Merge "audio policy: add new input audio device"
diff --git a/camera/Camera.cpp b/camera/Camera.cpp
index 22199fa..85f44f0 100644
--- a/camera/Camera.cpp
+++ b/camera/Camera.cpp
@@ -77,6 +77,32 @@
return CameraBaseT::connect(cameraId, clientPackageName, clientUid);
}
+status_t Camera::connectLegacy(int cameraId, int halVersion,
+ const String16& clientPackageName,
+ int clientUid,
+ sp<Camera>& camera)
+{
+ ALOGV("%s: connect legacy camera device", __FUNCTION__);
+ sp<Camera> c = new Camera(cameraId);
+ sp<ICameraClient> cl = c;
+ status_t status = NO_ERROR;
+ const sp<ICameraService>& cs = CameraBaseT::getCameraService();
+
+ if (cs != 0) {
+ status = cs.get()->connectLegacy(cl, cameraId, halVersion, clientPackageName,
+ clientUid, /*out*/c->mCamera);
+ }
+ if (status == OK && c->mCamera != 0) {
+ c->mCamera->asBinder()->linkToDeath(c);
+ c->mStatus = NO_ERROR;
+ camera = c;
+ } else {
+ ALOGW("An error occurred while connecting to camera: %d", cameraId);
+ c.clear();
+ }
+ return status;
+}
+
status_t Camera::reconnect()
{
ALOGV("reconnect");
diff --git a/camera/ICameraService.cpp b/camera/ICameraService.cpp
index 79c33f9..5485205 100644
--- a/camera/ICameraService.cpp
+++ b/camera/ICameraService.cpp
@@ -186,6 +186,29 @@
return status;
}
+ // connect to camera service (android.hardware.Camera)
+ virtual status_t connectLegacy(const sp<ICameraClient>& cameraClient, int cameraId,
+ int halVersion,
+ const String16 &clientPackageName, int clientUid,
+ /*out*/sp<ICamera>& device)
+ {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICameraService::getInterfaceDescriptor());
+ data.writeStrongBinder(cameraClient->asBinder());
+ data.writeInt32(cameraId);
+ data.writeInt32(halVersion);
+ data.writeString16(clientPackageName);
+ data.writeInt32(clientUid);
+ remote()->transact(BnCameraService::CONNECT_LEGACY, data, &reply);
+
+ if (readExceptionCode(reply)) return -EPROTO;
+ status_t status = reply.readInt32();
+ if (reply.readInt32() != 0) {
+ device = interface_cast<ICamera>(reply.readStrongBinder());
+ }
+ return status;
+ }
+
// connect to camera service (pro client)
virtual status_t connectPro(const sp<IProCameraCallbacks>& cameraCb, int cameraId,
const String16 &clientPackageName, int clientUid,
@@ -446,6 +469,27 @@
reply->writeInt32(supportsCameraApi(cameraId, apiVersion));
return NO_ERROR;
} break;
+ case CONNECT_LEGACY: {
+ CHECK_INTERFACE(ICameraService, data, reply);
+ sp<ICameraClient> cameraClient =
+ interface_cast<ICameraClient>(data.readStrongBinder());
+ int32_t cameraId = data.readInt32();
+ int32_t halVersion = data.readInt32();
+ const String16 clientName = data.readString16();
+ int32_t clientUid = data.readInt32();
+ sp<ICamera> camera;
+ status_t status = connectLegacy(cameraClient, cameraId, halVersion,
+ clientName, clientUid, /*out*/camera);
+ reply->writeNoException();
+ reply->writeInt32(status);
+ if (camera != NULL) {
+ reply->writeInt32(1);
+ reply->writeStrongBinder(camera->asBinder());
+ } else {
+ reply->writeInt32(0);
+ }
+ return NO_ERROR;
+ } break;
default:
return BBinder::onTransact(code, data, reply, flags);
}
diff --git a/cmds/screenrecord/FrameOutput.cpp b/cmds/screenrecord/FrameOutput.cpp
index 4da16bc..03e0062 100644
--- a/cmds/screenrecord/FrameOutput.cpp
+++ b/cmds/screenrecord/FrameOutput.cpp
@@ -71,7 +71,7 @@
sp<IGraphicBufferConsumer> consumer;
BufferQueue::createBufferQueue(&producer, &consumer);
mGlConsumer = new GLConsumer(consumer, mExtTextureName,
- GL_TEXTURE_EXTERNAL_OES);
+ GL_TEXTURE_EXTERNAL_OES, true, false);
mGlConsumer->setName(String8("virtual display"));
mGlConsumer->setDefaultBufferSize(width, height);
mGlConsumer->setDefaultMaxBufferCount(5);
diff --git a/cmds/screenrecord/Overlay.cpp b/cmds/screenrecord/Overlay.cpp
index c2a8f1b..7fef53d 100644
--- a/cmds/screenrecord/Overlay.cpp
+++ b/cmds/screenrecord/Overlay.cpp
@@ -14,6 +14,10 @@
* limitations under the License.
*/
+#include <assert.h>
+#include <inttypes.h>
+#include <stdlib.h>
+
#define LOG_TAG "ScreenRecord"
//#define LOG_NDEBUG 0
#include <utils/Log.h>
@@ -27,9 +31,6 @@
#include <GLES2/gl2.h>
#include <GLES2/gl2ext.h>
-#include <stdlib.h>
-#include <assert.h>
-
#include "screenrecord.h"
#include "Overlay.h"
#include "TextRenderer.h"
@@ -172,7 +173,7 @@
sp<IGraphicBufferConsumer> consumer;
BufferQueue::createBufferQueue(&mProducer, &consumer);
mGlConsumer = new GLConsumer(consumer, mExtTextureName,
- GL_TEXTURE_EXTERNAL_OES);
+ GL_TEXTURE_EXTERNAL_OES, true, false);
mGlConsumer->setName(String8("virtual display"));
mGlConsumer->setDefaultBufferSize(width, height);
mGlConsumer->setDefaultMaxBufferCount(5);
@@ -235,7 +236,7 @@
char textBuf[64];
getTimeString_l(monotonicNsec, textBuf, sizeof(textBuf));
- String8 timeStr(String8::format("%s f=%lld (%zd)",
+ String8 timeStr(String8::format("%s f=%" PRId64 " (%zd)",
textBuf, frameNumber, mTotalDroppedFrames));
mTextRenderer.drawString(mTexProgram, Program::kIdentity, 0, 0, timeStr);
diff --git a/cmds/screenrecord/TextRenderer.cpp b/cmds/screenrecord/TextRenderer.cpp
index 784055c..6a9176b 100644
--- a/cmds/screenrecord/TextRenderer.cpp
+++ b/cmds/screenrecord/TextRenderer.cpp
@@ -353,6 +353,6 @@
}
}
- ALOGV("goodPos=%d for str='%s'", goodPos, str);
+ ALOGV("goodPos=%zu for str='%s'", goodPos, str);
return const_cast<char*>(str + goodPos);
}
diff --git a/cmds/screenrecord/screenrecord.cpp b/cmds/screenrecord/screenrecord.cpp
index 02ed53a..02df1d2 100644
--- a/cmds/screenrecord/screenrecord.cpp
+++ b/cmds/screenrecord/screenrecord.cpp
@@ -14,6 +14,19 @@
* limitations under the License.
*/
+#include <assert.h>
+#include <ctype.h>
+#include <fcntl.h>
+#include <inttypes.h>
+#include <getopt.h>
+#include <signal.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <sys/wait.h>
+#include <termios.h>
+#include <unistd.h>
+
#define LOG_TAG "ScreenRecord"
#define ATRACE_TAG ATRACE_TAG_GRAPHICS
//#define LOG_NDEBUG 0
@@ -36,18 +49,6 @@
#include <media/stagefright/MediaMuxer.h>
#include <media/ICrypto.h>
-#include <stdlib.h>
-#include <unistd.h>
-#include <string.h>
-#include <stdio.h>
-#include <ctype.h>
-#include <fcntl.h>
-#include <signal.h>
-#include <getopt.h>
-#include <sys/wait.h>
-#include <termios.h>
-#include <assert.h>
-
#include "screenrecord.h"
#include "Overlay.h"
#include "FrameOutput.h"
@@ -354,7 +355,7 @@
case NO_ERROR:
// got a buffer
if ((flags & MediaCodec::BUFFER_FLAG_CODECCONFIG) != 0) {
- ALOGV("Got codec config buffer (%u bytes)", size);
+ ALOGV("Got codec config buffer (%zu bytes)", size);
if (muxer != NULL) {
// ignore this -- we passed the CSD into MediaMuxer when
// we got the format change notification
@@ -362,7 +363,7 @@
}
}
if (size != 0) {
- ALOGV("Got data in buffer %d, size=%d, pts=%lld",
+ ALOGV("Got data in buffer %zu, size=%zu, pts=%" PRId64,
bufIndex, size, ptsUsec);
{ // scope
@@ -473,7 +474,7 @@
ALOGV("Encoder stopping (req=%d)", gStopRequested);
if (gVerbose) {
- printf("Encoder stopping; recorded %u frames in %lld seconds\n",
+ printf("Encoder stopping; recorded %u frames in %" PRId64 " seconds\n",
debugNumFrames, nanoseconds_to_seconds(
systemTime(CLOCK_MONOTONIC) - startWhenNsec));
}
diff --git a/cmds/stagefright/sf2.cpp b/cmds/stagefright/sf2.cpp
index 3c0c7ec..0f729a3 100644
--- a/cmds/stagefright/sf2.cpp
+++ b/cmds/stagefright/sf2.cpp
@@ -211,28 +211,28 @@
int32_t what;
CHECK(msg->findInt32("what", &what));
- if (what == ACodec::kWhatFillThisBuffer) {
+ if (what == CodecBase::kWhatFillThisBuffer) {
onFillThisBuffer(msg);
- } else if (what == ACodec::kWhatDrainThisBuffer) {
+ } else if (what == CodecBase::kWhatDrainThisBuffer) {
if ((mNumOutputBuffersReceived++ % 16) == 0) {
printf(".");
fflush(stdout);
}
onDrainThisBuffer(msg);
- } else if (what == ACodec::kWhatEOS
- || what == ACodec::kWhatError) {
- printf((what == ACodec::kWhatEOS) ? "$\n" : "E\n");
+ } else if (what == CodecBase::kWhatEOS
+ || what == CodecBase::kWhatError) {
+ printf((what == CodecBase::kWhatEOS) ? "$\n" : "E\n");
printStatistics();
(new AMessage(kWhatStop, id()))->post();
- } else if (what == ACodec::kWhatFlushCompleted) {
+ } else if (what == CodecBase::kWhatFlushCompleted) {
mSeekState = SEEK_FLUSH_COMPLETED;
mCodec->signalResume();
(new AMessage(kWhatSeek, id()))->post(5000000ll);
- } else if (what == ACodec::kWhatOutputFormatChanged) {
- } else if (what == ACodec::kWhatShutdownCompleted) {
+ } else if (what == CodecBase::kWhatOutputFormatChanged) {
+ } else if (what == CodecBase::kWhatShutdownCompleted) {
mDecodeLooper->unregisterHandler(mCodec->id());
if (mDecodeLooper != looper()) {
@@ -240,12 +240,6 @@
}
looper()->stop();
- } else if (what == ACodec::kWhatError) {
- ALOGE("something went wrong, codec reported an error.");
-
- printf("E\n");
-
- (new AMessage(kWhatStop, id()))->post();
}
break;
}
diff --git a/cmds/stagefright/stagefright.cpp b/cmds/stagefright/stagefright.cpp
index b70afe6..81edcb4 100644
--- a/cmds/stagefright/stagefright.cpp
+++ b/cmds/stagefright/stagefright.cpp
@@ -942,7 +942,9 @@
sp<IGraphicBufferProducer> producer;
sp<IGraphicBufferConsumer> consumer;
BufferQueue::createBufferQueue(&producer, &consumer);
- sp<GLConsumer> texture = new GLConsumer(consumer, 0 /* tex */);
+ sp<GLConsumer> texture = new GLConsumer(consumer, 0 /* tex */,
+ GLConsumer::TEXTURE_EXTERNAL, true /* useFenceSync */,
+ false /* isControlledByApp */);
gSurface = new Surface(producer);
}
diff --git a/drm/drmserver/DrmManagerService.cpp b/drm/drmserver/DrmManagerService.cpp
index 2b71904..63341e0 100644
--- a/drm/drmserver/DrmManagerService.cpp
+++ b/drm/drmserver/DrmManagerService.cpp
@@ -34,7 +34,18 @@
static Vector<uid_t> trustedUids;
static bool isProtectedCallAllowed() {
- return true;
+ // TODO
+ // Following implementation is just for reference.
+ // Each OEM manufacturer should implement/replace with their own solutions.
+ IPCThreadState* ipcState = IPCThreadState::self();
+ uid_t uid = ipcState->getCallingUid();
+
+ for (unsigned int i = 0; i < trustedUids.size(); ++i) {
+ if (trustedUids[i] == uid) {
+ return true;
+ }
+ }
+ return false;
}
void DrmManagerService::instantiate() {
diff --git a/drm/mediadrm/plugins/clearkey/AesCtrDecryptor.cpp b/drm/mediadrm/plugins/clearkey/AesCtrDecryptor.cpp
new file mode 100644
index 0000000..01f8d65
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/AesCtrDecryptor.cpp
@@ -0,0 +1,67 @@
+/*
+ * Copyright (C) 2014 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 "ClearKeyCryptoPlugin"
+#include <utils/Log.h>
+
+#include <openssl/aes.h>
+
+#include "AesCtrDecryptor.h"
+
+namespace clearkeydrm {
+
+static const size_t kBlockBitCount = kBlockSize * 8;
+
+android::status_t AesCtrDecryptor::decrypt(const android::Vector<uint8_t>& key,
+ const Iv iv, const uint8_t* source,
+ uint8_t* destination,
+ const SubSample* subSamples,
+ size_t numSubSamples,
+ size_t* bytesDecryptedOut) {
+ uint32_t blockOffset = 0;
+ uint8_t previousEncryptedCounter[kBlockSize];
+ memset(previousEncryptedCounter, 0, kBlockSize);
+
+ size_t offset = 0;
+ AES_KEY opensslKey;
+ AES_set_encrypt_key(key.array(), kBlockBitCount, &opensslKey);
+ Iv opensslIv;
+ memcpy(opensslIv, iv, sizeof(opensslIv));
+
+ for (size_t i = 0; i < numSubSamples; ++i) {
+ const SubSample& subSample = subSamples[i];
+
+ if (subSample.mNumBytesOfClearData > 0) {
+ memcpy(destination + offset, source + offset,
+ subSample.mNumBytesOfClearData);
+ offset += subSample.mNumBytesOfClearData;
+ }
+
+ if (subSample.mNumBytesOfEncryptedData > 0) {
+ AES_ctr128_encrypt(source + offset, destination + offset,
+ subSample.mNumBytesOfEncryptedData, &opensslKey,
+ opensslIv, previousEncryptedCounter,
+ &blockOffset);
+ offset += subSample.mNumBytesOfEncryptedData;
+ }
+ }
+
+ *bytesDecryptedOut = offset;
+ return android::OK;
+}
+
+} // namespace clearkeydrm
diff --git a/drm/mediadrm/plugins/clearkey/AesCtrDecryptor.h b/drm/mediadrm/plugins/clearkey/AesCtrDecryptor.h
new file mode 100644
index 0000000..b416266
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/AesCtrDecryptor.h
@@ -0,0 +1,44 @@
+/*
+ * Copyright (C) 2014 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 CLEARKEY_AES_CTR_DECRYPTOR_H_
+#define CLEARKEY_AES_CTR_DECRYPTOR_H_
+
+#include <media/stagefright/foundation/ABase.h>
+#include <Utils.h>
+#include <utils/Errors.h>
+#include <utils/Vector.h>
+
+#include "ClearKeyTypes.h"
+
+namespace clearkeydrm {
+
+class AesCtrDecryptor {
+public:
+ AesCtrDecryptor() {}
+
+ android::status_t decrypt(const android::Vector<uint8_t>& key, const Iv iv,
+ const uint8_t* source, uint8_t* destination,
+ const SubSample* subSamples, size_t numSubSamples,
+ size_t* bytesDecryptedOut);
+
+private:
+ DISALLOW_EVIL_CONSTRUCTORS(AesCtrDecryptor);
+};
+
+} // namespace clearkeydrm
+
+#endif // CLEARKEY_AES_CTR_DECRYPTOR_H_
diff --git a/drm/mediadrm/plugins/clearkey/Android.mk b/drm/mediadrm/plugins/clearkey/Android.mk
new file mode 100644
index 0000000..22a85b4
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/Android.mk
@@ -0,0 +1,62 @@
+#
+# Copyright (C) 2014 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.
+#
+LOCAL_PATH:= $(call my-dir)
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES := \
+ AesCtrDecryptor.cpp \
+ ClearKeyUUID.cpp \
+ CreatePluginFactories.cpp \
+ CryptoFactory.cpp \
+ CryptoPlugin.cpp \
+ DrmFactory.cpp \
+ DrmPlugin.cpp \
+ InitDataParser.cpp \
+ JsonWebKey.cpp \
+ Session.cpp \
+ SessionLibrary.cpp \
+ Utils.cpp \
+
+LOCAL_C_INCLUDES := \
+ bionic \
+ external/jsmn \
+ external/openssl/include \
+ frameworks/av/drm/mediadrm/plugins/clearkey \
+ frameworks/av/include \
+ frameworks/native/include \
+
+LOCAL_MODULE := libdrmclearkeyplugin
+
+LOCAL_PROPRIETARY_MODULE := true
+LOCAL_MODULE_RELATIVE_PATH := mediadrm
+
+LOCAL_SHARED_LIBRARIES := \
+ libcrypto \
+ liblog \
+ libstagefright_foundation \
+ libutils \
+
+LOCAL_STATIC_LIBRARIES := \
+ libjsmn \
+
+LOCAL_MODULE_TAGS := optional
+
+include $(BUILD_SHARED_LIBRARY)
+
+#########################################################################
+# Build unit tests
+
+include $(LOCAL_PATH)/tests/Android.mk
diff --git a/drm/mediadrm/plugins/clearkey/ClearKeyTypes.h b/drm/mediadrm/plugins/clearkey/ClearKeyTypes.h
new file mode 100644
index 0000000..a28959a
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/ClearKeyTypes.h
@@ -0,0 +1,38 @@
+/*
+ * Copyright (C) 2014 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 CLEARKEY_TYPES_H_
+#define CLEARKEY_TYPES_H_
+
+#include <media/hardware/CryptoAPI.h>
+#include <openssl/aes.h>
+#include <utils/KeyedVector.h>
+#include <utils/Vector.h>
+
+namespace clearkeydrm {
+
+const uint8_t kBlockSize = AES_BLOCK_SIZE;
+typedef uint8_t KeyId[kBlockSize];
+typedef uint8_t Iv[kBlockSize];
+
+typedef android::CryptoPlugin::SubSample SubSample;
+
+typedef android::KeyedVector<android::Vector<uint8_t>,
+ android::Vector<uint8_t> > KeyMap;
+
+} // namespace clearkeydrm
+
+#endif // CLEARKEY_TYPES_H_
diff --git a/drm/mediadrm/plugins/clearkey/ClearKeyUUID.cpp b/drm/mediadrm/plugins/clearkey/ClearKeyUUID.cpp
new file mode 100644
index 0000000..ed050f7
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/ClearKeyUUID.cpp
@@ -0,0 +1,32 @@
+/*
+ * Copyright (C) 2014 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 <string.h>
+
+#include "ClearKeyUUID.h"
+
+namespace clearkeydrm {
+
+bool isClearKeyUUID(const uint8_t uuid[16]) {
+ static const uint8_t kClearKeyUUID[16] = {
+ 0x10,0x77,0xEF,0xEC,0xC0,0xB2,0x4D,0x02,
+ 0xAC,0xE3,0x3C,0x1E,0x52,0xE2,0xFB,0x4B
+ };
+
+ return !memcmp(uuid, kClearKeyUUID, sizeof(kClearKeyUUID));
+}
+
+} // namespace clearkeydrm
diff --git a/drm/mediadrm/plugins/clearkey/ClearKeyUUID.h b/drm/mediadrm/plugins/clearkey/ClearKeyUUID.h
new file mode 100644
index 0000000..ac99418
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/ClearKeyUUID.h
@@ -0,0 +1,28 @@
+/*
+ * Copyright (C) 2014 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 CLEARKEY_UUID_H_
+#define CLEARKEY_UUID_H_
+
+#include <stdint.h>
+
+namespace clearkeydrm {
+
+bool isClearKeyUUID(const uint8_t uuid[16]);
+
+} // namespace clearkeydrm
+
+#endif // CLEARKEY_UUID_H_
diff --git a/drm/mediadrm/plugins/clearkey/CreatePluginFactories.cpp b/drm/mediadrm/plugins/clearkey/CreatePluginFactories.cpp
new file mode 100644
index 0000000..ec1420e
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/CreatePluginFactories.cpp
@@ -0,0 +1,32 @@
+/*
+ * Copyright (C) 2014 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 "CreatePluginFactories.h"
+
+#include "CryptoFactory.h"
+#include "DrmFactory.h"
+
+extern "C" {
+
+android::DrmFactory* createDrmFactory() {
+ return new clearkeydrm::DrmFactory();
+}
+
+android::CryptoFactory* createCryptoFactory() {
+ return new clearkeydrm::CryptoFactory();
+}
+
+} // extern "C"
diff --git a/drm/mediadrm/plugins/clearkey/CreatePluginFactories.h b/drm/mediadrm/plugins/clearkey/CreatePluginFactories.h
new file mode 100644
index 0000000..d9acec1
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/CreatePluginFactories.h
@@ -0,0 +1,28 @@
+/*
+ * Copyright (C) 2014 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 CLEARKEY_CREATE_PLUGIN_FACTORIES_H_
+#define CLEARKEY_CREATE_PLUGIN_FACTORIES_H_
+
+#include <media/drm/DrmAPI.h>
+#include <media/hardware/CryptoAPI.h>
+
+extern "C" {
+ android::DrmFactory* createDrmFactory();
+ android::CryptoFactory* createCryptoFactory();
+}
+
+#endif // CLEARKEY_CREATE_PLUGIN_FACTORIES_H_
diff --git a/drm/mediadrm/plugins/clearkey/CryptoFactory.cpp b/drm/mediadrm/plugins/clearkey/CryptoFactory.cpp
new file mode 100644
index 0000000..ee3189b
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/CryptoFactory.cpp
@@ -0,0 +1,52 @@
+/*
+ * Copyright (C) 2014 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 "ClearKeyCryptoPlugin"
+#include <utils/Log.h>
+
+#include <utils/Errors.h>
+#include <utils/StrongPointer.h>
+
+#include "CryptoFactory.h"
+
+#include "ClearKeyUUID.h"
+#include "CryptoPlugin.h"
+#include "Session.h"
+#include "SessionLibrary.h"
+
+namespace clearkeydrm {
+
+bool CryptoFactory::isCryptoSchemeSupported(const uint8_t uuid[16]) const {
+ return isClearKeyUUID(uuid);
+}
+
+android::status_t CryptoFactory::createPlugin(
+ const uint8_t uuid[16],
+ const void* data, size_t size,
+ android::CryptoPlugin** plugin) {
+ if (!isCryptoSchemeSupported(uuid)) {
+ *plugin = NULL;
+ return android::BAD_VALUE;
+ }
+
+ android::sp<Session> session = SessionLibrary::get()->findSession(
+ data, size);
+ *plugin = new CryptoPlugin(session);
+ return android::OK;
+}
+
+} // namespace clearkeydrm
diff --git a/drm/mediadrm/plugins/clearkey/CryptoFactory.h b/drm/mediadrm/plugins/clearkey/CryptoFactory.h
new file mode 100644
index 0000000..568bc4b
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/CryptoFactory.h
@@ -0,0 +1,44 @@
+/*
+ * Copyright (C) 2014 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 CLEARKEY_CRYPTO_FACTORY_H_
+#define CLEARKEY_CRYPTO_FACTORY_H_
+
+#include <media/hardware/CryptoAPI.h>
+#include <media/stagefright/foundation/ABase.h>
+#include <utils/Errors.h>
+
+namespace clearkeydrm {
+
+class CryptoFactory : public android::CryptoFactory {
+public:
+ CryptoFactory() {}
+ virtual ~CryptoFactory() {}
+
+ virtual bool isCryptoSchemeSupported(const uint8_t uuid[16]) const;
+
+ virtual android::status_t createPlugin(
+ const uint8_t uuid[16],
+ const void* data, size_t size,
+ android::CryptoPlugin** plugin);
+
+private:
+ DISALLOW_EVIL_CONSTRUCTORS(CryptoFactory);
+};
+
+} // namespace clearkeydrm
+
+#endif // CLEARKEY_CRYPTO_FACTORY_H_
diff --git a/drm/mediadrm/plugins/clearkey/CryptoPlugin.cpp b/drm/mediadrm/plugins/clearkey/CryptoPlugin.cpp
new file mode 100644
index 0000000..adad136
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/CryptoPlugin.cpp
@@ -0,0 +1,83 @@
+/*
+ * Copyright (C) 2014 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 "ClearKeyCryptoPlugin"
+#include <utils/Log.h>
+
+#include <media/stagefright/MediaErrors.h>
+#include <utils/Errors.h>
+
+#include "CryptoPlugin.h"
+
+namespace clearkeydrm {
+
+using android::Vector;
+using android::AString;
+using android::status_t;
+
+// Returns negative values for error code and positive values for the size of
+// decrypted data. In theory, the output size can be larger than the input
+// size, but in practice this will never happen for AES-CTR.
+ssize_t CryptoPlugin::decrypt(bool secure, const KeyId keyId, const Iv iv,
+ Mode mode, const void* srcPtr,
+ const SubSample* subSamples, size_t numSubSamples,
+ void* dstPtr, AString* errorDetailMsg) {
+ if (secure) {
+ errorDetailMsg->setTo("Secure decryption is not supported with "
+ "ClearKey.");
+ return android::ERROR_DRM_CANNOT_HANDLE;
+ }
+
+ if (mode == kMode_Unencrypted) {
+ size_t offset = 0;
+ for (size_t i = 0; i < numSubSamples; ++i) {
+ const SubSample& subSample = subSamples[i];
+
+ if (subSample.mNumBytesOfEncryptedData != 0) {
+ errorDetailMsg->setTo(
+ "Encrypted subsamples found in allegedly unencrypted "
+ "data.");
+ return android::ERROR_DRM_DECRYPT;
+ }
+
+ if (subSample.mNumBytesOfClearData != 0) {
+ memcpy(reinterpret_cast<uint8_t*>(dstPtr) + offset,
+ reinterpret_cast<const uint8_t*>(srcPtr) + offset,
+ subSample.mNumBytesOfClearData);
+ offset += subSample.mNumBytesOfClearData;
+ }
+ }
+ return static_cast<ssize_t>(offset);
+ } else if (mode == kMode_AES_CTR) {
+ size_t bytesDecrypted;
+ status_t res = mSession->decrypt(keyId, iv, srcPtr, dstPtr, subSamples,
+ numSubSamples, &bytesDecrypted);
+ if (res == android::OK) {
+ return static_cast<ssize_t>(bytesDecrypted);
+ } else {
+ errorDetailMsg->setTo("Decryption Error");
+ return static_cast<ssize_t>(res);
+ }
+ } else {
+ errorDetailMsg->setTo(
+ "Selected encryption mode is not supported by the ClearKey DRM "
+ "Plugin.");
+ return android::ERROR_DRM_CANNOT_HANDLE;
+ }
+}
+
+} // namespace clearkeydrm
diff --git a/drm/mediadrm/plugins/clearkey/CryptoPlugin.h b/drm/mediadrm/plugins/clearkey/CryptoPlugin.h
new file mode 100644
index 0000000..002d9e0
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/CryptoPlugin.h
@@ -0,0 +1,56 @@
+/*
+ * Copyright (C) 2014 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 CLEARKEY_CRYPTO_PLUGIN_H_
+#define CLEARKEY_CRYPTO_PLUGIN_H_
+
+#include <media/hardware/CryptoAPI.h>
+#include <media/stagefright/foundation/ABase.h>
+#include <media/stagefright/foundation/AString.h>
+#include <utils/Errors.h>
+#include <utils/StrongPointer.h>
+
+#include "ClearKeyTypes.h"
+#include "Session.h"
+#include "Utils.h"
+
+namespace clearkeydrm {
+
+class CryptoPlugin : public android::CryptoPlugin {
+public:
+ CryptoPlugin(const android::sp<Session>& session) : mSession(session) {}
+ virtual ~CryptoPlugin() {}
+
+ virtual bool requiresSecureDecoderComponent(const char* mime) const {
+ UNUSED(mime);
+ return false;
+ }
+
+ virtual ssize_t decrypt(
+ bool secure, const KeyId keyId, const Iv iv,
+ Mode mode, const void* srcPtr,
+ const SubSample* subSamples, size_t numSubSamples,
+ void* dstPtr, android::AString* errorDetailMsg);
+
+private:
+ DISALLOW_EVIL_CONSTRUCTORS(CryptoPlugin);
+
+ android::sp<Session> mSession;
+};
+
+} // namespace clearkeydrm
+
+#endif // CLEARKEY_CRYPTO_PLUGIN_H_
diff --git a/drm/mediadrm/plugins/clearkey/DrmFactory.cpp b/drm/mediadrm/plugins/clearkey/DrmFactory.cpp
new file mode 100644
index 0000000..40275cf
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/DrmFactory.cpp
@@ -0,0 +1,52 @@
+/*
+ * Copyright (C) 2014 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 "ClearKeyCryptoPlugin"
+#include <utils/Log.h>
+
+#include <utils/Errors.h>
+
+#include "DrmFactory.h"
+
+#include "DrmPlugin.h"
+#include "ClearKeyUUID.h"
+#include "SessionLibrary.h"
+
+namespace clearkeydrm {
+
+bool DrmFactory::isCryptoSchemeSupported(const uint8_t uuid[16]) {
+ return isClearKeyUUID(uuid);
+}
+
+bool DrmFactory::isContentTypeSupported(const android::String8 &initDataType) {
+ // This should match the types handed by InitDataParser.
+ return initDataType == "cenc" ||
+ initDataType == "webm";
+}
+
+android::status_t DrmFactory::createDrmPlugin(
+ const uint8_t uuid[16], android::DrmPlugin** plugin) {
+ if (!isCryptoSchemeSupported(uuid)) {
+ *plugin = NULL;
+ return android::BAD_VALUE;
+ }
+
+ *plugin = new DrmPlugin(SessionLibrary::get());
+ return android::OK;
+}
+
+} // namespace clearkeydrm
diff --git a/drm/mediadrm/plugins/clearkey/DrmFactory.h b/drm/mediadrm/plugins/clearkey/DrmFactory.h
new file mode 100644
index 0000000..164d3d0
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/DrmFactory.h
@@ -0,0 +1,46 @@
+/*
+ * Copyright (C) 2014 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 CLEARKEY_DRM_FACTORY_H_
+#define CLEARKEY_DRM_FACTORY_H_
+
+#include <media/drm/DrmAPI.h>
+#include <media/stagefright/foundation/ABase.h>
+#include <utils/Errors.h>
+
+#include "Utils.h"
+
+namespace clearkeydrm {
+
+class DrmFactory : public android::DrmFactory {
+public:
+ DrmFactory() {}
+ virtual ~DrmFactory() {}
+
+ virtual bool isCryptoSchemeSupported(const uint8_t uuid[16]);
+
+ virtual bool isContentTypeSupported(const android::String8 &initDataType);
+
+ virtual android::status_t createDrmPlugin(
+ const uint8_t uuid[16], android::DrmPlugin** plugin);
+
+private:
+ DISALLOW_EVIL_CONSTRUCTORS(DrmFactory);
+};
+
+} // namespace clearkeydrm
+
+#endif // CLEARKEY_DRM_FACTORY_H_
diff --git a/drm/mediadrm/plugins/clearkey/DrmPlugin.cpp b/drm/mediadrm/plugins/clearkey/DrmPlugin.cpp
new file mode 100644
index 0000000..96fca94
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/DrmPlugin.cpp
@@ -0,0 +1,91 @@
+/*
+ * Copyright (C) 2014 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 "ClearKeyCryptoPlugin"
+#include <utils/Log.h>
+
+#include <media/stagefright/MediaErrors.h>
+#include <utils/StrongPointer.h>
+
+#include "DrmPlugin.h"
+
+#include "Session.h"
+
+namespace clearkeydrm {
+
+using android::sp;
+
+status_t DrmPlugin::openSession(Vector<uint8_t>& sessionId) {
+ sp<Session> session = mSessionLibrary->createSession();
+ sessionId = session->sessionId();
+ return android::OK;
+}
+
+status_t DrmPlugin::closeSession(const Vector<uint8_t>& sessionId) {
+ sp<Session> session = mSessionLibrary->findSession(sessionId);
+ mSessionLibrary->destroySession(session);
+ return android::OK;
+}
+
+status_t DrmPlugin::getKeyRequest(
+ const Vector<uint8_t>& scope,
+ const Vector<uint8_t>& initData,
+ const String8& initDataType,
+ KeyType keyType,
+ const KeyedVector<String8, String8>& optionalParameters,
+ Vector<uint8_t>& request,
+ String8& defaultUrl) {
+ UNUSED(optionalParameters);
+ if (keyType != kKeyType_Streaming) {
+ return android::ERROR_DRM_CANNOT_HANDLE;
+ }
+
+ sp<Session> session = mSessionLibrary->findSession(scope);
+ defaultUrl.clear();
+ return session->getKeyRequest(initData, initDataType, &request);
+}
+
+status_t DrmPlugin::provideKeyResponse(
+ const Vector<uint8_t>& scope,
+ const Vector<uint8_t>& response,
+ Vector<uint8_t>& keySetId) {
+ sp<Session> session = mSessionLibrary->findSession(scope);
+ status_t res = session->provideKeyResponse(response);
+ if (res == android::OK) {
+ keySetId.clear();
+ }
+ return res;
+}
+
+status_t DrmPlugin::getPropertyString(
+ const String8& name, String8& value) const {
+ if (name == "vendor") {
+ value = "Google";
+ } else if (name == "version") {
+ value = "1.0";
+ } else if (name == "description") {
+ value = "ClearKey CDM";
+ } else if (name == "algorithms") {
+ value = "";
+ } else {
+ ALOGE("App requested unknown string property %s", name.string());
+ return android::ERROR_DRM_CANNOT_HANDLE;
+ }
+ return android::OK;
+}
+
+} // namespace clearkeydrm
diff --git a/drm/mediadrm/plugins/clearkey/DrmPlugin.h b/drm/mediadrm/plugins/clearkey/DrmPlugin.h
new file mode 100644
index 0000000..bfbc6bf
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/DrmPlugin.h
@@ -0,0 +1,230 @@
+/*
+ * Copyright (C) 2014 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 CLEARKEY_DRM_PLUGIN_H_
+#define CLEARKEY_DRM_PLUGIN_H_
+
+#include <media/drm/DrmAPI.h>
+#include <media/stagefright/foundation/ABase.h>
+#include <media/stagefright/MediaErrors.h>
+#include <utils/Errors.h>
+#include <utils/KeyedVector.h>
+#include <utils/List.h>
+#include <utils/String8.h>
+#include <utils/Vector.h>
+
+#include "SessionLibrary.h"
+#include "Utils.h"
+
+namespace clearkeydrm {
+
+using android::KeyedVector;
+using android::List;
+using android::status_t;
+using android::String8;
+using android::Vector;
+
+class DrmPlugin : public android::DrmPlugin {
+public:
+ DrmPlugin(SessionLibrary* sessionLibrary)
+ : mSessionLibrary(sessionLibrary) {}
+ virtual ~DrmPlugin() {}
+
+ virtual status_t openSession(Vector<uint8_t>& sessionId);
+
+ virtual status_t closeSession(const Vector<uint8_t>& sessionId);
+
+ virtual status_t getKeyRequest(
+ const Vector<uint8_t>& scope,
+ const Vector<uint8_t>& initData,
+ const String8& initDataType,
+ KeyType keyType,
+ const KeyedVector<String8, String8>& optionalParameters,
+ Vector<uint8_t>& request,
+ String8& defaultUrl);
+
+ virtual status_t provideKeyResponse(
+ const Vector<uint8_t>& scope,
+ const Vector<uint8_t>& response,
+ Vector<uint8_t>& keySetId);
+
+ virtual status_t removeKeys(const Vector<uint8_t>& sessionId) {
+ UNUSED(sessionId);
+ return android::ERROR_DRM_CANNOT_HANDLE;
+ }
+
+ virtual status_t restoreKeys(
+ const Vector<uint8_t>& sessionId,
+ const Vector<uint8_t>& keySetId) {
+ UNUSED(sessionId);
+ UNUSED(keySetId);
+ return android::ERROR_DRM_CANNOT_HANDLE;
+ }
+
+ virtual status_t queryKeyStatus(
+ const Vector<uint8_t>& sessionId,
+ KeyedVector<String8, String8>& infoMap) const {
+ UNUSED(sessionId);
+ UNUSED(infoMap);
+ return android::ERROR_DRM_CANNOT_HANDLE;
+ }
+
+ virtual status_t getProvisionRequest(
+ const String8& cert_type,
+ const String8& cert_authority,
+ Vector<uint8_t>& request,
+ String8& defaultUrl) {
+ UNUSED(cert_type);
+ UNUSED(cert_authority);
+ UNUSED(request);
+ UNUSED(defaultUrl);
+ return android::ERROR_DRM_CANNOT_HANDLE;
+ }
+
+ virtual status_t provideProvisionResponse(
+ const Vector<uint8_t>& response,
+ Vector<uint8_t>& certificate,
+ Vector<uint8_t>& wrappedKey) {
+ UNUSED(response);
+ UNUSED(certificate);
+ UNUSED(wrappedKey);
+ return android::ERROR_DRM_CANNOT_HANDLE;
+ }
+
+ virtual status_t getSecureStops(List<Vector<uint8_t> >& secureStops) {
+ UNUSED(secureStops);
+ return android::ERROR_DRM_CANNOT_HANDLE;
+ }
+
+ virtual status_t releaseSecureStops(const Vector<uint8_t>& ssRelease) {
+ UNUSED(ssRelease);
+ return android::ERROR_DRM_CANNOT_HANDLE;
+ }
+
+ virtual status_t getPropertyString(
+ const String8& name, String8& value) const;
+
+ virtual status_t getPropertyByteArray(
+ const String8& name, Vector<uint8_t>& value) const {
+ UNUSED(name);
+ UNUSED(value);
+ return android::ERROR_DRM_CANNOT_HANDLE;
+ }
+
+ virtual status_t setPropertyString(
+ const String8& name, const String8& value) {
+ UNUSED(name);
+ UNUSED(value);
+ return android::ERROR_DRM_CANNOT_HANDLE;
+ }
+
+ virtual status_t setPropertyByteArray(
+ const String8& name, const Vector<uint8_t>& value) {
+ UNUSED(name);
+ UNUSED(value);
+ return android::ERROR_DRM_CANNOT_HANDLE;
+ }
+
+ virtual status_t setCipherAlgorithm(
+ const Vector<uint8_t>& sessionId, const String8& algorithm) {
+ UNUSED(sessionId);
+ UNUSED(algorithm);
+ return android::ERROR_DRM_CANNOT_HANDLE;
+ }
+
+ virtual status_t setMacAlgorithm(
+ const Vector<uint8_t>& sessionId, const String8& algorithm) {
+ UNUSED(sessionId);
+ UNUSED(algorithm);
+ return android::ERROR_DRM_CANNOT_HANDLE;
+ }
+
+ virtual status_t encrypt(
+ const Vector<uint8_t>& sessionId,
+ const Vector<uint8_t>& keyId,
+ const Vector<uint8_t>& input,
+ const Vector<uint8_t>& iv,
+ Vector<uint8_t>& output) {
+ UNUSED(sessionId);
+ UNUSED(keyId);
+ UNUSED(input);
+ UNUSED(iv);
+ UNUSED(output);
+ return android::ERROR_DRM_CANNOT_HANDLE;
+ }
+
+ virtual status_t decrypt(
+ const Vector<uint8_t>& sessionId,
+ const Vector<uint8_t>& keyId,
+ const Vector<uint8_t>& input,
+ const Vector<uint8_t>& iv,
+ Vector<uint8_t>& output) {
+ UNUSED(sessionId);
+ UNUSED(keyId);
+ UNUSED(input);
+ UNUSED(iv);
+ UNUSED(output);
+ return android::ERROR_DRM_CANNOT_HANDLE;
+ }
+
+ virtual status_t sign(
+ const Vector<uint8_t>& sessionId,
+ const Vector<uint8_t>& keyId,
+ const Vector<uint8_t>& message,
+ Vector<uint8_t>& signature) {
+ UNUSED(sessionId);
+ UNUSED(keyId);
+ UNUSED(message);
+ UNUSED(signature);
+ return android::ERROR_DRM_CANNOT_HANDLE;
+ }
+
+ virtual status_t verify(
+ const Vector<uint8_t>& sessionId,
+ const Vector<uint8_t>& keyId,
+ const Vector<uint8_t>& message,
+ const Vector<uint8_t>& signature, bool& match) {
+ UNUSED(sessionId);
+ UNUSED(keyId);
+ UNUSED(message);
+ UNUSED(signature);
+ UNUSED(match);
+ return android::ERROR_DRM_CANNOT_HANDLE;
+ }
+
+ virtual status_t signRSA(
+ const Vector<uint8_t>& sessionId,
+ const String8& algorithm,
+ const Vector<uint8_t>& message,
+ const Vector<uint8_t>& wrappedKey,
+ Vector<uint8_t>& signature) {
+ UNUSED(sessionId);
+ UNUSED(algorithm);
+ UNUSED(message);
+ UNUSED(wrappedKey);
+ UNUSED(signature);
+ return android::ERROR_DRM_CANNOT_HANDLE;
+ }
+
+private:
+ DISALLOW_EVIL_CONSTRUCTORS(DrmPlugin);
+
+ SessionLibrary* mSessionLibrary;
+};
+
+} // namespace clearkeydrm
+
+#endif // CLEARKEY_DRM_PLUGIN_H_
diff --git a/drm/mediadrm/plugins/clearkey/InitDataParser.cpp b/drm/mediadrm/plugins/clearkey/InitDataParser.cpp
new file mode 100644
index 0000000..c22d73a
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/InitDataParser.cpp
@@ -0,0 +1,147 @@
+/*
+ * Copyright (C) 2014 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 "ClearKeyCryptoPlugin"
+#include <utils/Log.h>
+
+#include <endian.h>
+#include <media/stagefright/foundation/AString.h>
+#include <media/stagefright/foundation/base64.h>
+#include <media/stagefright/MediaErrors.h>
+#include <string.h>
+
+#include "InitDataParser.h"
+
+#include "ClearKeyUUID.h"
+#include "Utils.h"
+
+namespace clearkeydrm {
+
+using android::AString;
+using android::String8;
+using android::Vector;
+
+namespace {
+ const size_t kKeyIdSize = 16;
+ const size_t kSystemIdSize = 16;
+}
+
+android::status_t InitDataParser::parse(const Vector<uint8_t>& initData,
+ const String8& initDataType,
+ Vector<uint8_t>* licenseRequest) {
+ // Build a list of the key IDs
+ Vector<const uint8_t*> keyIds;
+ if (initDataType == "cenc") {
+ android::status_t res = parsePssh(initData, &keyIds);
+ if (res != android::OK) {
+ return res;
+ }
+ } else if (initDataType == "webm") {
+ // WebM "init data" is just a single key ID
+ if (initData.size() != kKeyIdSize) {
+ return android::ERROR_DRM_CANNOT_HANDLE;
+ }
+ keyIds.push(initData.array());
+ } else {
+ return android::ERROR_DRM_CANNOT_HANDLE;
+ }
+
+ // Build the request
+ String8 requestJson = generateRequest(keyIds);
+ licenseRequest->clear();
+ licenseRequest->appendArray(
+ reinterpret_cast<const uint8_t*>(requestJson.string()),
+ requestJson.size());
+ return android::OK;
+}
+
+android::status_t InitDataParser::parsePssh(const Vector<uint8_t>& initData,
+ Vector<const uint8_t*>* keyIds) {
+ size_t readPosition = 0;
+
+ // Validate size field
+ uint32_t expectedSize = initData.size();
+ expectedSize = htonl(expectedSize);
+ if (memcmp(&initData[readPosition], &expectedSize,
+ sizeof(expectedSize)) != 0) {
+ return android::ERROR_DRM_CANNOT_HANDLE;
+ }
+ readPosition += sizeof(expectedSize);
+
+ // Validate PSSH box identifier
+ const char psshIdentifier[4] = {'p', 's', 's', 'h'};
+ if (memcmp(&initData[readPosition], psshIdentifier,
+ sizeof(psshIdentifier)) != 0) {
+ return android::ERROR_DRM_CANNOT_HANDLE;
+ }
+ readPosition += sizeof(psshIdentifier);
+
+ // Validate EME version number
+ const uint8_t psshVersion1[4] = {1, 0, 0, 0};
+ if (memcmp(&initData[readPosition], psshVersion1,
+ sizeof(psshVersion1)) != 0) {
+ return android::ERROR_DRM_CANNOT_HANDLE;
+ }
+ readPosition += sizeof(psshVersion1);
+
+ // Validate system ID
+ if (!isClearKeyUUID(&initData[readPosition])) {
+ return android::ERROR_DRM_CANNOT_HANDLE;
+ }
+ readPosition += kSystemIdSize;
+
+ // Read key ID count
+ uint32_t keyIdCount;
+ memcpy(&keyIdCount, &initData[readPosition], sizeof(keyIdCount));
+ keyIdCount = ntohl(keyIdCount);
+ readPosition += sizeof(keyIdCount);
+ if (readPosition + (keyIdCount * kKeyIdSize) !=
+ initData.size() - sizeof(uint32_t)) {
+ return android::ERROR_DRM_CANNOT_HANDLE;
+ }
+
+ // Calculate the key ID offsets
+ for (uint32_t i = 0; i < keyIdCount; ++i) {
+ size_t keyIdPosition = readPosition + (i * kKeyIdSize);
+ keyIds->push(&initData[keyIdPosition]);
+ }
+ return android::OK;
+}
+
+String8 InitDataParser::generateRequest(const Vector<const uint8_t*>& keyIds) {
+ const String8 kRequestPrefix("{\"kids\":[");
+ const String8 kRequestSuffix("],\"type\":\"temporary\"}");
+ const String8 kBase64Padding("=");
+
+ String8 request(kRequestPrefix);
+ AString encodedId;
+ for (size_t i = 0; i < keyIds.size(); ++i) {
+ encodedId.clear();
+ android::encodeBase64(keyIds[i], kKeyIdSize, &encodedId);
+ if (i != 0) {
+ request.append(",");
+ }
+ request.appendFormat("\"%s\"", encodedId.c_str());
+ }
+ request.append(kRequestSuffix);
+
+ // Android's Base64 encoder produces padding. EME forbids padding.
