeUs) {
+ return ERROR_OUT_OF_RANGE;
+ } else if (seekTimeUs < firstStartTimeUs) {
+ mIndex = 0;
+ } else {
+ // binary search
+ ssize_t low = 0;
+ ssize_t high = mTextVector.size() - 1;
+ ssize_t mid = 0;
+ int64_t currTimeUs;
+
+ while (low <= high) {
+ mid = low + (high - low)/2;
+ currTimeUs = mTextVector.keyAt(mid);
+ const int diff = currTimeUs - seekTimeUs;
+
+ if (diff == 0) {
+ break;
+ } else if (diff < 0) {
+ low = mid + 1;
+ } else {
+ if ((high == mid + 1)
+ && (seekTimeUs < mTextVector.keyAt(high))) {
+ break;
+ }
+ high = mid - 1;
+ }
+ }
+ mIndex = mid;
+ }
+ }
+ const TextInfo &info = mTextVector.valueAt(mIndex);
+ *startTimeUs = mTextVector.keyAt(mIndex);
+ *endTimeUs = info.endTimeUs;
+ mIndex++;
+
+ char *str = new char[info.textLen];
+ if (mSource->readAt(info.offset, str, info.textLen) < info.textLen) {
+ delete[] str;
+ return ERROR_IO;
+ }
+ text->append(str, info.textLen);
+ delete[] str;
+ return OK;
+}
+
+status_t TimedTextSRTSource::extractAndAppendLocalDescriptions(
+ int64_t timeUs, const AString &text, Parcel *parcel) {
+ const void *data = text.c_str();
+ size_t size = text.size();
+ int32_t flag = TextDescriptions::LOCAL_DESCRIPTIONS |
+ TextDescriptions::OUT_OF_BAND_TEXT_SRT;
+
+ if (size > 0) {
+ return TextDescriptions::getParcelOfDescriptions(
+ (const uint8_t *)data, size, flag, timeUs / 1000, parcel);
+ }
+ return OK;
+}
+
+sp<MetaData> TimedTextSRTSource::getFormat() {
+ return mMetaData;
+}
+
+} // namespace android
diff --git a/media/libstagefright/timedtext/TimedTextSRTSource.h b/media/libstagefright/timedtext/TimedTextSRTSource.h
new file mode 100644
index 0000000..e1371b8
--- /dev/null
+++ b/media/libstagefright/timedtext/TimedTextSRTSource.h
@@ -0,0 +1,78 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef TIMED_TEXT_SRT_SOURCE_H_
+#define TIMED_TEXT_SRT_SOURCE_H_
+
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaSource.h>
+#include <utils/Compat.h> // off64_t
+
+#include "TimedTextSource.h"
+
+namespace android {
+
+class AString;
+class DataSource;
+class MediaBuffer;
+class Parcel;
+
+class TimedTextSRTSource : public TimedTextSource {
+public:
+ TimedTextSRTSource(const sp<DataSource>& dataSource);
+ virtual status_t start();
+ virtual status_t stop();
+ virtual status_t read(
+ int64_t *startTimeUs,
+ int64_t *endTimeUs,
+ Parcel *parcel,
+ const MediaSource::ReadOptions *options = NULL);
+ virtual sp<MetaData> getFormat();
+
+protected:
+ virtual ~TimedTextSRTSource();
+
+private:
+ sp<DataSource> mSource;
+ sp<MetaData> mMetaData;
+
+ struct TextInfo {
+ int64_t endTimeUs;
+ // The offset of the text in the original file.
+ off64_t offset;
+ int textLen;
+ };
+
+ int mIndex;
+ KeyedVector<int64_t, TextInfo> mTextVector;
+
+ void reset();
+ status_t scanFile();
+ status_t getNextSubtitleInfo(
+ off64_t *offset, int64_t *startTimeUs, TextInfo *info);
+ status_t readNextLine(off64_t *offset, AString *data);
+ status_t getText(
+ const MediaSource::ReadOptions *options,
+ AString *text, int64_t *startTimeUs, int64_t *endTimeUs);
+ status_t extractAndAppendLocalDescriptions(
+ int64_t timeUs, const AString &text, Parcel *parcel);
+
+ DISALLOW_EVIL_CONSTRUCTORS(TimedTextSRTSource);
+};
+
+} // namespace android
+
+#endif // TIMED_TEXT_SRT_SOURCE_H_
diff --git a/media/libstagefright/timedtext/TimedTextSource.cpp b/media/libstagefright/timedtext/TimedTextSource.cpp
new file mode 100644
index 0000000..953f7b5
--- /dev/null
+++ b/media/libstagefright/timedtext/TimedTextSource.cpp
@@ -0,0 +1,66 @@
+ /*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "TimedTextSource"
+#include <utils/Log.h>
+
+#include <media/stagefright/foundation/ADebug.h> // CHECK_XX macro
+#include <media/stagefright/DataSource.h>
+#include <media/stagefright/MediaDefs.h> // for MEDIA_MIMETYPE_xxx
+#include <media/stagefright/MediaSource.h>
+#include <media/stagefright/MetaData.h>
+
+#include "TimedTextSource.h"
+
+#include "TimedText3GPPSource.h"
+#include "TimedTextSRTSource.h"
+
+namespace android {
+
+// static
+sp<TimedTextSource> TimedTextSource::CreateTimedTextSource(
+ const sp<MediaSource>& mediaSource) {
+ const char *mime;
+ CHECK(mediaSource->getFormat()->findCString(kKeyMIMEType, &mime));
+ if (strcasecmp(mime, MEDIA_MIMETYPE_TEXT_3GPP) == 0) {
+ return new TimedText3GPPSource(mediaSource);
+ }
+ ALOGE("Unsupported mime type for subtitle. : %s", mime);
+ return NULL;
+}
+
+// static
+sp<TimedTextSource> TimedTextSource::CreateTimedTextSource(
+ const sp<DataSource>& dataSource, FileType filetype) {
+ switch(filetype) {
+ case OUT_OF_BAND_FILE_SRT:
+ return new TimedTextSRTSource(dataSource);
+ case OUT_OF_BAND_FILE_SMI:
+ // TODO: Implement for SMI.
+ ALOGE("Supporting SMI is not implemented yet");
+ break;
+ default:
+ ALOGE("Undefined subtitle format. : %d", filetype);
+ }
+ return NULL;
+}
+
+sp<MetaData> TimedTextSource::getFormat() {
+ return NULL;
+}
+
+} // namespace android
diff --git a/media/libstagefright/timedtext/TimedTextSource.h b/media/libstagefright/timedtext/TimedTextSource.h
new file mode 100644
index 0000000..756cc31
--- /dev/null
+++ b/media/libstagefright/timedtext/TimedTextSource.h
@@ -0,0 +1,64 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef TIMED_TEXT_SOURCE_H_
+#define TIMED_TEXT_SOURCE_H_
+
+#include <media/stagefright/foundation/ABase.h> // for DISALLOW_XXX macro.
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaSource.h> // for MediaSource::ReadOptions
+#include <utils/RefBase.h>
+
+namespace android {
+
+class DataSource;
+class MetaData;
+class Parcel;
+
+class TimedTextSource : public RefBase {
+ public:
+ enum FileType {
+ OUT_OF_BAND_FILE_SRT = 1,
+ OUT_OF_BAND_FILE_SMI = 2,
+ };
+ static sp<TimedTextSource> CreateTimedTextSource(
+ const sp<MediaSource>& source);
+ static sp<TimedTextSource> CreateTimedTextSource(
+ const sp<DataSource>& source, FileType filetype);
+ TimedTextSource() {}
+ virtual status_t start() = 0;
+ virtual status_t stop() = 0;
+ // Returns subtitle parcel and its start time.
+ virtual status_t read(
+ int64_t *startTimeUs,
+ int64_t *endTimeUs,
+ Parcel *parcel,
+ const MediaSource::ReadOptions *options = NULL) = 0;
+ virtual status_t extractGlobalDescriptions(Parcel *parcel) {
+ return INVALID_OPERATION;
+ }
+ virtual sp<MetaData> getFormat();
+
+ protected:
+ virtual ~TimedTextSource() { }
+
+ private:
+ DISALLOW_EVIL_CONSTRUCTORS(TimedTextSource);
+};
+
+} // namespace android
+
+#endif // TIMED_TEXT_SOURCE_H_
diff --git a/media/libstagefright/yuv/YUVCanvas.cpp b/media/libstagefright/yuv/YUVCanvas.cpp
index 38aa779..4c9fee8 100644
--- a/media/libstagefright/yuv/YUVCanvas.cpp
+++ b/media/libstagefright/yuv/YUVCanvas.cpp
@@ -17,7 +17,7 @@
#define LOG_NDEBUG 0
#define LOG_TAG "YUVCanvas"
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/YUVCanvas.h>
#include <media/stagefright/YUVImage.h>
#include <ui/Rect.h>
diff --git a/media/libstagefright/yuv/YUVImage.cpp b/media/libstagefright/yuv/YUVImage.cpp
index 0d67c96..7b9000b 100644
--- a/media/libstagefright/yuv/YUVImage.cpp
+++ b/media/libstagefright/yuv/YUVImage.cpp
@@ -17,9 +17,9 @@
#define LOG_NDEBUG 0
#define LOG_TAG "YUVImage"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/YUVImage.h>
#include <ui/Rect.h>
-#include <media/stagefright/MediaDebug.h>
namespace android {
diff --git a/media/mediaserver/Android.mk b/media/mediaserver/Android.mk
index 0559812..4e9b4cf 100644
--- a/media/mediaserver/Android.mk
+++ b/media/mediaserver/Android.mk
@@ -11,12 +11,12 @@
libutils \
libbinder
-base := $(LOCAL_PATH)/../..
-
+# FIXME The duplicate audioflinger is temporary
LOCAL_C_INCLUDES := \
- $(base)/services/audioflinger \
- $(base)/services/camera/libcameraservice \
- $(base)/media/libmediaplayerservice
+ frameworks/native/services/audioflinger \
+ frameworks/base/services/audioflinger \
+ frameworks/base/services/camera/libcameraservice \
+ frameworks/base/media/libmediaplayerservice
LOCAL_MODULE:= mediaserver
diff --git a/media/mediaserver/main_mediaserver.cpp b/media/mediaserver/main_mediaserver.cpp
index f078192..6b1abb1 100644
--- a/media/mediaserver/main_mediaserver.cpp
+++ b/media/mediaserver/main_mediaserver.cpp
@@ -15,21 +15,19 @@
** limitations under the License.
*/
-// System headers required for setgroups, etc.
-#include <sys/types.h>
-#include <unistd.h>
-#include <grp.h>
+#define LOG_TAG "mediaserver"
+//#define LOG_NDEBUG 0
#include <binder/IPCThreadState.h>
#include <binder/ProcessState.h>
#include <binder/IServiceManager.h>
#include <utils/Log.h>
-#include <AudioFlinger.h>
-#include <CameraService.h>
-#include <MediaPlayerService.h>
-#include <AudioPolicyService.h>
-#include <private/android_filesystem_config.h>
+// from LOCAL_C_INCLUDES
+#include "AudioFlinger.h"
+#include "CameraService.h"
+#include "MediaPlayerService.h"
+#include "AudioPolicyService.h"
using namespace android;
diff --git a/media/mtp/Android.mk b/media/mtp/Android.mk
index e590bab..fc7fc4f 100644
--- a/media/mtp/Android.mk
+++ b/media/mtp/Android.mk
@@ -39,6 +39,9 @@
LOCAL_CFLAGS := -DMTP_DEVICE -DMTP_HOST
+# Needed for <bionic_time.h>
+LOCAL_C_INCLUDES := bionic/libc/private
+
LOCAL_SHARED_LIBRARIES := libutils libcutils libusbhost libbinder
include $(BUILD_SHARED_LIBRARY)
diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk
index fa49592..257f62c 100644
--- a/services/audioflinger/Android.mk
+++ b/services/audioflinger/Android.mk
@@ -6,18 +6,24 @@
AudioFlinger.cpp \
AudioMixer.cpp.arm \
AudioResampler.cpp.arm \
- AudioResamplerSinc.cpp.arm \
- AudioResamplerCubic.cpp.arm \
- AudioPolicyService.cpp
+ AudioPolicyService.cpp \
+ ServiceUtilities.cpp
+# AudioResamplerSinc.cpp.arm
+# AudioResamplerCubic.cpp.arm
LOCAL_C_INCLUDES := \
- system/media/audio_effects/include
+ $(call include-path-for, audio-effects) \
+ $(call include-path-for, audio-utils)
+# FIXME keep libmedia_native but remove libmedia after split
LOCAL_SHARED_LIBRARIES := \
+ libaudioutils \
+ libcommon_time_client \
libcutils \
libutils \
libbinder \
libmedia \
+ libmedia_native \
libhardware \
libhardware_legacy \
libeffects \
diff --git a/services/audioflinger/AudioBufferProvider.h b/services/audioflinger/AudioBufferProvider.h
index 81c5c39..43e4de7 100644
--- a/services/audioflinger/AudioBufferProvider.h
+++ b/services/audioflinger/AudioBufferProvider.h
@@ -17,8 +17,6 @@
#ifndef ANDROID_AUDIO_BUFFER_PROVIDER_H
#define ANDROID_AUDIO_BUFFER_PROVIDER_H
-#include <stdint.h>
-#include <sys/types.h>
#include <utils/Errors.h>
namespace android {
@@ -29,6 +27,7 @@
public:
struct Buffer {
+ Buffer() : raw(NULL), frameCount(0) { }
union {
void* raw;
short* i16;
@@ -38,8 +37,15 @@
};
virtual ~AudioBufferProvider() {}
-
- virtual status_t getNextBuffer(Buffer* buffer) = 0;
+
+ // value representing an invalid presentation timestamp
+ static const int64_t kInvalidPTS = 0x7FFFFFFFFFFFFFFFLL; // <stdint.h> is too painful
+
+ // pts is the local time when the next sample yielded by getNextBuffer
+ // will be rendered.
+ // Pass kInvalidPTS if the PTS is unknown or not applicable.
+ virtual status_t getNextBuffer(Buffer* buffer, int64_t pts = kInvalidPTS) = 0;
+
virtual void releaseBuffer(Buffer* buffer) = 0;
};
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index dab76a3..c0b18fa 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1,4 +1,4 @@
-/* //device/include/server/AudioFlinger/AudioFlinger.cpp
+/*
**
** Copyright 2007, The Android Open Source Project
**
@@ -35,10 +35,14 @@
#include <cutils/bitops.h>
#include <cutils/properties.h>
+#include <cutils/compiler.h>
-#include <media/AudioTrack.h>
-#include <media/AudioRecord.h>
+#undef ADD_BATTERY_DATA
+
+#ifdef ADD_BATTERY_DATA
#include <media/IMediaPlayerService.h>
+#include <media/IMediaDeathNotifier.h>
+#endif
#include <private/media/AudioTrackShared.h>
#include <private/media/AudioEffectShared.h>
@@ -48,27 +52,36 @@
#include "AudioMixer.h"
#include "AudioFlinger.h"
+#include "ServiceUtilities.h"
#include <media/EffectsFactoryApi.h>
#include <audio_effects/effect_visualizer.h>
#include <audio_effects/effect_ns.h>
#include <audio_effects/effect_aec.h>
-#include <cpustats/ThreadCpuUsage.h>
+#include <audio_utils/primitives.h>
+
#include <powermanager/PowerManager.h>
+
// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds
+#ifdef DEBUG_CPU_USAGE
+#include <cpustats/CentralTendencyStatistics.h>
+#include <cpustats/ThreadCpuUsage.h>
+#endif
+
+#include <common_time/cc_helper.h>
+#include <common_time/local_clock.h>
// ----------------------------------------------------------------------------
namespace android {
-static const char* kDeadlockedString = "AudioFlinger may be deadlocked\n";
-static const char* kHardwareLockedString = "Hardware lock is taken\n";
+static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
+static const char kHardwareLockedString[] = "Hardware lock is taken\n";
-//static const nsecs_t kStandbyTimeInNsecs = seconds(3);
static const float MAX_GAIN = 4096.0f;
-static const float MAX_GAIN_INT = 0x1000;
+static const uint32_t MAX_GAIN_INT = 0x1000;
// retry counts for buffer fill timeout
// 50 * ~20msecs = 1 second
@@ -80,49 +93,38 @@
static const int8_t kMaxTrackRetriesDirect = 2;
static const int kDumpLockRetries = 50;
-static const int kDumpLockSleep = 20000;
+static const int kDumpLockSleepUs = 20000;
-static const nsecs_t kWarningThrottle = seconds(5);
+// don't warn about blocked writes or record buffer overflows more often than this
+static const nsecs_t kWarningThrottleNs = seconds(5);
// RecordThread loop sleep time upon application overrun or audio HAL read error
static const int kRecordThreadSleepUs = 5000;
-static const nsecs_t kSetParametersTimeout = seconds(2);
+// maximum time to wait for setParameters to complete
+static const nsecs_t kSetParametersTimeoutNs = seconds(2);
// minimum sleep time for the mixer thread loop when tracks are active but in underrun
static const uint32_t kMinThreadSleepTimeUs = 5000;
// maximum divider applied to the active sleep time in the mixer thread loop
static const uint32_t kMaxThreadSleepTimeShift = 2;
+nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
// ----------------------------------------------------------------------------
-static bool recordingAllowed() {
- if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
- bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO"));
- if (!ok) ALOGE("Request requires android.permission.RECORD_AUDIO");
- return ok;
-}
-
-static bool settingsAllowed() {
- if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
- bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"));
- if (!ok) ALOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
- return ok;
-}
-
+#ifdef ADD_BATTERY_DATA
// To collect the amplifier usage
static void addBatteryData(uint32_t params) {
- sp<IBinder> binder =
- defaultServiceManager()->getService(String16("media.player"));
- sp<IMediaPlayerService> service = interface_cast<IMediaPlayerService>(binder);
- if (service.get() == NULL) {
- ALOGW("Cannot connect to the MediaPlayerService for battery tracking");
+ sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService();
+ if (service == NULL) {
+ // it already logged
return;
}
service->addBatteryData(params);
}
+#endif
static int load_audio_interface(const char *if_name, const hw_module_t **mod,
audio_hw_device_t **dev)
@@ -147,7 +149,7 @@
return rc;
}
-static const char *audio_interfaces[] = {
+static const char * const audio_interfaces[] = {
"primary",
"a2dp",
"usb",
@@ -158,8 +160,14 @@
AudioFlinger::AudioFlinger()
: BnAudioFlinger(),
- mPrimaryHardwareDev(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1),
- mBtNrecIsOff(false)
+ mPrimaryHardwareDev(NULL),
+ mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef()
+ mMasterVolume(1.0f),
+ mMasterVolumeSupportLvl(MVS_NONE),
+ mMasterMute(false),
+ mNextUniqueId(1),
+ mMode(AUDIO_MODE_INVALID),
+ mBtNrecIsOff(false)
{
}
@@ -170,7 +178,18 @@
Mutex::Autolock _l(mLock);
/* TODO: move all this work into an Init() function */
- mHardwareStatus = AUDIO_HW_IDLE;
+ char val_str[PROPERTY_VALUE_MAX] = { 0 };
+ if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
+ uint32_t int_val;
+ if (1 == sscanf(val_str, "%u", &int_val)) {
+ mStandbyTimeInNsecs = milliseconds(int_val);
+ ALOGI("Using %u mSec as standby time.", int_val);
+ } else {
+ mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
+ ALOGI("Using default %u mSec as standby time.",
+ (uint32_t)(mStandbyTimeInNsecs / 1000000));
+ }
+ }
for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
const hw_module_t *mod;
@@ -181,52 +200,83 @@
continue;
ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i],
- mod->name, mod->id);
+ mod->name, mod->id);
mAudioHwDevs.push(dev);
- if (!mPrimaryHardwareDev) {
+ if (mPrimaryHardwareDev == NULL) {
mPrimaryHardwareDev = dev;
ALOGI("Using '%s' (%s.%s) as the primary audio interface",
- mod->name, mod->id, audio_interfaces[i]);
+ mod->name, mod->id, audio_interfaces[i]);
}
}
- mHardwareStatus = AUDIO_HW_INIT;
-
- if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) {
+ if (mPrimaryHardwareDev == NULL) {
ALOGE("Primary audio interface not found");
- return;
+ // proceed, all later accesses to mPrimaryHardwareDev verify it's safe with initCheck()
}
+ // Currently (mPrimaryHardwareDev == NULL) == (mAudioHwDevs.size() == 0), but the way the
+ // primary HW dev is selected can change so these conditions might not always be equivalent.
+ // When that happens, re-visit all the code that assumes this.
+
+ AutoMutex lock(mHardwareLock);
+
+ // Determine the level of master volume support the primary audio HAL has,
+ // and set the initial master volume at the same time.
+ float initialVolume = 1.0;
+ mMasterVolumeSupportLvl = MVS_NONE;
+ if (0 == mPrimaryHardwareDev->init_check(mPrimaryHardwareDev)) {
+ audio_hw_device_t *dev = mPrimaryHardwareDev;
+
+ mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
+ if ((NULL != dev->get_master_volume) &&
+ (NO_ERROR == dev->get_master_volume(dev, &initialVolume))) {
+ mMasterVolumeSupportLvl = MVS_FULL;
+ } else {
+ mMasterVolumeSupportLvl = MVS_SETONLY;
+ initialVolume = 1.0;
+ }
+
+ mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
+ if ((NULL == dev->set_master_volume) ||
+ (NO_ERROR != dev->set_master_volume(dev, initialVolume))) {
+ mMasterVolumeSupportLvl = MVS_NONE;
+ }
+ mHardwareStatus = AUDIO_HW_IDLE;
+ }
+
+ // Set the mode for each audio HAL, and try to set the initial volume (if
+ // supported) for all of the non-primary audio HALs.
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
audio_hw_device_t *dev = mAudioHwDevs[i];
mHardwareStatus = AUDIO_HW_INIT;
rc = dev->init_check(dev);
+ mHardwareStatus = AUDIO_HW_IDLE;
if (rc == 0) {
- AutoMutex lock(mHardwareLock);
-
- mMode = AUDIO_MODE_NORMAL;
+ mMode = AUDIO_MODE_NORMAL; // assigned multiple times with same value
mHardwareStatus = AUDIO_HW_SET_MODE;
dev->set_mode(dev, mMode);
- mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
- dev->set_master_volume(dev, 1.0f);
+
+ if ((dev != mPrimaryHardwareDev) &&
+ (NULL != dev->set_master_volume)) {
+ mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
+ dev->set_master_volume(dev, initialVolume);
+ }
+
mHardwareStatus = AUDIO_HW_IDLE;
}
}
-}
-status_t AudioFlinger::initCheck() const
-{
- Mutex::Autolock _l(mLock);
- if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0)
- return NO_INIT;
- return NO_ERROR;
+ mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl)
+ ? initialVolume
+ : 1.0;
+ mMasterVolume = initialVolume;
+ mHardwareStatus = AUDIO_HW_IDLE;
}
AudioFlinger::~AudioFlinger()
{
- int num_devs = mAudioHwDevs.size();
while (!mRecordThreads.isEmpty()) {
// closeInput() will remove first entry from mRecordThreads
@@ -237,11 +287,10 @@
closeOutput(mPlaybackThreads.keyAt(0));
}
- for (int i = 0; i < num_devs; i++) {
- audio_hw_device_t *dev = mAudioHwDevs[i];
- audio_hw_device_close(dev);
+ for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
+ // no mHardwareLock needed, as there are no other references to this
+ audio_hw_device_close(mAudioHwDevs[i]);
}
- mAudioHwDevs.clear();
}
audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices)
@@ -263,21 +312,18 @@
result.append("Clients:\n");
for (size_t i = 0; i < mClients.size(); ++i) {
- wp<Client> wClient = mClients.valueAt(i);
- if (wClient != 0) {
- sp<Client> client = wClient.promote();
- if (client != 0) {
- snprintf(buffer, SIZE, " pid: %d\n", client->pid());
- result.append(buffer);
- }
+ sp<Client> client = mClients.valueAt(i).promote();
+ if (client != 0) {
+ snprintf(buffer, SIZE, " pid: %d\n", client->pid());
+ result.append(buffer);
}
}
result.append("Global session refs:\n");
- result.append(" session pid cnt\n");
+ result.append(" session pid count\n");
for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
AudioSessionRef *r = mAudioSessionRefs[i];
- snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt);
+ snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt);
result.append(buffer);
}
write(fd, result.string(), result.size());
@@ -290,9 +336,12 @@
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
- int hardwareStatus = mHardwareStatus;
+ hardware_call_state hardwareStatus = mHardwareStatus;
- snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus);
+ snprintf(buffer, SIZE, "Hardware status: %d\n"
+ "Standby Time mSec: %u\n",
+ hardwareStatus,
+ (uint32_t)(mStandbyTimeInNsecs / 1000000));
result.append(buffer);
write(fd, result.string(), result.size());
return NO_ERROR;
@@ -320,14 +369,14 @@
locked = true;
break;
}
- usleep(kDumpLockSleep);
+ usleep(kDumpLockSleepUs);
}
return locked;
}
status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
{
- if (checkCallingPermission(String16("android.permission.DUMP")) == false) {
+ if (!dumpAllowed()) {
dumpPermissionDenial(fd, args);
} else {
// get state of hardware lock
@@ -370,32 +419,47 @@
return NO_ERROR;
}
+sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid)
+{
+ // If pid is already in the mClients wp<> map, then use that entry
+ // (for which promote() is always != 0), otherwise create a new entry and Client.
+ sp<Client> client = mClients.valueFor(pid).promote();
+ if (client == 0) {
+ client = new Client(this, pid);
+ mClients.add(pid, client);
+ }
+
+ return client;
+}
// IAudioFlinger interface
sp<IAudioTrack> AudioFlinger::createTrack(
pid_t pid,
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
+ // FIXME dead, remove from IAudioFlinger
uint32_t flags,
const sp<IMemory>& sharedBuffer,
- int output,
+ audio_io_handle_t output,
+ bool isTimed,
int *sessionId,
status_t *status)
{
sp<PlaybackThread::Track> track;
sp<TrackHandle> trackHandle;
sp<Client> client;
- wp<Client> wclient;
status_t lStatus;
int lSessionId;
- if (streamType >= AUDIO_STREAM_CNT) {
- ALOGE("invalid stream type");
+ // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
+ // but if someone uses binder directly they could bypass that and cause us to crash
+ if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
+ ALOGE("createTrack() invalid stream type %d", streamType);
lStatus = BAD_VALUE;
goto Exit;
}
@@ -410,14 +474,7 @@
goto Exit;
}
- wclient = mClients.valueFor(pid);
-
- if (wclient != NULL) {
- client = wclient.promote();
- } else {
- client = new Client(this, pid);
- mClients.add(pid, client);
- }
+ client = registerPid_l(pid);
ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
@@ -427,6 +484,7 @@
// prevent same audio session on different output threads
uint32_t sessions = t->hasAudioSession(*sessionId);
if (sessions & PlaybackThread::TRACK_SESSION) {
+ ALOGE("createTrack() session ID %d already in use", *sessionId);
lStatus = BAD_VALUE;
goto Exit;
}
@@ -447,7 +505,7 @@
ALOGV("createTrack() lSessionId: %d", lSessionId);
track = thread->createTrack_l(client, streamType, sampleRate, format,
- channelMask, frameCount, sharedBuffer, lSessionId, &lStatus);
+ channelMask, frameCount, sharedBuffer, lSessionId, isTimed, &lStatus);
// move effect chain to this output thread if an effect on same session was waiting
// for a track to be created
@@ -467,13 +525,13 @@
}
Exit:
- if(status) {
+ if (status != NULL) {
*status = lStatus;
}
return trackHandle;
}
-uint32_t AudioFlinger::sampleRate(int output) const
+uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
@@ -484,7 +542,7 @@
return thread->sampleRate();
}
-int AudioFlinger::channelCount(int output) const
+int AudioFlinger::channelCount(audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
@@ -495,18 +553,18 @@
return thread->channelCount();
}
-uint32_t AudioFlinger::format(int output) const
+audio_format_t AudioFlinger::format(audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
ALOGW("format() unknown thread %d", output);
- return 0;
+ return AUDIO_FORMAT_INVALID;
}
return thread->format();
}
-size_t AudioFlinger::frameCount(int output) const
+size_t AudioFlinger::frameCount(audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
@@ -517,7 +575,7 @@
return thread->frameCount();
}
-uint32_t AudioFlinger::latency(int output) const
+uint32_t AudioFlinger::latency(audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
@@ -540,25 +598,34 @@
return PERMISSION_DENIED;
}
+ float swmv = value;
+
// when hw supports master volume, don't scale in sw mixer
- { // scope for the lock
- AutoMutex lock(mHardwareLock);
- mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
- if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) {
- value = 1.0f;
+ if (MVS_NONE != mMasterVolumeSupportLvl) {
+ for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
+ AutoMutex lock(mHardwareLock);
+ audio_hw_device_t *dev = mAudioHwDevs[i];
+
+ mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
+ if (NULL != dev->set_master_volume) {
+ dev->set_master_volume(dev, value);
+ }
+ mHardwareStatus = AUDIO_HW_IDLE;
}
- mHardwareStatus = AUDIO_HW_IDLE;
+
+ swmv = 1.0;
}
Mutex::Autolock _l(mLock);
- mMasterVolume = value;
- for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
- mPlaybackThreads.valueAt(i)->setMasterVolume(value);
+ mMasterVolume = value;
+ mMasterVolumeSW = swmv;
+ for (size_t i = 0; i < mPlaybackThreads.size(); i++)
+ mPlaybackThreads.valueAt(i)->setMasterVolume(swmv);
return NO_ERROR;
}
-status_t AudioFlinger::setMode(int mode)
+status_t AudioFlinger::setMode(audio_mode_t mode)
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
@@ -569,7 +636,7 @@
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
- if ((mode < 0) || (mode >= AUDIO_MODE_CNT)) {
+ if (uint32_t(mode) >= AUDIO_MODE_CNT) {
ALOGW("Illegal value: setMode(%d)", mode);
return BAD_VALUE;
}
@@ -584,8 +651,8 @@
if (NO_ERROR == ret) {
Mutex::Autolock _l(mLock);
mMode = mode;
- for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
- mPlaybackThreads.valueAt(i)->setMode(mode);
+ for (size_t i = 0; i < mPlaybackThreads.size(); i++)
+ mPlaybackThreads.valueAt(i)->setMode(mode);
}
return ret;
@@ -618,6 +685,7 @@
}
bool state = AUDIO_MODE_INVALID;
+ AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state);
mHardwareStatus = AUDIO_HW_IDLE;
@@ -632,31 +700,61 @@
}
Mutex::Autolock _l(mLock);
+ // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger
mMasterMute = muted;
- for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
- mPlaybackThreads.valueAt(i)->setMasterMute(muted);
+ for (size_t i = 0; i < mPlaybackThreads.size(); i++)
+ mPlaybackThreads.valueAt(i)->setMasterMute(muted);
return NO_ERROR;
}
float AudioFlinger::masterVolume() const
{
- return mMasterVolume;
+ Mutex::Autolock _l(mLock);
+ return masterVolume_l();
+}
+
+float AudioFlinger::masterVolumeSW() const
+{
+ Mutex::Autolock _l(mLock);
+ return masterVolumeSW_l();
}
bool AudioFlinger::masterMute() const
{
- return mMasterMute;
+ Mutex::Autolock _l(mLock);
+ return masterMute_l();
}
-status_t AudioFlinger::setStreamVolume(int stream, float value, int output)
+float AudioFlinger::masterVolume_l() const
+{
+ if (MVS_FULL == mMasterVolumeSupportLvl) {
+ float ret_val;
+ AutoMutex lock(mHardwareLock);
+
+ mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
+ ALOG_ASSERT((NULL != mPrimaryHardwareDev) &&
+ (NULL != mPrimaryHardwareDev->get_master_volume),
+ "can't get master volume");
+
+ mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ return ret_val;
+ }
+
+ return mMasterVolume;
+}
+
+status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
+ audio_io_handle_t output)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
- if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) {
+ if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
+ ALOGE("setStreamVolume() invalid stream %d", stream);
return BAD_VALUE;
}
@@ -672,8 +770,8 @@
mStreamTypes[stream].volume = value;
if (thread == NULL) {
- for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) {
- mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
+ for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+ mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
}
} else {
thread->setStreamVolume(stream, value);
@@ -682,29 +780,30 @@
return NO_ERROR;
}
-status_t AudioFlinger::setStreamMute(int stream, bool muted)
+status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
- if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT ||
+ if (uint32_t(stream) >= AUDIO_STREAM_CNT ||
uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
+ ALOGE("setStreamMute() invalid stream %d", stream);
return BAD_VALUE;
}
AutoMutex lock(mLock);
mStreamTypes[stream].mute = muted;
for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
- mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
+ mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
return NO_ERROR;
}
-float AudioFlinger::streamVolume(int stream, int output) const
+float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
{
- if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) {
+ if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
return 0.0f;
}
@@ -717,26 +816,25 @@
}
volume = thread->streamVolume(stream);
} else {
- volume = mStreamTypes[stream].volume;
+ volume = streamVolume_l(stream);
}
return volume;
}
-bool AudioFlinger::streamMute(int stream) const
+bool AudioFlinger::streamMute(audio_stream_type_t stream) const
{
- if (stream < 0 || stream >= (int)AUDIO_STREAM_CNT) {
+ if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
return true;
}
- return mStreamTypes[stream].mute;
+ AutoMutex lock(mLock);
+ return streamMute_l(stream);
}
-status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs)
+status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
{
- status_t result;
-
- ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d",
+ ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d",
ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid());
// check calling permissions
if (!settingsAllowed()) {
@@ -745,15 +843,17 @@
// ioHandle == 0 means the parameters are global to the audio hardware interface
if (ioHandle == 0) {
- AutoMutex lock(mHardwareLock);
- mHardwareStatus = AUDIO_SET_PARAMETER;
status_t final_result = NO_ERROR;
+ {
+ AutoMutex lock(mHardwareLock);
+ mHardwareStatus = AUDIO_HW_SET_PARAMETER;
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
audio_hw_device_t *dev = mAudioHwDevs[i];
- result = dev->set_parameters(dev, keyValuePairs.string());
+ status_t result = dev->set_parameters(dev, keyValuePairs.string());
final_result = result ?: final_result;
}
mHardwareStatus = AUDIO_HW_IDLE;
+ }
// disable AEC and NS if the device is a BT SCO headset supporting those pre processings
AudioParameter param = AudioParameter(keyValuePairs);
String8 value;
@@ -790,7 +890,7 @@
thread = checkPlaybackThread_l(ioHandle);
if (thread == NULL) {
thread = checkRecordThread_l(ioHandle);
- } else if (thread.get() == primaryPlaybackThread_l()) {
+ } else if (thread == primaryPlaybackThread_l()) {
// indicate output device change to all input threads for pre processing
AudioParameter param = AudioParameter(keyValuePairs);
int value;
@@ -802,25 +902,30 @@
}
}
}
- if (thread != NULL) {
- result = thread->setParameters(keyValuePairs);
- return result;
+ if (thread != 0) {
+ return thread->setParameters(keyValuePairs);
}
return BAD_VALUE;
}
-String8 AudioFlinger::getParameters(int ioHandle, const String8& keys)
+String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
{
-// ALOGV("getParameters() io %d, keys %s, tid %d, calling tid %d",
+// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d",
// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid());
if (ioHandle == 0) {
String8 out_s8;
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
+ char *s;
+ {
+ AutoMutex lock(mHardwareLock);
+ mHardwareStatus = AUDIO_HW_GET_PARAMETER;
audio_hw_device_t *dev = mAudioHwDevs[i];
- char *s = dev->get_parameters(dev, keys.string());
- out_s8 += String8(s);
+ s = dev->get_parameters(dev, keys.string());
+ mHardwareStatus = AUDIO_HW_IDLE;
+ }
+ out_s8 += String8(s ? s : "");
free(s);
}
return out_s8;
@@ -839,17 +944,21 @@
return String8("");
}
-size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
+size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return 0;
}
- return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount);
+ AutoMutex lock(mHardwareLock);
+ mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
+ size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ return size;
}
-unsigned int AudioFlinger::getInputFramesLost(int ioHandle)
+unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
{
if (ioHandle == 0) {
return 0;
@@ -877,14 +986,15 @@
}
AutoMutex lock(mHardwareLock);
- mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
+ mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value);
mHardwareStatus = AUDIO_HW_IDLE;
return ret;
}
-status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output)
+status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
+ audio_io_handle_t output) const
{
status_t status;
@@ -903,7 +1013,7 @@
Mutex::Autolock _l(mLock);
- int pid = IPCThreadState::self()->getCallingPid();
+ pid_t pid = IPCThreadState::self()->getCallingPid();
if (mNotificationClients.indexOfKey(pid) < 0) {
sp<NotificationClient> notificationClient = new NotificationClient(this,
client,
@@ -931,26 +1041,22 @@
{
Mutex::Autolock _l(mLock);
- int index = mNotificationClients.indexOfKey(pid);
- if (index >= 0) {
- sp <NotificationClient> client = mNotificationClients.valueFor(pid);
- ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid);
- mNotificationClients.removeItem(pid);
- }
+ mNotificationClients.removeItem(pid);
ALOGV("%d died, releasing its sessions", pid);
- int num = mAudioSessionRefs.size();
+ size_t num = mAudioSessionRefs.size();
bool removed = false;
- for (int i = 0; i< num; i++) {
+ for (size_t i = 0; i< num; ) {
AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
- ALOGV(" pid %d @ %d", ref->pid, i);
- if (ref->pid == pid) {
- ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid);
+ ALOGV(" pid %d @ %d", ref->mPid, i);
+ if (ref->mPid == pid) {
+ ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
mAudioSessionRefs.removeAt(i);
delete ref;
removed = true;
- i--;
num--;
+ } else {
+ i++;
}
}
if (removed) {
@@ -959,11 +1065,12 @@
}
// audioConfigChanged_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2)
+void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2)
{
size_t size = mNotificationClients.size();
for (size_t i = 0; i < size; i++) {
- mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2);
+ mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle,
+ param2);
}
}
@@ -977,19 +1084,24 @@
// ----------------------------------------------------------------------------
-AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device)
+AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
+ uint32_t device, type_t type)
: Thread(false),
- mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0),
- mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false),
- mDevice(device)
+ mType(type),
+ mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0),
+ // mChannelMask
+ mChannelCount(0),
+ mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID),
+ mParamStatus(NO_ERROR),
+ mStandby(false), mId(id),
+ mDevice(device),
+ mDeathRecipient(new PMDeathRecipient(this))
{
- mDeathRecipient = new PMDeathRecipient(this);
}
AudioFlinger::ThreadBase::~ThreadBase()
{
mParamCond.broadcast();
- mNewParameters.clear();
// do not lock the mutex in destructor
releaseWakeLock_l();
if (mPowerManager != 0) {
@@ -1000,40 +1112,26 @@
void AudioFlinger::ThreadBase::exit()
{
- // keep a strong ref on ourself so that we wont get
- // destroyed in the middle of requestExitAndWait()
- sp <ThreadBase> strongMe = this;
-
ALOGV("ThreadBase::exit");
{
- AutoMutex lock(&mLock);
- mExiting = true;
+ // This lock prevents the following race in thread (uniprocessor for illustration):
+ // if (!exitPending()) {
+ // // context switch from here to exit()
+ // // exit() calls requestExit(), what exitPending() observes
+ // // exit() calls signal(), which is dropped since no waiters
+ // // context switch back from exit() to here
+ // mWaitWorkCV.wait(...);
+ // // now thread is hung
+ // }
+ AutoMutex lock(mLock);
requestExit();
mWaitWorkCV.signal();
}
+ // When Thread::requestExitAndWait is made virtual and this method is renamed to
+ // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();"
requestExitAndWait();
}
-uint32_t AudioFlinger::ThreadBase::sampleRate() const
-{
- return mSampleRate;
-}
-
-int AudioFlinger::ThreadBase::channelCount() const
-{
- return (int)mChannelCount;
-}
-
-uint32_t AudioFlinger::ThreadBase::format() const
-{
- return mFormat;
-}
-
-size_t AudioFlinger::ThreadBase::frameCount() const
-{
- return mFrameCount;
-}
-
status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
{
status_t status;
@@ -1045,7 +1143,7 @@
mWaitWorkCV.signal();
// wait condition with timeout in case the thread loop has exited
// before the request could be processed
- if (mParamCond.waitRelative(mLock, kSetParametersTimeout) == NO_ERROR) {
+ if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) {
status = mParamStatus;
mWaitWorkCV.signal();
} else {
@@ -1063,9 +1161,9 @@
// sendConfigEvent_l() must be called with ThreadBase::mLock held
void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param)
{
- ConfigEvent *configEvent = new ConfigEvent();
- configEvent->mEvent = event;
- configEvent->mParam = param;
+ ConfigEvent configEvent;
+ configEvent.mEvent = event;
+ configEvent.mParam = param;
mConfigEvents.add(configEvent);
ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
mWaitWorkCV.signal();
@@ -1074,17 +1172,16 @@
void AudioFlinger::ThreadBase::processConfigEvents()
{
mLock.lock();
- while(!mConfigEvents.isEmpty()) {
+ while (!mConfigEvents.isEmpty()) {
ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
- ConfigEvent *configEvent = mConfigEvents[0];
+ ConfigEvent configEvent = mConfigEvents[0];
mConfigEvents.removeAt(0);
// release mLock before locking AudioFlinger mLock: lock order is always
// AudioFlinger then ThreadBase to avoid cross deadlock
mLock.unlock();
mAudioFlinger->mLock.lock();
- audioConfigChanged_l(configEvent->mEvent, configEvent->mParam);
+ audioConfigChanged_l(configEvent.mEvent, configEvent.mParam);
mAudioFlinger->mLock.unlock();
- delete configEvent;
mLock.lock();
}
mLock.unlock();
@@ -1102,6 +1199,10 @@
write(fd, buffer, strlen(buffer));
}
+ snprintf(buffer, SIZE, "io handle: %d\n", mId);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "TID: %d\n", getTid());
+ result.append(buffer);
snprintf(buffer, SIZE, "standby: %d\n", mStandby);
result.append(buffer);
snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate);
@@ -1114,7 +1215,7 @@
result.append(buffer);
snprintf(buffer, SIZE, "Format: %d\n", mFormat);
result.append(buffer);
- snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize);
+ snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize);
result.append(buffer);
snprintf(buffer, SIZE, "\nPending setParameters commands: \n");
@@ -1131,7 +1232,7 @@
snprintf(buffer, SIZE, " Index event param\n");
result.append(buffer);
for (size_t i = 0; i < mConfigEvents.size(); i++) {
- snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i]->mEvent, mConfigEvents[i]->mParam);
+ snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam);
result.append(buffer);
}
result.append("\n");
@@ -1236,8 +1337,7 @@
void AudioFlinger::ThreadBase::setEffectSuspended_l(
const effect_uuid_t *type, bool suspend, int sessionId)
{
- sp<EffectChain> chain;
- chain = getEffectChain_l(sessionId);
+ sp<EffectChain> chain = getEffectChain_l(sessionId);
if (chain != 0) {
if (type != NULL) {
chain->setEffectSuspended_l(type, suspend);
@@ -1251,7 +1351,7 @@
void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain)
{
- int index = mSuspendedSessions.indexOfKey(chain->sessionId());
+ ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId());
if (index < 0) {
return;
}
@@ -1260,13 +1360,13 @@
mSuspendedSessions.editValueAt(index);
for (size_t i = 0; i < sessionEffects.size(); i++) {
- sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i);
+ sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i);
for (int j = 0; j < desc->mRefCount; j++) {
if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) {
chain->setEffectSuspendedAll_l(true);
} else {
ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x",
- desc->mType.timeLow);
+ desc->mType.timeLow);
chain->setEffectSuspended_l(&desc->mType, true);
}
}
@@ -1277,7 +1377,7 @@
bool suspend,
int sessionId)
{
- int index = mSuspendedSessions.indexOfKey(sessionId);
+ ssize_t index = mSuspendedSessions.indexOfKey(sessionId);
KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects;
@@ -1301,7 +1401,7 @@
}
index = sessionEffects.indexOfKey(key);
- sp <SuspendedSessionDesc> desc;
+ sp<SuspendedSessionDesc> desc;
if (suspend) {
if (index >= 0) {
desc = sessionEffects.valueAt(index);
@@ -1365,24 +1465,39 @@
AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output,
- int id,
- uint32_t device)
- : ThreadBase(audioFlinger, id, device),
- mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output),
- mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false)
+ audio_io_handle_t id,
+ uint32_t device,
+ type_t type)
+ : ThreadBase(audioFlinger, id, device, type),
+ mMixBuffer(NULL), mSuspended(0), mBytesWritten(0),
+ // Assumes constructor is called by AudioFlinger with it's mLock held,
+ // but it would be safer to explicitly pass initial masterMute as parameter
+ mMasterMute(audioFlinger->masterMute_l()),
+ // mStreamTypes[] initialized in constructor body
+ mOutput(output),
+ // Assumes constructor is called by AudioFlinger with it's mLock held,
+ // but it would be safer to explicitly pass initial masterVolume as parameter
+ mMasterVolume(audioFlinger->masterVolumeSW_l()),
+ mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false),
+ mMixerStatus(MIXER_IDLE),
+ mPrevMixerStatus(MIXER_IDLE),
+ standbyDelay(AudioFlinger::mStandbyTimeInNsecs)
{
- snprintf(mName, kNameLength, "AudioOut_%d", id);
+ snprintf(mName, kNameLength, "AudioOut_%X", id);
readOutputParameters();
- mMasterVolume = mAudioFlinger->masterVolume();
- mMasterMute = mAudioFlinger->masterMute();
-
- for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
- mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream);
- mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream);
- mStreamTypes[stream].valid = true;
+ // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor
+ // There is no AUDIO_STREAM_MIN, and ++ operator does not compile
+ for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT;
+ stream = (audio_stream_type_t) (stream + 1)) {
+ mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream);
+ mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream);
+ // initialized by stream_type_t default constructor
+ // mStreamTypes[stream].valid = true;
}
+ // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here,
+ // because mAudioFlinger doesn't have one to copy from
}
AudioFlinger::PlaybackThread::~PlaybackThread()
@@ -1419,13 +1534,10 @@
result.append(buffer);
result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n");
for (size_t i = 0; i < mActiveTracks.size(); ++i) {
- wp<Track> wTrack = mActiveTracks[i];
- if (wTrack != 0) {
- sp<Track> track = wTrack.promote();
- if (track != 0) {
- track->dump(buffer, SIZE);
- result.append(buffer);
- }
+ sp<Track> track = mActiveTracks[i].promote();
+ if (track != 0) {
+ track->dump(buffer, SIZE);
+ result.append(buffer);
}
}
write(fd, result.string(), result.size());
@@ -1479,13 +1591,14 @@
// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
const sp<AudioFlinger::Client>& client,
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
+ bool isTimed,
status_t *status)
{
sp<Track> track;
@@ -1522,22 +1635,28 @@
// all tracks in same audio session must share the same routing strategy otherwise
// conflicts will happen when tracks are moved from one output to another by audio policy
// manager
- uint32_t strategy =
- AudioSystem::getStrategyForStream((audio_stream_type_t)streamType);
+ uint32_t strategy = AudioSystem::getStrategyForStream(streamType);
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> t = mTracks[i];
- if (t != 0) {
- if (sessionId == t->sessionId() &&
- strategy != AudioSystem::getStrategyForStream((audio_stream_type_t)t->type())) {
+ if (t != 0 && !t->isOutputTrack()) {
+ uint32_t actual = AudioSystem::getStrategyForStream(t->streamType());
+ if (sessionId == t->sessionId() && strategy != actual) {
+ ALOGE("createTrack_l() mismatched strategy; expected %u but found %u",
+ strategy, actual);
lStatus = BAD_VALUE;
goto Exit;
}
}
}
- track = new Track(this, client, streamType, sampleRate, format,
- channelMask, frameCount, sharedBuffer, sessionId);
- if (track->getCblk() == NULL || track->name() < 0) {
+ if (!isTimed) {
+ track = new Track(this, client, streamType, sampleRate, format,
+ channelMask, frameCount, sharedBuffer, sessionId);
+ } else {
+ track = TimedTrack::create(this, client, streamType, sampleRate, format,
+ channelMask, frameCount, sharedBuffer, sessionId);
+ }
+ if (track == NULL || track->getCblk() == NULL || track->name() < 0) {
lStatus = NO_MEMORY;
goto Exit;
}
@@ -1547,7 +1666,7 @@
if (chain != 0) {
ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
track->setMainBuffer(chain->inBuffer());
- chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type()));
+ chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType()));
chain->incTrackCnt();
}
@@ -1555,14 +1674,14 @@
// createTrack() was called by the client process.
if (!mStreamTypes[streamType].valid) {
ALOGW("createTrack_l() on thread %p: invalidating track on stream %d",
- this, streamType);
+ this, streamType);
android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags);
}
}
lStatus = NO_ERROR;
Exit:
- if(status) {
+ if (status) {
*status = lStatus;
}
return track;
@@ -1578,50 +1697,36 @@
}
}
-status_t AudioFlinger::PlaybackThread::setMasterVolume(float value)
+void AudioFlinger::PlaybackThread::setMasterVolume(float value)
{
+ Mutex::Autolock _l(mLock);
mMasterVolume = value;
- return NO_ERROR;
}
-status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted)
+void AudioFlinger::PlaybackThread::setMasterMute(bool muted)
{
- mMasterMute = muted;
- return NO_ERROR;
+ Mutex::Autolock _l(mLock);
+ setMasterMute_l(muted);
}
-float AudioFlinger::PlaybackThread::masterVolume() const
+void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
{
- return mMasterVolume;
-}
-
-bool AudioFlinger::PlaybackThread::masterMute() const
-{
- return mMasterMute;
-}
-
-status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value)
-{
+ Mutex::Autolock _l(mLock);
mStreamTypes[stream].volume = value;
- return NO_ERROR;
}
-status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted)
+void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
{
+ Mutex::Autolock _l(mLock);
mStreamTypes[stream].mute = muted;
- return NO_ERROR;
}
-float AudioFlinger::PlaybackThread::streamVolume(int stream) const
+float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const
{
+ Mutex::Autolock _l(mLock);
return mStreamTypes[stream].volume;
}
-bool AudioFlinger::PlaybackThread::streamMute(int stream) const
-{
- return mStreamTypes[stream].mute;
-}
-
// addTrack_l() must be called with ThreadBase::mLock held
status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
{
@@ -1691,7 +1796,7 @@
// audioConfigChanged_l() must be called with AudioFlinger::mLock held
void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
AudioSystem::OutputDescriptor desc;
- void *param2 = 0;
+ void *param2 = NULL;
ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param);
@@ -1721,12 +1826,12 @@
mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
mChannelCount = (uint16_t)popcount(mChannelMask);
mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
- mFrameSize = (uint16_t)audio_stream_frame_size(&mOutput->stream->common);
+ mFrameSize = audio_stream_frame_size(&mOutput->stream->common);
mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
// FIXME - Current mixer implementation only supports stereo output: Always
// Allocate a stereo buffer even if HW output is mono.
- if (mMixBuffer != NULL) delete[] mMixBuffer;
+ delete[] mMixBuffer;
mMixBuffer = new int16_t[mFrameCount * 2];
memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
@@ -1744,7 +1849,7 @@
status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames)
{
- if (halFrames == 0 || dspFrames == 0) {
+ if (halFrames == NULL || dspFrames == NULL) {
return BAD_VALUE;
}
Mutex::Autolock _l(mLock);
@@ -1787,14 +1892,14 @@
sp<Track> track = mTracks[i];
if (sessionId == track->sessionId() &&
!(track->mCblk->flags & CBLK_INVALID_MSK)) {
- return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type());
+ return AudioSystem::getStrategyForStream(track->streamType());
}
}
return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
}
-AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput()
+AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const
{
Mutex::Autolock _l(mLock);
return mOutput;
@@ -1831,13 +1936,11 @@
// ----------------------------------------------------------------------------
-AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
- : PlaybackThread(audioFlinger, output, id, device),
- mAudioMixer(0), mPrevMixerStatus(MIXER_IDLE)
+AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
+ audio_io_handle_t id, uint32_t device, type_t type)
+ : PlaybackThread(audioFlinger, output, id, device, type)
{
- mType = ThreadBase::MIXER;
mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
-
// FIXME - Current mixer implementation only supports stereo output
if (mChannelCount == 1) {
ALOGE("Invalid audio hardware channel count");
@@ -1849,234 +1952,388 @@
delete mAudioMixer;
}
-bool AudioFlinger::MixerThread::threadLoop()
+class CpuStats {
+public:
+ CpuStats();
+ void sample(const String8 &title);
+#ifdef DEBUG_CPU_USAGE
+private:
+ ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns
+ CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns
+
+ CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles
+
+ int mCpuNum; // thread's current CPU number
+ int mCpukHz; // frequency of thread's current CPU in kHz
+#endif
+};
+
+CpuStats::CpuStats()
+#ifdef DEBUG_CPU_USAGE
+ : mCpuNum(-1), mCpukHz(-1)
+#endif
+{
+}
+
+void CpuStats::sample(const String8 &title) {
+#ifdef DEBUG_CPU_USAGE
+ // get current thread's delta CPU time in wall clock ns
+ double wcNs;
+ bool valid = mCpuUsage.sampleAndEnable(wcNs);
+
+ // record sample for wall clock statistics
+ if (valid) {
+ mWcStats.sample(wcNs);
+ }
+
+ // get the current CPU number
+ int cpuNum = sched_getcpu();
+
+ // get the current CPU frequency in kHz
+ int cpukHz = mCpuUsage.getCpukHz(cpuNum);
+
+ // check if either CPU number or frequency changed
+ if (cpuNum != mCpuNum || cpukHz != mCpukHz) {
+ mCpuNum = cpuNum;
+ mCpukHz = cpukHz;
+ // ignore sample for purposes of cycles
+ valid = false;
+ }
+
+ // if no change in CPU number or frequency, then record sample for cycle statistics
+ if (valid && mCpukHz > 0) {
+ double cycles = wcNs * cpukHz * 0.000001;
+ mHzStats.sample(cycles);
+ }
+
+ unsigned n = mWcStats.n();
+ // mCpuUsage.elapsed() is expensive, so don't call it every loop
+ if ((n & 127) == 1) {
+ long long elapsed = mCpuUsage.elapsed();
+ if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
+ double perLoop = elapsed / (double) n;
+ double perLoop100 = perLoop * 0.01;
+ double perLoop1k = perLoop * 0.001;
+ double mean = mWcStats.mean();
+ double stddev = mWcStats.stddev();
+ double minimum = mWcStats.minimum();
+ double maximum = mWcStats.maximum();
+ double meanCycles = mHzStats.mean();
+ double stddevCycles = mHzStats.stddev();
+ double minCycles = mHzStats.minimum();
+ double maxCycles = mHzStats.maximum();
+ mCpuUsage.resetElapsed();
+ mWcStats.reset();
+ mHzStats.reset();
+ ALOGD("CPU usage for %s over past %.1f secs\n"
+ " (%u mixer loops at %.1f mean ms per loop):\n"
+ " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n"
+ " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n"
+ " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f",
+ title.string(),
+ elapsed * .000000001, n, perLoop * .000001,
+ mean * .001,
+ stddev * .001,
+ minimum * .001,
+ maximum * .001,
+ mean / perLoop100,
+ stddev / perLoop100,
+ minimum / perLoop100,
+ maximum / perLoop100,
+ meanCycles / perLoop1k,
+ stddevCycles / perLoop1k,
+ minCycles / perLoop1k,
+ maxCycles / perLoop1k);
+
+ }
+ }
+#endif
+};
+
+void AudioFlinger::PlaybackThread::checkSilentMode_l()
+{
+ if (!mMasterMute) {
+ char value[PROPERTY_VALUE_MAX];
+ if (property_get("ro.audio.silent", value, "0") > 0) {
+ char *endptr;
+ unsigned long ul = strtoul(value, &endptr, 0);
+ if (*endptr == '\0' && ul != 0) {
+ ALOGD("Silence is golden");
+ // The setprop command will not allow a property to be changed after
+ // the first time it is set, so we don't have to worry about un-muting.
+ setMasterMute_l(true);
+ }
+ }
+ }
+}
+
+bool AudioFlinger::PlaybackThread::threadLoop()
{
Vector< sp<Track> > tracksToRemove;
- uint32_t mixerStatus = MIXER_IDLE;
- nsecs_t standbyTime = systemTime();
- size_t mixBufferSize = mFrameCount * mFrameSize;
- // FIXME: Relaxed timing because of a certain device that can't meet latency
- // Should be reduced to 2x after the vendor fixes the driver issue
- // increase threshold again due to low power audio mode. The way this warning threshold is
- // calculated and its usefulness should be reconsidered anyway.
- nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15;
+
+ standbyTime = systemTime();
+
+ // MIXER
nsecs_t lastWarning = 0;
- bool longStandbyExit = false;
- uint32_t activeSleepTime = activeSleepTimeUs();
- uint32_t idleSleepTime = idleSleepTimeUs();
- uint32_t sleepTime = idleSleepTime;
- uint32_t sleepTimeShift = 0;
- Vector< sp<EffectChain> > effectChains;
-#ifdef DEBUG_CPU_USAGE
- ThreadCpuUsage cpu;
- const CentralTendencyStatistics& stats = cpu.statistics();
-#endif
+if (mType == MIXER) {
+ longStandbyExit = false;
+}
+
+ // DUPLICATING
+ // FIXME could this be made local to while loop?
+ writeFrames = 0;
+
+ cacheParameters_l();
+ sleepTime = idleSleepTime;
+
+if (mType == MIXER) {
+ sleepTimeShift = 0;
+}
+
+ CpuStats cpuStats;
+ const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid()));
acquireWakeLock();
while (!exitPending())
{
-#ifdef DEBUG_CPU_USAGE
- cpu.sampleAndEnable();
- unsigned n = stats.n();
- // cpu.elapsed() is expensive, so don't call it every loop
- if ((n & 127) == 1) {
- long long elapsed = cpu.elapsed();
- if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
- double perLoop = elapsed / (double) n;
- double perLoop100 = perLoop * 0.01;
- double mean = stats.mean();
- double stddev = stats.stddev();
- double minimum = stats.minimum();
- double maximum = stats.maximum();
- cpu.resetStatistics();
- ALOGI("CPU usage over past %.1f secs (%u mixer loops at %.1f mean ms per loop):\n us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f",
- elapsed * .000000001, n, perLoop * .000001,
- mean * .001,
- stddev * .001,
- minimum * .001,
- maximum * .001,
- mean / perLoop100,
- stddev / perLoop100,
- minimum / perLoop100,
- maximum / perLoop100);
- }
- }
-#endif
+ cpuStats.sample(myName);
+
+ Vector< sp<EffectChain> > effectChains;
+
processConfigEvents();
- mixerStatus = MIXER_IDLE;
{ // scope for mLock
Mutex::Autolock _l(mLock);
if (checkForNewParameters_l()) {
- mixBufferSize = mFrameCount * mFrameSize;
- // FIXME: Relaxed timing because of a certain device that can't meet latency
- // Should be reduced to 2x after the vendor fixes the driver issue
- // increase threshold again due to low power audio mode. The way this warning
- // threshold is calculated and its usefulness should be reconsidered anyway.
- maxPeriod = seconds(mFrameCount) / mSampleRate * 15;
- activeSleepTime = activeSleepTimeUs();
- idleSleepTime = idleSleepTimeUs();
+ cacheParameters_l();
}
- const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
+ saveOutputTracks();
// put audio hardware into standby after short delay
- if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
- mSuspended) {
+ if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
+ mSuspended > 0)) {
if (!mStandby) {
- ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended);
- mOutput->stream->common.standby(&mOutput->stream->common);
+
+ threadLoop_standby();
+
mStandby = true;
mBytesWritten = 0;
}
- if (!activeTracks.size() && mConfigEvents.isEmpty()) {
+ if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
// we're about to wait, flush the binder command buffer
IPCThreadState::self()->flushCommands();
+ clearOutputTracks();
+
if (exitPending()) break;
releaseWakeLock_l();
// wait until we have something to do...
- ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid());
+ ALOGV("%s going to sleep", myName.string());
mWaitWorkCV.wait(mLock);
- ALOGV("MixerThread %p TID %d waking up\n", this, gettid());
+ ALOGV("%s waking up", myName.string());
acquireWakeLock_l();
mPrevMixerStatus = MIXER_IDLE;
- if (mMasterMute == false) {
- char value[PROPERTY_VALUE_MAX];
- property_get("ro.audio.silent", value, "0");
- if (atoi(value)) {
- ALOGD("Silence is golden");
- setMasterMute(true);
- }
+
+ checkSilentMode_l();
+
+ standbyTime = systemTime() + standbyDelay;
+ sleepTime = idleSleepTime;
+ if (mType == MIXER) {
+ sleepTimeShift = 0;
}
- standbyTime = systemTime() + kStandbyTimeInNsecs;
- sleepTime = idleSleepTime;
- sleepTimeShift = 0;
continue;
}
}
- mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
+ mixer_state newMixerStatus = prepareTracks_l(&tracksToRemove);
+ // Shift in the new status; this could be a queue if it's
+ // useful to filter the mixer status over several cycles.
+ mPrevMixerStatus = mMixerStatus;
+ mMixerStatus = newMixerStatus;
// prevent any changes in effect chain list and in each effect chain
// during mixing and effect process as the audio buffers could be deleted
// or modified if an effect is created or deleted
lockEffectChains_l(effectChains);
- }
-
- if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
- // mix buffers...
- mAudioMixer->process();
- // increase sleep time progressively when application underrun condition clears.
- // Only increase sleep time if the mixer is ready for two consecutive times to avoid
- // that a steady state of alternating ready/not ready conditions keeps the sleep time
- // such that we would underrun the audio HAL.
- if ((sleepTime == 0) && (sleepTimeShift > 0)) {
- sleepTimeShift--;
- }
- sleepTime = 0;
- standbyTime = systemTime() + kStandbyTimeInNsecs;
- //TODO: delay standby when effects have a tail
- } else {
- // If no tracks are ready, sleep once for the duration of an output
- // buffer size, then write 0s to the output
- if (sleepTime == 0) {
- if (mixerStatus == MIXER_TRACKS_ENABLED) {
- sleepTime = activeSleepTime >> sleepTimeShift;
- if (sleepTime < kMinThreadSleepTimeUs) {
- sleepTime = kMinThreadSleepTimeUs;
- }
- // reduce sleep time in case of consecutive application underruns to avoid
- // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer
- // duration we would end up writing less data than needed by the audio HAL if
- // the condition persists.
- if (sleepTimeShift < kMaxThreadSleepTimeShift) {
- sleepTimeShift++;
- }
- } else {
- sleepTime = idleSleepTime;
- }
- } else if (mBytesWritten != 0 ||
- (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) {
- memset (mMixBuffer, 0, mixBufferSize);
- sleepTime = 0;
- ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start");
- }
- // TODO add standby time extension fct of effect tail
}
- if (mSuspended) {
+ if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) {
+ threadLoop_mix();
+ } else {
+ threadLoop_sleepTime();
+ }
+
+ if (mSuspended > 0) {
sleepTime = suspendSleepTimeUs();
}
+
+ // only process effects if we're going to write
+ if (sleepTime == 0) {
+ for (size_t i = 0; i < effectChains.size(); i ++) {
+ effectChains[i]->process_l();
+ }
+ }
+
+ // enable changes in effect chain
+ unlockEffectChains(effectChains);
+
// sleepTime == 0 means we must write to audio hardware
if (sleepTime == 0) {
- for (size_t i = 0; i < effectChains.size(); i ++) {
- effectChains[i]->process_l();
- }
- // enable changes in effect chain
- unlockEffectChains(effectChains);
- mLastWriteTime = systemTime();
- mInWrite = true;
- mBytesWritten += mixBufferSize;
- int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
- if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
- mNumWrites++;
- mInWrite = false;
+ threadLoop_write();
+
+if (mType == MIXER) {
+ // write blocked detection
nsecs_t now = systemTime();
nsecs_t delta = now - mLastWriteTime;
if (!mStandby && delta > maxPeriod) {
mNumDelayedWrites++;
- if ((now - lastWarning) > kWarningThrottle) {
+ if ((now - lastWarning) > kWarningThrottleNs) {
ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
ns2ms(delta), mNumDelayedWrites, this);
lastWarning = now;
}
+ // FIXME this is broken: longStandbyExit should be handled out of the if() and with
+ // a different threshold. Or completely removed for what it is worth anyway...
if (mStandby) {
longStandbyExit = true;
}
}
+}
+
mStandby = false;
} else {
- // enable changes in effect chain
- unlockEffectChains(effectChains);
usleep(sleepTime);
}
- // finally let go of all our tracks, without the lock held
+ // finally let go of removed track(s), without the lock held
// since we can't guarantee the destructors won't acquire that
// same lock.
tracksToRemove.clear();
+ // FIXME I don't understand the need for this here;
+ // it was in the original code but maybe the
+ // assignment in saveOutputTracks() makes this unnecessary?
+ clearOutputTracks();
+
// Effect chains will be actually deleted here if they were removed from
// mEffectChains list during mixing or effects processing
effectChains.clear();
+
+ // FIXME Note that the above .clear() is no longer necessary since effectChains
+ // is now local to this block, but will keep it for now (at least until merge done).
}
+if (mType == MIXER || mType == DIRECT) {
+ // put output stream into standby mode
if (!mStandby) {
mOutput->stream->common.standby(&mOutput->stream->common);
}
+}
+if (mType == DUPLICATING) {
+ // for DuplicatingThread, standby mode is handled by the outputTracks
+}
releaseWakeLock();
- ALOGV("MixerThread %p exiting", this);
+ ALOGV("Thread %p type %d exiting", this, mType);
return false;
}
+// shared by MIXER and DIRECT, overridden by DUPLICATING
+void AudioFlinger::PlaybackThread::threadLoop_write()
+{
+ // FIXME rewrite to reduce number of system calls
+ mLastWriteTime = systemTime();
+ mInWrite = true;
+ mBytesWritten += mixBufferSize;
+ int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
+ if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
+ mNumWrites++;
+ mInWrite = false;
+}
+
+// shared by MIXER and DIRECT, overridden by DUPLICATING
+void AudioFlinger::PlaybackThread::threadLoop_standby()
+{
+ ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended);
+ mOutput->stream->common.standby(&mOutput->stream->common);
+}
+
+void AudioFlinger::MixerThread::threadLoop_mix()
+{
+ // obtain the presentation timestamp of the next output buffer
+ int64_t pts;
+ status_t status = INVALID_OPERATION;
+
+ if (NULL != mOutput->stream->get_next_write_timestamp) {
+ status = mOutput->stream->get_next_write_timestamp(
+ mOutput->stream, &pts);
+ }
+
+ if (status != NO_ERROR) {
+ pts = AudioBufferProvider::kInvalidPTS;
+ }
+
+ // mix buffers...
+ mAudioMixer->process(pts);
+ // increase sleep time progressively when application underrun condition clears.
+ // Only increase sleep time if the mixer is ready for two consecutive times to avoid
+ // that a steady state of alternating ready/not ready conditions keeps the sleep time
+ // such that we would underrun the audio HAL.
+ if ((sleepTime == 0) && (sleepTimeShift > 0)) {
+ sleepTimeShift--;
+ }
+ sleepTime = 0;
+ standbyTime = systemTime() + standbyDelay;
+ //TODO: delay standby when effects have a tail
+}
+
+void AudioFlinger::MixerThread::threadLoop_sleepTime()
+{
+ // If no tracks are ready, sleep once for the duration of an output
+ // buffer size, then write 0s to the output
+ if (sleepTime == 0) {
+ if (mMixerStatus == MIXER_TRACKS_ENABLED) {
+ sleepTime = activeSleepTime >> sleepTimeShift;
+ if (sleepTime < kMinThreadSleepTimeUs) {
+ sleepTime = kMinThreadSleepTimeUs;
+ }
+ // reduce sleep time in case of consecutive application underruns to avoid
+ // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer
+ // duration we would end up writing less data than needed by the audio HAL if
+ // the condition persists.
+ if (sleepTimeShift < kMaxThreadSleepTimeShift) {
+ sleepTimeShift++;
+ }
+ } else {
+ sleepTime = idleSleepTime;
+ }
+ } else if (mBytesWritten != 0 ||
+ (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) {
+ memset (mMixBuffer, 0, mixBufferSize);
+ sleepTime = 0;
+ ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start");
+ }
+ // TODO add standby time extension fct of effect tail
+}
+
// prepareTracks_l() must be called with ThreadBase::mLock held
-uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove)
+AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l(
+ Vector< sp<Track> > *tracksToRemove)
{
- uint32_t mixerStatus = MIXER_IDLE;
+ mixer_state mixerStatus = MIXER_IDLE;
// find out which tracks need to be processed
- size_t count = activeTracks.size();
+ size_t count = mActiveTracks.size();
size_t mixedTracks = 0;
size_t tracksWithEffect = 0;
@@ -2096,15 +2353,16 @@
}
for (size_t i=0 ; i<count ; i++) {
- sp<Track> t = activeTracks[i].promote();
+ sp<Track> t = mActiveTracks[i].promote();
if (t == 0) continue;
+ // this const just means the local variable doesn't change
Track* const track = t.get();
audio_track_cblk_t* cblk = track->cblk();
// The first time a track is added we wait
// for all its buffers to be filled before processing it
- mAudioMixer->setActiveTrack(track->name());
+ int name = track->name();
// make sure that we have enough frames to mix one full buffer.
// enforce this condition only once to enable draining the buffer in case the client
// app does not call stop() and relies on underrun to stop:
@@ -2124,13 +2382,13 @@
// the minimum track buffer size is normally twice the number of frames necessary
// to fill one buffer and the resampler should not leave more than one buffer worth
// of unreleased frames after each pass, but just in case...
- LOG_ASSERT(minFrames <= cblk->frameCount);
+ ALOG_ASSERT(minFrames <= cblk->frameCount);
}
}
- if ((cblk->framesReady() >= minFrames) && track->isReady() &&
+ if ((track->framesReady() >= minFrames) && track->isReady() &&
!track->isPaused() && !track->isTerminated())
{
- //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this);
+ //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this);
mixedTracks++;
@@ -2143,8 +2401,8 @@
if (chain != 0) {
tracksWithEffect++;
} else {
- ALOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d",
- track->name(), track->sessionId());
+ ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d",
+ name, track->sessionId());
}
}
@@ -2157,7 +2415,7 @@
track->mState = TrackBase::ACTIVE;
param = AudioMixer::RAMP_VOLUME;
}
- mAudioMixer->setParameter(AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
+ mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
} else if (cblk->server != 0) {
// If the track is stopped before the first frame was mixed,
// do not apply ramp
@@ -2167,7 +2425,7 @@
// compute volume for this track
uint32_t vl, vr, va;
if (track->isMuted() || track->isPausing() ||
- mStreamTypes[track->type()].mute) {
+ mStreamTypes[track->streamType()].mute) {
vl = vr = va = 0;
if (track->isPausing()) {
track->setPaused();
@@ -2175,12 +2433,33 @@
} else {
// read original volumes with volume control
- float typeVolume = mStreamTypes[track->type()].volume;
+ float typeVolume = mStreamTypes[track->streamType()].volume;
float v = masterVolume * typeVolume;
- vl = (uint32_t)(v * cblk->volume[0]) << 12;
- vr = (uint32_t)(v * cblk->volume[1]) << 12;
+ uint32_t vlr = cblk->getVolumeLR();
+ vl = vlr & 0xFFFF;
+ vr = vlr >> 16;
+ // track volumes come from shared memory, so can't be trusted and must be clamped
+ if (vl > MAX_GAIN_INT) {
+ ALOGV("Track left volume out of range: %04X", vl);
+ vl = MAX_GAIN_INT;
+ }
+ if (vr > MAX_GAIN_INT) {
+ ALOGV("Track right volume out of range: %04X", vr);
+ vr = MAX_GAIN_INT;
+ }
+ // now apply the master volume and stream type volume
+ vl = (uint32_t)(v * vl) << 12;
+ vr = (uint32_t)(v * vr) << 12;
+ // assuming master volume and stream type volume each go up to 1.0,
+ // vl and vr are now in 8.24 format
- va = (uint32_t)(v * cblk->sendLevel);
+ uint16_t sendLevel = cblk->getSendLevel_U4_12();
+ // send level comes from shared memory and so may be corrupt
+ if (sendLevel > MAX_GAIN_INT) {
+ ALOGV("Track send level out of range: %04X", sendLevel);
+ sendLevel = MAX_GAIN_INT;
+ }
+ va = (uint32_t)(v * sendLevel);
}
// Delegate volume control to effect in track effect chain if needed
if (chain != 0 && chain->setVolume_l(&vl, &vr)) {
@@ -2197,38 +2476,40 @@
}
// Convert volumes from 8.24 to 4.12 format
- int16_t left, right, aux;
- uint32_t v_clamped = (vl + (1 << 11)) >> 12;
- if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
- left = int16_t(v_clamped);
- v_clamped = (vr + (1 << 11)) >> 12;
- if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
- right = int16_t(v_clamped);
+ // This additional clamping is needed in case chain->setVolume_l() overshot
+ vl = (vl + (1 << 11)) >> 12;
+ if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT;
+ vr = (vr + (1 << 11)) >> 12;
+ if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT;
- if (va > MAX_GAIN_INT) va = MAX_GAIN_INT;
- aux = int16_t(va);
+ if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for -
// XXX: these things DON'T need to be done each time
- mAudioMixer->setBufferProvider(track);
- mAudioMixer->enable(AudioMixer::MIXING);
+ mAudioMixer->setBufferProvider(name, track);
+ mAudioMixer->enable(name);
- mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left);
- mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right);
- mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux);
+ mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl);
+ mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr);
+ mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va);
mAudioMixer->setParameter(
+ name,
AudioMixer::TRACK,
AudioMixer::FORMAT, (void *)track->format());
mAudioMixer->setParameter(
+ name,
AudioMixer::TRACK,
AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
mAudioMixer->setParameter(
+ name,
AudioMixer::RESAMPLE,
AudioMixer::SAMPLE_RATE,
(void *)(cblk->sampleRate));
mAudioMixer->setParameter(
+ name,
AudioMixer::TRACK,
AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
mAudioMixer->setParameter(
+ name,
AudioMixer::TRACK,
AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
@@ -2242,7 +2523,7 @@
mixerStatus = MIXER_TRACKS_READY;
}
} else {
- //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this);
+ //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this);
if (track->isStopped()) {
track->reset();
}
@@ -2254,7 +2535,7 @@
// No buffers for this track. Give it a few chances to
// fill a buffer, then remove it from active list.
if (--(track->mRetryCount) <= 0) {
- ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this);
+ ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this);
tracksToRemove->add(track);
// indicate to client process that the track was disabled because of underrun
android_atomic_or(CBLK_DISABLED_ON, &cblk->flags);
@@ -2266,13 +2547,13 @@
mixerStatus = MIXER_TRACKS_ENABLED;
}
}
- mAudioMixer->disable(AudioMixer::MIXING);
+ mAudioMixer->disable(name);
}
}
// remove all the tracks that need to be...
count = tracksToRemove->size();
- if (UNLIKELY(count)) {
+ if (CC_UNLIKELY(count)) {
for (size_t i=0 ; i<count ; i++) {
const sp<Track>& track = tracksToRemove->itemAt(i);
mActiveTracks.remove(track);
@@ -2296,11 +2577,36 @@
memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t));
}
- mPrevMixerStatus = mixerStatus;
return mixerStatus;
}
-void AudioFlinger::MixerThread::invalidateTracks(int streamType)
+/*
+The derived values that are cached:
+ - mixBufferSize from frame count * frame size
+ - activeSleepTime from activeSleepTimeUs()
+ - idleSleepTime from idleSleepTimeUs()
+ - standbyDelay from mActiveSleepTimeUs (DIRECT only)
+ - maxPeriod from frame count and sample rate (MIXER only)
+
+The parameters that affect these derived values are:
+ - frame count
+ - frame size
+ - sample rate
+ - device type: A2DP or not
+ - device latency
+ - format: PCM or not
+ - active sleep time
+ - idle sleep time
+*/
+
+void AudioFlinger::PlaybackThread::cacheParameters_l()
+{
+ mixBufferSize = mFrameCount * mFrameSize;
+ activeSleepTime = activeSleepTimeUs();
+ idleSleepTime = idleSleepTimeUs();
+}
+
+void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType)
{
ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d",
this, streamType, mTracks.size());
@@ -2309,14 +2615,14 @@
size_t size = mTracks.size();
for (size_t i = 0; i < size; i++) {
sp<Track> t = mTracks[i];
- if (t->type() == streamType) {
+ if (t->streamType() == streamType) {
android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags);
t->mCblk->cv.signal();
}
}
}
-void AudioFlinger::PlaybackThread::setStreamValid(int streamType, bool valid)
+void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid)
{
ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d",
this, streamType, valid);
@@ -2353,7 +2659,7 @@
reconfig = true;
}
if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
- if (value != AUDIO_FORMAT_PCM_16_BIT) {
+ if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) {
status = BAD_VALUE;
} else {
reconfig = true;
@@ -2368,7 +2674,7 @@
}
if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
// do not accept frame count changes if tracks are open as the track buffer
- // size depends on frame count and correct behavior would not be garantied
+ // size depends on frame count and correct behavior would not be guaranteed
// if frame count is changed after track creation
if (!mTracks.isEmpty()) {
status = INVALID_OPERATION;
@@ -2377,6 +2683,7 @@
}
}
if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
+#ifdef ADD_BATTERY_DATA
// when changing the audio output device, call addBatteryData to notify
// the change
if ((int)mDevice != value) {
@@ -2397,6 +2704,7 @@
addBatteryData(params);
}
}
+#endif
// forward device change to effects that have requested to be
// aware of attached audio device.
@@ -2410,14 +2718,16 @@
status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
keyValuePair.string());
if (!mStandby && status == INVALID_OPERATION) {
- mOutput->stream->common.standby(&mOutput->stream->common);
- mStandby = true;
- mBytesWritten = 0;
- status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
+ mOutput->stream->common.standby(&mOutput->stream->common);
+ mStandby = true;
+ mBytesWritten = 0;
+ status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
keyValuePair.string());
}
if (status == NO_ERROR && reconfig) {
delete mAudioMixer;
+ // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't)
+ mAudioMixer = NULL;
readOutputParameters();
mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
for (size_t i = 0; i < mTracks.size() ; i++) {
@@ -2439,7 +2749,7 @@
mParamCond.signal();
// wait for condition with time out in case the thread calling ThreadBase::setParameters()
// already timed out waiting for the status and will never signal the condition.
- mWaitWorkCV.waitRelative(mLock, kSetParametersTimeout);
+ mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
}
return reconfig;
}
@@ -2468,42 +2778,193 @@
return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
}
-// ----------------------------------------------------------------------------
-AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
- : PlaybackThread(audioFlinger, output, id, device)
+void AudioFlinger::MixerThread::cacheParameters_l()
{
- mType = ThreadBase::DIRECT;
+ PlaybackThread::cacheParameters_l();
+
+ // FIXME: Relaxed timing because of a certain device that can't meet latency
+ // Should be reduced to 2x after the vendor fixes the driver issue
+ // increase threshold again due to low power audio mode. The way this warning
+ // threshold is calculated and its usefulness should be reconsidered anyway.
+ maxPeriod = seconds(mFrameCount) / mSampleRate * 15;
+}
+
+// ----------------------------------------------------------------------------
+AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger,
+ AudioStreamOut* output, audio_io_handle_t id, uint32_t device)
+ : PlaybackThread(audioFlinger, output, id, device, DIRECT)
+ // mLeftVolFloat, mRightVolFloat
+ // mLeftVolShort, mRightVolShort
+{
}
AudioFlinger::DirectOutputThread::~DirectOutputThread()
{
}
-
-static inline int16_t clamp16(int32_t sample)
+AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l(
+ Vector< sp<Track> > *tracksToRemove
+)
{
- if ((sample>>15) ^ (sample>>31))
- sample = 0x7FFF ^ (sample>>31);
- return sample;
+ sp<Track> trackToRemove;
+
+ mixer_state mixerStatus = MIXER_IDLE;
+
+ // find out which tracks need to be processed
+ if (mActiveTracks.size() != 0) {
+ sp<Track> t = mActiveTracks[0].promote();
+ // The track died recently
+ if (t == 0) return MIXER_IDLE;
+
+ Track* const track = t.get();
+ audio_track_cblk_t* cblk = track->cblk();
+
+ // The first time a track is added we wait
+ // for all its buffers to be filled before processing it
+ if (cblk->framesReady() && track->isReady() &&
+ !track->isPaused() && !track->isTerminated())
+ {
+ //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
+
+ if (track->mFillingUpStatus == Track::FS_FILLED) {
+ track->mFillingUpStatus = Track::FS_ACTIVE;
+ mLeftVolFloat = mRightVolFloat = 0;
+ mLeftVolShort = mRightVolShort = 0;
+ if (track->mState == TrackBase::RESUMING) {
+ track->mState = TrackBase::ACTIVE;
+ rampVolume = true;
+ }
+ } else if (cblk->server != 0) {
+ // If the track is stopped before the first frame was mixed,
+ // do not apply ramp
+ rampVolume = true;
+ }
+ // compute volume for this track
+ float left, right;
+ if (track->isMuted() || mMasterMute || track->isPausing() ||
+ mStreamTypes[track->streamType()].mute) {
+ left = right = 0;
+ if (track->isPausing()) {
+ track->setPaused();
+ }
+ } else {
+ float typeVolume = mStreamTypes[track->streamType()].volume;
+ float v = mMasterVolume * typeVolume;
+ uint32_t vlr = cblk->getVolumeLR();
+ float v_clamped = v * (vlr & 0xFFFF);
+ if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
+ left = v_clamped/MAX_GAIN;
+ v_clamped = v * (vlr >> 16);
+ if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
+ right = v_clamped/MAX_GAIN;
+ }
+
+ if (left != mLeftVolFloat || right != mRightVolFloat) {
+ mLeftVolFloat = left;
+ mRightVolFloat = right;
+
+ // If audio HAL implements volume control,
+ // force software volume to nominal value
+ if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) {
+ left = 1.0f;
+ right = 1.0f;
+ }
+
+ // Convert volumes from float to 8.24
+ uint32_t vl = (uint32_t)(left * (1 << 24));
+ uint32_t vr = (uint32_t)(right * (1 << 24));
+
+ // Delegate volume control to effect in track effect chain if needed
+ // only one effect chain can be present on DirectOutputThread, so if
+ // there is one, the track is connected to it
+ if (!mEffectChains.isEmpty()) {
+ // Do not ramp volume if volume is controlled by effect
+ if (mEffectChains[0]->setVolume_l(&vl, &vr)) {
+ rampVolume = false;
+ }
+ }
+
+ // Convert volumes from 8.24 to 4.12 format
+ uint32_t v_clamped = (vl + (1 << 11)) >> 12;
+ if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
+ leftVol = (uint16_t)v_clamped;
+ v_clamped = (vr + (1 << 11)) >> 12;
+ if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
+ rightVol = (uint16_t)v_clamped;
+ } else {
+ leftVol = mLeftVolShort;
+ rightVol = mRightVolShort;
+ rampVolume = false;
+ }
+
+ // reset retry count
+ track->mRetryCount = kMaxTrackRetriesDirect;
+ mActiveTrack = t;
+ mixerStatus = MIXER_TRACKS_READY;
+ } else {
+ //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
+ if (track->isStopped()) {
+ track->reset();
+ }
+ if (track->isTerminated() || track->isStopped() || track->isPaused()) {
+ // We have consumed all the buffers of this track.
+ // Remove it from the list of active tracks.
+ trackToRemove = track;
+ } else {
+ // No buffers for this track. Give it a few chances to
+ // fill a buffer, then remove it from active list.
+ if (--(track->mRetryCount) <= 0) {
+ ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
+ trackToRemove = track;
+ } else {
+ mixerStatus = MIXER_TRACKS_ENABLED;
+ }
+ }
+ }
+ }
+
+ // FIXME merge this with similar code for removing multiple tracks
+ // remove all the tracks that need to be...