+ request.removeAll(kBase64Padding);
+ return request;
+}
+
+} // namespace clearkeydrm
diff --git a/drm/mediadrm/plugins/clearkey/InitDataParser.h b/drm/mediadrm/plugins/clearkey/InitDataParser.h
new file mode 100644
index 0000000..9505d2a
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/InitDataParser.h
@@ -0,0 +1,47 @@
+/*
+ * Copyright (C) 2014 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 CLEARKEY_INIT_DATA_PARSER_H_
+#define CLEARKEY_INIT_DATA_PARSER_H_
+
+#include <media/stagefright/foundation/ABase.h>
+#include <utils/Errors.h>
+#include <utils/String8.h>
+#include <utils/Vector.h>
+
+namespace clearkeydrm {
+
+class InitDataParser {
+public:
+ InitDataParser() {}
+
+ android::status_t parse(const android::Vector<uint8_t>& initData,
+ const android::String8& initDataType,
+ android::Vector<uint8_t>* licenseRequest);
+
+private:
+ DISALLOW_EVIL_CONSTRUCTORS(InitDataParser);
+
+ android::status_t parsePssh(const android::Vector<uint8_t>& initData,
+ android::Vector<const uint8_t*>* keyIds);
+
+ android::String8 generateRequest(
+ const android::Vector<const uint8_t*>& keyIds);
+};
+
+} // namespace clearkeydrm
+
+#endif // CLEARKEY_INIT_DATA_PARSER_H_
diff --git a/drm/mediadrm/plugins/clearkey/JsonWebKey.cpp b/drm/mediadrm/plugins/clearkey/JsonWebKey.cpp
new file mode 100644
index 0000000..53ffae4
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/JsonWebKey.cpp
@@ -0,0 +1,269 @@
+/*
+ * Copyright (C) 2014 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 "JsonWebKey"
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/AString.h>
+#include <media/stagefright/foundation/base64.h>
+#include <utils/Log.h>
+
+#include "JsonWebKey.h"
+
+namespace {
+const android::String8 kKeysTag("keys");
+const android::String8 kKeyTypeTag("kty");
+const android::String8 kSymmetricKeyValue("oct");
+const android::String8 kKeyTag("k");
+const android::String8 kKeyIdTag("kid");
+const android::String8 kBase64Padding("=");
+}
+
+namespace clearkeydrm {
+
+using android::ABuffer;
+using android::AString;
+
+JsonWebKey::JsonWebKey() {
+}
+
+JsonWebKey::~JsonWebKey() {
+}
+
+/*
+ * Parses a JSON Web Key Set string, initializes a KeyMap with key id:key
+ * pairs from the JSON Web Key Set. Both key ids and keys are base64url
+ * encoded. The KeyMap contains base64url decoded key id:key pairs.
+ *
+ * @return Returns false for errors, true for success.
+ */
+bool JsonWebKey::extractKeysFromJsonWebKeySet(const String8& jsonWebKeySet,
+ KeyMap* keys) {
+
+ keys->clear();
+ if (!parseJsonWebKeySet(jsonWebKeySet, &mJsonObjects)) {
+ return false;
+ }
+
+ // mJsonObjects[0] contains the entire JSON Web Key Set, including
+ // all the base64 encoded keys. Each key is also stored separately as
+ // a JSON object in mJsonObjects[1..n] where n is the total
+ // number of keys in the set.
+ if (!isJsonWebKeySet(mJsonObjects[0])) {
+ return false;
+ }
+
+ String8 encodedKey, encodedKeyId;
+ Vector<uint8_t> decodedKey, decodedKeyId;
+
+ // mJsonObjects[1] contains the first JSON Web Key in the set
+ for (size_t i = 1; i < mJsonObjects.size(); ++i) {
+ encodedKeyId.clear();
+ encodedKey.clear();
+
+ if (!parseJsonObject(mJsonObjects[i], &mTokens))
+ return false;
+
+ if (findKey(mJsonObjects[i], &encodedKeyId, &encodedKey)) {
+ if (encodedKeyId.isEmpty() || encodedKey.isEmpty()) {
+ ALOGE("Must have both key id and key in the JsonWebKey set.");
+ continue;
+ }
+
+ if (!decodeBase64String(encodedKeyId, &decodedKeyId)) {
+ ALOGE("Failed to decode key id(%s)", encodedKeyId.string());
+ continue;
+ }
+
+ if (!decodeBase64String(encodedKey, &decodedKey)) {
+ ALOGE("Failed to decode key(%s)", encodedKey.string());
+ continue;
+ }
+
+ keys->add(decodedKeyId, decodedKey);
+ }
+ }
+ return true;
+}
+
+bool JsonWebKey::decodeBase64String(const String8& encodedText,
+ Vector<uint8_t>* decodedText) {
+
+ decodedText->clear();
+
+ // encodedText should not contain padding characters as per EME spec.
+ if (encodedText.find(kBase64Padding) != -1) {
+ return false;
+ }
+
+ // Since android::decodeBase64() requires padding characters,
+ // add them so length of encodedText is exactly a multiple of 4.
+ int remainder = encodedText.length() % 4;
+ String8 paddedText(encodedText);
+ if (remainder > 0) {
+ for (int i = 0; i < 4 - remainder; ++i) {
+ paddedText.append(kBase64Padding);
+ }
+ }
+
+ android::sp<ABuffer> buffer =
+ android::decodeBase64(AString(paddedText.string()));
+ if (buffer == NULL) {
+ ALOGE("Malformed base64 encoded content found.");
+ return false;
+ }
+
+ decodedText->appendArray(buffer->base(), buffer->size());
+ return true;
+}
+
+bool JsonWebKey::findKey(const String8& jsonObject, String8* keyId,
+ String8* encodedKey) {
+
+ String8 key, value;
+
+ // Only allow symmetric key, i.e. "kty":"oct" pair.
+ if (jsonObject.find(kKeyTypeTag) >= 0) {
+ findValue(kKeyTypeTag, &value);
+ if (0 != value.compare(kSymmetricKeyValue))
+ return false;
+ }
+
+ if (jsonObject.find(kKeyIdTag) >= 0) {
+ findValue(kKeyIdTag, keyId);
+ }
+
+ if (jsonObject.find(kKeyTag) >= 0) {
+ findValue(kKeyTag, encodedKey);
+ }
+ return true;
+}
+
+void JsonWebKey::findValue(const String8 &key, String8* value) {
+ value->clear();
+ const char* valueToken;
+ for (Vector<String8>::const_iterator nextToken = mTokens.begin();
+ nextToken != mTokens.end(); ++nextToken) {
+ if (0 == (*nextToken).compare(key)) {
+ if (nextToken + 1 == mTokens.end())
+ break;
+ valueToken = (*(nextToken + 1)).string();
+ value->setTo(valueToken);
+ nextToken++;
+ break;
+ }
+ }
+}
+
+bool JsonWebKey::isJsonWebKeySet(const String8& jsonObject) const {
+ if (jsonObject.find(kKeysTag) == -1) {
+ ALOGE("JSON Web Key does not contain keys.");
+ return false;
+ }
+ return true;
+}
+
+/*
+ * Parses a JSON objects string and initializes a vector of tokens.
+ *
+ * @return Returns false for errors, true for success.
+ */
+bool JsonWebKey::parseJsonObject(const String8& jsonObject,
+ Vector<String8>* tokens) {
+ jsmn_parser parser;
+
+ jsmn_init(&parser);
+ int numTokens = jsmn_parse(&parser,
+ jsonObject.string(), jsonObject.size(), NULL, 0);
+ if (numTokens < 0) {
+ ALOGE("Parser returns error code=%d", numTokens);
+ return false;
+ }
+
+ unsigned int jsmnTokensSize = numTokens * sizeof(jsmntok_t);
+ mJsmnTokens.clear();
+ mJsmnTokens.setCapacity(jsmnTokensSize);
+
+ jsmn_init(&parser);
+ int status = jsmn_parse(&parser, jsonObject.string(),
+ jsonObject.size(), mJsmnTokens.editArray(), numTokens);
+ if (status < 0) {
+ ALOGE("Parser returns error code=%d", status);
+ return false;
+ }
+
+ tokens->clear();
+ String8 token;
+ const char *pjs;
+ for (int j = 0; j < numTokens; ++j) {
+ pjs = jsonObject.string() + mJsmnTokens[j].start;
+ if (mJsmnTokens[j].type == JSMN_STRING ||
+ mJsmnTokens[j].type == JSMN_PRIMITIVE) {
+ token.setTo(pjs, mJsmnTokens[j].end - mJsmnTokens[j].start);
+ tokens->add(token);
+ }
+ }
+ return true;
+}
+
+/*
+ * Parses JSON Web Key Set string and initializes a vector of JSON objects.
+ *
+ * @return Returns false for errors, true for success.
+ */
+bool JsonWebKey::parseJsonWebKeySet(const String8& jsonWebKeySet,
+ Vector<String8>* jsonObjects) {
+ if (jsonWebKeySet.isEmpty()) {
+ ALOGE("Empty JSON Web Key");
+ return false;
+ }
+
+ // The jsmn parser only supports unicode encoding.
+ jsmn_parser parser;
+
+ // Computes number of tokens. A token marks the type, offset in
+ // the original string.
+ jsmn_init(&parser);
+ int numTokens = jsmn_parse(&parser,
+ jsonWebKeySet.string(), jsonWebKeySet.size(), NULL, 0);
+ if (numTokens < 0) {
+ ALOGE("Parser returns error code=%d", numTokens);
+ return false;
+ }
+
+ unsigned int jsmnTokensSize = numTokens * sizeof(jsmntok_t);
+ mJsmnTokens.setCapacity(jsmnTokensSize);
+
+ jsmn_init(&parser);
+ int status = jsmn_parse(&parser, jsonWebKeySet.string(),
+ jsonWebKeySet.size(), mJsmnTokens.editArray(), numTokens);
+ if (status < 0) {
+ ALOGE("Parser returns error code=%d", status);
+ return false;
+ }
+
+ String8 token;
+ const char *pjs;
+ for (int i = 0; i < numTokens; ++i) {
+ pjs = jsonWebKeySet.string() + mJsmnTokens[i].start;
+ if (mJsmnTokens[i].type == JSMN_OBJECT) {
+ token.setTo(pjs, mJsmnTokens[i].end - mJsmnTokens[i].start);
+ jsonObjects->add(token);
+ }
+ }
+ return true;
+}
+
+} // clearkeydrm
diff --git a/drm/mediadrm/plugins/clearkey/JsonWebKey.h b/drm/mediadrm/plugins/clearkey/JsonWebKey.h
new file mode 100644
index 0000000..6ae50ee
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/JsonWebKey.h
@@ -0,0 +1,62 @@
+/*
+ * Copyright (C) 2014 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 CLEARKEY_JSON_WEB_KEY_H_
+#define CLEARKEY_JSON_WEB_KEY_H_
+
+#include <media/stagefright/foundation/ABase.h>
+#include <utils/KeyedVector.h>
+#include <utils/String8.h>
+#include <utils/StrongPointer.h>
+
+#include "jsmn.h"
+#include "Utils.h"
+#include "ClearKeyTypes.h"
+
+namespace clearkeydrm {
+
+using android::KeyedVector;
+using android::sp;
+using android::String8;
+using android::Vector;
+
+class JsonWebKey {
+ public:
+ JsonWebKey();
+ virtual ~JsonWebKey();
+
+ bool extractKeysFromJsonWebKeySet(const String8& jsonWebKeySet,
+ KeyMap* keys);
+
+ private:
+ Vector<jsmntok_t> mJsmnTokens;
+ Vector<String8> mJsonObjects;
+ Vector<String8> mTokens;
+
+ bool decodeBase64String(const String8& encodedText,
+ Vector<uint8_t>* decodedText);
+ bool findKey(const String8& jsonObject, String8* keyId,
+ String8* encodedKey);
+ void findValue(const String8 &key, String8* value);
+ bool isJsonWebKeySet(const String8& jsonObject) const;
+ bool parseJsonObject(const String8& jsonObject, Vector<String8>* tokens);
+ bool parseJsonWebKeySet(const String8& jsonWebKeySet, Vector<String8>* jsonObjects);
+
+ DISALLOW_EVIL_CONSTRUCTORS(JsonWebKey);
+};
+
+} // namespace clearkeydrm
+
+#endif // CLEARKEY_JSON_WEB_KEY_H_
diff --git a/drm/mediadrm/plugins/clearkey/Session.cpp b/drm/mediadrm/plugins/clearkey/Session.cpp
new file mode 100644
index 0000000..95016f5
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/Session.cpp
@@ -0,0 +1,85 @@
+/*
+ * Copyright (C) 2014 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 "ClearKeyCryptoPlugin"
+#include <utils/Log.h>
+
+#include <media/stagefright/MediaErrors.h>
+#include <utils/String8.h>
+
+#include "Session.h"
+
+#include "AesCtrDecryptor.h"
+#include "InitDataParser.h"
+#include "JsonWebKey.h"
+
+namespace clearkeydrm {
+
+using android::Mutex;
+using android::String8;
+using android::Vector;
+using android::status_t;
+
+status_t Session::getKeyRequest(
+ const Vector<uint8_t>& initData,
+ const String8& initDataType,
+ Vector<uint8_t>* keyRequest) const {
+ InitDataParser parser;
+ return parser.parse(initData, initDataType, keyRequest);
+}
+
+status_t Session::provideKeyResponse(const Vector<uint8_t>& response) {
+ String8 responseString(
+ reinterpret_cast<const char*>(response.array()), response.size());
+ KeyMap keys;
+
+ Mutex::Autolock lock(mMapLock);
+ JsonWebKey parser;
+ if (parser.extractKeysFromJsonWebKeySet(responseString, &keys)) {
+ for (size_t i = 0; i < keys.size(); ++i) {
+ const KeyMap::key_type& keyId = keys.keyAt(i);
+ const KeyMap::value_type& key = keys.valueAt(i);
+ mKeyMap.add(keyId, key);
+ }
+ return android::OK;
+ } else {
+ return android::ERROR_DRM_UNKNOWN;
+ }
+}
+
+status_t Session::decrypt(
+ const KeyId keyId, const Iv iv, const void* source,
+ void* destination, const SubSample* subSamples,
+ size_t numSubSamples, size_t* bytesDecryptedOut) {
+ Mutex::Autolock lock(mMapLock);
+
+ Vector<uint8_t> keyIdVector;
+ keyIdVector.appendArray(keyId, kBlockSize);
+ if (mKeyMap.indexOfKey(keyIdVector) < 0) {
+ return android::ERROR_DRM_NO_LICENSE;
+ }
+
+ const Vector<uint8_t>& key = mKeyMap.valueFor(keyIdVector);
+ AesCtrDecryptor decryptor;
+ return decryptor.decrypt(
+ key, iv,
+ reinterpret_cast<const uint8_t*>(source),
+ reinterpret_cast<uint8_t*>(destination), subSamples,
+ numSubSamples, bytesDecryptedOut);
+}
+
+} // namespace clearkeydrm
diff --git a/drm/mediadrm/plugins/clearkey/Session.h b/drm/mediadrm/plugins/clearkey/Session.h
new file mode 100644
index 0000000..cab0dc3
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/Session.h
@@ -0,0 +1,64 @@
+/*
+ * Copyright (C) 2014 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 CLEARKEY_SESSION_H_
+#define CLEARKEY_SESSION_H_
+
+#include <media/stagefright/foundation/ABase.h>
+#include <utils/Errors.h>
+#include <utils/Mutex.h>
+#include <utils/RefBase.h>
+#include <utils/String8.h>
+#include <utils/Vector.h>
+
+#include "ClearKeyTypes.h"
+#include "Utils.h"
+
+namespace clearkeydrm {
+
+class Session : public android::RefBase {
+public:
+ explicit Session(const android::Vector<uint8_t>& sessionId)
+ : mSessionId(sessionId) {}
+ virtual ~Session() {}
+
+ const android::Vector<uint8_t>& sessionId() const { return mSessionId; }
+
+ android::status_t getKeyRequest(
+ const android::Vector<uint8_t>& initData,
+ const android::String8& initDataType,
+ android::Vector<uint8_t>* keyRequest) const;
+
+ android::status_t provideKeyResponse(
+ const android::Vector<uint8_t>& response);
+
+ android::status_t decrypt(
+ const KeyId keyId, const Iv iv, const void* source,
+ void* destination, const SubSample* subSamples,
+ size_t numSubSamples, size_t* bytesDecryptedOut);
+
+private:
+ DISALLOW_EVIL_CONSTRUCTORS(Session);
+
+ const android::Vector<uint8_t> mSessionId;
+
+ android::Mutex mMapLock;
+ KeyMap mKeyMap;
+};
+
+} // namespace clearkeydrm
+
+#endif // CLEARKEY_SESSION_H_
diff --git a/drm/mediadrm/plugins/clearkey/SessionLibrary.cpp b/drm/mediadrm/plugins/clearkey/SessionLibrary.cpp
new file mode 100644
index 0000000..d047c53
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/SessionLibrary.cpp
@@ -0,0 +1,78 @@
+/*
+ * Copyright (C) 2014 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 "ClearKeyCryptoPlugin"
+#include <utils/Log.h>
+
+#include <utils/String8.h>
+
+#include "SessionLibrary.h"
+
+namespace clearkeydrm {
+
+using android::Mutex;
+using android::sp;
+using android::String8;
+using android::Vector;
+
+Mutex SessionLibrary::sSingletonLock;
+SessionLibrary* SessionLibrary::sSingleton = NULL;
+
+SessionLibrary* SessionLibrary::get() {
+ Mutex::Autolock lock(sSingletonLock);
+
+ if (sSingleton == NULL) {
+ ALOGD("Instantiating Session Library Singleton.");
+ sSingleton = new SessionLibrary();
+ }
+
+ return sSingleton;
+}
+
+const sp<Session>& SessionLibrary::createSession() {
+ Mutex::Autolock lock(mSessionsLock);
+
+ String8 sessionIdString = String8::format("%u", mNextSessionId);
+ mNextSessionId += 1;
+ Vector<uint8_t> sessionId;
+ sessionId.appendArray(
+ reinterpret_cast<const uint8_t*>(sessionIdString.string()),
+ sessionIdString.size());
+
+ mSessions.add(sessionId, new Session(sessionId));
+ return mSessions.valueFor(sessionId);
+}
+
+const sp<Session>& SessionLibrary::findSession(
+ const Vector<uint8_t>& sessionId) {
+ Mutex::Autolock lock(mSessionsLock);
+ return mSessions.valueFor(sessionId);
+}
+
+const sp<Session>& SessionLibrary::findSession(
+ const void* data, size_t size) {
+ Vector<uint8_t> sessionId;
+ sessionId.appendArray(reinterpret_cast<const uint8_t*>(data), size);
+ return findSession(sessionId);
+}
+
+void SessionLibrary::destroySession(const sp<Session>& session) {
+ Mutex::Autolock lock(mSessionsLock);\
+ mSessions.removeItem(session->sessionId());
+}
+
+} // namespace clearkeydrm
diff --git a/drm/mediadrm/plugins/clearkey/SessionLibrary.h b/drm/mediadrm/plugins/clearkey/SessionLibrary.h
new file mode 100644
index 0000000..56c8828
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/SessionLibrary.h
@@ -0,0 +1,59 @@
+/*
+ * Copyright (C) 2014 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 CLEARKEY_SESSION_LIBRARY_H_
+#define CLEARKEY_SESSION_LIBRARY_H_
+
+#include <utils/KeyedVector.h>
+#include <utils/Mutex.h>
+#include <utils/StrongPointer.h>
+#include <utils/Vector.h>
+
+#include "Session.h"
+#include "Utils.h"
+
+namespace clearkeydrm {
+
+class SessionLibrary {
+public:
+ static SessionLibrary* get();
+
+ const android::sp<Session>& createSession();
+
+ const android::sp<Session>& findSession(
+ const android::Vector<uint8_t>& sessionId);
+
+ const android::sp<Session>& findSession(const void* data, size_t size);
+
+ void destroySession(const android::sp<Session>& session);
+
+private:
+ DISALLOW_EVIL_CONSTRUCTORS(SessionLibrary);
+
+ SessionLibrary() : mNextSessionId(1) {}
+
+ static android::Mutex sSingletonLock;
+ static SessionLibrary* sSingleton;
+
+ android::Mutex mSessionsLock;
+ uint32_t mNextSessionId;
+ android::KeyedVector<android::Vector<uint8_t>, android::sp<Session> >
+ mSessions;
+};
+
+} // namespace clearkeydrm
+
+#endif // CLEARKEY_SESSION_LIBRARY_H_
diff --git a/drm/mediadrm/plugins/clearkey/Utils.cpp b/drm/mediadrm/plugins/clearkey/Utils.cpp
new file mode 100644
index 0000000..93c643b
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/Utils.cpp
@@ -0,0 +1,30 @@
+/*
+ * Copyright (C) 2014 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 "Utils.h"
+
+namespace android {
+
+bool operator<(const Vector<uint8_t> &lhs, const Vector<uint8_t> &rhs) {
+ if (lhs.size() < rhs.size()) {
+ return true;
+ } else if (lhs.size() > rhs.size()) {
+ return false;
+ }
+ return memcmp((void *)lhs.array(), (void *)rhs.array(), rhs.size()) < 0;
+}
+
+} // namespace android
diff --git a/drm/mediadrm/plugins/clearkey/Utils.h b/drm/mediadrm/plugins/clearkey/Utils.h
new file mode 100644
index 0000000..2543124
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/Utils.h
@@ -0,0 +1,32 @@
+/*
+ * Copyright (C) 2014 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 CLEARKEY_UTILS_H_
+#define CLEARKEY_UTILS_H_
+
+#include <utils/Vector.h>
+
+// Add a comparison operator for this Vector specialization so that it can be
+// used as a key in a KeyedVector.
+namespace android {
+
+bool operator<(const Vector<uint8_t> &lhs, const Vector<uint8_t> &rhs);
+
+} // namespace android
+
+#define UNUSED(x) (void)(x);
+
+#endif // CLEARKEY_UTILS_H_
diff --git a/drm/mediadrm/plugins/clearkey/tests/AesCtrDecryptorUnittest.cpp b/drm/mediadrm/plugins/clearkey/tests/AesCtrDecryptorUnittest.cpp
new file mode 100644
index 0000000..039e402
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/tests/AesCtrDecryptorUnittest.cpp
@@ -0,0 +1,421 @@
+/*
+ * Copyright (C) 2014 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 <gtest/gtest.h>
+#include <string.h>
+
+#include <utils/String8.h>
+#include <utils/Vector.h>
+
+#include "AesCtrDecryptor.h"
+
+namespace clearkeydrm {
+
+using namespace android;
+
+class AesCtrDecryptorTest : public ::testing::Test {
+ protected:
+ typedef uint8_t Key[kBlockSize];
+
+ status_t attemptDecrypt(const Key& key, const Iv& iv, const uint8_t* source,
+ uint8_t* destination, const SubSample* subSamples,
+ size_t numSubSamples, size_t* bytesDecryptedOut) {
+ Vector<uint8_t> keyVector;
+ keyVector.appendArray(key, kBlockSize);
+
+ AesCtrDecryptor decryptor;
+ return decryptor.decrypt(keyVector, iv, source, destination, subSamples,
+ numSubSamples, bytesDecryptedOut);
+ }
+
+ template <size_t totalSize>
+ void attemptDecryptExpectingSuccess(const Key& key, const Iv& iv,
+ const uint8_t* encrypted,
+ const uint8_t* decrypted,
+ const SubSample* subSamples,
+ size_t numSubSamples) {
+ uint8_t outputBuffer[totalSize] = {};
+ size_t bytesDecrypted = 0;
+ ASSERT_EQ(android::OK, attemptDecrypt(key, iv, encrypted, outputBuffer,
+ subSamples, numSubSamples,
+ &bytesDecrypted));
+ EXPECT_EQ(totalSize, bytesDecrypted);
+ EXPECT_EQ(0, memcmp(outputBuffer, decrypted, totalSize));
+ }
+};
+
+TEST_F(AesCtrDecryptorTest, DecryptsContiguousEncryptedBlock) {
+ const size_t kTotalSize = 64;
+ const size_t kNumSubsamples = 1;
+
+ // Test vectors from NIST-800-38A
+ Key key = {
+ 0x2b, 0x7e, 0x15, 0x16, 0x28, 0xae, 0xd2, 0xa6,
+ 0xab, 0xf7, 0x15, 0x88, 0x09, 0xcf, 0x4f, 0x3c
+ };
+
+ Iv iv = {
+ 0xf0, 0xf1, 0xf2, 0xf3, 0xf4, 0xf5, 0xf6, 0xf7,
+ 0xf8, 0xf9, 0xfa, 0xfb, 0xfc, 0xfd, 0xfe, 0xff
+ };
+
+ uint8_t encrypted[kTotalSize] = {
+ 0x87, 0x4d, 0x61, 0x91, 0xb6, 0x20, 0xe3, 0x26,
+ 0x1b, 0xef, 0x68, 0x64, 0x99, 0x0d, 0xb6, 0xce,
+ 0x98, 0x06, 0xf6, 0x6b, 0x79, 0x70, 0xfd, 0xff,
+ 0x86, 0x17, 0x18, 0x7b, 0xb9, 0xff, 0xfd, 0xff,
+ 0x5a, 0xe4, 0xdf, 0x3e, 0xdb, 0xd5, 0xd3, 0x5e,
+ 0x5b, 0x4f, 0x09, 0x02, 0x0d, 0xb0, 0x3e, 0xab,
+ 0x1e, 0x03, 0x1d, 0xda, 0x2f, 0xbe, 0x03, 0xd1,
+ 0x79, 0x21, 0x70, 0xa0, 0xf3, 0x00, 0x9c, 0xee
+ };
+
+ uint8_t decrypted[kTotalSize] = {
+ 0x6b, 0xc1, 0xbe, 0xe2, 0x2e, 0x40, 0x9f, 0x96,
+ 0xe9, 0x3d, 0x7e, 0x11, 0x73, 0x93, 0x17, 0x2a,
+ 0xae, 0x2d, 0x8a, 0x57, 0x1e, 0x03, 0xac, 0x9c,
+ 0x9e, 0xb7, 0x6f, 0xac, 0x45, 0xaf, 0x8e, 0x51,
+ 0x30, 0xc8, 0x1c, 0x46, 0xa3, 0x5c, 0xe4, 0x11,
+ 0xe5, 0xfb, 0xc1, 0x19, 0x1a, 0x0a, 0x52, 0xef,
+ 0xf6, 0x9f, 0x24, 0x45, 0xdf, 0x4f, 0x9b, 0x17,
+ 0xad, 0x2b, 0x41, 0x7b, 0xe6, 0x6c, 0x37, 0x10
+ };
+
+ SubSample subSamples[kNumSubsamples] = {
+ {0, 64}
+ };
+
+ attemptDecryptExpectingSuccess<kTotalSize>(key, iv, encrypted, decrypted,
+ subSamples, kNumSubsamples);
+}
+
+TEST_F(AesCtrDecryptorTest, DecryptsAlignedBifurcatedEncryptedBlock) {
+ const size_t kTotalSize = 64;
+ const size_t kNumSubsamples = 2;
+
+ // Test vectors from NIST-800-38A
+ Key key = {
+ 0x2b, 0x7e, 0x15, 0x16, 0x28, 0xae, 0xd2, 0xa6,
+ 0xab, 0xf7, 0x15, 0x88, 0x09, 0xcf, 0x4f, 0x3c
+ };
+
+ Iv iv = {
+ 0xf0, 0xf1, 0xf2, 0xf3, 0xf4, 0xf5, 0xf6, 0xf7,
+ 0xf8, 0xf9, 0xfa, 0xfb, 0xfc, 0xfd, 0xfe, 0xff
+ };
+
+ uint8_t encrypted[kTotalSize] = {
+ 0x87, 0x4d, 0x61, 0x91, 0xb6, 0x20, 0xe3, 0x26,
+ 0x1b, 0xef, 0x68, 0x64, 0x99, 0x0d, 0xb6, 0xce,
+ 0x98, 0x06, 0xf6, 0x6b, 0x79, 0x70, 0xfd, 0xff,
+ 0x86, 0x17, 0x18, 0x7b, 0xb9, 0xff, 0xfd, 0xff,
+ 0x5a, 0xe4, 0xdf, 0x3e, 0xdb, 0xd5, 0xd3, 0x5e,
+ 0x5b, 0x4f, 0x09, 0x02, 0x0d, 0xb0, 0x3e, 0xab,
+ 0x1e, 0x03, 0x1d, 0xda, 0x2f, 0xbe, 0x03, 0xd1,
+ 0x79, 0x21, 0x70, 0xa0, 0xf3, 0x00, 0x9c, 0xee
+ };
+
+ uint8_t decrypted[kTotalSize] = {
+ 0x6b, 0xc1, 0xbe, 0xe2, 0x2e, 0x40, 0x9f, 0x96,
+ 0xe9, 0x3d, 0x7e, 0x11, 0x73, 0x93, 0x17, 0x2a,
+ 0xae, 0x2d, 0x8a, 0x57, 0x1e, 0x03, 0xac, 0x9c,
+ 0x9e, 0xb7, 0x6f, 0xac, 0x45, 0xaf, 0x8e, 0x51,
+ 0x30, 0xc8, 0x1c, 0x46, 0xa3, 0x5c, 0xe4, 0x11,
+ 0xe5, 0xfb, 0xc1, 0x19, 0x1a, 0x0a, 0x52, 0xef,
+ 0xf6, 0x9f, 0x24, 0x45, 0xdf, 0x4f, 0x9b, 0x17,
+ 0xad, 0x2b, 0x41, 0x7b, 0xe6, 0x6c, 0x37, 0x10
+ };
+
+ SubSample subSamples[kNumSubsamples] = {
+ {0, 32},
+ {0, 32}
+ };
+
+ attemptDecryptExpectingSuccess<kTotalSize>(key, iv, encrypted, decrypted,
+ subSamples, kNumSubsamples);
+}
+
+TEST_F(AesCtrDecryptorTest, DecryptsUnalignedBifurcatedEncryptedBlock) {
+ const size_t kTotalSize = 64;
+ const size_t kNumSubsamples = 2;
+
+ // Test vectors from NIST-800-38A
+ Key key = {
+ 0x2b, 0x7e, 0x15, 0x16, 0x28, 0xae, 0xd2, 0xa6,
+ 0xab, 0xf7, 0x15, 0x88, 0x09, 0xcf, 0x4f, 0x3c
+ };
+
+ Iv iv = {
+ 0xf0, 0xf1, 0xf2, 0xf3, 0xf4, 0xf5, 0xf6, 0xf7,
+ 0xf8, 0xf9, 0xfa, 0xfb, 0xfc, 0xfd, 0xfe, 0xff
+ };
+
+ uint8_t encrypted[kTotalSize] = {
+ 0x87, 0x4d, 0x61, 0x91, 0xb6, 0x20, 0xe3, 0x26,
+ 0x1b, 0xef, 0x68, 0x64, 0x99, 0x0d, 0xb6, 0xce,
+ 0x98, 0x06, 0xf6, 0x6b, 0x79, 0x70, 0xfd, 0xff,
+ 0x86, 0x17, 0x18, 0x7b, 0xb9, 0xff, 0xfd, 0xff,
+ 0x5a, 0xe4, 0xdf, 0x3e, 0xdb, 0xd5, 0xd3, 0x5e,
+ 0x5b, 0x4f, 0x09, 0x02, 0x0d, 0xb0, 0x3e, 0xab,
+ 0x1e, 0x03, 0x1d, 0xda, 0x2f, 0xbe, 0x03, 0xd1,
+ 0x79, 0x21, 0x70, 0xa0, 0xf3, 0x00, 0x9c, 0xee
+ };
+
+ uint8_t decrypted[kTotalSize] = {
+ 0x6b, 0xc1, 0xbe, 0xe2, 0x2e, 0x40, 0x9f, 0x96,
+ 0xe9, 0x3d, 0x7e, 0x11, 0x73, 0x93, 0x17, 0x2a,
+ 0xae, 0x2d, 0x8a, 0x57, 0x1e, 0x03, 0xac, 0x9c,
+ 0x9e, 0xb7, 0x6f, 0xac, 0x45, 0xaf, 0x8e, 0x51,
+ 0x30, 0xc8, 0x1c, 0x46, 0xa3, 0x5c, 0xe4, 0x11,
+ 0xe5, 0xfb, 0xc1, 0x19, 0x1a, 0x0a, 0x52, 0xef,
+ 0xf6, 0x9f, 0x24, 0x45, 0xdf, 0x4f, 0x9b, 0x17,
+ 0xad, 0x2b, 0x41, 0x7b, 0xe6, 0x6c, 0x37, 0x10
+ };
+
+ SubSample subSamples[kNumSubsamples] = {
+ {0, 29},
+ {0, 35}
+ };
+
+ attemptDecryptExpectingSuccess<kTotalSize>(key, iv, encrypted, decrypted,
+ subSamples, kNumSubsamples);
+}
+
+TEST_F(AesCtrDecryptorTest, DecryptsOneMixedSubSample) {
+ const size_t kTotalSize = 72;
+ const size_t kNumSubsamples = 1;
+
+ // Based on test vectors from NIST-800-38A
+ Key key = {
+ 0x2b, 0x7e, 0x15, 0x16, 0x28, 0xae, 0xd2, 0xa6,
+ 0xab, 0xf7, 0x15, 0x88, 0x09, 0xcf, 0x4f, 0x3c
+ };
+
+ Iv iv = {
+ 0xf0, 0xf1, 0xf2, 0xf3, 0xf4, 0xf5, 0xf6, 0xf7,
+ 0xf8, 0xf9, 0xfa, 0xfb, 0xfc, 0xfd, 0xfe, 0xff
+ };
+
+ uint8_t encrypted[kTotalSize] = {
+ // 8 clear bytes
+ 0xf0, 0x13, 0xca, 0xc7, 0x00, 0x64, 0x0b, 0xbb,
+ // 64 encrypted bytes
+ 0x87, 0x4d, 0x61, 0x91, 0xb6, 0x20, 0xe3, 0x26,
+ 0x1b, 0xef, 0x68, 0x64, 0x99, 0x0d, 0xb6, 0xce,
+ 0x98, 0x06, 0xf6, 0x6b, 0x79, 0x70, 0xfd, 0xff,
+ 0x86, 0x17, 0x18, 0x7b, 0xb9, 0xff, 0xfd, 0xff,
+ 0x5a, 0xe4, 0xdf, 0x3e, 0xdb, 0xd5, 0xd3, 0x5e,
+ 0x5b, 0x4f, 0x09, 0x02, 0x0d, 0xb0, 0x3e, 0xab,
+ 0x1e, 0x03, 0x1d, 0xda, 0x2f, 0xbe, 0x03, 0xd1,
+ 0x79, 0x21, 0x70, 0xa0, 0xf3, 0x00, 0x9c, 0xee
+ };
+
+ uint8_t decrypted[kTotalSize] = {
+ 0xf0, 0x13, 0xca, 0xc7, 0x00, 0x64, 0x0b, 0xbb,
+ 0x6b, 0xc1, 0xbe, 0xe2, 0x2e, 0x40, 0x9f, 0x96,
+ 0xe9, 0x3d, 0x7e, 0x11, 0x73, 0x93, 0x17, 0x2a,
+ 0xae, 0x2d, 0x8a, 0x57, 0x1e, 0x03, 0xac, 0x9c,
+ 0x9e, 0xb7, 0x6f, 0xac, 0x45, 0xaf, 0x8e, 0x51,
+ 0x30, 0xc8, 0x1c, 0x46, 0xa3, 0x5c, 0xe4, 0x11,
+ 0xe5, 0xfb, 0xc1, 0x19, 0x1a, 0x0a, 0x52, 0xef,
+ 0xf6, 0x9f, 0x24, 0x45, 0xdf, 0x4f, 0x9b, 0x17,
+ 0xad, 0x2b, 0x41, 0x7b, 0xe6, 0x6c, 0x37, 0x10
+ };
+
+ SubSample subSamples[kNumSubsamples] = {
+ {8, 64}
+ };
+
+ attemptDecryptExpectingSuccess<kTotalSize>(key, iv, encrypted, decrypted,
+ subSamples, kNumSubsamples);
+}
+
+TEST_F(AesCtrDecryptorTest, DecryptsAlignedMixedSubSamples) {
+ const size_t kTotalSize = 80;
+ const size_t kNumSubsamples = 2;
+
+ // Based on test vectors from NIST-800-38A
+ Key key = {
+ 0x2b, 0x7e, 0x15, 0x16, 0x28, 0xae, 0xd2, 0xa6,
+ 0xab, 0xf7, 0x15, 0x88, 0x09, 0xcf, 0x4f, 0x3c
+ };
+
+ Iv iv = {
+ 0xf0, 0xf1, 0xf2, 0xf3, 0xf4, 0xf5, 0xf6, 0xf7,
+ 0xf8, 0xf9, 0xfa, 0xfb, 0xfc, 0xfd, 0xfe, 0xff
+ };
+
+ uint8_t encrypted[kTotalSize] = {
+ // 8 clear bytes
+ 0xf0, 0x13, 0xca, 0xc7, 0x00, 0x64, 0x0b, 0xbb,
+ // 32 encrypted bytes
+ 0x87, 0x4d, 0x61, 0x91, 0xb6, 0x20, 0xe3, 0x26,
+ 0x1b, 0xef, 0x68, 0x64, 0x99, 0x0d, 0xb6, 0xce,
+ 0x98, 0x06, 0xf6, 0x6b, 0x79, 0x70, 0xfd, 0xff,
+ 0x86, 0x17, 0x18, 0x7b, 0xb9, 0xff, 0xfd, 0xff,
+ // 8 clear bytes
+ 0x94, 0xba, 0x88, 0x2e, 0x0e, 0x12, 0x11, 0x55,
+ // 32 encrypted bytes
+ 0x5a, 0xe4, 0xdf, 0x3e, 0xdb, 0xd5, 0xd3, 0x5e,
+ 0x5b, 0x4f, 0x09, 0x02, 0x0d, 0xb0, 0x3e, 0xab,
+ 0x1e, 0x03, 0x1d, 0xda, 0x2f, 0xbe, 0x03, 0xd1,
+ 0x79, 0x21, 0x70, 0xa0, 0xf3, 0x00, 0x9c, 0xee
+ };
+
+ uint8_t decrypted[kTotalSize] = {
+ 0xf0, 0x13, 0xca, 0xc7, 0x00, 0x64, 0x0b, 0xbb,
+ 0x6b, 0xc1, 0xbe, 0xe2, 0x2e, 0x40, 0x9f, 0x96,
+ 0xe9, 0x3d, 0x7e, 0x11, 0x73, 0x93, 0x17, 0x2a,
+ 0xae, 0x2d, 0x8a, 0x57, 0x1e, 0x03, 0xac, 0x9c,
+ 0x9e, 0xb7, 0x6f, 0xac, 0x45, 0xaf, 0x8e, 0x51,
+ 0x94, 0xba, 0x88, 0x2e, 0x0e, 0x12, 0x11, 0x55,
+ 0x30, 0xc8, 0x1c, 0x46, 0xa3, 0x5c, 0xe4, 0x11,
+ 0xe5, 0xfb, 0xc1, 0x19, 0x1a, 0x0a, 0x52, 0xef,
+ 0xf6, 0x9f, 0x24, 0x45, 0xdf, 0x4f, 0x9b, 0x17,
+ 0xad, 0x2b, 0x41, 0x7b, 0xe6, 0x6c, 0x37, 0x10
+ };
+
+ SubSample subSamples[kNumSubsamples] = {
+ {8, 32},
+ {8, 32}
+ };
+
+ attemptDecryptExpectingSuccess<kTotalSize>(key, iv, encrypted, decrypted,
+ subSamples, kNumSubsamples);
+}
+
+TEST_F(AesCtrDecryptorTest, DecryptsUnalignedMixedSubSamples) {
+ const size_t kTotalSize = 80;
+ const size_t kNumSubsamples = 2;
+
+ // Based on test vectors from NIST-800-38A
+ Key key = {
+ 0x2b, 0x7e, 0x15, 0x16, 0x28, 0xae, 0xd2, 0xa6,
+ 0xab, 0xf7, 0x15, 0x88, 0x09, 0xcf, 0x4f, 0x3c
+ };
+
+ Iv iv = {
+ 0xf0, 0xf1, 0xf2, 0xf3, 0xf4, 0xf5, 0xf6, 0xf7,
+ 0xf8, 0xf9, 0xfa, 0xfb, 0xfc, 0xfd, 0xfe, 0xff
+ };
+
+ uint8_t encrypted[kTotalSize] = {
+ // 8 clear bytes
+ 0xf0, 0x13, 0xca, 0xc7, 0x00, 0x64, 0x0b, 0xbb,
+ // 30 encrypted bytes
+ 0x87, 0x4d, 0x61, 0x91, 0xb6, 0x20, 0xe3, 0x26,
+ 0x1b, 0xef, 0x68, 0x64, 0x99, 0x0d, 0xb6, 0xce,
+ 0x98, 0x06, 0xf6, 0x6b, 0x79, 0x70, 0xfd, 0xff,
+ 0x86, 0x17, 0x18, 0x7b, 0xb9, 0xff,
+ // 8 clear bytes
+ 0x94, 0xba, 0x88, 0x2e, 0x0e, 0x12, 0x11, 0x55,
+ // 34 encrypted bytes
+ 0xfd, 0xff, 0x5a, 0xe4, 0xdf, 0x3e, 0xdb, 0xd5,
+ 0xd3, 0x5e, 0x5b, 0x4f, 0x09, 0x02, 0x0d, 0xb0,
+ 0x3e, 0xab, 0x1e, 0x03, 0x1d, 0xda, 0x2f, 0xbe,
+ 0x03, 0xd1, 0x79, 0x21, 0x70, 0xa0, 0xf3, 0x00,
+ 0x9c, 0xee
+ };
+
+ uint8_t decrypted[kTotalSize] = {
+ 0xf0, 0x13, 0xca, 0xc7, 0x00, 0x64, 0x0b, 0xbb,
+ 0x6b, 0xc1, 0xbe, 0xe2, 0x2e, 0x40, 0x9f, 0x96,
+ 0xe9, 0x3d, 0x7e, 0x11, 0x73, 0x93, 0x17, 0x2a,
+ 0xae, 0x2d, 0x8a, 0x57, 0x1e, 0x03, 0xac, 0x9c,
+ 0x9e, 0xb7, 0x6f, 0xac, 0x45, 0xaf, 0x94, 0xba,
+ 0x88, 0x2e, 0x0e, 0x12, 0x11, 0x55, 0x8e, 0x51,
+ 0x30, 0xc8, 0x1c, 0x46, 0xa3, 0x5c, 0xe4, 0x11,
+ 0xe5, 0xfb, 0xc1, 0x19, 0x1a, 0x0a, 0x52, 0xef,
+ 0xf6, 0x9f, 0x24, 0x45, 0xdf, 0x4f, 0x9b, 0x17,
+ 0xad, 0x2b, 0x41, 0x7b, 0xe6, 0x6c, 0x37, 0x10
+ };
+
+ SubSample subSamples[kNumSubsamples] = {
+ {8, 30},
+ {8, 34}
+ };
+
+ attemptDecryptExpectingSuccess<kTotalSize>(key, iv, encrypted, decrypted,
+ subSamples, kNumSubsamples);
+}
+
+TEST_F(AesCtrDecryptorTest, DecryptsComplexMixedSubSamples) {
+ const size_t kTotalSize = 72;
+ const size_t kNumSubsamples = 6;
+
+ // Based on test vectors from NIST-800-38A
+ Key key = {
+ 0x2b, 0x7e, 0x15, 0x16, 0x28, 0xae, 0xd2, 0xa6,
+ 0xab, 0xf7, 0x15, 0x88, 0x09, 0xcf, 0x4f, 0x3c
+ };
+
+ Iv iv = {
+ 0xf0, 0xf1, 0xf2, 0xf3, 0xf4, 0xf5, 0xf6, 0xf7,
+ 0xf8, 0xf9, 0xfa, 0xfb, 0xfc, 0xfd, 0xfe, 0xff
+ };
+
+ uint8_t encrypted[kTotalSize] = {
+ // 4 clear bytes
+ 0xf0, 0x13, 0xca, 0xc7,
+ // 1 encrypted bytes
+ 0x87,
+ // 9 encrypted bytes
+ 0x4d, 0x61, 0x91, 0xb6, 0x20, 0xe3, 0x26, 0x1b,
+ 0xef,
+ // 11 clear bytes
+ 0x81, 0x4f, 0x24, 0x87, 0x0e, 0xde, 0xba, 0xad,
+ 0x11, 0x9b, 0x46,
+ // 20 encrypted bytes
+ 0x68, 0x64, 0x99, 0x0d, 0xb6, 0xce,
+ 0x98, 0x06, 0xf6, 0x6b, 0x79, 0x70, 0xfd, 0xff,
+ 0x86, 0x17, 0x18, 0x7b, 0xb9, 0xff,
+ // 8 clear bytes
+ 0x94, 0xba, 0x88, 0x2e, 0x0e, 0x12, 0x11, 0x55,
+ // 3 clear bytes
+ 0x10, 0xf5, 0x22,
+ // 14 encrypted bytes
+ 0xfd, 0xff, 0x5a, 0xe4, 0xdf, 0x3e, 0xdb, 0xd5,
+ 0xd3, 0x5e, 0x5b, 0x4f, 0x09, 0x02,
+ // 2 clear bytes
+ 0x02, 0x01
+ };
+
+ uint8_t decrypted[kTotalSize] = {
+ 0xf0, 0x13, 0xca, 0xc7, 0x6b, 0xc1, 0xbe, 0xe2,
+ 0x2e, 0x40, 0x9f, 0x96, 0xe9, 0x3d, 0x81, 0x4f,
+ 0x24, 0x87, 0x0e, 0xde, 0xba, 0xad, 0x11, 0x9b,
+ 0x46, 0x7e, 0x11, 0x73, 0x93, 0x17, 0x2a, 0xae,
+ 0x2d, 0x8a, 0x57, 0x1e, 0x03, 0xac, 0x9c, 0x9e,
+ 0xb7, 0x6f, 0xac, 0x45, 0xaf, 0x94, 0xba, 0x88,
+ 0x2e, 0x0e, 0x12, 0x11, 0x55, 0x10, 0xf5, 0x22,
+ 0x8e, 0x51, 0x30, 0xc8, 0x1c, 0x46, 0xa3, 0x5c,
+ 0xe4, 0x11, 0xe5, 0xfb, 0xc1, 0x19, 0x02, 0x01
+ };
+
+ SubSample subSamples[kNumSubsamples] = {
+ {4, 1},
+ {0, 9},
+ {11, 20},
+ {8, 0},
+ {3, 14},
+ {2, 0}
+ };
+
+ attemptDecryptExpectingSuccess<kTotalSize>(key, iv, encrypted, decrypted,
+ subSamples, kNumSubsamples);
+}
+
+} // namespace clearkeydrm
diff --git a/drm/mediadrm/plugins/clearkey/tests/Android.mk b/drm/mediadrm/plugins/clearkey/tests/Android.mk
new file mode 100644
index 0000000..ac5bb21
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/tests/Android.mk
@@ -0,0 +1,52 @@
+#
+# Copyright (C) 2014 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.