+ if (CC_UNLIKELY(trackToRemove != 0)) {
+ tracksToRemove->add(trackToRemove);
+ mActiveTracks.remove(trackToRemove);
+ if (!mEffectChains.isEmpty()) {
+ ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(),
+ trackToRemove->sessionId());
+ mEffectChains[0]->decActiveTrackCnt();
+ }
+ if (trackToRemove->isTerminated()) {
+ removeTrack_l(trackToRemove);
+ }
+ }
+
+ return mixerStatus;
}
-static inline
-int32_t mul(int16_t in, int16_t v)
+void AudioFlinger::DirectOutputThread::threadLoop_mix()
{
-#if defined(__arm__) && !defined(__thumb__)
- int32_t out;
- asm( "smulbb %[out], %[in], %[v] \n"
- : [out]"=r"(out)
- : [in]"%r"(in), [v]"r"(v)
- : );
- return out;
-#else
- return in * int32_t(v);
-#endif
-}
+ AudioBufferProvider::Buffer buffer;
+ size_t frameCount = mFrameCount;
+ int8_t *curBuf = (int8_t *)mMixBuffer;
+ // output audio to hardware
+ while (frameCount) {
+ buffer.frameCount = frameCount;
+ mActiveTrack->getNextBuffer(&buffer);
+ if (CC_UNLIKELY(buffer.raw == NULL)) {
+ memset(curBuf, 0, frameCount * mFrameSize);
+ break;
+ }
+ memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
+ frameCount -= buffer.frameCount;
+ curBuf += buffer.frameCount * mFrameSize;
+ mActiveTrack->releaseBuffer(&buffer);
+ }
+ sleepTime = 0;
+ standbyTime = systemTime() + standbyDelay;
+ mActiveTrack.clear();
-void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp)
-{
+ // apply volume
+
// Do not apply volume on compressed audio
if (!audio_is_linear_pcm(mFormat)) {
return;
@@ -2514,14 +2975,14 @@
size_t count = mFrameCount * mChannelCount;
uint8_t *src = (uint8_t *)mMixBuffer + count-1;
int16_t *dst = mMixBuffer + count-1;
- while(count--) {
+ while (count--) {
*dst-- = (int16_t)(*src--^0x80) << 8;
}
}
- size_t frameCount = mFrameCount;
+ frameCount = mFrameCount;
int16_t *out = mMixBuffer;
- if (ramp) {
+ if (rampVolume) {
if (mChannelCount == 1) {
int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
int32_t vlInc = d / (int32_t)frameCount;
@@ -2567,7 +3028,7 @@
size_t count = mFrameCount * mChannelCount;
int16_t *src = mMixBuffer;
uint8_t *dst = (uint8_t *)mMixBuffer;
- while(count--) {
+ while (count--) {
*dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80;
}
}
@@ -2576,289 +3037,18 @@
mRightVolShort = rightVol;
}
-bool AudioFlinger::DirectOutputThread::threadLoop()
+void AudioFlinger::DirectOutputThread::threadLoop_sleepTime()
{
- uint32_t mixerStatus = MIXER_IDLE;
- sp<Track> trackToRemove;
- sp<Track> activeTrack;
- nsecs_t standbyTime = systemTime();
- int8_t *curBuf;
- size_t mixBufferSize = mFrameCount*mFrameSize;
- uint32_t activeSleepTime = activeSleepTimeUs();
- uint32_t idleSleepTime = idleSleepTimeUs();
- uint32_t sleepTime = idleSleepTime;
- // use shorter standby delay as on normal output to release
- // hardware resources as soon as possible
- nsecs_t standbyDelay = microseconds(activeSleepTime*2);
-
- acquireWakeLock();
-
- while (!exitPending())
- {
- bool rampVolume;
- uint16_t leftVol;
- uint16_t rightVol;
- Vector< sp<EffectChain> > effectChains;
-
- processConfigEvents();
-
- mixerStatus = MIXER_IDLE;
-
- { // scope for the mLock
-
- Mutex::Autolock _l(mLock);
-
- if (checkForNewParameters_l()) {
- mixBufferSize = mFrameCount*mFrameSize;
- activeSleepTime = activeSleepTimeUs();
- idleSleepTime = idleSleepTimeUs();
- standbyDelay = microseconds(activeSleepTime*2);
- }
-
- // put audio hardware into standby after short delay
- if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
- mSuspended) {
- // wait until we have something to do...
- if (!mStandby) {
- ALOGV("Audio hardware entering standby, mixer %p\n", this);
- mOutput->stream->common.standby(&mOutput->stream->common);
- mStandby = true;
- mBytesWritten = 0;
- }
-
- if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
- // we're about to wait, flush the binder command buffer
- IPCThreadState::self()->flushCommands();
-
- if (exitPending()) break;
-
- releaseWakeLock_l();
- ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid());
- mWaitWorkCV.wait(mLock);
- ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid());
- acquireWakeLock_l();
-
- if (mMasterMute == false) {
- char value[PROPERTY_VALUE_MAX];
- property_get("ro.audio.silent", value, "0");
- if (atoi(value)) {
- ALOGD("Silence is golden");
- setMasterMute(true);
- }
- }
-
- standbyTime = systemTime() + standbyDelay;
- sleepTime = idleSleepTime;
- continue;
- }
- }
-
- effectChains = mEffectChains;
-
- // find out which tracks need to be processed
- if (mActiveTracks.size() != 0) {
- sp<Track> t = mActiveTracks[0].promote();
- if (t == 0) continue;
-
- Track* const track = t.get();
- audio_track_cblk_t* cblk = track->cblk();
-
- // The first time a track is added we wait
- // for all its buffers to be filled before processing it
- if (cblk->framesReady() && track->isReady() &&
- !track->isPaused() && !track->isTerminated())
- {
- //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
-
- if (track->mFillingUpStatus == Track::FS_FILLED) {
- track->mFillingUpStatus = Track::FS_ACTIVE;
- mLeftVolFloat = mRightVolFloat = 0;
- mLeftVolShort = mRightVolShort = 0;
- if (track->mState == TrackBase::RESUMING) {
- track->mState = TrackBase::ACTIVE;
- rampVolume = true;
- }
- } else if (cblk->server != 0) {
- // If the track is stopped before the first frame was mixed,
- // do not apply ramp
- rampVolume = true;
- }
- // compute volume for this track
- float left, right;
- if (track->isMuted() || mMasterMute || track->isPausing() ||
- mStreamTypes[track->type()].mute) {
- left = right = 0;
- if (track->isPausing()) {
- track->setPaused();
- }
- } else {
- float typeVolume = mStreamTypes[track->type()].volume;
- float v = mMasterVolume * typeVolume;
- float v_clamped = v * cblk->volume[0];
- if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
- left = v_clamped/MAX_GAIN;
- v_clamped = v * cblk->volume[1];
- if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
- right = v_clamped/MAX_GAIN;
- }
-
- if (left != mLeftVolFloat || right != mRightVolFloat) {
- mLeftVolFloat = left;
- mRightVolFloat = right;
-
- // If audio HAL implements volume control,
- // force software volume to nominal value
- if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) {
- left = 1.0f;
- right = 1.0f;
- }
-
- // Convert volumes from float to 8.24
- uint32_t vl = (uint32_t)(left * (1 << 24));
- uint32_t vr = (uint32_t)(right * (1 << 24));
-
- // Delegate volume control to effect in track effect chain if needed
- // only one effect chain can be present on DirectOutputThread, so if
- // there is one, the track is connected to it
- if (!effectChains.isEmpty()) {
- // Do not ramp volume if volume is controlled by effect
- if(effectChains[0]->setVolume_l(&vl, &vr)) {
- rampVolume = false;
- }
- }
-
- // Convert volumes from 8.24 to 4.12 format
- uint32_t v_clamped = (vl + (1 << 11)) >> 12;
- if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
- leftVol = (uint16_t)v_clamped;
- v_clamped = (vr + (1 << 11)) >> 12;
- if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
- rightVol = (uint16_t)v_clamped;
- } else {
- leftVol = mLeftVolShort;
- rightVol = mRightVolShort;
- rampVolume = false;
- }
-
- // reset retry count
- track->mRetryCount = kMaxTrackRetriesDirect;
- activeTrack = t;
- mixerStatus = MIXER_TRACKS_READY;
- } else {
- //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
- if (track->isStopped()) {
- track->reset();
- }
- if (track->isTerminated() || track->isStopped() || track->isPaused()) {
- // We have consumed all the buffers of this track.
- // Remove it from the list of active tracks.
- trackToRemove = track;
- } else {
- // No buffers for this track. Give it a few chances to
- // fill a buffer, then remove it from active list.
- if (--(track->mRetryCount) <= 0) {
- ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
- trackToRemove = track;
- } else {
- mixerStatus = MIXER_TRACKS_ENABLED;
- }
- }
- }
- }
-
- // remove all the tracks that need to be...
- if (UNLIKELY(trackToRemove != 0)) {
- mActiveTracks.remove(trackToRemove);
- if (!effectChains.isEmpty()) {
- ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(),
- trackToRemove->sessionId());
- effectChains[0]->decActiveTrackCnt();
- }
- if (trackToRemove->isTerminated()) {
- removeTrack_l(trackToRemove);
- }
- }
-
- lockEffectChains_l(effectChains);
- }
-
- if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
- AudioBufferProvider::Buffer buffer;
- size_t frameCount = mFrameCount;
- curBuf = (int8_t *)mMixBuffer;
- // output audio to hardware
- while (frameCount) {
- buffer.frameCount = frameCount;
- activeTrack->getNextBuffer(&buffer);
- if (UNLIKELY(buffer.raw == 0)) {
- memset(curBuf, 0, frameCount * mFrameSize);
- break;
- }
- memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
- frameCount -= buffer.frameCount;
- curBuf += buffer.frameCount * mFrameSize;
- activeTrack->releaseBuffer(&buffer);
- }
- sleepTime = 0;
- standbyTime = systemTime() + standbyDelay;
+ if (sleepTime == 0) {
+ if (mMixerStatus == MIXER_TRACKS_ENABLED) {
+ sleepTime = activeSleepTime;
} else {
- if (sleepTime == 0) {
- if (mixerStatus == MIXER_TRACKS_ENABLED) {
- sleepTime = activeSleepTime;
- } else {
- sleepTime = idleSleepTime;
- }
- } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) {
- memset (mMixBuffer, 0, mFrameCount * mFrameSize);
- sleepTime = 0;
- }
+ sleepTime = idleSleepTime;
}
-
- if (mSuspended) {
- sleepTime = suspendSleepTimeUs();
- }
- // sleepTime == 0 means we must write to audio hardware
- if (sleepTime == 0) {
- if (mixerStatus == MIXER_TRACKS_READY) {
- applyVolume(leftVol, rightVol, rampVolume);
- }
- for (size_t i = 0; i < effectChains.size(); i ++) {
- effectChains[i]->process_l();
- }
- unlockEffectChains(effectChains);
-
- mLastWriteTime = systemTime();
- mInWrite = true;
- mBytesWritten += mixBufferSize;
- int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
- if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
- mNumWrites++;
- mInWrite = false;
- mStandby = false;
- } else {
- unlockEffectChains(effectChains);
- usleep(sleepTime);
- }
-
- // finally let go of removed track, without the lock held
- // since we can't guarantee the destructors won't acquire that
- // same lock.
- trackToRemove.clear();
- activeTrack.clear();
-
- // Effect chains will be actually deleted here if they were removed from
- // mEffectChains list during mixing or effects processing
- effectChains.clear();
+ } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) {
+ memset (mMixBuffer, 0, mFrameCount * mFrameSize);
+ sleepTime = 0;
}
-
- if (!mStandby) {
- mOutput->stream->common.standby(&mOutput->stream->common);
- }
-
- releaseWakeLock();
-
- ALOGV("DirectOutputThread %p exiting", this);
- return false;
}
// getTrackName_l() must be called with ThreadBase::mLock held
@@ -2897,10 +3087,10 @@
status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
keyValuePair.string());
if (!mStandby && status == INVALID_OPERATION) {
- mOutput->stream->common.standby(&mOutput->stream->common);
- mStandby = true;
- mBytesWritten = 0;
- status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
+ mOutput->stream->common.standby(&mOutput->stream->common);
+ mStandby = true;
+ mBytesWritten = 0;
+ status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
keyValuePair.string());
}
if (status == NO_ERROR && reconfig) {
@@ -2915,7 +3105,7 @@
mParamCond.signal();
// wait for condition with time out in case the thread calling ThreadBase::setParameters()
// already timed out waiting for the status and will never signal the condition.
- mWaitWorkCV.waitRelative(mLock, kSetParametersTimeout);
+ mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
}
return reconfig;
}
@@ -2953,13 +3143,22 @@
return time;
}
+void AudioFlinger::DirectOutputThread::cacheParameters_l()
+{
+ PlaybackThread::cacheParameters_l();
+
+ // use shorter standby delay as on normal output to release
+ // hardware resources as soon as possible
+ standbyDelay = microseconds(activeSleepTime*2);
+}
// ----------------------------------------------------------------------------
-AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id)
- : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX)
+AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
+ AudioFlinger::MixerThread* mainThread, audio_io_handle_t id)
+ : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING),
+ mWaitTimeMs(UINT_MAX)
{
- mType = ThreadBase::DUPLICATING;
addOutputTrack(mainThread);
}
@@ -2968,166 +3167,74 @@
for (size_t i = 0; i < mOutputTracks.size(); i++) {
mOutputTracks[i]->destroy();
}
- mOutputTracks.clear();
}
-bool AudioFlinger::DuplicatingThread::threadLoop()
+void AudioFlinger::DuplicatingThread::threadLoop_mix()
{
- Vector< sp<Track> > tracksToRemove;
- uint32_t mixerStatus = MIXER_IDLE;
- nsecs_t standbyTime = systemTime();
- size_t mixBufferSize = mFrameCount*mFrameSize;
- SortedVector< sp<OutputTrack> > outputTracks;
- uint32_t writeFrames = 0;
- uint32_t activeSleepTime = activeSleepTimeUs();
- uint32_t idleSleepTime = idleSleepTimeUs();
- uint32_t sleepTime = idleSleepTime;
- Vector< sp<EffectChain> > effectChains;
-
- acquireWakeLock();
-
- while (!exitPending())
- {
- processConfigEvents();
-
- mixerStatus = MIXER_IDLE;
- { // scope for the mLock
-
- Mutex::Autolock _l(mLock);
-
- if (checkForNewParameters_l()) {
- mixBufferSize = mFrameCount*mFrameSize;
- updateWaitTime();
- activeSleepTime = activeSleepTimeUs();
- idleSleepTime = idleSleepTimeUs();
- }
-
- const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
-
- for (size_t i = 0; i < mOutputTracks.size(); i++) {
- outputTracks.add(mOutputTracks[i]);
- }
-
- // put audio hardware into standby after short delay
- if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
- mSuspended) {
- if (!mStandby) {
- for (size_t i = 0; i < outputTracks.size(); i++) {
- outputTracks[i]->stop();
- }
- mStandby = true;
- mBytesWritten = 0;
- }
-
- if (!activeTracks.size() && mConfigEvents.isEmpty()) {
- // we're about to wait, flush the binder command buffer
- IPCThreadState::self()->flushCommands();
- outputTracks.clear();
-
- if (exitPending()) break;
-
- releaseWakeLock_l();
- ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid());
- mWaitWorkCV.wait(mLock);
- ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid());
- acquireWakeLock_l();
-
- mPrevMixerStatus = MIXER_IDLE;
- if (mMasterMute == false) {
- char value[PROPERTY_VALUE_MAX];
- property_get("ro.audio.silent", value, "0");
- if (atoi(value)) {
- ALOGD("Silence is golden");
- setMasterMute(true);
- }
- }
-
- standbyTime = systemTime() + kStandbyTimeInNsecs;
- sleepTime = idleSleepTime;
- continue;
- }
- }
-
- mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
-
- // prevent any changes in effect chain list and in each effect chain
- // during mixing and effect process as the audio buffers could be deleted
- // or modified if an effect is created or deleted
- lockEffectChains_l(effectChains);
- }
-
- if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
- // mix buffers...
- if (outputsReady(outputTracks)) {
- mAudioMixer->process();
- } else {
- memset(mMixBuffer, 0, mixBufferSize);
- }
- sleepTime = 0;
- writeFrames = mFrameCount;
- } else {
- if (sleepTime == 0) {
- if (mixerStatus == MIXER_TRACKS_ENABLED) {
- sleepTime = activeSleepTime;
- } else {
- sleepTime = idleSleepTime;
- }
- } else if (mBytesWritten != 0) {
- // flush remaining overflow buffers in output tracks
- for (size_t i = 0; i < outputTracks.size(); i++) {
- if (outputTracks[i]->isActive()) {
- sleepTime = 0;
- writeFrames = 0;
- memset(mMixBuffer, 0, mixBufferSize);
- break;
- }
- }
- }
- }
-
- if (mSuspended) {
- sleepTime = suspendSleepTimeUs();
- }
- // sleepTime == 0 means we must write to audio hardware
- if (sleepTime == 0) {
- for (size_t i = 0; i < effectChains.size(); i ++) {
- effectChains[i]->process_l();
- }
- // enable changes in effect chain
- unlockEffectChains(effectChains);
-
- standbyTime = systemTime() + kStandbyTimeInNsecs;
- for (size_t i = 0; i < outputTracks.size(); i++) {
- outputTracks[i]->write(mMixBuffer, writeFrames);
- }
- mStandby = false;
- mBytesWritten += mixBufferSize;
- } else {
- // enable changes in effect chain
- unlockEffectChains(effectChains);
- usleep(sleepTime);
- }
-
- // finally let go of all our tracks, without the lock held
- // since we can't guarantee the destructors won't acquire that
- // same lock.
- tracksToRemove.clear();
- outputTracks.clear();
-
- // Effect chains will be actually deleted here if they were removed from
- // mEffectChains list during mixing or effects processing
- effectChains.clear();
+ // mix buffers...
+ if (outputsReady(outputTracks)) {
+ mAudioMixer->process(AudioBufferProvider::kInvalidPTS);
+ } else {
+ memset(mMixBuffer, 0, mixBufferSize);
}
+ sleepTime = 0;
+ writeFrames = mFrameCount;
+}
- releaseWakeLock();
+void AudioFlinger::DuplicatingThread::threadLoop_sleepTime()
+{
+ if (sleepTime == 0) {
+ if (mMixerStatus == MIXER_TRACKS_ENABLED) {
+ sleepTime = activeSleepTime;
+ } else {
+ sleepTime = idleSleepTime;
+ }
+ } else if (mBytesWritten != 0) {
+ // flush remaining overflow buffers in output tracks
+ for (size_t i = 0; i < outputTracks.size(); i++) {
+ if (outputTracks[i]->isActive()) {
+ sleepTime = 0;
+ writeFrames = 0;
+ memset(mMixBuffer, 0, mixBufferSize);
+ break;
+ }
+ }
+ }
+}
- return false;
+void AudioFlinger::DuplicatingThread::threadLoop_write()
+{
+ standbyTime = systemTime() + standbyDelay;
+ for (size_t i = 0; i < outputTracks.size(); i++) {
+ outputTracks[i]->write(mMixBuffer, writeFrames);
+ }
+ mBytesWritten += mixBufferSize;
+}
+
+void AudioFlinger::DuplicatingThread::threadLoop_standby()
+{
+ // DuplicatingThread implements standby by stopping all tracks
+ for (size_t i = 0; i < outputTracks.size(); i++) {
+ outputTracks[i]->stop();
+ }
+}
+
+void AudioFlinger::DuplicatingThread::saveOutputTracks()
+{
+ outputTracks = mOutputTracks;
+}
+
+void AudioFlinger::DuplicatingThread::clearOutputTracks()
+{
+ outputTracks.clear();
}
void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
{
+ Mutex::Autolock _l(mLock);
+ // FIXME explain this formula
int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
- OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread,
+ OutputTrack *outputTrack = new OutputTrack(thread,
this,
mSampleRate,
mFormat,
@@ -3137,7 +3244,7 @@
thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f);
mOutputTracks.add(outputTrack);
ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
- updateWaitTime();
+ updateWaitTime_l();
}
}
@@ -3145,22 +3252,23 @@
{
Mutex::Autolock _l(mLock);
for (size_t i = 0; i < mOutputTracks.size(); i++) {
- if (mOutputTracks[i]->thread() == (ThreadBase *)thread) {
+ if (mOutputTracks[i]->thread() == thread) {
mOutputTracks[i]->destroy();
mOutputTracks.removeAt(i);
- updateWaitTime();
+ updateWaitTime_l();
return;
}
}
ALOGV("removeOutputTrack(): unkonwn thread: %p", thread);
}
-void AudioFlinger::DuplicatingThread::updateWaitTime()
+// caller must hold mLock
+void AudioFlinger::DuplicatingThread::updateWaitTime_l()
{
mWaitTimeMs = UINT_MAX;
for (size_t i = 0; i < mOutputTracks.size(); i++) {
sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
- if (strong != NULL) {
+ if (strong != 0) {
uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
if (waitTimeMs < mWaitTimeMs) {
mWaitTimeMs = waitTimeMs;
@@ -3170,10 +3278,10 @@
}
-bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks)
+bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks)
{
for (size_t i = 0; i < outputTracks.size(); i++) {
- sp <ThreadBase> thread = outputTracks[i]->thread().promote();
+ sp<ThreadBase> thread = outputTracks[i]->thread().promote();
if (thread == 0) {
ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get());
return false;
@@ -3192,45 +3300,55 @@
return (mWaitTimeMs * 1000) / 2;
}
+void AudioFlinger::DuplicatingThread::cacheParameters_l()
+{
+ // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first
+ updateWaitTime_l();
+
+ MixerThread::cacheParameters_l();
+}
+
// ----------------------------------------------------------------------------
// TrackBase constructor must be called with AudioFlinger::mLock held
AudioFlinger::ThreadBase::TrackBase::TrackBase(
- const wp<ThreadBase>& thread,
+ ThreadBase *thread,
const sp<Client>& client,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
- uint32_t flags,
const sp<IMemory>& sharedBuffer,
int sessionId)
: RefBase(),
mThread(thread),
mClient(client),
- mCblk(0),
+ mCblk(NULL),
+ // mBuffer
+ // mBufferEnd
mFrameCount(0),
mState(IDLE),
- mClientTid(-1),
mFormat(format),
- mFlags(flags & ~SYSTEM_FLAGS_MASK),
+ mStepServerFailed(false),
mSessionId(sessionId)
+ // mChannelCount
+ // mChannelMask
{
ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
// ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
- size_t size = sizeof(audio_track_cblk_t);
- uint8_t channelCount = popcount(channelMask);
- size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
- if (sharedBuffer == 0) {
- size += bufferSize;
- }
+ size_t size = sizeof(audio_track_cblk_t);
+ uint8_t channelCount = popcount(channelMask);
+ size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
+ if (sharedBuffer == 0) {
+ size += bufferSize;
+ }
- if (client != NULL) {
+ if (client != NULL) {
mCblkMemory = client->heap()->allocate(size);
if (mCblkMemory != 0) {
mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
- if (mCblk) { // construct the shared structure in-place.
+ if (mCblk != NULL) { // construct the shared structure in-place.
new(mCblk) audio_track_cblk_t();
// clear all buffers
mCblk->frameCount = frameCount;
@@ -3253,45 +3371,52 @@
client->heap()->dump("AudioTrack");
return;
}
- } else {
- mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
- if (mCblk) { // construct the shared structure in-place.
- new(mCblk) audio_track_cblk_t();
- // clear all buffers
- mCblk->frameCount = frameCount;
- mCblk->sampleRate = sampleRate;
- mChannelCount = channelCount;
- mChannelMask = channelMask;
- mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
- memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
- // Force underrun condition to avoid false underrun callback until first data is
- // written to buffer (other flags are cleared)
- mCblk->flags = CBLK_UNDERRUN_ON;
- mBufferEnd = (uint8_t *)mBuffer + bufferSize;
- }
- }
+ } else {
+ mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
+ // construct the shared structure in-place.
+ new(mCblk) audio_track_cblk_t();
+ // clear all buffers
+ mCblk->frameCount = frameCount;
+ mCblk->sampleRate = sampleRate;
+ mChannelCount = channelCount;
+ mChannelMask = channelMask;
+ mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
+ memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
+ // Force underrun condition to avoid false underrun callback until first data is
+ // written to buffer (other flags are cleared)
+ mCblk->flags = CBLK_UNDERRUN_ON;
+ mBufferEnd = (uint8_t *)mBuffer + bufferSize;
+ }
}
AudioFlinger::ThreadBase::TrackBase::~TrackBase()
{
- if (mCblk) {
- mCblk->~audio_track_cblk_t(); // destroy our shared-structure.
- if (mClient == NULL) {
+ if (mCblk != NULL) {
+ if (mClient == 0) {
delete mCblk;
+ } else {
+ mCblk->~audio_track_cblk_t(); // destroy our shared-structure.
}
}
- mCblkMemory.clear(); // and free the shared memory
- if (mClient != NULL) {
+ mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to
+ if (mClient != 0) {
+ // Client destructor must run with AudioFlinger mutex locked
Mutex::Autolock _l(mClient->audioFlinger()->mLock);
+ // If the client's reference count drops to zero, the associated destructor
+ // must run with AudioFlinger lock held. Thus the explicit clear() rather than
+ // relying on the automatic clear() at end of scope.
mClient.clear();
}
}
+// AudioBufferProvider interface
+// getNextBuffer() = 0;
+// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack
void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
{
- buffer->raw = 0;
+ buffer->raw = NULL;
mFrameCount = buffer->frameCount;
- step();
+ (void) step(); // ignore return value of step()
buffer->frameCount = 0;
}
@@ -3302,7 +3427,7 @@
result = cblk->stepServer(mFrameCount);
if (!result) {
ALOGV("stepServer failed acquiring cblk mutex");
- mFlags |= STEPSERVER_FAILED;
+ mStepServerFailed = true;
}
return result;
}
@@ -3314,40 +3439,28 @@
cblk->server = 0;
cblk->userBase = 0;
cblk->serverBase = 0;
- mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK);
+ mStepServerFailed = false;
ALOGV("TrackBase::reset");
}
-sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const
-{
- return mCblkMemory;
-}
-
int AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
return (int)mCblk->sampleRate;
}
-int AudioFlinger::ThreadBase::TrackBase::channelCount() const {
- return (const int)mChannelCount;
-}
-
-uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const {
- return mChannelMask;
-}
-
void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
audio_track_cblk_t* cblk = this->cblk();
- int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize;
- int8_t *bufferEnd = bufferStart + frames * cblk->frameSize;
+ size_t frameSize = cblk->frameSize;
+ int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize;
+ int8_t *bufferEnd = bufferStart + frames * frameSize;
// Check validity of returned pointer in case the track control block would have been corrupted.
if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
- ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) {
+ ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) {
ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \
server %d, serverBase %d, user %d, userBase %d",
bufferStart, bufferEnd, mBuffer, mBufferEnd,
cblk->server, cblk->serverBase, cblk->user, cblk->userBase);
- return 0;
+ return NULL;
}
return bufferStart;
@@ -3357,32 +3470,28 @@
// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
AudioFlinger::PlaybackThread::Track::Track(
- const wp<ThreadBase>& thread,
+ PlaybackThread *thread,
const sp<Client>& client,
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId)
- : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId),
+ : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId),
mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL),
mAuxEffectId(0), mHasVolumeController(false)
{
if (mCblk != NULL) {
- sp<ThreadBase> baseThread = thread.promote();
- if (baseThread != 0) {
- PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get();
- mName = playbackThread->getTrackName_l();
- mMainBuffer = playbackThread->mixBuffer();
+ if (thread != NULL) {
+ mName = thread->getTrackName_l();
+ mMainBuffer = thread->mixBuffer();
}
- ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+ ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid());
if (mName < 0) {
ALOGE("no more track names available");
}
- mVolume[0] = 1.0f;
- mVolume[1] = 1.0f;
mStreamType = streamType;
// NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
// 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
@@ -3404,7 +3513,7 @@
{
// NOTE: destroyTrack_l() can remove a strong reference to this Track
// by removing it from mTracks vector, so there is a risk that this Tracks's
- // desctructor is called. As the destructor needs to lock mLock,
+ // destructor is called. As the destructor needs to lock mLock,
// we must acquire a strong reference on this Track before locking mLock
// here so that the destructor is called only when exiting this function.
// On the other hand, as long as Track::destroy() is only called by
@@ -3416,12 +3525,12 @@
if (thread != 0) {
if (!isOutputTrack()) {
if (mState == ACTIVE || mState == RESUMING) {
- AudioSystem::stopOutput(thread->id(),
- (audio_stream_type_t)mStreamType,
- mSessionId);
+ AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
+#ifdef ADD_BATTERY_DATA
// to track the speaker usage
addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
+#endif
}
AudioSystem::releaseOutput(thread->id());
}
@@ -3434,9 +3543,10 @@
void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
{
+ uint32_t vlr = mCblk->getVolumeLR();
snprintf(buffer, size, " %05d %05d %03u %03u 0x%08x %05u %04u %1d %1d %1d %05u %05u %05u 0x%08x 0x%08x 0x%08x 0x%08x\n",
mName - AudioMixer::TRACK0,
- (mClient == NULL) ? getpid() : mClient->pid(),
+ (mClient == 0) ? getpid_cached : mClient->pid(),
mStreamType,
mFormat,
mChannelMask,
@@ -3446,30 +3556,32 @@
mMute,
mFillingUpStatus,
mCblk->sampleRate,
- mCblk->volume[0],
- mCblk->volume[1],
+ vlr & 0xFFFF,
+ vlr >> 16,
mCblk->server,
mCblk->user,
(int)mMainBuffer,
(int)mAuxBuffer);
}
-status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+// AudioBufferProvider interface
+status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
+ AudioBufferProvider::Buffer* buffer, int64_t pts)
{
- audio_track_cblk_t* cblk = this->cblk();
- uint32_t framesReady;
- uint32_t framesReq = buffer->frameCount;
+ audio_track_cblk_t* cblk = this->cblk();
+ uint32_t framesReady;
+ uint32_t framesReq = buffer->frameCount;
- // Check if last stepServer failed, try to step now
- if (mFlags & TrackBase::STEPSERVER_FAILED) {
- if (!step()) goto getNextBuffer_exit;
- ALOGV("stepServer recovered");
- mFlags &= ~TrackBase::STEPSERVER_FAILED;
- }
+ // Check if last stepServer failed, try to step now
+ if (mStepServerFailed) {
+ if (!step()) goto getNextBuffer_exit;
+ ALOGV("stepServer recovered");
+ mStepServerFailed = false;
+ }
- framesReady = cblk->framesReady();
+ framesReady = cblk->framesReady();
- if (LIKELY(framesReady)) {
+ if (CC_LIKELY(framesReady)) {
uint32_t s = cblk->server;
uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
@@ -3481,24 +3593,28 @@
framesReq = bufferEnd - s;
}
- buffer->raw = getBuffer(s, framesReq);
- if (buffer->raw == 0) goto getNextBuffer_exit;
+ buffer->raw = getBuffer(s, framesReq);
+ if (buffer->raw == NULL) goto getNextBuffer_exit;
- buffer->frameCount = framesReq;
+ buffer->frameCount = framesReq;
return NO_ERROR;
- }
+ }
getNextBuffer_exit:
- buffer->raw = 0;
- buffer->frameCount = 0;
- ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
- return NOT_ENOUGH_DATA;
+ buffer->raw = NULL;
+ buffer->frameCount = 0;
+ ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
+ return NOT_ENOUGH_DATA;
+}
+
+uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const {
+ return mCblk->framesReady();
}
bool AudioFlinger::PlaybackThread::Track::isReady() const {
if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true;
- if (mCblk->framesReady() >= mCblk->frameCount ||
+ if (framesReady() >= mCblk->frameCount ||
(mCblk->flags & CBLK_FORCEREADY_MSK)) {
mFillingUpStatus = FS_FILLED;
android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
@@ -3507,15 +3623,15 @@
return false;
}
-status_t AudioFlinger::PlaybackThread::Track::start()
+status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid)
{
status_t status = NO_ERROR;
- ALOGV("start(%d), calling thread %d session %d",
- mName, IPCThreadState::self()->getCallingPid(), mSessionId);
+ ALOGV("start(%d), calling pid %d session %d tid %d",
+ mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid);
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
- int state = mState;
+ track_state state = mState;
// here the track could be either new, or restarted
// in both cases "unstop" the track
if (mState == PAUSED) {
@@ -3528,15 +3644,15 @@
if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
thread->mLock.unlock();
- status = AudioSystem::startOutput(thread->id(),
- (audio_stream_type_t)mStreamType,
- mSessionId);
+ status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId);
thread->mLock.lock();
+#ifdef ADD_BATTERY_DATA
// to track the speaker usage
if (status == NO_ERROR) {
addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart);
}
+#endif
}
if (status == NO_ERROR) {
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
@@ -3552,11 +3668,11 @@
void AudioFlinger::PlaybackThread::Track::stop()
{
- ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+ ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
- int state = mState;
+ track_state state = mState;
if (mState > STOPPED) {
mState = STOPPED;
// If the track is not active (PAUSED and buffers full), flush buffers
@@ -3568,20 +3684,20 @@
}
if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
thread->mLock.unlock();
- AudioSystem::stopOutput(thread->id(),
- (audio_stream_type_t)mStreamType,
- mSessionId);
+ AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
thread->mLock.lock();
+#ifdef ADD_BATTERY_DATA
// to track the speaker usage
addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
+#endif
}
}
}
void AudioFlinger::PlaybackThread::Track::pause()
{
- ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+ ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
@@ -3590,13 +3706,13 @@
ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
if (!isOutputTrack()) {
thread->mLock.unlock();
- AudioSystem::stopOutput(thread->id(),
- (audio_stream_type_t)mStreamType,
- mSessionId);
+ AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
thread->mLock.lock();
+#ifdef ADD_BATTERY_DATA
// to track the speaker usage
addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
+#endif
}
}
}
@@ -3644,19 +3760,13 @@
mMute = muted;
}
-void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right)
-{
- mVolume[0] = left;
- mVolume[1] = right;
-}
-
status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
{
status_t status = DEAD_OBJECT;
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
- PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
- status = playbackThread->attachAuxEffect(this, EffectId);
+ PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+ status = playbackThread->attachAuxEffect(this, EffectId);
}
return status;
}
@@ -3667,31 +3777,422 @@
mAuxBuffer = buffer;
}
+// timed audio tracks
+
+sp<AudioFlinger::PlaybackThread::TimedTrack>
+AudioFlinger::PlaybackThread::TimedTrack::create(
+ PlaybackThread *thread,
+ const sp<Client>& client,
+ audio_stream_type_t streamType,
+ uint32_t sampleRate,
+ audio_format_t format,
+ uint32_t channelMask,
+ int frameCount,
+ const sp<IMemory>& sharedBuffer,
+ int sessionId) {
+ if (!client->reserveTimedTrack())
+ return NULL;
+
+ sp<TimedTrack> track = new TimedTrack(
+ thread, client, streamType, sampleRate, format, channelMask, frameCount,
+ sharedBuffer, sessionId);
+
+ if (track == NULL) {
+ client->releaseTimedTrack();
+ return NULL;
+ }
+
+ return track;
+}
+
+AudioFlinger::PlaybackThread::TimedTrack::TimedTrack(
+ PlaybackThread *thread,
+ const sp<Client>& client,
+ audio_stream_type_t streamType,
+ uint32_t sampleRate,
+ audio_format_t format,
+ uint32_t channelMask,
+ int frameCount,
+ const sp<IMemory>& sharedBuffer,
+ int sessionId)
+ : Track(thread, client, streamType, sampleRate, format, channelMask,
+ frameCount, sharedBuffer, sessionId),
+ mTimedSilenceBuffer(NULL),
+ mTimedSilenceBufferSize(0),
+ mTimedAudioOutputOnTime(false),
+ mMediaTimeTransformValid(false)
+{
+ LocalClock lc;
+ mLocalTimeFreq = lc.getLocalFreq();
+
+ mLocalTimeToSampleTransform.a_zero = 0;
+ mLocalTimeToSampleTransform.b_zero = 0;
+ mLocalTimeToSampleTransform.a_to_b_numer = sampleRate;
+ mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq;
+ LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer,
+ &mLocalTimeToSampleTransform.a_to_b_denom);
+}
+
+AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() {
+ mClient->releaseTimedTrack();
+ delete [] mTimedSilenceBuffer;
+}
+
+status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer(
+ size_t size, sp<IMemory>* buffer) {
+
+ Mutex::Autolock _l(mTimedBufferQueueLock);
+
+ trimTimedBufferQueue_l();
+
+ // lazily initialize the shared memory heap for timed buffers
+ if (mTimedMemoryDealer == NULL) {
+ const int kTimedBufferHeapSize = 512 << 10;
+
+ mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize,
+ "AudioFlingerTimed");
+ if (mTimedMemoryDealer == NULL)
+ return NO_MEMORY;
+ }
+
+ sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size);
+ if (newBuffer == NULL) {
+ newBuffer = mTimedMemoryDealer->allocate(size);
+ if (newBuffer == NULL)
+ return NO_MEMORY;
+ }
+
+ *buffer = newBuffer;
+ return NO_ERROR;
+}
+
+// caller must hold mTimedBufferQueueLock
+void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() {
+ int64_t mediaTimeNow;
+ {
+ Mutex::Autolock mttLock(mMediaTimeTransformLock);
+ if (!mMediaTimeTransformValid)
+ return;
+
+ int64_t targetTimeNow;
+ status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME)
+ ? mCCHelper.getCommonTime(&targetTimeNow)
+ : mCCHelper.getLocalTime(&targetTimeNow);
+
+ if (OK != res)
+ return;
+
+ if (!mMediaTimeTransform.doReverseTransform(targetTimeNow,
+ &mediaTimeNow)) {
+ return;
+ }
+ }
+
+ size_t trimIndex;
+ for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) {
+ if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow)
+ break;
+ }
+
+ if (trimIndex) {
+ mTimedBufferQueue.removeItemsAt(0, trimIndex);
+ }
+}
+
+status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer(
+ const sp<IMemory>& buffer, int64_t pts) {
+
+ {
+ Mutex::Autolock mttLock(mMediaTimeTransformLock);
+ if (!mMediaTimeTransformValid)
+ return INVALID_OPERATION;
+ }
+
+ Mutex::Autolock _l(mTimedBufferQueueLock);
+
+ mTimedBufferQueue.add(TimedBuffer(buffer, pts));
+
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform(
+ const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) {
+
+ ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__,
+ xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom,
+ target);
+
+ if (!(target == TimedAudioTrack::LOCAL_TIME ||
+ target == TimedAudioTrack::COMMON_TIME)) {
+ return BAD_VALUE;
+ }
+
+ Mutex::Autolock lock(mMediaTimeTransformLock);
+ mMediaTimeTransform = xform;
+ mMediaTimeTransformTarget = target;
+ mMediaTimeTransformValid = true;
+
+ return NO_ERROR;
+}
+
+#define min(a, b) ((a) < (b) ? (a) : (b))
+
+// implementation of getNextBuffer for tracks whose buffers have timestamps
+status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
+ AudioBufferProvider::Buffer* buffer, int64_t pts)
+{
+ if (pts == AudioBufferProvider::kInvalidPTS) {
+ buffer->raw = 0;
+ buffer->frameCount = 0;
+ return INVALID_OPERATION;
+ }
+
+ Mutex::Autolock _l(mTimedBufferQueueLock);
+
+ while (true) {
+
+ // if we have no timed buffers, then fail
+ if (mTimedBufferQueue.isEmpty()) {
+ buffer->raw = 0;
+ buffer->frameCount = 0;
+ return NOT_ENOUGH_DATA;
+ }
+
+ TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
+
+ // calculate the PTS of the head of the timed buffer queue expressed in
+ // local time
+ int64_t headLocalPTS;
+ {
+ Mutex::Autolock mttLock(mMediaTimeTransformLock);
+
+ ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid");
+
+ if (mMediaTimeTransform.a_to_b_denom == 0) {
+ // the transform represents a pause, so yield silence
+ timedYieldSilence(buffer->frameCount, buffer);
+ return NO_ERROR;
+ }
+
+ int64_t transformedPTS;
+ if (!mMediaTimeTransform.doForwardTransform(head.pts(),
+ &transformedPTS)) {
+ // the transform failed. this shouldn't happen, but if it does
+ // then just drop this buffer
+ ALOGW("timedGetNextBuffer transform failed");
+ buffer->raw = 0;
+ buffer->frameCount = 0;
+ mTimedBufferQueue.removeAt(0);
+ return NO_ERROR;
+ }
+
+ if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) {
+ if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS,
+ &headLocalPTS)) {
+ buffer->raw = 0;
+ buffer->frameCount = 0;
+ return INVALID_OPERATION;
+ }
+ } else {
+ headLocalPTS = transformedPTS;
+ }
+ }
+
+ // adjust the head buffer's PTS to reflect the portion of the head buffer
+ // that has already been consumed
+ int64_t effectivePTS = headLocalPTS +
+ ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate());
+
+ // Calculate the delta in samples between the head of the input buffer
+ // queue and the start of the next output buffer that will be written.
+ // If the transformation fails because of over or underflow, it means
+ // that the sample's position in the output stream is so far out of
+ // whack that it should just be dropped.
+ int64_t sampleDelta;
+ if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) {
+ ALOGV("*** head buffer is too far from PTS: dropped buffer");
+ mTimedBufferQueue.removeAt(0);
+ continue;
+ }
+ if (!mLocalTimeToSampleTransform.doForwardTransform(
+ (effectivePTS - pts) << 32, &sampleDelta)) {
+ ALOGV("*** too late during sample rate transform: dropped buffer");
+ mTimedBufferQueue.removeAt(0);
+ continue;
+ }
+
+ ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]",
+ __PRETTY_FUNCTION__, head.pts(), head.position(), pts,
+ static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)),
+ static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF));
+
+ // if the delta between the ideal placement for the next input sample and
+ // the current output position is within this threshold, then we will
+ // concatenate the next input samples to the previous output
+ const int64_t kSampleContinuityThreshold =
+ (static_cast<int64_t>(sampleRate()) << 32) / 10;
+
+ // if this is the first buffer of audio that we're emitting from this track
+ // then it should be almost exactly on time.
+ const int64_t kSampleStartupThreshold = 1LL << 32;
+
+ if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) ||
+ (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) {
+ // the next input is close enough to being on time, so concatenate it
+ // with the last output
+ timedYieldSamples(buffer);
+
+ ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
+ return NO_ERROR;
+ } else if (sampleDelta > 0) {
+ // the gap between the current output position and the proper start of
+ // the next input sample is too big, so fill it with silence
+ uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32;
+
+ timedYieldSilence(framesUntilNextInput, buffer);
+ ALOGV("*** silence: frameCount=%u", buffer->frameCount);
+ return NO_ERROR;
+ } else {
+ // the next input sample is late
+ uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32));
+ size_t onTimeSamplePosition =
+ head.position() + lateFrames * mCblk->frameSize;
+
+ if (onTimeSamplePosition > head.buffer()->size()) {
+ // all the remaining samples in the head are too late, so
+ // drop it and move on
+ ALOGV("*** too late: dropped buffer");
+ mTimedBufferQueue.removeAt(0);
+ continue;
+ } else {
+ // skip over the late samples
+ head.setPosition(onTimeSamplePosition);
+
+ // yield the available samples
+ timedYieldSamples(buffer);
+
+ ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
+ return NO_ERROR;
+ }
+ }
+ }
+}
+
+// Yield samples from the timed buffer queue head up to the given output
+// buffer's capacity.