+#
+# ----------------------------------------------------------------
+# Builds ClearKey Drm Tests
+#
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+LOCAL_MODULE := ClearKeyDrmUnitTest
+LOCAL_MODULE_TAGS := tests
+
+LOCAL_SRC_FILES := \
+ AesCtrDecryptorUnittest.cpp \
+ InitDataParserUnittest.cpp \
+ JsonWebKeyUnittest.cpp \
+
+LOCAL_C_INCLUDES := \
+ bionic \
+ external/gtest/include \
+ external/jsmn \
+ external/openssl/include \
+ external/stlport/stlport \
+ frameworks/av/drm/mediadrm/plugins/clearkey \
+ frameworks/av/include \
+ frameworks/native/include \
+
+LOCAL_STATIC_LIBRARIES := \
+ libgtest \
+ libgtest_main \
+
+LOCAL_SHARED_LIBRARIES := \
+ libcrypto \
+ libdrmclearkeyplugin \
+ liblog \
+ libstagefright_foundation \
+ libstlport \
+ libutils \
+
+include $(BUILD_NATIVE_TEST)
diff --git a/drm/mediadrm/plugins/clearkey/tests/InitDataParserUnittest.cpp b/drm/mediadrm/plugins/clearkey/tests/InitDataParserUnittest.cpp
new file mode 100644
index 0000000..4ba65ed
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/tests/InitDataParserUnittest.cpp
@@ -0,0 +1,235 @@
+/*
+ * Copyright (C) 2014 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 <gtest/gtest.h>
+#include <string.h>
+
+#include <media/stagefright/foundation/AString.h>
+#include <media/stagefright/foundation/base64.h>
+#include <utils/String8.h>
+#include <utils/Vector.h>
+
+#include "InitDataParser.h"
+
+namespace clearkeydrm {
+
+using namespace android;
+
+namespace {
+ const size_t kKeyIdSize = 16;
+ const String8 kCencType("cenc");
+ const String8 kWebMType("webm");
+ const String8 kBase64Padding("=");
+}
+
+class InitDataParserTest : public ::testing::Test {
+ protected:
+ status_t attemptParse(const Vector<uint8_t>& initData,
+ const String8& initDataType,
+ Vector<uint8_t>* licenseRequest) {
+ InitDataParser parser;
+ return parser.parse(initData, initDataType, licenseRequest);
+ }
+
+ void attemptParseExpectingSuccess(const Vector<uint8_t>& initData,
+ const String8& initDataType,
+ const Vector<String8>& expectedKeys) {
+ const String8 kRequestPrefix("{\"kids\":[");
+ const String8 kRequestSuffix("],\"type\":\"temporary\"}");
+ Vector<uint8_t> request;
+ ASSERT_EQ(android::OK, attemptParse(initData, initDataType, &request));
+
+ String8 requestString(reinterpret_cast<const char*>(request.array()),
+ request.size());
+ EXPECT_EQ(0, requestString.find(kRequestPrefix));
+ EXPECT_EQ(requestString.size() - kRequestSuffix.size(),
+ requestString.find(kRequestSuffix));
+ for (size_t i = 0; i < expectedKeys.size(); ++i) {
+ AString encodedIdAString;
+ android::encodeBase64(expectedKeys[i], kKeyIdSize,
+ &encodedIdAString);
+ String8 encodedId(encodedIdAString.c_str());
+ encodedId.removeAll(kBase64Padding);
+ EXPECT_TRUE(requestString.contains(encodedId));
+ }
+ }
+
+ void attemptParseExpectingFailure(const Vector<uint8_t>& initData,
+ const String8& initDataType) {
+ Vector<uint8_t> request;
+ ASSERT_NE(android::OK, attemptParse(initData, initDataType, &request));
+ EXPECT_EQ(0, request.size());
+ }
+};
+
+TEST_F(InitDataParserTest, ParsesSingleKeyPssh) {
+ uint8_t pssh[52] = {
+ 0, 0, 0, 52, // Total Size
+ 'p', 's', 's', 'h', // PSSH
+ 1, 0, 0, 0, // Version
+ 0x10, 0x77, 0xef, 0xec, 0xc0, 0xb2, 0x4d, 0x02, // System ID
+ 0xac, 0xe3, 0x3c, 0x1e, 0x52, 0xe2, 0xfb, 0x4b,
+ 0, 0, 0, 1, // Key Count
+ 0x30, 0x31, 0x32, 0x33, 0x34, 0x35, 0x36, 0x37, // Key ID #1
+ 0x38, 0x39, 0x30, 0x41, 0x42, 0x43, 0x44, 0x45, // "01234567890ABCDE"
+ 0, 0, 0, 0 // Data Size (always 0)
+ };
+ Vector<uint8_t> initData;
+ initData.appendArray(pssh, 52);
+
+ Vector<String8> expectedKeys;
+ expectedKeys.push(String8("01234567890ABCDE"));
+
+ attemptParseExpectingSuccess(initData, kCencType, expectedKeys);
+}
+
+TEST_F(InitDataParserTest, ParsesMultipleKeyPssh) {
+ uint8_t pssh[84] = {
+ 0, 0, 0, 84, // Total Size
+ 'p', 's', 's', 'h', // PSSH
+ 1, 0, 0, 0, // Version
+ 0x10, 0x77, 0xef, 0xec, 0xc0, 0xb2, 0x4d, 0x02, // System ID
+ 0xac, 0xe3, 0x3c, 0x1e, 0x52, 0xe2, 0xfb, 0x4b,
+ 0, 0, 0, 3, // Key Count
+ 0x30, 0x31, 0x32, 0x33, 0x34, 0x35, 0x36, 0x37, // Key ID #1
+ 0x38, 0x39, 0x30, 0x41, 0x42, 0x43, 0x44, 0x45, // "01234567890ABCDE"
+ 0x43, 0x6c, 0x65, 0x61, 0x72, 0x4b, 0x65, 0x79, // Key ID #2
+ 0x43, 0x6c, 0x65, 0x61, 0x72, 0x4b, 0x65, 0x79, // "ClearKeyClearKey"
+ 0x20, 0x47, 0x4f, 0x4f, 0x47, 0x4c, 0x45, 0x20, // Key ID #3
+ 0x20, 0x47, 0x4f, 0x4f, 0x47, 0x4c, 0x45, 0x20, // " GOOGLE GOOGLE "
+ 0, 0, 0, 0 // Data Size (always 0)
+ };
+ Vector<uint8_t> initData;
+ initData.appendArray(pssh, 84);
+
+ Vector<String8> expectedKeys;
+ expectedKeys.push(String8("01234567890ABCDE"));
+ expectedKeys.push(String8("ClearKeyClearKey"));
+ expectedKeys.push(String8(" GOOGLE GOOGLE "));
+
+ attemptParseExpectingSuccess(initData, kCencType, expectedKeys);
+}
+
+TEST_F(InitDataParserTest, ParsesWebM) {
+ uint8_t initDataRaw[16] = {
+ 0x30, 0x31, 0x32, 0x33, 0x34, 0x35, 0x36, 0x37, // Key ID
+ 0x38, 0x39, 0x30, 0x41, 0x42, 0x43, 0x44, 0x45, // "01234567890ABCDE"
+ };
+ Vector<uint8_t> initData;
+ initData.appendArray(initDataRaw, 16);
+
+ Vector<String8> expectedKeys;
+ expectedKeys.push(String8("01234567890ABCDE"));
+
+ attemptParseExpectingSuccess(initData, kWebMType, expectedKeys);
+}
+
+TEST_F(InitDataParserTest, FailsForPsshTooSmall) {
+ uint8_t pssh[16] = {
+ 0, 0, 0, 52,
+ 'p', 's', 's', 'h',
+ 1, 0, 0, 0,
+ 0x10, 0x77, 0xef, 0xec
+ };
+ Vector<uint8_t> initData;
+ initData.appendArray(pssh, 16);
+
+ attemptParseExpectingFailure(initData, kCencType);
+}
+
+TEST_F(InitDataParserTest, FailsForWebMTooSmall) {
+ uint8_t initDataRaw[8] = {
+ 0x30, 0x31, 0x32, 0x33, 0x34, 0x35, 0x36, 0x37
+ };
+ Vector<uint8_t> initData;
+ initData.appendArray(initDataRaw, 8);
+
+ attemptParseExpectingFailure(initData, kWebMType);
+}
+
+TEST_F(InitDataParserTest, FailsForPsshBadSystemId) {
+ uint8_t pssh[52] = {
+ 0, 0, 0, 52, // Total Size
+ 'p', 's', 's', 'h', // PSSH
+ 1, 0, 0, 0, // Version
+ 0xac, 0xe3, 0x3c, 0x1e, 0x52, 0xe2, 0xfb, 0x4b, // System ID
+ 0x10, 0x77, 0xef, 0xec, 0xc0, 0xb2, 0x4d, 0x02,
+ 0, 0, 0, 1, // Key Count
+ 0x30, 0x31, 0x32, 0x33, 0x34, 0x35, 0x36, 0x37, // Key ID #1
+ 0x38, 0x39, 0x30, 0x41, 0x42, 0x43, 0x44, 0x45, // "01234567890ABCDE"
+ 0, 0, 0, 0 // Data Size (always 0)
+ };
+ Vector<uint8_t> initData;
+ initData.appendArray(pssh, 52);
+
+ attemptParseExpectingFailure(initData, kCencType);
+}
+
+TEST_F(InitDataParserTest, FailsForPsshBadSize) {
+ uint8_t pssh[52] = {
+ 0, 0, 70, 200, // Total Size
+ 'p', 's', 's', 'h', // PSSH
+ 1, 0, 0, 0, // Version
+ 0x10, 0x77, 0xef, 0xec, 0xc0, 0xb2, 0x4d, 0x02, // System ID
+ 0xac, 0xe3, 0x3c, 0x1e, 0x52, 0xe2, 0xfb, 0x4b,
+ 0, 0, 0, 1, // Key Count
+ 0x30, 0x31, 0x32, 0x33, 0x34, 0x35, 0x36, 0x37, // Key ID #1
+ 0x38, 0x39, 0x30, 0x41, 0x42, 0x43, 0x44, 0x45, // "01234567890ABCDE"
+ 0, 0, 0, 0 // Data Size (always 0)
+ };
+ Vector<uint8_t> initData;
+ initData.appendArray(pssh, 52);
+
+ attemptParseExpectingFailure(initData, kCencType);
+}
+
+TEST_F(InitDataParserTest, FailsForPsshWrongVersion) {
+ uint8_t pssh[52] = {
+ 0, 0, 0, 52, // Total Size
+ 'p', 's', 's', 'h', // PSSH
+ 0, 0, 0, 0, // Version
+ 0x10, 0x77, 0xef, 0xec, 0xc0, 0xb2, 0x4d, 0x02, // System ID
+ 0xac, 0xe3, 0x3c, 0x1e, 0x52, 0xe2, 0xfb, 0x4b,
+ 0, 0, 0, 1, // Key Count
+ 0x30, 0x31, 0x32, 0x33, 0x34, 0x35, 0x36, 0x37, // Key ID #1
+ 0x38, 0x39, 0x30, 0x41, 0x42, 0x43, 0x44, 0x45, // "01234567890ABCDE"
+ 0, 0, 0, 0 // Data Size (always 0)
+ };
+ Vector<uint8_t> initData;
+ initData.appendArray(pssh, 52);
+
+ attemptParseExpectingFailure(initData, kCencType);
+}
+
+TEST_F(InitDataParserTest, FailsForPsshBadKeyCount) {
+ uint8_t pssh[52] = {
+ 0, 0, 0, 52, // Total Size
+ 'p', 's', 's', 'h', // PSSH
+ 1, 0, 0, 0, // Version
+ 0x10, 0x77, 0xef, 0xec, 0xc0, 0xb2, 0x4d, 0x02, // System ID
+ 0xac, 0xe3, 0x3c, 0x1e, 0x52, 0xe2, 0xfb, 0x4b,
+ 0, 0, 0, 7, // Key Count
+ 0x30, 0x31, 0x32, 0x33, 0x34, 0x35, 0x36, 0x37, // Key ID #1
+ 0x38, 0x39, 0x30, 0x41, 0x42, 0x43, 0x44, 0x45, // "01234567890ABCDE"
+ 0, 0, 0, 0 // Data Size (always 0)
+ };
+ Vector<uint8_t> initData;
+ initData.appendArray(pssh, 52);
+
+ attemptParseExpectingFailure(initData, kCencType);
+}
+
+} // namespace clearkeydrm
diff --git a/drm/mediadrm/plugins/clearkey/tests/JsonWebKeyUnittest.cpp b/drm/mediadrm/plugins/clearkey/tests/JsonWebKeyUnittest.cpp
new file mode 100644
index 0000000..c3b0d84
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/tests/JsonWebKeyUnittest.cpp
@@ -0,0 +1,321 @@
+/*
+ * Copyright (C) 2014 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 <utils/Log.h>
+
+#include "JsonWebKey.h"
+
+#include "gtest/gtest.h"
+#include "Utils.h"
+
+namespace clearkeydrm {
+using android::String8;
+using android::Vector;
+
+class JsonWebKeyTest : public ::testing::Test {
+ protected:
+ JsonWebKey* jwk;
+
+ JsonWebKeyTest() {
+ jwk = new JsonWebKey;
+ }
+
+ virtual ~JsonWebKeyTest() {
+ if (jwk)
+ delete jwk;
+ }
+};
+
+void stringFromVector(const Vector<uint8_t>& input,
+ String8* converted) {
+ converted->clear();
+ if (input.isEmpty()) {
+ return;
+ }
+
+ for (size_t i = 0; i < input.size(); ++i) {
+ converted->appendFormat("%c", input.itemAt(i));
+ }
+}
+
+void verifyKeys(const KeyMap& keys, const String8* clearKeys) {
+ if (keys.isEmpty()) {
+ return;
+ }
+
+ String8 keyString;
+ for (size_t i = 0; i < keys.size(); ++i) {
+ stringFromVector(keys.valueAt(i), &keyString);
+ EXPECT_EQ(keyString, clearKeys[i]);
+ }
+}
+
+TEST_F(JsonWebKeyTest, NoSymmetricKey) {
+ const String8 js(
+ "{"
+ "[{"
+ "\"kty\":\"rsa\","
+ "\"alg\":\"A128KW1\","
+ "\"kid\":\"Y2xlYXJrZXlrZXlpZDAx\","
+ "\"k\":\"1-GawgguFyGrWKav7AX4VKUg\""
+ "}]"
+ "}");
+
+ KeyMap keys;
+ EXPECT_FALSE(jwk->extractKeysFromJsonWebKeySet(js, &keys));
+ EXPECT_TRUE(keys.isEmpty());
+}
+
+TEST_F(JsonWebKeyTest, NoKeysTag) {
+ const String8 js(
+ "{"
+ "[{"
+ "\"kty\":\"oct\","
+ "\"alg\":\"A128KW1\","
+ "\"kid\":\"Y2xlYXJrZXlrZXlpZDAx\","
+ "\"k\":\"1-GawgguFyGrWKav7AX4VKUg\""
+ "},"
+ "{"
+ "\"kty\":\"oct\","
+ "\"alg\":\"A128KW2\","
+ "\"k\":\"R29vZCBkYXkh\","
+ "\"kid\":\"Y2xlYXJrZXlrZXlpZDAy\""
+ "}]"
+ "}");
+
+ KeyMap keys;
+ EXPECT_FALSE(jwk->extractKeysFromJsonWebKeySet(js, &keys));
+ EXPECT_TRUE(keys.isEmpty());
+}
+
+TEST_F(JsonWebKeyTest, NoKeyId) {
+ const String8 js(
+ "{"
+ "\"keys\":"
+ "[{"
+ "\"kty\":\"oct\""
+ "\"alg\":\"A128KW1\""
+ "\"k\":\"SGVsbG8gRnJpZW5kISE=\""
+ "}"
+ "{"
+ "\"kty\":\"oct\""
+ "\"alg\":\"A128KW2\""
+ "\"k\":\"R29vZCBkYXkh\""
+ "\"kid\":\"Y2xlYXJrZXlrZXlpZDAy\""
+ "}]"
+ "}");
+
+ KeyMap keys;
+ EXPECT_TRUE(jwk->extractKeysFromJsonWebKeySet(js, &keys));
+ EXPECT_TRUE(keys.size() == 1);
+
+ const String8 clearKeys("Good day!");
+ verifyKeys(keys, &clearKeys);
+}
+
+TEST_F(JsonWebKeyTest, NoKey) {
+ const String8 js(
+ "{"
+ "\"keys\":"
+ "[{"
+ "\"kty\":\"oct\""
+ "\"alg\":\"A128KW1\""
+ "\"kid\":\"`\""
+ "}"
+ "{"
+ "\"kty\":\"oct\""
+ "\"alg\":\"A128KW2\""
+ "\"k\":\"R29vZCBkYXkh\""
+ "\"kid\":\"Y2xlYXJrZXlrZXlpZDAy\""
+ "}]"
+ "}");
+
+ KeyMap keys;
+ EXPECT_TRUE(jwk->extractKeysFromJsonWebKeySet(js, &keys));
+ EXPECT_TRUE(keys.size() == 1);
+
+ const String8 clearKeys("Good day!");
+ verifyKeys(keys, &clearKeys);
+}
+
+TEST_F(JsonWebKeyTest, MalformedKey) {
+ const String8 js(
+ "{"
+ "\"keys\":"
+ "[{"
+ "\"kty\":\"oct\""
+ "\"alg\":\"A128KW1\""
+ "\"k\":\"GawgguFyGrWKav7AX4V???\""
+ "\"kid\":\"67ef0gd8pvfd0=\""
+ "}"
+ "{"
+ "\"kty\":\"oct\""
+ "\"alg\":\"A128KW1\""
+ "\"k\":\"GawgguFyGrWKav7AX4V???\""
+ "\"kid\":"
+ "}"
+ "{"
+ "\"kty\":\"oct\""
+ "\"alg\":\"A128KW1\""
+ ":\"GawgguFyGrWKav7AX4V???\""
+ "\"kid\":\"67ef0gd8pvfd0=\""
+ "}"
+ "{"
+ "\"kty\":\"oct\""
+ "\"alg\":\"A128KW3\""
+ "\"kid\":\"Y2xlYXJrZXlrZXlpZDAz\""
+ "\"k\":\"R29vZCBkYXkh\""
+ "}]"
+ "}");
+
+ KeyMap keys;
+ EXPECT_TRUE(jwk->extractKeysFromJsonWebKeySet(js, &keys));
+ EXPECT_TRUE(keys.size() == 1);
+
+ const String8 clearKeys("Good day!");
+ verifyKeys(keys, &clearKeys);
+}
+
+TEST_F(JsonWebKeyTest, EmptyJsonWebKey) {
+ const String8 js;
+ KeyMap keys;
+ EXPECT_FALSE(jwk->extractKeysFromJsonWebKeySet(js, &keys));
+ EXPECT_TRUE(keys.isEmpty());
+}
+
+TEST_F(JsonWebKeyTest, MalformedJsonWebKey) {
+ // Missing begin array '['
+ const String8 js(
+ "{"
+ "\"keys\":"
+ "{"
+ "\"kty\":\"oct\""
+ "\"alg\":\"A128KW1\""
+ "\"k\":\"GawgguFyGrWKav7AX4VKUg\""
+ "\"kid\":\"67ef0gd8pvfd0=\""
+ "}"
+ "]"
+ "}");
+
+ KeyMap keys;
+ EXPECT_FALSE(jwk->extractKeysFromJsonWebKeySet(js, &keys));
+ EXPECT_TRUE(keys.isEmpty());
+}
+
+TEST_F(JsonWebKeyTest, SameKeyId) {
+ const String8 js(
+ "{"
+ "\"keys\":"
+ "[{"
+ "\"kty\":\"oct\""
+ "\"alg\":\"A128KW1\""
+ "\"kid\":\"Y2xlYXJrZXlrZXlpZDAx\""
+ "\"k\":\"SGVsbG8gRnJpZW5kISE\""
+ "}"
+ "{"
+ "\"kty\":\"oct\""
+ "\"alg\":\"A128KW1\""
+ "\"k\":\"SGVsbG8gRnJpZW5kIQ\""
+ "\"kid\":\"Y2xlYXJrZXlrZXlpZDAx\""
+ "}"
+ "{"
+ "\"kty\":\"oct\""
+ "\"alg\":\"A128KW3\""
+ "\"kid\":\"Y2xlYXJrZXlrZXlpZDAz\""
+ "\"k\":\"R29vZCBkYXkh\""
+ "}]"
+ "}");
+
+ KeyMap keys;
+ jwk->extractKeysFromJsonWebKeySet(js, &keys);
+ EXPECT_TRUE(keys.size() == 2);
+
+ const String8 clearKeys[] =
+ { String8("Hello Friend!"), String8("Good day!") };
+ verifyKeys(keys, clearKeys);
+}
+
+TEST_F(JsonWebKeyTest, ExtractWellFormedKeys) {
+ const String8 js(
+ "{"
+ "\"keys\":"
+ "[{"
+ "\"kty\":\"oct\""
+ "\"alg\":\"A128KW1\""
+ "}"
+ "{"
+ "\"kty\":\"oct\""
+ "\"alg\":\"A128KW2\""
+ "\"k\":\"SGVsbG8gRnJpZW5kIQ\""
+ "\"kid\":\"Y2xlYXJrZXlrZXlpZDAy\""
+ "}"
+ "{"
+ "\"kty\":\"oct\""
+ "\"alg\":\"A128KW3\""
+ "\"kid\":\"Y2xlYXJrZXlrZXlpZDAz\""
+ "\"k\":\"R29vZCBkYXkh\""
+ "}]"
+ "}");
+
+ KeyMap keys;
+ jwk->extractKeysFromJsonWebKeySet(js, &keys);
+ EXPECT_TRUE(keys.size() == 2);
+
+ const String8 clearKeys[] =
+ { String8("Hello Friend!"), String8("Good day!") };
+ verifyKeys(keys, clearKeys);
+}
+
+TEST_F(JsonWebKeyTest, ExtractKeys) {
+ const String8 js(
+ "{"
+ "\"keys\":"
+ "[{"
+ "\"kid\":\"Y2xlYXJrZXlrZXlpZDAx\""
+ "\"k\":\"SGVsbG8gRnJpZW5kISE\""
+ "\"kty\":\"oct\""
+ "\"alg\":\"A128KW1\""
+ "}"
+ "{"
+ "\"kty\":\"oct\""
+ "\"alg\":\"A128KW2\""
+ "\"k\":\"SGVsbG8gRnJpZW5kIQ\""
+ "\"kid\":\"Y2xlYXJrZXlrZXlpZDAy\""
+ "}"
+ "{"
+ "\"kty\":\"rsa\""
+ "\"alg\":\"A128KW-rsa\""
+ "\"k\":\"R29vZCBkYXkh\""
+ "\"kid\":\"rsa-67ef0gd8pvfd0=\""
+ "}"
+ "{"
+ "\"alg\":\"A128KW3\""
+ "\"kid\":\"Y2xlYXJrZXlrZXlpZDAz\""
+ "\"k\":\"R29vZCBkYXkh\""
+ "\"kty\":\"oct\""
+ "}]"
+ "}");
+
+ KeyMap keys;
+ jwk->extractKeysFromJsonWebKeySet(js, &keys);
+ EXPECT_TRUE(keys.size() == 3);
+
+ const String8 clearKeys[] =
+ { String8("Hello Friend!!"), String8("Hello Friend!"),
+ String8("Good day!") };
+ verifyKeys(keys, clearKeys);
+}
+
+} // namespace clearkeydrm
diff --git a/include/camera/Camera.h b/include/camera/Camera.h
index 79682b8..2b60842 100644
--- a/include/camera/Camera.h
+++ b/include/camera/Camera.h
@@ -74,6 +74,10 @@
const String16& clientPackageName,
int clientUid);
+ static status_t connectLegacy(int cameraId, int halVersion,
+ const String16& clientPackageName,
+ int clientUid, sp<Camera>& camera);
+
virtual ~Camera();
status_t reconnect();
diff --git a/include/camera/ICameraService.h b/include/camera/ICameraService.h
index d7e5dac..f7f06bb 100644
--- a/include/camera/ICameraService.h
+++ b/include/camera/ICameraService.h
@@ -52,6 +52,7 @@
GET_CAMERA_VENDOR_TAG_DESCRIPTOR,
GET_LEGACY_PARAMETERS,
SUPPORTS_CAMERA_API,
+ CONNECT_LEGACY,
};
enum {
@@ -63,6 +64,10 @@
API_VERSION_2 = 2,
};
+ enum {
+ CAMERA_HAL_API_VERSION_UNSPECIFIED = -1
+ };
+
public:
DECLARE_META_INTERFACE(CameraService);
@@ -125,6 +130,18 @@
*/
virtual status_t supportsCameraApi(
int cameraId, int apiVersion) = 0;
+
+ /**
+ * Connect the device as a legacy device for a given HAL version.
+ * For halVersion, use CAMERA_API_DEVICE_VERSION_* for a particular
+ * version, or CAMERA_HAL_API_VERSION_UNSPECIFIED for a service-selected version.
+ */
+ virtual status_t connectLegacy(const sp<ICameraClient>& cameraClient,
+ int cameraId, int halVersion,
+ const String16& clientPackageName,
+ int clientUid,
+ /*out*/
+ sp<ICamera>& device) = 0;
};
// ----------------------------------------------------------------------------
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index 6a68c94..f9c7efd 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -240,6 +240,11 @@
*/
uint32_t getSampleRate() const { return mSampleRate; }
+ /* Return the notification frame count.
+ * This is approximately how often the callback is invoked, for transfer type TRANSFER_CALLBACK.
+ */
+ size_t notificationFrames() const { return mNotificationFramesAct; }
+
/* Sets marker position. When record reaches the number of frames specified,
* a callback with event type EVENT_MARKER is called. Calling setMarkerPosition
* with marker == 0 cancels marker notification callback.
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index a4722cb..e1aab41 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -99,6 +99,8 @@
// to be non-zero if status == NO_ERROR
static status_t getOutputSamplingRate(uint32_t* samplingRate,
audio_stream_type_t stream);
+ static status_t getOutputSamplingRateForAttr(uint32_t* samplingRate,
+ const audio_attributes_t *attr);
static status_t getOutputFrameCount(size_t* frameCount,
audio_stream_type_t stream);
static status_t getOutputLatency(uint32_t* latency,
@@ -106,7 +108,7 @@
static status_t getSamplingRate(audio_io_handle_t output,
uint32_t* samplingRate);
// returns the number of frames per audio HAL write buffer. Corresponds to
- // audio_stream->get_buffer_size()/audio_stream_frame_size()
+ // audio_stream->get_buffer_size()/audio_stream_out_frame_size()
static status_t getFrameCount(audio_io_handle_t output,
size_t* frameCount);
// returns the audio output stream latency in ms. Corresponds to
@@ -212,7 +214,12 @@
audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO,
audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
const audio_offload_info_t *offloadInfo = NULL);
-
+ static audio_io_handle_t getOutputForAttr(const audio_attributes_t *attr,
+ uint32_t samplingRate = 0,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
+ audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO,
+ audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+ const audio_offload_info_t *offloadInfo = NULL);
static status_t startOutput(audio_io_handle_t output,
audio_stream_type_t stream,
int session);
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 79db323..a3cc396 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -188,7 +188,8 @@
transfer_type transferType = TRANSFER_DEFAULT,
const audio_offload_info_t *offloadInfo = NULL,
int uid = -1,
- pid_t pid = -1);
+ pid_t pid = -1,
+ const audio_attributes_t* pAttributes = NULL);
/* Creates an audio track and registers it with AudioFlinger.
* With this constructor, the track is configured for static buffer mode.
@@ -214,7 +215,8 @@
transfer_type transferType = TRANSFER_DEFAULT,
const audio_offload_info_t *offloadInfo = NULL,
int uid = -1,
- pid_t pid = -1);
+ pid_t pid = -1,
+ const audio_attributes_t* pAttributes = NULL);
/* Terminates the AudioTrack and unregisters it from AudioFlinger.
* Also destroys all resources associated with the AudioTrack.
@@ -253,7 +255,8 @@
transfer_type transferType = TRANSFER_DEFAULT,
const audio_offload_info_t *offloadInfo = NULL,
int uid = -1,
- pid_t pid = -1);
+ pid_t pid = -1,
+ const audio_attributes_t* pAttributes = NULL);
/* Result of constructing the AudioTrack. This must be checked for successful initialization
* before using any AudioTrack API (except for set()), because using
@@ -586,6 +589,11 @@
AudioTrack(const AudioTrack& other);
AudioTrack& operator = (const AudioTrack& other);
+ void setAttributesFromStreamType(audio_stream_type_t streamType);
+ void setStreamTypeFromAttributes(audio_attributes_t& aa);
+ /* paa is guaranteed non-NULL */
+ bool isValidAttributes(const audio_attributes_t *paa);
+
/* a small internal class to handle the callback */
class AudioTrackThread : public Thread
{
@@ -626,6 +634,8 @@
nsecs_t processAudioBuffer();
bool isOffloaded() const;
+ bool isDirect() const;
+ bool isOffloadedOrDirect() const;
// caller must hold lock on mLock for all _l methods
@@ -642,6 +652,13 @@
bool isOffloaded_l() const
{ return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; }
+ bool isOffloadedOrDirect_l() const
+ { return (mFlags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|
+ AUDIO_OUTPUT_FLAG_DIRECT)) != 0; }
+
+ bool isDirect_l() const
+ { return (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0; }
+
// Next 4 fields may be changed if IAudioTrack is re-created, but always != 0
sp<IAudioTrack> mAudioTrack;
sp<IMemory> mCblkMemory;
@@ -667,6 +684,7 @@
transfer_type mTransfer;
audio_offload_info_t mOffloadInfoCopy;
const audio_offload_info_t* mOffloadInfo;
+ audio_attributes_t mAttributes;
// mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. For 8-bit PCM data, it's
// twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer.
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index c742810..a8f4605 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -50,6 +50,7 @@
TRACK_TIMED = 1, // client requests a TimedAudioTrack
TRACK_FAST = 2, // client requests a fast AudioTrack or AudioRecord
TRACK_OFFLOAD = 4, // client requests offload to hw codec
+ TRACK_DIRECT = 8, // client requests a direct output
};
typedef uint32_t track_flags_t;
@@ -88,6 +89,7 @@
track_flags_t *flags,
pid_t tid, // -1 means unused, otherwise must be valid non-0
int *sessionId,
+ size_t *notificationFrames,
sp<IMemory>& cblk,
sp<IMemory>& buffers, // return value 0 means it follows cblk
status_t *status) = 0;
@@ -96,9 +98,6 @@
* and therefore can be cached.
*/
virtual uint32_t sampleRate(audio_io_handle_t output) const = 0;
-#if 0
- virtual int channelCount(audio_io_handle_t output) const = 0;
-#endif
virtual audio_format_t format(audio_io_handle_t output) const = 0;
virtual size_t frameCount(audio_io_handle_t output) const = 0;
diff --git a/include/media/IAudioPolicyService.h b/include/media/IAudioPolicyService.h
index d422aa3..959e4c3 100644
--- a/include/media/IAudioPolicyService.h
+++ b/include/media/IAudioPolicyService.h
@@ -56,6 +56,12 @@
audio_channel_mask_t channelMask = 0,
audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
const audio_offload_info_t *offloadInfo = NULL) = 0;
+ virtual audio_io_handle_t getOutputForAttr(const audio_attributes_t *attr,
+ uint32_t samplingRate = 0,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
+ audio_channel_mask_t channelMask = 0,
+ audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+ const audio_offload_info_t *offloadInfo = NULL) = 0;
virtual status_t startOutput(audio_io_handle_t output,
audio_stream_type_t stream,
int session = 0) = 0;
diff --git a/include/media/mediaplayer.h b/include/media/mediaplayer.h
index 3ca3095..e756368 100644
--- a/include/media/mediaplayer.h
+++ b/include/media/mediaplayer.h
@@ -160,6 +160,9 @@
// 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
+
+ // Set a Parcel containing the value of a parcelled Java AudioAttribute instance
+ KEY_PARAMETER_AUDIO_ATTRIBUTES = 1400 // set only
};
// Keep INVOKE_ID_* in sync with MediaPlayer.java.
@@ -169,7 +172,7 @@
INVOKE_ID_ADD_EXTERNAL_SOURCE_FD = 3,
INVOKE_ID_SELECT_TRACK = 4,
INVOKE_ID_UNSELECT_TRACK = 5,
- INVOKE_ID_SET_VIDEO_SCALING_MODE = 6,
+ INVOKE_ID_SET_VIDEO_SCALING_MODE = 6
};
// Keep MEDIA_TRACK_TYPE_* in sync with MediaPlayer.java.
@@ -259,6 +262,7 @@
status_t attachNewPlayer(const sp<IMediaPlayer>& player);
status_t reset_l();
status_t doSetRetransmitEndpoint(const sp<IMediaPlayer>& player);
+ status_t checkStateForKeySet_l(int key);
sp<IMediaPlayer> mPlayer;
thread_id_t mLockThreadId;
diff --git a/include/media/stagefright/ACodec.h b/include/media/stagefright/ACodec.h
index 8ec7f1c..142b7cb 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 <media/stagefright/CodecBase.h>
#include <media/stagefright/SkipCutBuffer.h>
#include <OMX_Audio.h>
@@ -32,42 +33,32 @@
struct ABuffer;
struct MemoryDealer;
-struct ACodec : public AHierarchicalStateMachine {
- enum {
- kWhatFillThisBuffer = 'fill',
- kWhatDrainThisBuffer = 'drai',
- kWhatEOS = 'eos ',
- kWhatShutdownCompleted = 'scom',
- kWhatFlushCompleted = 'fcom',
- kWhatOutputFormatChanged = 'outC',
- kWhatError = 'erro',
- kWhatComponentAllocated = 'cAll',
- kWhatComponentConfigured = 'cCon',
- kWhatInputSurfaceCreated = 'isfc',
- kWhatSignaledInputEOS = 'seos',
- kWhatBuffersAllocated = 'allc',
- kWhatOMXDied = 'OMXd',
- };
-
+struct ACodec : public AHierarchicalStateMachine, public CodecBase {
ACodec();
- void setNotificationMessage(const sp<AMessage> &msg);
+ virtual void setNotificationMessage(const sp<AMessage> &msg);
+
void initiateSetup(const sp<AMessage> &msg);
- void signalFlush();
- void signalResume();
- void initiateShutdown(bool keepComponentAllocated = false);
- void signalSetParameters(const sp<AMessage> &msg);
- void signalEndOfInputStream();
+ virtual void initiateAllocateComponent(const sp<AMessage> &msg);
+ virtual void initiateConfigureComponent(const sp<AMessage> &msg);
+ virtual void initiateCreateInputSurface();
+ virtual void initiateStart();
+ virtual void initiateShutdown(bool keepComponentAllocated = false);
- void initiateAllocateComponent(const sp<AMessage> &msg);
- void initiateConfigureComponent(const sp<AMessage> &msg);
- void initiateCreateInputSurface();
- void initiateStart();
+ virtual void signalFlush();
+ virtual void signalResume();
- void signalRequestIDRFrame();
+ virtual void signalSetParameters(const sp<AMessage> &msg);
+ virtual void signalEndOfInputStream();
+ virtual void signalRequestIDRFrame();
- struct PortDescription : public RefBase {
+ // AHierarchicalStateMachine implements the message handling
+ virtual void onMessageReceived(const sp<AMessage> &msg) {
+ handleMessage(msg);
+ }
+
+ struct PortDescription : public CodecBase::PortDescription {
size_t countBuffers();
IOMX::buffer_id bufferIDAt(size_t index) const;
sp<ABuffer> bufferAt(size_t index) const;
@@ -117,6 +108,7 @@
kWhatRequestIDRFrame = 'ridr',
kWhatSetParameters = 'setP',
kWhatSubmitOutputMetaDataBufferIfEOS = 'subm',
+ kWhatOMXDied = 'OMXd',
};
enum {
@@ -278,6 +270,7 @@
status_t setupMPEG4EncoderParameters(const sp<AMessage> &msg);
status_t setupH263EncoderParameters(const sp<AMessage> &msg);
status_t setupAVCEncoderParameters(const sp<AMessage> &msg);
+ status_t setupHEVCEncoderParameters(const sp<AMessage> &msg);
status_t setupVPXEncoderParameters(const sp<AMessage> &msg);
status_t verifySupportForProfileAndLevel(int32_t profile, int32_t level);
diff --git a/include/media/stagefright/CodecBase.h b/include/media/stagefright/CodecBase.h
new file mode 100644
index 0000000..1bf27a6
--- /dev/null
+++ b/include/media/stagefright/CodecBase.h
@@ -0,0 +1,88 @@
+/*
+ * Copyright (C) 2014 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 CODEC_BASE_H_
+
+#define CODEC_BASE_H_
+
+#include <stdint.h>
+#include <media/IOMX.h>
+
+#include <media/stagefright/foundation/AHandler.h>
+
+namespace android {
+
+struct ABuffer;
+
+struct CodecBase : public AHandler {
+ enum {
+ kWhatFillThisBuffer = 'fill',
+ kWhatDrainThisBuffer = 'drai',
+ kWhatEOS = 'eos ',
+ kWhatShutdownCompleted = 'scom',
+ kWhatFlushCompleted = 'fcom',
+ kWhatOutputFormatChanged = 'outC',
+ kWhatError = 'erro',
+ kWhatComponentAllocated = 'cAll',
+ kWhatComponentConfigured = 'cCon',
+ kWhatInputSurfaceCreated = 'isfc',
+ kWhatSignaledInputEOS = 'seos',
+ kWhatBuffersAllocated = 'allc',
+ };
+
+ virtual void setNotificationMessage(const sp<AMessage> &msg) = 0;
+
+ virtual void initiateAllocateComponent(const sp<AMessage> &msg) = 0;
+ virtual void initiateConfigureComponent(const sp<AMessage> &msg) = 0;
+ virtual void initiateCreateInputSurface() = 0;
+ virtual void initiateStart() = 0;
+ virtual void initiateShutdown(bool keepComponentAllocated = false) = 0;
+
+ // require an explicit message handler
+ virtual void onMessageReceived(const sp<AMessage> &msg) = 0;
+
+ virtual void signalFlush() = 0;
+ virtual void signalResume() = 0;
+
+ virtual void signalRequestIDRFrame() = 0;
+ virtual void signalSetParameters(const sp<AMessage> &msg) = 0;
+ virtual void signalEndOfInputStream() = 0;
+
+ struct PortDescription : public RefBase {
+ virtual size_t countBuffers() = 0;
+ virtual IOMX::buffer_id bufferIDAt(size_t index) const = 0;
+ virtual sp<ABuffer> bufferAt(size_t index) const = 0;
+
+ protected:
+ PortDescription();
+ virtual ~PortDescription();
+
+ private:
+ DISALLOW_EVIL_CONSTRUCTORS(PortDescription);
+ };
+
+protected:
+ CodecBase();
+ virtual ~CodecBase();
+
+private:
+ DISALLOW_EVIL_CONSTRUCTORS(CodecBase);
+};
+
+} // namespace android
+
+#endif // CODEC_BASE_H_
+
diff --git a/include/media/stagefright/MediaCodec.h b/include/media/stagefright/MediaCodec.h
index a1e32c9..3e6eefb 100644
--- a/include/media/stagefright/MediaCodec.h
+++ b/include/media/stagefright/MediaCodec.h
@@ -26,9 +26,9 @@
namespace android {
struct ABuffer;
-struct ACodec;
struct AMessage;
struct AString;
+struct CodecBase;
struct ICrypto;
struct SoftwareRenderer;
struct Surface;
@@ -195,7 +195,7 @@
State mState;
sp<ALooper> mLooper;
sp<ALooper> mCodecLooper;
- sp<ACodec> mCodec;
+ sp<CodecBase> mCodec;
AString mComponentName;
uint32_t mReplyID;
uint32_t mFlags;
diff --git a/include/media/stagefright/foundation/ABitReader.h b/include/media/stagefright/foundation/ABitReader.h
index 5510b12..c3bf0ff 100644
--- a/include/media/stagefright/foundation/ABitReader.h
+++ b/include/media/stagefright/foundation/ABitReader.h
@@ -25,8 +25,10 @@
namespace android {
-struct ABitReader {
+class ABitReader {
+public:
ABitReader(const uint8_t *data, size_t size);
+ virtual ~ABitReader();
uint32_t getBits(size_t n);
void skipBits(size_t n);
@@ -37,18 +39,32 @@
const uint8_t *data() const;
-private:
+protected:
const uint8_t *mData;
size_t mSize;
uint32_t mReservoir; // left-aligned bits
size_t mNumBitsLeft;
- void fillReservoir();
+ virtual void fillReservoir();
DISALLOW_EVIL_CONSTRUCTORS(ABitReader);
};
+class NALBitReader : public ABitReader {
+public:
+ NALBitReader(const uint8_t *data, size_t size);
+
+ bool atLeastNumBitsLeft(size_t n) const;
+
+private:
+ int32_t mNumZeros;
+
+ virtual void fillReservoir();
+
+ DISALLOW_EVIL_CONSTRUCTORS(NALBitReader);
+};
+
} // namespace android
#endif // A_BIT_READER_H_
diff --git a/include/media/stagefright/foundation/AHierarchicalStateMachine.h b/include/media/stagefright/foundation/AHierarchicalStateMachine.h
index d2e6b28..3bb7d75 100644
--- a/include/media/stagefright/foundation/AHierarchicalStateMachine.h
+++ b/include/media/stagefright/foundation/AHierarchicalStateMachine.h
@@ -43,13 +43,13 @@
DISALLOW_EVIL_CONSTRUCTORS(AState);
};
-struct AHierarchicalStateMachine : public AHandler {
+struct AHierarchicalStateMachine {
AHierarchicalStateMachine();
protected:
virtual ~AHierarchicalStateMachine();
- virtual void onMessageReceived(const sp<AMessage> &msg);
+ virtual void handleMessage(const sp<AMessage> &msg);
// Only to be called in response to a message.