+//
+// Caller must hold mTimedBufferQueueLock
+void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples(
+ AudioBufferProvider::Buffer* buffer) {
+
+ const TimedBuffer& head = mTimedBufferQueue[0];
+
+ buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) +
+ head.position());
+
+ uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) /
+ mCblk->frameSize);
+ size_t framesRequested = buffer->frameCount;
+ buffer->frameCount = min(framesLeftInHead, framesRequested);
+
+ mTimedAudioOutputOnTime = true;
+}
+
+// Yield samples of silence up to the given output buffer's capacity
+//
+// Caller must hold mTimedBufferQueueLock
+void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence(
+ uint32_t numFrames, AudioBufferProvider::Buffer* buffer) {
+
+ // lazily allocate a buffer filled with silence
+ if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) {
+ delete [] mTimedSilenceBuffer;
+ mTimedSilenceBufferSize = numFrames * mCblk->frameSize;
+ mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize];
+ memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize);
+ }
+
+ buffer->raw = mTimedSilenceBuffer;
+ size_t framesRequested = buffer->frameCount;
+ buffer->frameCount = min(numFrames, framesRequested);
+
+ mTimedAudioOutputOnTime = false;
+}
+
+// AudioBufferProvider interface
+void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer(
+ AudioBufferProvider::Buffer* buffer) {
+
+ Mutex::Autolock _l(mTimedBufferQueueLock);
+
+ // If the buffer which was just released is part of the buffer at the head
+ // of the queue, be sure to update the amt of the buffer which has been
+ // consumed. If the buffer being returned is not part of the head of the
+ // queue, its either because the buffer is part of the silence buffer, or
+ // because the head of the timed queue was trimmed after the mixer called
+ // getNextBuffer but before the mixer called releaseBuffer.
+ if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) {
+ TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
+
+ void* start = head.buffer()->pointer();
+ void* end = (char *) head.buffer()->pointer() + head.buffer()->size();
+
+ if ((buffer->raw >= start) && (buffer->raw <= end)) {
+ head.setPosition(head.position() +
+ (buffer->frameCount * mCblk->frameSize));
+ if (static_cast<size_t>(head.position()) >= head.buffer()->size()) {
+ mTimedBufferQueue.removeAt(0);
+ }
+ }
+ }
+
+ buffer->raw = 0;
+ buffer->frameCount = 0;
+}
+
+uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const {
+ Mutex::Autolock _l(mTimedBufferQueueLock);
+
+ uint32_t frames = 0;
+ for (size_t i = 0; i < mTimedBufferQueue.size(); i++) {
+ const TimedBuffer& tb = mTimedBufferQueue[i];
+ frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize;
+ }
+
+ return frames;
+}
+
+AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer()
+ : mPTS(0), mPosition(0) {}
+
+AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer(
+ const sp<IMemory>& buffer, int64_t pts)
+ : mBuffer(buffer), mPTS(pts), mPosition(0) {}
+
// ----------------------------------------------------------------------------
// RecordTrack constructor must be called with AudioFlinger::mLock held
AudioFlinger::RecordThread::RecordTrack::RecordTrack(
- const wp<ThreadBase>& thread,
+ RecordThread *thread,
const sp<Client>& client,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
- uint32_t flags,
int sessionId)
: TrackBase(thread, client, sampleRate, format,
- channelMask, frameCount, flags, 0, sessionId),
+ channelMask, frameCount, 0 /*sharedBuffer*/, sessionId),
mOverflow(false)
{
if (mCblk != NULL) {
- ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
- if (format == AUDIO_FORMAT_PCM_16_BIT) {
- mCblk->frameSize = mChannelCount * sizeof(int16_t);
- } else if (format == AUDIO_FORMAT_PCM_8_BIT) {
- mCblk->frameSize = mChannelCount * sizeof(int8_t);
- } else {
- mCblk->frameSize = sizeof(int8_t);
- }
+ ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
+ if (format == AUDIO_FORMAT_PCM_16_BIT) {
+ mCblk->frameSize = mChannelCount * sizeof(int16_t);
+ } else if (format == AUDIO_FORMAT_PCM_8_BIT) {
+ mCblk->frameSize = mChannelCount * sizeof(int8_t);
+ } else {
+ mCblk->frameSize = sizeof(int8_t);
+ }
}
}
@@ -3703,22 +4204,23 @@
}
}
-status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+// AudioBufferProvider interface
+status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts)
{
audio_track_cblk_t* cblk = this->cblk();
uint32_t framesAvail;
uint32_t framesReq = buffer->frameCount;
- // Check if last stepServer failed, try to step now
- if (mFlags & TrackBase::STEPSERVER_FAILED) {
+ // Check if last stepServer failed, try to step now
+ if (mStepServerFailed) {
if (!step()) goto getNextBuffer_exit;
ALOGV("stepServer recovered");
- mFlags &= ~TrackBase::STEPSERVER_FAILED;
+ mStepServerFailed = false;
}
framesAvail = cblk->framesAvailable_l();
- if (LIKELY(framesAvail)) {
+ if (CC_LIKELY(framesAvail)) {
uint32_t s = cblk->server;
uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
@@ -3730,24 +4232,24 @@
}
buffer->raw = getBuffer(s, framesReq);
- if (buffer->raw == 0) goto getNextBuffer_exit;
+ if (buffer->raw == NULL) goto getNextBuffer_exit;
buffer->frameCount = framesReq;
return NO_ERROR;
}
getNextBuffer_exit:
- buffer->raw = 0;
+ buffer->raw = NULL;
buffer->frameCount = 0;
return NOT_ENOUGH_DATA;
}
-status_t AudioFlinger::RecordThread::RecordTrack::start()
+status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid)
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
RecordThread *recordThread = (RecordThread *)thread.get();
- return recordThread->start(this);
+ return recordThread->start(this, tid);
} else {
return BAD_VALUE;
}
@@ -3769,7 +4271,7 @@
void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
{
snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n",
- (mClient == NULL) ? getpid() : mClient->pid(),
+ (mClient == 0) ? getpid_cached : mClient->pid(),
mFormat,
mChannelMask,
mSessionId,
@@ -3784,21 +4286,19 @@
// ----------------------------------------------------------------------------
AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
- const wp<ThreadBase>& thread,
+ PlaybackThread *playbackThread,
DuplicatingThread *sourceThread,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount)
- : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0),
+ : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0),
mActive(false), mSourceThread(sourceThread)
{
- PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get();
if (mCblk != NULL) {
mCblk->flags |= CBLK_DIRECTION_OUT;
mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
- mCblk->volume[0] = mCblk->volume[1] = 0x1000;
mOutBuffer.frameCount = 0;
playbackThread->mTracks.add(this);
ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \
@@ -3815,9 +4315,9 @@
clearBufferQueue();
}
-status_t AudioFlinger::PlaybackThread::OutputTrack::start()
+status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid)
{
- status_t status = Track::start();
+ status_t status = Track::start(tid);
if (status != NO_ERROR) {
return status;
}
@@ -3847,7 +4347,7 @@
uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
if (!mActive && frames != 0) {
- start();
+ start(0);
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
MixerThread *mixerThread = (MixerThread *)thread.get();
@@ -3882,7 +4382,7 @@
if (mOutBuffer.frameCount == 0) {
mOutBuffer.frameCount = pInBuffer->frameCount;
nsecs_t startTime = systemTime();
- if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) {
+ if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) {
ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get());
outputBufferFull = true;
break;
@@ -3971,13 +4471,13 @@
goto start_loop_here;
while (framesAvail == 0) {
active = mActive;
- if (UNLIKELY(!active)) {
+ if (CC_UNLIKELY(!active)) {
ALOGV("Not active and NO_MORE_BUFFERS");
- return AudioTrack::NO_MORE_BUFFERS;
+ return NO_MORE_BUFFERS;
}
result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
if (result != NO_ERROR) {
- return AudioTrack::NO_MORE_BUFFERS;
+ return NO_MORE_BUFFERS;
}
// read the server count again
start_loop_here:
@@ -3986,7 +4486,7 @@
}
// if (framesAvail < framesReq) {
-// return AudioTrack::NO_MORE_BUFFERS;
+// return NO_MORE_BUFFERS;
// }
if (framesReq > framesAvail) {
@@ -4009,10 +4509,9 @@
void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
{
size_t size = mBufferQueue.size();
- Buffer *pBuffer;
for (size_t i = 0; i < size; i++) {
- pBuffer = mBufferQueue.itemAt(i);
+ Buffer *pBuffer = mBufferQueue.itemAt(i);
delete [] pBuffer->mBuffer;
delete pBuffer;
}
@@ -4024,8 +4523,10 @@
AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
: RefBase(),
mAudioFlinger(audioFlinger),
+ // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below
mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
- mPid(pid)
+ mPid(pid),
+ mTimedTrackCount(0)
{
// 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
}
@@ -4036,31 +4537,53 @@
mAudioFlinger->removeClient_l(mPid);
}
-const sp<MemoryDealer>& AudioFlinger::Client::heap() const
+sp<MemoryDealer> AudioFlinger::Client::heap() const
{
return mMemoryDealer;
}
+// Reserve one of the limited slots for a timed audio track associated
+// with this client
+bool AudioFlinger::Client::reserveTimedTrack()
+{
+ const int kMaxTimedTracksPerClient = 4;
+
+ Mutex::Autolock _l(mTimedTrackLock);
+
+ if (mTimedTrackCount >= kMaxTimedTracksPerClient) {
+ ALOGW("can not create timed track - pid %d has exceeded the limit",
+ mPid);
+ return false;
+ }
+
+ mTimedTrackCount++;
+ return true;
+}
+
+// Release a slot for a timed audio track
+void AudioFlinger::Client::releaseTimedTrack()
+{
+ Mutex::Autolock _l(mTimedTrackLock);
+ mTimedTrackCount--;
+}
+
// ----------------------------------------------------------------------------
AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
const sp<IAudioFlingerClient>& client,
pid_t pid)
- : mAudioFlinger(audioFlinger), mPid(pid), mClient(client)
+ : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
{
}
AudioFlinger::NotificationClient::~NotificationClient()
{
- mClient.clear();
}
void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
{
sp<NotificationClient> keep(this);
- {
- mAudioFlinger->removeNotificationClient(mPid);
- }
+ mAudioFlinger->removeNotificationClient(mPid);
}
// ----------------------------------------------------------------------------
@@ -4079,8 +4602,12 @@
mTrack->destroy();
}
-status_t AudioFlinger::TrackHandle::start() {
- return mTrack->start();
+sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
+ return mTrack->getCblk();
+}
+
+status_t AudioFlinger::TrackHandle::start(pid_t tid) {
+ return mTrack->start(tid);
}
void AudioFlinger::TrackHandle::stop() {
@@ -4099,19 +4626,43 @@
mTrack->pause();
}
-void AudioFlinger::TrackHandle::setVolume(float left, float right) {
- mTrack->setVolume(left, right);
-}
-
-sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
- return mTrack->getCblk();
-}
-
status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
{
return mTrack->attachAuxEffect(EffectId);
}
+status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size,
+ sp<IMemory>* buffer) {
+ if (!mTrack->isTimedTrack())
+ return INVALID_OPERATION;
+
+ PlaybackThread::TimedTrack* tt =
+ reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
+ return tt->allocateTimedBuffer(size, buffer);
+}
+
+status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer,
+ int64_t pts) {
+ if (!mTrack->isTimedTrack())
+ return INVALID_OPERATION;
+
+ PlaybackThread::TimedTrack* tt =
+ reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
+ return tt->queueTimedBuffer(buffer, pts);
+}
+
+status_t AudioFlinger::TrackHandle::setMediaTimeTransform(
+ const LinearTransform& xform, int target) {
+
+ if (!mTrack->isTimedTrack())
+ return INVALID_OPERATION;
+
+ PlaybackThread::TimedTrack* tt =
+ reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
+ return tt->setMediaTimeTransform(
+ xform, static_cast<TimedAudioTrack::TargetTimeline>(target));
+}
+
status_t AudioFlinger::TrackHandle::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
@@ -4122,11 +4673,12 @@
sp<IAudioRecord> AudioFlinger::openRecord(
pid_t pid,
- int input,
+ audio_io_handle_t input,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
+ // FIXME dead, remove from IAudioFlinger
uint32_t flags,
int *sessionId,
status_t *status)
@@ -4134,7 +4686,6 @@
sp<RecordThread::RecordTrack> recordTrack;
sp<RecordHandle> recordHandle;
sp<Client> client;
- wp<Client> wclient;
status_t lStatus;
RecordThread *thread;
size_t inFrameCount;
@@ -4155,13 +4706,7 @@
goto Exit;
}
- wclient = mClients.valueFor(pid);
- if (wclient != NULL) {
- client = wclient.promote();
- } else {
- client = new Client(this, pid);
- mClients.add(pid, client);
- }
+ client = registerPid_l(pid);
// If no audio session id is provided, create one here
if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
@@ -4178,7 +4723,6 @@
format,
channelMask,
frameCount,
- flags,
lSessionId,
&lStatus);
}
@@ -4213,9 +4757,13 @@
stop();
}
-status_t AudioFlinger::RecordHandle::start() {
+sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
+ return mRecordTrack->getCblk();
+}
+
+status_t AudioFlinger::RecordHandle::start(pid_t tid) {
ALOGV("RecordHandle::start()");
- return mRecordTrack->start();
+ return mRecordTrack->start(tid);
}
void AudioFlinger::RecordHandle::stop() {
@@ -4223,10 +4771,6 @@
mRecordTrack->stop();
}
-sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
- return mRecordTrack->getCblk();
-}
-
status_t AudioFlinger::RecordHandle::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
@@ -4239,17 +4783,18 @@
AudioStreamIn *input,
uint32_t sampleRate,
uint32_t channels,
- int id,
+ audio_io_handle_t id,
uint32_t device) :
- ThreadBase(audioFlinger, id, device),
- mInput(input), mTrack(NULL), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0)
+ ThreadBase(audioFlinger, id, device, RECORD),
+ mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL),
+ // mRsmpInIndex and mInputBytes set by readInputParameters()
+ mReqChannelCount(popcount(channels)),
+ mReqSampleRate(sampleRate)
+ // mBytesRead is only meaningful while active, and so is cleared in start()
+ // (but might be better to also clear here for dump?)
{
- mType = ThreadBase::RECORD;
+ snprintf(mName, kNameLength, "AudioIn_%X", id);
- snprintf(mName, kNameLength, "AudioIn_%d", id);
-
- mReqChannelCount = popcount(channels);
- mReqSampleRate = sampleRate;
readInputParameters();
}
@@ -4257,10 +4802,8 @@
AudioFlinger::RecordThread::~RecordThread()
{
delete[] mRsmpInBuffer;
- if (mResampler != 0) {
- delete mResampler;
- delete[] mRsmpOutBuffer;
- }
+ delete mResampler;
+ delete[] mRsmpOutBuffer;
}
void AudioFlinger::RecordThread::onFirstRef()
@@ -4349,9 +4892,9 @@
}
buffer.frameCount = mFrameCount;
- if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
+ if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
size_t framesOut = buffer.frameCount;
- if (mResampler == 0) {
+ if (mResampler == NULL) {
// no resampling
while (framesOut) {
size_t framesIn = mFrameCount - mRsmpInIndex;
@@ -4416,7 +4959,7 @@
// ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer()
// are 32 bit aligned which should be always true.
if (mChannelCount == 2 && mReqChannelCount == 1) {
- AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
+ ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
// the resampler always outputs stereo samples: do post stereo to mono conversion
int16_t *src = (int16_t *)mRsmpOutBuffer;
int16_t *dst = buffer.i16;
@@ -4425,7 +4968,7 @@
src += 2;
}
} else {
- AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
+ ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
}
}
@@ -4436,7 +4979,7 @@
else {
if (!mActiveTrack->setOverflow()) {
nsecs_t now = systemTime();
- if ((now - lastWarning) > kWarningThrottle) {
+ if ((now - lastWarning) > kWarningThrottleNs) {
ALOGW("RecordThread: buffer overflow");
lastWarning = now;
}
@@ -4469,10 +5012,9 @@
sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l(
const sp<AudioFlinger::Client>& client,
uint32_t sampleRate,
- int format,
+ audio_format_t format,
int channelMask,
int frameCount,
- uint32_t flags,
int sessionId,
status_t *status)
{
@@ -4489,9 +5031,9 @@
Mutex::Autolock _l(mLock);
track = new RecordTrack(this, client, sampleRate,
- format, channelMask, frameCount, flags, sessionId);
+ format, channelMask, frameCount, sessionId);
- if (track->getCblk() == NULL) {
+ if (track->getCblk() == 0) {
lStatus = NO_MEMORY;
goto Exit;
}
@@ -4512,13 +5054,13 @@
return track;
}
-status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack)
+status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid)
{
- ALOGV("RecordThread::start");
- sp <ThreadBase> strongMe = this;
+ ALOGV("RecordThread::start tid=%d", tid);
+ sp<ThreadBase> strongMe = this;
status_t status = NO_ERROR;
{
- AutoMutex lock(&mLock);
+ AutoMutex lock(mLock);
if (mActiveTrack != 0) {
if (recordTrack != mActiveTrack.get()) {
status = -EBUSY;
@@ -4547,7 +5089,7 @@
ALOGV("Signal record thread");
mWaitWorkCV.signal();
// do not wait for mStartStopCond if exiting
- if (mExiting) {
+ if (exitPending()) {
mActiveTrack.clear();
status = INVALID_OPERATION;
goto startError;
@@ -4568,13 +5110,13 @@
void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
ALOGV("RecordThread::stop");
- sp <ThreadBase> strongMe = this;
+ sp<ThreadBase> strongMe = this;
{
- AutoMutex lock(&mLock);
+ AutoMutex lock(mLock);
if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) {
mActiveTrack->mState = TrackBase::PAUSING;
// do not wait for mStartStopCond if exiting
- if (mExiting) {
+ if (exitPending()) {
return;
}
mStartStopCond.wait(mLock);
@@ -4594,7 +5136,6 @@
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
- pid_t pid = 0;
snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
result.append(buffer);
@@ -4609,7 +5150,7 @@
result.append(buffer);
snprintf(buffer, SIZE, "In size: %d\n", mInputBytes);
result.append(buffer);
- snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0));
+ snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL));
result.append(buffer);
snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount);
result.append(buffer);
@@ -4628,7 +5169,8 @@
return NO_ERROR;
}
-status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+// AudioBufferProvider interface
+status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts)
{
size_t framesReq = buffer->frameCount;
size_t framesReady = mFrameCount - mRsmpInIndex;
@@ -4644,7 +5186,7 @@
mInput->stream->common.standby(&mInput->stream->common);
usleep(kRecordThreadSleepUs);
}
- buffer->raw = 0;
+ buffer->raw = NULL;
buffer->frameCount = 0;
return NOT_ENOUGH_DATA;
}
@@ -4666,6 +5208,7 @@
return NO_ERROR;
}
+// AudioBufferProvider interface
void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
{
mRsmpInIndex += buffer->frameCount;
@@ -4681,7 +5224,7 @@
String8 keyValuePair = mNewParameters[0];
AudioParameter param = AudioParameter(keyValuePair);
int value;
- int reqFormat = mFormat;
+ audio_format_t reqFormat = mFormat;
int reqSamplingRate = mReqSampleRate;
int reqChannelCount = mReqChannelCount;
@@ -4690,7 +5233,7 @@
reconfig = true;
}
if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
- reqFormat = value;
+ reqFormat = (audio_format_t) value;
reconfig = true;
}
if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
@@ -4699,7 +5242,7 @@
}
if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
// do not accept frame count changes if tracks are open as the track buffer
- // size depends on frame count and correct behavior would not be garantied
+ // size depends on frame count and correct behavior would not be guaranteed
// if frame count is changed after track creation
if (mActiveTrack != 0) {
status = INVALID_OPERATION;
@@ -4734,16 +5277,17 @@
if (status == NO_ERROR) {
status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
if (status == INVALID_OPERATION) {
- mInput->stream->common.standby(&mInput->stream->common);
- status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
+ mInput->stream->common.standby(&mInput->stream->common);
+ status = mInput->stream->common.set_parameters(&mInput->stream->common,
+ keyValuePair.string());
}
if (reconfig) {
if (status == BAD_VALUE &&
reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) &&
- (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) &&
- (reqChannelCount < 3)) {
+ popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 &&
+ (reqChannelCount <= FCC_2)) {
status = NO_ERROR;
}
if (status == NO_ERROR) {
@@ -4759,7 +5303,7 @@
mParamCond.signal();
// wait for condition with time out in case the thread calling ThreadBase::setParameters()
// already timed out waiting for the status and will never signal the condition.
- mWaitWorkCV.waitRelative(mLock, kSetParametersTimeout);
+ mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
}
return reconfig;
}
@@ -4782,7 +5326,7 @@
void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
AudioSystem::OutputDescriptor desc;
- void *param2 = 0;
+ void *param2 = NULL;
switch (event) {
case AudioSystem::INPUT_OPENED:
@@ -4804,25 +5348,27 @@
void AudioFlinger::RecordThread::readInputParameters()
{
- if (mRsmpInBuffer) delete mRsmpInBuffer;
- if (mRsmpOutBuffer) delete mRsmpOutBuffer;
- if (mResampler) delete mResampler;
- mResampler = 0;
+ delete mRsmpInBuffer;
+ // mRsmpInBuffer is always assigned a new[] below
+ delete mRsmpOutBuffer;
+ mRsmpOutBuffer = NULL;
+ delete mResampler;
+ mResampler = NULL;
mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
mChannelCount = (uint16_t)popcount(mChannelMask);
mFormat = mInput->stream->common.get_format(&mInput->stream->common);
- mFrameSize = (uint16_t)audio_stream_frame_size(&mInput->stream->common);
+ mFrameSize = audio_stream_frame_size(&mInput->stream->common);
mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common);
mFrameCount = mInputBytes / mFrameSize;
mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
- if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3)
+ if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2)
{
int channelCount;
- // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid
- // stereo to mono post process as the resampler always outputs stereo.
+ // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid
+ // stereo to mono post process as the resampler always outputs stereo.
if (mChannelCount == 1 && mReqChannelCount == 2) {
channelCount = 1;
} else {
@@ -4873,7 +5419,7 @@
return mTrack;
}
-AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput()
+AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const
{
Mutex::Autolock _l(mLock);
return mInput;
@@ -4899,18 +5445,17 @@
// ----------------------------------------------------------------------------
-int AudioFlinger::openOutput(uint32_t *pDevices,
+audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices,
uint32_t *pSamplingRate,
- uint32_t *pFormat,
+ audio_format_t *pFormat,
uint32_t *pChannels,
uint32_t *pLatencyMs,
- uint32_t flags)
+ audio_policy_output_flags_t flags)
{
status_t status;
PlaybackThread *thread = NULL;
- mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
- uint32_t format = pFormat ? *pFormat : 0;
+ audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT;
uint32_t channels = pChannels ? *pChannels : 0;
uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
audio_stream_out_t *outStream;
@@ -4933,8 +5478,10 @@
if (outHwDev == NULL)
return 0;
- status = outHwDev->open_output_stream(outHwDev, *pDevices, (int *)&format,
+ mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
+ status = outHwDev->open_output_stream(outHwDev, *pDevices, &format,
&channels, &samplingRate, &outStream);
+ mHardwareStatus = AUDIO_HW_IDLE;
ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
outStream,
samplingRate,
@@ -4942,10 +5489,9 @@
channels,
status);
- mHardwareStatus = AUDIO_HW_IDLE;
if (outStream != NULL) {
AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream);
- int id = nextUniqueId();
+ audio_io_handle_t id = nextUniqueId();
if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) ||
(format != AUDIO_FORMAT_PCM_16_BIT) ||
@@ -4958,10 +5504,10 @@
}
mPlaybackThreads.add(id, thread);
- if (pSamplingRate) *pSamplingRate = samplingRate;
- if (pFormat) *pFormat = format;
- if (pChannels) *pChannels = channels;
- if (pLatencyMs) *pLatencyMs = thread->latency();
+ if (pSamplingRate != NULL) *pSamplingRate = samplingRate;
+ if (pFormat != NULL) *pFormat = format;
+ if (pChannels != NULL) *pChannels = channels;
+ if (pLatencyMs != NULL) *pLatencyMs = thread->latency();
// notify client processes of the new output creation
thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
@@ -4971,7 +5517,8 @@
return 0;
}
-int AudioFlinger::openDuplicateOutput(int output1, int output2)
+audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
+ audio_io_handle_t output2)
{
Mutex::Autolock _l(mLock);
MixerThread *thread1 = checkMixerThread_l(output1);
@@ -4982,7 +5529,7 @@
return 0;
}
- int id = nextUniqueId();
+ audio_io_handle_t id = nextUniqueId();
DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
thread->addOutputTrack(thread2);
mPlaybackThreads.add(id, thread);
@@ -4991,11 +5538,11 @@
return id;
}
-status_t AudioFlinger::closeOutput(int output)
+status_t AudioFlinger::closeOutput(audio_io_handle_t output)
{
// keep strong reference on the playback thread so that
// it is not destroyed while exit() is executed
- sp <PlaybackThread> thread;
+ sp<PlaybackThread> thread;
{
Mutex::Autolock _l(mLock);
thread = checkPlaybackThread_l(output);
@@ -5013,14 +5560,16 @@
}
}
}
- void *param2 = 0;
- audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2);
+ audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL);
mPlaybackThreads.removeItem(output);
}
thread->exit();
+ // The thread entity (active unit of execution) is no longer running here,
+ // but the ThreadBase container still exists.
if (thread->type() != ThreadBase::DUPLICATING) {
AudioStreamOut *out = thread->clearOutput();
+ ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
// from now on thread->mOutput is NULL
out->hwDev->close_output_stream(out->hwDev, out->stream);
delete out;
@@ -5028,7 +5577,7 @@
return NO_ERROR;
}
-status_t AudioFlinger::suspendOutput(int output)
+status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
@@ -5043,7 +5592,7 @@
return NO_ERROR;
}
-status_t AudioFlinger::restoreOutput(int output)
+status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
@@ -5059,19 +5608,19 @@
return NO_ERROR;
}
-int AudioFlinger::openInput(uint32_t *pDevices,
+audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices,
uint32_t *pSamplingRate,
- uint32_t *pFormat,
+ audio_format_t *pFormat,
uint32_t *pChannels,
- uint32_t acoustics)
+ audio_in_acoustics_t acoustics)
{
status_t status;
RecordThread *thread = NULL;
uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
- uint32_t format = pFormat ? *pFormat : 0;
+ audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT;
uint32_t channels = pChannels ? *pChannels : 0;
uint32_t reqSamplingRate = samplingRate;
- uint32_t reqFormat = format;
+ audio_format_t reqFormat = format;
uint32_t reqChannels = channels;
audio_stream_in_t *inStream;
audio_hw_device_t *inHwDev;
@@ -5086,9 +5635,9 @@
if (inHwDev == NULL)
return 0;
- status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format,
+ status = inHwDev->open_input_stream(inHwDev, *pDevices, &format,
&channels, &samplingRate,
- (audio_in_acoustics_t)acoustics,
+ acoustics,
&inStream);
ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d",
inStream,
@@ -5104,18 +5653,18 @@
if (inStream == NULL && status == BAD_VALUE &&
reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT &&
(samplingRate <= 2 * reqSamplingRate) &&
- (popcount(channels) < 3) && (popcount(reqChannels) < 3)) {
+ (popcount(channels) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) {
ALOGV("openInput() reopening with proposed sampling rate and channels");
- status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format,
+ status = inHwDev->open_input_stream(inHwDev, *pDevices, &format,
&channels, &samplingRate,
- (audio_in_acoustics_t)acoustics,
+ acoustics,
&inStream);
}
if (inStream != NULL) {
AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
- int id = nextUniqueId();
+ audio_io_handle_t id = nextUniqueId();
// Start record thread
// RecorThread require both input and output device indication to forward to audio
// pre processing modules
@@ -5128,9 +5677,9 @@
device);
mRecordThreads.add(id, thread);
ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
- if (pSamplingRate) *pSamplingRate = reqSamplingRate;
- if (pFormat) *pFormat = format;
- if (pChannels) *pChannels = reqChannels;
+ if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate;
+ if (pFormat != NULL) *pFormat = format;
+ if (pChannels != NULL) *pChannels = reqChannels;
input->stream->common.standby(&input->stream->common);
@@ -5142,11 +5691,11 @@
return 0;
}
-status_t AudioFlinger::closeInput(int input)
+status_t AudioFlinger::closeInput(audio_io_handle_t input)
{
// keep strong reference on the record thread so that
// it is not destroyed while exit() is executed
- sp <RecordThread> thread;
+ sp<RecordThread> thread;
{
Mutex::Autolock _l(mLock);
thread = checkRecordThread_l(input);
@@ -5155,13 +5704,15 @@
}
ALOGV("closeInput() %d", input);
- void *param2 = 0;
- audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2);
+ audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL);
mRecordThreads.removeItem(input);
}
thread->exit();
+ // The thread entity (active unit of execution) is no longer running here,
+ // but the ThreadBase container still exists.
AudioStreamIn *in = thread->clearInput();
+ ALOG_ASSERT(in != NULL, "in shouldn't be NULL");
// from now on thread->mInput is NULL
in->hwDev->close_input_stream(in->hwDev, in->stream);
delete in;
@@ -5169,7 +5720,7 @@
return NO_ERROR;
}
-status_t AudioFlinger::setStreamOutput(uint32_t stream, int output)
+status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output)
{
Mutex::Autolock _l(mLock);
MixerThread *dstThread = checkMixerThread_l(output);
@@ -5185,8 +5736,7 @@
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
- if (thread != dstThread &&
- thread->type() != ThreadBase::DIRECT) {
+ if (thread != dstThread && thread->type() != ThreadBase::DIRECT) {
MixerThread *srcThread = (MixerThread *)thread;
srcThread->setStreamValid(stream, false);
srcThread->invalidateTracks(stream);
@@ -5205,37 +5755,33 @@
void AudioFlinger::acquireAudioSessionId(int audioSession)
{
Mutex::Autolock _l(mLock);
- int caller = IPCThreadState::self()->getCallingPid();
+ pid_t caller = IPCThreadState::self()->getCallingPid();
ALOGV("acquiring %d from %d", audioSession, caller);
- int num = mAudioSessionRefs.size();
- for (int i = 0; i< num; i++) {
+ size_t num = mAudioSessionRefs.size();
+ for (size_t i = 0; i< num; i++) {
AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
- if (ref->sessionid == audioSession && ref->pid == caller) {
- ref->cnt++;
- ALOGV(" incremented refcount to %d", ref->cnt);
+ if (ref->mSessionid == audioSession && ref->mPid == caller) {
+ ref->mCnt++;
+ ALOGV(" incremented refcount to %d", ref->mCnt);
return;
}
}
- AudioSessionRef *ref = new AudioSessionRef();
- ref->sessionid = audioSession;
- ref->pid = caller;
- ref->cnt = 1;
- mAudioSessionRefs.push(ref);
- ALOGV(" added new entry for %d", ref->sessionid);
+ mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
+ ALOGV(" added new entry for %d", audioSession);
}
void AudioFlinger::releaseAudioSessionId(int audioSession)
{
Mutex::Autolock _l(mLock);
- int caller = IPCThreadState::self()->getCallingPid();
+ pid_t caller = IPCThreadState::self()->getCallingPid();
ALOGV("releasing %d from %d", audioSession, caller);
- int num = mAudioSessionRefs.size();
- for (int i = 0; i< num; i++) {
+ size_t num = mAudioSessionRefs.size();
+ for (size_t i = 0; i< num; i++) {
AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
- if (ref->sessionid == audioSession && ref->pid == caller) {
- ref->cnt--;
- ALOGV(" decremented refcount to %d", ref->cnt);
- if (ref->cnt == 0) {
+ if (ref->mSessionid == audioSession && ref->mPid == caller) {
+ ref->mCnt--;
+ ALOGV(" decremented refcount to %d", ref->mCnt);
+ if (ref->mCnt == 0) {
mAudioSessionRefs.removeAt(i);
delete ref;
purgeStaleEffects_l();
@@ -5280,9 +5826,9 @@
bool found = false;
for (size_t k = 0; k < numsessionrefs; k++) {
AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
- if (ref->sessionid == sessionid) {
+ if (ref->mSessionid == sessionid) {
ALOGV(" session %d still exists for %d with %d refs",
- sessionid, ref->pid, ref->cnt);
+ sessionid, ref->mPid, ref->mCnt);
found = true;
break;
}
@@ -5311,35 +5857,22 @@
}
// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
-AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const
+AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
{
- PlaybackThread *thread = NULL;
- if (mPlaybackThreads.indexOfKey(output) >= 0) {
- thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get();
- }
- return thread;
+ return mPlaybackThreads.valueFor(output).get();
}
// checkMixerThread_l() must be called with AudioFlinger::mLock held
-AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const
+AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
{
PlaybackThread *thread = checkPlaybackThread_l(output);
- if (thread != NULL) {
- if (thread->type() == ThreadBase::DIRECT) {
- thread = NULL;
- }
- }
- return (MixerThread *)thread;
+ return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
}
// checkRecordThread_l() must be called with AudioFlinger::mLock held
-AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const
+AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
{
- RecordThread *thread = NULL;
- if (mRecordThreads.indexOfKey(input) >= 0) {
- thread = (RecordThread *)mRecordThreads.valueFor(input).get();
- }
- return thread;
+ return mRecordThreads.valueFor(input).get();
}
uint32_t AudioFlinger::nextUniqueId()
@@ -5347,7 +5880,7 @@
return android_atomic_inc(&mNextUniqueId);
}
-AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l()
+AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
{
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
@@ -5359,7 +5892,7 @@
return NULL;
}
-uint32_t AudioFlinger::primaryOutputDevice_l()
+uint32_t AudioFlinger::primaryOutputDevice_l() const
{
PlaybackThread *thread = primaryPlaybackThread_l();
@@ -5376,19 +5909,20 @@
// ----------------------------------------------------------------------------
-status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects)
+status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
{
Mutex::Autolock _l(mLock);
return EffectQueryNumberEffects(numEffects);
}
-status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor)
+status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
{
Mutex::Autolock _l(mLock);
return EffectQueryEffect(index, descriptor);
}
-status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor)
+status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
+ effect_descriptor_t *descriptor) const
{
Mutex::Autolock _l(mLock);
return EffectGetDescriptor(pUuid, descriptor);
@@ -5399,7 +5933,7 @@
effect_descriptor_t *pDesc,
const sp<IEffectClient>& effectClient,
int32_t priority,
- int io,
+ audio_io_handle_t io,
int sessionId,
status_t *status,
int *id,
@@ -5408,10 +5942,8 @@
status_t lStatus = NO_ERROR;
sp<EffectHandle> handle;
effect_descriptor_t desc;
- sp<Client> client;
- wp<Client> wclient;
- ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d",
+ ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d",
pid, effectClient.get(), priority, sessionId, io);
if (pDesc == NULL) {
@@ -5427,7 +5959,7 @@
// Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
// that can only be created by audio policy manager (running in same process)
- if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) {
+ if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
lStatus = PERMISSION_DENIED;
goto Exit;
}
@@ -5529,7 +6061,7 @@
// because of code checking output when entering the function.
// Note: io is never 0 when creating an effect on an input
if (io == 0) {
- // look for the thread where the specified audio session is present
+ // look for the thread where the specified audio session is present
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
io = mPlaybackThreads.keyAt(i);
@@ -5537,12 +6069,12 @@
}
}
if (io == 0) {
- for (size_t i = 0; i < mRecordThreads.size(); i++) {
- if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
- io = mRecordThreads.keyAt(i);
- break;
- }
- }
+ for (size_t i = 0; i < mRecordThreads.size(); i++) {
+ if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
+ io = mRecordThreads.keyAt(i);
+ break;
+ }
+ }
}
// If no output thread contains the requested session ID, default to
// first output. The effect chain will be moved to the correct output
@@ -5562,14 +6094,7 @@
}
}
- wclient = mClients.valueFor(pid);
-
- if (wclient != NULL) {
- client = wclient.promote();
- } else {
- client = new Client(this, pid);
- mClients.add(pid, client);
- }
+ sp<Client> client = registerPid_l(pid);
// create effect on selected output thread
handle = thread->createEffect_l(client, effectClient, priority, sessionId,
@@ -5580,13 +6105,14 @@
}
Exit:
- if(status) {
+ if (status != NULL) {
*status = lStatus;
}
return handle;
}
-status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput)
+status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput,
+ audio_io_handle_t dstOutput)
{
ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
sessionId, srcOutput, dstOutput);
@@ -5637,7 +6163,7 @@
// transfer all effects one by one so that new effect chain is created on new thread with
// correct buffer sizes and audio parameters and effect engines reconfigured accordingly
- int dstOutput = dstThread->id();
+ audio_io_handle_t dstOutput = dstThread->id();
sp<EffectChain> dstChain;
uint32_t strategy = 0; // prevent compiler warning
sp<EffectModule> effect = chain->getEffectFromId_l(0);
@@ -5709,10 +6235,7 @@
goto Exit;
}
// Only Pre processor effects are allowed on input threads and only on input threads
- if ((mType == RECORD &&
- (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) ||
- (mType != RECORD &&
- (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) {
+ if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) {
ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d",
desc->name, desc->flags, mType);
lStatus = BAD_VALUE;
@@ -5737,7 +6260,7 @@
effect = chain->getEffectFromDesc_l(desc);
}
- ALOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get());
+ ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get());
if (effect == 0) {
int id = mAudioFlinger->nextUniqueId();
@@ -5765,7 +6288,7 @@
// create effect handle and connect it to effect module
handle = new EffectHandle(effect, client, effectClient, priority);
lStatus = effect->addHandle(handle);
- if (enabled) {
+ if (enabled != NULL) {
*enabled = (int)effect->isEnabled();
}
}
@@ -5785,7 +6308,7 @@
handle.clear();
}
- if(status) {
+ if (status != NULL) {
*status = lStatus;
}
return handle;
@@ -5793,13 +6316,8 @@
sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId)
{
- sp<EffectModule> effect;
-
sp<EffectChain> chain = getEffectChain_l(sessionId);
- if (chain != 0) {
- effect = chain->getEffectFromId_l(effectId);
- }
- return effect;
+ return chain != 0 ? chain->getEffectFromId_l(effectId) : 0;
}
// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
@@ -5860,7 +6378,7 @@
}
void AudioFlinger::ThreadBase::lockEffectChains_l(
- Vector<sp <AudioFlinger::EffectChain> >& effectChains)
+ Vector< sp<AudioFlinger::EffectChain> >& effectChains)
{
effectChains = mEffectChains;
for (size_t i = 0; i < mEffectChains.size(); i++) {
@@ -5869,7 +6387,7 @@
}
void AudioFlinger::ThreadBase::unlockEffectChains(
- Vector<sp <AudioFlinger::EffectChain> >& effectChains)
+ const Vector< sp<AudioFlinger::EffectChain> >& effectChains)
{
for (size_t i = 0; i < effectChains.size(); i++) {
effectChains[i]->unlock();
@@ -5884,19 +6402,16 @@
sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId)
{
- sp<EffectChain> chain;
-
size_t size = mEffectChains.size();
for (size_t i = 0; i < size; i++) {
if (mEffectChains[i]->sessionId() == sessionId) {
- chain = mEffectChains[i];
- break;
+ return mEffectChains[i];
}
}
- return chain;
+ return 0;
}
-void AudioFlinger::ThreadBase::setMode(uint32_t mode)
+void AudioFlinger::ThreadBase::setMode(audio_mode_t mode)
{
Mutex::Autolock _l(mLock);
size_t size = mEffectChains.size();
@@ -5907,13 +6422,13 @@
void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect,
const wp<EffectHandle>& handle,
- bool unpiniflast) {
+ bool unpinIfLast) {
Mutex::Autolock _l(mLock);
ALOGV("disconnectEffect() %p effect %p", this, effect.get());
// delete the effect module if removing last handle on it
if (effect->removeHandle(handle) == 0) {
- if (!effect->isPinned() || unpiniflast) {
+ if (!effect->isPinned() || unpinIfLast) {
removeEffect_l(effect);
AudioSystem::unregisterEffect(effect->id());
}
@@ -6048,7 +6563,7 @@
void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
{
- for (size_t i = 0; i < mTracks.size(); ++i) {
+ for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> track = mTracks[i];
if (track->auxEffectId() == effectId) {
attachAuxEffect_l(track, 0);
@@ -6093,18 +6608,17 @@
#undef LOG_TAG
#define LOG_TAG "AudioFlinger::EffectModule"
-AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread,
+AudioFlinger::EffectModule::EffectModule(ThreadBase *thread,
const wp<AudioFlinger::EffectChain>& chain,
effect_descriptor_t *desc,
int id,
int sessionId)
- : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL),
+ : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL),
mStatus(NO_INIT), mState(IDLE), mSuspended(false)
{
ALOGV("Constructor %p", this);
int lStatus;
- sp<ThreadBase> thread = mThread.promote();
- if (thread == 0) {
+ if (thread == NULL) {
return;
}
@@ -6152,12 +6666,11 @@
}
}
-status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle)
+status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle)
{
status_t status;
Mutex::Autolock _l(mLock);
- // First handle in mHandles has highest priority and controls the effect module
int priority = handle->priority();
size_t size = mHandles.size();
sp<EffectHandle> h;
@@ -6199,7 +6712,7 @@
bool enabled = false;
EffectHandle *hdl = handle.unsafe_get();
- if (hdl) {
+ if (hdl != NULL) {
ALOGV("removeHandle() unsafe_get OK");
enabled = hdl->enabled();
}
@@ -6226,23 +6739,19 @@
sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle()
{
Mutex::Autolock _l(mLock);
- sp<EffectHandle> handle;
- if (mHandles.size() != 0) {
- handle = mHandles[0].promote();
- }
- return handle;
+ return mHandles.size() != 0 ? mHandles[0].promote() : 0;
}
-void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast)
+void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast)
{
- ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get());
+ ALOGV("disconnect() %p handle %p", this, handle.unsafe_get());
// keep a strong reference on this EffectModule to avoid calling the
// destructor before we exit
sp<EffectModule> keep(this);
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
- thread->disconnectEffect(keep, handle, unpiniflast);
+ thread->disconnectEffect(keep, handle, unpinIfLast);
}
}
}
@@ -6296,7 +6805,7 @@
if (isProcessEnabled()) {
// do 32 bit to 16 bit conversion for auxiliary effect input buffer
if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
- AudioMixer::ditherAndClamp(mConfig.inputCfg.buffer.s32,
+ ditherAndClamp(mConfig.inputCfg.buffer.s32,
mConfig.inputCfg.buffer.s32,
mConfig.inputCfg.buffer.frameCount/2);
}
@@ -6401,7 +6910,7 @@
status_t cmdStatus;
uint32_t size = sizeof(int);
status_t status = (*mEffectInterface)->command(mEffectInterface,
- EFFECT_CMD_CONFIGURE,
+ EFFECT_CMD_SET_CONFIG,
sizeof(effect_config_t),
&mConfig,
&size,
@@ -6585,7 +7094,7 @@
return NO_ERROR;
}
-bool AudioFlinger::EffectModule::isEnabled()
+bool AudioFlinger::EffectModule::isEnabled() const
{
switch (mState) {
case RESTART:
@@ -6601,7 +7110,7 @@
}
}
-bool AudioFlinger::EffectModule::isProcessEnabled()
+bool AudioFlinger::EffectModule::isProcessEnabled() const
{
switch (mState) {
case RESTART:
@@ -6694,7 +7203,7 @@
return status;
}
-status_t AudioFlinger::EffectModule::setMode(uint32_t mode)
+status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode)
{
Mutex::Autolock _l(mLock);
status_t status = NO_ERROR;
@@ -6703,7 +7212,7 @@
uint32_t size = sizeof(status_t);
status = (*mEffectInterface)->command(mEffectInterface,
EFFECT_CMD_SET_AUDIO_MODE,
- sizeof(int),
+ sizeof(audio_mode_t),
&mode,
&size,
&cmdStatus);
@@ -6719,7 +7228,8 @@
Mutex::Autolock _l(mLock);
mSuspended = suspended;
}
-bool AudioFlinger::EffectModule::suspended()
+
+bool AudioFlinger::EffectModule::suspended() const
{
Mutex::Autolock _l(mLock);
return mSuspended;
@@ -6834,10 +7344,10 @@
if (mCblkMemory != 0) {
mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer());
- if (mCblk) {
+ if (mCblk != NULL) {
new(mCblk) effect_param_cblk_t();
mBuffer = (uint8_t *)mCblk + bufOffset;
- }
+ }
} else {
ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t));
return;
@@ -6913,13 +7423,13 @@
disconnect(true);
}
-void AudioFlinger::EffectHandle::disconnect(bool unpiniflast)
+void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast)
{
- ALOGV("disconnect(%s)", unpiniflast ? "true" : "false");
+ ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false");
if (mEffect == 0) {
return;
}
- mEffect->disconnect(this, unpiniflast);
+ mEffect->disconnect(this, unpinIfLast);
if (mHasControl && mEnabled) {
sp<ThreadBase> thread = mEffect->thread().promote();
@@ -6931,10 +7441,12 @@
// release sp on module => module destructor can be called now
mEffect.clear();
if (mClient != 0) {
- if (mCblk) {
+ if (mCblk != NULL) {
+ // unlike ~TrackBase(), mCblk is never a local new, so don't delete
mCblk->~effect_param_cblk_t(); // destroy our shared-structure.