void changeState(const sp<AState> &state);
diff --git a/include/media/stagefright/foundation/AString.h b/include/media/stagefright/foundation/AString.h
index 622028e..0edaa1c 100644
--- a/include/media/stagefright/foundation/AString.h
+++ b/include/media/stagefright/foundation/AString.h
@@ -77,6 +77,8 @@
bool startsWith(const char *prefix) const;
bool endsWith(const char *suffix) const;
+ bool startsWithIgnoreCase(const char *prefix) const;
+ bool endsWithIgnoreCase(const char *suffix) const;
void tolower();
diff --git a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
index db5c78f..695767d 100644
--- a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
+++ b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
@@ -19,11 +19,13 @@
#define ARRAY_SIZE(array) (sizeof array / sizeof array[0])
//#define LOG_NDEBUG 0
-#include <cutils/log.h>
#include <assert.h>
+#include <inttypes.h>
+#include <new>
#include <stdlib.h>
#include <string.h>
-#include <new>
+
+#include <cutils/log.h>
#include "EffectBundle.h"
@@ -560,11 +562,12 @@
MemTab.Region[i].pBaseAddress = malloc(MemTab.Region[i].Size);
if (MemTab.Region[i].pBaseAddress == LVM_NULL){
- ALOGV("\tLVM_ERROR :LvmBundle_init CreateInstance Failed to allocate %ld bytes "
- "for region %u\n", MemTab.Region[i].Size, i );
+ ALOGV("\tLVM_ERROR :LvmBundle_init CreateInstance Failed to allocate %" PRIu32
+ " bytes for region %u\n", MemTab.Region[i].Size, i );
bMallocFailure = LVM_TRUE;
}else{
- ALOGV("\tLvmBundle_init CreateInstance allocated %ld bytes for region %u at %p\n",
+ ALOGV("\tLvmBundle_init CreateInstance allocated %" PRIu32
+ " bytes for region %u at %p\n",
MemTab.Region[i].Size, i, MemTab.Region[i].pBaseAddress);
}
}
@@ -576,11 +579,11 @@
if(bMallocFailure == LVM_TRUE){
for (int i=0; i<LVM_NR_MEMORY_REGIONS; i++){
if (MemTab.Region[i].pBaseAddress == LVM_NULL){
- ALOGV("\tLVM_ERROR :LvmBundle_init CreateInstance Failed to allocate %ld bytes "
- "for region %u Not freeing\n", MemTab.Region[i].Size, i );
+ ALOGV("\tLVM_ERROR :LvmBundle_init CreateInstance Failed to allocate %" PRIu32
+ " bytes for region %u Not freeing\n", MemTab.Region[i].Size, i );
}else{
- ALOGV("\tLVM_ERROR :LvmBundle_init CreateInstance Failed: but allocated %ld bytes "
- "for region %u at %p- free\n",
+ ALOGV("\tLVM_ERROR :LvmBundle_init CreateInstance Failed: but allocated %" PRIu32
+ " bytes for region %u at %p- free\n",
MemTab.Region[i].Size, i, MemTab.Region[i].pBaseAddress);
free(MemTab.Region[i].pBaseAddress);
}
@@ -889,16 +892,16 @@
for (int i=0; i<LVM_NR_MEMORY_REGIONS; i++){
if (MemTab.Region[i].Size != 0){
if (MemTab.Region[i].pBaseAddress != NULL){
- ALOGV("\tLvmEffect_free - START freeing %ld bytes for region %u at %p\n",
+ ALOGV("\tLvmEffect_free - START freeing %" PRIu32 " bytes for region %u at %p\n",
MemTab.Region[i].Size, i, MemTab.Region[i].pBaseAddress);
free(MemTab.Region[i].pBaseAddress);
- ALOGV("\tLvmEffect_free - END freeing %ld bytes for region %u at %p\n",
+ ALOGV("\tLvmEffect_free - END freeing %" PRIu32 " bytes for region %u at %p\n",
MemTab.Region[i].Size, i, MemTab.Region[i].pBaseAddress);
}else{
- ALOGV("\tLVM_ERROR : LvmEffect_free - trying to free with NULL pointer %ld bytes "
- "for region %u at %p ERROR\n",
+ ALOGV("\tLVM_ERROR : LvmEffect_free - trying to free with NULL pointer %" PRIu32
+ " bytes for region %u at %p ERROR\n",
MemTab.Region[i].Size, i, MemTab.Region[i].pBaseAddress);
}
}
diff --git a/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp b/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp
index c6d3759..13f1a0d 100644
--- a/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp
+++ b/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp
@@ -19,11 +19,13 @@
#define ARRAY_SIZE(array) (sizeof array / sizeof array[0])
//#define LOG_NDEBUG 0
-#include <cutils/log.h>
#include <assert.h>
+#include <inttypes.h>
+#include <new>
#include <stdlib.h>
#include <string.h>
-#include <new>
+
+#include <cutils/log.h>
#include "EffectReverb.h"
// from Reverb/lib
#include "LVREV.h"
@@ -269,7 +271,7 @@
pContext->InFrames32 = (LVM_INT32 *)malloc(LVREV_MAX_FRAME_SIZE * sizeof(LVM_INT32) * 2);
pContext->OutFrames32 = (LVM_INT32 *)malloc(LVREV_MAX_FRAME_SIZE * sizeof(LVM_INT32) * 2);
- ALOGV("\tEffectCreate %p, size %d", pContext, sizeof(ReverbContext));
+ ALOGV("\tEffectCreate %p, size %zu", pContext, sizeof(ReverbContext));
ALOGV("\tEffectCreate end\n");
return 0;
} /* end EffectCreate */
@@ -570,15 +572,15 @@
for (int i=0; i<LVM_NR_MEMORY_REGIONS; i++){
if (MemTab.Region[i].Size != 0){
if (MemTab.Region[i].pBaseAddress != NULL){
- ALOGV("\tfree() - START freeing %ld bytes for region %u at %p\n",
+ ALOGV("\tfree() - START freeing %" PRIu32 " bytes for region %u at %p\n",
MemTab.Region[i].Size, i, MemTab.Region[i].pBaseAddress);
free(MemTab.Region[i].pBaseAddress);
- ALOGV("\tfree() - END freeing %ld bytes for region %u at %p\n",
+ ALOGV("\tfree() - END freeing %" PRIu32 " bytes for region %u at %p\n",
MemTab.Region[i].Size, i, MemTab.Region[i].pBaseAddress);
}else{
- ALOGV("\tLVM_ERROR : free() - trying to free with NULL pointer %ld bytes "
+ ALOGV("\tLVM_ERROR : free() - trying to free with NULL pointer %" PRIu32 " bytes "
"for region %u at %p ERROR\n",
MemTab.Region[i].Size, i, MemTab.Region[i].pBaseAddress);
}
@@ -771,11 +773,12 @@
MemTab.Region[i].pBaseAddress = malloc(MemTab.Region[i].Size);
if (MemTab.Region[i].pBaseAddress == LVM_NULL){
- ALOGV("\tLVREV_ERROR :Reverb_init CreateInstance Failed to allocate %ld "
- "bytes for region %u\n", MemTab.Region[i].Size, i );
+ ALOGV("\tLVREV_ERROR :Reverb_init CreateInstance Failed to allocate %" PRIu32
+ " bytes for region %u\n", MemTab.Region[i].Size, i );
bMallocFailure = LVM_TRUE;
}else{
- ALOGV("\tReverb_init CreateInstance allocate %ld bytes for region %u at %p\n",
+ ALOGV("\tReverb_init CreateInstance allocate %" PRIu32
+ " bytes for region %u at %p\n",
MemTab.Region[i].Size, i, MemTab.Region[i].pBaseAddress);
}
}
@@ -787,11 +790,11 @@
if(bMallocFailure == LVM_TRUE){
for (int i=0; i<LVM_NR_MEMORY_REGIONS; i++){
if (MemTab.Region[i].pBaseAddress == LVM_NULL){
- ALOGV("\tLVM_ERROR :Reverb_init CreateInstance Failed to allocate %ld bytes "
- "for region %u - Not freeing\n", MemTab.Region[i].Size, i );
+ ALOGV("\tLVM_ERROR :Reverb_init CreateInstance Failed to allocate %" PRIu32
+ " bytes for region %u - Not freeing\n", MemTab.Region[i].Size, i );
}else{
- ALOGV("\tLVM_ERROR :Reverb_init CreateInstance Failed: but allocated %ld bytes "
- "for region %u at %p- free\n",
+ ALOGV("\tLVM_ERROR :Reverb_init CreateInstance Failed: but allocated %" PRIu32
+ " bytes for region %u at %p- free\n",
MemTab.Region[i].Size, i, MemTab.Region[i].pBaseAddress);
free(MemTab.Region[i].pBaseAddress);
}
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 50b444a..3ee5809 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -18,7 +18,9 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "AudioRecord"
+#include <inttypes.h>
#include <sys/resource.h>
+
#include <binder/IPCThreadState.h>
#include <media/AudioRecord.h>
#include <utils/Log.h>
@@ -468,7 +470,7 @@
if (frameCount == 0) {
frameCount = minFrameCount;
} else if (frameCount < minFrameCount) {
- ALOGE("frameCount %u < minFrameCount %u", frameCount, minFrameCount);
+ ALOGE("frameCount %zu < minFrameCount %zu", frameCount, minFrameCount);
return BAD_VALUE;
}
@@ -493,6 +495,10 @@
size_t temp = frameCount; // temp may be replaced by a revised value of frameCount,
// but we will still need the original value also
int originalSessionId = mSessionId;
+
+ // The notification frame count is the period between callbacks, as suggested by the server.
+ size_t notificationFrames;
+
sp<IMemory> iMem; // for cblk
sp<IMemory> bufferMem;
sp<IAudioRecord> record = audioFlinger->openRecord(input,
@@ -502,6 +508,7 @@
&trackFlags,
tid,
&mSessionId,
+ ¬ificationFrames,
iMem,
bufferMem,
&status);
@@ -555,17 +562,17 @@
mCblk = cblk;
// note that temp is the (possibly revised) value of frameCount
if (temp < frameCount || (frameCount == 0 && temp == 0)) {
- ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp);
+ ALOGW("Requested frameCount %zu but received frameCount %zu", frameCount, temp);
}
frameCount = temp;
mAwaitBoost = false;
if (mFlags & AUDIO_INPUT_FLAG_FAST) {
if (trackFlags & IAudioFlinger::TRACK_FAST) {
- ALOGV("AUDIO_INPUT_FLAG_FAST successful; frameCount %u", frameCount);
+ ALOGV("AUDIO_INPUT_FLAG_FAST successful; frameCount %zu", frameCount);
mAwaitBoost = true;
} else {
- ALOGV("AUDIO_INPUT_FLAG_FAST denied by server; frameCount %u", frameCount);
+ ALOGV("AUDIO_INPUT_FLAG_FAST denied by server; frameCount %zu", frameCount);
// once denied, do not request again if IAudioRecord is re-created
mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST);
}
@@ -740,7 +747,7 @@
if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
// sanity-check. user is most-likely passing an error code, and it would
// make the return value ambiguous (actualSize vs error).
- ALOGE("AudioRecord::read(buffer=%p, size=%u (%d)", buffer, userSize, userSize);
+ ALOGE("AudioRecord::read(buffer=%p, size=%zu (%zu)", buffer, userSize, userSize);
return BAD_VALUE;
}
@@ -921,10 +928,10 @@
size_t nonContig;
status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig);
LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0),
- "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount);
+ "obtainBuffer() err=%d frameCount=%zu", err, audioBuffer.frameCount);
requested = &ClientProxy::kNonBlocking;
size_t avail = audioBuffer.frameCount + nonContig;
- ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d",
+ ALOGV("obtainBuffer(%u) returned %zu = %zu + %zu err %d",
mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
if (err != NO_ERROR) {
if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) {
@@ -952,8 +959,8 @@
// Sanity check on returned size
if (ssize_t(readSize) < 0 || readSize > reqSize) {
- ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes",
- reqSize, (int) readSize);
+ ALOGE("EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes",
+ reqSize, ssize_t(readSize));
return NS_NEVER;
}
@@ -1092,7 +1099,7 @@
ns = 1000000000LL;
// fall through
default:
- LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns);
+ LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns);
pauseInternal(ns);
return true;
}
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 15b32ff..a47d45c 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -245,6 +245,19 @@
return getSamplingRate(output, samplingRate);
}
+status_t AudioSystem::getOutputSamplingRateForAttr(uint32_t* samplingRate,
+ const audio_attributes_t *attr)
+{
+ if (attr == NULL) {
+ return BAD_VALUE;
+ }
+ audio_io_handle_t output = getOutputForAttr(attr);
+ if (output == 0) {
+ return PERMISSION_DENIED;
+ }
+ return getSamplingRate(output, samplingRate);
+}
+
status_t AudioSystem::getSamplingRate(audio_io_handle_t output,
uint32_t* samplingRate)
{
@@ -310,7 +323,7 @@
return BAD_VALUE;
}
- ALOGV("getFrameCount() output %d, frameCount %d", output, *frameCount);
+ ALOGV("getFrameCount() output %d, frameCount %zu", output, *frameCount);
return NO_ERROR;
}
@@ -476,7 +489,7 @@
OutputDescriptor *outputDesc = new OutputDescriptor(*desc);
gOutputs.add(ioHandle, outputDesc);
- ALOGV("ioConfigChanged() new output samplingRate %u, format %#x channel mask %#x frameCount %u "
+ ALOGV("ioConfigChanged() new output samplingRate %u, format %#x channel mask %#x frameCount %zu "
"latency %d",
outputDesc->samplingRate, outputDesc->format, outputDesc->channelMask,
outputDesc->frameCount, outputDesc->latency);
@@ -501,7 +514,7 @@
desc = (const OutputDescriptor *)param2;
ALOGV("ioConfigChanged() new config for output %d samplingRate %u, format %#x channel mask %#x "
- "frameCount %d latency %d",
+ "frameCount %zu latency %d",
ioHandle, desc->samplingRate, desc->format,
desc->channelMask, desc->frameCount, desc->latency);
OutputDescriptor *outputDesc = gOutputs.valueAt(index);
@@ -633,6 +646,19 @@
return aps->getOutput(stream, samplingRate, format, channelMask, flags, offloadInfo);
}
+audio_io_handle_t AudioSystem::getOutputForAttr(const audio_attributes_t *attr,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo)
+{
+ if (attr == NULL) return 0;
+ const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+ if (aps == 0) return 0;
+ return aps->getOutputForAttr(attr, samplingRate, format, channelMask, flags, offloadInfo);
+}
+
status_t AudioSystem::startOutput(audio_io_handle_t output,
audio_stream_type_t stream,
int session)
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index e6827ee..b5c9125 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -15,12 +15,13 @@
** limitations under the License.
*/
-
//#define LOG_NDEBUG 0
#define LOG_TAG "AudioTrack"
+#include <inttypes.h>
#include <math.h>
#include <sys/resource.h>
+
#include <audio_utils/primitives.h>
#include <binder/IPCThreadState.h>
#include <media/AudioTrack.h>
@@ -89,7 +90,7 @@
streamType, sampleRate);
return BAD_VALUE;
}
- ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d",
+ ALOGV("getMinFrameCount=%zu: afFrameCount=%zu, minBufCount=%d, afSampleRate=%d, afLatency=%d",
*frameCount, afFrameCount, minBufCount, afSampleRate, afLatency);
return NO_ERROR;
}
@@ -103,6 +104,10 @@
mPreviousSchedulingGroup(SP_DEFAULT),
mPausedPosition(0)
{
+ mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN;
+ mAttributes.usage = AUDIO_USAGE_UNKNOWN;
+ mAttributes.flags = 0x0;
+ strcpy(mAttributes.tags, "");
}
AudioTrack::AudioTrack(
@@ -119,7 +124,8 @@
transfer_type transferType,
const audio_offload_info_t *offloadInfo,
int uid,
- pid_t pid)
+ pid_t pid,
+ const audio_attributes_t* pAttributes)
: mStatus(NO_INIT),
mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
@@ -129,7 +135,7 @@
mStatus = set(streamType, sampleRate, format, channelMask,
frameCount, flags, cbf, user, notificationFrames,
0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType,
- offloadInfo, uid, pid);
+ offloadInfo, uid, pid, pAttributes);
}
AudioTrack::AudioTrack(
@@ -146,7 +152,8 @@
transfer_type transferType,
const audio_offload_info_t *offloadInfo,
int uid,
- pid_t pid)
+ pid_t pid,
+ const audio_attributes_t* pAttributes)
: mStatus(NO_INIT),
mIsTimed(false),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
@@ -156,7 +163,7 @@
mStatus = set(streamType, sampleRate, format, channelMask,
0 /*frameCount*/, flags, cbf, user, notificationFrames,
sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo,
- uid, pid);
+ uid, pid, pAttributes);
}
AudioTrack::~AudioTrack()
@@ -199,7 +206,8 @@
transfer_type transferType,
const audio_offload_info_t *offloadInfo,
int uid,
- pid_t pid)
+ pid_t pid,
+ const audio_attributes_t* pAttributes)
{
ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
"flags #%x, notificationFrames %u, sessionId %d, transferType %d",
@@ -245,7 +253,7 @@
ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(),
sharedBuffer->size());
- ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags);
+ ALOGV("set() streamType %d frameCount %zu flags %04x", streamType, frameCount, flags);
AutoMutex lock(mLock);
@@ -259,18 +267,33 @@
if (streamType == AUDIO_STREAM_DEFAULT) {
streamType = AUDIO_STREAM_MUSIC;
}
- if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
- ALOGE("Invalid stream type %d", streamType);
- return BAD_VALUE;
+
+ if (pAttributes == NULL) {
+ if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
+ ALOGE("Invalid stream type %d", streamType);
+ return BAD_VALUE;
+ }
+ setAttributesFromStreamType(streamType);
+ mStreamType = streamType;
+ } else {
+ if (!isValidAttributes(pAttributes)) {
+ ALOGE("Invalid attributes: usage=%d content=%d flags=0x%x tags=[%s]",
+ pAttributes->usage, pAttributes->content_type, pAttributes->flags,
+ pAttributes->tags);
+ }
+ // stream type shouldn't be looked at, this track has audio attributes
+ memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t));
+ setStreamTypeFromAttributes(mAttributes);
+ ALOGV("Building AudioTrack with attributes: usage=%d content=%d flags=0x%x tags=[%s]",
+ mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags);
}
- mStreamType = streamType;
status_t status;
if (sampleRate == 0) {
- status = AudioSystem::getOutputSamplingRate(&sampleRate, streamType);
+ status = AudioSystem::getOutputSamplingRateForAttr(&sampleRate, &mAttributes);
if (status != NO_ERROR) {
ALOGE("Could not get output sample rate for stream type %d; status %d",
- streamType, status);
+ mStreamType, status);
return status;
}
}
@@ -314,7 +337,7 @@
((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST);
}
// only allow deep buffering for music stream type
- if (streamType != AUDIO_STREAM_MUSIC) {
+ if (mStreamType != AUDIO_STREAM_MUSIC) {
flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
}
@@ -615,12 +638,12 @@
status_t AudioTrack::setSampleRate(uint32_t rate)
{
- if (mIsTimed || isOffloaded()) {
+ if (mIsTimed || isOffloadedOrDirect()) {
return INVALID_OPERATION;
}
uint32_t afSamplingRate;
- if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) {
+ if (AudioSystem::getOutputSamplingRateForAttr(&afSamplingRate, &mAttributes) != NO_ERROR) {
return NO_INIT;
}
// Resampler implementation limits input sampling rate to 2 x output sampling rate.
@@ -646,7 +669,7 @@
// sample rate can be updated during playback by the offloaded decoder so we need to
// query the HAL and update if needed.
// FIXME use Proxy return channel to update the rate from server and avoid polling here
- if (isOffloaded_l()) {
+ if (isOffloadedOrDirect_l()) {
if (mOutput != AUDIO_IO_HANDLE_NONE) {
uint32_t sampleRate = 0;
status_t status = AudioSystem::getSamplingRate(mOutput, &sampleRate);
@@ -660,7 +683,7 @@
status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
{
- if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) {
+ if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) {
return INVALID_OPERATION;
}
@@ -694,7 +717,7 @@
status_t AudioTrack::setMarkerPosition(uint32_t marker)
{
// The only purpose of setting marker position is to get a callback
- if (mCbf == NULL || isOffloaded()) {
+ if (mCbf == NULL || isOffloadedOrDirect()) {
return INVALID_OPERATION;
}
@@ -707,7 +730,7 @@
status_t AudioTrack::getMarkerPosition(uint32_t *marker) const
{
- if (isOffloaded()) {
+ if (isOffloadedOrDirect()) {
return INVALID_OPERATION;
}
if (marker == NULL) {
@@ -723,7 +746,7 @@
status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
{
// The only purpose of setting position update period is to get a callback
- if (mCbf == NULL || isOffloaded()) {
+ if (mCbf == NULL || isOffloadedOrDirect()) {
return INVALID_OPERATION;
}
@@ -736,7 +759,7 @@
status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const
{
- if (isOffloaded()) {
+ if (isOffloadedOrDirect()) {
return INVALID_OPERATION;
}
if (updatePeriod == NULL) {
@@ -751,7 +774,7 @@
status_t AudioTrack::setPosition(uint32_t position)
{
- if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) {
+ if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) {
return INVALID_OPERATION;
}
if (position > mFrameCount) {
@@ -784,10 +807,10 @@
}
AutoMutex lock(mLock);
- if (isOffloaded_l()) {
+ if (isOffloadedOrDirect_l()) {
uint32_t dspFrames = 0;
- if ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING)) {
+ if (isOffloaded_l() && ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING))) {
ALOGV("getPosition called in paused state, return cached position %u", mPausedPosition);
*position = mPausedPosition;
return NO_ERROR;
@@ -822,7 +845,7 @@
status_t AudioTrack::reload()
{
- if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) {
+ if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) {
return INVALID_OPERATION;
}
@@ -867,12 +890,12 @@
return NO_INIT;
}
- audio_io_handle_t output = AudioSystem::getOutput(mStreamType, mSampleRate, mFormat,
+ audio_io_handle_t output = AudioSystem::getOutputForAttr(&mAttributes, mSampleRate, mFormat,
mChannelMask, mFlags, mOffloadInfo);
if (output == AUDIO_IO_HANDLE_NONE) {
- ALOGE("Could not get audio output for stream type %d, sample rate %u, format %#x, "
- "channel mask %#x, flags %#x",
- mStreamType, mSampleRate, mFormat, mChannelMask, mFlags);
+ ALOGE("Could not get audio output for stream type %d, usage %d, sample rate %u, format %#x,"
+ " channel mask %#x, flags %#x",
+ mStreamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask, mFlags);
return BAD_VALUE;
}
{
@@ -973,14 +996,14 @@
// Ensure that buffer depth covers at least audio hardware latency
uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
- ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d",
+ ALOGV("afFrameCount=%zu, minBufCount=%d, afSampleRate=%u, afLatency=%d",
afFrameCount, minBufCount, afSampleRate, afLatency);
if (minBufCount <= nBuffering) {
minBufCount = nBuffering;
}
size_t minFrameCount = (afFrameCount*mSampleRate*minBufCount)/afSampleRate;
- ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u"
+ ALOGV("minFrameCount: %zu, afFrameCount=%zu, minBufCount=%d, sampleRate=%u, afSampleRate=%u"
", afLatency=%d",
minFrameCount, afFrameCount, minBufCount, mSampleRate, afSampleRate, afLatency);
@@ -988,7 +1011,7 @@
frameCount = minFrameCount;
} else if (frameCount < minFrameCount) {
// not ALOGW because it happens all the time when playing key clicks over A2DP
- ALOGV("Minimum buffer size corrected from %d to %d",
+ ALOGV("Minimum buffer size corrected from %zu to %zu",
frameCount, minFrameCount);
frameCount = minFrameCount;
}
@@ -1018,6 +1041,10 @@
trackFlags |= IAudioFlinger::TRACK_OFFLOAD;
}
+ if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
+ trackFlags |= IAudioFlinger::TRACK_DIRECT;
+ }
+
size_t temp = frameCount; // temp may be replaced by a revised value of frameCount,
// but we will still need the original value also
sp<IAudioTrack> track = audioFlinger->createTrack(mStreamType,
@@ -1071,14 +1098,14 @@
// In current design, AudioTrack client checks and ensures frame count validity before
// passing it to AudioFlinger so AudioFlinger should not return a different value except
// for fast track as it uses a special method of assigning frame count.
- ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp);
+ ALOGW("Requested frameCount %zu but received frameCount %zu", frameCount, temp);
}
frameCount = temp;
mAwaitBoost = false;
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
if (trackFlags & IAudioFlinger::TRACK_FAST) {
- ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount);
+ ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu", frameCount);
mAwaitBoost = true;
if (mSharedBuffer == 0) {
// Theoretically double-buffering is not required for fast tracks,
@@ -1089,7 +1116,7 @@
}
}
} else {
- ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount);
+ ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu", frameCount);
// once denied, do not request again if IAudioTrack is re-created
mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
if (mSharedBuffer == 0) {
@@ -1109,6 +1136,16 @@
//return NO_INIT;
}
}
+ if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
+ if (trackFlags & IAudioFlinger::TRACK_DIRECT) {
+ ALOGV("AUDIO_OUTPUT_FLAG_DIRECT successful");
+ } else {
+ ALOGW("AUDIO_OUTPUT_FLAG_DIRECT denied by server");
+ mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_DIRECT);
+ // FIXME This is a warning, not an error, so don't return error status
+ //return NO_INIT;
+ }
+ }
// We retain a copy of the I/O handle, but don't own the reference
mOutput = output;
@@ -1304,6 +1341,16 @@
return INVALID_OPERATION;
}
+ if (isDirect()) {
+ AutoMutex lock(mLock);
+ int32_t flags = android_atomic_and(
+ ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
+ &mCblk->mFlags);
+ if (flags & CBLK_INVALID) {
+ return DEAD_OBJECT;
+ }
+ }
+
if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
// Sanity-check: user is most-likely passing an error code, and it would
// make the return value ambiguous (actualSize vs error).
@@ -1452,7 +1499,7 @@
// for offloaded tracks restoreTrack_l() will just update the sequence and clear
// AudioSystem cache. We should not exit here but after calling the callback so
// that the upper layers can recreate the track
- if (!isOffloaded_l() || (mSequence == mObservedSequence)) {
+ if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) {
status_t status = restoreTrack_l("processAudioBuffer");
mLock.unlock();
// Run again immediately, but with a new IAudioTrack
@@ -1578,7 +1625,7 @@
mObservedSequence = sequence;
mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL);
// for offloaded tracks, just wait for the upper layers to recreate the track
- if (isOffloaded()) {
+ if (isOffloadedOrDirect()) {
return NS_INACTIVE;
}
}
@@ -1636,10 +1683,10 @@
size_t nonContig;
status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig);
LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0),
- "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount);
+ "obtainBuffer() err=%d frameCount=%zu", err, audioBuffer.frameCount);
requested = &ClientProxy::kNonBlocking;
size_t avail = audioBuffer.frameCount + nonContig;
- ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d",
+ ALOGV("obtainBuffer(%u) returned %zu = %zu + %zu err %d",
mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
if (err != NO_ERROR) {
if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR ||
@@ -1674,8 +1721,8 @@
// Sanity check on returned size
if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) {
- ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes",
- reqSize, (int) writtenSize);
+ ALOGE("EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes",
+ reqSize, ssize_t(writtenSize));
return NS_NEVER;
}
@@ -1736,7 +1783,7 @@
status_t AudioTrack::restoreTrack_l(const char *from)
{
ALOGW("dead IAudioTrack, %s, creating a new one from %s()",
- isOffloaded_l() ? "Offloaded" : "PCM", from);
+ isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from);
++mSequence;
status_t result;
@@ -1744,7 +1791,7 @@
// output parameters in createTrack_l()
AudioSystem::clearAudioConfigCache();
- if (isOffloaded_l()) {
+ if (isOffloadedOrDirect_l()) {
// FIXME re-creation of offloaded tracks is not yet implemented
return DEAD_OBJECT;
}
@@ -1830,6 +1877,19 @@
return isOffloaded_l();
}
+bool AudioTrack::isDirect() const
+{
+ AutoMutex lock(mLock);
+ return isDirect_l();
+}
+
+bool AudioTrack::isOffloadedOrDirect() const
+{
+ AutoMutex lock(mLock);
+ return isOffloadedOrDirect_l();
+}
+
+
status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const
{
@@ -1858,6 +1918,136 @@
return mProxy->getUnderrunFrames();
}
+void AudioTrack::setAttributesFromStreamType(audio_stream_type_t streamType) {
+ mAttributes.flags = 0x0;
+
+ switch(streamType) {
+ case AUDIO_STREAM_DEFAULT:
+ case AUDIO_STREAM_MUSIC:
+ mAttributes.content_type = AUDIO_CONTENT_TYPE_MUSIC;
+ mAttributes.usage = AUDIO_USAGE_MEDIA;
+ break;
+ case AUDIO_STREAM_VOICE_CALL:
+ mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH;
+ mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION;
+ break;
+ case AUDIO_STREAM_ENFORCED_AUDIBLE:
+ mAttributes.flags |= AUDIO_FLAG_AUDIBILITY_ENFORCED;
+ // intended fall through, attributes in common with STREAM_SYSTEM
+ case AUDIO_STREAM_SYSTEM:
+ mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+ mAttributes.usage = AUDIO_USAGE_ASSISTANCE_SONIFICATION;
+ break;
+ case AUDIO_STREAM_RING:
+ mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+ mAttributes.usage = AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE;
+ break;
+ case AUDIO_STREAM_ALARM:
+ mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+ mAttributes.usage = AUDIO_USAGE_ALARM;
+ break;
+ case AUDIO_STREAM_NOTIFICATION:
+ mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+ mAttributes.usage = AUDIO_USAGE_NOTIFICATION;
+ break;
+ case AUDIO_STREAM_BLUETOOTH_SCO:
+ mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH;
+ mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION;
+ mAttributes.flags |= AUDIO_FLAG_SCO;
+ break;
+ case AUDIO_STREAM_DTMF:
+ mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+ mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING;
+ break;
+ case AUDIO_STREAM_TTS:
+ mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH;
+ mAttributes.usage = AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY;
+ break;
+ default:
+ ALOGE("invalid stream type %d when converting to attributes", streamType);
+ }
+}
+
+void AudioTrack::setStreamTypeFromAttributes(audio_attributes_t& aa) {
+ // flags to stream type mapping
+ if ((aa.flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) {
+ mStreamType = AUDIO_STREAM_ENFORCED_AUDIBLE;
+ return;
+ }
+ if ((aa.flags & AUDIO_FLAG_SCO) == AUDIO_FLAG_SCO) {
+ mStreamType = AUDIO_STREAM_BLUETOOTH_SCO;
+ return;
+ }
+
+ // usage to stream type mapping
+ switch (aa.usage) {
+ case AUDIO_USAGE_MEDIA:
+ case AUDIO_USAGE_GAME:
+ case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
+ case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
+ mStreamType = AUDIO_STREAM_MUSIC;
+ return;
+ case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
+ mStreamType = AUDIO_STREAM_SYSTEM;
+ return;
+ case AUDIO_USAGE_VOICE_COMMUNICATION:
+ mStreamType = AUDIO_STREAM_VOICE_CALL;
+ return;
+
+ case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
+ mStreamType = AUDIO_STREAM_DTMF;
+ return;
+
+ case AUDIO_USAGE_ALARM:
+ mStreamType = AUDIO_STREAM_ALARM;
+ return;
+ case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
+ mStreamType = AUDIO_STREAM_RING;
+ return;
+
+ case AUDIO_USAGE_NOTIFICATION:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
+ case AUDIO_USAGE_NOTIFICATION_EVENT:
+ mStreamType = AUDIO_STREAM_NOTIFICATION;
+ return;
+
+ case AUDIO_USAGE_UNKNOWN:
+ default:
+ mStreamType = AUDIO_STREAM_MUSIC;
+ }
+}
+
+bool AudioTrack::isValidAttributes(const audio_attributes_t *paa) {
+ // has flags that map to a strategy?
+ if ((paa->flags & (AUDIO_FLAG_AUDIBILITY_ENFORCED | AUDIO_FLAG_SCO)) != 0) {
+ return true;
+ }
+
+ // has known usage?
+ switch (paa->usage) {
+ case AUDIO_USAGE_UNKNOWN:
+ case AUDIO_USAGE_MEDIA:
+ case AUDIO_USAGE_VOICE_COMMUNICATION:
+ case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
+ case AUDIO_USAGE_ALARM:
+ case AUDIO_USAGE_NOTIFICATION:
+ case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
+ case AUDIO_USAGE_NOTIFICATION_EVENT:
+ case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
+ case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
+ case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
+ case AUDIO_USAGE_GAME:
+ break;
+ default:
+ return false;
+ }
+ return true;
+}
// =========================================================================
void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused)
@@ -1918,7 +2108,7 @@
ns = 1000000000LL;
// fall through
default:
- LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns);
+ LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns);
pauseInternal(ns);
return true;
}
diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp
index 0dbfa62..eec025e 100644
--- a/media/libmedia/AudioTrackShared.cpp
+++ b/media/libmedia/AudioTrackShared.cpp
@@ -135,7 +135,7 @@
// pipe should not be overfull
if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
if (mIsOut) {
- ALOGE("Shared memory control block is corrupt (filled=%d, mFrameCount=%u); "
+ ALOGE("Shared memory control block is corrupt (filled=%zd, mFrameCount=%zu); "
"shutting down", filled, mFrameCount);
mIsShutdown = true;
status = NO_INIT;
@@ -338,7 +338,7 @@
ssize_t filled = rear - front;
// pipe should not be overfull
if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
- ALOGE("Shared memory control block is corrupt (filled=%d); shutting down", filled);
+ ALOGE("Shared memory control block is corrupt (filled=%zd); shutting down", filled);
return 0;
}
return (size_t)filled;
@@ -555,7 +555,7 @@
ssize_t filled = rear - front;
// pipe should not already be overfull
if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
- ALOGE("Shared memory control block is corrupt (filled=%d); shutting down", filled);
+ ALOGE("Shared memory control block is corrupt (filled=%zd); shutting down", filled);
mIsShutdown = true;
}
if (mIsShutdown) {
@@ -642,7 +642,7 @@
}
// FIXME AudioRecord wakeup needs to be optimized; it currently wakes up client every time
if (!mIsOut || (mAvailToClient + stepCount >= minimum)) {
- ALOGV("mAvailToClient=%u stepCount=%u minimum=%u", mAvailToClient, stepCount, minimum);
+ ALOGV("mAvailToClient=%zu stepCount=%zu minimum=%zu", mAvailToClient, stepCount, minimum);
int32_t old = android_atomic_or(CBLK_FUTEX_WAKE, &cblk->mFutex);
if (!(old & CBLK_FUTEX_WAKE)) {
(void) syscall(__NR_futex, &cblk->mFutex,
@@ -675,7 +675,7 @@
ssize_t filled = rear - cblk->u.mStreaming.mFront;
// pipe should not already be overfull
if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
- ALOGE("Shared memory control block is corrupt (filled=%d); shutting down", filled);
+ ALOGE("Shared memory control block is corrupt (filled=%zd); shutting down", filled);
mIsShutdown = true;
return 0;
}
@@ -834,7 +834,7 @@
size_t newPosition = position + stepCount;
int32_t setFlags = 0;
if (!(position <= newPosition && newPosition <= mFrameCount)) {
- ALOGW("%s newPosition %u outside [%u, %u]", __func__, newPosition, position, mFrameCount);
+ ALOGW("%s newPosition %zu outside [%zu, %zu]", __func__, newPosition, position, mFrameCount);
newPosition = mFrameCount;
} else if (mState.mLoopCount != 0 && newPosition == mState.mLoopEnd) {
if (mState.mLoopCount == -1 || --mState.mLoopCount != 0) {
diff --git a/media/libmedia/CharacterEncodingDetector.cpp b/media/libmedia/CharacterEncodingDetector.cpp
index 4992798..7d1ddfd 100644
--- a/media/libmedia/CharacterEncodingDetector.cpp
+++ b/media/libmedia/CharacterEncodingDetector.cpp
@@ -112,7 +112,7 @@
if (allprintable) {
// since 'buf' is empty, ICU would return a UTF-8 matcher with low confidence, so
// no need to even call it
- ALOGV("all tags are printable, assuming ascii (%d)", strlen(buf));
+ ALOGV("all tags are printable, assuming ascii (%zu)", strlen(buf));
} else {
ucsdet_setText(csd, buf, strlen(buf), &status);
int32_t matches;
@@ -267,11 +267,11 @@
Vector<const UCharsetMatch*> matches;
UErrorCode status = U_ZERO_ERROR;
- ALOGV("%d matches", nummatches);
+ ALOGV("%zu matches", nummatches);
for (size_t i = 0; i < nummatches; i++) {
const char *encname = ucsdet_getName(ucma[i], &status);
int confidence = ucsdet_getConfidence(ucma[i], &status);
- ALOGV("%d: %s %d", i, encname, confidence);
+ ALOGV("%zu: %s %d", i, encname, confidence);
matches.push_back(ucma[i]);
}
@@ -287,7 +287,7 @@
return matches[0];
}
- ALOGV("considering %d matches", num);
+ ALOGV("considering %zu matches", num);
// keep track of how many "special" characters result when converting the input using each
// encoding
@@ -315,7 +315,7 @@
freqcoverage = frequent_ja_coverage;
}
- ALOGV("%d: %s %d", i, encname, confidence);
+ ALOGV("%zu: %s %d", i, encname, confidence);
UConverter *conv = ucnv_open(encname, &status);
const char *source = input;
const char *sourceLimit = input + len;
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index 687fa76..5cf42f7 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -175,6 +175,7 @@
track_flags_t *flags,
pid_t tid,
int *sessionId,
+ size_t *notificationFrames,
sp<IMemory>& cblk,
sp<IMemory>& buffers,
status_t *status)
@@ -214,6 +215,10 @@
if (sessionId != NULL) {
*sessionId = lSessionId;
}
+ size_t lNotificationFrames = (size_t) reply.readInt64();
+ if (notificationFrames != NULL) {
+ *notificationFrames = lNotificationFrames;
+ }
lStatus = reply.readInt32();
record = interface_cast<IAudioRecord>(reply.readStrongBinder());
cblk = interface_cast<IMemory>(reply.readStrongBinder());
@@ -959,16 +964,19 @@
track_flags_t flags = (track_flags_t) data.readInt32();
pid_t tid = (pid_t) data.readInt32();
int sessionId = data.readInt32();
+ size_t notificationFrames = 0;
sp<IMemory> cblk;
sp<IMemory> buffers;
status_t status;
sp<IAudioRecord> record = openRecord(input,
sampleRate, format, channelMask, &frameCount, &flags, tid, &sessionId,
+ ¬ificationFrames,
cblk, buffers, &status);
LOG_ALWAYS_FATAL_IF((record != 0) != (status == NO_ERROR));
reply->writeInt64(frameCount);
reply->writeInt32(flags);
reply->writeInt32(sessionId);
+ reply->writeInt64(notificationFrames);
reply->writeInt32(status);
reply->writeStrongBinder(record->asBinder());
reply->writeStrongBinder(cblk->asBinder());
diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp
index 77d131b..41a9065 100644
--- a/media/libmedia/IAudioPolicyService.cpp
+++ b/media/libmedia/IAudioPolicyService.cpp
@@ -64,7 +64,8 @@
RELEASE_AUDIO_PATCH,
LIST_AUDIO_PATCHES,
SET_AUDIO_PORT_CONFIG,
- REGISTER_CLIENT
+ REGISTER_CLIENT,
+ GET_OUTPUT_FOR_ATTR
};
class BpAudioPolicyService : public BpInterface<IAudioPolicyService>
@@ -155,6 +156,36 @@
return static_cast <audio_io_handle_t> (reply.readInt32());
}
+ virtual audio_io_handle_t getOutputForAttr(
+ const audio_attributes_t *attr,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo)
+ {
+ Parcel data, reply;
+ data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+ if (attr == NULL) {
+ ALOGE("Writing NULL audio attributes - shouldn't happen");
+ return (audio_io_handle_t) 0;
+ }
+ data.write(attr, sizeof(audio_attributes_t));
+ data.writeInt32(samplingRate);
+ data.writeInt32(static_cast <uint32_t>(format));
+ data.writeInt32(channelMask);
+ data.writeInt32(static_cast <uint32_t>(flags));
+ // hasOffloadInfo
+ if (offloadInfo == NULL) {
+ data.writeInt32(0);
+ } else {
+ data.writeInt32(1);
+ data.write(offloadInfo, sizeof(audio_offload_info_t));
+ }
+ remote()->transact(GET_OUTPUT_FOR_ATTR, data, &reply);
+ return static_cast <audio_io_handle_t> (reply.readInt32());
+ }
+
virtual status_t startOutput(audio_io_handle_t output,
audio_stream_type_t stream,
int session)
@@ -614,6 +645,30 @@
return NO_ERROR;
} break;
+ case GET_OUTPUT_FOR_ATTR: {
+ CHECK_INTERFACE(IAudioPolicyService, data, reply);
+ audio_attributes_t *attr = (audio_attributes_t *) calloc(1, sizeof(audio_attributes_t));
+ data.read(attr, sizeof(audio_attributes_t));
+ uint32_t samplingRate = data.readInt32();
+ audio_format_t format = (audio_format_t) data.readInt32();
+ audio_channel_mask_t channelMask = data.readInt32();
+ audio_output_flags_t flags =
+ static_cast <audio_output_flags_t>(data.readInt32());
+ bool hasOffloadInfo = data.readInt32() != 0;
+ audio_offload_info_t offloadInfo;
+ if (hasOffloadInfo) {
+ data.read(&offloadInfo, sizeof(audio_offload_info_t));
+ }
+ audio_io_handle_t output = getOutputForAttr(attr,
+ samplingRate,
+ format,
+ channelMask,
+ flags,
+ hasOffloadInfo ? &offloadInfo : NULL);
+ reply->writeInt32(static_cast <int>(output));
+ return NO_ERROR;
+ } break;
+
case START_OUTPUT: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
audio_io_handle_t output = static_cast <audio_io_handle_t>(data.readInt32());
diff --git a/media/libmedia/IMediaMetadataRetriever.cpp b/media/libmedia/IMediaMetadataRetriever.cpp