}
- mCblkMemory.clear(); // and free the shared memory
+ mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to
+ // Client destructor must run with AudioFlinger mutex locked
Mutex::Autolock _l(mClient->audioFlinger()->mLock);
mClient.clear();
}
@@ -7016,10 +7528,6 @@
return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
}
-sp<IMemory> AudioFlinger::EffectHandle::getCblk() const {
- return mCblkMemory;
-}
-
void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled)
{
ALOGV("setControl %p control %d", this, hasControl);
@@ -7061,10 +7569,10 @@
void AudioFlinger::EffectHandle::dump(char* buffer, size_t size)
{
- bool locked = mCblk ? tryLock(mCblk->lock) : false;
+ bool locked = mCblk != NULL && tryLock(mCblk->lock);
snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n",
- (mClient == NULL) ? getpid() : mClient->pid(),
+ (mClient == 0) ? getpid_cached : mClient->pid(),
mPriority,
mHasControl,
!locked,
@@ -7080,15 +7588,14 @@
#undef LOG_TAG
#define LOG_TAG "AudioFlinger::EffectChain"
-AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread,
+AudioFlinger::EffectChain::EffectChain(ThreadBase *thread,
int sessionId)
- : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0),
+ : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0),
mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX),
mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX)
{
mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
- sp<ThreadBase> thread = mThread.promote();
- if (thread == 0) {
+ if (thread == NULL) {
return;
}
mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) /
@@ -7106,48 +7613,42 @@
// getEffectFromDesc_l() must be called with ThreadBase::mLock held
sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor)
{
- sp<EffectModule> effect;
size_t size = mEffects.size();
for (size_t i = 0; i < size; i++) {
if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) {
- effect = mEffects[i];
- break;
+ return mEffects[i];
}
}
- return effect;
+ return 0;
}
// getEffectFromId_l() must be called with ThreadBase::mLock held
sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id)
{
- sp<EffectModule> effect;
size_t size = mEffects.size();
for (size_t i = 0; i < size; i++) {
// by convention, return first effect if id provided is 0 (0 is never a valid id)
if (id == 0 || mEffects[i]->id() == id) {
- effect = mEffects[i];
- break;
+ return mEffects[i];
}
}
- return effect;
+ return 0;
}
// getEffectFromType_l() must be called with ThreadBase::mLock held
sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l(
const effect_uuid_t *type)
{
- sp<EffectModule> effect;
size_t size = mEffects.size();
for (size_t i = 0; i < size; i++) {
if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) {
- effect = mEffects[i];
- break;
+ return mEffects[i];
}
}
- return effect;
+ return 0;
}
// Must be called with EffectChain::mLock locked
@@ -7238,12 +7739,12 @@
// Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is
// already present
- int size = (int)mEffects.size();
- int idx_insert = size;
- int idx_insert_first = -1;
- int idx_insert_last = -1;
+ size_t size = mEffects.size();
+ size_t idx_insert = size;
+ ssize_t idx_insert_first = -1;
+ ssize_t idx_insert_last = -1;
- for (int i = 0; i < size; i++) {
+ for (size_t i = 0; i < size; i++) {
effect_descriptor_t d = mEffects[i]->desc();
uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK;
uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK;
@@ -7312,11 +7813,10 @@
size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect)
{
Mutex::Autolock _l(mLock);
- int size = (int)mEffects.size();
- int i;
+ size_t size = mEffects.size();
uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK;
- for (i = 0; i < size; i++) {
+ for (size_t i = 0; i < size; i++) {
if (effect == mEffects[i]) {
// calling stop here will remove pre-processing effect from the audio HAL.
// This is safe as we hold the EffectChain mutex which guarantees that we are not in
@@ -7352,7 +7852,7 @@
}
// setMode_l() must be called with PlaybackThread::mLock held
-void AudioFlinger::EffectChain::setMode_l(uint32_t mode)
+void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode)
{
size_t size = mEffects.size();
for (size_t i = 0; i < size; i++) {
@@ -7463,7 +7963,7 @@
sp<SuspendedEffectDesc> desc;
// use effect type UUID timelow as key as there is no real risk of identical
// timeLow fields among effect type UUIDs.
- int index = mSuspendedEffects.indexOfKey(type->timeLow);
+ ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow);
if (suspend) {
if (index >= 0) {
desc = mSuspendedEffects.valueAt(index);
@@ -7513,7 +8013,7 @@
{
sp<SuspendedEffectDesc> desc;
- int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
+ ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
if (suspend) {
if (index >= 0) {
desc = mSuspendedEffects.valueAt(index);
@@ -7523,7 +8023,8 @@
ALOGV("setEffectSuspendedAll_l() add entry for 0");
}
if (desc->mRefCount++ == 0) {
- Vector< sp<EffectModule> > effects = getSuspendEligibleEffects();
+ Vector< sp<EffectModule> > effects;
+ getSuspendEligibleEffects(effects);
for (size_t i = 0; i < effects.size(); i++) {
setEffectSuspended_l(&effects[i]->desc().type, true);
}
@@ -7574,33 +8075,27 @@
return true;
}
-Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects()
+void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects)
{
- Vector< sp<EffectModule> > effects;
+ effects.clear();
for (size_t i = 0; i < mEffects.size(); i++) {
- if (!isEffectEligibleForSuspend(mEffects[i]->desc())) {
- continue;
+ if (isEffectEligibleForSuspend(mEffects[i]->desc())) {
+ effects.add(mEffects[i]);
}
- effects.add(mEffects[i]);
}
- return effects;
}
sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled(
const effect_uuid_t *type)
{
- sp<EffectModule> effect;
- effect = getEffectFromType_l(type);
- if (effect != 0 && !effect->isEnabled()) {
- effect.clear();
- }
- return effect;
+ sp<EffectModule> effect = getEffectFromType_l(type);
+ return effect != 0 && effect->isEnabled() ? effect : 0;
}
void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
bool enabled)
{
- int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
+ ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
if (enabled) {
if (index < 0) {
// if the effect is not suspend check if all effects are suspended
@@ -7619,7 +8114,7 @@
}
}
ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x",
- effect->desc().type.timeLow);
+ effect->desc().type.timeLow);
sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
// if effect is requested to suspended but was not yet enabled, supend it now.
if (desc->mEffect == 0) {
@@ -7632,7 +8127,7 @@
return;
}
ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x",
- effect->desc().type.timeLow);
+ effect->desc().type.timeLow);
sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
desc->mEffect.clear();
effect->setSuspended(false);
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 9bd2c7f..0e4b24a 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -1,4 +1,4 @@
-/* //device/include/server/AudioFlinger/AudioFlinger.h
+/*
**
** Copyright 2007, The Android Open Source Project
**
@@ -22,10 +22,13 @@
#include <sys/types.h>
#include <limits.h>
+#include <common_time/cc_helper.h>
+
#include <media/IAudioFlinger.h>
#include <media/IAudioFlingerClient.h>
#include <media/IAudioTrack.h>
#include <media/IAudioRecord.h>
+#include <media/AudioSystem.h>
#include <media/AudioTrack.h>
#include <utils/Atomic.h>
@@ -55,97 +58,118 @@
// ----------------------------------------------------------------------------
-#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
-#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
+// AudioFlinger has a hard-coded upper limit of 2 channels for capture and playback.
+// There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect.
+// Adding full support for > 2 channel capture or playback would require more than simply changing
+// this #define. There is an independent hard-coded upper limit in AudioMixer;
+// removing that AudioMixer limit would be necessary but insufficient to support > 2 channels.
+// The macro FCC_2 highlights some (but not all) places where there is are 2-channel assumptions.
+// Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc.
+#define FCC_2 2 // FCC_2 = Fixed Channel Count 2
-
-// ----------------------------------------------------------------------------
-
-static const nsecs_t kStandbyTimeInNsecs = seconds(3);
+static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3);
class AudioFlinger :
public BinderService<AudioFlinger>,
public BnAudioFlinger
{
- friend class BinderService<AudioFlinger>;
+ friend class BinderService<AudioFlinger>; // for AudioFlinger()
public:
- static char const* getServiceName() { return "media.audio_flinger"; }
+ static const char* getServiceName() { return "media.audio_flinger"; }
virtual status_t dump(int fd, const Vector<String16>& args);
- // IAudioFlinger interface
+ // IAudioFlinger interface, in binder opcode order
virtual sp<IAudioTrack> createTrack(
pid_t pid,
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
- int output,
+ audio_io_handle_t output,
+ bool isTimed,
int *sessionId,
status_t *status);
- virtual uint32_t sampleRate(int output) const;
- virtual int channelCount(int output) const;
- virtual uint32_t format(int output) const;
- virtual size_t frameCount(int output) const;
- virtual uint32_t latency(int output) const;
+ virtual sp<IAudioRecord> openRecord(
+ pid_t pid,
+ audio_io_handle_t input,
+ uint32_t sampleRate,
+ audio_format_t format,
+ uint32_t channelMask,
+ int frameCount,
+ uint32_t flags,
+ int *sessionId,
+ status_t *status);
+
+ virtual uint32_t sampleRate(audio_io_handle_t output) const;
+ virtual int channelCount(audio_io_handle_t output) const;
+ virtual audio_format_t format(audio_io_handle_t output) const;
+ virtual size_t frameCount(audio_io_handle_t output) const;
+ virtual uint32_t latency(audio_io_handle_t output) const;
virtual status_t setMasterVolume(float value);
virtual status_t setMasterMute(bool muted);
virtual float masterVolume() const;
+ virtual float masterVolumeSW() const;
virtual bool masterMute() const;
- virtual status_t setStreamVolume(int stream, float value, int output);
- virtual status_t setStreamMute(int stream, bool muted);
+ virtual status_t setStreamVolume(audio_stream_type_t stream, float value,
+ audio_io_handle_t output);
+ virtual status_t setStreamMute(audio_stream_type_t stream, bool muted);
- virtual float streamVolume(int stream, int output) const;
- virtual bool streamMute(int stream) const;
+ virtual float streamVolume(audio_stream_type_t stream,
+ audio_io_handle_t output) const;
+ virtual bool streamMute(audio_stream_type_t stream) const;
- virtual status_t setMode(int mode);
+ virtual status_t setMode(audio_mode_t mode);
virtual status_t setMicMute(bool state);
virtual bool getMicMute() const;
- virtual status_t setParameters(int ioHandle, const String8& keyValuePairs);
- virtual String8 getParameters(int ioHandle, const String8& keys);
+ virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);
+ virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const;
virtual void registerClient(const sp<IAudioFlingerClient>& client);
- virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount);
- virtual unsigned int getInputFramesLost(int ioHandle);
+ virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const;
- virtual int openOutput(uint32_t *pDevices,
+ virtual audio_io_handle_t openOutput(uint32_t *pDevices,
uint32_t *pSamplingRate,
- uint32_t *pFormat,
+ audio_format_t *pFormat,
uint32_t *pChannels,
uint32_t *pLatencyMs,
- uint32_t flags);
+ audio_policy_output_flags_t flags);
- virtual int openDuplicateOutput(int output1, int output2);
+ virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
+ audio_io_handle_t output2);
- virtual status_t closeOutput(int output);
+ virtual status_t closeOutput(audio_io_handle_t output);
- virtual status_t suspendOutput(int output);
+ virtual status_t suspendOutput(audio_io_handle_t output);
- virtual status_t restoreOutput(int output);
+ virtual status_t restoreOutput(audio_io_handle_t output);
- virtual int openInput(uint32_t *pDevices,
+ virtual audio_io_handle_t openInput(uint32_t *pDevices,
uint32_t *pSamplingRate,
- uint32_t *pFormat,
+ audio_format_t *pFormat,
uint32_t *pChannels,
- uint32_t acoustics);
+ audio_in_acoustics_t acoustics);
- virtual status_t closeInput(int input);
+ virtual status_t closeInput(audio_io_handle_t input);
- virtual status_t setStreamOutput(uint32_t stream, int output);
+ virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output);
virtual status_t setVoiceVolume(float volume);
- virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output);
+ virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
+ audio_io_handle_t output) const;
+
+ virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const;
virtual int newAudioSessionId();
@@ -153,54 +177,25 @@
virtual void releaseAudioSessionId(int audioSession);
- virtual status_t queryNumberEffects(uint32_t *numEffects);
+ virtual status_t queryNumberEffects(uint32_t *numEffects) const;
- virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor);
+ virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const;
- virtual status_t getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor);
+ virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid,
+ effect_descriptor_t *descriptor) const;
virtual sp<IEffect> createEffect(pid_t pid,
effect_descriptor_t *pDesc,
const sp<IEffectClient>& effectClient,
int32_t priority,
- int io,
+ audio_io_handle_t io,
int sessionId,
status_t *status,
int *id,
int *enabled);
- virtual status_t moveEffects(int sessionId, int srcOutput, int dstOutput);
-
- enum hardware_call_state {
- AUDIO_HW_IDLE = 0,
- AUDIO_HW_INIT,
- AUDIO_HW_OUTPUT_OPEN,
- AUDIO_HW_OUTPUT_CLOSE,
- AUDIO_HW_INPUT_OPEN,
- AUDIO_HW_INPUT_CLOSE,
- AUDIO_HW_STANDBY,
- AUDIO_HW_SET_MASTER_VOLUME,
- AUDIO_HW_GET_ROUTING,
- AUDIO_HW_SET_ROUTING,
- AUDIO_HW_GET_MODE,
- AUDIO_HW_SET_MODE,
- AUDIO_HW_GET_MIC_MUTE,
- AUDIO_HW_SET_MIC_MUTE,
- AUDIO_SET_VOICE_VOLUME,
- AUDIO_SET_PARAMETER,
- };
-
- // record interface
- virtual sp<IAudioRecord> openRecord(
- pid_t pid,
- int input,
- uint32_t sampleRate,
- uint32_t format,
- uint32_t channelMask,
- int frameCount,
- uint32_t flags,
- int *sessionId,
- status_t *status);
+ virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput,
+ audio_io_handle_t dstOutput);
virtual status_t onTransact(
uint32_t code,
@@ -208,19 +203,29 @@
Parcel* reply,
uint32_t flags);
- uint32_t getMode() { return mMode; }
-
- bool btNrecIsOff() { return mBtNrecIsOff; }
+ // end of IAudioFlinger interface
private:
+ audio_mode_t getMode() const { return mMode; }
+
+ bool btNrecIsOff() const { return mBtNrecIsOff; }
+
AudioFlinger();
virtual ~AudioFlinger();
- status_t initCheck() const;
+ // call in any IAudioFlinger method that accesses mPrimaryHardwareDev
+ status_t initCheck() const { return mPrimaryHardwareDev == NULL ? NO_INIT : NO_ERROR; }
+
+ // RefBase
virtual void onFirstRef();
+
audio_hw_device_t* findSuitableHwDev_l(uint32_t devices);
void purgeStaleEffects_l();
+ // standby delay for MIXER and DUPLICATING playback threads is read from property
+ // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs
+ static nsecs_t mStandbyTimeInNsecs;
+
// Internal dump utilites.
status_t dumpPermissionDenial(int fd, const Vector<String16>& args);
status_t dumpClients(int fd, const Vector<String16>& args);
@@ -231,16 +236,22 @@
public:
Client(const sp<AudioFlinger>& audioFlinger, pid_t pid);
virtual ~Client();
- const sp<MemoryDealer>& heap() const;
+ sp<MemoryDealer> heap() const;
pid_t pid() const { return mPid; }
- sp<AudioFlinger> audioFlinger() { return mAudioFlinger; }
+ sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; }
+
+ bool reserveTimedTrack();
+ void releaseTimedTrack();
private:
Client(const Client&);
Client& operator = (const Client&);
- sp<AudioFlinger> mAudioFlinger;
- sp<MemoryDealer> mMemoryDealer;
- pid_t mPid;
+ const sp<AudioFlinger> mAudioFlinger;
+ const sp<MemoryDealer> mMemoryDealer;
+ const pid_t mPid;
+
+ Mutex mTimedTrackLock;
+ int mTimedTrackCount;
};
// --- Notification Client ---
@@ -251,7 +262,7 @@
pid_t pid);
virtual ~NotificationClient();
- sp<IAudioFlingerClient> client() { return mClient; }
+ sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; }
// IBinder::DeathRecipient
virtual void binderDied(const wp<IBinder>& who);
@@ -260,9 +271,9 @@
NotificationClient(const NotificationClient&);
NotificationClient& operator = (const NotificationClient&);
- sp<AudioFlinger> mAudioFlinger;
- pid_t mPid;
- sp<IAudioFlingerClient> mClient;
+ const sp<AudioFlinger> mAudioFlinger;
+ const pid_t mPid;
+ const sp<IAudioFlingerClient> mAudioFlingerClient;
};
class TrackHandle;
@@ -282,17 +293,17 @@
class ThreadBase : public Thread {
public:
- ThreadBase (const sp<AudioFlinger>& audioFlinger, int id, uint32_t device);
- virtual ~ThreadBase();
-
- enum type {
+ enum type_t {
MIXER, // Thread class is MixerThread
DIRECT, // Thread class is DirectOutputThread
DUPLICATING, // Thread class is DuplicatingThread
RECORD // Thread class is RecordThread
};
+ ThreadBase (const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, uint32_t device, type_t type);
+ virtual ~ThreadBase();
+
status_t dumpBase(int fd, const Vector<String16>& args);
status_t dumpEffectChains(int fd, const Vector<String16>& args);
@@ -305,6 +316,8 @@
enum track_state {
IDLE,
TERMINATED,
+ // These are order-sensitive; do not change order without reviewing the impact.
+ // In particular there are assumptions about > STOPPED.
STOPPED,
RESUMING,
ACTIVE,
@@ -312,52 +325,39 @@
PAUSED
};
- enum track_flags {
- STEPSERVER_FAILED = 0x01, // StepServer could not acquire cblk->lock mutex
- SYSTEM_FLAGS_MASK = 0x0000ffffUL,
- // The upper 16 bits are used for track-specific flags.
- };
-
- TrackBase(const wp<ThreadBase>& thread,
+ TrackBase(ThreadBase *thread,
const sp<Client>& client,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
- uint32_t flags,
const sp<IMemory>& sharedBuffer,
int sessionId);
- ~TrackBase();
+ virtual ~TrackBase();
- virtual status_t start() = 0;
+ virtual status_t start(pid_t tid) = 0;
virtual void stop() = 0;
- sp<IMemory> getCblk() const;
+ sp<IMemory> getCblk() const { return mCblkMemory; }
audio_track_cblk_t* cblk() const { return mCblk; }
- int sessionId() { return mSessionId; }
+ int sessionId() const { return mSessionId; }
protected:
- friend class ThreadBase;
- friend class RecordHandle;
- friend class PlaybackThread;
- friend class RecordThread;
- friend class MixerThread;
- friend class DirectOutputThread;
-
TrackBase(const TrackBase&);
TrackBase& operator = (const TrackBase&);
- virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) = 0;
+ // AudioBufferProvider interface
+ virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0;
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
- uint32_t format() const {
+ audio_format_t format() const {
return mFormat;
}
- int channelCount() const ;
+ int channelCount() const { return mChannelCount; }
- uint32_t channelMask() const;
+ uint32_t channelMask() const { return mChannelMask; }
- int sampleRate() const;
+ int sampleRate() const; // FIXME inline after cblk sr moved
void* getBuffer(uint32_t offset, uint32_t frames) const;
@@ -372,19 +372,18 @@
bool step();
void reset();
- wp<ThreadBase> mThread;
- sp<Client> mClient;
+ const wp<ThreadBase> mThread;
+ /*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const
sp<IMemory> mCblkMemory;
audio_track_cblk_t* mCblk;
void* mBuffer;
void* mBufferEnd;
uint32_t mFrameCount;
// we don't really need a lock for these
- int mState;
- int mClientTid;
- uint32_t mFormat;
- uint32_t mFlags;
- int mSessionId;
+ track_state mState;
+ const audio_format_t mFormat;
+ bool mStepServerFailed;
+ const int mSessionId;
uint8_t mChannelCount;
uint32_t mChannelMask;
};
@@ -413,12 +412,14 @@
};
virtual status_t initCheck() const = 0;
- int type() const { return mType; }
- uint32_t sampleRate() const;
- int channelCount() const;
- uint32_t format() const;
- size_t frameCount() const;
+ type_t type() const { return mType; }
+ uint32_t sampleRate() const { return mSampleRate; }
+ int channelCount() const { return mChannelCount; }
+ audio_format_t format() const { return mFormat; }
+ size_t frameCount() const { return mFrameCount; }
void wakeUp() { mWaitWorkCV.broadcast(); }
+ // Should be "virtual status_t requestExitAndWait()" and override same
+ // method in Thread, but Thread::requestExitAndWait() is not yet virtual.
void exit();
virtual bool checkForNewParameters_l() = 0;
virtual status_t setParameters(const String8& keyValuePairs);
@@ -427,9 +428,9 @@
void sendConfigEvent(int event, int param = 0);
void sendConfigEvent_l(int event, int param = 0);
void processConfigEvents();
- int id() const { return mId;}
- bool standby() { return mStandby; }
- uint32_t device() { return mDevice; }
+ audio_io_handle_t id() const { return mId;}
+ bool standby() const { return mStandby; }
+ uint32_t device() const { return mDevice; }
virtual audio_stream_t* stream() = 0;
sp<EffectHandle> createEffect_l(
@@ -442,7 +443,7 @@
status_t *status);
void disconnectEffect(const sp< EffectModule>& effect,
const wp<EffectHandle>& handle,
- bool unpiniflast);
+ bool unpinIfLast);
// return values for hasAudioSession (bit field)
enum effect_state {
@@ -460,14 +461,15 @@
virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0;
// remove an effect chain from the chain list (mEffectChains)
virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0;
- // lock mall effect chains Mutexes. Must be called before releasing the
+ // lock all effect chains Mutexes. Must be called before releasing the
// ThreadBase mutex before processing the mixer and effects. This guarantees the
// integrity of the chains during the process.
- void lockEffectChains_l(Vector<sp <EffectChain> >& effectChains);
+ // Also sets the parameter 'effectChains' to current value of mEffectChains.
+ void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains);
// unlock effect chains after process
- void unlockEffectChains(Vector<sp <EffectChain> >& effectChains);
+ void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains);
// set audio mode to all effect chains
- void setMode(uint32_t mode);
+ void setMode(audio_mode_t mode);
// get effect module with corresponding ID on specified audio session
sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId);
// add and effect module. Also creates the effect chain is none exists for
@@ -525,96 +527,122 @@
// check if some effects must be suspended when an effect chain is added
void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain);
- friend class AudioFlinger;
- friend class Track;
- friend class TrackBase;
- friend class PlaybackThread;
- friend class MixerThread;
- friend class DirectOutputThread;
- friend class DuplicatingThread;
- friend class RecordThread;
- friend class RecordTrack;
+ friend class AudioFlinger; // for mEffectChains
- int mType;
+ const type_t mType;
+
+ // Used by parameters, config events, addTrack_l, exit
Condition mWaitWorkCV;
- sp<AudioFlinger> mAudioFlinger;
+
+ const sp<AudioFlinger> mAudioFlinger;
uint32_t mSampleRate;
size_t mFrameCount;
uint32_t mChannelMask;
uint16_t mChannelCount;
- uint16_t mFrameSize;
- uint32_t mFormat;
+ size_t mFrameSize;
+ audio_format_t mFormat;
+
+ // Parameter sequence by client: binder thread calling setParameters():
+ // 1. Lock mLock
+ // 2. Append to mNewParameters
+ // 3. mWaitWorkCV.signal
+ // 4. mParamCond.waitRelative with timeout
+ // 5. read mParamStatus
+ // 6. mWaitWorkCV.signal
+ // 7. Unlock
+ //
+ // Parameter sequence by server: threadLoop calling checkForNewParameters_l():
+ // 1. Lock mLock
+ // 2. If there is an entry in mNewParameters proceed ...
+ // 2. Read first entry in mNewParameters
+ // 3. Process
+ // 4. Remove first entry from mNewParameters
+ // 5. Set mParamStatus
+ // 6. mParamCond.signal
+ // 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus)
+ // 8. Unlock
Condition mParamCond;
Vector<String8> mNewParameters;
status_t mParamStatus;
- Vector<ConfigEvent *> mConfigEvents;
+
+ Vector<ConfigEvent> mConfigEvents;
bool mStandby;
- int mId;
- bool mExiting;
+ const audio_io_handle_t mId;
Vector< sp<EffectChain> > mEffectChains;
uint32_t mDevice; // output device for PlaybackThread
// input + output devices for RecordThread
- static const int kNameLength = 32;
+ static const int kNameLength = 16; // prctl(PR_SET_NAME) limit
char mName[kNameLength];
sp<IPowerManager> mPowerManager;
sp<IBinder> mWakeLockToken;
- sp<PMDeathRecipient> mDeathRecipient;
+ const sp<PMDeathRecipient> mDeathRecipient;
// list of suspended effects per session and per type. The first vector is
// keyed by session ID, the second by type UUID timeLow field
KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions;
};
+ struct stream_type_t {
+ stream_type_t()
+ : volume(1.0f),
+ mute(false),
+ valid(true)
+ {
+ }
+ float volume;
+ bool mute;
+ bool valid;
+ };
+
// --- PlaybackThread ---
class PlaybackThread : public ThreadBase {
public:
enum mixer_state {
- MIXER_IDLE,
- MIXER_TRACKS_ENABLED,
- MIXER_TRACKS_READY
+ MIXER_IDLE, // no active tracks
+ MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready
+ MIXER_TRACKS_READY // at least one active track, and at least one track has data
+ // standby mode does not have an enum value
+ // suspend by audio policy manager is orthogonal to mixer state
};
// playback track
class Track : public TrackBase {
public:
- Track( const wp<ThreadBase>& thread,
+ Track( PlaybackThread *thread,
const sp<Client>& client,
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId);
- ~Track();
+ virtual ~Track();
void dump(char* buffer, size_t size);
- virtual status_t start();
+ virtual status_t start(pid_t tid);
virtual void stop();
void pause();
void flush();
void destroy();
void mute(bool);
- void setVolume(float left, float right);
int name() const {
return mName;
}
- int type() const {
+ audio_stream_type_t streamType() const {
return mStreamType;
}
status_t attachAuxEffect(int EffectId);
void setAuxBuffer(int EffectId, int32_t *buffer);
- int32_t *auxBuffer() { return mAuxBuffer; }
+ int32_t *auxBuffer() const { return mAuxBuffer; }
void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; }
- int16_t *mainBuffer() { return mMainBuffer; }
- int auxEffectId() { return mAuxEffectId; }
-
+ int16_t *mainBuffer() const { return mMainBuffer; }
+ int auxEffectId() const { return mAuxEffectId; }
protected:
- friend class ThreadBase;
- friend class TrackHandle;
+ // for numerous
friend class PlaybackThread;
friend class MixerThread;
friend class DirectOutputThread;
@@ -622,8 +650,13 @@
Track(const Track&);
Track& operator = (const Track&);
- virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
- bool isMuted() { return mMute; }
+ // AudioBufferProvider interface
+ virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS);
+ // releaseBuffer() not overridden
+
+ virtual uint32_t framesReady() const;
+
+ bool isMuted() const { return mMute; }
bool isPausing() const {
return mState == PAUSING;
}
@@ -638,8 +671,11 @@
return (mStreamType == AUDIO_STREAM_CNT);
}
+ public:
+ virtual bool isTimedTrack() const { return false; }
+ protected:
+
// we don't really need a lock for these
- float mVolume[2];
volatile bool mMute;
// FILLED state is used for suppressing volume ramp at begin of playing
enum {FS_FILLING, FS_FILLED, FS_ACTIVE};
@@ -647,7 +683,7 @@
int8_t mRetryCount;
sp<IMemory> mSharedBuffer;
bool mResetDone;
- int mStreamType;
+ audio_stream_type_t mStreamType;
int mName;
int16_t *mMainBuffer;
int32_t *mAuxBuffer;
@@ -655,6 +691,80 @@
bool mHasVolumeController;
}; // end of Track
+ class TimedTrack : public Track {
+ public:
+ static sp<TimedTrack> create(PlaybackThread *thread,
+ const sp<Client>& client,
+ audio_stream_type_t streamType,
+ uint32_t sampleRate,
+ audio_format_t format,
+ uint32_t channelMask,
+ int frameCount,
+ const sp<IMemory>& sharedBuffer,
+ int sessionId);
+ ~TimedTrack();
+
+ class TimedBuffer {
+ public:
+ TimedBuffer();
+ TimedBuffer(const sp<IMemory>& buffer, int64_t pts);
+ const sp<IMemory>& buffer() const { return mBuffer; }
+ int64_t pts() const { return mPTS; }
+ int position() const { return mPosition; }
+ void setPosition(int pos) { mPosition = pos; }
+ private:
+ sp<IMemory> mBuffer;
+ int64_t mPTS;
+ int mPosition;
+ };
+
+ virtual bool isTimedTrack() const { return true; }
+
+ virtual uint32_t framesReady() const;
+
+ // AudioBufferProvider interface
+ virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts);
+ virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
+
+ void timedYieldSamples(AudioBufferProvider::Buffer* buffer);
+ void timedYieldSilence(uint32_t numFrames,
+ AudioBufferProvider::Buffer* buffer);
+
+ status_t allocateTimedBuffer(size_t size,
+ sp<IMemory>* buffer);
+ status_t queueTimedBuffer(const sp<IMemory>& buffer,
+ int64_t pts);
+ status_t setMediaTimeTransform(const LinearTransform& xform,
+ TimedAudioTrack::TargetTimeline target);
+ void trimTimedBufferQueue_l();
+
+ private:
+ TimedTrack(PlaybackThread *thread,
+ const sp<Client>& client,
+ audio_stream_type_t streamType,
+ uint32_t sampleRate,
+ audio_format_t format,
+ uint32_t channelMask,
+ int frameCount,
+ const sp<IMemory>& sharedBuffer,
+ int sessionId);
+
+ uint64_t mLocalTimeFreq;
+ LinearTransform mLocalTimeToSampleTransform;
+ sp<MemoryDealer> mTimedMemoryDealer;
+ Vector<TimedBuffer> mTimedBufferQueue;
+ uint8_t* mTimedSilenceBuffer;
+ uint32_t mTimedSilenceBufferSize;
+ mutable Mutex mTimedBufferQueueLock;
+ bool mTimedAudioOutputOnTime;
+ CCHelper mCCHelper;
+
+ Mutex mMediaTimeTransformLock;
+ LinearTransform mMediaTimeTransform;
+ bool mMediaTimeTransformValid;
+ TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget;
+ };
+
// playback track
class OutputTrack : public Track {
@@ -665,23 +775,27 @@
int16_t *mBuffer;
};
- OutputTrack( const wp<ThreadBase>& thread,
+ OutputTrack(PlaybackThread *thread,
DuplicatingThread *sourceThread,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount);
- ~OutputTrack();
+ virtual ~OutputTrack();
- virtual status_t start();
+ virtual status_t start(pid_t tid);
virtual void stop();
bool write(int16_t* data, uint32_t frames);
- bool bufferQueueEmpty() { return (mBufferQueue.size() == 0) ? true : false; }
- bool isActive() { return mActive; }
- wp<ThreadBase>& thread() { return mThread; }
+ bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; }
+ bool isActive() const { return mActive; }
+ const wp<ThreadBase>& thread() const { return mThread; }
private:
+ enum {
+ NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value
+ };
+
status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs);
void clearBufferQueue();
@@ -691,56 +805,73 @@
Vector < Buffer* > mBufferQueue;
AudioBufferProvider::Buffer mOutBuffer;
bool mActive;
- DuplicatingThread* mSourceThread;
+ DuplicatingThread* const mSourceThread; // for waitTimeMs() in write()
}; // end of OutputTrack
- PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device);
+ PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
+ audio_io_handle_t id, uint32_t device, type_t type);
virtual ~PlaybackThread();
- virtual status_t dump(int fd, const Vector<String16>& args);
+ status_t dump(int fd, const Vector<String16>& args);
// Thread virtuals
virtual status_t readyToRun();
+ virtual bool threadLoop();
+
+ // RefBase
virtual void onFirstRef();
- virtual status_t initCheck() const { return (mOutput == 0) ? NO_INIT : NO_ERROR; }
+protected:
+ // Code snippets that were lifted up out of threadLoop()
+ virtual void threadLoop_mix() = 0;
+ virtual void threadLoop_sleepTime() = 0;
+ virtual void threadLoop_write();
+ virtual void threadLoop_standby();
- virtual uint32_t latency() const;
+ // prepareTracks_l reads and writes mActiveTracks, and also returns the
+ // pending set of tracks to remove via Vector 'tracksToRemove'. The caller is
+ // responsible for clearing or destroying this Vector later on, when it
+ // is safe to do so. That will drop the final ref count and destroy the tracks.
+ virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0;
- virtual status_t setMasterVolume(float value);
- virtual status_t setMasterMute(bool muted);
+public:
- virtual float masterVolume() const;
- virtual bool masterMute() const;
+ virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; }
- virtual status_t setStreamVolume(int stream, float value);
- virtual status_t setStreamMute(int stream, bool muted);
+ // return estimated latency in milliseconds, as reported by HAL
+ uint32_t latency() const;
- virtual float streamVolume(int stream) const;
- virtual bool streamMute(int stream) const;
+ void setMasterVolume(float value);
+ void setMasterMute(bool muted);
+
+ void setStreamVolume(audio_stream_type_t stream, float value);
+ void setStreamMute(audio_stream_type_t stream, bool muted);
+
+ float streamVolume(audio_stream_type_t stream) const;
sp<Track> createTrack_l(
const sp<AudioFlinger::Client>& client,
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
+ bool isTimed,
status_t *status);
- AudioStreamOut* getOutput();
+ AudioStreamOut* getOutput() const;
AudioStreamOut* clearOutput();
virtual audio_stream_t* stream();
void suspend() { mSuspended++; }
- void restore() { if (mSuspended) mSuspended--; }
- bool isSuspended() { return (mSuspended != 0); }
+ void restore() { if (mSuspended > 0) mSuspended--; }
+ bool isSuspended() const { return (mSuspended > 0); }
virtual String8 getParameters(const String8& keys);
virtual void audioConfigChanged_l(int event, int param = 0);
- virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames);
- int16_t *mixBuffer() { return mMixBuffer; };
+ status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames);
+ int16_t *mixBuffer() const { return mMixBuffer; };
virtual void detachAuxEffect_l(int effectId);
status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track,
@@ -753,25 +884,19 @@
virtual uint32_t hasAudioSession(int sessionId);
virtual uint32_t getStrategyForSession_l(int sessionId);
- void setStreamValid(int streamType, bool valid);
-
- struct stream_type_t {
- stream_type_t()
- : volume(1.0f),
- mute(false),
- valid(true)
- {
- }
- float volume;
- bool mute;
- bool valid;
- };
+ void setStreamValid(audio_stream_type_t streamType, bool valid);
protected:
int16_t* mMixBuffer;
- int mSuspended;
+ uint32_t mSuspended; // suspend count, > 0 means suspended
int mBytesWritten;
+ private:
+ // mMasterMute is in both PlaybackThread and in AudioFlinger. When a
+ // PlaybackThread needs to find out if master-muted, it checks it's local
+ // copy rather than the one in AudioFlinger. This optimization saves a lock.