index 432d890..38f717c 100644
--- a/media/libmedia/IMediaMetadataRetriever.cpp
+++ b/media/libmedia/IMediaMetadataRetriever.cpp
@@ -15,8 +15,10 @@
** limitations under the License.
*/
+#include <inttypes.h>
#include <stdint.h>
#include <sys/types.h>
+
#include <binder/Parcel.h>
#include <media/IMediaHTTPService.h>
#include <media/IMediaMetadataRetriever.h>
@@ -125,7 +127,7 @@
sp<IMemory> getFrameAtTime(int64_t timeUs, int option)
{
- ALOGV("getTimeAtTime: time(%lld us) and option(%d)", timeUs, option);
+ ALOGV("getTimeAtTime: time(%" PRId64 " us) and option(%d)", timeUs, option);
Parcel data, reply;
data.writeInterfaceToken(IMediaMetadataRetriever::getInterfaceDescriptor());
data.writeInt64(timeUs);
@@ -237,7 +239,7 @@
CHECK_INTERFACE(IMediaMetadataRetriever, data, reply);
int64_t timeUs = data.readInt64();
int option = data.readInt32();
- ALOGV("getTimeAtTime: time(%lld us) and option(%d)", timeUs, option);
+ ALOGV("getTimeAtTime: time(%" PRId64 " us) and option(%d)", timeUs, option);
#ifndef DISABLE_GROUP_SCHEDULE_HACK
setSchedPolicy(data);
#endif
diff --git a/media/libmedia/IMediaRecorder.cpp b/media/libmedia/IMediaRecorder.cpp
index 8e58162..95af006 100644
--- a/media/libmedia/IMediaRecorder.cpp
+++ b/media/libmedia/IMediaRecorder.cpp
@@ -17,6 +17,10 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "IMediaRecorder"
+
+#include <inttypes.h>
+#include <unistd.h>
+
#include <utils/Log.h>
#include <binder/Parcel.h>
#include <camera/ICamera.h>
@@ -24,8 +28,6 @@
#include <media/IMediaRecorder.h>
#include <gui/Surface.h>
#include <gui/IGraphicBufferProducer.h>
-#include <unistd.h>
-
namespace android {
@@ -167,7 +169,7 @@
}
status_t setOutputFile(int fd, int64_t offset, int64_t length) {
- ALOGV("setOutputFile(%d, %lld, %lld)", fd, offset, length);
+ ALOGV("setOutputFile(%d, %" PRId64 ", %" PRId64 ")", fd, offset, length);
Parcel data, reply;
data.writeInterfaceToken(IMediaRecorder::getInterfaceDescriptor());
data.writeFileDescriptor(fd);
diff --git a/media/libmedia/MediaProfiles.cpp b/media/libmedia/MediaProfiles.cpp
index 28238c4..e9e453b 100644
--- a/media/libmedia/MediaProfiles.cpp
+++ b/media/libmedia/MediaProfiles.cpp
@@ -475,7 +475,7 @@
}
void MediaProfiles::initRequiredProfileRefs(const Vector<int>& cameraIds) {
- ALOGV("Number of camera ids: %d", cameraIds.size());
+ ALOGV("Number of camera ids: %zu", cameraIds.size());
CHECK(cameraIds.size() > 0);
mRequiredProfileRefs = new RequiredProfiles[cameraIds.size()];
for (size_t i = 0, n = cameraIds.size(); i < n; ++i) {
@@ -602,14 +602,14 @@
int index = getCamcorderProfileIndex(cameraId, profile->mQuality);
if (index != -1) {
- ALOGV("Profile quality %d for camera %d already exists",
+ ALOGV("Profile quality %d for camera %zu already exists",
profile->mQuality, cameraId);
CHECK(index == refIndex);
continue;
}
// Insert the new profile
- ALOGV("Add a profile: quality %d=>%d for camera %d",
+ ALOGV("Add a profile: quality %d=>%d for camera %zu",
mCamcorderProfiles[info->mRefProfileIndex]->mQuality,
profile->mQuality, cameraId);
diff --git a/media/libmedia/SoundPool.cpp b/media/libmedia/SoundPool.cpp
index a55e09c..2aa0592 100644
--- a/media/libmedia/SoundPool.cpp
+++ b/media/libmedia/SoundPool.cpp
@@ -16,6 +16,9 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "SoundPool"
+
+#include <inttypes.h>
+
#include <utils/Log.h>
#define USE_SHARED_MEM_BUFFER
@@ -212,7 +215,7 @@
int SoundPool::load(int fd, int64_t offset, int64_t length, int priority __unused)
{
- ALOGV("load: fd=%d, offset=%lld, length=%lld, priority=%d",
+ ALOGV("load: fd=%d, offset=%" PRId64 ", length=%" PRId64 ", priority=%d",
fd, offset, length, priority);
Mutex::Autolock lock(&mLock);
sp<Sample> sample = new Sample(++mNextSampleID, fd, offset, length);
@@ -462,7 +465,8 @@
mFd = dup(fd);
mOffset = offset;
mLength = length;
- ALOGV("create sampleID=%d, fd=%d, offset=%lld, length=%lld", mSampleID, mFd, mLength, mOffset);
+ ALOGV("create sampleID=%d, fd=%d, offset=%" PRId64 " length=%" PRId64,
+ mSampleID, mFd, mLength, mOffset);
}
void Sample::init()
@@ -516,7 +520,7 @@
ALOGE("Unable to load sample: %s", mUrl);
goto error;
}
- ALOGV("pointer = %p, size = %u, sampleRate = %u, numChannels = %d",
+ ALOGV("pointer = %p, size = %zu, sampleRate = %u, numChannels = %d",
mHeap->getBase(), mSize, sampleRate, numChannels);
if (sampleRate > kMaxSampleRate) {
diff --git a/media/libmedia/mediametadataretriever.cpp b/media/libmedia/mediametadataretriever.cpp
index 1d6bb6f..39a239d 100644
--- a/media/libmedia/mediametadataretriever.cpp
+++ b/media/libmedia/mediametadataretriever.cpp
@@ -18,6 +18,8 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "MediaMetadataRetriever"
+#include <inttypes.h>
+
#include <binder/IServiceManager.h>
#include <binder/IPCThreadState.h>
#include <media/mediametadataretriever.h>
@@ -114,7 +116,7 @@
status_t MediaMetadataRetriever::setDataSource(int fd, int64_t offset, int64_t length)
{
- ALOGV("setDataSource(%d, %lld, %lld)", fd, offset, length);
+ ALOGV("setDataSource(%d, %" PRId64 ", %" PRId64 ")", fd, offset, length);
Mutex::Autolock _l(mLock);
if (mRetriever == 0) {
ALOGE("retriever is not initialized");
@@ -129,7 +131,7 @@
sp<IMemory> MediaMetadataRetriever::getFrameAtTime(int64_t timeUs, int option)
{
- ALOGV("getFrameAtTime: time(%lld us) option(%d)", timeUs, option);
+ ALOGV("getFrameAtTime: time(%" PRId64 " us) option(%d)", timeUs, option);
Mutex::Autolock _l(mLock);
if (mRetriever == 0) {
ALOGE("retriever is not initialized");
diff --git a/media/libmedia/mediaplayer.cpp b/media/libmedia/mediaplayer.cpp
index 0be01a9..889bd7f 100644
--- a/media/libmedia/mediaplayer.cpp
+++ b/media/libmedia/mediaplayer.cpp
@@ -17,12 +17,14 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "MediaPlayer"
-#include <utils/Log.h>
-#include <sys/types.h>
-#include <sys/stat.h>
-#include <unistd.h>
#include <fcntl.h>
+#include <inttypes.h>
+#include <sys/stat.h>
+#include <sys/types.h>
+#include <unistd.h>
+
+#include <utils/Log.h>
#include <binder/IServiceManager.h>
#include <binder/IPCThreadState.h>
@@ -157,7 +159,7 @@
status_t MediaPlayer::setDataSource(int fd, int64_t offset, int64_t length)
{
- ALOGV("setDataSource(%d, %lld, %lld)", fd, offset, length);
+ ALOGV("setDataSource(%d, %" PRId64 ", %" PRId64 ")", fd, offset, length);
status_t err = UNKNOWN_ERROR;
const sp<IMediaPlayerService>& service(getMediaPlayerService());
if (service != 0) {
@@ -194,7 +196,7 @@
(mCurrentState != MEDIA_PLAYER_STATE_ERROR) &&
((mCurrentState & MEDIA_PLAYER_IDLE) != MEDIA_PLAYER_IDLE);
if ((mPlayer != NULL) && hasBeenInitialized) {
- ALOGV("invoke %d", request.dataSize());
+ ALOGV("invoke %zu", request.dataSize());
return mPlayer->invoke(request, reply);
}
ALOGE("invoke failed: wrong state %X", mCurrentState);
@@ -622,10 +624,32 @@
return mPlayer->attachAuxEffect(effectId);
}
+// always call with lock held
+status_t MediaPlayer::checkStateForKeySet_l(int key)
+{
+ switch(key) {
+ case KEY_PARAMETER_AUDIO_ATTRIBUTES:
+ if (mCurrentState & ( MEDIA_PLAYER_PREPARED | MEDIA_PLAYER_STARTED |
+ MEDIA_PLAYER_PAUSED | MEDIA_PLAYER_PLAYBACK_COMPLETE) ) {
+ // Can't change the audio attributes after prepare
+ ALOGE("trying to set audio attributes called in state %d", mCurrentState);
+ return INVALID_OPERATION;
+ }
+ break;
+ default:
+ // parameter doesn't require player state check
+ break;
+ }
+ return OK;
+}
+
status_t MediaPlayer::setParameter(int key, const Parcel& request)
{
ALOGV("MediaPlayer::setParameter(%d)", key);
Mutex::Autolock _l(mLock);
+ if (checkStateForKeySet_l(key) != OK) {
+ return INVALID_OPERATION;
+ }
if (mPlayer != NULL) {
return mPlayer->setParameter(key, request);
}
@@ -818,7 +842,7 @@
audio_format_t* pFormat,
const sp<IMemoryHeap>& heap, size_t *pSize)
{
- ALOGV("decode(%d, %lld, %lld)", fd, offset, length);
+ ALOGV("decode(%d, %" PRId64 ", %" PRId64 ")", fd, offset, length);
status_t status;
const sp<IMediaPlayerService>& service = getMediaPlayerService();
if (service != 0) {
diff --git a/media/libmedia/mediarecorder.cpp b/media/libmedia/mediarecorder.cpp
index 3710e46..c8192e9 100644
--- a/media/libmedia/mediarecorder.cpp
+++ b/media/libmedia/mediarecorder.cpp
@@ -17,6 +17,9 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "MediaRecorder"
+
+#include <inttypes.h>
+
#include <utils/Log.h>
#include <media/mediarecorder.h>
#include <binder/IServiceManager.h>
@@ -286,7 +289,7 @@
status_t MediaRecorder::setOutputFile(int fd, int64_t offset, int64_t length)
{
- ALOGV("setOutputFile(%d, %lld, %lld)", fd, offset, length);
+ ALOGV("setOutputFile(%d, %" PRId64 ", %" PRId64 ")", fd, offset, length);
if (mMediaRecorder == NULL) {
ALOGE("media recorder is not initialized yet");
return INVALID_OPERATION;
diff --git a/media/libmediaplayerservice/MediaPlayerFactory.cpp b/media/libmediaplayerservice/MediaPlayerFactory.cpp
index 9b239b1..e9c5e8e 100644
--- a/media/libmediaplayerservice/MediaPlayerFactory.cpp
+++ b/media/libmediaplayerservice/MediaPlayerFactory.cpp
@@ -176,11 +176,11 @@
class StagefrightPlayerFactory :
public MediaPlayerFactory::IFactory {
public:
- virtual float scoreFactory(const sp<IMediaPlayer>& client,
+ virtual float scoreFactory(const sp<IMediaPlayer>& /*client*/,
int fd,
int64_t offset,
- int64_t length,
- float curScore) {
+ int64_t /*length*/,
+ float /*curScore*/) {
char buf[20];
lseek(fd, offset, SEEK_SET);
read(fd, buf, sizeof(buf));
@@ -203,7 +203,7 @@
class NuPlayerFactory : public MediaPlayerFactory::IFactory {
public:
- virtual float scoreFactory(const sp<IMediaPlayer>& client,
+ virtual float scoreFactory(const sp<IMediaPlayer>& /*client*/,
const char* url,
float curScore) {
static const float kOurScore = 0.8;
@@ -235,9 +235,9 @@
return 0.0;
}
- virtual float scoreFactory(const sp<IMediaPlayer>& client,
- const sp<IStreamSource> &source,
- float curScore) {
+ virtual float scoreFactory(const sp<IMediaPlayer>& /*client*/,
+ const sp<IStreamSource>& /*source*/,
+ float /*curScore*/) {
return 1.0;
}
@@ -249,7 +249,7 @@
class SonivoxPlayerFactory : public MediaPlayerFactory::IFactory {
public:
- virtual float scoreFactory(const sp<IMediaPlayer>& client,
+ virtual float scoreFactory(const sp<IMediaPlayer>& /*client*/,
const char* url,
float curScore) {
static const float kOurScore = 0.4;
@@ -280,7 +280,7 @@
return 0.0;
}
- virtual float scoreFactory(const sp<IMediaPlayer>& client,
+ virtual float scoreFactory(const sp<IMediaPlayer>& /*client*/,
int fd,
int64_t offset,
int64_t length,
@@ -318,9 +318,9 @@
class TestPlayerFactory : public MediaPlayerFactory::IFactory {
public:
- virtual float scoreFactory(const sp<IMediaPlayer>& client,
+ virtual float scoreFactory(const sp<IMediaPlayer>& /*client*/,
const char* url,
- float curScore) {
+ float /*curScore*/) {
if (TestPlayerStub::canBeUsed(url)) {
return 1.0;
}
diff --git a/media/libmediaplayerservice/MediaPlayerFactory.h b/media/libmediaplayerservice/MediaPlayerFactory.h
index fe8972b..5ddde19 100644
--- a/media/libmediaplayerservice/MediaPlayerFactory.h
+++ b/media/libmediaplayerservice/MediaPlayerFactory.h
@@ -29,19 +29,19 @@
public:
virtual ~IFactory() { }
- virtual float scoreFactory(const sp<IMediaPlayer>& client,
- const char* url,
- float curScore) { return 0.0; }
+ virtual float scoreFactory(const sp<IMediaPlayer>& /*client*/,
+ const char* /*url*/,
+ float /*curScore*/) { return 0.0; }
- virtual float scoreFactory(const sp<IMediaPlayer>& client,
- int fd,
- int64_t offset,
- int64_t length,
- float curScore) { return 0.0; }
+ virtual float scoreFactory(const sp<IMediaPlayer>& /*client*/,
+ int /*fd*/,
+ int64_t /*offset*/,
+ int64_t /*length*/,
+ float /*curScore*/) { return 0.0; }
- virtual float scoreFactory(const sp<IMediaPlayer>& client,
- const sp<IStreamSource> &source,
- float curScore) { return 0.0; }
+ virtual float scoreFactory(const sp<IMediaPlayer>& /*client*/,
+ const sp<IStreamSource> &/*source*/,
+ float /*curScore*/) { return 0.0; }
virtual sp<MediaPlayerBase> createPlayer() = 0;
};
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index 778eb9a..7218467 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -186,6 +186,60 @@
} // anonymous namespace
+namespace {
+using android::Parcel;
+using android::String16;
+
+// marshalling tag indicating flattened utf16 tags
+// keep in sync with frameworks/base/media/java/android/media/AudioAttributes.java
+const int32_t kAudioAttributesMarshallTagFlattenTags = 1;
+
+// Audio attributes format in a parcel:
+//
+// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// | usage |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// | content_type |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// | flags |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// | kAudioAttributesMarshallTagFlattenTags | // ignore tags if not found
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// | flattened tags in UTF16 |
+// | ... |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//
+// @param p Parcel that contains audio attributes.
+// @param[out] attributes On exit points to an initialized audio_attributes_t structure
+// @param[out] status On exit contains the status code to be returned.
+void unmarshallAudioAttributes(const Parcel& parcel, audio_attributes_t *attributes)
+{
+ attributes->usage = (audio_usage_t) parcel.readInt32();
+ attributes->content_type = (audio_content_type_t) parcel.readInt32();
+ attributes->flags = (audio_flags_mask_t) parcel.readInt32();
+ const bool hasFlattenedTag = (parcel.readInt32() == kAudioAttributesMarshallTagFlattenTags);
+ if (hasFlattenedTag) {
+ // the tags are UTF16, convert to UTF8
+ String16 tags = parcel.readString16();
+ ssize_t realTagSize = utf16_to_utf8_length(tags.string(), tags.size());
+ if (realTagSize <= 0) {
+ strcpy(attributes->tags, "");
+ } else {
+ // copy the flattened string into the attributes as the destination for the conversion:
+ // copying array size -1, array for tags was calloc'd, no need to NULL-terminate it
+ size_t tagSize = realTagSize > AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - 1 ?
+ AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - 1 : realTagSize;
+ utf16_to_utf8(tags.string(), tagSize, attributes->tags);
+ }
+ } else {
+ ALOGE("unmarshallAudioAttributes() received unflattened tags, ignoring tag values");
+ strcpy(attributes->tags, "");
+ }
+}
+} // anonymous namespace
+
+
namespace android {
static bool checkPermission(const char* permissionString) {
@@ -307,7 +361,7 @@
return new RemoteDisplay(client, iface.string());
}
-status_t MediaPlayerService::AudioCache::dump(int fd, const Vector<String16>& args) const
+status_t MediaPlayerService::AudioCache::dump(int fd, const Vector<String16>& /*args*/) const
{
const size_t SIZE = 256;
char buffer[SIZE];
@@ -508,6 +562,7 @@
mAudioSessionId = audioSessionId;
mUID = uid;
mRetransmitEndpointValid = false;
+ mAudioAttributes = NULL;
#if CALLBACK_ANTAGONIZER
ALOGD("create Antagonizer");
@@ -522,6 +577,9 @@
wp<Client> client(this);
disconnect();
mService->removeClient(client);
+ if (mAudioAttributes != NULL) {
+ free(mAudioAttributes);
+ }
}
void MediaPlayerService::Client::disconnect()
@@ -587,7 +645,7 @@
if (!p->hardwareOutput()) {
mAudioOutput = new AudioOutput(mAudioSessionId, IPCThreadState::self()->getCallingUid(),
- mPid);
+ mPid, mAudioAttributes);
static_cast<MediaPlayerInterface*>(p.get())->setAudioSink(mAudioOutput);
}
@@ -673,8 +731,8 @@
ALOGV("st_dev = %llu", sb.st_dev);
ALOGV("st_mode = %u", sb.st_mode);
- ALOGV("st_uid = %lu", sb.st_uid);
- ALOGV("st_gid = %lu", sb.st_gid);
+ ALOGV("st_uid = %lu", static_cast<unsigned long>(sb.st_uid));
+ ALOGV("st_gid = %lu", static_cast<unsigned long>(sb.st_gid));
ALOGV("st_size = %llu", sb.st_size);
if (offset >= sb.st_size) {
@@ -803,7 +861,7 @@
}
status_t MediaPlayerService::Client::getMetadata(
- bool update_only, bool apply_filter, Parcel *reply)
+ bool update_only, bool /*apply_filter*/, Parcel *reply)
{
sp<MediaPlayerBase> player = getPlayer();
if (player == 0) return UNKNOWN_ERROR;
@@ -968,6 +1026,22 @@
return NO_ERROR;
}
+status_t MediaPlayerService::Client::setAudioAttributes_l(const Parcel &parcel)
+{
+ if (mAudioAttributes != NULL) { free(mAudioAttributes); }
+ mAudioAttributes = (audio_attributes_t *) calloc(1, sizeof(audio_attributes_t));
+ unmarshallAudioAttributes(parcel, mAudioAttributes);
+
+ ALOGV("setAudioAttributes_l() usage=%d content=%d flags=0x%x tags=%s",
+ mAudioAttributes->usage, mAudioAttributes->content_type, mAudioAttributes->flags,
+ mAudioAttributes->tags);
+
+ if (mAudioOutput != 0) {
+ mAudioOutput->setAudioAttributes(mAudioAttributes);
+ }
+ return NO_ERROR;
+}
+
status_t MediaPlayerService::Client::setLooping(int loop)
{
ALOGV("[%d] setLooping(%d)", mConnId, loop);
@@ -1016,9 +1090,17 @@
status_t MediaPlayerService::Client::setParameter(int key, const Parcel &request) {
ALOGV("[%d] setParameter(%d)", mConnId, key);
- sp<MediaPlayerBase> p = getPlayer();
- if (p == 0) return UNKNOWN_ERROR;
- return p->setParameter(key, request);
+ switch (key) {
+ case KEY_PARAMETER_AUDIO_ATTRIBUTES:
+ {
+ Mutex::Autolock l(mLock);
+ return setAudioAttributes_l(request);
+ }
+ default:
+ sp<MediaPlayerBase> p = getPlayer();
+ if (p == 0) { return UNKNOWN_ERROR; }
+ return p->setParameter(key, request);
+ }
}
status_t MediaPlayerService::Client::getParameter(int key, Parcel *reply) {
@@ -1300,7 +1382,8 @@
#undef LOG_TAG
#define LOG_TAG "AudioSink"
-MediaPlayerService::AudioOutput::AudioOutput(int sessionId, int uid, int pid)
+MediaPlayerService::AudioOutput::AudioOutput(int sessionId, int uid, int pid,
+ const audio_attributes_t* attr)
: mCallback(NULL),
mCallbackCookie(NULL),
mCallbackData(NULL),
@@ -1319,6 +1402,7 @@
mAuxEffectId = 0;
mSendLevel = 0.0;
setMinBufferCount();
+ mAttributes = attr;
}
MediaPlayerService::AudioOutput::~AudioOutput()
@@ -1408,6 +1492,10 @@
return mTrack->getParameters(keys);
}
+void MediaPlayerService::AudioOutput::setAudioAttributes(const audio_attributes_t * attributes) {
+ mAttributes = attributes;
+}
+
void MediaPlayerService::AudioOutput::deleteRecycledTrack()
{
ALOGV("deleteRecycledTrack");
@@ -1557,7 +1645,8 @@
AudioTrack::TRANSFER_CALLBACK,
offloadInfo,
mUid,
- mPid);
+ mPid,
+ mAttributes);
} else {
t = new AudioTrack(
mStreamType,
@@ -1573,13 +1662,18 @@
AudioTrack::TRANSFER_DEFAULT,
NULL, // offload info
mUid,
- mPid);
+ mPid,
+ mAttributes);
}
if ((t == 0) || (t->initCheck() != NO_ERROR)) {
ALOGE("Unable to create audio track");
delete newcbd;
return NO_INIT;
+ } else {
+ // successful AudioTrack initialization implies a legacy stream type was generated
+ // from the audio attributes
+ mStreamType = t->streamType();
}
}
@@ -1926,8 +2020,8 @@
status_t MediaPlayerService::AudioCache::open(
uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
audio_format_t format, int bufferCount,
- AudioCallback cb, void *cookie, audio_output_flags_t flags,
- const audio_offload_info_t *offloadInfo)
+ AudioCallback cb, void *cookie, audio_output_flags_t /*flags*/,
+ const audio_offload_info_t* /*offloadInfo*/)
{
ALOGV("open(%u, %d, 0x%x, %d, %d)", sampleRate, channelCount, channelMask, format, bufferCount);
if (mHeap->getHeapID() < 0) {
@@ -1994,7 +2088,7 @@
}
void MediaPlayerService::AudioCache::notify(
- void* cookie, int msg, int ext1, int ext2, const Parcel *obj)
+ void* cookie, int msg, int ext1, int ext2, const Parcel* /*obj*/)
{
ALOGV("notify(%p, %d, %d, %d)", cookie, msg, ext1, ext2);
AudioCache* p = static_cast<AudioCache*>(cookie);
diff --git a/media/libmediaplayerservice/MediaPlayerService.h b/media/libmediaplayerservice/MediaPlayerService.h
index 448f27a..2eca6a0 100644
--- a/media/libmediaplayerservice/MediaPlayerService.h
+++ b/media/libmediaplayerservice/MediaPlayerService.h
@@ -72,7 +72,8 @@
class CallbackData;
public:
- AudioOutput(int sessionId, int uid, int pid);
+ AudioOutput(int sessionId, int uid, int pid,
+ const audio_attributes_t * attr);
virtual ~AudioOutput();
virtual bool ready() const { return mTrack != 0; }
@@ -104,6 +105,7 @@
void setAudioStreamType(audio_stream_type_t streamType) {
mStreamType = streamType; }
virtual audio_stream_type_t getAudioStreamType() const { return mStreamType; }
+ void setAudioAttributes(const audio_attributes_t * attributes);
void setVolume(float left, float right);
virtual status_t setPlaybackRatePermille(int32_t ratePermille);
@@ -133,6 +135,7 @@
CallbackData * mCallbackData;
uint64_t mBytesWritten;
audio_stream_type_t mStreamType;
+ const audio_attributes_t *mAttributes;
float mLeftVolume;
float mRightVolume;
int32_t mPlaybackRatePermille;
@@ -410,6 +413,8 @@
// Disconnect from the currently connected ANativeWindow.
void disconnectNativeWindow();
+ status_t setAudioAttributes_l(const Parcel &request);
+
mutable Mutex mLock;
sp<MediaPlayerBase> mPlayer;
sp<MediaPlayerService> mService;
@@ -420,6 +425,7 @@
bool mLoop;
int32_t mConnId;
int mAudioSessionId;
+ audio_attributes_t * mAudioAttributes;
uid_t mUID;
sp<ANativeWindow> mConnectedWindow;
sp<IBinder> mConnectedWindowBinder;
diff --git a/media/libmediaplayerservice/MetadataRetrieverClient.cpp b/media/libmediaplayerservice/MetadataRetrieverClient.cpp
index 80c7e0a..fa28451 100644
--- a/media/libmediaplayerservice/MetadataRetrieverClient.cpp
+++ b/media/libmediaplayerservice/MetadataRetrieverClient.cpp
@@ -57,7 +57,7 @@
disconnect();
}
-status_t MetadataRetrieverClient::dump(int fd, const Vector<String16>& args) const
+status_t MetadataRetrieverClient::dump(int fd, const Vector<String16>& /*args*/) const
{
const size_t SIZE = 256;
char buffer[SIZE];
@@ -147,8 +147,8 @@
}
ALOGV("st_dev = %llu", sb.st_dev);
ALOGV("st_mode = %u", sb.st_mode);
- ALOGV("st_uid = %lu", sb.st_uid);
- ALOGV("st_gid = %lu", sb.st_gid);
+ ALOGV("st_uid = %lu", static_cast<unsigned long>(sb.st_uid));
+ ALOGV("st_gid = %lu", static_cast<unsigned long>(sb.st_gid));
ALOGV("st_size = %llu", sb.st_size);
if (offset >= sb.st_size) {
diff --git a/media/libmediaplayerservice/MidiFile.cpp b/media/libmediaplayerservice/MidiFile.cpp
index deeddd1..749ef96 100644
--- a/media/libmediaplayerservice/MidiFile.cpp
+++ b/media/libmediaplayerservice/MidiFile.cpp
@@ -114,7 +114,7 @@
}
status_t MidiFile::setDataSource(
- const sp<IMediaHTTPService> &httpService,
+ const sp<IMediaHTTPService> & /*httpService*/,
const char* path,
const KeyedVector<String8, String8> *) {
ALOGV("MidiFile::setDataSource url=%s", path);
diff --git a/media/libmediaplayerservice/MidiFile.h b/media/libmediaplayerservice/MidiFile.h
index 12802ba..82e4e88 100644
--- a/media/libmediaplayerservice/MidiFile.h
+++ b/media/libmediaplayerservice/MidiFile.h
@@ -38,7 +38,7 @@
virtual status_t setDataSource(int fd, int64_t offset, int64_t length);
virtual status_t setVideoSurfaceTexture(
- const sp<IGraphicBufferProducer>& bufferProducer)
+ const sp<IGraphicBufferProducer>& /*bufferProducer*/)
{ return UNKNOWN_ERROR; }
virtual status_t prepare();
virtual status_t prepareAsync();
@@ -53,13 +53,13 @@
virtual status_t reset();
virtual status_t setLooping(int loop);
virtual player_type playerType() { return SONIVOX_PLAYER; }
- virtual status_t invoke(const Parcel& request, Parcel *reply) {
+ virtual status_t invoke(const Parcel& /*request*/, Parcel* /*reply*/) {
return INVALID_OPERATION;
}
- virtual status_t setParameter(int key, const Parcel &request) {
+ virtual status_t setParameter(int /*key*/, const Parcel &/*request*/) {
return INVALID_OPERATION;
}
- virtual status_t getParameter(int key, Parcel *reply) {
+ virtual status_t getParameter(int /*key*/, Parcel* /*reply*/) {
return INVALID_OPERATION;
}
diff --git a/media/libmediaplayerservice/nuplayer/GenericSource.cpp b/media/libmediaplayerservice/nuplayer/GenericSource.cpp
index 06aac33..388f77a 100644
--- a/media/libmediaplayerservice/nuplayer/GenericSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/GenericSource.cpp
@@ -67,6 +67,14 @@
CHECK(extractor != NULL);
+ sp<MetaData> fileMeta = extractor->getMetaData();
+ if (fileMeta != NULL) {
+ int64_t duration;
+ if (fileMeta->findInt64(kKeyDuration, &duration)) {
+ mDurationUs = duration;
+ }
+ }
+
for (size_t i = 0; i < extractor->countTracks(); ++i) {
sp<MetaData> meta = extractor->getTrackMetaData(i);
@@ -264,7 +272,9 @@
if (seeking) {
track->mPackets->queueDiscontinuity(
- ATSParser::DISCONTINUITY_SEEK, NULL);
+ ATSParser::DISCONTINUITY_SEEK,
+ NULL,
+ true /* discard */);
}
track->mPackets->queueAccessUnit(buffer);
@@ -272,7 +282,9 @@
} else if (err == INFO_FORMAT_CHANGED) {
#if 0
track->mPackets->queueDiscontinuity(
- ATSParser::DISCONTINUITY_FORMATCHANGE, NULL);
+ ATSParser::DISCONTINUITY_FORMATCHANGE,
+ NULL,
+ false /* discard */);
#endif
} else {
track->mPackets->signalEOS(err);
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
index 5abfb71..dd73cc4 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
@@ -674,9 +674,9 @@
bool hasCC = false;
- ABitReader br(sei->data() + 1, sei->size() - 1);
+ NALBitReader br(sei->data() + 1, sei->size() - 1);
// sei_message()
- while (br.numBitsLeft() >= 16) { // at least 16-bit for sei_message()
+ while (br.atLeastNumBitsLeft(16)) { // at least 16-bit for sei_message()
uint32_t payload_type = 0;
size_t payload_size = 0;
uint8_t last_byte;
diff --git a/media/libmediaplayerservice/nuplayer/RTSPSource.cpp b/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
index 94800ba..2338b23 100644
--- a/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
@@ -505,7 +505,10 @@
TrackInfo *info = &mTracks.editItemAt(trackIndex);
sp<AnotherPacketSource> source = info->mSource;
if (source != NULL) {
- source->queueDiscontinuity(ATSParser::DISCONTINUITY_SEEK, NULL);
+ source->queueDiscontinuity(
+ ATSParser::DISCONTINUITY_SEEK,
+ NULL,
+ true /* discard */);
}
break;
diff --git a/media/libnbaio/MonoPipe.cpp b/media/libnbaio/MonoPipe.cpp
index 4adf018..0b65861 100644
--- a/media/libnbaio/MonoPipe.cpp
+++ b/media/libnbaio/MonoPipe.cpp
@@ -14,6 +14,8 @@
* limitations under the License.
*/
+#include <inttypes.h>
+
#define LOG_TAG "MonoPipe"
//#define LOG_NDEBUG 0
@@ -87,7 +89,7 @@
static const uint64_t kUnsignedHiBitsMask = ~(0xFFFFFFFFull);
if ((N & kSignedHiBitsMask) || (D & kUnsignedHiBitsMask)) {
ALOGE("Cannot reduce sample rate to local clock frequency ratio to fit"
- " in a 32/32 bit rational. (max reduction is 0x%016llx/0x%016llx"
+ " in a 32/32 bit rational. (max reduction is 0x%016" PRIx64 "/0x%016" PRIx64
"). getNextWriteTimestamp calls will be non-functional", N, D);
return;
}
@@ -308,7 +310,7 @@
// error, but then zero out the ratio in the linear transform so
// that we don't try to do any conversions from now on. This
// MonoPipe's getNextWriteTimestamp is now broken for good.
- ALOGE("Overflow when attempting to convert %d audio frames to"
+ ALOGE("Overflow when attempting to convert %zu audio frames to"
" duration in local time. getNextWriteTimestamp will fail from"
" now on.", audFrames);
mSamplesToLocalTime.a_to_b_numer = 0;
diff --git a/media/libnbaio/NBAIO.cpp b/media/libnbaio/NBAIO.cpp
index ff3284c..d641e74 100644
--- a/media/libnbaio/NBAIO.cpp
+++ b/media/libnbaio/NBAIO.cpp
@@ -137,7 +137,7 @@
ssize_t NBAIO_Port::negotiate(const NBAIO_Format offers[], size_t numOffers,
NBAIO_Format counterOffers[], size_t& numCounterOffers)
{
- ALOGV("negotiate offers=%p numOffers=%u countersOffers=%p numCounterOffers=%u",
+ ALOGV("negotiate offers=%p numOffers=%zu countersOffers=%p numCounterOffers=%zu",
offers, numOffers, counterOffers, numCounterOffers);
if (Format_isValid(mFormat)) {
for (size_t i = 0; i < numOffers; ++i) {
diff --git a/media/libstagefright/AACWriter.cpp b/media/libstagefright/AACWriter.cpp
index deee8e7..353920e 100644
--- a/media/libstagefright/AACWriter.cpp
+++ b/media/libstagefright/AACWriter.cpp
@@ -14,6 +14,12 @@
* limitations under the License.
*/
+#include <fcntl.h>
+#include <inttypes.h>
+#include <sys/prctl.h>
+#include <sys/stat.h>
+#include <sys/types.h>
+
//#define LOG_NDEBUG 0
#define LOG_TAG "AACWriter"
#include <utils/Log.h>
@@ -27,10 +33,6 @@
#include <media/stagefright/MediaSource.h>
#include <media/stagefright/MetaData.h>
#include <media/mediarecorder.h>
-#include <sys/prctl.h>
-#include <sys/types.h>
-#include <sys/stat.h>
-#include <fcntl.h>
namespace android {
@@ -348,7 +350,7 @@
mResumed = false;
}
timestampUs -= previousPausedDurationUs;
- ALOGV("time stamp: %lld, previous paused duration: %lld",
+ ALOGV("time stamp: %" PRId64 ", previous paused duration: %" PRId64,
timestampUs, previousPausedDurationUs);
if (timestampUs > maxTimestampUs) {
maxTimestampUs = timestampUs;
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index 9ab1417..9c64d72 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -547,7 +547,7 @@
}
sp<AMessage> notify = mNotify->dup();
- notify->setInt32("what", ACodec::kWhatBuffersAllocated);
+ notify->setInt32("what", CodecBase::kWhatBuffersAllocated);
notify->setInt32("portIndex", portIndex);
@@ -2035,6 +2035,10 @@
err = setupAVCEncoderParameters(msg);
break;
+ case OMX_VIDEO_CodingHEVC:
+ err = setupHEVCEncoderParameters(msg);
+ break;
+
case OMX_VIDEO_CodingVP8:
case OMX_VIDEO_CodingVP9:
err = setupVPXEncoderParameters(msg);
@@ -2371,6 +2375,62 @@
return configureBitrate(bitrate, bitrateMode);
}
+status_t ACodec::setupHEVCEncoderParameters(const sp<AMessage> &msg) {
+ int32_t bitrate, iFrameInterval;
+ if (!msg->findInt32("bitrate", &bitrate)
+ || !msg->findInt32("i-frame-interval", &iFrameInterval)) {
+ return INVALID_OPERATION;
+ }
+
+ OMX_VIDEO_CONTROLRATETYPE bitrateMode = getBitrateMode(msg);
+
+ 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_HEVCTYPE hevcType;
+ InitOMXParams(&hevcType);
+ hevcType.nPortIndex = kPortIndexOutput;
+
+ status_t err = OK;
+ err = mOMX->getParameter(
+ mNode, (OMX_INDEXTYPE)OMX_IndexParamVideoHevc, &hevcType, sizeof(hevcType));
+ if (err != OK) {
+ return err;
+ }
+
+ 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;
+ }
+
+ hevcType.eProfile = static_cast<OMX_VIDEO_HEVCPROFILETYPE>(profile);
+ hevcType.eLevel = static_cast<OMX_VIDEO_HEVCLEVELTYPE>(level);
+ }
+
+ // TODO: Need OMX structure definition for setting iFrameInterval
+
+ err = mOMX->setParameter(
+ mNode, (OMX_INDEXTYPE)OMX_IndexParamVideoHevc, &hevcType, sizeof(hevcType));
+ if (err != OK) {
+ return err;
+ }
+
+ return configureBitrate(bitrate, bitrateMode);
+}
+
status_t ACodec::setupVPXEncoderParameters(const sp<AMessage> &msg) {
int32_t bitrate;
int32_t iFrameInterval = 0;
@@ -2945,7 +3005,7 @@
void ACodec::signalError(OMX_ERRORTYPE error, status_t internalError) {
sp<AMessage> notify = mNotify->dup();
- notify->setInt32("what", ACodec::kWhatError);
+ notify->setInt32("what", CodecBase::kWhatError);
notify->setInt32("omx-error", error);
notify->setInt32("err", internalError);
notify->post();
@@ -3338,7 +3398,7 @@
CHECK_EQ((int)info->mStatus, (int)BufferInfo::OWNED_BY_US);
sp<AMessage> notify = mCodec->mNotify->dup();
- notify->setInt32("what", ACodec::kWhatFillThisBuffer);
+ notify->setInt32("what", CodecBase::kWhatFillThisBuffer);
notify->setInt32("buffer-id", info->mBufferID);
info->mData->meta()->clear();
@@ -3633,7 +3693,7 @@
info->mData->meta()->setInt64("timeUs", timeUs);
sp<AMessage> notify = mCodec->mNotify->dup();
- notify->setInt32("what", ACodec::kWhatDrainThisBuffer);
+ notify->setInt32("what", CodecBase::kWhatDrainThisBuffer);
notify->setInt32("buffer-id", info->mBufferID);
notify->setBuffer("buffer", info->mData);
notify->setInt32("flags", flags);
@@ -3650,7 +3710,7 @@
ALOGV("[%s] saw output EOS", mCodec->mComponentName.c_str());
sp<AMessage> notify = mCodec->mNotify->dup();
- notify->setInt32("what", ACodec::kWhatEOS);
+ notify->setInt32("what", CodecBase::kWhatEOS);
notify->setInt32("err", mCodec->mInputEOSResult);
notify->post();
@@ -3831,7 +3891,7 @@
"cannot keep component allocated on shutdown in Uninitialized state");
sp<AMessage> notify = mCodec->mNotify->dup();
- notify->setInt32("what", ACodec::kWhatShutdownCompleted);
+ notify->setInt32("what", CodecBase::kWhatShutdownCompleted);
notify->post();
handled = true;
@@ -3841,7 +3901,7 @@
case ACodec::kWhatFlush:
{
sp<AMessage> notify = mCodec->mNotify->dup();
- notify->setInt32("what", ACodec::kWhatFlushCompleted);
+ notify->setInt32("what", CodecBase::kWhatFlushCompleted);
notify->post();
handled = true;
@@ -3963,7 +4023,7 @@
{
sp<AMessage> notify = mCodec->mNotify->dup();
- notify->setInt32("what", ACodec::kWhatComponentAllocated);
+ notify->setInt32("what", CodecBase::kWhatComponentAllocated);
notify->setString("componentName", mCodec->mComponentName.c_str());
notify->post();
}
@@ -4013,7 +4073,7 @@
if (mCodec->mExplicitShutdown) {
sp<AMessage> notify = mCodec->mNotify->dup();
- notify->setInt32("what", ACodec::kWhatShutdownCompleted);
+ notify->setInt32("what", CodecBase::kWhatShutdownCompleted);
notify->post();
mCodec->mExplicitShutdown = false;
}
@@ -4060,7 +4120,7 @@
case ACodec::kWhatFlush:
{
sp<AMessage> notify = mCodec->mNotify->dup();
- notify->setInt32("what", ACodec::kWhatFlushCompleted);
+ notify->setInt32("what", CodecBase::kWhatFlushCompleted);
notify->post();
handled = true;
@@ -4109,7 +4169,7 @@
{
sp<AMessage> notify = mCodec->mNotify->dup();
- notify->setInt32("what", ACodec::kWhatComponentConfigured);
+ notify->setInt32("what", CodecBase::kWhatComponentConfigured);
notify->setMessage("input-format", mCodec->mInputFormat);
notify->setMessage("output-format", mCodec->mOutputFormat);
notify->post();
@@ -4123,7 +4183,7 @@
ALOGV("onCreateInputSurface");
sp<AMessage> notify = mCodec->mNotify->dup();
- notify->setInt32("what", ACodec::kWhatInputSurfaceCreated);
+ notify->setInt32("what", CodecBase::kWhatInputSurfaceCreated);
sp<IGraphicBufferProducer> bufferProducer;
status_t err;
@@ -4277,7 +4337,7 @@
{
// We haven't even started yet, so we're flushed alright...
sp<AMessage> notify = mCodec->mNotify->dup();
- notify->setInt32("what", ACodec::kWhatFlushCompleted);
+ notify->setInt32("what", CodecBase::kWhatFlushCompleted);
notify->post();
return true;
}
@@ -4338,7 +4398,7 @@
{
// We haven't even started yet, so we're flushed alright...
sp<AMessage> notify = mCodec->mNotify->dup();
- notify->setInt32("what", ACodec::kWhatFlushCompleted);
+ notify->setInt32("what", CodecBase::kWhatFlushCompleted);
notify->post();
return true;
@@ -4641,7 +4701,7 @@
void ACodec::onSignalEndOfInputStream() {
sp<AMessage> notify = mNotify->dup();
- notify->setInt32("what", ACodec::kWhatSignaledInputEOS);
+ notify->setInt32("what", CodecBase::kWhatSignaledInputEOS);
status_t err = mOMX->signalEndOfInputStream(mNode);
if (err != OK) {
@@ -5071,7 +5131,7 @@
mCodec->waitUntilAllPossibleNativeWindowBuffersAreReturnedToUs();
sp<AMessage> notify = mCodec->mNotify->dup();
- notify->setInt32("what", ACodec::kWhatFlushCompleted);
+ notify->setInt32("what", CodecBase::kWhatFlushCompleted);
notify->post();
mCodec->mPortEOS[kPortIndexInput] =
diff --git a/media/libstagefright/AMRWriter.cpp b/media/libstagefright/AMRWriter.cpp
index 653ca36..9aa7d95 100644
--- a/media/libstagefright/AMRWriter.cpp
+++ b/media/libstagefright/AMRWriter.cpp
@@ -14,6 +14,12 @@
* limitations under the License.
*/
+#include <fcntl.h>
+#include <inttypes.h>
+#include <sys/prctl.h>
+#include <sys/stat.h>
+#include <sys/types.h>
+
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/AMRWriter.h>
#include <media/stagefright/MediaBuffer.h>
@@ -22,10 +28,6 @@
#include <media/stagefright/MediaSource.h>
#include <media/stagefright/MetaData.h>
#include <media/mediarecorder.h>
-#include <sys/prctl.h>
-#include <sys/types.h>
-#include <sys/stat.h>
-#include <fcntl.h>
namespace android {
@@ -235,7 +237,7 @@
mResumed = false;
}
timestampUs -= previousPausedDurationUs;
- ALOGV("time stamp: %lld, previous paused duration: %lld",
+ ALOGV("time stamp: %" PRId64 ", previous paused duration: %" PRId64,
timestampUs, previousPausedDurationUs);
if (timestampUs > maxTimestampUs) {
maxTimestampUs = timestampUs;
diff --git a/media/libstagefright/Android.mk b/media/libstagefright/Android.mk
index d9e39ff..99c8e9f 100644
--- a/media/libstagefright/Android.mk
+++ b/media/libstagefright/Android.mk
@@ -15,6 +15,7 @@
CameraSource.cpp \
CameraSourceTimeLapse.cpp \
ClockEstimator.cpp \
+ CodecBase.cpp \
DataSource.cpp \
DataURISource.cpp \
DRMExtractor.cpp \
@@ -68,6 +69,7 @@
$(TOP)/external/flac/include \
$(TOP)/external/tremolo \
$(TOP)/external/openssl/include \
+ $(TOP)/external/libvpx/libwebm \
LOCAL_SHARED_LIBRARIES := \
libbinder \
diff --git a/media/libstagefright/AudioPlayer.cpp b/media/libstagefright/AudioPlayer.cpp
index 2669849..fdac8fc 100644
--- a/media/libstagefright/AudioPlayer.cpp
+++ b/media/libstagefright/AudioPlayer.cpp
@@ -14,6 +14,8 @@
* limitations under the License.
*/
+#include <inttypes.h>
+
//#define LOG_NDEBUG 0
#define LOG_TAG "AudioPlayer"
#include <utils/Log.h>
@@ -566,12 +568,12 @@
int64_t timeToCompletionUs =
(1000000ll * numFramesPendingPlayout) / mSampleRate;
- ALOGV("total number of frames played: %lld (%lld us)",
+ ALOGV("total number of frames played: %" PRId64 " (%lld us)",
(mNumFramesPlayed + numAdditionalFrames),
1000000ll * (mNumFramesPlayed + numAdditionalFrames)
/ mSampleRate);
- ALOGV("%d frames left to play, %lld us (%.2f secs)",
+ ALOGV("%d frames left to play, %" PRId64 " us (%.2f secs)",
numFramesPendingPlayout,
timeToCompletionUs, timeToCompletionUs / 1E6);
@@ -628,7 +630,7 @@
mPositionTimeRealUs =
((mNumFramesPlayed + size_done / mFrameSize) * 1000000)
/ mSampleRate;
- ALOGV("buffer->size() = %d, "
+ ALOGV("buffer->size() = %zu, "
"mPositionTimeMediaUs=%.2f mPositionTimeRealUs=%.2f",
mInputBuffer->range_length(),
mPositionTimeMediaUs / 1E6, mPositionTimeRealUs / 1E6);
@@ -746,7 +748,7 @@
// HAL position is relative to the first buffer we sent at mStartPosUs
const int64_t renderedDuration = mStartPosUs + playedUs;
- ALOGV("getOutputPlayPositionUs_l %lld", renderedDuration);
+ ALOGV("getOutputPlayPositionUs_l %" PRId64, renderedDuration);
return renderedDuration;
}
@@ -758,7 +760,7 @@
return mSeekTimeUs;
}
mPositionTimeRealUs = getOutputPlayPositionUs_l();
- ALOGV("getMediaTimeUs getOutputPlayPositionUs_l() mPositionTimeRealUs %lld",
+ ALOGV("getMediaTimeUs getOutputPlayPositionUs_l() mPositionTimeRealUs %" PRId64,
mPositionTimeRealUs);
return mPositionTimeRealUs;
}
@@ -796,7 +798,7 @@
status_t AudioPlayer::seekTo(int64_t time_us) {
Mutex::Autolock autoLock(mLock);
- ALOGV("seekTo( %lld )", time_us);
+ ALOGV("seekTo( %" PRId64 " )", time_us);
mSeeking = true;
mPositionTimeRealUs = mPositionTimeMediaUs = -1;
diff --git a/media/libstagefright/AudioSource.cpp b/media/libstagefright/AudioSource.cpp
index d0e0e8e..d9aed01 100644
--- a/media/libstagefright/AudioSource.cpp
+++ b/media/libstagefright/AudioSource.cpp
@@ -14,6 +14,9 @@
* limitations under the License.
*/
+#include <inttypes.h>
+#include <stdlib.h>
+
//#define LOG_NDEBUG 0
#define LOG_TAG "AudioSource"
#include <utils/Log.h>
@@ -26,7 +29,6 @@
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/ALooper.h>
#include <cutils/properties.h>
-#include <stdlib.h>
namespace android {
@@ -136,7 +138,7 @@
}
void AudioSource::waitOutstandingEncodingFrames_l() {
- ALOGV("waitOutstandingEncodingFrames_l: %lld", mNumClientOwnedBuffers);
+ ALOGV("waitOutstandingEncodingFrames_l: %" PRId64, mNumClientOwnedBuffers);
while (mNumClientOwnedBuffers > 0) {
mFrameEncodingCompletionCondition.wait(mLock);
}
@@ -269,7 +271,7 @@
status_t AudioSource::dataCallback(const AudioRecord::Buffer& audioBuffer) {
int64_t timeUs = systemTime() / 1000ll;
- ALOGV("dataCallbackTimestamp: %lld us", timeUs);
+ ALOGV("dataCallbackTimestamp: %" PRId64 " us", timeUs);
Mutex::Autolock autoLock(mLock);
if (!mStarted) {
ALOGW("Spurious callback from AudioRecord. Drop the audio data.");
@@ -279,7 +281,7 @@
// Drop retrieved and previously lost audio data.
if (mNumFramesReceived == 0 && timeUs < mStartTimeUs) {
(void) mRecord->getInputFramesLost();
- ALOGV("Drop audio data at %lld/%lld us", timeUs, mStartTimeUs);
+ ALOGV("Drop audio data at %" PRId64 "/%" PRId64 " us", timeUs, mStartTimeUs);
return OK;
}
diff --git a/media/libstagefright/AwesomePlayer.cpp b/media/libstagefright/AwesomePlayer.cpp
index d679be1..63799e1 100644
--- a/media/libstagefright/AwesomePlayer.cpp
+++ b/media/libstagefright/AwesomePlayer.cpp
@@ -19,7 +19,9 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "AwesomePlayer"
#define ATRACE_TAG ATRACE_TAG_VIDEO
+
#include <inttypes.h>
+
#include <utils/Log.h>
#include <utils/Trace.h>
@@ -408,6 +410,13 @@
totalBitRate += bitrate;
}
+ sp<MetaData> fileMeta = mExtractor->getMetaData();
+ if (fileMeta != NULL) {
+ int64_t duration;
+ if (fileMeta->findInt64(kKeyDuration, &duration)) {
+ mDurationUs = duration;
+ }
+ }
mBitrate = totalBitRate;
@@ -1708,7 +1717,7 @@
}
if (mAudioPlayer != NULL) {
- ALOGV("seeking audio to %lld us (%.2f secs).", videoTimeUs, videoTimeUs / 1E6);
+ ALOGV("seeking audio to %" PRId64 " us (%.2f secs).", videoTimeUs, videoTimeUs / 1E6);
// If we don't have a video time, seek audio to the originally
// requested seek time instead.
@@ -1772,7 +1781,7 @@
if (!mVideoBuffer) {
MediaSource::ReadOptions options;
if (mSeeking != NO_SEEK) {
- ALOGV("seeking to %lld us (%.2f secs)", mSeekTimeUs, mSeekTimeUs / 1E6);
+ ALOGV("seeking to %" PRId64 " us (%.2f secs)", mSeekTimeUs, mSeekTimeUs / 1E6);
options.setSeekTo(
mSeekTimeUs,
@@ -1842,7 +1851,7 @@
if (mSeeking == SEEK_VIDEO_ONLY) {
if (mSeekTimeUs > timeUs) {
- ALOGI("XXX mSeekTimeUs = %lld us, timeUs = %lld us",
+ ALOGI("XXX mSeekTimeUs = %" PRId64 " us, timeUs = %" PRId64 " us",
mSeekTimeUs, timeUs);
}
}
@@ -1940,13 +1949,13 @@
if (latenessUs > 40000) {
// We're more than 40ms late.
- ALOGV("we're late by %lld us (%.2f secs)",
+ ALOGV("we're late by %" PRId64 " us (%.2f secs)",
latenessUs, latenessUs / 1E6);
if (!(mFlags & SLOW_DECODER_HACK)
|| mSinceLastDropped > FRAME_DROP_FREQ)
{
- ALOGV("we're late by %lld us (%.2f secs) dropping "
+ ALOGV("we're late by %" PRId64 " us (%.2f secs) dropping "
"one after %d frames",
latenessUs, latenessUs / 1E6, mSinceLastDropped);
@@ -2308,12 +2317,12 @@
if (finalStatus != OK
|| (metaDataSize >= 0
- && cachedDataRemaining >= metaDataSize)
+ && (off64_t)cachedDataRemaining >= metaDataSize)
|| (mFlags & PREPARE_CANCELLED)) {
break;
}
- ALOGV("now cached %d bytes of data", cachedDataRemaining);
+ ALOGV("now cached %zu bytes of data", cachedDataRemaining);
if (metaDataSize < 0
&& cachedDataRemaining >= kMinBytesForSniffing) {
@@ -2692,7 +2701,7 @@
status_t AwesomePlayer::selectTrack(size_t trackIndex, bool select) {
ATRACE_CALL();
- ALOGV("selectTrack: trackIndex = %d and select=%d", trackIndex, select);
+ ALOGV("selectTrack: trackIndex = %zu and select=%d", trackIndex, select);
Mutex::Autolock autoLock(mLock);
size_t trackCount = mExtractor->countTracks();
if (mTextDriver != NULL) {
diff --git a/media/libstagefright/CameraSource.cpp b/media/libstagefright/CameraSource.cpp
index b31e9e8..2b50763 100644
--- a/media/libstagefright/CameraSource.cpp
+++ b/media/libstagefright/CameraSource.cpp
@@ -14,6 +14,8 @@
* limitations under the License.
*/
+#include <inttypes.h>
+
//#define LOG_NDEBUG 0
#define LOG_TAG "CameraSource"
#include <utils/Log.h>
@@ -77,7 +79,7 @@
void CameraSourceListener::postData(int32_t msgType, const sp<IMemory> &dataPtr,
camera_frame_metadata_t * /* metadata */) {
- ALOGV("postData(%d, ptr:%p, size:%d)",
+ ALOGV("postData(%d, ptr:%p, size:%zu)",
msgType, dataPtr->pointer(), dataPtr->size());
sp<CameraSource> source = mSource.promote();
@@ -711,7 +713,7 @@
if (NO_ERROR !=
mFrameCompleteCondition.waitRelative(mLock,
mTimeBetweenFrameCaptureUs * 1000LL + CAMERA_SOURCE_TIMEOUT_NS)) {
- ALOGW("Timed out waiting for outstanding frames being encoded: %d",
+ ALOGW("Timed out waiting for outstanding frames being encoded: %zu",
mFramesBeingEncoded.size());
}
}
@@ -722,7 +724,7 @@
}
if (mCollectStats) {
- ALOGI("Frames received/encoded/dropped: %d/%d/%d in %lld us",
+ ALOGI("Frames received/encoded/dropped: %d/%d/%d in %" PRId64 " us",
mNumFramesReceived, mNumFramesEncoded, mNumFramesDropped,
mLastFrameTimestampUs - mFirstFrameTimeUs);
}
@@ -809,7 +811,7 @@
ALOGW("camera recording proxy is gone");
return ERROR_END_OF_STREAM;
}
- ALOGW("Timed out waiting for incoming camera video frames: %lld us",
+ ALOGW("Timed out waiting for incoming camera video frames: %" PRId64 " us",
mLastFrameTimestampUs);
}
}
@@ -832,10 +834,10 @@
void CameraSource::dataCallbackTimestamp(int64_t timestampUs,
int32_t msgType, const sp<IMemory> &data) {
- ALOGV("dataCallbackTimestamp: timestamp %lld us", timestampUs);
+ ALOGV("dataCallbackTimestamp: timestamp %" PRId64 " us", timestampUs);
Mutex::Autolock autoLock(mLock);
if (!mStarted || (mNumFramesReceived == 0 && timestampUs < mStartTimeUs)) {
- ALOGV("Drop frame at %lld/%lld us", timestampUs, mStartTimeUs);
+ ALOGV("Drop frame at %" PRId64 "/%" PRId64 " us", timestampUs, mStartTimeUs);
releaseOneRecordingFrame(data);
return;
}
@@ -874,7 +876,7 @@
mFramesReceived.push_back(data);
int64_t timeUs = mStartTimeUs + (timestampUs - mFirstFrameTimeUs);
mFrameTimes.push_back(timeUs);
- ALOGV("initial delay: %lld, current time stamp: %lld",
+ ALOGV("initial delay: %" PRId64 ", current time stamp: %" PRId64,
mStartTimeUs, timeUs);
mFrameAvailableCondition.signal();
}
diff --git a/media/libstagefright/CameraSourceTimeLapse.cpp b/media/libstagefright/CameraSourceTimeLapse.cpp
index 15ba967..0acd9d0 100644
--- a/media/libstagefright/CameraSourceTimeLapse.cpp
+++ b/media/libstagefright/CameraSourceTimeLapse.cpp
@@ -14,6 +14,8 @@
* limitations under the License.
*/
+#include <inttypes.h>
+
//#define LOG_NDEBUG 0
#define LOG_TAG "CameraSourceTimeLapse"
@@ -79,7 +81,7 @@
mSkipCurrentFrame(false) {
mTimeBetweenFrameCaptureUs = timeBetweenFrameCaptureUs;
- ALOGD("starting time lapse mode: %lld us",
+ ALOGD("starting time lapse mode: %" PRId64 " us",
mTimeBetweenFrameCaptureUs);
mVideoWidth = videoSize.width;
@@ -266,7 +268,7 @@
// Really make sure that this video recording frame will not be dropped.
if (*timestampUs < mStartTimeUs) {
- ALOGI("set timestampUs to start time stamp %lld us", mStartTimeUs);
+ ALOGI("set timestampUs to start time stamp %" PRId64 " us", mStartTimeUs);
*timestampUs = mStartTimeUs;
}
return false;
diff --git a/media/libstagefright/CodecBase.cpp b/media/libstagefright/CodecBase.cpp
new file mode 100644
index 0000000..f729d4d
--- /dev/null
+++ b/media/libstagefright/CodecBase.cpp
@@ -0,0 +1,38 @@
+/*
+ * Copyright (C) 2014 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 "CodecBase"
+
+#include <inttypes.h>
+
+#include <media/stagefright/CodecBase.h>
+
+namespace android {
+
+CodecBase::CodecBase() {
+}
+
+CodecBase::~CodecBase() {
+}
+
+CodecBase::PortDescription::PortDescription() {
+}
+
+CodecBase::PortDescription::~PortDescription() {
+}
+
+} // namespace android
diff --git a/media/libstagefright/DataURISource.cpp b/media/libstagefright/DataURISource.cpp
index 377bc85..2c39314 100644
--- a/media/libstagefright/DataURISource.cpp
+++ b/media/libstagefright/DataURISource.cpp
@@ -85,7 +85,7 @@
}
ssize_t DataURISource::readAt(off64_t offset, void *data, size_t size) {
- if (offset >= mBuffer->size()) {
+ if ((offset < 0) || (offset >= (off64_t)mBuffer->size())) {
return 0;
}
diff --git a/media/libstagefright/ESDS.cpp b/media/libstagefright/ESDS.cpp
index 4a0c35c..427bf7b 100644
--- a/media/libstagefright/ESDS.cpp
+++ b/media/libstagefright/ESDS.cpp
@@ -91,7 +91,7 @@
}
while (more);
- ALOGV("tag=0x%02x data_size=%d", *tag, *data_size);
+ ALOGV("tag=0x%02x data_size=%zu", *tag, *data_size);
if (*data_size > size) {
return ERROR_MALFORMED;
diff --git a/media/libstagefright/MPEG2TSWriter.cpp b/media/libstagefright/MPEG2TSWriter.cpp
index 78c12e1..9856f92 100644
--- a/media/libstagefright/MPEG2TSWriter.cpp
+++ b/media/libstagefright/MPEG2TSWriter.cpp
@@ -682,7 +682,7 @@
break;
}
- ALOGV("writing access unit at time %.2f secs (index %d)",
+ ALOGV("writing access unit at time %.2f secs (index %zu)",
minTimeUs / 1E6, minIndex);
source = mSources.editItemAt(minIndex);
diff --git a/media/libstagefright/MPEG4Extractor.cpp b/media/libstagefright/MPEG4Extractor.cpp
index 297f4fc..207acc8 100644
--- a/media/libstagefright/MPEG4Extractor.cpp
+++ b/media/libstagefright/MPEG4Extractor.cpp
@@ -16,17 +16,19 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "MPEG4Extractor"
+
+#include <ctype.h>
+#include <inttypes.h>
+#include <stdint.h>
+#include <stdlib.h>
+#include <string.h>
+
#include <utils/Log.h>
#include "include/MPEG4Extractor.h"
#include "include/SampleTable.h"
#include "include/ESDS.h"
-#include <ctype.h>
-#include <stdint.h>
-#include <stdlib.h>
-#include <string.h>
-
#include <media/stagefright/foundation/ABitReader.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
@@ -51,6 +53,7 @@
int32_t timeScale,
const sp<SampleTable> &sampleTable,
Vector<SidxEntry> &sidx,
+ const Trex *trex,
off64_t firstMoofOffset);
virtual status_t start(MetaData *params = NULL);
@@ -74,6 +77,7 @@
uint32_t mCurrentSampleIndex;
uint32_t mCurrentFragmentIndex;
Vector<SidxEntry> &mSegments;
+ const Trex *mTrex;
off64_t mFirstMoofOffset;
off64_t mCurrentMoofOffset;
off64_t mNextMoofOffset;
@@ -141,6 +145,7 @@
off64_t offset;
size_t size;
uint32_t duration;
+ int32_t compositionOffset;
uint8_t iv[16];
Vector<size_t> clearsizes;
Vector<size_t> encryptedsizes;
@@ -343,8 +348,7 @@
}
MPEG4Extractor::MPEG4Extractor(const sp<DataSource> &source)
- : mSidxDuration(0),
- mMoofOffset(0),
+ : mMoofOffset(0),
mDataSource(source),
mInitCheck(NO_INIT),
mHasVideo(false),
@@ -369,7 +373,7 @@
SINF *sinf = mFirstSINF;
while (sinf) {
SINF *next = sinf->next;
- delete sinf->IPMPData;
+ delete[] sinf->IPMPData;
delete sinf;
sinf = next;
}
@@ -409,7 +413,7 @@
track = track->next;
}
- ALOGV("MPEG4Extractor::countTracks: %d tracks", n);
+ ALOGV("MPEG4Extractor::countTracks: %zu tracks", n);
return n;
}
@@ -694,7 +698,10 @@
return ERROR_MALFORMED;
}
sinf->len = dataLen - 3;
- sinf->IPMPData = new char[sinf->len];
+ sinf->IPMPData = new (std::nothrow) char[sinf->len];
+ if (sinf->IPMPData == NULL) {
+ return ERROR_MALFORMED;
+ }
data_offset += 2;
if (mDataSource->readAt(data_offset, sinf->IPMPData, sinf->len) < sinf->len) {
@@ -787,7 +794,7 @@
}
} else if (chunk_size < 8) {
// The smallest valid chunk is 8 bytes long.
- ALOGE("invalid chunk size: %d", int(chunk_size));
+ ALOGE("invalid chunk size: %" PRIu64, chunk_size);
return ERROR_MALFORMED;
}
@@ -798,7 +805,7 @@
#if 0
static const char kWhitespace[] = " ";
const char *indent = &kWhitespace[sizeof(kWhitespace) - 1 - 2 * depth];
- printf("%sfound chunk '%s' of size %lld\n", indent, chunk, chunk_size);
+ printf("%sfound chunk '%s' of size %" PRIu64 "\n", indent, chunk, chunk_size);
char buffer[256];
size_t n = chunk_size;
@@ -854,7 +861,7 @@
case FOURCC('e', 'd', 't', 's'):
{
if (chunk_type == FOURCC('s', 't', 'b', 'l')) {
- ALOGV("sampleTable chunk is %d bytes long.", (size_t)chunk_size);
+ ALOGV("sampleTable chunk is %" PRIu64 " bytes long.", chunk_size);
if (mDataSource->flags()
& (DataSource::kWantsPrefetching
@@ -1112,7 +1119,10 @@
return ERROR_MALFORMED;
}
- pssh.data = new uint8_t[pssh.datalen];
+ pssh.data = new (std::nothrow) uint8_t[pssh.datalen];
+ if (pssh.data == NULL) {
+ return ERROR_MALFORMED;
+ }
ALOGV("allocated pssh @ %p", pssh.data);
ssize_t requested = (ssize_t) pssh.datalen;
if (mDataSource->readAt(data_offset + 24, pssh.data, requested) < requested) {
@@ -1157,6 +1167,8 @@
mLastTrack->timescale = ntohl(timescale);
+ // 14496-12 says all ones means indeterminate, but some files seem to use
+ // 0 instead. We treat both the same.
int64_t duration = 0;
if (version == 1) {
if (mDataSource->readAt(
@@ -1164,7 +1176,9 @@
< (ssize_t)sizeof(duration)) {
return ERROR_IO;
}
- duration = ntoh64(duration);
+ if (duration != -1) {
+ duration = ntoh64(duration);
+ }
} else {
uint32_t duration32;
if (mDataSource->readAt(
@@ -1172,13 +1186,14 @@
< (ssize_t)sizeof(duration32)) {
return ERROR_IO;
}
- // ffmpeg sets duration to -1, which is incorrect.
if (duration32 != 0xffffffff) {
duration = ntohl(duration32);
}
}
- mLastTrack->meta->setInt64(
- kKeyDuration, (duration * 1000000) / mLastTrack->timescale);
+ if (duration != 0) {
+ mLastTrack->meta->setInt64(
+ kKeyDuration, (duration * 1000000) / mLastTrack->timescale);
+ }
uint8_t lang[2];
off64_t lang_offset;
@@ -1718,11 +1733,11 @@
{
*offset += chunk_size;
- if (chunk_data_size < 24) {
+ if (chunk_data_size < 32) {
return ERROR_MALFORMED;
}
- uint8_t header[24];
+ uint8_t header[32];
if (mDataSource->readAt(
data_offset, header, sizeof(header))
< (ssize_t)sizeof(header)) {
@@ -1730,14 +1745,27 @@
}
uint64_t creationTime;
+ uint64_t duration = 0;
if (header[0] == 1) {
creationTime = U64_AT(&header[4]);
mHeaderTimescale = U32_AT(&header[20]);
+ duration = U64_AT(&header[24]);
+ if (duration == 0xffffffffffffffff) {
+ duration = 0;
+ }
} else if (header[0] != 0) {
return ERROR_MALFORMED;
} else {
creationTime = U32_AT(&header[4]);
mHeaderTimescale = U32_AT(&header[12]);
+ uint32_t d32 = U32_AT(&header[16]);
+ if (d32 == 0xffffffff) {
+ d32 = 0;
+ }
+ duration = d32;
+ }
+ if (duration != 0) {
+ mFileMetaData->setInt64(kKeyDuration, duration * 1000000 / mHeaderTimescale);
}
String8 s;
@@ -1748,6 +1776,50 @@
break;
}
+ case FOURCC('m', 'e', 'h', 'd'):
+ {
+ *offset += chunk_size;
+
+ if (chunk_data_size < 8) {
+ return ERROR_MALFORMED;
+ }
+
+ uint8_t flags[4];
+ if (mDataSource->readAt(
+ data_offset, flags, sizeof(flags))
+ < (ssize_t)sizeof(flags)) {
+ return ERROR_IO;
+ }
+
+ uint64_t duration = 0;
+ if (flags[0] == 1) {
+ // 64 bit
+ if (chunk_data_size < 12) {
+ return ERROR_MALFORMED;
+ }
+ mDataSource->getUInt64(data_offset + 4, &duration);
+ if (duration == 0xffffffffffffffff) {
+ duration = 0;
+ }
+ } else if (flags[0] == 0) {
+ // 32 bit
+ uint32_t d32;
+ mDataSource->getUInt32(data_offset + 4, &d32);
+ if (d32 == 0xffffffff) {
+ d32 = 0;
+ }
+ duration = d32;
+ } else {
+ return ERROR_MALFORMED;
+ }
+
+ if (duration != 0) {
+ mFileMetaData->setInt64(kKeyDuration, duration * 1000000 / mHeaderTimescale);
+ }
+
+ break;
+ }
+
case FOURCC('m', 'd', 'a', 't'):
{
ALOGV("mdat chunk, drm: %d", mIsDrm);
@@ -1784,6 +1856,26 @@
break;
}
+ case FOURCC('t', 'r', 'e', 'x'):
+ {
+ *offset += chunk_size;
+
+ if (chunk_data_size < 24) {
+ return ERROR_IO;
+ }
+ uint32_t duration;
+ Trex trex;
+ if (!mDataSource->getUInt32(data_offset + 4, &trex.track_ID) ||
+ !mDataSource->getUInt32(data_offset + 8, &trex.default_sample_description_index) ||
+ !mDataSource->getUInt32(data_offset + 12, &trex.default_sample_duration) ||
+ !mDataSource->getUInt32(data_offset + 16, &trex.default_sample_size) ||
+ !mDataSource->getUInt32(data_offset + 20, &trex.default_sample_flags)) {
+ return ERROR_IO;
+ }
+ mTrex.add(trex);
+ break;
+ }
+
case FOURCC('t', 'x', '3', 'g'):
{
uint32_t type;
@@ -1794,7 +1886,10 @@
size = 0;
}
- uint8_t *buffer = new uint8_t[size + chunk_size];
+ uint8_t *buffer = new (std::nothrow) uint8_t[size + chunk_size];
+ if (buffer == NULL) {
+ return ERROR_MALFORMED;
+ }
if (size > 0) {
memcpy(buffer, data, size);
@@ -1959,7 +2054,7 @@
offset += 16;
size -= 16;
}
- ALOGV("sidx pres/off: %Ld/%Ld", earliestPresentationTime, firstOffset);
+ ALOGV("sidx pres/off: %" PRIu64 "/%" PRIu64, earliestPresentationTime, firstOffset);
if (size < 4) {
return -EINVAL;
@@ -2005,12 +2100,11 @@
mSidxEntries.add(se);
}
- mSidxDuration = total_duration * 1000000 / timeScale;
- ALOGV("duration: %lld", mSidxDuration);
+ uint64_t sidxDuration = total_duration * 1000000 / timeScale;
int64_t metaDuration;
if (!mLastTrack->meta->findInt64(kKeyDuration, &metaDuration) || metaDuration == 0) {
- mLastTrack->meta->setInt64(kKeyDuration, mSidxDuration);
+ mLastTrack->meta->setInt64(kKeyDuration, sidxDuration);
}
return OK;
}
@@ -2111,7 +2205,10 @@
return ERROR_MALFORMED;
}
- uint8_t *buffer = new uint8_t[size + 1];
+ uint8_t *buffer = new (std::nothrow) uint8_t[size + 1];
+ if (buffer == NULL) {
+ return ERROR_MALFORMED;
+ }
if (mDataSource->readAt(
offset, buffer, size) != (ssize_t)size) {
delete[] buffer;
@@ -2298,7 +2395,10 @@
return ERROR_MALFORMED;
}
- uint8_t *buffer = new uint8_t[size];
+ uint8_t *buffer = new (std::nothrow) uint8_t[size];
+ if (buffer == NULL) {
+ return ERROR_MALFORMED;
+ }
if (mDataSource->readAt(
offset, buffer, size) != (ssize_t)size) {
delete[] buffer;
@@ -2477,11 +2577,24 @@
return NULL;
}
- ALOGV("getTrack called, pssh: %d", mPssh.size());
+
+ Trex *trex = NULL;
+ int32_t trackId;
+ if (track->meta->findInt32(kKeyTrackID, &trackId)) {
+ for (size_t i = 0; i < mTrex.size(); i++) {
+ Trex *t = &mTrex.editItemAt(index);
+ if (t->track_ID == (uint32_t) trackId) {
+ trex = t;
+ break;
+ }
+ }
+ }
+
+ ALOGV("getTrack called, pssh: %zu", mPssh.size());
return new MPEG4Source(
track->meta, mDataSource, track->timescale, track->sampleTable,
- mSidxEntries, mMoofOffset);
+ mSidxEntries, trex, mMoofOffset);
}
// static
@@ -2811,6 +2924,7 @@
int32_t timeScale,
const sp<SampleTable> &sampleTable,
Vector<SidxEntry> &sidx,
+ const Trex *trex,
off64_t firstMoofOffset)
: mFormat(format),
mDataSource(dataSource),
@@ -2819,6 +2933,7 @@
mCurrentSampleIndex(0),
mCurrentFragmentIndex(0),
mSegments(sidx),
+ mTrex(trex),
mFirstMoofOffset(firstMoofOffset),
mCurrentMoofOffset(firstMoofOffset),
mCurrentTime(0),
@@ -2835,6 +2950,8 @@
mWantsNALFragments(false),
mSrcBuffer(NULL) {
+ memset(&mTrackFragmentHeaderInfo, 0, sizeof(mTrackFragmentHeaderInfo));
+
mFormat->findInt32(kKeyCryptoMode, &mCryptoMode);
mDefaultIVSize = 0;
mFormat->findInt32(kKeyCryptoDefaultIVSize, &mDefaultIVSize);
@@ -2917,7 +3034,11 @@
mGroup->add_buffer(new MediaBuffer(max_size));
- mSrcBuffer = new uint8_t[max_size];
+ mSrcBuffer = new (std::nothrow) uint8_t[max_size];
+ if (mSrcBuffer == NULL) {
+ // file probably specified a bad max size
+ return ERROR_MALFORMED;
+ }
mStarted = true;
@@ -3402,8 +3523,8 @@
} else if (mTrackFragmentHeaderInfo.mFlags
& TrackFragmentHeaderInfo::kDefaultSampleDurationPresent) {
sampleDuration = mTrackFragmentHeaderInfo.mDefaultSampleDuration;
- } else {
- sampleDuration = mTrackFragmentHeaderInfo.mDefaultSampleDuration;
+ } else if (mTrex) {
+ sampleDuration = mTrex->default_sample_duration;
}
if (flags & kSampleSizePresent) {
@@ -3430,7 +3551,7 @@
sampleCtsOffset = 0;
}
- if (size < sampleCount * bytesPerSample) {
+ if (size < (off64_t)sampleCount * bytesPerSample) {
return -EINVAL;
}
@@ -3464,7 +3585,7 @@
offset += 4;
}
- ALOGV("adding sample %d at offset 0x%08llx, size %u, duration %u, "
+ ALOGV("adding sample %d at offset 0x%08" PRIx64 ", size %u, duration %u, "
" flags 0x%08x", i + 1,
dataOffset, sampleSize, sampleDuration,
(flags & kFirstSampleFlagsPresent) && i == 0
@@ -3472,6 +3593,7 @@
tmp.offset = dataOffset;
tmp.size = sampleSize;
tmp.duration = sampleDuration;
+ tmp.compositionOffset = sampleCtsOffset;
mCurrentSamples.add(tmp);
dataOffset += sampleSize;
@@ -3874,7 +3996,7 @@
const Sample *smpl = &mCurrentSamples[mCurrentSampleIndex];
offset = smpl->offset;
size = smpl->size;
- cts = mCurrentTime;
+ cts = mCurrentTime + smpl->compositionOffset;
mCurrentTime += smpl->duration;
isSyncSample = (mCurrentSampleIndex == 0); // XXX
@@ -4181,7 +4303,7 @@
char chunkstring[5];
MakeFourCCString(chunkType, chunkstring);
- ALOGV("saw chunk type %s, size %lld @ %lld", chunkstring, chunkSize, offset);
+ ALOGV("saw chunk type %s, size %" PRIu64 " @ %lld", chunkstring, chunkSize, offset);
switch (chunkType) {
case FOURCC('f', 't', 'y', 'p'):
{
diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp
index 24e53b3..4b8440b 100644
--- a/media/libstagefright/MPEG4Writer.cpp
+++ b/media/libstagefright/MPEG4Writer.cpp
@@ -16,13 +16,17 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "MPEG4Writer"
-#include <inttypes.h>
-#include <utils/Log.h>
#include <arpa/inet.h>
-
+#include <fcntl.h>
+#include <inttypes.h>
#include <pthread.h>
#include <sys/prctl.h>
+#include <sys/stat.h>
+#include <sys/types.h>
+#include <unistd.h>
+
+#include <utils/Log.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MPEG4Writer.h>
@@ -34,10 +38,6 @@
#include <media/stagefright/Utils.h>
#include <media/mediarecorder.h>
#include <cutils/properties.h>
-#include <sys/types.h>
-#include <sys/stat.h>
-#include <fcntl.h>
-#include <unistd.h>
#include "include/ESDS.h"
@@ -441,7 +441,7 @@
// At most 2 tracks can be supported.
if (mTracks.size() >= 2) {
- ALOGE("Too many tracks (%d) to add", mTracks.size());
+ ALOGE("Too many tracks (%zu) to add", mTracks.size());
return ERROR_UNSUPPORTED;
}
@@ -555,8 +555,8 @@
size = MAX_MOOV_BOX_SIZE;
}
- ALOGI("limits: %lld/%lld bytes/us, bit rate: %d bps and the estimated"
- " moov size %lld bytes",
+ ALOGI("limits: %" PRId64 "/%" PRId64 " bytes/us, bit rate: %d bps and the"
+ " estimated moov size %" PRId64 " bytes",
mMaxFileSizeLimitBytes, mMaxFileDurationLimitUs, bitRate, size);
return factor * size;
}
@@ -592,8 +592,8 @@
// If file size is set to be larger than the 32 bit file
// size limit, treat it as an error.
if (mMaxFileSizeLimitBytes > kMax32BitFileSize) {
- ALOGW("32-bit file size limit (%lld bytes) too big. "
- "It is changed to %lld bytes",
+ ALOGW("32-bit file size limit (%" PRId64 " bytes) too big. "
+ "It is changed to %" PRId64 " bytes",
mMaxFileSizeLimitBytes, kMax32BitFileSize);
mMaxFileSizeLimitBytes = kMax32BitFileSize;
}
@@ -854,7 +854,7 @@
}
if (mTracks.size() > 1) {
- ALOGD("Duration from tracks range is [%lld, %lld] us",
+ ALOGD("Duration from tracks range is [%" PRId64 ", %" PRId64 "] us",
minDurationUs, maxDurationUs);
}
@@ -1321,12 +1321,12 @@
}
void MPEG4Writer::setStartTimestampUs(int64_t timeUs) {
- ALOGI("setStartTimestampUs: %lld", timeUs);
+ ALOGI("setStartTimestampUs: %" PRId64, timeUs);
CHECK_GE(timeUs, 0ll);
Mutex::Autolock autoLock(mLock);
if (mStartTimestampUs < 0 || mStartTimestampUs > timeUs) {
mStartTimestampUs = timeUs;
- ALOGI("Earliest track starting time: %lld", mStartTimestampUs);
+ ALOGI("Earliest track starting time: %" PRId64, mStartTimestampUs);
}
}
@@ -1527,7 +1527,7 @@
{
int64_t timeUs;
if (params && params->findInt64(kKeyTrackTimeStatus, &timeUs)) {
- ALOGV("Receive request to track progress status for every %lld us", timeUs);
+ ALOGV("Receive request to track progress status for every %" PRId64 " us", timeUs);
mTrackEveryTimeDurationUs = timeUs;
mTrackingProgressStatus = true;
}
@@ -1561,7 +1561,7 @@
}
void MPEG4Writer::writeChunkToFile(Chunk* chunk) {
- ALOGV("writeChunkToFile: %lld from %s track",
+ ALOGV("writeChunkToFile: %" PRId64 " from %s track",
chunk->mTimeStampUs, chunk->mTrack->isAudio()? "audio": "video");
int32_t isFirstSample = true;
@@ -1737,7 +1737,7 @@
startTimeOffsetUs = kInitialDelayTimeUs;
}
startTimeUs += startTimeOffsetUs;
- ALOGI("Start time offset: %lld us", startTimeOffsetUs);
+ ALOGI("Start time offset: %" PRId64 " us", startTimeOffsetUs);
}
meta->setInt64(kKeyTime, startTimeUs);
@@ -1817,7 +1817,7 @@
static const uint8_t *findNextStartCode(
const uint8_t *data, size_t length) {
- ALOGV("findNextStartCode: %p %d", data, length);
+ ALOGV("findNextStartCode: %p %zu", data, length);
size_t bytesLeft = length;
while (bytesLeft > 4 &&
@@ -2238,7 +2238,7 @@
}
timestampUs = decodingTimeUs;
- ALOGV("decoding time: %lld and ctts offset time: %lld",
+ ALOGV("decoding time: %" PRId64 " and ctts offset time: %" PRId64,
timestampUs, cttsOffsetTimeUs);
// Update ctts box table if necessary
@@ -2291,7 +2291,7 @@
return ERROR_MALFORMED;
}
- ALOGV("%s media time stamp: %lld and previous paused duration %lld",
+ ALOGV("%s media time stamp: %" PRId64 " and previous paused duration %" PRId64,
trackName, timestampUs, previousPausedDurationUs);
if (timestampUs > mTrackDurationUs) {
mTrackDurationUs = timestampUs;
@@ -2306,7 +2306,7 @@
((timestampUs * mTimeScale + 500000LL) / 1000000LL -
(lastTimestampUs * mTimeScale + 500000LL) / 1000000LL);
if (currDurationTicks < 0ll) {
- ALOGE("timestampUs %lld < lastTimestampUs %lld for %s track",
+ ALOGE("timestampUs %" PRId64 " < lastTimestampUs %" PRId64 " for %s track",
timestampUs, lastTimestampUs, trackName);
copy->release();
return UNKNOWN_ERROR;
@@ -2347,7 +2347,7 @@
}
previousSampleSize = sampleSize;
}
- ALOGV("%s timestampUs/lastTimestampUs: %lld/%lld",
+ ALOGV("%s timestampUs/lastTimestampUs: %" PRId64 "/%" PRId64,
trackName, timestampUs, lastTimestampUs);
lastDurationUs = timestampUs - lastTimestampUs;
lastDurationTicks = currDurationTicks;
@@ -2455,7 +2455,7 @@
ALOGI("Received total/0-length (%d/%d) buffers and encoded %d frames. - %s",
count, nZeroLengthFrames, mStszTableEntries->count(), trackName);
if (mIsAudio) {
- ALOGI("Audio track drift time: %lld us", mOwner->getDriftTimeUs());
+ ALOGI("Audio track drift time: %" PRId64 " us", mOwner->getDriftTimeUs());
}
if (err == ERROR_END_OF_STREAM) {
@@ -2538,11 +2538,11 @@
}
void MPEG4Writer::Track::trackProgressStatus(int64_t timeUs, status_t err) {
- ALOGV("trackProgressStatus: %lld us", timeUs);
+ ALOGV("trackProgressStatus: %" PRId64 " us", timeUs);
if (mTrackEveryTimeDurationUs > 0 &&
timeUs - mPreviousTrackTimeUs >= mTrackEveryTimeDurationUs) {
- ALOGV("Fire time tracking progress status at %lld us", timeUs);
+ ALOGV("Fire time tracking progress status at %" PRId64 " us", timeUs);
mOwner->trackProgressStatus(mTrackId, timeUs - mPreviousTrackTimeUs, err);
mPreviousTrackTimeUs = timeUs;
}
@@ -2576,13 +2576,13 @@
}
void MPEG4Writer::setDriftTimeUs(int64_t driftTimeUs) {
- ALOGV("setDriftTimeUs: %lld us", driftTimeUs);
+ ALOGV("setDriftTimeUs: %" PRId64 " us", driftTimeUs);
Mutex::Autolock autolock(mLock);
mDriftTimeUs = driftTimeUs;
}
int64_t MPEG4Writer::getDriftTimeUs() {
- ALOGV("getDriftTimeUs: %lld us", mDriftTimeUs);
+ ALOGV("getDriftTimeUs: %" PRId64 " us", mDriftTimeUs);
Mutex::Autolock autolock(mLock);
return mDriftTimeUs;
}
@@ -3038,7 +3038,7 @@
return;
}
- ALOGV("ctts box has %d entries with range [%lld, %lld]",
+ ALOGV("ctts box has %d entries with range [%" PRId64 ", %" PRId64 "]",
mCttsTableEntries->count(), mMinCttsOffsetTimeUs, mMaxCttsOffsetTimeUs);
mOwner->beginBox("ctts");
diff --git a/media/libstagefright/MediaBuffer.cpp b/media/libstagefright/MediaBuffer.cpp
index 8af0880..1f80a47 100644
--- a/media/libstagefright/MediaBuffer.cpp
+++ b/media/libstagefright/MediaBuffer.cpp
@@ -134,7 +134,7 @@
void MediaBuffer::set_range(size_t offset, size_t length) {
if ((mGraphicBuffer == NULL) && (offset + length > mSize)) {
- ALOGE("offset = %d, length = %d, mSize = %d", offset, length, mSize);
+ ALOGE("offset = %zu, length = %zu, mSize = %zu", offset, length, mSize);
}
CHECK((mGraphicBuffer != NULL) || (offset + length <= mSize));
diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index b9c5904..6e3d6f8 100644
--- a/media/libstagefright/MediaCodec.cpp
+++ b/media/libstagefright/MediaCodec.cpp
@@ -67,7 +67,7 @@
MediaCodec::MediaCodec(const sp<ALooper> &looper)
: mState(UNINITIALIZED),
mLooper(looper),
- mCodec(new ACodec),
+ mCodec(NULL),
mReplyID(0),
mFlags(0),
mSoftRenderer(NULL),
@@ -103,6 +103,7 @@
// quickly, violating the OpenMAX specs, until that is remedied
// we need to invest in an extra looper to free the main event
// queue.
+ mCodec = new ACodec;
bool needDedicatedLooper = false;
if (nameIsType && !strncasecmp(name, "video/", 6)) {
needDedicatedLooper = true;
@@ -541,7 +542,7 @@
CHECK(msg->findInt32("what", &what));
switch (what) {
- case ACodec::kWhatError:
+ case CodecBase::kWhatError:
{
int32_t omxError, internalError;
CHECK(msg->findInt32("omx-error", &omxError));
@@ -638,7 +639,7 @@
break;
}
- case ACodec::kWhatComponentAllocated:
+ case CodecBase::kWhatComponentAllocated:
{
CHECK_EQ(mState, INITIALIZING);
setState(INITIALIZED);
@@ -661,7 +662,7 @@
break;
}
- case ACodec::kWhatComponentConfigured:
+ case CodecBase::kWhatComponentConfigured:
{
CHECK_EQ(mState, CONFIGURING);
setState(CONFIGURED);
@@ -676,9 +677,9 @@
break;
}
- case ACodec::kWhatInputSurfaceCreated:
+ case CodecBase::kWhatInputSurfaceCreated:
{
- // response to ACodec::kWhatCreateInputSurface
+ // response to initiateCreateInputSurface()
status_t err = NO_ERROR;
sp<AMessage> response = new AMessage();
if (!msg->findInt32("err", &err)) {
@@ -694,9 +695,9 @@
break;
}
- case ACodec::kWhatSignaledInputEOS:
+ case CodecBase::kWhatSignaledInputEOS:
{
- // response to ACodec::kWhatSignalEndOfInputStream
+ // response to signalEndOfInputStream()
sp<AMessage> response = new AMessage();
status_t err;
if (msg->findInt32("err", &err)) {
@@ -707,7 +708,7 @@
}
- case ACodec::kWhatBuffersAllocated:
+ case CodecBase::kWhatBuffersAllocated:
{
int32_t portIndex;
CHECK(msg->findInt32("portIndex", &portIndex));
@@ -725,8 +726,8 @@
sp<RefBase> obj;
CHECK(msg->findObject("portDesc", &obj));
- sp<ACodec::PortDescription> portDesc =
- static_cast<ACodec::PortDescription *>(obj.get());
+ sp<CodecBase::PortDescription> portDesc =
+ static_cast<CodecBase::PortDescription *>(obj.get());
size_t numBuffers = portDesc->countBuffers();
@@ -759,7 +760,7 @@
break;
}
- case ACodec::kWhatOutputFormatChanged:
+ case CodecBase::kWhatOutputFormatChanged:
{
ALOGV("codec output format changed");
@@ -810,7 +811,7 @@
break;
}
- case ACodec::kWhatFillThisBuffer:
+ case CodecBase::kWhatFillThisBuffer:
{
/* size_t index = */updateBuffers(kPortIndexInput, msg);
@@ -857,7 +858,7 @@
break;
}
- case ACodec::kWhatDrainThisBuffer:
+ case CodecBase::kWhatDrainThisBuffer:
{
/* size_t index = */updateBuffers(kPortIndexOutput, msg);
@@ -908,14 +909,14 @@
break;
}
- case ACodec::kWhatEOS:
+ case CodecBase::kWhatEOS:
{
// We already notify the client of this by using the
// corresponding flag in "onOutputBufferReady".
break;
}
- case ACodec::kWhatShutdownCompleted:
+ case CodecBase::kWhatShutdownCompleted:
{
if (mState == STOPPING) {
setState(INITIALIZED);
@@ -928,7 +929,7 @@
break;
}
- case ACodec::kWhatFlushCompleted:
+ case CodecBase::kWhatFlushCompleted:
{
CHECK_EQ(mState, FLUSHING);
setState(STARTED);
@@ -1456,7 +1457,7 @@
++i;
}
- ALOGV("Found %u pieces of codec specific data.", mCSD.size());
+ ALOGV("Found %zu pieces of codec specific data.", mCSD.size());
}
status_t MediaCodec::queueCSDInputBuffer(size_t bufferIndex) {
diff --git a/media/libstagefright/MediaCodecSource.cpp b/media/libstagefright/MediaCodecSource.cpp
index 924173c..9868ecf 100644
--- a/media/libstagefright/MediaCodecSource.cpp
+++ b/media/libstagefright/MediaCodecSource.cpp
@@ -17,6 +17,9 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "MediaCodecSource"
#define DEBUG_DRIFT_TIME 0
+
+#include <inttypes.h>
+
#include <gui/IGraphicBufferProducer.h>
#include <gui/Surface.h>
#include <media/ICrypto.h>
@@ -677,7 +680,7 @@
}
mbuf->meta_data()->setInt64(kKeyDecodingTime, decodingTimeUs);
- ALOGV("[video] time %lld us (%.2f secs), dts/pts diff %lld",
+ ALOGV("[video] time %" PRId64 " us (%.2f secs), dts/pts diff %" PRId64,
timeUs, timeUs / 1E6, decodingTimeUs - timeUs);
} else {
int64_t driftTimeUs = 0;
@@ -687,7 +690,7 @@
mDriftTimeQueue.erase(mDriftTimeQueue.begin());
mbuf->meta_data()->setInt64(kKeyDriftTime, driftTimeUs);
#endif // DEBUG_DRIFT_TIME
- ALOGV("[audio] time %lld us (%.2f secs), drift %lld",
+ ALOGV("[audio] time %" PRId64 " us (%.2f secs), drift %" PRId64,
timeUs, timeUs / 1E6, driftTimeUs);
}
mbuf->meta_data()->setInt64(kKeyTime, timeUs);
diff --git a/media/libstagefright/MediaMuxer.cpp b/media/libstagefright/MediaMuxer.cpp
index 90335ee..c7c6f34 100644
--- a/media/libstagefright/MediaMuxer.cpp
+++ b/media/libstagefright/MediaMuxer.cpp
@@ -176,7 +176,7 @@
}
if (trackIndex >= mTrackList.size()) {
- ALOGE("WriteSampleData() get an invalid index %d", trackIndex);
+ ALOGE("WriteSampleData() get an invalid index %zu", trackIndex);
return -EINVAL;
}
diff --git a/media/libstagefright/NuCachedSource2.cpp b/media/libstagefright/NuCachedSource2.cpp
index 61cf0ad..c1feff8 100644
--- a/media/libstagefright/NuCachedSource2.cpp
+++ b/media/libstagefright/NuCachedSource2.cpp
@@ -14,6 +14,8 @@
* limitations under the License.