bool mMasterMute;
+ void setMasterMute_l(bool muted) { mMasterMute = muted; }
+ protected:
SortedVector< wp<Track> > mActiveTracks;
virtual int getTrackName_l() = 0;
@@ -780,15 +905,19 @@
virtual uint32_t idleSleepTimeUs() = 0;
virtual uint32_t suspendSleepTimeUs() = 0;
+ // Code snippets that are temporarily lifted up out of threadLoop() until the merge
+ void checkSilentMode_l();
+
+ // Non-trivial for DUPLICATING only
+ virtual void saveOutputTracks() { }
+ virtual void clearOutputTracks() { }
+
+ // Cache various calculated values, at threadLoop() entry and after a parameter change
+ virtual void cacheParameters_l();
+
private:
- friend class AudioFlinger;
- friend class OutputTrack;
- friend class Track;
- friend class TrackBase;
- friend class MixerThread;
- friend class DirectOutputThread;
- friend class DuplicatingThread;
+ friend class AudioFlinger; // for numerous
PlaybackThread(const Client&);
PlaybackThread& operator = (const PlaybackThread&);
@@ -803,52 +932,82 @@
status_t dumpTracks(int fd, const Vector<String16>& args);
SortedVector< sp<Track> > mTracks;
- // mStreamTypes[] uses 1 additionnal stream type internally for the OutputTrack used by DuplicatingThread
+ // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread
stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1];
- AudioStreamOut* mOutput;
+ AudioStreamOut *mOutput;
float mMasterVolume;
nsecs_t mLastWriteTime;
int mNumWrites;
int mNumDelayedWrites;
bool mInWrite;
+
+ // FIXME rename these former local variables of threadLoop to standard "m" names
+ nsecs_t standbyTime;
+ size_t mixBufferSize;
+
+ // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l()
+ uint32_t activeSleepTime;
+ uint32_t idleSleepTime;
+
+ uint32_t sleepTime;
+
+ // mixer status returned by prepareTracks_l()
+ mixer_state mMixerStatus; // current cycle
+ mixer_state mPrevMixerStatus; // previous cycle
+
+ // FIXME move these declarations into the specific sub-class that needs them
+ // MIXER only
+ bool longStandbyExit;
+ uint32_t sleepTimeShift;
+
+ // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value
+ nsecs_t standbyDelay;
+
+ // MIXER only
+ nsecs_t maxPeriod;
+
+ // DUPLICATING only
+ uint32_t writeFrames;
};
class MixerThread : public PlaybackThread {
public:
MixerThread (const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output,
- int id,
- uint32_t device);
+ audio_io_handle_t id,
+ uint32_t device,
+ type_t type = MIXER);
virtual ~MixerThread();
// Thread virtuals
- virtual bool threadLoop();
- void invalidateTracks(int streamType);
+ void invalidateTracks(audio_stream_type_t streamType);
virtual bool checkForNewParameters_l();
virtual status_t dumpInternals(int fd, const Vector<String16>& args);
protected:
- uint32_t prepareTracks_l(const SortedVector< wp<Track> >& activeTracks,
- Vector< sp<Track> > *tracksToRemove);
+ virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
virtual int getTrackName_l();
virtual void deleteTrackName_l(int name);
virtual uint32_t idleSleepTimeUs();
virtual uint32_t suspendSleepTimeUs();
+ virtual void cacheParameters_l();
+
+ // threadLoop snippets
+ virtual void threadLoop_mix();
+ virtual void threadLoop_sleepTime();
AudioMixer* mAudioMixer;
- uint32_t mPrevMixerStatus; // previous status (mixer_state) returned by
- // prepareTracks_l()
};
class DirectOutputThread : public PlaybackThread {
public:
- DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device);
- ~DirectOutputThread();
+ DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
+ audio_io_handle_t id, uint32_t device);
+ virtual ~DirectOutputThread();
// Thread virtuals
- virtual bool threadLoop();
virtual bool checkForNewParameters_l();
@@ -858,23 +1017,38 @@
virtual uint32_t activeSleepTimeUs();
virtual uint32_t idleSleepTimeUs();
virtual uint32_t suspendSleepTimeUs();
+ virtual void cacheParameters_l();
- private:
- void applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp);
+ // threadLoop snippets
+ virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
+ virtual void threadLoop_mix();
+ virtual void threadLoop_sleepTime();
+ // volumes last sent to audio HAL with stream->set_volume()
+ // FIXME use standard representation and names
float mLeftVolFloat;
float mRightVolFloat;
uint16_t mLeftVolShort;
uint16_t mRightVolShort;
+
+ // FIXME rename these former local variables of threadLoop to standard names
+ // next 3 were local to the while !exitingPending loop
+ bool rampVolume;
+ uint16_t leftVol;
+ uint16_t rightVol;
+
+private:
+ // prepareTracks_l() tells threadLoop_mix() the name of the single active track
+ sp<Track> mActiveTrack;
};
class DuplicatingThread : public MixerThread {
public:
- DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, int id);
- ~DuplicatingThread();
+ DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread,
+ audio_io_handle_t id);
+ virtual ~DuplicatingThread();
// Thread virtuals
- virtual bool threadLoop();
void addOutputTrack(MixerThread* thread);
void removeOutputTrack(MixerThread* thread);
uint32_t waitTimeMs() { return mWaitTimeMs; }
@@ -882,51 +1056,74 @@
virtual uint32_t activeSleepTimeUs();
private:
- bool outputsReady(SortedVector< sp<OutputTrack> > &outputTracks);
- void updateWaitTime();
+ bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks);
+ protected:
+ // threadLoop snippets
+ virtual void threadLoop_mix();
+ virtual void threadLoop_sleepTime();
+ virtual void threadLoop_write();
+ virtual void threadLoop_standby();
+ virtual void cacheParameters_l();
- SortedVector < sp<OutputTrack> > mOutputTracks;
+ private:
+ // called from threadLoop, addOutputTrack, removeOutputTrack
+ virtual void updateWaitTime_l();
+ protected:
+ virtual void saveOutputTracks();
+ virtual void clearOutputTracks();
+ private:
+
uint32_t mWaitTimeMs;
+ SortedVector < sp<OutputTrack> > outputTracks;
+ SortedVector < sp<OutputTrack> > mOutputTracks;
};
- PlaybackThread *checkPlaybackThread_l(int output) const;
- MixerThread *checkMixerThread_l(int output) const;
- RecordThread *checkRecordThread_l(int input) const;
- float streamVolumeInternal(int stream) const { return mStreamTypes[stream].volume; }
- void audioConfigChanged_l(int event, int ioHandle, void *param2);
+ PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const;
+ MixerThread *checkMixerThread_l(audio_io_handle_t output) const;
+ RecordThread *checkRecordThread_l(audio_io_handle_t input) const;
+ // no range check, AudioFlinger::mLock held
+ bool streamMute_l(audio_stream_type_t stream) const
+ { return mStreamTypes[stream].mute; }
+ // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held
+ float streamVolume_l(audio_stream_type_t stream) const
+ { return mStreamTypes[stream].volume; }
+ void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2);
+ // allocate an audio_io_handle_t, session ID, or effect ID
uint32_t nextUniqueId();
+
status_t moveEffectChain_l(int sessionId,
- AudioFlinger::PlaybackThread *srcThread,
- AudioFlinger::PlaybackThread *dstThread,
+ PlaybackThread *srcThread,
+ PlaybackThread *dstThread,
bool reRegister);
- PlaybackThread *primaryPlaybackThread_l();
- uint32_t primaryOutputDevice_l();
+ // return thread associated with primary hardware device, or NULL
+ PlaybackThread *primaryPlaybackThread_l() const;
+ uint32_t primaryOutputDevice_l() const;
- friend class AudioBuffer;
-
+ // server side of the client's IAudioTrack
class TrackHandle : public android::BnAudioTrack {
public:
TrackHandle(const sp<PlaybackThread::Track>& track);
virtual ~TrackHandle();
- virtual status_t start();
+ virtual sp<IMemory> getCblk() const;
+ virtual status_t start(pid_t tid);
virtual void stop();
virtual void flush();
virtual void mute(bool);
virtual void pause();
- virtual void setVolume(float left, float right);
- virtual sp<IMemory> getCblk() const;
virtual status_t attachAuxEffect(int effectId);
+ virtual status_t allocateTimedBuffer(size_t size,
+ sp<IMemory>* buffer);
+ virtual status_t queueTimedBuffer(const sp<IMemory>& buffer,
+ int64_t pts);
+ virtual status_t setMediaTimeTransform(const LinearTransform& xform,
+ int target);
virtual status_t onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
private:
- sp<PlaybackThread::Track> mTrack;
+ const sp<PlaybackThread::Track> mTrack;
};
- friend class Client;
- friend class PlaybackThread::Track;
-
-
void removeClient_l(pid_t pid);
void removeNotificationClient(pid_t pid);
@@ -939,17 +1136,16 @@
// record track
class RecordTrack : public TrackBase {
public:
- RecordTrack(const wp<ThreadBase>& thread,
+ RecordTrack(RecordThread *thread,
const sp<Client>& client,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
- uint32_t flags,
int sessionId);
- ~RecordTrack();
+ virtual ~RecordTrack();
- virtual status_t start();
+ virtual status_t start(pid_t tid);
virtual void stop();
bool overflow() { bool tmp = mOverflow; mOverflow = false; return tmp; }
@@ -958,13 +1154,14 @@
void dump(char* buffer, size_t size);
private:
- friend class AudioFlinger;
- friend class RecordThread;
+ friend class AudioFlinger; // for mState
RecordTrack(const RecordTrack&);
RecordTrack& operator = (const RecordTrack&);
- virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
+ // AudioBufferProvider interface
+ virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS);
+ // releaseBuffer() not overridden
bool mOverflow;
};
@@ -974,34 +1171,39 @@
AudioStreamIn *input,
uint32_t sampleRate,
uint32_t channels,
- int id,
+ audio_io_handle_t id,
uint32_t device);
- ~RecordThread();
+ virtual ~RecordThread();
+ // Thread
virtual bool threadLoop();
virtual status_t readyToRun();
+
+ // RefBase
virtual void onFirstRef();
- virtual status_t initCheck() const { return (mInput == 0) ? NO_INIT : NO_ERROR; }
+ virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; }
sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l(
const sp<AudioFlinger::Client>& client,
uint32_t sampleRate,
- int format,
+ audio_format_t format,
int channelMask,
int frameCount,
- uint32_t flags,
int sessionId,
status_t *status);
status_t start(RecordTrack* recordTrack);
+ status_t start(RecordTrack* recordTrack, pid_t tid);
void stop(RecordTrack* recordTrack);
status_t dump(int fd, const Vector<String16>& args);
- AudioStreamIn* getInput();
+ AudioStreamIn* getInput() const;
AudioStreamIn* clearInput();
virtual audio_stream_t* stream();
- virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
+ // AudioBufferProvider interface
+ virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts);
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
+
virtual bool checkForNewParameters_l();
virtual String8 getParameters(const String8& keys);
virtual void audioConfigChanged_l(int event, int param = 0);
@@ -1024,22 +1226,23 @@
int16_t *mRsmpInBuffer;
size_t mRsmpInIndex;
size_t mInputBytes;
- int mReqChannelCount;
- uint32_t mReqSampleRate;
+ const int mReqChannelCount;
+ const uint32_t mReqSampleRate;
ssize_t mBytesRead;
};
+ // server side of the client's IAudioRecord
class RecordHandle : public android::BnAudioRecord {
public:
RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack);
virtual ~RecordHandle();
- virtual status_t start();
- virtual void stop();
virtual sp<IMemory> getCblk() const;
+ virtual status_t start(pid_t tid);
+ virtual void stop();
virtual status_t onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
private:
- sp<RecordThread::RecordTrack> mRecordTrack;
+ const sp<RecordThread::RecordTrack> mRecordTrack;
};
//--- Audio Effect Management
@@ -1060,12 +1263,12 @@
// the attached track(s) to accumulate their auxiliary channel.
class EffectModule: public RefBase {
public:
- EffectModule(const wp<ThreadBase>& wThread,
+ EffectModule(ThreadBase *thread,
const wp<AudioFlinger::EffectChain>& chain,
effect_descriptor_t *desc,
int id,
int sessionId);
- ~EffectModule();
+ virtual ~EffectModule();
enum effect_state {
IDLE,
@@ -1077,7 +1280,7 @@
DESTROYED
};
- int id() { return mId; }
+ int id() const { return mId; }
void process();
void updateState();
status_t command(uint32_t cmdCode,
@@ -1089,18 +1292,18 @@
void reset_l();
status_t configure();
status_t init();
- uint32_t state() {
+ effect_state state() const {
return mState;
}
uint32_t status() {
return mStatus;
}
- int sessionId() {
+ int sessionId() const {
return mSessionId;
}
status_t setEnabled(bool enabled);
- bool isEnabled();
- bool isProcessEnabled();
+ bool isEnabled() const;
+ bool isProcessEnabled() const;
void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; }
int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; }
@@ -1108,10 +1311,10 @@
int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; }
void setChain(const wp<EffectChain>& chain) { mChain = chain; }
void setThread(const wp<ThreadBase>& thread) { mThread = thread; }
- wp<ThreadBase>& thread() { return mThread; }
+ const wp<ThreadBase>& thread() { return mThread; }
- status_t addHandle(sp<EffectHandle>& handle);
- void disconnect(const wp<EffectHandle>& handle, bool unpiniflast);
+ status_t addHandle(const sp<EffectHandle>& handle);
+ void disconnect(const wp<EffectHandle>& handle, bool unpinIfLast);
size_t removeHandle (const wp<EffectHandle>& handle);
effect_descriptor_t& desc() { return mDescriptor; }
@@ -1119,22 +1322,21 @@
status_t setDevice(uint32_t device);
status_t setVolume(uint32_t *left, uint32_t *right, bool controller);
- status_t setMode(uint32_t mode);
+ status_t setMode(audio_mode_t mode);
status_t start();
status_t stop();
void setSuspended(bool suspended);
- bool suspended();
+ bool suspended() const;
sp<EffectHandle> controlHandle();
- bool isPinned() { return mPinned; }
+ bool isPinned() const { return mPinned; }
void unPin() { mPinned = false; }
status_t dump(int fd, const Vector<String16>& args);
protected:
- friend class EffectHandle;
- friend class AudioFlinger;
+ friend class AudioFlinger; // for mHandles
bool mPinned;
// Maximum time allocated to effect engines to complete the turn off sequence
@@ -1146,7 +1348,7 @@
status_t start_l();
status_t stop_l();
- Mutex mLock; // mutex for process, commands and handles list protection
+mutable Mutex mLock; // mutex for process, commands and handles list protection
wp<ThreadBase> mThread; // parent thread
wp<EffectChain> mChain; // parent effect chain
int mId; // this instance unique ID
@@ -1154,9 +1356,10 @@
effect_descriptor_t mDescriptor;// effect descriptor received from effect engine
effect_config_t mConfig; // input and output audio configuration
effect_handle_t mEffectInterface; // Effect module C API
- status_t mStatus; // initialization status
- uint32_t mState; // current activation state (effect_state)
+ status_t mStatus; // initialization status
+ effect_state mState; // current activation state
Vector< wp<EffectHandle> > mHandles; // list of client handles
+ // First handle in mHandles has highest priority and controls the effect module
uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after
// sending disable command.
uint32_t mDisableWaitCnt; // current process() calls count during disable period.
@@ -1187,8 +1390,10 @@
uint32_t *replySize,
void *pReplyData);
virtual void disconnect();
- virtual void disconnect(bool unpiniflast);
- virtual sp<IMemory> getCblk() const;
+ private:
+ void disconnect(bool unpinIfLast);
+ public:
+ virtual sp<IMemory> getCblk() const { return mCblkMemory; }
virtual status_t onTransact(uint32_t code, const Parcel& data,
Parcel* reply, uint32_t flags);
@@ -1204,25 +1409,24 @@
uint32_t replySize,
void *pReplyData);
void setEnabled(bool enabled);
- bool enabled() { return mEnabled; }
+ bool enabled() const { return mEnabled; }
// Getters
- int id() { return mEffect->id(); }
- int priority() { return mPriority; }
- bool hasControl() { return mHasControl; }
- sp<EffectModule> effect() { return mEffect; }
+ int id() const { return mEffect->id(); }
+ int priority() const { return mPriority; }
+ bool hasControl() const { return mHasControl; }
+ sp<EffectModule> effect() const { return mEffect; }
void dump(char* buffer, size_t size);
protected:
- friend class AudioFlinger;
- friend class EffectModule;
+ friend class AudioFlinger; // for mEffect, mHasControl, mEnabled
EffectHandle(const EffectHandle&);
EffectHandle& operator =(const EffectHandle&);
sp<EffectModule> mEffect; // pointer to controlled EffectModule
sp<IEffectClient> mEffectClient; // callback interface for client notifications
- sp<Client> mClient; // client for shared memory allocation
+ /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect()
sp<IMemory> mCblkMemory; // shared memory for control block
effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory
uint8_t* mBuffer; // pointer to parameter area in shared memory
@@ -1242,7 +1446,8 @@
class EffectChain: public RefBase {
public:
EffectChain(const wp<ThreadBase>& wThread, int sessionId);
- ~EffectChain();
+ EffectChain(ThreadBase *thread, int sessionId);
+ virtual ~EffectChain();
// special key used for an entry in mSuspendedEffects keyed vector
// corresponding to a suspend all request.
@@ -1264,7 +1469,7 @@
status_t addEffect_l(const sp<EffectModule>& handle);
size_t removeEffect_l(const sp<EffectModule>& handle);
- int sessionId() { return mSessionId; }
+ int sessionId() const { return mSessionId; }
void setSessionId(int sessionId) { mSessionId = sessionId; }
sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor);
@@ -1272,34 +1477,34 @@
sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type);
bool setVolume_l(uint32_t *left, uint32_t *right);
void setDevice_l(uint32_t device);
- void setMode_l(uint32_t mode);
+ void setMode_l(audio_mode_t mode);
void setInBuffer(int16_t *buffer, bool ownsBuffer = false) {
mInBuffer = buffer;
mOwnInBuffer = ownsBuffer;
}
- int16_t *inBuffer() {
+ int16_t *inBuffer() const {
return mInBuffer;
}
void setOutBuffer(int16_t *buffer) {
mOutBuffer = buffer;
}
- int16_t *outBuffer() {
+ int16_t *outBuffer() const {
return mOutBuffer;
}
void incTrackCnt() { android_atomic_inc(&mTrackCnt); }
void decTrackCnt() { android_atomic_dec(&mTrackCnt); }
- int32_t trackCnt() { return mTrackCnt;}
+ int32_t trackCnt() const { return mTrackCnt;}
void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt);
mTailBufferCount = mMaxTailBuffers; }
void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); }
- int32_t activeTrackCnt() { return mActiveTrackCnt;}
+ int32_t activeTrackCnt() const { return mActiveTrackCnt;}
- uint32_t strategy() { return mStrategy; }
+ uint32_t strategy() const { return mStrategy; }
void setStrategy(uint32_t strategy)
- { mStrategy = strategy; }
+ { mStrategy = strategy; }
// suspend effect of the given type
void setEffectSuspended_l(const effect_uuid_t *type,
@@ -1313,7 +1518,7 @@
status_t dump(int fd, const Vector<String16>& args);
protected:
- friend class AudioFlinger;
+ friend class AudioFlinger; // for mThread, mEffects
EffectChain(const EffectChain&);
EffectChain& operator =(const EffectChain&);
@@ -1328,7 +1533,8 @@
// get a list of effect modules to suspend when an effect of the type
// passed is enabled.
- Vector< sp<EffectModule> > getSuspendEligibleEffects();
+ void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects);
+
// get an effect module if it is currently enable
sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type);
// true if the effect whose descriptor is passed can be suspended
@@ -1338,7 +1544,7 @@
wp<ThreadBase> mThread; // parent mixer thread
Mutex mLock; // mutex protecting effect list
- Vector<sp<EffectModule> > mEffects; // list of effect modules
+ Vector< sp<EffectModule> > mEffects; // list of effect modules
int mSessionId; // audio session ID
int16_t *mInBuffer; // chain input buffer
int16_t *mOutBuffer; // chain output buffer
@@ -1359,54 +1565,119 @@
KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects;
};
+ // AudioStreamOut and AudioStreamIn are immutable, so their fields are const.
+ // For emphasis, we could also make all pointers to them be "const *",
+ // but that would clutter the code unnecessarily.
+
struct AudioStreamOut {
- audio_hw_device_t *hwDev;
- audio_stream_out_t *stream;
+ audio_hw_device_t* const hwDev;
+ audio_stream_out_t* const stream;
AudioStreamOut(audio_hw_device_t *dev, audio_stream_out_t *out) :
hwDev(dev), stream(out) {}
};
struct AudioStreamIn {
- audio_hw_device_t *hwDev;
- audio_stream_in_t *stream;
+ audio_hw_device_t* const hwDev;
+ audio_stream_in_t* const stream;
AudioStreamIn(audio_hw_device_t *dev, audio_stream_in_t *in) :
hwDev(dev), stream(in) {}
};
+ // for mAudioSessionRefs only
struct AudioSessionRef {
- int sessionid;
- pid_t pid;
- int cnt;
+ AudioSessionRef(int sessionid, pid_t pid) :
+ mSessionid(sessionid), mPid(pid), mCnt(1) {}
+ const int mSessionid;
+ const pid_t mPid;
+ int mCnt;
};
- friend class RecordThread;
- friend class PlaybackThread;
+ enum master_volume_support {
+ // MVS_NONE:
+ // Audio HAL has no support for master volume, either setting or
+ // getting. All master volume control must be implemented in SW by the
+ // AudioFlinger mixing core.
+ MVS_NONE,
+
+ // MVS_SETONLY:
+ // Audio HAL has support for setting master volume, but not for getting
+ // master volume (original HAL design did not include a getter).
+ // AudioFlinger needs to keep track of the last set master volume in
+ // addition to needing to set an initial, default, master volume at HAL
+ // load time.
+ MVS_SETONLY,
+
+ // MVS_FULL:
+ // Audio HAL has support both for setting and getting master volume.
+ // AudioFlinger should send all set and get master volume requests
+ // directly to the HAL.
+ MVS_FULL,
+ };
mutable Mutex mLock;
- DefaultKeyedVector< pid_t, wp<Client> > mClients;
+ DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client()
mutable Mutex mHardwareLock;
- audio_hw_device_t* mPrimaryHardwareDev;
+
+ // These two fields are immutable after onFirstRef(), so no lock needed to access
+ audio_hw_device_t* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL
Vector<audio_hw_device_t*> mAudioHwDevs;
- mutable int mHardwareStatus;
+
+ // for dump, indicates which hardware operation is currently in progress (but not stream ops)
+ enum hardware_call_state {
+ AUDIO_HW_IDLE = 0, // no operation in progress
+ AUDIO_HW_INIT, // init_check
+ AUDIO_HW_OUTPUT_OPEN, // open_output_stream
+ AUDIO_HW_OUTPUT_CLOSE, // unused
+ AUDIO_HW_INPUT_OPEN, // unused
+ AUDIO_HW_INPUT_CLOSE, // unused
+ AUDIO_HW_STANDBY, // unused
+ AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume
+ AUDIO_HW_GET_ROUTING, // unused
+ AUDIO_HW_SET_ROUTING, // unused
+ AUDIO_HW_GET_MODE, // unused
+ AUDIO_HW_SET_MODE, // set_mode
+ AUDIO_HW_GET_MIC_MUTE, // get_mic_mute
+ AUDIO_HW_SET_MIC_MUTE, // set_mic_mute
+ AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume
+ AUDIO_HW_SET_PARAMETER, // set_parameters
+ AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size
+ AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume
+ AUDIO_HW_GET_PARAMETER, // get_parameters
+ };
+
+ mutable hardware_call_state mHardwareStatus; // for dump only
- DefaultKeyedVector< int, sp<PlaybackThread> > mPlaybackThreads;
- PlaybackThread::stream_type_t mStreamTypes[AUDIO_STREAM_CNT];
+ DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads;
+ stream_type_t mStreamTypes[AUDIO_STREAM_CNT];
+
+ // both are protected by mLock
float mMasterVolume;
+ float mMasterVolumeSW;
+ master_volume_support mMasterVolumeSupportLvl;
bool mMasterMute;
- DefaultKeyedVector< int, sp<RecordThread> > mRecordThreads;
+ DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads;
DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients;
- volatile int32_t mNextUniqueId;
- uint32_t mMode;
+ volatile int32_t mNextUniqueId; // updated by android_atomic_inc
+ audio_mode_t mMode;
bool mBtNrecIsOff;
+ // protected by mLock
Vector<AudioSessionRef*> mAudioSessionRefs;
+
+ float masterVolume_l() const;
+ float masterVolumeSW_l() const { return mMasterVolumeSW; }
+ bool masterMute_l() const { return mMasterMute; }
+
+private:
+ sp<Client> registerPid_l(pid_t pid); // always returns non-0
+
};
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 9dda256..1ec238b 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -1,4 +1,4 @@
-/* //device/include/server/AudioFlinger/AudioMixer.cpp
+/*
**
** Copyright 2007, The Android Open Source Project
**
@@ -27,240 +27,236 @@
#include <utils/Log.h>
#include <cutils/bitops.h>
+#include <cutils/compiler.h>
+#include <utils/Debug.h>
#include <system/audio.h>
+#include <audio_utils/primitives.h>
+#include <common_time/local_clock.h>
+#include <common_time/cc_helper.h>
+
#include "AudioMixer.h"
namespace android {
-// ----------------------------------------------------------------------------
-
-static inline int16_t clamp16(int32_t sample)
-{
- if ((sample>>15) ^ (sample>>31))
- sample = 0x7FFF ^ (sample>>31);
- return sample;
-}
// ----------------------------------------------------------------------------
AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate)
- : mActiveTrack(0), mTrackNames(0), mSampleRate(sampleRate)
+ : mTrackNames(0), mSampleRate(sampleRate)
{
+ // AudioMixer is not yet capable of multi-channel beyond stereo
+ COMPILE_TIME_ASSERT_FUNCTION_SCOPE(2 == MAX_NUM_CHANNELS);
+
+ LocalClock lc;
+
mState.enabledTracks= 0;
mState.needsChanged = 0;
mState.frameCount = frameCount;
- mState.outputTemp = 0;
- mState.resampleTemp = 0;
mState.hook = process__nop;
+ mState.outputTemp = NULL;
+ mState.resampleTemp = NULL;
+ // mState.reserved
track_t* t = mState.tracks;
- for (int i=0 ; i<32 ; i++) {
+ for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) {
t->needs = 0;
t->volume[0] = UNITY_GAIN;
t->volume[1] = UNITY_GAIN;
+ // no initialization needed
+ // t->prevVolume[0]
+ // t->prevVolume[1]
t->volumeInc[0] = 0;
t->volumeInc[1] = 0;
t->auxLevel = 0;
t->auxInc = 0;
+ // no initialization needed
+ // t->prevAuxLevel
+ // t->frameCount
t->channelCount = 2;
- t->enabled = 0;
+ t->enabled = false;
t->format = 16;
t->channelMask = AUDIO_CHANNEL_OUT_STEREO;
- t->buffer.raw = 0;
- t->bufferProvider = 0;
- t->hook = 0;
- t->resampler = 0;
+ t->bufferProvider = NULL;
+ t->buffer.raw = NULL;
+ // t->buffer.frameCount
+ t->hook = NULL;
+ t->in = NULL;
+ t->resampler = NULL;
t->sampleRate = mSampleRate;
- t->in = 0;
t->mainBuffer = NULL;
t->auxBuffer = NULL;
+ t->localTimeFreq = lc.getLocalFreq();
t++;
}
}
- AudioMixer::~AudioMixer()
- {
- track_t* t = mState.tracks;
- for (int i=0 ; i<32 ; i++) {
- delete t->resampler;
- t++;
- }
- delete [] mState.outputTemp;
- delete [] mState.resampleTemp;
- }
-
- int AudioMixer::getTrackName()
- {
- uint32_t names = mTrackNames;
- uint32_t mask = 1;
- int n = 0;
- while (names & mask) {
- mask <<= 1;
- n++;
+AudioMixer::~AudioMixer()
+{
+ track_t* t = mState.tracks;
+ for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) {
+ delete t->resampler;
+ t++;
}
- if (mask) {
+ delete [] mState.outputTemp;
+ delete [] mState.resampleTemp;
+}
+
+int AudioMixer::getTrackName()
+{
+ uint32_t names = ~mTrackNames;
+ if (names != 0) {
+ int n = __builtin_ctz(names);
ALOGV("add track (%d)", n);
- mTrackNames |= mask;
+ mTrackNames |= 1 << n;
return TRACK0 + n;
}
return -1;
- }
+}
- void AudioMixer::invalidateState(uint32_t mask)
- {
+void AudioMixer::invalidateState(uint32_t mask)
+{
if (mask) {
mState.needsChanged |= mask;
mState.hook = process__validate;
}
}
- void AudioMixer::deleteTrackName(int name)
- {
+void AudioMixer::deleteTrackName(int name)
+{
name -= TRACK0;
- if (uint32_t(name) < MAX_NUM_TRACKS) {
- ALOGV("deleteTrackName(%d)", name);
- track_t& track(mState.tracks[ name ]);
- if (track.enabled != 0) {
- track.enabled = 0;
- invalidateState(1<<name);
- }
- if (track.resampler) {
- // delete the resampler
- delete track.resampler;
- track.resampler = 0;
- track.sampleRate = mSampleRate;
- invalidateState(1<<name);
- }
- track.volumeInc[0] = 0;
- track.volumeInc[1] = 0;
- mTrackNames &= ~(1<<name);
+ ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
+ ALOGV("deleteTrackName(%d)", name);
+ track_t& track(mState.tracks[ name ]);
+ if (track.enabled) {
+ track.enabled = false;
+ invalidateState(1<<name);
}
- }
-
-status_t AudioMixer::enable(int name)
-{
- switch (name) {
- case MIXING: {
- if (mState.tracks[ mActiveTrack ].enabled != 1) {
- mState.tracks[ mActiveTrack ].enabled = 1;
- ALOGV("enable(%d)", mActiveTrack);
- invalidateState(1<<mActiveTrack);
- }
- } break;
- default:
- return NAME_NOT_FOUND;
+ if (track.resampler != NULL) {
+ // delete the resampler
+ delete track.resampler;
+ track.resampler = NULL;
+ track.sampleRate = mSampleRate;
+ invalidateState(1<<name);
}
- return NO_ERROR;
+ track.volumeInc[0] = 0;
+ track.volumeInc[1] = 0;
+ mTrackNames &= ~(1<<name);
}
-status_t AudioMixer::disable(int name)
+void AudioMixer::enable(int name)
{
- switch (name) {
- case MIXING: {
- if (mState.tracks[ mActiveTrack ].enabled != 0) {
- mState.tracks[ mActiveTrack ].enabled = 0;
- ALOGV("disable(%d)", mActiveTrack);
- invalidateState(1<<mActiveTrack);
- }
- } break;
- default:
- return NAME_NOT_FOUND;
+ name -= TRACK0;
+ ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
+ track_t& track = mState.tracks[name];
+
+ if (!track.enabled) {
+ track.enabled = true;
+ ALOGV("enable(%d)", name);
+ invalidateState(1 << name);
}
- return NO_ERROR;
}
-status_t AudioMixer::setActiveTrack(int track)
+void AudioMixer::disable(int name)
{
- if (uint32_t(track-TRACK0) >= MAX_NUM_TRACKS) {
- return BAD_VALUE;
+ name -= TRACK0;
+ ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
+ track_t& track = mState.tracks[name];
+
+ if (track.enabled) {
+ track.enabled = false;
+ ALOGV("disable(%d)", name);
+ invalidateState(1 << name);
}
- mActiveTrack = track - TRACK0;
- return NO_ERROR;
}
-status_t AudioMixer::setParameter(int target, int name, void *value)
+void AudioMixer::setParameter(int name, int target, int param, void *value)
{
+ name -= TRACK0;
+ ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
+ track_t& track = mState.tracks[name];
+
int valueInt = (int)value;
int32_t *valueBuf = (int32_t *)value;
switch (target) {
- case TRACK:
- if (name == CHANNEL_MASK) {
- uint32_t mask = (uint32_t)value;
- if (mState.tracks[ mActiveTrack ].channelMask != mask) {
- uint8_t channelCount = popcount(mask);
- if ((channelCount <= MAX_NUM_CHANNELS) && (channelCount)) {
- mState.tracks[ mActiveTrack ].channelMask = mask;
- mState.tracks[ mActiveTrack ].channelCount = channelCount;
- ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", mask);
- invalidateState(1<<mActiveTrack);
- return NO_ERROR;
- }
- } else {
- return NO_ERROR;
- }
- }
- if (name == MAIN_BUFFER) {
- if (mState.tracks[ mActiveTrack ].mainBuffer != valueBuf) {
- mState.tracks[ mActiveTrack ].mainBuffer = valueBuf;
- ALOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf);
- invalidateState(1<<mActiveTrack);
- }
- return NO_ERROR;
- }
- if (name == AUX_BUFFER) {
- if (mState.tracks[ mActiveTrack ].auxBuffer != valueBuf) {
- mState.tracks[ mActiveTrack ].auxBuffer = valueBuf;
- ALOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf);
- invalidateState(1<<mActiveTrack);
- }
- return NO_ERROR;
- }
- break;
- case RESAMPLE:
- if (name == SAMPLE_RATE) {
- if (valueInt > 0) {
- track_t& track = mState.tracks[ mActiveTrack ];
- if (track.setResampler(uint32_t(valueInt), mSampleRate)) {
- ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
- uint32_t(valueInt));
- invalidateState(1<<mActiveTrack);
- }
- return NO_ERROR;
+ case TRACK:
+ switch (param) {
+ case CHANNEL_MASK: {
+ uint32_t mask = (uint32_t)value;
+ if (track.channelMask != mask) {
+ uint32_t channelCount = popcount(mask);
+ ALOG_ASSERT((channelCount <= MAX_NUM_CHANNELS) && (channelCount),
+ "bad channel count %u", channelCount);
+ track.channelMask = mask;
+ track.channelCount = channelCount;
+ ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", mask);
+ invalidateState(1 << name);
}
- }
- if (name == RESET) {
- track_t& track = mState.tracks[ mActiveTrack ];
- track.resetResampler();
- invalidateState(1<<mActiveTrack);
- return NO_ERROR;
+ } break;
+ case MAIN_BUFFER:
+ if (track.mainBuffer != valueBuf) {
+ track.mainBuffer = valueBuf;
+ ALOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf);
+ invalidateState(1 << name);
+ }
+ break;
+ case AUX_BUFFER:
+ if (track.auxBuffer != valueBuf) {
+ track.auxBuffer = valueBuf;
+ ALOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf);
+ invalidateState(1 << name);
+ }
+ break;
+ default:
+ LOG_FATAL("bad param");
}
break;
+
+ case RESAMPLE:
+ switch (param) {
+ case SAMPLE_RATE:
+ ALOG_ASSERT(valueInt > 0, "bad sample rate %d", valueInt);
+ if (track.setResampler(uint32_t(valueInt), mSampleRate)) {
+ ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
+ uint32_t(valueInt));
+ invalidateState(1 << name);
+ }
+ break;
+ case RESET:
+ track.resetResampler();
+ invalidateState(1 << name);
+ break;
+ default:
+ LOG_FATAL("bad param");
+ }
+ break;
+
case RAMP_VOLUME:
case VOLUME:
- if ((uint32_t(name-VOLUME0) < MAX_NUM_CHANNELS)) {
- track_t& track = mState.tracks[ mActiveTrack ];
- if (track.volume[name-VOLUME0] != valueInt) {
+ switch (param) {
+ case VOLUME0:
+ case VOLUME1:
+ if (track.volume[param-VOLUME0] != valueInt) {
ALOGV("setParameter(VOLUME, VOLUME0/1: %04x)", valueInt);
- track.prevVolume[name-VOLUME0] = track.volume[name-VOLUME0] << 16;
- track.volume[name-VOLUME0] = valueInt;
+ track.prevVolume[param-VOLUME0] = track.volume[param-VOLUME0] << 16;
+ track.volume[param-VOLUME0] = valueInt;
if (target == VOLUME) {
- track.prevVolume[name-VOLUME0] = valueInt << 16;
- track.volumeInc[name-VOLUME0] = 0;
+ track.prevVolume[param-VOLUME0] = valueInt << 16;
+ track.volumeInc[param-VOLUME0] = 0;
} else {
- int32_t d = (valueInt<<16) - track.prevVolume[name-VOLUME0];
+ int32_t d = (valueInt<<16) - track.prevVolume[param-VOLUME0];
int32_t volInc = d / int32_t(mState.frameCount);
- track.volumeInc[name-VOLUME0] = volInc;
+ track.volumeInc[param-VOLUME0] = volInc;
if (volInc == 0) {
- track.prevVolume[name-VOLUME0] = valueInt << 16;
+ track.prevVolume[param-VOLUME0] = valueInt << 16;
}
}
- invalidateState(1<<mActiveTrack);
+ invalidateState(1 << name);
}
- return NO_ERROR;
- } else if (name == AUXLEVEL) {
- track_t& track = mState.tracks[ mActiveTrack ];
+ break;
+ case AUXLEVEL:
+ //ALOG_ASSERT(0 <= valueInt && valueInt <= MAX_GAIN_INT, "bad aux level %d", valueInt);
if (track.auxLevel != valueInt) {
ALOGV("setParameter(VOLUME, AUXLEVEL: %04x)", valueInt);
track.prevAuxLevel = track.auxLevel << 16;
@@ -276,13 +272,17 @@
track.prevAuxLevel = valueInt << 16;
}
}
- invalidateState(1<<mActiveTrack);
+ invalidateState(1 << name);
}
- return NO_ERROR;
+ break;
+ default:
+ LOG_FATAL("bad param");
}
break;
+
+ default:
+ LOG_FATAL("bad target");
}
- return BAD_VALUE;
}
bool AudioMixer::track_t::setResampler(uint32_t value, uint32_t devSampleRate)
@@ -290,9 +290,10 @@
if (value!=devSampleRate || resampler) {
if (sampleRate != value) {
sampleRate = value;
- if (resampler == 0) {
+ if (resampler == NULL) {
resampler = AudioResampler::create(
format, channelCount, devSampleRate);
+ resampler->setLocalTimeFreq(localTimeFreq);
}
return true;
}
@@ -300,22 +301,10 @@
return false;
}
-bool AudioMixer::track_t::doesResample() const
-{
- return resampler != 0;
-}
-
-void AudioMixer::track_t::resetResampler()
-{
- if (resampler != 0) {
- resampler->reset();
- }
-}
-
inline
void AudioMixer::track_t::adjustVolumeRamp(bool aux)
{
- for (int i=0 ; i<2 ; i++) {
+ for (uint32_t i=0 ; i<MAX_NUM_CHANNELS ; i++) {
if (((volumeInc[i]>0) && (((prevVolume[i]+volumeInc[i])>>16) >= volume[i])) ||
((volumeInc[i]<0) && (((prevVolume[i]+volumeInc[i])>>16) <= volume[i]))) {
volumeInc[i] = 0;
@@ -331,39 +320,31 @@
}
}
-size_t AudioMixer::track_t::getUnreleasedFrames()
-{
- if (resampler != NULL) {
- return resampler->getUnreleasedFrames();
- }
- return 0;
-}
-
-size_t AudioMixer::getUnreleasedFrames(int name)
+size_t AudioMixer::getUnreleasedFrames(int name) const
{
name -= TRACK0;
if (uint32_t(name) < MAX_NUM_TRACKS) {
- track_t& track(mState.tracks[name]);
- return track.getUnreleasedFrames();
+ return mState.tracks[name].getUnreleasedFrames();
}
return 0;
}
-status_t AudioMixer::setBufferProvider(AudioBufferProvider* buffer)
+void AudioMixer::setBufferProvider(int name, AudioBufferProvider* bufferProvider)
{
- mState.tracks[ mActiveTrack ].bufferProvider = buffer;
- return NO_ERROR;
+ name -= TRACK0;
+ ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
+ mState.tracks[name].bufferProvider = bufferProvider;
}
-void AudioMixer::process()
+void AudioMixer::process(int64_t pts)
{
- mState.hook(&mState);
+ mState.hook(&mState, pts);
}
-void AudioMixer::process__validate(state_t* state)
+void AudioMixer::process__validate(state_t* state, int64_t pts)
{
ALOGW_IF(!state->needsChanged,
"in process__validate() but nothing's invalid");
@@ -386,9 +367,9 @@
// compute everything we need...