*/
+#include <inttypes.h>
+
//#define LOG_NDEBUG 0
#define LOG_TAG "NuCachedSource2"
#include <utils/Log.h>
@@ -135,7 +137,7 @@
}
void PageCache::copy(size_t from, void *data, size_t size) {
- ALOGV("copy from %d size %d", from, size);
+ ALOGV("copy from %zu size %zu", from, size);
if (size == 0) {
return;
@@ -333,7 +335,7 @@
mNumRetriesLeft = 0;
}
- ALOGE("source returned error %d, %d retries left", n, mNumRetriesLeft);
+ ALOGE("source returned error %zd, %d retries left", n, mNumRetriesLeft);
mCache->releasePage(page);
} else if (n == 0) {
ALOGI("ERROR_END_OF_STREAM");
@@ -464,14 +466,14 @@
size_t actualBytes = mCache->releaseFromStart(maxBytes);
mCacheOffset += actualBytes;
- ALOGI("restarting prefetcher, totalSize = %d", mCache->totalSize());
+ ALOGI("restarting prefetcher, totalSize = %zu", mCache->totalSize());
mFetching = true;
}
ssize_t NuCachedSource2::readAt(off64_t offset, void *data, size_t size) {
Mutex::Autolock autoSerializer(mSerializer);
- ALOGV("readAt offset %lld, size %d", offset, size);
+ ALOGV("readAt offset %lld, size %zu", offset, size);
Mutex::Autolock autoLock(mLock);
@@ -539,7 +541,7 @@
ssize_t NuCachedSource2::readInternal(off64_t offset, void *data, size_t size) {
CHECK_LE(size, (size_t)mHighwaterThresholdBytes);
- ALOGV("readInternal offset %lld size %d", offset, size);
+ ALOGV("readInternal offset %lld size %zu", offset, size);
Mutex::Autolock autoLock(mLock);
@@ -679,7 +681,7 @@
mKeepAliveIntervalUs = kDefaultKeepAliveIntervalUs;
}
- ALOGV("lowwater = %d bytes, highwater = %d bytes, keepalive = %lld us",
+ ALOGV("lowwater = %zu bytes, highwater = %zu bytes, keepalive = %" PRId64 " us",
mLowwaterThresholdBytes,
mHighwaterThresholdBytes,
mKeepAliveIntervalUs);
diff --git a/media/libstagefright/NuMediaExtractor.cpp b/media/libstagefright/NuMediaExtractor.cpp
index 64f56e9..f24cf3a 100644
--- a/media/libstagefright/NuMediaExtractor.cpp
+++ b/media/libstagefright/NuMediaExtractor.cpp
@@ -389,7 +389,7 @@
info->mFinalResult = err;
if (info->mFinalResult != ERROR_END_OF_STREAM) {
- ALOGW("read on track %d failed with error %d",
+ ALOGW("read on track %zu failed with error %d",
info->mTrackIndex, err);
}
diff --git a/media/libstagefright/OggExtractor.cpp b/media/libstagefright/OggExtractor.cpp
index f3eeb03..8c15929 100644
--- a/media/libstagefright/OggExtractor.cpp
+++ b/media/libstagefright/OggExtractor.cpp
@@ -338,7 +338,7 @@
const TOCEntry &entry = mTableOfContents.itemAt(left);
- ALOGV("seeking to entry %d / %d at offset %lld",
+ ALOGV("seeking to entry %zu / %zu at offset %lld",
left, mTableOfContents.size(), entry.mPageOffset);
return seekToOffset(entry.mPageOffset);
@@ -381,7 +381,7 @@
ssize_t n;
if ((n = mSource->readAt(offset, header, sizeof(header)))
< (ssize_t)sizeof(header)) {
- ALOGV("failed to read %zu bytes at offset 0x%016llx, got %d bytes",
+ ALOGV("failed to read %zu bytes at offset 0x%016llx, got %zd bytes",
sizeof(header), offset, n);
if (n < 0) {
@@ -505,7 +505,7 @@
packetSize);
if (n < (ssize_t)packetSize) {
- ALOGV("failed to read %zu bytes at 0x%016llx, got %d bytes",
+ ALOGV("failed to read %zu bytes at 0x%016llx, got %zd bytes",
packetSize, dataOffset, n);
return ERROR_IO;
}
@@ -546,7 +546,7 @@
buffer = NULL;
}
- ALOGV("readPage returned %d", n);
+ ALOGV("readPage returned %zd", n);
return n < 0 ? n : (status_t)ERROR_END_OF_STREAM;
}
@@ -590,7 +590,7 @@
if ((err = readNextPacket(&packet)) != OK) {
return err;
}
- ALOGV("read packet of size %d\n", packet->range_length());
+ ALOGV("read packet of size %zu\n", packet->range_length());
err = verifyHeader(packet, 1);
packet->release();
packet = NULL;
@@ -601,7 +601,7 @@
if ((err = readNextPacket(&packet)) != OK) {
return err;
}
- ALOGV("read packet of size %d\n", packet->range_length());
+ ALOGV("read packet of size %zu\n", packet->range_length());
err = verifyHeader(packet, 3);
packet->release();
packet = NULL;
@@ -612,7 +612,7 @@
if ((err = readNextPacket(&packet)) != OK) {
return err;
}
- ALOGV("read packet of size %d\n", packet->range_length());
+ ALOGV("read packet of size %zu\n", packet->range_length());
err = verifyHeader(packet, 5);
packet->release();
packet = NULL;
@@ -903,7 +903,7 @@
return;
}
- ALOGV("got flac of size %d", flacSize);
+ ALOGV("got flac of size %zu", flacSize);
uint32_t picType;
uint32_t typeLen;
@@ -953,7 +953,7 @@
goto exit;
}
- ALOGV("got image data, %d trailing bytes",
+ ALOGV("got image data, %zu trailing bytes",
flacSize - 32 - typeLen - descLen - dataLen);
fileMeta->setData(
diff --git a/media/libstagefright/StagefrightMetadataRetriever.cpp b/media/libstagefright/StagefrightMetadataRetriever.cpp
index af96225..8cc41e7 100644
--- a/media/libstagefright/StagefrightMetadataRetriever.cpp
+++ b/media/libstagefright/StagefrightMetadataRetriever.cpp
@@ -16,7 +16,9 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "StagefrightMetadataRetriever"
+
#include <inttypes.h>
+
#include <utils/Log.h>
#include "include/StagefrightMetadataRetriever.h"
@@ -87,7 +89,7 @@
int fd, int64_t offset, int64_t length) {
fd = dup(fd);
- ALOGV("setDataSource(%d, %lld, %lld)", fd, offset, length);
+ ALOGV("setDataSource(%d, %" PRId64 ", %" PRId64 ")", fd, offset, length);
mParsedMetaData = false;
mMetaData.clear();
@@ -242,7 +244,7 @@
const char *mime;
CHECK(trackMeta->findCString(kKeyMIMEType, &mime));
- ALOGV("thumbNailTime = %lld us, timeUs = %lld us, mime = %s",
+ ALOGV("thumbNailTime = %" PRId64 " us, timeUs = %" PRId64 " us, mime = %s",
thumbNailTime, timeUs, mime);
}
}
@@ -325,7 +327,7 @@
VideoFrame *StagefrightMetadataRetriever::getFrameAtTime(
int64_t timeUs, int option) {
- ALOGV("getFrameAtTime: %lld us option: %d", timeUs, option);
+ ALOGV("getFrameAtTime: %" PRId64 " us option: %d", timeUs, option);
if (mExtractor.get() == NULL) {
ALOGV("no extractor.");
diff --git a/media/libstagefright/SurfaceMediaSource.cpp b/media/libstagefright/SurfaceMediaSource.cpp
index 62aea36..4e1c65c 100644
--- a/media/libstagefright/SurfaceMediaSource.cpp
+++ b/media/libstagefright/SurfaceMediaSource.cpp
@@ -16,6 +16,8 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "SurfaceMediaSource"
+#include <inttypes.h>
+
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/SurfaceMediaSource.h>
#include <media/stagefright/MediaDefs.h>
@@ -179,7 +181,7 @@
}
status_t SurfaceMediaSource::setMaxAcquiredBufferCount(size_t count) {
- ALOGV("setMaxAcquiredBufferCount(%d)", count);
+ ALOGV("setMaxAcquiredBufferCount(%zu)", count);
Mutex::Autolock lock(mMutex);
CHECK_GT(count, 1);
@@ -209,7 +211,7 @@
mFrameAvailableCondition.signal();
while (mNumPendingBuffers > 0) {
- ALOGI("Still waiting for %d buffers to be returned.",
+ ALOGI("Still waiting for %zu buffers to be returned.",
mNumPendingBuffers);
#if DEBUG_PENDING_BUFFERS
@@ -269,7 +271,7 @@
memcpy(data, &type, 4);
memcpy(data + 4, &bufferHandle, sizeof(buffer_handle_t));
- ALOGV("handle = %p, , offset = %d, length = %d",
+ ALOGV("handle = %p, , offset = %zu, length = %zu",
bufferHandle, (*buffer)->range_length(), (*buffer)->range_offset());
}
@@ -363,7 +365,7 @@
(*buffer)->setObserver(this);
(*buffer)->add_ref();
(*buffer)->meta_data()->setInt64(kKeyTime, mCurrentTimestamp / 1000);
- ALOGV("Frames encoded = %d, timestamp = %lld, time diff = %lld",
+ ALOGV("Frames encoded = %d, timestamp = %" PRId64 ", time diff = %" PRId64,
mNumFramesEncoded, mCurrentTimestamp / 1000,
mCurrentTimestamp / 1000 - prevTimeStamp / 1000);
diff --git a/media/libstagefright/TimedEventQueue.cpp b/media/libstagefright/TimedEventQueue.cpp
index 3d2eb1f..da50c56 100644
--- a/media/libstagefright/TimedEventQueue.cpp
+++ b/media/libstagefright/TimedEventQueue.cpp
@@ -17,7 +17,11 @@
#undef __STRICT_ANSI__
#define __STDINT_LIMITS
#define __STDC_LIMIT_MACROS
+
+#include <inttypes.h>
#include <stdint.h>
+#include <sys/prctl.h>
+#include <sys/time.h>
//#define LOG_NDEBUG 0
#define LOG_TAG "TimedEventQueue"
@@ -26,9 +30,6 @@
#include "include/TimedEventQueue.h"
-#include <sys/prctl.h>
-#include <sys/time.h>
-
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/ALooper.h>
#include <binder/IServiceManager.h>
@@ -258,7 +259,7 @@
static int64_t kMaxTimeoutUs = 10000000ll; // 10 secs
bool timeoutCapped = false;
if (delay_us > kMaxTimeoutUs) {
- ALOGW("delay_us exceeds max timeout: %lld us", delay_us);
+ ALOGW("delay_us exceeds max timeout: %" PRId64 " us", delay_us);
// We'll never block for more than 10 secs, instead
// we will split up the full timeout into chunks of
diff --git a/media/libstagefright/Utils.cpp b/media/libstagefright/Utils.cpp
index 7ff31a1..d53051e 100644
--- a/media/libstagefright/Utils.cpp
+++ b/media/libstagefright/Utils.cpp
@@ -335,7 +335,7 @@
// there can't be another param here, so use all the rest
i = csd0->size();
}
- ALOGV("block at %d, last was %d", i, lastparamoffset);
+ ALOGV("block at %zu, last was %d", i, lastparamoffset);
if (lastparamoffset > 0) {
int size = i - lastparamoffset;
avcc[avccidx++] = size >> 8;
@@ -366,7 +366,7 @@
// there can't be another param here, so use all the rest
i = csd1->size();
}
- ALOGV("block at %d, last was %d", i, lastparamoffset);
+ ALOGV("block at %zu, last was %d", i, lastparamoffset);
if (lastparamoffset > 0) {
int size = i - lastparamoffset;
avcc[avccidx++] = size >> 8;
diff --git a/media/libstagefright/VBRISeeker.cpp b/media/libstagefright/VBRISeeker.cpp
index af858b9..e988f6d 100644
--- a/media/libstagefright/VBRISeeker.cpp
+++ b/media/libstagefright/VBRISeeker.cpp
@@ -16,6 +16,9 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "VBRISeeker"
+
+#include <inttypes.h>
+
#include <utils/Log.h>
#include "include/VBRISeeker.h"
@@ -75,7 +78,7 @@
size_t entrySize = U16_AT(&vbriHeader[22]);
size_t scale = U16_AT(&vbriHeader[20]);
- ALOGV("%d entries, scale=%d, size_per_entry=%d",
+ ALOGV("%zu entries, scale=%zu, size_per_entry=%zu",
numEntries,
scale,
entrySize);
@@ -119,7 +122,7 @@
seeker->mSegments.push(numBytes);
- ALOGV("entry #%d: %u offset 0x%016llx", i, numBytes, offset);
+ ALOGV("entry #%zu: %u offset 0x%016llx", i, numBytes, offset);
offset += numBytes;
}
@@ -160,7 +163,7 @@
*pos += mSegments.itemAt(segmentIndex++);
}
- ALOGV("getOffsetForTime %lld us => 0x%016llx", *timeUs, *pos);
+ ALOGV("getOffsetForTime %" PRId64 " us => 0x%016llx", *timeUs, *pos);
*timeUs = nowUs;
diff --git a/media/libstagefright/WAVExtractor.cpp b/media/libstagefright/WAVExtractor.cpp
index fe9058b..7124fd3 100644
--- a/media/libstagefright/WAVExtractor.cpp
+++ b/media/libstagefright/WAVExtractor.cpp
@@ -414,7 +414,7 @@
} else {
pos = (seekTimeUs * mSampleRate) / 1000000 * mNumChannels * (mBitsPerSample >> 3);
}
- if (pos > mSize) {
+ if (pos > (off64_t)mSize) {
pos = mSize;
}
mCurrentPos = pos + mOffset;
diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
index 64bf2b6..ab30865 100644
--- a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
+++ b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
@@ -65,10 +65,9 @@
mInputBufferCount(0),
mOutputBufferCount(0),
mSignalledError(false),
+ mLastInHeader(NULL),
+ mCurrentInputTime(0),
mOutputPortSettingsChange(NONE) {
- for (unsigned int i = 0; i < kNumDelayBlocksMax; i++) {
- mAnchorTimeUs[i] = 0;
- }
initPorts();
CHECK_EQ(initDecoder(), (status_t)OK);
}
@@ -496,14 +495,11 @@
} else {
mEndOfInput = false;
}
- if (inHeader->nOffset == 0) { // TODO: does nOffset != 0 happen?
- mAnchorTimeUs[mInputBufferCount % kNumDelayBlocksMax] =
- inHeader->nTimeStamp;
- }
if (inHeader->nFilledLen == 0) {
inInfo->mOwnedByUs = false;
inQueue.erase(inQueue.begin());
+ mLastInHeader = NULL;
inInfo = NULL;
notifyEmptyBufferDone(inHeader);
inHeader = NULL;
@@ -568,6 +564,18 @@
INT prevSampleRate = mStreamInfo->sampleRate;
INT prevNumChannels = mStreamInfo->numChannels;
+ if (inHeader != mLastInHeader) {
+ mLastInHeader = inHeader;
+ mCurrentInputTime = inHeader->nTimeStamp;
+ } else {
+ if (mStreamInfo->sampleRate) {
+ mCurrentInputTime += mStreamInfo->aacSamplesPerFrame *
+ 1000000ll / mStreamInfo->sampleRate;
+ } else {
+ ALOGW("no sample rate yet");
+ }
+ }
+ mAnchorTimes.add(mCurrentInputTime);
aacDecoder_Fill(mAACDecoder,
inBuffer,
inBufferLength,
@@ -671,6 +679,7 @@
inInfo->mOwnedByUs = false;
mInputBufferCount++;
inQueue.erase(inQueue.begin());
+ mLastInHeader = NULL;
inInfo = NULL;
notifyEmptyBufferDone(inHeader);
inHeader = NULL;
@@ -687,11 +696,12 @@
inInfo->mOwnedByUs = false;
mInputBufferCount++;
inQueue.erase(inQueue.begin());
+ mLastInHeader = NULL;
inInfo = NULL;
notifyEmptyBufferDone(inHeader);
inHeader = NULL;
} else {
- ALOGW("inHeader->nFilledLen = %d", inHeader->nFilledLen);
+ ALOGV("inHeader->nFilledLen = %d", inHeader->nFilledLen);
}
}
}
@@ -779,8 +789,8 @@
outHeader->nFlags = 0;
}
- outHeader->nTimeStamp = mAnchorTimeUs[mOutputBufferCount
- % kNumDelayBlocksMax];
+ outHeader->nTimeStamp = mAnchorTimes.isEmpty() ? 0 : mAnchorTimes.itemAt(0);
+ mAnchorTimes.removeAt(0);
mOutputBufferCount++;
outInfo->mOwnedByUs = false;
@@ -820,8 +830,8 @@
outHeader->nFilledLen = 0;
outHeader->nFlags = OMX_BUFFERFLAG_EOS;
- outHeader->nTimeStamp = mAnchorTimeUs[mOutputBufferCount
- % kNumDelayBlocksMax];
+ outHeader->nTimeStamp = mAnchorTimes.itemAt(0);
+ mAnchorTimes.removeAt(0);
mOutputBufferCount++;
outInfo->mOwnedByUs = false;
@@ -842,6 +852,8 @@
// depend on fragments from the last one decoded.
// drain all existing data
drainDecoder();
+ mAnchorTimes.clear();
+ mLastInHeader = NULL;
} else {
while (outputDelayRingBufferSamplesAvailable() > 0) {
int32_t ns = outputDelayRingBufferGetSamples(0,
@@ -896,6 +908,8 @@
mOutputDelayRingBufferReadPos = 0;
mEndOfInput = false;
mEndOfOutput = false;
+ mAnchorTimes.clear();
+ mLastInHeader = NULL;
// To make the codec behave the same before and after a reset, we need to invalidate the
// streaminfo struct. This does that:
diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.h b/media/libstagefright/codecs/aacdec/SoftAAC2.h
index 5cde03a..865bd15 100644
--- a/media/libstagefright/codecs/aacdec/SoftAAC2.h
+++ b/media/libstagefright/codecs/aacdec/SoftAAC2.h
@@ -58,7 +58,9 @@
size_t mInputBufferCount;
size_t mOutputBufferCount;
bool mSignalledError;
- int64_t mAnchorTimeUs[kNumDelayBlocksMax];
+ OMX_BUFFERHEADERTYPE *mLastInHeader;
+ int64_t mCurrentInputTime;
+ Vector<int64_t> mAnchorTimes;
CDrcPresModeWrapper mDrcWrap;
diff --git a/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp b/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp
index dc38ea8..cabd6bd 100644
--- a/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp
+++ b/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp
@@ -379,7 +379,7 @@
}
default:
{
- ALOGE("Wrong number of temporal layers %u", mTemporalLayers);
+ ALOGE("Wrong number of temporal layers %zu", mTemporalLayers);
return UNKNOWN_ERROR;
}
}
diff --git a/media/libstagefright/foundation/ABitReader.cpp b/media/libstagefright/foundation/ABitReader.cpp
index 5499c32..beb5cc0 100644
--- a/media/libstagefright/foundation/ABitReader.cpp
+++ b/media/libstagefright/foundation/ABitReader.cpp
@@ -27,6 +27,9 @@
mNumBitsLeft(0) {
}
+ABitReader::~ABitReader() {
+}
+
void ABitReader::fillReservoir() {
CHECK_GT(mSize, 0u);
@@ -99,4 +102,69 @@
return mData - (mNumBitsLeft + 7) / 8;
}
+NALBitReader::NALBitReader(const uint8_t *data, size_t size)
+ : ABitReader(data, size),
+ mNumZeros(0) {
+}
+
+bool NALBitReader::atLeastNumBitsLeft(size_t n) const {
+ // check against raw size and reservoir bits first
+ size_t numBits = numBitsLeft();
+ if (n > numBits) {
+ return false;
+ }
+
+ ssize_t numBitsRemaining = n - mNumBitsLeft;
+
+ size_t size = mSize;
+ const uint8_t *data = mData;
+ int32_t numZeros = mNumZeros;
+ while (size > 0 && numBitsRemaining > 0) {
+ bool isEmulationPreventionByte = (numZeros >= 2 && *data == 3);
+
+ if (*data == 0) {
+ ++numZeros;
+ } else {
+ numZeros = 0;
+ }
+
+ if (!isEmulationPreventionByte) {
+ numBitsRemaining -= 8;
+ }
+
+ ++data;
+ --size;
+ }
+
+ return (numBitsRemaining <= 0);
+}
+
+void NALBitReader::fillReservoir() {
+ CHECK_GT(mSize, 0u);
+
+ mReservoir = 0;
+ size_t i = 0;
+ while (mSize > 0 && i < 4) {
+ bool isEmulationPreventionByte = (mNumZeros >= 2 && *mData == 3);
+
+ if (*mData == 0) {
+ ++mNumZeros;
+ } else {
+ mNumZeros = 0;
+ }
+
+ // skip emulation_prevention_three_byte
+ if (!isEmulationPreventionByte) {
+ mReservoir = (mReservoir << 8) | *mData;
+ ++i;
+ }
+
+ ++mData;
+ --mSize;
+ }
+
+ mNumBitsLeft = 8 * i;
+ mReservoir <<= 32 - mNumBitsLeft;
+}
+
} // namespace android
diff --git a/media/libstagefright/foundation/AHierarchicalStateMachine.cpp b/media/libstagefright/foundation/AHierarchicalStateMachine.cpp
index f7a00d8..5f7c70d 100644
--- a/media/libstagefright/foundation/AHierarchicalStateMachine.cpp
+++ b/media/libstagefright/foundation/AHierarchicalStateMachine.cpp
@@ -51,7 +51,7 @@
AHierarchicalStateMachine::~AHierarchicalStateMachine() {
}
-void AHierarchicalStateMachine::onMessageReceived(const sp<AMessage> &msg) {
+void AHierarchicalStateMachine::handleMessage(const sp<AMessage> &msg) {
sp<AState> save = mState;
sp<AState> cur = mState;
diff --git a/media/libstagefright/foundation/AString.cpp b/media/libstagefright/foundation/AString.cpp
index f2d501e..894f65c 100644
--- a/media/libstagefright/foundation/AString.cpp
+++ b/media/libstagefright/foundation/AString.cpp
@@ -328,6 +328,20 @@
return !strcmp(mData + mSize - suffixLen, suffix);
}
+bool AString::startsWithIgnoreCase(const char *prefix) const {
+ return !strncasecmp(mData, prefix, strlen(prefix));
+}
+
+bool AString::endsWithIgnoreCase(const char *suffix) const {
+ size_t suffixLen = strlen(suffix);
+
+ if (mSize < suffixLen) {
+ return false;
+ }
+
+ return !strcasecmp(mData + mSize - suffixLen, suffix);
+}
+
AString StringPrintf(const char *format, ...) {
va_list ap;
va_start(ap, format);
diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp
index 326d85b..10437c9 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.cpp
+++ b/media/libstagefright/httplive/PlaylistFetcher.cpp
@@ -588,7 +588,10 @@
void PlaylistFetcher::queueDiscontinuity(
ATSParser::DiscontinuityType type, const sp<AMessage> &extra) {
for (size_t i = 0; i < mPacketSources.size(); ++i) {
- mPacketSources.valueAt(i)->queueDiscontinuity(type, extra);
+ // do not discard buffer upon #EXT-X-DISCONTINUITY tag
+ // (seek will discard buffer by abandoning old fetchers)
+ mPacketSources.valueAt(i)->queueDiscontinuity(
+ type, extra, false /* discard */);
}
}
@@ -723,8 +726,7 @@
firstSeqNumberInPlaylist = 0;
}
- bool seekDiscontinuity = false;
- bool explicitDiscontinuity = false;
+ bool discontinuity = false;
const int32_t lastSeqNumberInPlaylist =
firstSeqNumberInPlaylist + (int32_t)mPlaylist->size() - 1;
@@ -792,7 +794,7 @@
if (mSeqNumber < firstSeqNumberInPlaylist) {
mSeqNumber = firstSeqNumberInPlaylist;
}
- explicitDiscontinuity = true;
+ discontinuity = true;
// fall through
} else {
@@ -817,7 +819,7 @@
int32_t val;
if (itemMeta->findInt32("discontinuity", &val) && val != 0) {
- explicitDiscontinuity = true;
+ discontinuity = true;
}
int64_t range_offset, range_length;
@@ -877,7 +879,7 @@
return;
}
- if (mStartup || seekDiscontinuity || explicitDiscontinuity) {
+ if (mStartup || discontinuity) {
// Signal discontinuity.
if (mPlaylist->isComplete() || mPlaylist->isEvent()) {
@@ -887,15 +889,14 @@
mNextPTSTimeUs = getSegmentStartTimeUs(mSeqNumber);
}
- if (seekDiscontinuity || explicitDiscontinuity) {
- ALOGI("queueing discontinuity (seek=%d, explicit=%d)",
- seekDiscontinuity, explicitDiscontinuity);
+ if (discontinuity) {
+ ALOGI("queueing discontinuity (explicit=%d)", discontinuity);
queueDiscontinuity(
- explicitDiscontinuity
- ? ATSParser::DISCONTINUITY_FORMATCHANGE
- : ATSParser::DISCONTINUITY_SEEK,
+ ATSParser::DISCONTINUITY_FORMATCHANGE,
NULL /* extra */);
+
+ discontinuity = false;
}
}
diff --git a/media/libstagefright/include/MPEG4Extractor.h b/media/libstagefright/include/MPEG4Extractor.h
index 7b4bc6d..1fe6fcf 100644
--- a/media/libstagefright/include/MPEG4Extractor.h
+++ b/media/libstagefright/include/MPEG4Extractor.h
@@ -39,6 +39,14 @@
uint32_t mDurationUs;
};
+struct Trex {
+ uint32_t track_ID;
+ uint32_t default_sample_description_index;
+ uint32_t default_sample_duration;
+ uint32_t default_sample_size;
+ uint32_t default_sample_flags;
+};
+
class MPEG4Extractor : public MediaExtractor {
public:
// Extractor assumes ownership of "source".
@@ -74,11 +82,12 @@
};
Vector<SidxEntry> mSidxEntries;
- uint64_t mSidxDuration;
off64_t mMoofOffset;
Vector<PsshInfo> mPssh;
+ Vector<Trex> mTrex;
+
sp<DataSource> mDataSource;
status_t mInitCheck;
bool mHasVideo;
diff --git a/media/libstagefright/matroska/MatroskaExtractor.cpp b/media/libstagefright/matroska/MatroskaExtractor.cpp
index d7bec59..2587ec7 100644
--- a/media/libstagefright/matroska/MatroskaExtractor.cpp
+++ b/media/libstagefright/matroska/MatroskaExtractor.cpp
@@ -20,8 +20,6 @@
#include "MatroskaExtractor.h"
-#include "mkvparser.hpp"
-
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/hexdump.h>
#include <media/stagefright/DataSource.h>
@@ -89,7 +87,7 @@
////////////////////////////////////////////////////////////////////////////////
struct BlockIterator {
- BlockIterator(MatroskaExtractor *extractor, unsigned long trackNum);
+ BlockIterator(MatroskaExtractor *extractor, unsigned long trackNum, unsigned long index);
bool eos() const;
@@ -106,6 +104,7 @@
private:
MatroskaExtractor *mExtractor;
long long mTrackNum;
+ unsigned long mIndex;
const mkvparser::Cluster *mCluster;
const mkvparser::BlockEntry *mBlockEntry;
@@ -157,6 +156,53 @@
MatroskaSource &operator=(const MatroskaSource &);
};
+const mkvparser::Track* MatroskaExtractor::TrackInfo::getTrack() const {
+ return mExtractor->mSegment->GetTracks()->GetTrackByNumber(mTrackNum);
+}
+
+// This function does exactly the same as mkvparser::Cues::Find, except that it
+// searches in our own track based vectors. We should not need this once mkvparser
+// adds the same functionality.
+const mkvparser::CuePoint::TrackPosition *MatroskaExtractor::TrackInfo::find(
+ long long timeNs) const {
+ ALOGV("mCuePoints.size %zu", mCuePoints.size());
+ if (mCuePoints.empty()) {
+ return NULL;
+ }
+
+ const mkvparser::CuePoint* cp = mCuePoints.itemAt(0);
+ const mkvparser::Track* track = getTrack();
+ if (timeNs <= cp->GetTime(mExtractor->mSegment)) {
+ return cp->Find(track);
+ }
+
+ // Binary searches through relevant cues; assumes cues are ordered by timecode.
+ // If we do detect out-of-order cues, return NULL.
+ size_t lo = 0;
+ size_t hi = mCuePoints.size();
+ while (lo < hi) {
+ const size_t mid = lo + (hi - lo) / 2;
+ const mkvparser::CuePoint* const midCp = mCuePoints.itemAt(mid);
+ const long long cueTimeNs = midCp->GetTime(mExtractor->mSegment);
+ if (cueTimeNs <= timeNs) {
+ lo = mid + 1;
+ } else {
+ hi = mid;
+ }
+ }
+
+ if (lo == 0) {
+ return NULL;
+ }
+
+ cp = mCuePoints.itemAt(lo - 1);
+ if (cp->GetTime(mExtractor->mSegment) > timeNs) {
+ return NULL;
+ }
+
+ return cp->Find(track);
+}
+
MatroskaSource::MatroskaSource(
const sp<MatroskaExtractor> &extractor, size_t index)
: mExtractor(extractor),
@@ -164,7 +210,8 @@
mType(OTHER),
mIsAudio(false),
mBlockIter(mExtractor.get(),
- mExtractor->mTracks.itemAt(index).mTrackNum),
+ mExtractor->mTracks.itemAt(index).mTrackNum,
+ index),
mNALSizeLen(0) {
sp<MetaData> meta = mExtractor->mTracks.itemAt(index).mMeta;
@@ -214,9 +261,10 @@
////////////////////////////////////////////////////////////////////////////////
BlockIterator::BlockIterator(
- MatroskaExtractor *extractor, unsigned long trackNum)
+ MatroskaExtractor *extractor, unsigned long trackNum, unsigned long index)
: mExtractor(extractor),
mTrackNum(trackNum),
+ mIndex(index),
mCluster(NULL),
mBlockEntry(NULL),
mBlockEntryIndex(0) {
@@ -364,9 +412,20 @@
}
const mkvparser::CuePoint* pCP;
+ mkvparser::Tracks const *pTracks = pSegment->GetTracks();
+ unsigned long int trackCount = pTracks->GetTracksCount();
while (!pCues->DoneParsing()) {
pCues->LoadCuePoint();
pCP = pCues->GetLast();
+ CHECK(pCP);
+
+ for (size_t index = 0; index < trackCount; ++index) {
+ const mkvparser::Track *pTrack = pTracks->GetTrackByIndex(index);
+ if (pTrack && pTrack->GetType() == 1 && pCP->Find(pTrack)) { // VIDEO_TRACK
+ MatroskaExtractor::TrackInfo& track = mExtractor->mTracks.editItemAt(index);
+ track.mCuePoints.push_back(pCP);
+ }
+ }
if (pCP->GetTime(pSegment) >= seekTimeNs) {
ALOGV("Parsed past relevant Cue");
@@ -374,22 +433,25 @@
}
}
- // The Cue index is built around video keyframes
- mkvparser::Tracks const *pTracks = pSegment->GetTracks();
- const mkvparser::Track *pTrack = NULL;
- for (size_t index = 0; index < pTracks->GetTracksCount(); ++index) {
- pTrack = pTracks->GetTrackByIndex(index);
- if (pTrack && pTrack->GetType() == 1) { // VIDEO_TRACK
- ALOGV("Video track located at %zu", index);
- break;
+ const mkvparser::CuePoint::TrackPosition *pTP = NULL;
+ const mkvparser::Track *thisTrack = pTracks->GetTrackByIndex(mIndex);
+ if (thisTrack->GetType() == 1) { // video
+ MatroskaExtractor::TrackInfo& track = mExtractor->mTracks.editItemAt(mIndex);
+ pTP = track.find(seekTimeNs);
+ } else {
+ // The Cue index is built around video keyframes
+ for (size_t index = 0; index < trackCount; ++index) {
+ const mkvparser::Track *pTrack = pTracks->GetTrackByIndex(index);
+ if (pTrack && pTrack->GetType() == 1 && pCues->Find(seekTimeNs, pTrack, pCP, pTP)) {
+ ALOGV("Video track located at %zu", index);
+ break;
+ }
}
}
+
// Always *search* based on the video track, but finalize based on mTrackNum
- const mkvparser::CuePoint::TrackPosition* pTP;
- if (pTrack && pTrack->GetType() == 1) {
- pCues->Find(seekTimeNs, pTrack, pCP, pTP);
- } else {
+ if (!pTP) {
ALOGE("Did not locate the video track for seeking");
return;
}
@@ -410,10 +472,13 @@
if (isAudio || block()->IsKey()) {
// Accept the first key frame
- *actualFrameTimeUs = (block()->GetTime(mCluster) + 500LL) / 1000LL;
- ALOGV("Requested seek point: %" PRId64 " actual: %" PRId64,
- seekTimeUs, *actualFrameTimeUs);
- break;
+ int64_t frameTimeUs = (block()->GetTime(mCluster) + 500LL) / 1000LL;
+ if (thisTrack->GetType() == 1 || frameTimeUs >= seekTimeUs) {
+ *actualFrameTimeUs = frameTimeUs;
+ ALOGV("Requested seek point: %" PRId64 " actual: %" PRId64,
+ seekTimeUs, *actualFrameTimeUs);
+ break;
+ }
}
}
}
@@ -964,6 +1029,7 @@
TrackInfo *trackInfo = &mTracks.editItemAt(mTracks.size() - 1);
trackInfo->mTrackNum = track->GetNumber();
trackInfo->mMeta = meta;
+ trackInfo->mExtractor = this;
}
}
@@ -978,7 +1044,7 @@
continue;
}
- BlockIterator iter(this, info->mTrackNum);
+ BlockIterator iter(this, info->mTrackNum, i);
int32_t j = 0;
int64_t thumbnailTimeUs = 0;
size_t maxBlockSize = 0;
diff --git a/media/libstagefright/matroska/MatroskaExtractor.h b/media/libstagefright/matroska/MatroskaExtractor.h
index cf200f3..db36bf8 100644
--- a/media/libstagefright/matroska/MatroskaExtractor.h
+++ b/media/libstagefright/matroska/MatroskaExtractor.h
@@ -18,14 +18,12 @@
#define MATROSKA_EXTRACTOR_H_
+#include "mkvparser.hpp"
+
#include <media/stagefright/MediaExtractor.h>
#include <utils/Vector.h>
#include <utils/threads.h>
-namespace mkvparser {
-struct Segment;
-};
-
namespace android {
struct AMessage;
@@ -58,6 +56,11 @@
struct TrackInfo {
unsigned long mTrackNum;
sp<MetaData> mMeta;
+ const MatroskaExtractor *mExtractor;
+ Vector<const mkvparser::CuePoint*> mCuePoints;
+
+ const mkvparser::Track* getTrack() const;
+ const mkvparser::CuePoint::TrackPosition *find(long long timeNs) const;
};
Mutex mLock;
diff --git a/media/libstagefright/mpeg2ts/ATSParser.cpp b/media/libstagefright/mpeg2ts/ATSParser.cpp
index 338e899..3d241e0 100644
--- a/media/libstagefright/mpeg2ts/ATSParser.cpp
+++ b/media/libstagefright/mpeg2ts/ATSParser.cpp
@@ -669,7 +669,7 @@
}
if (mSource != NULL) {
- mSource->queueDiscontinuity(type, extra);
+ mSource->queueDiscontinuity(type, extra, true);
}
}
diff --git a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
index 021b640..871824a 100644
--- a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
+++ b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
@@ -217,22 +217,25 @@
void AnotherPacketSource::queueDiscontinuity(
ATSParser::DiscontinuityType type,
- const sp<AMessage> &extra) {
+ const sp<AMessage> &extra,
+ bool discard) {
Mutex::Autolock autoLock(mLock);
- // Leave only discontinuities in the queue.
- List<sp<ABuffer> >::iterator it = mBuffers.begin();
- while (it != mBuffers.end()) {
- sp<ABuffer> oldBuffer = *it;
+ if (discard) {
+ // Leave only discontinuities in the queue.
+ List<sp<ABuffer> >::iterator it = mBuffers.begin();
+ while (it != mBuffers.end()) {
+ sp<ABuffer> oldBuffer = *it;
- int32_t oldDiscontinuityType;
- if (!oldBuffer->meta()->findInt32(
- "discontinuity", &oldDiscontinuityType)) {
- it = mBuffers.erase(it);
- continue;
+ int32_t oldDiscontinuityType;
+ if (!oldBuffer->meta()->findInt32(
+ "discontinuity", &oldDiscontinuityType)) {
+ it = mBuffers.erase(it);
+ continue;
+ }
+
+ ++it;
}
-
- ++it;
}
mEOSResult = OK;
diff --git a/media/libstagefright/mpeg2ts/AnotherPacketSource.h b/media/libstagefright/mpeg2ts/AnotherPacketSource.h
index 9b193a2..06c49bd 100644
--- a/media/libstagefright/mpeg2ts/AnotherPacketSource.h
+++ b/media/libstagefright/mpeg2ts/AnotherPacketSource.h
@@ -54,7 +54,9 @@
void queueAccessUnit(const sp<ABuffer> &buffer);
void queueDiscontinuity(
- ATSParser::DiscontinuityType type, const sp<AMessage> &extra);
+ ATSParser::DiscontinuityType type,
+ const sp<AMessage> &extra,
+ bool discard);
void signalEOS(status_t result);
diff --git a/media/libstagefright/omx/GraphicBufferSource.cpp b/media/libstagefright/omx/GraphicBufferSource.cpp
index 16f6c58..67e6d7b 100644
--- a/media/libstagefright/omx/GraphicBufferSource.cpp
+++ b/media/libstagefright/omx/GraphicBufferSource.cpp
@@ -114,7 +114,7 @@
void GraphicBufferSource::omxExecuting() {
Mutex::Autolock autoLock(mMutex);
- ALOGV("--> executing; avail=%d, codec vec size=%zd",
+ ALOGV("--> executing; avail=%zu, codec vec size=%zd",
mNumFramesAvailable, mCodecBuffers.size());
CHECK(!mExecuting);
mExecuting = true;
@@ -136,7 +136,7 @@
}
}
- ALOGV("done loading initial frames, avail=%d", mNumFramesAvailable);
+ ALOGV("done loading initial frames, avail=%zu", mNumFramesAvailable);
// If EOS has already been signaled, and there are no more frames to
// submit, try to send EOS now as well.
@@ -188,7 +188,7 @@
mLooper.clear();
}
- ALOGV("--> loaded; avail=%d eos=%d eosSent=%d",
+ ALOGV("--> loaded; avail=%zu eos=%d eosSent=%d",
mNumFramesAvailable, mEndOfStream, mEndOfStreamSent);
// Codec is no longer executing. Discard all codec-related state.
@@ -291,7 +291,7 @@
if (mNumFramesAvailable) {
// Fill this codec buffer.
CHECK(!mEndOfStreamSent);
- ALOGV("buffer freed, %d frames avail (eos=%d)",
+ ALOGV("buffer freed, %zu frames avail (eos=%d)",
mNumFramesAvailable, mEndOfStream);
fillCodecBuffer_l();
} else if (mEndOfStream) {
@@ -320,7 +320,8 @@
ssize_t index = mOriginalTimeUs.indexOfKey(header->nTimeStamp);
if (index >= 0) {
ALOGV("OUT timestamp: %lld -> %lld",
- header->nTimeStamp, mOriginalTimeUs[index]);
+ static_cast<long long>(header->nTimeStamp),
+ static_cast<long long>(mOriginalTimeUs[index]));
header->nTimeStamp = mOriginalTimeUs[index];
mOriginalTimeUs.removeItemsAt(index);
} else {
@@ -331,7 +332,7 @@
}
if (mOriginalTimeUs.size() > BufferQueue::NUM_BUFFER_SLOTS) {
// something terribly wrong must have happened, giving up...
- ALOGE("mOriginalTimeUs has too many entries (%d)",
+ ALOGE("mOriginalTimeUs has too many entries (%zu)",
mOriginalTimeUs.size());
mMaxTimestampGapUs = -1ll;
}
@@ -388,12 +389,12 @@
int cbi = findAvailableCodecBuffer_l();
if (cbi < 0) {
// No buffers available, bail.
- ALOGV("fillCodecBuffer_l: no codec buffers, avail now %d",
+ ALOGV("fillCodecBuffer_l: no codec buffers, avail now %zu",
mNumFramesAvailable);
return false;
}
- ALOGV("fillCodecBuffer_l: acquiring buffer, avail=%d",
+ ALOGV("fillCodecBuffer_l: acquiring buffer, avail=%zu",
mNumFramesAvailable);
BufferQueue::BufferItem item;
status_t err = mConsumer->acquireBuffer(&item, 0);
@@ -540,7 +541,7 @@
status_t GraphicBufferSource::signalEndOfInputStream() {
Mutex::Autolock autoLock(mMutex);
- ALOGV("signalEndOfInputStream: exec=%d avail=%d eos=%d",
+ ALOGV("signalEndOfInputStream: exec=%d avail=%zu eos=%d",
mExecuting, mNumFramesAvailable, mEndOfStream);
if (mEndOfStream) {
@@ -580,7 +581,7 @@
/ mTimePerCaptureUs;
if (nFrames <= 0) {
// skip this frame as it's too close to previous capture
- ALOGV("skipping frame, timeUs %lld", timeUs);
+ ALOGV("skipping frame, timeUs %lld", static_cast<long long>(timeUs));
return -1;
}
mPrevCaptureUs = mPrevCaptureUs + nFrames * mTimePerCaptureUs;
@@ -588,7 +589,9 @@
}
ALOGV("timeUs %lld, captureUs %lld, frameUs %lld",
- timeUs, mPrevCaptureUs, mPrevFrameUs);
+ static_cast<long long>(timeUs),
+ static_cast<long long>(mPrevCaptureUs),
+ static_cast<long long>(mPrevFrameUs));
return mPrevFrameUs;
} else if (mMaxTimestampGapUs > 0ll) {
@@ -615,7 +618,9 @@
mPrevOriginalTimeUs = originalTimeUs;
mPrevModifiedTimeUs = timeUs;
mOriginalTimeUs.add(timeUs, originalTimeUs);
- ALOGV("IN timestamp: %lld -> %lld", originalTimeUs, timeUs);
+ ALOGV("IN timestamp: %lld -> %lld",
+ static_cast<long long>(originalTimeUs),
+ static_cast<long long>(timeUs));
}
return timeUs;
@@ -723,7 +728,7 @@
void GraphicBufferSource::onFrameAvailable() {
Mutex::Autolock autoLock(mMutex);
- ALOGV("onFrameAvailable exec=%d avail=%d",
+ ALOGV("onFrameAvailable exec=%d avail=%zu",
mExecuting, mNumFramesAvailable);
if (mEndOfStream || mSuspended) {
diff --git a/media/libstagefright/omx/OMXMaster.cpp b/media/libstagefright/omx/OMXMaster.cpp
index 6b6d0ab..ae3cb33 100644
--- a/media/libstagefright/omx/OMXMaster.cpp
+++ b/media/libstagefright/omx/OMXMaster.cpp
@@ -91,7 +91,7 @@
}
if (err != OMX_ErrorNoMore) {
- ALOGE("OMX plugin failed w/ error 0x%08x after registering %d "
+ ALOGE("OMX plugin failed w/ error 0x%08x after registering %zu "
"components", err, mPluginByComponentName.size());
}
}
diff --git a/media/mtp/MtpDataPacket.cpp b/media/mtp/MtpDataPacket.cpp
index c4f87a0..e6e19e3 100644
--- a/media/mtp/MtpDataPacket.cpp
+++ b/media/mtp/MtpDataPacket.cpp
@@ -363,7 +363,7 @@
}
int MtpDataPacket::writeData(int fd, void* data, uint32_t length) {
- allocate(length);
+ allocate(length + MTP_CONTAINER_HEADER_SIZE);
memcpy(mBuffer + MTP_CONTAINER_HEADER_SIZE, data, length);
length += MTP_CONTAINER_HEADER_SIZE;
MtpPacket::putUInt32(MTP_CONTAINER_LENGTH_OFFSET, length);
diff --git a/media/mtp/MtpServer.cpp b/media/mtp/MtpServer.cpp
index 157f2ce..aa43967 100644
--- a/media/mtp/MtpServer.cpp
+++ b/media/mtp/MtpServer.cpp
@@ -325,6 +325,14 @@
mSendObjectHandle = kInvalidObjectHandle;
}
+ int containertype = mRequest.getContainerType();
+ if (containertype != MTP_CONTAINER_TYPE_COMMAND) {
+ ALOGE("wrong container type %d", containertype);
+ return false;
+ }
+
+ ALOGV("got command %s (%x)", MtpDebug::getOperationCodeName(operation), operation);
+
switch (operation) {
case MTP_OPERATION_GET_DEVICE_INFO:
response = doGetDeviceInfo();
@@ -415,7 +423,8 @@
response = doEndEditObject();
break;
default:
- ALOGE("got unsupported command %s", MtpDebug::getOperationCodeName(operation));
+ ALOGE("got unsupported command %s (%x)",
+ MtpDebug::getOperationCodeName(operation), operation);
response = MTP_RESPONSE_OPERATION_NOT_SUPPORTED;
break;
}
@@ -793,7 +802,7 @@
int result = mDatabase->getObjectFilePath(handle, pathBuf, fileLength, format);
if (result != MTP_RESPONSE_OK)
return result;
- if (offset + length > fileLength)
+ if (offset + length > (uint64_t)fileLength)
length = fileLength - offset;
const char* filePath = (const char *)pathBuf;
@@ -950,22 +959,28 @@
fchmod(mfr.fd, mFilePermission);
umask(mask);
- if (initialData > 0)
+ if (initialData > 0) {
ret = write(mfr.fd, mData.getData(), initialData);
+ }
- if (mSendObjectFileSize - initialData > 0) {
- mfr.offset = initialData;
- if (mSendObjectFileSize == 0xFFFFFFFF) {
- // tell driver to read until it receives a short packet
- mfr.length = 0xFFFFFFFF;
- } else {
- mfr.length = mSendObjectFileSize - initialData;
+ if (ret < 0) {
+ ALOGE("failed to write initial data");
+ result = MTP_RESPONSE_GENERAL_ERROR;
+ } else {
+ if (mSendObjectFileSize - initialData > 0) {
+ mfr.offset = initialData;
+ if (mSendObjectFileSize == 0xFFFFFFFF) {
+ // tell driver to read until it receives a short packet
+ mfr.length = 0xFFFFFFFF;
+ } else {
+ mfr.length = mSendObjectFileSize - initialData;
+ }
+
+ ALOGV("receiving %s\n", (const char *)mSendObjectFilePath);
+ // transfer the file
+ ret = ioctl(mFD, MTP_RECEIVE_FILE, (unsigned long)&mfr);
+ ALOGV("MTP_RECEIVE_FILE returned %d\n", ret);
}
-
- ALOGV("receiving %s\n", (const char *)mSendObjectFilePath);
- // transfer the file
- ret = ioctl(mFD, MTP_RECEIVE_FILE, (unsigned long)&mfr);
- ALOGV("MTP_RECEIVE_FILE returned %d\n", ret);
}
close(mfr.fd);
@@ -990,7 +1005,7 @@
static void deleteRecursive(const char* path) {
char pathbuf[PATH_MAX];
- int pathLength = strlen(path);
+ size_t pathLength = strlen(path);
if (pathLength >= sizeof(pathbuf) - 1) {
ALOGE("path too long: %s\n", path);
}
@@ -1112,12 +1127,13 @@
// can't start writing past the end of the file
if (offset > edit->mSize) {
- ALOGD("writing past end of object, offset: %lld, edit->mSize: %lld", offset, edit->mSize);
+ ALOGD("writing past end of object, offset: %" PRIu64 ", edit->mSize: %" PRIu64,
+ offset, edit->mSize);
return MTP_RESPONSE_GENERAL_ERROR;
}
const char* filePath = (const char *)edit->mPath;
- ALOGV("receiving partial %s %lld %" PRIu32 "\n", filePath, offset, length);
+ ALOGV("receiving partial %s %" PRIu64 " %" PRIu32, filePath, offset, length);
// read the header, and possibly some data
int ret = mData.read(mFD);
@@ -1131,15 +1147,19 @@
length -= initialData;
}
- if (length > 0) {
- mtp_file_range mfr;
- mfr.fd = edit->mFD;
- mfr.offset = offset;
- mfr.length = length;
+ if (ret < 0) {
+ ALOGE("failed to write initial data");
+ } else {
+ if (length > 0) {
+ mtp_file_range mfr;
+ mfr.fd = edit->mFD;
+ mfr.offset = offset;
+ mfr.length = length;
- // transfer the file
- ret = ioctl(mFD, MTP_RECEIVE_FILE, (unsigned long)&mfr);
- ALOGV("MTP_RECEIVE_FILE returned %d", ret);
+ // transfer the file
+ ret = ioctl(mFD, MTP_RECEIVE_FILE, (unsigned long)&mfr);
+ ALOGV("MTP_RECEIVE_FILE returned %d", ret);
+ }
}
if (ret < 0) {
mResponse.setParameter(1, 0);
diff --git a/media/mtp/MtpStorageInfo.cpp b/media/mtp/MtpStorageInfo.cpp
index dcd37cd..2b1a9ae 100644
--- a/media/mtp/MtpStorageInfo.cpp
+++ b/media/mtp/MtpStorageInfo.cpp
@@ -16,6 +16,8 @@
#define LOG_TAG "MtpStorageInfo"
+#include <inttypes.h>
+
#include "MtpDebug.h"
#include "MtpDataPacket.h"
#include "MtpStorageInfo.h"
@@ -63,7 +65,7 @@
void MtpStorageInfo::print() {
ALOGD("Storage Info %08X:\n\tmStorageType: %d\n\tmFileSystemType: %d\n\tmAccessCapability: %d\n",
mStorageID, mStorageType, mFileSystemType, mAccessCapability);
- ALOGD("\tmMaxCapacity: %lld\n\tmFreeSpaceBytes: %lld\n\tmFreeSpaceObjects: %d\n",
+ ALOGD("\tmMaxCapacity: %" PRIu64 "\n\tmFreeSpaceBytes: %" PRIu64 "\n\tmFreeSpaceObjects: %d\n",
mMaxCapacity, mFreeSpaceBytes, mFreeSpaceObjects);
ALOGD("\tmStorageDescription: %s\n\tmVolumeIdentifier: %s\n",
mStorageDescription, mVolumeIdentifier);
diff --git a/media/ndk/NdkMediaCodec.cpp b/media/ndk/NdkMediaCodec.cpp
index c5d8858..ed00b72 100644
--- a/media/ndk/NdkMediaCodec.cpp
+++ b/media/ndk/NdkMediaCodec.cpp
@@ -14,6 +14,8 @@
* limitations under the License.