int countActiveTracks = 0;
- int all16BitsStereoNoResample = 1;
- int resampling = 0;
- int volumeRamp = 0;
+ bool all16BitsStereoNoResample = true;
+ bool resampling = false;
+ bool volumeRamp = false;
uint32_t en = state->enabledTracks;
while (en) {
const int i = 31 - __builtin_clz(en);
@@ -405,7 +386,7 @@
}
if (t.volumeInc[0]|t.volumeInc[1]) {
- volumeRamp = 1;
+ volumeRamp = true;
} else if (!t.doesResample() && t.volumeRL == 0) {
n |= NEEDS_MUTE_ENABLED;
}
@@ -415,16 +396,16 @@
t.hook = track__nop;
} else {
if ((n & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) {
- all16BitsStereoNoResample = 0;
+ all16BitsStereoNoResample = false;
}
if ((n & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) {
- all16BitsStereoNoResample = 0;
- resampling = 1;
+ all16BitsStereoNoResample = false;
+ resampling = true;
t.hook = track__genericResample;
} else {
if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){
t.hook = track__16BitsMono;
- all16BitsStereoNoResample = 0;
+ all16BitsStereoNoResample = false;
}
if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_2){
t.hook = track__16BitsStereo;
@@ -447,11 +428,11 @@
} else {
if (state->outputTemp) {
delete [] state->outputTemp;
- state->outputTemp = 0;
+ state->outputTemp = NULL;
}
if (state->resampleTemp) {
delete [] state->resampleTemp;
- state->resampleTemp = 0;
+ state->resampleTemp = NULL;
}
state->hook = process__genericNoResampling;
if (all16BitsStereoNoResample && !volumeRamp) {
@@ -467,115 +448,33 @@
countActiveTracks, state->enabledTracks,
all16BitsStereoNoResample, resampling, volumeRamp);
- state->hook(state);
+ state->hook(state, pts);
- // Now that the volume ramp has been done, set optimal state and
- // track hooks for subsequent mixer process
- if (countActiveTracks) {
- int allMuted = 1;
- uint32_t en = state->enabledTracks;
- while (en) {
- const int i = 31 - __builtin_clz(en);
- en &= ~(1<<i);
- track_t& t = state->tracks[i];
- if (!t.doesResample() && t.volumeRL == 0)
- {
- t.needs |= NEEDS_MUTE_ENABLED;
- t.hook = track__nop;
- } else {
- allMuted = 0;
- }
- }
- if (allMuted) {
- state->hook = process__nop;
- } else if (all16BitsStereoNoResample) {
- if (countActiveTracks == 1) {
- state->hook = process__OneTrack16BitsStereoNoResampling;
- }
- }
- }
-}
-
-static inline
-int32_t mulAdd(int16_t in, int16_t v, int32_t a)
-{
-#if defined(__arm__) && !defined(__thumb__)
- int32_t out;
- asm( "smlabb %[out], %[in], %[v], %[a] \n"
- : [out]"=r"(out)
- : [in]"%r"(in), [v]"r"(v), [a]"r"(a)
- : );
- return out;
-#else
- return a + in * int32_t(v);
-#endif
-}
-
-static inline
-int32_t mul(int16_t in, int16_t v)
-{
-#if defined(__arm__) && !defined(__thumb__)
- int32_t out;
- asm( "smulbb %[out], %[in], %[v] \n"
- : [out]"=r"(out)
- : [in]"%r"(in), [v]"r"(v)
- : );
- return out;
-#else
- return in * int32_t(v);
-#endif
-}
-
-static inline
-int32_t mulAddRL(int left, uint32_t inRL, uint32_t vRL, int32_t a)
-{
-#if defined(__arm__) && !defined(__thumb__)
- int32_t out;
- if (left) {
- asm( "smlabb %[out], %[inRL], %[vRL], %[a] \n"
- : [out]"=r"(out)
- : [inRL]"%r"(inRL), [vRL]"r"(vRL), [a]"r"(a)
- : );
- } else {
- asm( "smlatt %[out], %[inRL], %[vRL], %[a] \n"
- : [out]"=r"(out)
- : [inRL]"%r"(inRL), [vRL]"r"(vRL), [a]"r"(a)
- : );
+ // Now that the volume ramp has been done, set optimal state and
+ // track hooks for subsequent mixer process
+ if (countActiveTracks) {
+ bool allMuted = true;
+ uint32_t en = state->enabledTracks;
+ while (en) {
+ const int i = 31 - __builtin_clz(en);
+ en &= ~(1<<i);
+ track_t& t = state->tracks[i];
+ if (!t.doesResample() && t.volumeRL == 0)
+ {
+ t.needs |= NEEDS_MUTE_ENABLED;
+ t.hook = track__nop;
+ } else {
+ allMuted = false;
+ }
+ }
+ if (allMuted) {
+ state->hook = process__nop;
+ } else if (all16BitsStereoNoResample) {
+ if (countActiveTracks == 1) {
+ state->hook = process__OneTrack16BitsStereoNoResampling;
+ }
+ }
}
- return out;
-#else
- if (left) {
- return a + int16_t(inRL&0xFFFF) * int16_t(vRL&0xFFFF);
- } else {
- return a + int16_t(inRL>>16) * int16_t(vRL>>16);
- }
-#endif
-}
-
-static inline
-int32_t mulRL(int left, uint32_t inRL, uint32_t vRL)
-{
-#if defined(__arm__) && !defined(__thumb__)
- int32_t out;
- if (left) {
- asm( "smulbb %[out], %[inRL], %[vRL] \n"
- : [out]"=r"(out)
- : [inRL]"%r"(inRL), [vRL]"r"(vRL)
- : );
- } else {
- asm( "smultt %[out], %[inRL], %[vRL] \n"
- : [out]"=r"(out)
- : [inRL]"%r"(inRL), [vRL]"r"(vRL)
- : );
- }
- return out;
-#else
- if (left) {
- return int16_t(inRL&0xFFFF) * int16_t(vRL&0xFFFF);
- } else {
- return int16_t(inRL>>16) * int16_t(vRL>>16);
- }
-#endif
}
@@ -591,13 +490,13 @@
t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN);
memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
t->resampler->resample(temp, outFrameCount, t->bufferProvider);
- if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc) {
+ if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
volumeRampStereo(t, out, outFrameCount, temp, aux);
} else {
volumeStereo(t, out, outFrameCount, temp, aux);
}
} else {
- if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]) {
+ if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) {
t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN);
memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
t->resampler->resample(temp, outFrameCount, t->bufferProvider);
@@ -628,7 +527,7 @@
// (vl + vlInc*frameCount)/65536.0f, frameCount);
// ramp volume
- if UNLIKELY(aux != NULL) {
+ if (CC_UNLIKELY(aux != NULL)) {
int32_t va = t->prevAuxLevel;
const int32_t vaInc = t->auxInc;
int32_t l;
@@ -655,7 +554,7 @@
}
t->prevVolume[0] = vl;
t->prevVolume[1] = vr;
- t->adjustVolumeRamp((aux != NULL));
+ t->adjustVolumeRamp(aux != NULL);
}
void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
@@ -663,8 +562,8 @@
const int16_t vl = t->volume[0];
const int16_t vr = t->volume[1];
- if UNLIKELY(aux != NULL) {
- const int16_t va = (int16_t)t->auxLevel;
+ if (CC_UNLIKELY(aux != NULL)) {
+ const int16_t va = t->auxLevel;
do {
int16_t l = (int16_t)(*temp++ >> 12);
int16_t r = (int16_t)(*temp++ >> 12);
@@ -688,13 +587,13 @@
void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
{
- int16_t const *in = static_cast<int16_t const *>(t->in);
+ const int16_t *in = static_cast<const int16_t *>(t->in);
- if UNLIKELY(aux != NULL) {
+ if (CC_UNLIKELY(aux != NULL)) {
int32_t l;
int32_t r;
// ramp gain
- if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc) {
+ if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
int32_t vl = t->prevVolume[0];
int32_t vr = t->prevVolume[1];
int32_t va = t->prevAuxLevel;
@@ -727,7 +626,7 @@
const uint32_t vrl = t->volumeRL;
const int16_t va = (int16_t)t->auxLevel;
do {
- uint32_t rl = *reinterpret_cast<uint32_t const *>(in);
+ uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
int16_t a = (int16_t)(((int32_t)in[0] + in[1]) >> 1);
in += 2;
out[0] = mulAddRL(1, rl, vrl, out[0]);
@@ -739,7 +638,7 @@
}
} else {
// ramp gain
- if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]) {
+ if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) {
int32_t vl = t->prevVolume[0];
int32_t vr = t->prevVolume[1];
const int32_t vlInc = t->volumeInc[0];
@@ -765,7 +664,7 @@
else {
const uint32_t vrl = t->volumeRL;
do {
- uint32_t rl = *reinterpret_cast<uint32_t const *>(in);
+ uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
in += 2;
out[0] = mulAddRL(1, rl, vrl, out[0]);
out[1] = mulAddRL(0, rl, vrl, out[1]);
@@ -778,11 +677,11 @@
void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
{
- int16_t const *in = static_cast<int16_t const *>(t->in);
+ const int16_t *in = static_cast<int16_t const *>(t->in);
- if UNLIKELY(aux != NULL) {
+ if (CC_UNLIKELY(aux != NULL)) {
// ramp gain
- if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc) {
+ if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
int32_t vl = t->prevVolume[0];
int32_t vr = t->prevVolume[1];
int32_t va = t->prevAuxLevel;
@@ -825,7 +724,7 @@
}
} else {
// ramp gain
- if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]) {
+ if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) {
int32_t vl = t->prevVolume[0];
int32_t vr = t->prevVolume[1];
const int32_t vlInc = t->volumeInc[0];
@@ -862,21 +761,8 @@
t->in = in;
}
-void AudioMixer::ditherAndClamp(int32_t* out, int32_t const *sums, size_t c)
-{
- for (size_t i=0 ; i<c ; i++) {
- int32_t l = *sums++;
- int32_t r = *sums++;
- int32_t nl = l >> 12;
- int32_t nr = r >> 12;
- l = clamp16(nl);
- r = clamp16(nr);
- *out++ = (r<<16) | (l & 0xFFFF);
- }
-}
-
// no-op case
-void AudioMixer::process__nop(state_t* state)
+void AudioMixer::process__nop(state_t* state, int64_t pts)
{
uint32_t e0 = state->enabledTracks;
size_t bufSize = state->frameCount * sizeof(int16_t) * MAX_NUM_CHANNELS;
@@ -891,7 +777,7 @@
i = 31 - __builtin_clz(e2);
e2 &= ~(1<<i);
track_t& t2 = state->tracks[i];
- if UNLIKELY(t2.mainBuffer != t1.mainBuffer) {
+ if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) {
e1 &= ~(1<<i);
}
}
@@ -906,8 +792,10 @@
size_t outFrames = state->frameCount;
while (outFrames) {
t1.buffer.frameCount = outFrames;
- t1.bufferProvider->getNextBuffer(&t1.buffer);
- if (!t1.buffer.raw) break;
+ int64_t outputPTS = calculateOutputPTS(
+ t1, pts, state->frameCount - outFrames);
+ t1.bufferProvider->getNextBuffer(&t1.buffer, outputPTS);
+ if (t1.buffer.raw == NULL) break;
outFrames -= t1.buffer.frameCount;
t1.bufferProvider->releaseBuffer(&t1.buffer);
}
@@ -916,7 +804,7 @@
}
// generic code without resampling
-void AudioMixer::process__genericNoResampling(state_t* state)
+void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts)
{
int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32)));
@@ -928,7 +816,7 @@
e0 &= ~(1<<i);
track_t& t = state->tracks[i];
t.buffer.frameCount = state->frameCount;
- t.bufferProvider->getNextBuffer(&t.buffer);
+ t.bufferProvider->getNextBuffer(&t.buffer, pts);
t.frameCount = t.buffer.frameCount;
t.in = t.buffer.raw;
// t.in == NULL can happen if the track was flushed just after having
@@ -949,7 +837,7 @@
j = 31 - __builtin_clz(e2);
e2 &= ~(1<<j);
track_t& t2 = state->tracks[j];
- if UNLIKELY(t2.mainBuffer != t1.mainBuffer) {
+ if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) {
e1 &= ~(1<<j);
}
}
@@ -966,23 +854,25 @@
track_t& t = state->tracks[i];
size_t outFrames = BLOCKSIZE;
int32_t *aux = NULL;
- if UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) {
+ if (CC_UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED)) {
aux = t.auxBuffer + numFrames;
}
while (outFrames) {
size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount;
if (inFrames) {
- (t.hook)(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames, state->resampleTemp, aux);
+ t.hook(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames, state->resampleTemp, aux);
t.frameCount -= inFrames;
outFrames -= inFrames;
- if UNLIKELY(aux != NULL) {
+ if (CC_UNLIKELY(aux != NULL)) {
aux += inFrames;
}
}
if (t.frameCount == 0 && outFrames) {
t.bufferProvider->releaseBuffer(&t.buffer);
t.buffer.frameCount = (state->frameCount - numFrames) - (BLOCKSIZE - outFrames);
- t.bufferProvider->getNextBuffer(&t.buffer);
+ int64_t outputPTS = calculateOutputPTS(
+ t, pts, numFrames + (BLOCKSIZE - outFrames));
+ t.bufferProvider->getNextBuffer(&t.buffer, outputPTS);
t.in = t.buffer.raw;
if (t.in == NULL) {
enabledTracks &= ~(1<<i);
@@ -1010,9 +900,10 @@
}
- // generic code with resampling
-void AudioMixer::process__genericResampling(state_t* state)
+// generic code with resampling
+void AudioMixer::process__genericResampling(state_t* state, int64_t pts)
{
+ // this const just means that local variable outTemp doesn't change
int32_t* const outTemp = state->outputTemp;
const size_t size = sizeof(int32_t) * MAX_NUM_CHANNELS * state->frameCount;
@@ -1030,7 +921,7 @@
j = 31 - __builtin_clz(e2);
e2 &= ~(1<<j);
track_t& t2 = state->tracks[j];
- if UNLIKELY(t2.mainBuffer != t1.mainBuffer) {
+ if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) {
e1 &= ~(1<<j);
}
}
@@ -1042,7 +933,7 @@
e1 &= ~(1<<i);
track_t& t = state->tracks[i];
int32_t *aux = NULL;
- if UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) {
+ if (CC_UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED)) {
aux = t.auxBuffer;
}
@@ -1050,23 +941,25 @@
// acquire/release the buffers because it's done by
// the resampler.
if ((t.needs & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) {
- (t.hook)(&t, outTemp, numFrames, state->resampleTemp, aux);
+ t.resampler->setPTS(pts);
+ t.hook(&t, outTemp, numFrames, state->resampleTemp, aux);
} else {
size_t outFrames = 0;
while (outFrames < numFrames) {
t.buffer.frameCount = numFrames - outFrames;
- t.bufferProvider->getNextBuffer(&t.buffer);
+ int64_t outputPTS = calculateOutputPTS(t, pts, outFrames);
+ t.bufferProvider->getNextBuffer(&t.buffer, outputPTS);
t.in = t.buffer.raw;
// t.in == NULL can happen if the track was flushed just after having
// been enabled for mixing.
if (t.in == NULL) break;
- if UNLIKELY(aux != NULL) {
+ if (CC_UNLIKELY(aux != NULL)) {
aux += outFrames;
}
- (t.hook)(&t, outTemp + outFrames*MAX_NUM_CHANNELS, t.buffer.frameCount, state->resampleTemp, aux);
+ t.hook(&t, outTemp + outFrames*MAX_NUM_CHANNELS, t.buffer.frameCount, state->resampleTemp, aux);
outFrames += t.buffer.frameCount;
t.bufferProvider->releaseBuffer(&t.buffer);
}
@@ -1077,9 +970,15 @@
}
// one track, 16 bits stereo without resampling is the most common case
-void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state)
+void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state,
+ int64_t pts)
{
+ // This method is only called when state->enabledTracks has exactly
+ // one bit set. The asserts below would verify this, but are commented out
+ // since the whole point of this method is to optimize performance.
+ //ALOG_ASSERT(0 != state->enabledTracks, "no tracks enabled");
const int i = 31 - __builtin_clz(state->enabledTracks);
+ //ALOG_ASSERT((1 << i) == state->enabledTracks, "more than 1 track enabled");
const track_t& t = state->tracks[i];
AudioBufferProvider::Buffer& b(t.buffer);
@@ -1092,8 +991,9 @@
const uint32_t vrl = t.volumeRL;
while (numFrames) {
b.frameCount = numFrames;
- t.bufferProvider->getNextBuffer(&b);
- int16_t const *in = b.i16;
+ int64_t outputPTS = calculateOutputPTS(t, pts, out - t.mainBuffer);
+ t.bufferProvider->getNextBuffer(&b, outputPTS);
+ const int16_t *in = b.i16;
// in == NULL can happen if the track was flushed just after having
// been enabled for mixing.
@@ -1105,11 +1005,11 @@
}
size_t outFrames = b.frameCount;
- if (UNLIKELY(uint32_t(vl) > UNITY_GAIN || uint32_t(vr) > UNITY_GAIN)) {
+ if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN || uint32_t(vr) > UNITY_GAIN)) {
// volume is boosted, so we might need to clamp even though
// we process only one track.
do {
- uint32_t rl = *reinterpret_cast<uint32_t const *>(in);
+ uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
in += 2;
int32_t l = mulRL(1, rl, vrl) >> 12;
int32_t r = mulRL(0, rl, vrl) >> 12;
@@ -1120,7 +1020,7 @@
} while (--outFrames);
} else {
do {
- uint32_t rl = *reinterpret_cast<uint32_t const *>(in);
+ uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
in += 2;
int32_t l = mulRL(1, rl, vrl) >> 12;
int32_t r = mulRL(0, rl, vrl) >> 12;
@@ -1132,10 +1032,12 @@
}
}
+#if 0
// 2 tracks is also a common case
// NEVER used in current implementation of process__validate()
// only use if the 2 tracks have the same output buffer
-void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state)
+void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state,
+ int64_t pts)
{
int i;
uint32_t en = state->enabledTracks;
@@ -1149,12 +1051,12 @@
const track_t& t1 = state->tracks[i];
AudioBufferProvider::Buffer& b1(t1.buffer);
- int16_t const *in0;
+ const int16_t *in0;
const int16_t vl0 = t0.volume[0];
const int16_t vr0 = t0.volume[1];
size_t frameCount0 = 0;
- int16_t const *in1;
+ const int16_t *in1;
const int16_t vl1 = t1.volume[0];
const int16_t vr1 = t1.volume[1];
size_t frameCount1 = 0;
@@ -1162,14 +1064,16 @@
//FIXME: only works if two tracks use same buffer
int32_t* out = t0.mainBuffer;
size_t numFrames = state->frameCount;
- int16_t const *buff = NULL;
+ const int16_t *buff = NULL;
while (numFrames) {
if (frameCount0 == 0) {
b0.frameCount = numFrames;
- t0.bufferProvider->getNextBuffer(&b0);
+ int64_t outputPTS = calculateOutputPTS(t0, pts,
+ out - t0.mainBuffer);
+ t0.bufferProvider->getNextBuffer(&b0, outputPTS);
if (b0.i16 == NULL) {
if (buff == NULL) {
buff = new int16_t[MAX_NUM_CHANNELS * state->frameCount];
@@ -1183,14 +1087,16 @@
}
if (frameCount1 == 0) {
b1.frameCount = numFrames;
- t1.bufferProvider->getNextBuffer(&b1);
+ int64_t outputPTS = calculateOutputPTS(t1, pts,
+ out - t0.mainBuffer);
+ t1.bufferProvider->getNextBuffer(&b1, outputPTS);
if (b1.i16 == NULL) {
if (buff == NULL) {
buff = new int16_t[MAX_NUM_CHANNELS * state->frameCount];
}
in1 = buff;
b1.frameCount = numFrames;
- } else {
+ } else {
in1 = b1.i16;
}
frameCount1 = b1.frameCount;
@@ -1225,11 +1131,18 @@
}
}
- if (buff != NULL) {
- delete [] buff;
- }
+ delete [] buff;
+}
+#endif
+
+int64_t AudioMixer::calculateOutputPTS(const track_t& t, int64_t basePTS,
+ int outputFrameIndex)
+{
+ if (AudioBufferProvider::kInvalidPTS == basePTS)
+ return AudioBufferProvider::kInvalidPTS;
+
+ return basePTS + ((outputFrameIndex * t.localTimeFreq) / t.sampleRate);
}
// ----------------------------------------------------------------------------
}; // namespace android
-
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index 0137185..b210212 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -1,4 +1,4 @@
-/* //device/include/server/AudioFlinger/AudioMixer.h
+/*
**
** Copyright 2007, The Android Open Source Project
**
@@ -28,17 +28,12 @@
// ----------------------------------------------------------------------------
-#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
-#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
-
-// ----------------------------------------------------------------------------
-
class AudioMixer
{
public:
AudioMixer(size_t frameCount, uint32_t sampleRate);
- ~AudioMixer();
+ /*virtual*/ ~AudioMixer(); // non-virtual saves a v-table, restore if sub-classed
static const uint32_t MAX_NUM_TRACKS = 32;
static const uint32_t MAX_NUM_CHANNELS = 2;
@@ -47,11 +42,10 @@
enum { // names
- // track units (32 units)
+ // track names (MAX_NUM_TRACKS units)
TRACK0 = 0x1000,
- // enable/disable
- MIXING = 0x2000,
+ // 0x2000 is unused
// setParameter targets
TRACK = 0x3000,
@@ -65,33 +59,31 @@
FORMAT = 0x4001,
MAIN_BUFFER = 0x4002,
AUX_BUFFER = 0x4003,
- // for TARGET RESAMPLE
+ // for target RESAMPLE
SAMPLE_RATE = 0x4100,
RESET = 0x4101,
- // for TARGET VOLUME (8 channels max)
+ // for target RAMP_VOLUME and VOLUME (8 channels max)
VOLUME0 = 0x4200,
VOLUME1 = 0x4201,
AUXLEVEL = 0x4210,
};
+ // For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS
int getTrackName();
void deleteTrackName(int name);
- status_t enable(int name);
- status_t disable(int name);
+ void enable(int name);
+ void disable(int name);
- status_t setActiveTrack(int track);
- status_t setParameter(int target, int name, void *value);
+ void setParameter(int name, int target, int param, void *value);
- status_t setBufferProvider(AudioBufferProvider* bufferProvider);
- void process();
+ void setBufferProvider(int name, AudioBufferProvider* bufferProvider);
+ void process(int64_t pts);
uint32_t trackNames() const { return mTrackNames; }
- static void ditherAndClamp(int32_t* out, int32_t const *sums, size_t c);
-
- size_t getUnreleasedFrames(int name);
+ size_t getUnreleasedFrames(int name) const;
private:
@@ -119,15 +111,9 @@
NEEDS_AUX_ENABLED = 0x00010000,
};
- static inline int32_t applyVolume(int32_t in, int32_t v) {
- return in * v;
- }
-
-
struct state_t;
struct track_t;
- typedef void (*mix_t)(state_t* state);
typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux);
static const int BLOCKSIZE = 16; // 4 cache lines
@@ -135,41 +121,58 @@
uint32_t needs;
union {
- int16_t volume[2]; // [0]3.12 fixed point
+ int16_t volume[MAX_NUM_CHANNELS]; // [0]3.12 fixed point
int32_t volumeRL;
};
- int32_t prevVolume[2];
+ int32_t prevVolume[MAX_NUM_CHANNELS];
- int32_t volumeInc[2];
- int32_t auxLevel;
+ // 16-byte boundary
+
+ int32_t volumeInc[MAX_NUM_CHANNELS];
int32_t auxInc;
int32_t prevAuxLevel;
+ // 16-byte boundary
+
+ int16_t auxLevel; // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
uint16_t frameCount;
- uint8_t channelCount : 4;
- uint8_t enabled : 1;
- uint8_t reserved0 : 3;
- uint8_t format;
- uint32_t channelMask;
+ uint8_t channelCount; // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
+ uint8_t format; // always 16
+ uint16_t enabled; // actually bool
+ uint32_t channelMask; // currently under-used
AudioBufferProvider* bufferProvider;
- mutable AudioBufferProvider::Buffer buffer;
+
+ // 16-byte boundary
+
+ mutable AudioBufferProvider::Buffer buffer; // 8 bytes
hook_t hook;
- void const* in; // current location in buffer
+ const void* in; // current location in buffer
+
+ // 16-byte boundary
AudioResampler* resampler;
uint32_t sampleRate;
int32_t* mainBuffer;
int32_t* auxBuffer;
+ // 16-byte boundary
+
+ uint64_t localTimeFreq;
+
+ int64_t padding;
+
+ // 16-byte boundary
+
bool setResampler(uint32_t sampleRate, uint32_t devSampleRate);
- bool doesResample() const;
- void resetResampler();
+ bool doesResample() const { return resampler != NULL; }
+ void resetResampler() { if (resampler != NULL) resampler->reset(); }
void adjustVolumeRamp(bool aux);
- size_t getUnreleasedFrames();
+ size_t getUnreleasedFrames() const { return resampler != NULL ?
+ resampler->getUnreleasedFrames() : 0; };
};
// pad to 32-bytes to fill cache line
@@ -177,14 +180,14 @@
uint32_t enabledTracks;
uint32_t needsChanged;
size_t frameCount;
- mix_t hook;
+ void (*hook)(state_t* state, int64_t pts); // one of process__*, never NULL
int32_t *outputTemp;
int32_t *resampleTemp;
int32_t reserved[2];
- track_t tracks[32]; __attribute__((aligned(32)));
+ track_t tracks[MAX_NUM_TRACKS]; __attribute__((aligned(32)));
};
- int mActiveTrack;
+ // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc.
uint32_t mTrackNames;
const uint32_t mSampleRate;
@@ -199,12 +202,19 @@
static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
- static void process__validate(state_t* state);
- static void process__nop(state_t* state);
- static void process__genericNoResampling(state_t* state);
- static void process__genericResampling(state_t* state);
- static void process__OneTrack16BitsStereoNoResampling(state_t* state);
- static void process__TwoTracks16BitsStereoNoResampling(state_t* state);
+ static void process__validate(state_t* state, int64_t pts);
+ static void process__nop(state_t* state, int64_t pts);
+ static void process__genericNoResampling(state_t* state, int64_t pts);
+ static void process__genericResampling(state_t* state, int64_t pts);
+ static void process__OneTrack16BitsStereoNoResampling(state_t* state,
+ int64_t pts);
+#if 0
+ static void process__TwoTracks16BitsStereoNoResampling(state_t* state,
+ int64_t pts);
+#endif
+
+ static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS,
+ int outputFrameIndex);
};
// ----------------------------------------------------------------------------
diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp
index 6be669b..c23eb04 100644
--- a/services/audioflinger/AudioPolicyService.cpp
+++ b/services/audioflinger/AudioPolicyService.cpp
@@ -30,8 +30,8 @@
#include <utils/String16.h>
#include <utils/threads.h>
#include "AudioPolicyService.h"
+#include "ServiceUtilities.h"
#include <cutils/properties.h>
-#include <dlfcn.h>
#include <hardware_legacy/power.h>
#include <media/AudioEffect.h>
#include <media/EffectsFactoryApi.h>
@@ -44,18 +44,11 @@
namespace android {
-static const char *kDeadlockedString = "AudioPolicyService may be deadlocked\n";
-static const char *kCmdDeadlockedString = "AudioPolicyService command thread may be deadlocked\n";
+static const char kDeadlockedString[] = "AudioPolicyService may be deadlocked\n";
+static const char kCmdDeadlockedString[] = "AudioPolicyService command thread may be deadlocked\n";
static const int kDumpLockRetries = 50;
-static const int kDumpLockSleep = 20000;
-
-static bool checkPermission() {
- if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
- bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"));
- if (!ok) ALOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
- return ok;
-}
+static const int kDumpLockSleepUs = 20000;
namespace {
extern struct audio_policy_service_ops aps_ops;
@@ -76,7 +69,7 @@
// start tone playback thread
mTonePlaybackThread = new AudioCommandThread(String8(""));
// start audio commands thread
- mAudioCommandThread = new AudioCommandThread(String8("ApmCommandThread"));
+ mAudioCommandThread = new AudioCommandThread(String8("ApmCommand"));
/* instantiate the audio policy manager */
rc = hw_get_module(AUDIO_POLICY_HARDWARE_MODULE_ID, &module);
@@ -123,19 +116,7 @@
// release audio pre processing resources
for (size_t i = 0; i < mInputSources.size(); i++) {
- InputSourceDesc *source = mInputSources.valueAt(i);
- Vector <EffectDesc *> effects = source->mEffects;
- for (size_t j = 0; j < effects.size(); j++) {
- delete effects[j]->mName;
- Vector <effect_param_t *> params = effects[j]->mParams;
- for (size_t k = 0; k < params.size(); k++) {
- delete params[k];
- }
- params.clear();
- delete effects[j];
- }
- effects.clear();
- delete source;
+ delete mInputSources.valueAt(i);
}
mInputSources.clear();
@@ -145,9 +126,9 @@
}
mInputs.clear();
- if (mpAudioPolicy && mpAudioPolicyDev)
+ if (mpAudioPolicy != NULL && mpAudioPolicyDev != NULL)
mpAudioPolicyDev->destroy_audio_policy(mpAudioPolicyDev, mpAudioPolicy);
- if (mpAudioPolicyDev)
+ if (mpAudioPolicyDev != NULL)
audio_policy_dev_close(mpAudioPolicyDev);
}
@@ -158,7 +139,7 @@
if (mpAudioPolicy == NULL) {
return NO_INIT;
}
- if (!checkPermission()) {
+ if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if (!audio_is_output_device(device) && !audio_is_input_device(device)) {
@@ -186,15 +167,15 @@
device_address);
}
-status_t AudioPolicyService::setPhoneState(int state)
+status_t AudioPolicyService::setPhoneState(audio_mode_t state)
{
if (mpAudioPolicy == NULL) {
return NO_INIT;
}
- if (!checkPermission()) {
+ if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
- if (state < 0 || state >= AUDIO_MODE_CNT) {
+ if (uint32_t(state) >= AUDIO_MODE_CNT) {
return BAD_VALUE;
}
@@ -208,26 +189,13 @@
return NO_ERROR;
}
-status_t AudioPolicyService::setRingerMode(uint32_t mode, uint32_t mask)
-{
- if (mpAudioPolicy == NULL) {
- return NO_INIT;
- }
- if (!checkPermission()) {
- return PERMISSION_DENIED;
- }
-
- mpAudioPolicy->set_ringer_mode(mpAudioPolicy, mode, mask);
- return NO_ERROR;
-}
-
status_t AudioPolicyService::setForceUse(audio_policy_force_use_t usage,
audio_policy_forced_cfg_t config)
{
if (mpAudioPolicy == NULL) {
return NO_INIT;
}
- if (!checkPermission()) {
+ if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if (usage < 0 || usage >= AUDIO_POLICY_FORCE_USE_CNT) {
@@ -255,7 +223,7 @@
audio_io_handle_t AudioPolicyService::getOutput(audio_stream_type_t stream,
uint32_t samplingRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channels,
audio_policy_output_flags_t flags)
{
@@ -301,9 +269,9 @@
mpAudioPolicy->release_output(mpAudioPolicy, output);
}
-audio_io_handle_t AudioPolicyService::getInput(int inputSource,
+audio_io_handle_t AudioPolicyService::getInput(audio_source_t inputSource,
uint32_t samplingRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channels,
audio_in_acoustics_t acoustics,
int audioSession)
@@ -311,6 +279,10 @@
if (mpAudioPolicy == NULL) {
return 0;
}
+ // already checked by client, but double-check in case the client wrapper is bypassed
+ if (uint32_t(inputSource) >= AUDIO_SOURCE_CNT) {
+ return 0;
+ }
Mutex::Autolock _l(mLock);
audio_io_handle_t input = mpAudioPolicy->get_input(mpAudioPolicy, inputSource, samplingRate,
format, channels, acoustics);
@@ -319,15 +291,14 @@
return input;
}
// create audio pre processors according to input source
- ssize_t index = mInputSources.indexOfKey((audio_source_t)inputSource);
+ ssize_t index = mInputSources.indexOfKey(inputSource);
if (index < 0) {
return input;
}
ssize_t idx = mInputs.indexOfKey(input);
InputDesc *inputDesc;
if (idx < 0) {
- inputDesc = new InputDesc();
- inputDesc->mSessionId = audioSession;
+ inputDesc = new InputDesc(audioSession);
mInputs.add(input, inputDesc);
} else {
inputDesc = mInputs.valueAt(idx);
@@ -386,7 +357,6 @@
}
InputDesc *inputDesc = mInputs.valueAt(index);
setPreProcessorEnabled(inputDesc, false);
- inputDesc->mEffects.clear();
delete inputDesc;
mInputs.removeItemsAt(index);
}
@@ -398,40 +368,58 @@
if (mpAudioPolicy == NULL) {
return NO_INIT;
}
- if (!checkPermission()) {
+ if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
- if (stream < 0 || stream >= AUDIO_STREAM_CNT) {
+ if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
return BAD_VALUE;
}
mpAudioPolicy->init_stream_volume(mpAudioPolicy, stream, indexMin, indexMax);
return NO_ERROR;
}
-status_t AudioPolicyService::setStreamVolumeIndex(audio_stream_type_t stream, int index)
+status_t AudioPolicyService::setStreamVolumeIndex(audio_stream_type_t stream,
+ int index,
+ audio_devices_t device)
{
if (mpAudioPolicy == NULL) {
return NO_INIT;
}
- if (!checkPermission()) {
+ if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
- if (stream < 0 || stream >= AUDIO_STREAM_CNT) {
+ if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
return BAD_VALUE;
}
- return mpAudioPolicy->set_stream_volume_index(mpAudioPolicy, stream, index);
+ if (mpAudioPolicy->set_stream_volume_index_for_device) {
+ return mpAudioPolicy->set_stream_volume_index_for_device(mpAudioPolicy,
+ stream,
+ index,
+ device);
+ } else {
+ return mpAudioPolicy->set_stream_volume_index(mpAudioPolicy, stream, index);
+ }
}
-status_t AudioPolicyService::getStreamVolumeIndex(audio_stream_type_t stream, int *index)
+status_t AudioPolicyService::getStreamVolumeIndex(audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device)
{
if (mpAudioPolicy == NULL) {
return NO_INIT;
}
- if (stream < 0 || stream >= AUDIO_STREAM_CNT) {
+ if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
return BAD_VALUE;
}
- return mpAudioPolicy->get_stream_volume_index(mpAudioPolicy, stream, index);
+ if (mpAudioPolicy->get_stream_volume_index_for_device) {
+ return mpAudioPolicy->get_stream_volume_index_for_device(mpAudioPolicy,
+ stream,
+ index,
+ device);
+ } else {
+ return mpAudioPolicy->get_stream_volume_index(mpAudioPolicy, stream, index);
+ }
}
uint32_t AudioPolicyService::getStrategyForStream(audio_stream_type_t stream)
@@ -442,10 +430,12 @@
return mpAudioPolicy->get_strategy_for_stream(mpAudioPolicy, stream);
}
-uint32_t AudioPolicyService::getDevicesForStream(audio_stream_type_t stream)
+//audio policy: use audio_device_t appropriately
+
+audio_devices_t AudioPolicyService::getDevicesForStream(audio_stream_type_t stream)
{
if (mpAudioPolicy == NULL) {
- return 0;
+ return (audio_devices_t)0;
}
return mpAudioPolicy->get_devices_for_stream(mpAudioPolicy, stream);
}
@@ -487,7 +477,7 @@
return mpAudioPolicy->set_effect_enabled(mpAudioPolicy, id, enabled);
}
-bool AudioPolicyService::isStreamActive(int stream, uint32_t inPastMs) const
+bool AudioPolicyService::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const
{
if (mpAudioPolicy == NULL) {
return 0;
@@ -534,7 +524,7 @@
}
void AudioPolicyService::binderDied(const wp<IBinder>& who) {
- ALOGW("binderDied() %p, tid %d, calling tid %d", who.unsafe_get(), gettid(),
+ ALOGW("binderDied() %p, tid %d, calling pid %d", who.unsafe_get(), gettid(),
IPCThreadState::self()->getCallingPid());
}
@@ -546,7 +536,7 @@
locked = true;
break;
}
- usleep(kDumpLockSleep);
+ usleep(kDumpLockSleepUs);
}
return locked;
}
@@ -570,7 +560,7 @@
status_t AudioPolicyService::dump(int fd, const Vector<String16>& args)
{
- if (checkCallingPermission(String16("android.permission.DUMP")) == false) {
+ if (!dumpAllowed()) {
dumpPermissionDenial(fd);
} else {
bool locked = tryLock(mLock);
@@ -580,10 +570,10 @@
}
dumpInternals(fd);
- if (mAudioCommandThread != NULL) {
+ if (mAudioCommandThread != 0) {
mAudioCommandThread->dump(fd);
}
- if (mTonePlaybackThread != NULL) {
+ if (mTonePlaybackThread != 0) {
mTonePlaybackThread->dump(fd);
}
@@ -610,12 +600,11 @@
return NO_ERROR;
}
-void AudioPolicyService::setPreProcessorEnabled(InputDesc *inputDesc, bool enabled)
+void AudioPolicyService::setPreProcessorEnabled(const InputDesc *inputDesc, bool enabled)
{
- Vector<sp<AudioEffect> > fxVector = inputDesc->mEffects;
+ const Vector<sp<AudioEffect> > &fxVector = inputDesc->mEffects;
for (size_t i = 0; i < fxVector.size(); i++) {
- sp<AudioEffect> fx = fxVector.itemAt(i);
- fx->setEnabled(enabled);
+ fxVector.itemAt(i)->setEnabled(enabled);
}
}
@@ -641,7 +630,7 @@
release_wake_lock(mName.string());
}
mAudioCommands.clear();
- if (mpToneGenerator != NULL) delete mpToneGenerator;
+ delete mpToneGenerator;
}
void AudioPolicyService::AudioCommandThread::onFirstRef()
@@ -649,7 +638,7 @@
if (mName != "") {
run(mName.string(), ANDROID_PRIORITY_AUDIO);
} else {
- run("AudioCommandThread", ANDROID_PRIORITY_AUDIO);
+ run("AudioCommand", ANDROID_PRIORITY_AUDIO);
}
}
@@ -660,7 +649,7 @@
mLock.lock();
while (!exitPending())
{
- while(!mAudioCommands.isEmpty()) {
+ while (!mAudioCommands.isEmpty()) {
nsecs_t curTime = systemTime();
// commands are sorted by increasing time stamp: execute them from index 0 and up
if (mAudioCommands[0]->mTime <= curTime) {
@@ -674,8 +663,7 @@
ToneData *data = (ToneData *)command->mParam;
ALOGV("AudioCommandThread() processing start tone %d on stream %d",
data->mType, data->mStream);
- if (mpToneGenerator != NULL)
- delete mpToneGenerator;
+ delete mpToneGenerator;
mpToneGenerator = new ToneGenerator(data->mStream, 1.0);
mpToneGenerator->startTone(data->mType);
delete data;
@@ -705,16 +693,16 @@
delete data;
}break;
case SET_PARAMETERS: {
- ParametersData *data = (ParametersData *)command->mParam;
- ALOGV("AudioCommandThread() processing set parameters string %s, io %d",
- data->mKeyValuePairs.string(), data->mIO);
- command->mStatus = AudioSystem::setParameters(data->mIO, data->mKeyValuePairs);
- if (command->mWaitStatus) {
- command->mCond.signal();
- mWaitWorkCV.wait(mLock);
- }
- delete data;
- }break;
+ ParametersData *data = (ParametersData *)command->mParam;
+ ALOGV("AudioCommandThread() processing set parameters string %s, io %d",
+ data->mKeyValuePairs.string(), data->mIO);
+ command->mStatus = AudioSystem::setParameters(data->mIO, data->mKeyValuePairs);
+ if (command->mWaitStatus) {
+ command->mCond.signal();
+ mWaitWorkCV.wait(mLock);
+ }
+ delete data;
+ }break;
case SET_VOICE_VOLUME: {
VoiceVolumeData *data = (VoiceVolumeData *)command->mParam;
ALOGV("AudioCommandThread() processing set voice volume volume %f",
@@ -767,7 +755,7 @@
snprintf(buffer, SIZE, "- Commands:\n");
result = String8(buffer);
result.append(" Command Time Wait pParam\n");
- for (int i = 0; i < (int)mAudioCommands.size(); i++) {
+ for (size_t i = 0; i < mAudioCommands.size(); i++) {
mAudioCommands[i]->dump(buffer, SIZE);
result.append(buffer);
}
@@ -782,7 +770,8 @@
return NO_ERROR;
}
-void AudioPolicyService::AudioCommandThread::startToneCommand(int type, int stream)
+void AudioPolicyService::AudioCommandThread::startToneCommand(ToneGenerator::tone_type type,
+ audio_stream_type_t stream)
{
AudioCommand *command = new AudioCommand();
command->mCommand = START_TONE;
@@ -809,9 +798,9 @@
mWaitWorkCV.signal();
}
-status_t AudioPolicyService::AudioCommandThread::volumeCommand(int stream,
+status_t AudioPolicyService::AudioCommandThread::volumeCommand(audio_stream_type_t stream,
float volume,
- int output,
+ audio_io_handle_t output,
int delayMs)
{
status_t status = NO_ERROR;
@@ -841,7 +830,7 @@
return status;
}
-status_t AudioPolicyService::AudioCommandThread::parametersCommand(int ioHandle,
+status_t AudioPolicyService::AudioCommandThread::parametersCommand(audio_io_handle_t ioHandle,
const char *keyValuePairs,
int delayMs)
{
@@ -900,7 +889,7 @@
// insertCommand_l() must be called with mLock held
void AudioPolicyService::AudioCommandThread::insertCommand_l(AudioCommand *command, int delayMs)
{
- ssize_t i;
+ ssize_t i; // not size_t because i will count down to -1
Vector <AudioCommand *> removedCommands;
command->mTime = systemTime() + milliseconds(delayMs);
@@ -927,19 +916,19 @@
AudioParameter param = AudioParameter(data->mKeyValuePairs);
AudioParameter param2 = AudioParameter(data2->mKeyValuePairs);
for (size_t j = 0; j < param.size(); j++) {
- String8 key;
- String8 value;
- param.getAt(j, key, value);
- for (size_t k = 0; k < param2.size(); k++) {
- String8 key2;
- String8 value2;
- param2.getAt(k, key2, value2);
- if (key2 == key) {
- param2.remove(key2);
- ALOGV("Filtering out parameter %s", key2.string());
- break;
- }
- }
+ String8 key;
+ String8 value;
+ param.getAt(j, key, value);
+ for (size_t k = 0; k < param2.size(); k++) {
+ String8 key2;
+ String8 value2;
+ param2.getAt(k, key2, value2);
+ if (key2 == key) {
+ param2.remove(key2);
+ ALOGV("Filtering out parameter %s", key2.string());
+ break;
+ }
+ }
}
// if all keys have been filtered out, remove the command.
// otherwise, update the key value pairs
@@ -1011,7 +1000,7 @@
const char *keyValuePairs,
int delayMs)
{
- mAudioCommandThread->parametersCommand((int)ioHandle, keyValuePairs,
+ mAudioCommandThread->parametersCommand(ioHandle, keyValuePairs,
delayMs);
}
@@ -1020,8 +1009,8 @@
audio_io_handle_t output,
int delayMs)
{
- return (int)mAudioCommandThread->volumeCommand((int)stream, volume,
- (int)output, delayMs);
+ return (int)mAudioCommandThread->volumeCommand(stream, volume,
+ output, delayMs);
}
int AudioPolicyService::startTone(audio_policy_tone_t tone,
@@ -1031,7 +1020,7 @@
ALOGE("startTone: illegal tone requested (%d)", tone);
if (stream != AUDIO_STREAM_VOICE_CALL)
ALOGE("startTone: illegal stream (%d) requested for tone %d", stream,
- tone);
+ tone);
mTonePlaybackThread->startToneCommand(ToneGenerator::TONE_SUP_CALL_WAITING,
AUDIO_STREAM_VOICE_CALL);
return 0;
@@ -1052,7 +1041,7 @@
// Audio pre-processing configuration
// ----------------------------------------------------------------------------
-const char *AudioPolicyService::kInputSourceNames[AUDIO_SOURCE_CNT -1] = {
+/*static*/ const char * const AudioPolicyService::kInputSourceNames[AUDIO_SOURCE_CNT -1] = {
MIC_SRC_TAG,
VOICE_UL_SRC_TAG,
VOICE_DL_SRC_TAG,
@@ -1152,7 +1141,7 @@
if (param == NULL && value == NULL) {
// try to parse simple parameter form {int int}
param = root->first_child;
- if (param) {
+ if (param != NULL) {
// Note: that a pair of random strings is read as 0 0
int *ptr = (int *)fx_param->data;
int *ptr2 = (int *)((char *)param + sizeof(effect_param_t));
@@ -1241,7 +1230,7 @@
node = node->next;
continue;
}
- EffectDesc *effect = new EffectDesc(*effects[i]);
+ EffectDesc *effect = new EffectDesc(*effects[i]); // deep copy
loadEffectParameters(node, effect->mParams);
ALOGV("loadInputSource() adding effect %s uuid %08x", effect->mName, effect->mUuid.timeLow);
source->mEffects.add(effect);
@@ -1292,11 +1281,7 @@
ALOGW("loadEffect() invalid uuid %s", node->value);
return NULL;
}
- EffectDesc *effect = new EffectDesc();
- effect->mName = strdup(root->name);
- memcpy(&effect->mUuid, &uuid, sizeof(effect_uuid_t));
-
- return effect;
+ return new EffectDesc(root->name, uuid);
}
status_t AudioPolicyService::loadEffects(cnode *root, Vector <EffectDesc *>& effects)
@@ -1348,13 +1333,13 @@
static audio_io_handle_t aps_open_output(void *service,
uint32_t *pDevices,
uint32_t *pSamplingRate,
- uint32_t *pFormat,
+ audio_format_t *pFormat,
uint32_t *pChannels,
uint32_t *pLatencyMs,
audio_policy_output_flags_t flags)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == NULL) {
+ if (af == 0) {
ALOGW("%s: could not get AudioFlinger", __func__);
return 0;
}
@@ -1368,7 +1353,7 @@
audio_io_handle_t output2)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == NULL) {
+ if (af == 0) {
ALOGW("%s: could not get AudioFlinger", __func__);
return 0;
}
@@ -1378,7 +1363,7 @@
static int aps_close_output(void *service, audio_io_handle_t output)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == NULL)
+ if (af == 0)
return PERMISSION_DENIED;
return af->closeOutput(output);
@@ -1387,7 +1372,7 @@
static int aps_suspend_output(void *service, audio_io_handle_t output)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == NULL) {
+ if (af == 0) {
ALOGW("%s: could not get AudioFlinger", __func__);
return PERMISSION_DENIED;
}
@@ -1398,7 +1383,7 @@
static int aps_restore_output(void *service, audio_io_handle_t output)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == NULL) {
+ if (af == 0) {
ALOGW("%s: could not get AudioFlinger", __func__);
return PERMISSION_DENIED;
}
@@ -1409,12 +1394,12 @@
static audio_io_handle_t aps_open_input(void *service,
uint32_t *pDevices,
uint32_t *pSamplingRate,
- uint32_t *pFormat,
+ audio_format_t *pFormat,
uint32_t *pChannels,
- uint32_t acoustics)
+ audio_in_acoustics_t acoustics)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == NULL) {
+ if (af == 0) {
ALOGW("%s: could not get AudioFlinger", __func__);
return 0;
}
@@ -1426,7 +1411,7 @@
static int aps_close_input(void *service, audio_io_handle_t input)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == NULL)
+ if (af == 0)
return PERMISSION_DENIED;
return af->closeInput(input);
@@ -1436,7 +1421,7 @@
audio_io_handle_t output)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == NULL)
+ if (af == 0)
return PERMISSION_DENIED;
return af->setStreamOutput(stream, output);
@@ -1447,10 +1432,10 @@
audio_io_handle_t dst_output)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == NULL)
+ if (af == 0)
return PERMISSION_DENIED;
- return af->moveEffects(session, (int)src_output, (int)dst_output);
+ return af->moveEffects(session, src_output, dst_output);
}
static char * aps_get_parameters(void *service, audio_io_handle_t io_handle,
diff --git a/services/audioflinger/AudioPolicyService.h b/services/audioflinger/AudioPolicyService.h
index d898a53..9ed905d 100644
--- a/services/audioflinger/AudioPolicyService.h
+++ b/services/audioflinger/AudioPolicyService.h
@@ -19,6 +19,7 @@
#include <cutils/misc.h>
#include <cutils/config_utils.h>
+#include <utils/String8.h>
#include <utils/Vector.h>
#include <utils/SortedVector.h>
#include <binder/BinderService.h>
@@ -31,8 +32,6 @@
namespace android {
-class String8;
-
// ----------------------------------------------------------------------------
class AudioPolicyService :
@@ -59,16 +58,15 @@
virtual audio_policy_dev_state_t getDeviceConnectionState(
audio_devices_t device,
const char *device_address);
- virtual status_t setPhoneState(int state);
- virtual status_t setRingerMode(uint32_t mode, uint32_t mask);
+ virtual status_t setPhoneState(audio_mode_t state);
virtual status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config);
virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
virtual audio_io_handle_t getOutput(audio_stream_type_t stream,
uint32_t samplingRate = 0,
- uint32_t format = AUDIO_FORMAT_DEFAULT,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
uint32_t channels = 0,
audio_policy_output_flags_t flags =
- AUDIO_POLICY_OUTPUT_FLAG_INDIRECT);
+ AUDIO_POLICY_OUTPUT_FLAG_NONE);
virtual status_t startOutput(audio_io_handle_t output,
audio_stream_type_t stream,
int session = 0);
@@ -76,12 +74,12 @@
audio_stream_type_t stream,
int session = 0);
virtual void releaseOutput(audio_io_handle_t output);
- virtual audio_io_handle_t getInput(int inputSource,
+ virtual audio_io_handle_t getInput(audio_source_t inputSource,
uint32_t samplingRate = 0,
- uint32_t format = AUDIO_FORMAT_DEFAULT,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
uint32_t channels = 0,
audio_in_acoustics_t acoustics =
- (audio_in_acoustics_t)0,
+ (audio_in_acoustics_t)0 /*AUDIO_IN_ACOUSTICS_NONE*/,
int audioSession = 0);
virtual status_t startInput(audio_io_handle_t input);
virtual status_t stopInput(audio_io_handle_t input);
@@ -89,11 +87,15 @@
virtual status_t initStreamVolume(audio_stream_type_t stream,
int indexMin,
int indexMax);
- virtual status_t setStreamVolumeIndex(audio_stream_type_t stream, int index);
- virtual status_t getStreamVolumeIndex(audio_stream_type_t stream, int *index);
+ virtual status_t setStreamVolumeIndex(audio_stream_type_t stream,
+ int index,
+ audio_devices_t device);
+ virtual status_t getStreamVolumeIndex(audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device);
virtual uint32_t getStrategyForStream(audio_stream_type_t stream);
- virtual uint32_t getDevicesForStream(audio_stream_type_t stream);
+ virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream);
virtual audio_io_handle_t getOutputForEffect(effect_descriptor_t *desc);
virtual status_t registerEffect(effect_descriptor_t *desc,
@@ -103,7 +105,7 @@
int id);
virtual status_t unregisterEffect(int id);
virtual status_t setEffectEnabled(int id, bool enabled);
- virtual bool isStreamActive(int stream, uint32_t inPastMs = 0) const;
+ virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const;
virtual status_t queryDefaultPreProcessing(int audioSession,
effect_descriptor_t *descriptors,
@@ -169,10 +171,13 @@
virtual bool threadLoop();
void exit();
- void startToneCommand(int type = 0, int stream = 0);
+ void startToneCommand(ToneGenerator::tone_type type,
+ audio_stream_type_t stream);
void stopToneCommand();
- status_t volumeCommand(int stream, float volume, int output, int delayMs = 0);
- status_t parametersCommand(int ioHandle, const char *keyValuePairs, int delayMs = 0);
+ status_t volumeCommand(audio_stream_type_t stream, float volume,
+ audio_io_handle_t output, int delayMs = 0);
+ status_t parametersCommand(audio_io_handle_t ioHandle,
+ const char *keyValuePairs, int delayMs = 0);
status_t voiceVolumeCommand(float volume, int delayMs = 0);
void insertCommand_l(AudioCommand *command, int delayMs = 0);
@@ -196,20 +201,20 @@
class ToneData {
public:
- int mType; // tone type (START_TONE only)
- int mStream; // stream type (START_TONE only)
+ ToneGenerator::tone_type mType; // tone type (START_TONE only)
+ audio_stream_type_t mStream; // stream type (START_TONE only)
};
class VolumeData {
public:
- int mStream;
+ audio_stream_type_t mStream;
float mVolume;
- int mIO;
+ audio_io_handle_t mIO;
};
class ParametersData {
public:
- int mIO;
+ audio_io_handle_t mIO;
String8 mKeyValuePairs;
};
@@ -228,8 +233,33 @@
class EffectDesc {
public:
- EffectDesc() {}
- virtual ~EffectDesc() {}
+ EffectDesc(const char *name, const effect_uuid_t& uuid) :
+ mName(strdup(name)),
+ mUuid(uuid) { }
+ EffectDesc(const EffectDesc& orig) :
+ mName(strdup(orig.mName)),
+ mUuid(orig.mUuid) {
+ // deep copy mParams
+ for (size_t k = 0; k < orig.mParams.size(); k++) {
+ effect_param_t *origParam = orig.mParams[k];
+ // psize and vsize are rounded up to an int boundary for allocation
+ size_t origSize = sizeof(effect_param_t) +
+ ((origParam->psize + 3) & ~3) +
+ ((origParam->vsize + 3) & ~3);
+ effect_param_t *dupParam = (effect_param_t *) malloc(origSize);
+ memcpy(dupParam, origParam, origSize);
+ // This works because the param buffer allocation is also done by
+ // multiples of 4 bytes originally. In theory we should memcpy only
+ // the actual param size, that is without rounding vsize.
+ mParams.add(dupParam);
+ }
+ }
+ /*virtual*/ ~EffectDesc() {
+ free(mName);
+ for (size_t k = 0; k < mParams.size(); k++) {
+ free(mParams[k]);
+ }
+ }
char *mName;
effect_uuid_t mUuid;
Vector <effect_param_t *> mParams;
@@ -238,22 +268,26 @@
class InputSourceDesc {
public:
InputSourceDesc() {}
- virtual ~InputSourceDesc() {}
+ /*virtual*/ ~InputSourceDesc() {
+ for (size_t j = 0; j < mEffects.size(); j++) {
+ delete mEffects[j];
+ }
+ }
Vector <EffectDesc *> mEffects;
};
class InputDesc {
public:
- InputDesc() {}
- virtual ~InputDesc() {}
- int mSessionId;
+ InputDesc(int session) : mSessionId(session) {}
+ /*virtual*/ ~InputDesc() {}
+ const int mSessionId;
Vector< sp<AudioEffect> >mEffects;
};
- static const char *kInputSourceNames[AUDIO_SOURCE_CNT -1];
+ static const char * const kInputSourceNames[AUDIO_SOURCE_CNT -1];
- void setPreProcessorEnabled(InputDesc *inputDesc, bool enabled);
+ void setPreProcessorEnabled(const InputDesc *inputDesc, bool enabled);
status_t loadPreProcessorConfig(const char *path);
status_t loadEffects(cnode *root, Vector <EffectDesc *>& effects);
EffectDesc *loadEffect(cnode *root);
@@ -277,8 +311,8 @@
mutable Mutex mLock; // prevents concurrent access to AudioPolicy manager functions changing
// device connection state or routing
- sp <AudioCommandThread> mAudioCommandThread; // audio commands thread
- sp <AudioCommandThread> mTonePlaybackThread; // tone playback thread
+ sp<AudioCommandThread> mAudioCommandThread; // audio commands thread
+ sp<AudioCommandThread> mTonePlaybackThread; // tone playback thread
struct audio_policy_device *mpAudioPolicyDev;
struct audio_policy *mpAudioPolicy;
KeyedVector< audio_source_t, InputSourceDesc* > mInputSources;
diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index 4586b54..fbb54cf 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -23,8 +23,10 @@
#include <cutils/log.h>
#include <cutils/properties.h>
#include "AudioResampler.h"
+#if 0
#include "AudioResamplerSinc.h"
#include "AudioResamplerCubic.h"
+#endif
#ifdef __arm__
#include <machine/cpu-features.h>
@@ -99,6 +101,7 @@
ALOGV("Create linear Resampler");
resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate);
break;
+#if 0
case MED_QUALITY:
ALOGV("Create cubic Resampler");
resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate);
@@ -107,6 +110,7 @@
ALOGV("Create sinc Resampler");
resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate);
break;
+#endif
}
// initialize resampler
@@ -118,7 +122,8 @@
int32_t sampleRate) :
mBitDepth(bitDepth), mChannelCount(inChannelCount),
mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputIndex(0),
- mPhaseFraction(0) {
+ mPhaseFraction(0), mLocalTimeFreq(0),
+ mPTS(AudioBufferProvider::kInvalidPTS) {
// sanity check on format
if ((bitDepth != 16) ||(inChannelCount < 1) || (inChannelCount > 2)) {
ALOGE("Unsupported sample format, %d bits, %d channels", bitDepth,
@@ -130,12 +135,6 @@
mVolume[0] = mVolume[1] = 0;
mBuffer.frameCount = 0;
- // save format for quick lookup
- if (inChannelCount == 1) {
- mFormat = MONO_16_BIT;
- } else {
- mFormat = STEREO_16_BIT;
- }
}
AudioResampler::~AudioResampler() {
@@ -152,6 +151,23 @@
mVolume[1] = right;
}
+void AudioResampler::setLocalTimeFreq(uint64_t freq) {
+ mLocalTimeFreq = freq;
+}
+
+void AudioResampler::setPTS(int64_t pts) {
+ mPTS = pts;
+}
+
+int64_t AudioResampler::calculateOutputPTS(int outputFrameIndex) {
+
+ if (mPTS == AudioBufferProvider::kInvalidPTS) {
+ return AudioBufferProvider::kInvalidPTS;
+ } else {
+ return mPTS + ((outputFrameIndex * mLocalTimeFreq) / mSampleRate);
+ }
+}
+
void AudioResampler::reset() {
mInputIndex = 0;
mPhaseFraction = 0;
@@ -190,7 +206,7 @@
size_t outputSampleCount = outFrameCount * 2;
size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
- // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d\n",
+ // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",
// outFrameCount, inputIndex, phaseFraction, phaseIncrement);
while (outputIndex < outputSampleCount) {
@@ -198,26 +214,27 @@
// buffer is empty, fetch a new one
while (mBuffer.frameCount == 0) {
mBuffer.frameCount = inFrameCount;
- provider->getNextBuffer(&mBuffer);
+ provider->getNextBuffer(&mBuffer,
+ calculateOutputPTS(outputIndex / 2));
if (mBuffer.raw == NULL) {
goto resampleStereo16_exit;
}
- // ALOGE("New buffer fetched: %d frames\n", mBuffer.frameCount);
+ // ALOGE("New buffer fetched: %d frames", mBuffer.frameCount);
if (mBuffer.frameCount > inputIndex) break;
inputIndex -= mBuffer.frameCount;
mX0L = mBuffer.i16[mBuffer.frameCount*2-2];
mX0R = mBuffer.i16[mBuffer.frameCount*2-1];
provider->releaseBuffer(&mBuffer);
- // mBuffer.frameCount == 0 now so we reload a new buffer
+ // mBuffer.frameCount == 0 now so we reload a new buffer
}
int16_t *in = mBuffer.i16;
// handle boundary case
while (inputIndex == 0) {
- // ALOGE("boundary case\n");
+ // ALOGE("boundary case");
out[outputIndex++] += vl * Interp(mX0L, in[0], phaseFraction);
out[outputIndex++] += vr * Interp(mX0R, in[1], phaseFraction);
Advance(&inputIndex, &phaseFraction, phaseIncrement);
@@ -226,7 +243,7 @@
}
// process input samples
- // ALOGE("general case\n");
+ // ALOGE("general case");
#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
if (inputIndex + 2 < mBuffer.frameCount) {
@@ -248,7 +265,7 @@
Advance(&inputIndex, &phaseFraction, phaseIncrement);
}
- // ALOGE("loop done - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
+ // ALOGE("loop done - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
// if done with buffer, save samples
if (inputIndex >= mBuffer.frameCount) {
@@ -265,7 +282,7 @@
}
}
- // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
+ // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
resampleStereo16_exit:
// save state
@@ -286,19 +303,20 @@
size_t outputSampleCount = outFrameCount * 2;
size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
- // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d\n",
+ // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",
// outFrameCount, inputIndex, phaseFraction, phaseIncrement);
while (outputIndex < outputSampleCount) {
// buffer is empty, fetch a new one
while (mBuffer.frameCount == 0) {
mBuffer.frameCount = inFrameCount;
- provider->getNextBuffer(&mBuffer);
+ provider->getNextBuffer(&mBuffer,
+ calculateOutputPTS(outputIndex / 2));
if (mBuffer.raw == NULL) {
mInputIndex = inputIndex;
mPhaseFraction = phaseFraction;
goto resampleMono16_exit;
}
- // ALOGE("New buffer fetched: %d frames\n", mBuffer.frameCount);
+ // ALOGE("New buffer fetched: %d frames", mBuffer.frameCount);
if (mBuffer.frameCount > inputIndex) break;
inputIndex -= mBuffer.frameCount;
@@ -310,7 +328,7 @@
// handle boundary case
while (inputIndex == 0) {
- // ALOGE("boundary case\n");
+ // ALOGE("boundary case");
int32_t sample = Interp(mX0L, in[0], phaseFraction);
out[outputIndex++] += vl * sample;
out[outputIndex++] += vr * sample;
@@ -320,7 +338,7 @@
}
// process input samples
- // ALOGE("general case\n");
+ // ALOGE("general case");
#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
if (inputIndex + 2 < mBuffer.frameCount) {
@@ -343,7 +361,7 @@
}
- // ALOGE("loop done - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
+ // ALOGE("loop done - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
// if done with buffer, save samples
if (inputIndex >= mBuffer.frameCount) {
@@ -359,7 +377,7 @@
}
}
- // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
+ // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
resampleMono16_exit:
// save state
@@ -390,6 +408,7 @@
* phaseFraction : phase fraction for next interpolation
*
*******************************************************************/
+__attribute__((noinline))
void AudioResamplerOrder1::AsmMono16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
uint32_t &phaseFraction, uint32_t phaseIncrement)
@@ -500,6 +519,7 @@
* phaseFraction : phase fraction for next interpolation
*
*******************************************************************/
+__attribute__((noinline))
void AudioResamplerOrder1::AsmStereo16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
uint32_t &phaseFraction, uint32_t phaseIncrement)
@@ -600,6 +620,5 @@
// ----------------------------------------------------------------------------
-}
-; // namespace android
+} // namespace android
diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h
index ffa690a..1610e00 100644
--- a/services/audioflinger/AudioResampler.h
+++ b/services/audioflinger/AudioResampler.h
@@ -33,7 +33,7 @@
// HIGH_QUALITY: fixed multi-tap FIR (e.g. 48KHz->44.1KHz)
// NOTE: high quality SRC will only be supported for
// certain fixed rate conversions. Sample rate cannot be
- // changed dynamically.
+ // changed dynamically.
enum src_quality {
DEFAULT=0,
LOW_QUALITY=1,
@@ -49,12 +49,16 @@
virtual void init() = 0;
virtual void setSampleRate(int32_t inSampleRate);
virtual void setVolume(int16_t left, int16_t right);
+ virtual void setLocalTimeFreq(uint64_t freq);
+
+ // set the PTS of the next buffer output by the resampler
+ virtual void setPTS(int64_t pts);
virtual void resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider) = 0;
virtual void reset();
- virtual size_t getUnreleasedFrames() { return mInputIndex; }
+ virtual size_t getUnreleasedFrames() const { return mInputIndex; }
protected:
// number of bits for phase fraction - 30 bits allows nearly 2x downsampling
@@ -66,16 +70,17 @@
// multiplier to calculate fixed point phase increment
static const double kPhaseMultiplier = 1L << kNumPhaseBits;
- enum format {MONO_16_BIT, STEREO_16_BIT};
AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate);
// prevent copying
AudioResampler(const AudioResampler&);
AudioResampler& operator=(const AudioResampler&);
- int32_t mBitDepth;
- int32_t mChannelCount;
- int32_t mSampleRate;
+ int64_t calculateOutputPTS(int outputFrameIndex);
+
+ const int32_t mBitDepth;
+ const int32_t mChannelCount;
+ const int32_t mSampleRate;
int32_t mInSampleRate;
AudioBufferProvider::Buffer mBuffer;
union {
@@ -83,10 +88,11 @@
uint32_t mVolumeRL;
};
int16_t mTargetVolume[2];
- format mFormat;
size_t mInputIndex;
int32_t mPhaseIncrement;
uint32_t mPhaseFraction;
+ uint64_t mLocalTimeFreq;
+ int64_t mPTS;
};
// ----------------------------------------------------------------------------
diff --git a/services/audioflinger/AudioResamplerCubic.cpp b/services/audioflinger/AudioResamplerCubic.cpp
index 47205ba..18e59e9 100644
--- a/services/audioflinger/AudioResamplerCubic.cpp
+++ b/services/audioflinger/AudioResamplerCubic.cpp
@@ -65,7 +65,7 @@
// fetch first buffer
if (mBuffer.frameCount == 0) {
mBuffer.frameCount = inFrameCount;
- provider->getNextBuffer(&mBuffer);
+ provider->getNextBuffer(&mBuffer, mPTS);
if (mBuffer.raw == NULL)
return;
// ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
@@ -95,11 +95,12 @@
inputIndex = 0;
provider->releaseBuffer(&mBuffer);
mBuffer.frameCount = inFrameCount;
- provider->getNextBuffer(&mBuffer);
+ provider->getNextBuffer(&mBuffer,
+ calculateOutputPTS(outputIndex / 2));
if (mBuffer.raw == NULL)
goto save_state; // ugly, but efficient
in = mBuffer.i16;
- // ALOGW("New buffer: offset=%p, frames=%d\n", mBuffer.raw, mBuffer.frameCount);
+ // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
}
// advance sample state
@@ -130,10 +131,10 @@
// fetch first buffer
if (mBuffer.frameCount == 0) {
mBuffer.frameCount = inFrameCount;
- provider->getNextBuffer(&mBuffer);
+ provider->getNextBuffer(&mBuffer, mPTS);
if (mBuffer.raw == NULL)
return;
- // ALOGW("New buffer: offset=%p, frames=%d\n", mBuffer.raw, mBuffer.frameCount);
+ // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
}
int16_t *in = mBuffer.i16;
@@ -160,7 +161,8 @@
inputIndex = 0;
provider->releaseBuffer(&mBuffer);
mBuffer.frameCount = inFrameCount;
- provider->getNextBuffer(&mBuffer);
+ provider->getNextBuffer(&mBuffer,
+ calculateOutputPTS(outputIndex / 2));
if (mBuffer.raw == NULL)
goto save_state; // ugly, but efficient
// ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
@@ -181,4 +183,3 @@
// ----------------------------------------------------------------------------
}
; // namespace android
-
diff --git a/services/audioflinger/AudioResamplerCubic.h b/services/audioflinger/AudioResamplerCubic.h
index b72b62a..892785a 100644
--- a/services/audioflinger/AudioResamplerCubic.h
+++ b/services/audioflinger/AudioResamplerCubic.h
@@ -55,7 +55,7 @@
p->y1 = p->y2;
p->y2 = p->y3;
p->y3 = in;
- p->a = (3 * (p->y1 - p->y2) - p->y0 + p->y3) >> 1;
+ p->a = (3 * (p->y1 - p->y2) - p->y0 + p->y3) >> 1;
p->b = (p->y2 << 1) + p->y0 - (((5 * p->y1 + p->y3)) >> 1);
p->c = (p->y2 - p->y0) >> 1;
}
diff --git a/services/audioflinger/AudioResamplerSinc.cpp b/services/audioflinger/AudioResamplerSinc.cpp
index 9e5e254..76662d8 100644
--- a/services/audioflinger/AudioResamplerSinc.cpp
+++ b/services/audioflinger/AudioResamplerSinc.cpp
@@ -199,33 +199,33 @@
size_t outputSampleCount = outFrameCount * 2;
size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
- AudioBufferProvider::Buffer& buffer(mBuffer);
while (outputIndex < outputSampleCount) {
// buffer is empty, fetch a new one
- while (buffer.frameCount == 0) {
- buffer.frameCount = inFrameCount;
- provider->getNextBuffer(&buffer);
- if (buffer.raw == NULL) {
+ while (mBuffer.frameCount == 0) {
+ mBuffer.frameCount = inFrameCount;
+ provider->getNextBuffer(&mBuffer,
+ calculateOutputPTS(outputIndex / 2));
+ if (mBuffer.raw == NULL) {
goto resample_exit;
}
const uint32_t phaseIndex = phaseFraction >> kNumPhaseBits;
if (phaseIndex == 1) {
// read one frame
- read<CHANNELS>(impulse, phaseFraction, buffer.i16, inputIndex);
+ read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);
} else if (phaseIndex == 2) {
// read 2 frames
- read<CHANNELS>(impulse, phaseFraction, buffer.i16, inputIndex);
+ read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);
inputIndex++;
if (inputIndex >= mBuffer.frameCount) {
inputIndex -= mBuffer.frameCount;
- provider->releaseBuffer(&buffer);
+ provider->releaseBuffer(&mBuffer);
} else {
- read<CHANNELS>(impulse, phaseFraction, buffer.i16, inputIndex);
+ read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);
}
- }
+ }
}
- int16_t *in = buffer.i16;
- const size_t frameCount = buffer.frameCount;
+ int16_t *in = mBuffer.i16;
+ const size_t frameCount = mBuffer.frameCount;
// Always read-in the first samples from the input buffer
int16_t* head = impulse + halfNumCoefs*CHANNELS;
@@ -247,7 +247,7 @@
if (inputIndex >= frameCount)
break; // need a new buffer
read<CHANNELS>(impulse, phaseFraction, in, inputIndex);
- } else if(phaseIndex == 2) { // maximum value
+ } else if (phaseIndex == 2) { // maximum value
inputIndex++;
if (inputIndex >= frameCount)
break; // 0 frame available, 2 frames needed
@@ -264,7 +264,7 @@
// if done with buffer, save samples
if (inputIndex >= frameCount) {
inputIndex -= frameCount;
- provider->releaseBuffer(&buffer);
+ provider->releaseBuffer(&mBuffer);
}
}
@@ -284,7 +284,7 @@
**/
void AudioResamplerSinc::read(
int16_t*& impulse, uint32_t& phaseFraction,
- int16_t const* in, size_t inputIndex)
+ const int16_t* in, size_t inputIndex)
{
const uint32_t phaseIndex = phaseFraction >> kNumPhaseBits;
impulse += CHANNELS;
@@ -302,7 +302,7 @@
template<int CHANNELS>
void AudioResamplerSinc::filterCoefficient(
- int32_t& l, int32_t& r, uint32_t phase, int16_t const *samples)
+ int32_t& l, int32_t& r, uint32_t phase, const int16_t *samples)
{
// compute the index of the coefficient on the positive side and
// negative side
@@ -317,9 +317,9 @@
l = 0;
r = 0;
- int32_t const* coefs = mFirCoefs;
- int16_t const *sP = samples;
- int16_t const *sN = samples+CHANNELS;
+ const int32_t* coefs = mFirCoefs;
+ const int16_t *sP = samples;
+ const int16_t *sN = samples+CHANNELS;
for (unsigned int i=0 ; i<halfNumCoefs/4 ; i++) {
interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP);
interpolate<CHANNELS>(l, r, coefs+indexN, lerpN, sN);
@@ -339,13 +339,13 @@
template<int CHANNELS>
void AudioResamplerSinc::interpolate(
int32_t& l, int32_t& r,
- int32_t const* coefs, int16_t lerp, int16_t const* samples)
+ const int32_t* coefs, int16_t lerp, const int16_t* samples)
{
int32_t c0 = coefs[0];
int32_t c1 = coefs[1];
int32_t sinc = mulAdd(lerp, (c1-c0)<<1, c0);
if (CHANNELS == 2) {
- uint32_t rl = *reinterpret_cast<uint32_t const*>(samples);
+ uint32_t rl = *reinterpret_cast<const uint32_t*>(samples);
l = mulAddRL(1, rl, sinc, l);
r = mulAddRL(0, rl, sinc, r);
} else {
@@ -355,4 +355,3 @@
// ----------------------------------------------------------------------------
}; // namespace android
-
diff --git a/services/audioflinger/AudioResamplerSinc.h b/services/audioflinger/AudioResamplerSinc.h
index e6cb90b..f0a07b8 100644
--- a/services/audioflinger/AudioResamplerSinc.h
+++ b/services/audioflinger/AudioResamplerSinc.h
@@ -31,7 +31,7 @@
public:
AudioResamplerSinc(int bitDepth, int inChannelCount, int32_t sampleRate);
- ~AudioResamplerSinc();
+ virtual ~AudioResamplerSinc();
virtual void resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
@@ -44,22 +44,22 @@
template<int CHANNELS>
inline void filterCoefficient(
- int32_t& l, int32_t& r, uint32_t phase, int16_t const *samples);
+ int32_t& l, int32_t& r, uint32_t phase, const int16_t *samples);
template<int CHANNELS>
inline void interpolate(
int32_t& l, int32_t& r,
- int32_t const* coefs, int16_t lerp, int16_t const* samples);
+ const int32_t* coefs, int16_t lerp, const int16_t* samples);
template<int CHANNELS>
inline void read(int16_t*& impulse, uint32_t& phaseFraction,
- int16_t const* in, size_t inputIndex);
+ const int16_t* in, size_t inputIndex);
int16_t *mState;
int16_t *mImpulse;
int16_t *mRingFull;
- int32_t const * mFirCoefs;
+ const int32_t * mFirCoefs;
static const int32_t mFirCoefsDown[];
static const int32_t mFirCoefsUp[];
diff --git a/services/audioflinger/ServiceUtilities.cpp b/services/audioflinger/ServiceUtilities.cpp
new file mode 100644
index 0000000..6a58852
--- /dev/null
+++ b/services/audioflinger/ServiceUtilities.cpp
@@ -0,0 +1,55 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <binder/IPCThreadState.h>
+#include <binder/IServiceManager.h>
+#include <binder/PermissionCache.h>
+#include "ServiceUtilities.h"
+
+namespace android {
+
+// This optimization assumes mediaserver process doesn't fork, which it doesn't
+const pid_t getpid_cached = getpid();
+
+bool recordingAllowed() {
+ if (getpid_cached == IPCThreadState::self()->getCallingPid()) return true;
+ static const String16 sRecordAudio("android.permission.RECORD_AUDIO");
+ // don't use PermissionCache; this is not a system permission
+ bool ok = checkCallingPermission(sRecordAudio);
+ if (!ok) ALOGE("Request requires android.permission.RECORD_AUDIO");
+ return ok;
+}
+
+bool settingsAllowed() {
+ if (getpid_cached == IPCThreadState::self()->getCallingPid()) return true;
+ static const String16 sAudioSettings("android.permission.MODIFY_AUDIO_SETTINGS");
+ // don't use PermissionCache; this is not a system permission
+ bool ok = checkCallingPermission(sAudioSettings);
+ if (!ok) ALOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
+ return ok;
+}
+
+bool dumpAllowed() {
+ // don't optimize for same pid, since mediaserver never dumps itself
+ static const String16 sDump("android.permission.DUMP");
+ // OK to use PermissionCache; this is a system permission
+ bool ok = PermissionCache::checkCallingPermission(sDump);
+ // convention is for caller to dump an error message to fd instead of logging here
+ //if (!ok) ALOGE("Request requires android.permission.DUMP");
+ return ok;
+}
+
+} // namespace android
diff --git a/include/media/thread_init.h b/services/audioflinger/ServiceUtilities.h
similarity index 73%
rename from include/media/thread_init.h
rename to services/audioflinger/ServiceUtilities.h
index 2feac86..f77ec5b 100644
--- a/include/media/thread_init.h
+++ b/services/audioflinger/ServiceUtilities.h
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2008 The Android Open Source Project
+ * Copyright (C) 2012 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
@@ -14,11 +14,14 @@
* limitations under the License.
*/
-#ifndef THREAD_INIT_H
-#define THREAD_INIT_H
+#include <unistd.h>
-bool InitializeForThread();
-void UninitializeForThread();
+namespace android {
-#endif /* THREAD_INIT_H*/
-
+extern const pid_t getpid_cached;
+
+bool recordingAllowed();
+bool settingsAllowed();
+bool dumpAllowed();
+
+}
diff --git a/services/camera/libcameraservice/Android.mk b/services/camera/libcameraservice/Android.mk
index e35435e..3cae1f5 100644
--- a/services/camera/libcameraservice/Android.mk
+++ b/services/camera/libcameraservice/Android.mk
@@ -15,6 +15,7 @@
libbinder \
libcutils \
libmedia \
+ libmedia_native \
libcamera_client \
libgui \
libhardware
diff --git a/services/camera/libcameraservice/CameraHardwareInterface.h b/services/camera/libcameraservice/CameraHardwareInterface.h
index 34087b5..87a0802 100644
--- a/services/camera/libcameraservice/CameraHardwareInterface.h
+++ b/services/camera/libcameraservice/CameraHardwareInterface.h
@@ -21,8 +21,6 @@
#include <binder/MemoryBase.h>
#include <binder/MemoryHeapBase.h>
#include <utils/RefBase.h>
-#include <surfaceflinger/ISurface.h>
-#include <ui/android_native_buffer.h>
#include <ui/GraphicBuffer.h>
#include <camera/Camera.h>
#include <camera/CameraParameters.h>
@@ -635,6 +633,12 @@
return native_window_set_crop(a, &crop);
}
+ static int __set_timestamp(struct preview_stream_ops *w,
+ int64_t timestamp) {
+ ANativeWindow *a = anw(w);
+ return native_window_set_buffers_timestamp(a, timestamp);
+ }
+
static int __set_usage(struct preview_stream_ops* w, int usage)
{
ANativeWindow *a = anw(w);
@@ -664,6 +668,7 @@
mHalPreviewWindow.nw.set_buffer_count = __set_buffer_count;
mHalPreviewWindow.nw.set_buffers_geometry = __set_buffers_geometry;
mHalPreviewWindow.nw.set_crop = __set_crop;
+ mHalPreviewWindow.nw.set_timestamp = __set_timestamp;
mHalPreviewWindow.nw.set_usage = __set_usage;
mHalPreviewWindow.nw.set_swap_interval = __set_swap_interval;
diff --git a/services/camera/libcameraservice/CameraService.cpp b/services/camera/libcameraservice/CameraService.cpp
index 918f31e..22836e3 100644
--- a/services/camera/libcameraservice/CameraService.cpp
+++ b/services/camera/libcameraservice/CameraService.cpp
@@ -21,6 +21,7 @@
#include <stdio.h>
#include <sys/types.h>
#include <pthread.h>
+#include <time.h>
#include <binder/IPCThreadState.h>
#include <binder/IServiceManager.h>
@@ -29,10 +30,11 @@
#include <cutils/atomic.h>
#include <cutils/properties.h>
#include <gui/SurfaceTextureClient.h>
+#include <gui/Surface.h>
#include <hardware/hardware.h>
#include <media/AudioSystem.h>
#include <media/mediaplayer.h>
-#include <surfaceflinger/ISurface.h>
+#include <utils/Condition.h>
#include <utils/Errors.h>
#include <utils/Log.h>
#include <utils/String16.h>
@@ -42,6 +44,8 @@
namespace android {
+#define WAIT_RELEASE_TIMEOUT 250 // 250ms
+
// ----------------------------------------------------------------------------
// Logging support -- this is for debugging only
// Use "adb shell dumpsys media.camera -v 1" to change it.
@@ -64,6 +68,13 @@
return IPCThreadState::self()->getCallingUid();
}
+static long long getTimeInMs() {
+ struct timeval t;
+ t.tv_sec = t.tv_usec = 0;
+ gettimeofday(&t, NULL);
+ return t.tv_sec * 1000LL + t.tv_usec / 1000;
+}
+
// ----------------------------------------------------------------------------
// This is ugly and only safe if we never re-create the CameraService, but
@@ -131,7 +142,7 @@
}
sp<ICamera> CameraService::connect(
- const sp<ICameraClient>& cameraClient, int cameraId) {
+ const sp<ICameraClient>& cameraClient, int cameraId, bool force, bool keep) {
int callingPid = getCallingPid();
sp<CameraHardwareInterface> hardware = NULL;
@@ -157,27 +168,73 @@
return NULL;
}
- Mutex::Autolock lock(mServiceLock);
- if (mClient[cameraId] != 0) {
- client = mClient[cameraId].promote();
- if (client != 0) {
- if (cameraClient->asBinder() == client->getCameraClient()->asBinder()) {
- LOG1("CameraService::connect X (pid %d) (the same client)",
- callingPid);
- return client;
- } else {
- ALOGW("CameraService::connect X (pid %d) rejected (existing client).",
- callingPid);
- return NULL;
- }
- }
- mClient[cameraId].clear();
+ if (keep && !checkCallingPermission(String16("android.permission.KEEP_CAMERA"))) {
+ ALOGE("connect X (pid %d) rejected (no KEEP_CAMERA permission).", callingPid);
+ return NULL;
}
- if (mBusy[cameraId]) {
- ALOGW("CameraService::connect X (pid %d) rejected"
- " (camera %d is still busy).", callingPid, cameraId);
- return NULL;
+ Mutex::Autolock lock(mServiceLock);
+ // Check if there is an existing client.
+ client = mClient[cameraId].promote();
+ if (client != 0 &&
+ cameraClient->asBinder() == client->getCameraClient()->asBinder()) {
+ LOG1("connect X (pid %d) (the same client)", callingPid);
+ return client;
+ }
+
+ if (!force) {
+ if (mClient[cameraId].promote() != 0) {
+ ALOGW("connect X (pid %d) rejected (existing client).", callingPid);
+ return NULL;
+ }
+ mClient[cameraId].clear();
+ if (mBusy[cameraId]) {
+ ALOGW("connect X (pid %d) rejected (camera %d is still busy).",
+ callingPid, cameraId);
+ return NULL;
+ }
+ } else { // force == true
+ int i = 0;
+ long long start_time = getTimeInMs();
+ while (i < mNumberOfCameras) {
+ if (getTimeInMs() - start_time >= 3000LL) {
+ ALOGE("connect X (pid %d) rejected (timeout 3s)", callingPid);
+ return NULL;
+ }
+
+ client = mClient[i].promote();
+ if (client != 0) {
+ if (client->keep()) {
+ ALOGW("connect X (pid %d) rejected (existing client wants to keeps the camera)",
+ callingPid);
+ return NULL;
+ } else {
+ ALOGW("New client (pid %d, id=%d). Disconnect the existing client (id=%d).",
+ callingPid, cameraId, i);
+ // Do not hold mServiceLock because disconnect will try to get it.
+ mServiceLock.unlock();
+ client->notifyCallback(CAMERA_MSG_ERROR, CAMERA_ERROR_RELEASED, 0, &i);
+ client->waitRelease(WAIT_RELEASE_TIMEOUT);
+ client->disconnectInternal(false);
+ mServiceLock.lock();
+ // Restart from the first client because a new client may have connected
+ // when mServiceLock is unlocked.
+ i = 0;
+ continue;
+ }
+ }
+
+ if (mBusy[i]) {
+ // Give the client a chance to release the hardware.
+ mServiceLock.unlock();
+ usleep(10 * 1000);
+ mServiceLock.lock();
+ i = 0; // Restart from the first client
+ continue;
+ }
+
+ i++;
+ }
}
struct camera_info info;
@@ -195,9 +252,15 @@
return NULL;
}
- client = new Client(this, cameraClient, hardware, cameraId, info.facing, callingPid);
+ client = new Client(this, cameraClient, hardware, cameraId, info.facing,
+ callingPid, keep);
+ // We need to clear the hardware here. After the destructor of mServiceLock
+ // finishes, a new client may connect and disconnect this client. If this
+ // reference is not cleared, the destructor of CameraHardwareInterface
+ // cannot run. The new client will not be able to connect.
+ hardware.clear();
mClient[cameraId] = client;
- LOG1("CameraService::connect X");
+ LOG1("CameraService::connect X (id %d)", cameraId);
return client;
}
@@ -331,9 +394,9 @@
CameraService::Client::Client(const sp<CameraService>& cameraService,
const sp<ICameraClient>& cameraClient,
const sp<CameraHardwareInterface>& hardware,
- int cameraId, int cameraFacing, int clientPid) {
+ int cameraId, int cameraFacing, int clientPid, bool keep) {
int callingPid = getCallingPid();
- LOG1("Client::Client E (pid %d)", callingPid);
+ LOG1("Client::Client E (pid %d, id %d)", callingPid, cameraId);
mCameraService = cameraService;
mCameraClient = cameraClient;
@@ -341,6 +404,7 @@
mCameraId = cameraId;
mCameraFacing = cameraFacing;
mClientPid = clientPid;
+ mKeep = keep;
mMsgEnabled = 0;
mSurface = 0;
mPreviewWindow = 0;
@@ -351,7 +415,7 @@
// Enable zoom, error, focus, and metadata messages by default
enableMsgType(CAMERA_MSG_ERROR | CAMERA_MSG_ZOOM | CAMERA_MSG_FOCUS |
- CAMERA_MSG_PREVIEW_METADATA);
+ CAMERA_MSG_PREVIEW_METADATA | CAMERA_MSG_FOCUS_MOVE);
// Callback is disabled by default
mPreviewCallbackFlag = CAMERA_FRAME_CALLBACK_FLAG_NOOP;
@@ -359,7 +423,7 @@
mPlayShutterSound = true;
cameraService->setCameraBusy(cameraId);
cameraService->loadSound();
- LOG1("Client::Client X (pid %d)", callingPid);
+ LOG1("Client::Client X (pid %d, id %d)", callingPid, cameraId);
}
// tear down the client
@@ -468,18 +532,24 @@
}
void CameraService::Client::disconnect() {
+ disconnectInternal(true);
+}
+
+void CameraService::Client::disconnectInternal(bool needCheckPid) {
int callingPid = getCallingPid();
- LOG1("disconnect E (pid %d)", callingPid);
+ LOG1("disconnectInternal E (pid %d)", callingPid);
Mutex::Autolock lock(mLock);
- if (checkPid() != NO_ERROR) {
- ALOGW("different client - don't disconnect");
- return;
- }
+ if (needCheckPid) {
+ if (checkPid() != NO_ERROR) {
+ ALOGW("different client - don't disconnect");
+ return;
+ }
- if (mClientPid <= 0) {
- LOG1("camera is unlocked (mClientPid = %d), don't tear down hardware", mClientPid);
- return;
+ if (mClientPid <= 0) {
+ LOG1("camera is unlocked (mClientPid = %d), don't tear down hardware", mClientPid);
+ return;
+ }
}
// Make sure disconnect() is done once and once only, whether it is called
@@ -506,8 +576,16 @@
mCameraService->removeClient(mCameraClient);
mCameraService->setCameraFree(mCameraId);
+ mReleaseCondition.signal();
- LOG1("disconnect X (pid %d)", callingPid);
+ LOG1("disconnectInternal X (pid %d)", callingPid);
+}
+
+void CameraService::Client::waitRelease(int ms) {
+ Mutex::Autolock lock(mLock);
+ if (mHardware != 0) {
+ mReleaseCondition.waitRelative(mLock, ms * 1000000);
+ }
}
// ----------------------------------------------------------------------------
@@ -874,6 +952,9 @@
return OK;
} else if (cmd == CAMERA_CMD_PLAY_RECORDING_SOUND) {
mCameraService->playSound(SOUND_RECORDING);
+ } else if (cmd == CAMERA_CMD_PING) {
+ // If mHardware is 0, checkPidAndHardware will return error.
+ return OK;
}
return mHardware->sendCommand(cmd, arg1, arg2);
@@ -1217,6 +1298,10 @@
return -1;
}
+// Whether the client wants to keep the camera from taking
+bool CameraService::Client::keep() const {
+ return mKeep;
+}
// ----------------------------------------------------------------------------
diff --git a/services/camera/libcameraservice/CameraService.h b/services/camera/libcameraservice/CameraService.h
index bad41f5..457c79b 100644
--- a/services/camera/libcameraservice/CameraService.h
+++ b/services/camera/libcameraservice/CameraService.h
@@ -46,7 +46,8 @@
virtual int32_t getNumberOfCameras();
virtual status_t getCameraInfo(int cameraId,
struct CameraInfo* cameraInfo);
- virtual sp<ICamera> connect(const sp<ICameraClient>& cameraClient, int cameraId);
+ virtual sp<ICamera> connect(const sp<ICameraClient>& cameraClient, int cameraId,
+ bool force, bool keep);
virtual void removeClient(const sp<ICameraClient>& cameraClient);
virtual sp<Client> getClientById(int cameraId);
@@ -114,7 +115,8 @@
const sp<CameraHardwareInterface>& hardware,
int cameraId,
int cameraFacing,
- int clientPid);
+ int clientPid,
+ bool keep);
~Client();
// return our camera client
@@ -172,12 +174,19 @@
const sp<IBinder>& binder,
const sp<ANativeWindow>& window);
+ void disconnectInternal(bool needCheckPid);
+ bool keep() const;
+ void waitRelease(int ms);
+
+
// these are initialized in the constructor.
sp<CameraService> mCameraService; // immutable after constructor
sp<ICameraClient> mCameraClient;
int mCameraId; // immutable after constructor
int mCameraFacing; // immutable after constructor
pid_t mClientPid;
+ // Client wants to keep the camera from taking by other clients.
+ bool mKeep;
sp<CameraHardwareInterface> mHardware; // cleared after disconnect()
int mPreviewCallbackFlag;
int mOrientation; // Current display orientation
@@ -185,6 +194,8 @@
// Ensures atomicity among the public methods
mutable Mutex mLock;
+ // This will get notified when the hardware is released.
+ Condition mReleaseCondition;
// This is a binder of Surface or SurfaceTexture.
sp<IBinder> mSurface;
sp<ANativeWindow> mPreviewWindow;
diff --git a/services/camera/tests/CameraServiceTest/CameraServiceTest.cpp b/services/camera/tests/CameraServiceTest/CameraServiceTest.cpp
index 1055538..e417b79 100644
--- a/services/camera/tests/CameraServiceTest/CameraServiceTest.cpp
+++ b/services/camera/tests/CameraServiceTest/CameraServiceTest.cpp
@@ -22,7 +22,6 @@
#include <sys/types.h>
#include <sys/wait.h>
#include <unistd.h>
-#include <surfaceflinger/ISurface.h>
#include <camera/Camera.h>
#include <camera/CameraParameters.h>
#include <ui/GraphicBuffer.h>