*/
+#include <inttypes.h>
+
//#define LOG_NDEBUG 0
#define LOG_TAG "NdkMediaCodec"
@@ -257,7 +259,7 @@
if (mData->mCodec->getInputBuffers(&abufs) == 0) {
size_t n = abufs.size();
if (idx >= n) {
- ALOGE("buffer index %d out of range", idx);
+ ALOGE("buffer index %zu out of range", idx);
return NULL;
}
if (out_size != NULL) {
@@ -275,7 +277,7 @@
if (mData->mCodec->getOutputBuffers(&abufs) == 0) {
size_t n = abufs.size();
if (idx >= n) {
- ALOGE("buffer index %d out of range", idx);
+ ALOGE("buffer index %zu out of range", idx);
return NULL;
}
if (out_size != NULL) {
@@ -345,7 +347,7 @@
EXPORT
media_status_t AMediaCodec_releaseOutputBufferAtTime(
AMediaCodec *mData, size_t idx, int64_t timestampNs) {
- ALOGV("render @ %lld", timestampNs);
+ ALOGV("render @ %" PRId64, timestampNs);
return translate_error(mData->mCodec->renderOutputBufferAndRelease(idx, timestampNs));
}
@@ -413,7 +415,7 @@
size_t cryptosize = sizeof(AMediaCodecCryptoInfo) + sizeof(size_t) * numsubsamples * 2;
AMediaCodecCryptoInfo *ret = (AMediaCodecCryptoInfo*) malloc(cryptosize);
if (!ret) {
- ALOGE("couldn't allocate %d bytes", cryptosize);
+ ALOGE("couldn't allocate %zu bytes", cryptosize);
return NULL;
}
ret->numsubsamples = numsubsamples;
diff --git a/media/ndk/NdkMediaExtractor.cpp b/media/ndk/NdkMediaExtractor.cpp
index 492e002..970a43c 100644
--- a/media/ndk/NdkMediaExtractor.cpp
+++ b/media/ndk/NdkMediaExtractor.cpp
@@ -133,13 +133,13 @@
EXPORT
media_status_t AMediaExtractor_selectTrack(AMediaExtractor *mData, size_t idx) {
- ALOGV("selectTrack(%z)", idx);
+ ALOGV("selectTrack(%zu)", idx);
return translate_error(mData->mImpl->selectTrack(idx));
}
EXPORT
media_status_t AMediaExtractor_unselectTrack(AMediaExtractor *mData, size_t idx) {
- ALOGV("unselectTrack(%z)", idx);
+ ALOGV("unselectTrack(%zu)", idx);
return translate_error(mData->mImpl->unselectTrack(idx));
}
diff --git a/media/ndk/NdkMediaFormat.cpp b/media/ndk/NdkMediaFormat.cpp
index 67dc2c2..a354d58 100644
--- a/media/ndk/NdkMediaFormat.cpp
+++ b/media/ndk/NdkMediaFormat.cpp
@@ -17,6 +17,7 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "NdkMediaFormat"
+#include <inttypes.h>
#include "NdkMediaFormat.h"
@@ -89,21 +90,21 @@
{
int32_t val;
f->findInt32(name, &val);
- ret.appendFormat("int32(%d)", val);
+ ret.appendFormat("int32(%" PRId32 ")", val);
break;
}
case AMessage::kTypeInt64:
{
int64_t val;
f->findInt64(name, &val);
- ret.appendFormat("int64(%lld)", val);
+ ret.appendFormat("int64(%" PRId64 ")", val);
break;
}
case AMessage::kTypeSize:
{
size_t val;
f->findSize(name, &val);
- ret.appendFormat("size_t(%d)", val);
+ ret.appendFormat("size_t(%zu)", val);
break;
}
case AMessage::kTypeFloat:
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 527fd65..2124f85 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -668,17 +668,6 @@
return thread->sampleRate();
}
-int AudioFlinger::channelCount(audio_io_handle_t output) const
-{
- Mutex::Autolock _l(mLock);
- PlaybackThread *thread = checkPlaybackThread_l(output);
- if (thread == NULL) {
- ALOGW("channelCount() unknown thread %d", output);
- return 0;
- }
- return thread->channelCount();
-}
-
audio_format_t AudioFlinger::format(audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
@@ -1344,6 +1333,7 @@
IAudioFlinger::track_flags_t *flags,
pid_t tid,
int *sessionId,
+ size_t *notificationFrames,
sp<IMemory>& cblk,
sp<IMemory>& buffers,
status_t *status)
@@ -1418,7 +1408,7 @@
// TODO: the uid should be passed in as a parameter to openRecord
recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
- frameCount, lSessionId,
+ frameCount, lSessionId, notificationFrames,
IPCThreadState::self()->getCallingUid(),
flags, tid, &lStatus);
LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0));
@@ -1621,6 +1611,19 @@
mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
audio_stream_out_t *outStream = NULL;
+
+ // FOR TESTING ONLY:
+ // Enable increased sink precision for mixing mode if kEnableExtendedPrecision is true.
+ if (kEnableExtendedPrecision && // Check only for Normal Mixing mode
+ !(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) {
+ // Update format
+ //config.format = AUDIO_FORMAT_PCM_FLOAT;
+ //config.format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
+ //config.format = AUDIO_FORMAT_PCM_32_BIT;
+ //config.format = AUDIO_FORMAT_PCM_8_24_BIT;
+ // ALOGV("openOutput() upgrading format to %#08x", config.format);
+ }
+
status_t status = hwDevHal->open_output_stream(hwDevHal,
id,
*pDevices,
@@ -1644,9 +1647,9 @@
if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
thread = new OffloadThread(this, output, id, *pDevices);
ALOGV("openOutput() created offload output: ID %d thread %p", id, thread);
- } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) ||
- (config.format != AUDIO_FORMAT_PCM_16_BIT) ||
- (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) {
+ } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT)
+ || !isValidPcmSinkFormat(config.format)
+ || (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) {
thread = new DirectOutputThread(this, output, id, *pDevices);
ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
} else {
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 6e73a14..4df0921 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -120,12 +120,12 @@
IAudioFlinger::track_flags_t *flags,
pid_t tid,
int *sessionId,
+ size_t *notificationFrames,
sp<IMemory>& cblk,
sp<IMemory>& buffers,
status_t *status /*non-NULL*/);
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;
@@ -326,6 +326,24 @@
audio_devices_t devices);
void purgeStaleEffects_l();
+ // Set kEnableExtendedPrecision to true to use extended precision in MixerThread
+ static const bool kEnableExtendedPrecision = false;
+
+ // Returns true if format is permitted for the PCM sink in the MixerThread
+ static inline bool isValidPcmSinkFormat(audio_format_t format) {
+ switch (format) {
+ case AUDIO_FORMAT_PCM_16_BIT:
+ return true;
+ case AUDIO_FORMAT_PCM_FLOAT:
+ case AUDIO_FORMAT_PCM_24_BIT_PACKED:
+ case AUDIO_FORMAT_PCM_32_BIT:
+ case AUDIO_FORMAT_PCM_8_24_BIT:
+ return kEnableExtendedPrecision;
+ default:
+ return false;
+ }
+ }
+
// 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;
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index d73292e..4dbbc43 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -40,8 +40,36 @@
#include <media/EffectsFactoryApi.h>
+#include "AudioMixerOps.h"
#include "AudioMixer.h"
+// Use the FCC_2 macro for code assuming Fixed Channel Count of 2 and
+// whose stereo assumption may need to be revisited later.
+#ifndef FCC_2
+#define FCC_2 2
+#endif
+
+/* VERY_VERY_VERBOSE_LOGGING will show exactly which process hook and track hook is
+ * being used. This is a considerable amount of log spam, so don't enable unless you
+ * are verifying the hook based code.
+ */
+//#define VERY_VERY_VERBOSE_LOGGING
+#ifdef VERY_VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+//define ALOGVV printf // for test-mixer.cpp
+#else
+#define ALOGVV(a...) do { } while (0)
+#endif
+
+// Set kUseNewMixer to true to use the new mixer engine. Otherwise the
+// original code will be used. This is false for now.
+static const bool kUseNewMixer = false;
+
+// Set kUseFloat to true to allow floating input into the mixer engine.
+// If kUseNewMixer is false, this is ignored or may be overridden internally
+// because of downmix/upmix support.
+static const bool kUseFloat = true;
+
namespace android {
// ----------------------------------------------------------------------------
@@ -300,15 +328,19 @@
t->downmixerBufferProvider = NULL;
t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
t->mFormat = format;
- t->mMixerInFormat = AUDIO_FORMAT_PCM_16_BIT;
- if (t->mFormat != t->mMixerInFormat) {
- prepareTrackForReformat(t, n);
- }
- status_t status = initTrackDownmix(&mState.tracks[n], n, channelMask);
+ t->mMixerInFormat = kUseFloat && kUseNewMixer
+ ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
+ // Check the downmixing (or upmixing) requirements.
+ status_t status = initTrackDownmix(t, n, channelMask);
if (status != OK) {
ALOGE("AudioMixer::getTrackName invalid channelMask (%#x)", channelMask);
return -1;
}
+ // initTrackDownmix() may change the input format requirement.
+ // If you desire floating point input to the mixer, it may change
+ // to integer because the downmixer requires integer to process.
+ ALOGVV("mMixerFormat:%#x mMixerInFormat:%#x\n", t->mMixerFormat, t->mMixerInFormat);
+ prepareTrackForReformat(t, n);
mTrackNames |= 1 << n;
return TRACK0 + n;
}
@@ -443,6 +475,7 @@
}// end of scope for local variables that are not used in goto label "noDownmixForActiveTrack"
// initialization successful:
+ pTrack->mMixerInFormat = AUDIO_FORMAT_PCM_16_BIT; // 16 bit input is required for downmix
pTrack->downmixerBufferProvider = pDbp;
reconfigureBufferProviders(pTrack);
return NO_ERROR;
@@ -467,12 +500,15 @@
{
ALOGV("AudioMixer::prepareTrackForReformat(%d) with format %#x", trackName, pTrack->mFormat);
// discard the previous reformatter if there was one
- unprepareTrackForReformat(pTrack, trackName);
- pTrack->mReformatBufferProvider = new ReformatBufferProvider(
- audio_channel_count_from_out_mask(pTrack->channelMask),
- pTrack->mFormat, pTrack->mMixerInFormat);
- reconfigureBufferProviders(pTrack);
- return NO_ERROR;
+ unprepareTrackForReformat(pTrack, trackName);
+ // only configure reformatter if needed
+ if (pTrack->mFormat != pTrack->mMixerInFormat) {
+ pTrack->mReformatBufferProvider = new ReformatBufferProvider(
+ audio_channel_count_from_out_mask(pTrack->channelMask),
+ pTrack->mFormat, pTrack->mMixerInFormat);
+ reconfigureBufferProviders(pTrack);
+ }
+ return NO_ERROR;
}
void AudioMixer::reconfigureBufferProviders(track_t* pTrack)
@@ -596,8 +632,15 @@
track.channelMask = mask;
track.channelCount = channelCount;
// the mask has changed, does this track need a downmixer?
- initTrackDownmix(&mState.tracks[name], name, mask);
+ // update to try using our desired format (if we aren't already using it)
+ track.mMixerInFormat = kUseFloat && kUseNewMixer
+ ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
+ status_t status = initTrackDownmix(&mState.tracks[name], name, mask);
+ ALOGE_IF(status != OK,
+ "Invalid channel mask %#x, initTrackDownmix returned %d",
+ mask, status);
ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", mask);
+ prepareTrackForReformat(&track, name); // format may have changed
invalidateState(1 << name);
}
} break;
@@ -621,11 +664,7 @@
ALOG_ASSERT(audio_is_linear_pcm(format), "Invalid format %#x", format);
track.mFormat = format;
ALOGV("setParameter(TRACK, FORMAT, %#x)", format);
- //if (track.mFormat != track.mMixerInFormat)
- {
- ALOGD("Reformatting!");
- prepareTrackForReformat(&track, name);
- }
+ prepareTrackForReformat(&track, name);
invalidateState(1 << name);
}
} break;
@@ -723,9 +762,10 @@
} else {
quality = AudioResampler::DEFAULT_QUALITY;
}
- const int bits = mMixerInFormat == AUDIO_FORMAT_PCM_16_BIT ? 16 : /* FLOAT */ 32;
+
+ ALOGVV("Creating resampler with %d bits\n", bits);
resampler = AudioResampler::create(
- bits,
+ mMixerInFormat,
// the resampler sees the number of channels after the downmixer, if any
(int) (downmixerBufferProvider != NULL ? MAX_NUM_CHANNELS : channelCount),
devSampleRate, quality);
@@ -848,16 +888,19 @@
if (n & NEEDS_RESAMPLE) {
all16BitsStereoNoResample = false;
resampling = true;
- t.hook = track__genericResample;
+ t.hook = getTrackHook(TRACKTYPE_RESAMPLE, FCC_2,
+ t.mMixerInFormat, t.mMixerFormat);
ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
"Track %d needs downmix + resample", i);
} else {
if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){
- t.hook = track__16BitsMono;
+ t.hook = getTrackHook(TRACKTYPE_NORESAMPLEMONO, FCC_2,
+ t.mMixerInFormat, t.mMixerFormat);
all16BitsStereoNoResample = false;
}
if ((n & NEEDS_CHANNEL_COUNT__MASK) >= NEEDS_CHANNEL_2){
- t.hook = track__16BitsStereo;
+ t.hook = getTrackHook(TRACKTYPE_NORESAMPLE, FCC_2,
+ t.mMixerInFormat, t.mMixerFormat);
ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
"Track %d needs downmix", i);
}
@@ -888,7 +931,10 @@
state->hook = process__genericNoResampling;
if (all16BitsStereoNoResample && !volumeRamp) {
if (countActiveTracks == 1) {
- state->hook = process__OneTrack16BitsStereoNoResampling;
+ const int i = 31 - __builtin_clz(state->enabledTracks);
+ track_t& t = state->tracks[i];
+ state->hook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK, FCC_2,
+ t.mMixerInFormat, t.mMixerFormat);
}
}
}
@@ -931,6 +977,7 @@
void AudioMixer::track__genericResample(track_t* t, int32_t* out, size_t outFrameCount,
int32_t* temp, int32_t* aux)
{
+ ALOGVV("track__genericResample\n");
t->resampler->setSampleRate(t->sampleRate);
// ramp gain - resample to temp buffer and scale/mix in 2nd step
@@ -1042,6 +1089,7 @@
void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount,
int32_t* temp __unused, int32_t* aux)
{
+ ALOGVV("track__16BitsStereo\n");
const int16_t *in = static_cast<const int16_t *>(t->in);
if (CC_UNLIKELY(aux != NULL)) {
@@ -1133,6 +1181,7 @@
void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount,
int32_t* temp __unused, int32_t* aux)
{
+ ALOGVV("track__16BitsMono\n");
const int16_t *in = static_cast<int16_t const *>(t->in);
if (CC_UNLIKELY(aux != NULL)) {
@@ -1220,6 +1269,7 @@
// no-op case
void AudioMixer::process__nop(state_t* state, int64_t pts)
{
+ ALOGVV("process__nop\n");
uint32_t e0 = state->enabledTracks;
size_t sampleCount = state->frameCount * MAX_NUM_CHANNELS;
while (e0) {
@@ -1267,6 +1317,7 @@
// generic code without resampling
void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts)
{
+ ALOGVV("process__genericNoResampling\n");
int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32)));
// acquire each track's buffer
@@ -1349,18 +1400,12 @@
}
}
}
- switch (t1.mMixerFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- memcpy_to_float_from_q4_27(reinterpret_cast<float *>(out), outTemp, BLOCKSIZE * 2);
- out += BLOCKSIZE * 2; // output is 2 floats/frame.
- break;
- case AUDIO_FORMAT_PCM_16_BIT:
- ditherAndClamp(out, outTemp, BLOCKSIZE);
- out += BLOCKSIZE; // output is 1 int32_t (2 int16_t samples)/frame
- break;
- default:
- LOG_ALWAYS_FATAL("bad mixer format: %d", t1.mMixerFormat);
- }
+
+ convertMixerFormat(out, t1.mMixerFormat, outTemp, t1.mMixerInFormat,
+ BLOCKSIZE * FCC_2);
+ // TODO: fix ugly casting due to choice of out pointer type
+ out = reinterpret_cast<int32_t*>((uint8_t*)out
+ + BLOCKSIZE * FCC_2 * audio_bytes_per_sample(t1.mMixerFormat));
numFrames += BLOCKSIZE;
} while (numFrames < state->frameCount);
}
@@ -1379,6 +1424,7 @@
// generic code with resampling
void AudioMixer::process__genericResampling(state_t* state, int64_t pts)
{
+ ALOGVV("process__genericResampling\n");
// 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;
@@ -1442,16 +1488,7 @@
}
}
}
- switch (t1.mMixerFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- memcpy_to_float_from_q4_27(reinterpret_cast<float*>(out), outTemp, numFrames*2);
- break;
- case AUDIO_FORMAT_PCM_16_BIT:
- ditherAndClamp(out, outTemp, numFrames);
- break;
- default:
- LOG_ALWAYS_FATAL("bad mixer format: %d", t1.mMixerFormat);
- }
+ convertMixerFormat(out, t1.mMixerFormat, outTemp, t1.mMixerInFormat, numFrames * FCC_2);
}
}
@@ -1459,6 +1496,7 @@
void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state,
int64_t pts)
{
+ ALOGVV("process__OneTrack16BitsStereoNoResampling\n");
// 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.
@@ -1683,5 +1721,275 @@
ALOGW_IF(!sIsMultichannelCapable, "unable to find downmix effect");
}
+/* This process hook is called when there is a single track without
+ * aux buffer, volume ramp, or resampling.
+ * TODO: Update the hook selection: this can properly handle aux and ramp.
+ */
+template <int MIXTYPE, int NCHAN, typename TO, typename TI, typename TA>
+void AudioMixer::process_NoResampleOneTrack(state_t* state, int64_t pts)
+{
+ ALOGVV("process_NoResampleOneTrack\n");
+ // CLZ is faster than CTZ on ARM, though really not sure if true after 31 - clz.
+ const int i = 31 - __builtin_clz(state->enabledTracks);
+ ALOG_ASSERT((1 << i) == state->enabledTracks, "more than 1 track enabled");
+ track_t *t = &state->tracks[i];
+ TO* out = reinterpret_cast<TO*>(t->mainBuffer);
+ TA* aux = reinterpret_cast<TA*>(t->auxBuffer);
+ const bool ramp = t->needsRamp();
+
+ for (size_t numFrames = state->frameCount; numFrames; ) {
+ AudioBufferProvider::Buffer& b(t->buffer);
+ // get input buffer
+ b.frameCount = numFrames;
+ const int64_t outputPTS = calculateOutputPTS(*t, pts, state->frameCount - numFrames);
+ t->bufferProvider->getNextBuffer(&b, outputPTS);
+ const TI *in = reinterpret_cast<TI*>(b.raw);
+
+ // in == NULL can happen if the track was flushed just after having
+ // been enabled for mixing.
+ if (in == NULL || (((uintptr_t)in) & 3)) {
+ memset(out, 0, numFrames
+ * NCHAN * audio_bytes_per_sample(t->mMixerFormat));
+ ALOGE_IF((((uintptr_t)in) & 3), "process_NoResampleOneTrack: bus error: "
+ "buffer %p track %p, channels %d, needs %#x",
+ in, t, t->channelCount, t->needs);
+ return;
+ }
+
+ const size_t outFrames = b.frameCount;
+ if (ramp) {
+ volumeRampMulti<MIXTYPE_MULTI_SAVEONLY, NCHAN>(out, outFrames, in, aux,
+ t->prevVolume, t->volumeInc, &t->prevAuxLevel, t->auxInc);
+ } else {
+ volumeMulti<MIXTYPE_MULTI_SAVEONLY, NCHAN>(out, outFrames, in, aux,
+ t->volume, t->auxLevel);
+ }
+ out += outFrames * NCHAN;
+ if (aux != NULL) {
+ aux += NCHAN;
+ }
+ numFrames -= b.frameCount;
+
+ // release buffer
+ t->bufferProvider->releaseBuffer(&b);
+ }
+ if (ramp) {
+ t->adjustVolumeRamp(aux != NULL);
+ }
+}
+
+/* This track hook is called to do resampling then mixing,
+ * pulling from the track's upstream AudioBufferProvider.
+ */
+template <int MIXTYPE, int NCHAN, typename TO, typename TI, typename TA>
+void AudioMixer::track__Resample(track_t* t, TO* out, size_t outFrameCount, TO* temp, TA* aux)
+{
+ ALOGVV("track__Resample\n");
+ t->resampler->setSampleRate(t->sampleRate);
+
+ const bool ramp = t->needsRamp();
+ if (ramp || aux != NULL) {
+ // if ramp: resample with unity gain to temp buffer and scale/mix in 2nd step.
+ // if aux != NULL: resample with unity gain to temp buffer then apply send level.
+
+ t->resampler->setVolume(UNITY_GAIN_INT, UNITY_GAIN_INT);
+ memset(temp, 0, outFrameCount * NCHAN * sizeof(TO));
+ t->resampler->resample((int32_t*)temp, outFrameCount, t->bufferProvider);
+ if (ramp) {
+ volumeRampMulti<MIXTYPE_MULTI, NCHAN>(out, outFrameCount, temp, aux,
+ t->prevVolume, t->volumeInc, &t->prevAuxLevel, t->auxInc);
+ t->adjustVolumeRamp(aux != NULL);
+ } else {
+ volumeMulti<MIXTYPE_MULTI, NCHAN>(out, outFrameCount, temp, aux,
+ t->volume, t->auxLevel);
+ }
+ } else { // constant volume gain
+ t->resampler->setVolume(t->volume[0], t->volume[1]);
+ t->resampler->resample((int32_t*)out, outFrameCount, t->bufferProvider);
+ }
+}
+
+/* This track hook is called to mix a track, when no resampling is required.
+ * The input buffer should be present in t->in.
+ */
+template <int MIXTYPE, int NCHAN, typename TO, typename TI, typename TA>
+void AudioMixer::track__NoResample(track_t* t, TO* out, size_t frameCount,
+ TO* temp __unused, TA* aux)
+{
+ ALOGVV("track__NoResample\n");
+ const TI *in = static_cast<const TI *>(t->in);
+
+ if (t->needsRamp()) {
+ volumeRampMulti<MIXTYPE, NCHAN>(out, frameCount, in, aux,
+ t->prevVolume, t->volumeInc, &t->prevAuxLevel, t->auxInc);
+ t->adjustVolumeRamp(aux != NULL);
+ } else {
+ volumeMulti<MIXTYPE, NCHAN>(out, frameCount, in, aux, t->volume, t->auxLevel);
+ }
+ // MIXTYPE_MONOEXPAND reads a single input channel and expands to NCHAN output channels.
+ // MIXTYPE_MULTI reads NCHAN input channels and places to NCHAN output channels.
+ in += (MIXTYPE == MIXTYPE_MONOEXPAND) ? frameCount : frameCount * NCHAN;
+ t->in = in;
+}
+
+/* The Mixer engine generates either int32_t (Q4_27) or float data.
+ * We use this function to convert the engine buffers
+ * to the desired mixer output format, either int16_t (Q.15) or float.
+ */
+void AudioMixer::convertMixerFormat(void *out, audio_format_t mixerOutFormat,
+ void *in, audio_format_t mixerInFormat, size_t sampleCount)
+{
+ switch (mixerInFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ switch (mixerOutFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ memcpy(out, in, sampleCount * sizeof(float)); // MEMCPY. TODO optimize out
+ break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ memcpy_to_i16_from_float((int16_t*)out, (float*)in, sampleCount);
+ break;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+ break;
+ }
+ break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ switch (mixerOutFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ memcpy_to_float_from_q4_27((float*)out, (int32_t*)in, sampleCount);
+ break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ // two int16_t are produced per iteration
+ ditherAndClamp((int32_t*)out, (int32_t*)in, sampleCount >> 1);
+ break;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+ break;
+ }
+ break;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+ break;
+ }
+}
+
+/* Returns the proper track hook to use for mixing the track into the output buffer.
+ */
+AudioMixer::hook_t AudioMixer::getTrackHook(int trackType, int channels,
+ audio_format_t mixerInFormat, audio_format_t mixerOutFormat __unused)
+{
+ if (!kUseNewMixer && channels == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
+ switch (trackType) {
+ case TRACKTYPE_NOP:
+ return track__nop;
+ case TRACKTYPE_RESAMPLE:
+ return track__genericResample;
+ case TRACKTYPE_NORESAMPLEMONO:
+ return track__16BitsMono;
+ case TRACKTYPE_NORESAMPLE:
+ return track__16BitsStereo;
+ default:
+ LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
+ break;
+ }
+ }
+ LOG_ALWAYS_FATAL_IF(channels != FCC_2); // TODO: must be stereo right now
+ switch (trackType) {
+ case TRACKTYPE_NOP:
+ return track__nop;
+ case TRACKTYPE_RESAMPLE:
+ switch (mixerInFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ return (AudioMixer::hook_t)
+ track__Resample<MIXTYPE_MULTI, 2, float, float, int32_t>;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ return (AudioMixer::hook_t)\
+ track__Resample<MIXTYPE_MULTI, 2, int32_t, int16_t, int32_t>;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+ break;
+ }
+ break;
+ case TRACKTYPE_NORESAMPLEMONO:
+ switch (mixerInFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ return (AudioMixer::hook_t)
+ track__NoResample<MIXTYPE_MONOEXPAND, 2, float, float, int32_t>;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ return (AudioMixer::hook_t)
+ track__NoResample<MIXTYPE_MONOEXPAND, 2, int32_t, int16_t, int32_t>;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+ break;
+ }
+ break;
+ case TRACKTYPE_NORESAMPLE:
+ switch (mixerInFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ return (AudioMixer::hook_t)
+ track__NoResample<MIXTYPE_MULTI, 2, float, float, int32_t>;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ return (AudioMixer::hook_t)
+ track__NoResample<MIXTYPE_MULTI, 2, int32_t, int16_t, int32_t>;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+ break;
+ }
+ break;
+ default:
+ LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
+ break;
+ }
+ return NULL;
+}
+
+/* Returns the proper process hook for mixing tracks. Currently works only for
+ * PROCESSTYPE_NORESAMPLEONETRACK, a mix involving one track, no resampling.
+ */
+AudioMixer::process_hook_t AudioMixer::getProcessHook(int processType, int channels,
+ audio_format_t mixerInFormat, audio_format_t mixerOutFormat)
+{
+ if (processType != PROCESSTYPE_NORESAMPLEONETRACK) { // Only NORESAMPLEONETRACK
+ LOG_ALWAYS_FATAL("bad processType: %d", processType);
+ return NULL;
+ }
+ if (!kUseNewMixer && channels == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
+ return process__OneTrack16BitsStereoNoResampling;
+ }
+ LOG_ALWAYS_FATAL_IF(channels != FCC_2); // TODO: must be stereo right now
+ switch (mixerInFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ switch (mixerOutFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY, 2,
+ float, float, int32_t>;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY, 2,
+ int16_t, float, int32_t>;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+ break;
+ }
+ break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ switch (mixerOutFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY, 2,
+ float, int16_t, int32_t>;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY, 2,
+ int16_t, int16_t, int32_t>;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+ break;
+ }
+ break;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+ break;
+ }
+ return NULL;
+}
+
// ----------------------------------------------------------------------------
}; // namespace android
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index 766ff60..e6de00c 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -221,6 +221,7 @@
// 16-byte boundary
+ bool needsRamp() { return (volumeInc[0] | volumeInc[1] | auxInc) != 0; }
bool setResampler(uint32_t sampleRate, uint32_t devSampleRate);
bool doesResample() const { return resampler != NULL; }
void resetResampler() { if (resampler != NULL) resampler->reset(); }
@@ -229,12 +230,14 @@
resampler->getUnreleasedFrames() : 0; };
};
+ typedef void (*process_hook_t)(state_t* state, int64_t pts);
+
// pad to 32-bytes to fill cache line
struct state_t {
uint32_t enabledTracks;
uint32_t needsChanged;
size_t frameCount;
- void (*hook)(state_t* state, int64_t pts); // one of process__*, never NULL
+ process_hook_t hook; // one of process__*, never NULL
int32_t *outputTemp;
int32_t *resampleTemp;
NBLog::Writer* mLog;
@@ -345,6 +348,38 @@
static uint64_t sLocalTimeFreq;
static pthread_once_t sOnceControl;
static void sInitRoutine();
+
+ // multi-format process hooks
+ template <int MIXTYPE, int NCHAN, typename TO, typename TI, typename TA>
+ static void process_NoResampleOneTrack(state_t* state, int64_t pts);
+
+ // multi-format track hooks
+ template <int MIXTYPE, int NCHAN, typename TO, typename TI, typename TA>
+ static void track__Resample(track_t* t, TO* out, size_t frameCount,
+ TO* temp __unused, TA* aux);
+ template <int MIXTYPE, int NCHAN, typename TO, typename TI, typename TA>
+ static void track__NoResample(track_t* t, TO* out, size_t frameCount,
+ TO* temp __unused, TA* aux);
+
+ static void convertMixerFormat(void *out, audio_format_t mixerOutFormat,
+ void *in, audio_format_t mixerInFormat, size_t sampleCount);
+
+ // hook types
+ enum {
+ PROCESSTYPE_NORESAMPLEONETRACK,
+ };
+ enum {
+ TRACKTYPE_NOP,
+ TRACKTYPE_RESAMPLE,
+ TRACKTYPE_NORESAMPLE,
+ TRACKTYPE_NORESAMPLEMONO,
+ };
+
+ // functions for determining the proper process and track hooks.
+ static process_hook_t getProcessHook(int processType, int channels,
+ audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
+ static hook_t getTrackHook(int trackType, int channels,
+ audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
};
// ----------------------------------------------------------------------------
diff --git a/services/audioflinger/AudioMixerOps.h b/services/audioflinger/AudioMixerOps.h
new file mode 100644
index 0000000..de92946
--- /dev/null
+++ b/services/audioflinger/AudioMixerOps.h
@@ -0,0 +1,361 @@
+/*
+ * Copyright (C) 2014 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_AUDIO_MIXER_OPS_H
+#define ANDROID_AUDIO_MIXER_OPS_H
+
+namespace android {
+
+/* Behavior of is_same<>::value is true if the types are identical,
+ * false otherwise. Identical to the STL std::is_same.
+ */
+template<typename T, typename U>
+struct is_same
+{
+ static const bool value = false;
+};
+
+template<typename T>
+struct is_same<T, T> // partial specialization
+{
+ static const bool value = true;
+};
+
+
+/* MixMul is a multiplication operator to scale an audio input signal
+ * by a volume gain, with the formula:
+ *
+ * O(utput) = I(nput) * V(olume)
+ *
+ * The output, input, and volume may have different types.
+ * There are 27 variants, of which 14 are actually defined in an
+ * explicitly templated class.
+ *
+ * The following type variables and the underlying meaning:
+ *
+ * Output type TO: int32_t (Q4.27) or int16_t (Q.15) or float [-1,1]
+ * Input signal type TI: int32_t (Q4.27) or int16_t (Q.15) or float [-1,1]
+ * Volume type TV: int32_t (U4.28) or int16_t (U4.12) or float [-1,1]
+ *
+ * For high precision audio, only the <TO, TI, TV> = <float, float, float>
+ * needs to be accelerated. This is perhaps the easiest form to do quickly as well.
+ */
+
+template <typename TO, typename TI, typename TV>
+inline TO MixMul(TI value, TV volume) {
+ COMPILE_TIME_ASSERT_FUNCTION_SCOPE(false);
+ // should not be here :-).
+ // To avoid mistakes, this template is always specialized.
+ return value * volume;
+}
+
+template <>
+inline int32_t MixMul<int32_t, int16_t, int16_t>(int16_t value, int16_t volume) {
+ return value * volume;
+}
+
+template <>
+inline int32_t MixMul<int32_t, int32_t, int16_t>(int32_t value, int16_t volume) {
+ return (value >> 12) * volume;
+}
+
+template <>
+inline int32_t MixMul<int32_t, int16_t, int32_t>(int16_t value, int32_t volume) {
+ return value * (volume >> 16);
+}
+
+template <>
+inline int32_t MixMul<int32_t, int32_t, int32_t>(int32_t value, int32_t volume) {
+ return (value >> 12) * (volume >> 16);
+}
+
+template <>
+inline float MixMul<float, float, int16_t>(float value, int16_t volume) {
+ static const float norm = 1. / (1 << 12);
+ return value * volume * norm;
+}
+
+template <>
+inline float MixMul<float, float, int32_t>(float value, int32_t volume) {
+ static const float norm = 1. / (1 << 28);
+ return value * volume * norm;
+}
+
+template <>
+inline int16_t MixMul<int16_t, float, int16_t>(float value, int16_t volume) {
+ return clamp16_from_float(MixMul<float, float, int16_t>(value, volume));
+}
+
+template <>
+inline int16_t MixMul<int16_t, float, int32_t>(float value, int32_t volume) {
+ return clamp16_from_float(MixMul<float, float, int32_t>(value, volume));
+}
+
+template <>
+inline float MixMul<float, int16_t, int16_t>(int16_t value, int16_t volume) {
+ static const float norm = 1. / (1 << (15 + 12));
+ return static_cast<float>(value) * static_cast<float>(volume) * norm;
+}
+
+template <>
+inline float MixMul<float, int16_t, int32_t>(int16_t value, int32_t volume) {
+ static const float norm = 1. / (1ULL << (15 + 28));
+ return static_cast<float>(value) * static_cast<float>(volume) * norm;
+}
+
+template <>
+inline int16_t MixMul<int16_t, int16_t, int16_t>(int16_t value, int16_t volume) {
+ return clamp16(MixMul<int32_t, int16_t, int16_t>(value, volume) >> 12);
+}
+
+template <>
+inline int16_t MixMul<int16_t, int32_t, int16_t>(int32_t value, int16_t volume) {
+ return clamp16(MixMul<int32_t, int32_t, int16_t>(value, volume) >> 12);
+}
+
+template <>
+inline int16_t MixMul<int16_t, int16_t, int32_t>(int16_t value, int32_t volume) {
+ return clamp16(MixMul<int32_t, int16_t, int32_t>(value, volume) >> 12);
+}
+
+template <>
+inline int16_t MixMul<int16_t, int32_t, int32_t>(int32_t value, int32_t volume) {
+ return clamp16(MixMul<int32_t, int32_t, int32_t>(value, volume) >> 12);
+}
+
+/*
+ * MixAccum is used to add into an accumulator register of a possibly different
+ * type. The TO and TI types are the same as MixMul.
+ */
+
+template <typename TO, typename TI>
+inline void MixAccum(TO *auxaccum, TI value) {
+ if (!is_same<TO, TI>::value) {
+ LOG_ALWAYS_FATAL("MixAccum type not properly specialized: %d %d\n",
+ sizeof(TO), sizeof(TI));
+ }
+ *auxaccum += value;
+}
+
+template<>
+inline void MixAccum<float, int16_t>(float *auxaccum, int16_t value) {
+ static const float norm = 1. / (1 << 15);
+ *auxaccum += norm * value;
+}
+
+template<>
+inline void MixAccum<float, int32_t>(float *auxaccum, int32_t value) {
+ static const float norm = 1. / (1 << 27);
+ *auxaccum += norm * value;
+}
+
+template<>
+inline void MixAccum<int32_t, int16_t>(int32_t *auxaccum, int16_t value) {
+ *auxaccum += value << 12;
+}
+
+template<>
+inline void MixAccum<int32_t, float>(int32_t *auxaccum, float value) {
+ *auxaccum += clampq4_27_from_float(value);
+}
+
+/* MixMulAux is just like MixMul except it combines with
+ * an accumulator operation MixAccum.
+ */
+
+template <typename TO, typename TI, typename TV, typename TA>
+inline TO MixMulAux(TI value, TV volume, TA *auxaccum) {
+ MixAccum<TA, TI>(auxaccum, value);
+ return MixMul<TO, TI, TV>(value, volume);
+}
+
+/* MIXTYPE is used to determine how the samples in the input frame
+ * are mixed with volume gain into the output frame.
+ * See the volumeRampMulti functions below for more details.
+ */
+enum {
+ MIXTYPE_MULTI,
+ MIXTYPE_MONOEXPAND,
+ MIXTYPE_MULTI_SAVEONLY,
+};
+
+/*
+ * The volumeRampMulti and volumeRamp functions take a MIXTYPE
+ * which indicates the per-frame mixing and accumulation strategy.
+ *
+ * MIXTYPE_MULTI:
+ * NCHAN represents number of input and output channels.
+ * TO: int32_t (Q4.27) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TV: int32_t (U4.28) or int16_t (U4.12) or float
+ * vol: represents a volume array.
+ *
+ * This accumulates into the out pointer.
+ *
+ * MIXTYPE_MONOEXPAND:
+ * Single input channel. NCHAN represents number of output channels.
+ * TO: int32_t (Q4.27) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TV: int32_t (U4.28) or int16_t (U4.12) or float
+ * Input channel count is 1.
+ * vol: represents volume array.
+ *
+ * This accumulates into the out pointer.
+ *
+ * MIXTYPE_MULTI_SAVEONLY:
+ * NCHAN represents number of input and output channels.
+ * TO: int16_t (Q.15) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TV: int32_t (U4.28) or int16_t (U4.12) or float
+ * vol: represents a volume array.
+ *
+ * MIXTYPE_MULTI_SAVEONLY does not accumulate into the out pointer.
+ */
+
+template <int MIXTYPE, int NCHAN,
+ typename TO, typename TI, typename TV, typename TA, typename TAV>
+inline void volumeRampMulti(TO* out, size_t frameCount,
+ const TI* in, TA* aux, TV *vol, const TV *volinc, TAV *vola, TAV volainc)
+{
+#ifdef ALOGVV
+ ALOGVV("volumeRampMulti, MIXTYPE:%d\n", MIXTYPE);
+#endif
+ if (aux != NULL) {
+ do {
+ TA auxaccum = 0;
+ switch (MIXTYPE) {
+ case MIXTYPE_MULTI:
+ for (int i = 0; i < NCHAN; ++i) {
+ *out++ += MixMulAux<TO, TI, TV, TA>(*in++, vol[i], &auxaccum);
+ vol[i] += volinc[i];
+ }
+ break;
+ case MIXTYPE_MULTI_SAVEONLY:
+ for (int i = 0; i < NCHAN; ++i) {
+ *out++ = MixMulAux<TO, TI, TV, TA>(*in++, vol[i], &auxaccum);
+ vol[i] += volinc[i];
+ }
+ break;
+ case MIXTYPE_MONOEXPAND:
+ for (int i = 0; i < NCHAN; ++i) {
+ *out++ += MixMulAux<TO, TI, TV, TA>(*in, vol[i], &auxaccum);
+ vol[i] += volinc[i];
+ }
+ in++;
+ break;
+ default:
+ LOG_ALWAYS_FATAL("invalid mixtype %d", MIXTYPE);
+ break;
+ }
+ auxaccum /= NCHAN;
+ *aux++ += MixMul<TA, TA, TAV>(auxaccum, *vola);
+ vola[0] += volainc;
+ } while (--frameCount);
+ } else {
+ do {
+ switch (MIXTYPE) {
+ case MIXTYPE_MULTI:
+ for (int i = 0; i < NCHAN; ++i) {
+ *out++ += MixMul<TO, TI, TV>(*in++, vol[i]);
+ vol[i] += volinc[i];
+ }
+ break;
+ case MIXTYPE_MULTI_SAVEONLY:
+ for (int i = 0; i < NCHAN; ++i) {
+ *out++ = MixMul<TO, TI, TV>(*in++, vol[i]);
+ vol[i] += volinc[i];
+ }
+ break;
+ case MIXTYPE_MONOEXPAND:
+ for (int i = 0; i < NCHAN; ++i) {
+ *out++ += MixMul<TO, TI, TV>(*in, vol[i]);
+ vol[i] += volinc[i];
+ }
+ in++;
+ break;
+ default:
+ LOG_ALWAYS_FATAL("invalid mixtype %d", MIXTYPE);
+ break;
+ }
+ } while (--frameCount);
+ }
+}
+
+template <int MIXTYPE, int NCHAN,
+ typename TO, typename TI, typename TV, typename TA, typename TAV>
+inline void volumeMulti(TO* out, size_t frameCount,
+ const TI* in, TA* aux, const TV *vol, TAV vola)
+{
+#ifdef ALOGVV
+ ALOGVV("volumeMulti MIXTYPE:%d\n", MIXTYPE);
+#endif
+ if (aux != NULL) {
+ do {
+ TA auxaccum = 0;
+ switch (MIXTYPE) {
+ case MIXTYPE_MULTI:
+ for (int i = 0; i < NCHAN; ++i) {
+ *out++ += MixMulAux<TO, TI, TV, TA>(*in++, vol[i], &auxaccum);
+ }
+ break;
+ case MIXTYPE_MULTI_SAVEONLY:
+ for (int i = 0; i < NCHAN; ++i) {
+ *out++ = MixMulAux<TO, TI, TV, TA>(*in++, vol[i], &auxaccum);
+ }
+ break;
+ case MIXTYPE_MONOEXPAND:
+ for (int i = 0; i < NCHAN; ++i) {
+ *out++ += MixMulAux<TO, TI, TV, TA>(*in, vol[i], &auxaccum);
+ }
+ in++;
+ break;
+ default:
+ LOG_ALWAYS_FATAL("invalid mixtype %d", MIXTYPE);
+ break;
+ }
+ auxaccum /= NCHAN;
+ *aux++ += MixMul<TA, TA, TAV>(auxaccum, vola);
+ } while (--frameCount);
+ } else {
+ do {
+ switch (MIXTYPE) {
+ case MIXTYPE_MULTI:
+ for (int i = 0; i < NCHAN; ++i) {
+ *out++ += MixMul<TO, TI, TV>(*in++, vol[i]);
+ }
+ break;
+ case MIXTYPE_MULTI_SAVEONLY:
+ for (int i = 0; i < NCHAN; ++i) {
+ *out++ = MixMul<TO, TI, TV>(*in++, vol[i]);
+ }
+ break;
+ case MIXTYPE_MONOEXPAND:
+ for (int i = 0; i < NCHAN; ++i) {
+ *out++ += MixMul<TO, TI, TV>(*in, vol[i]);
+ }
+ in++;
+ break;
+ default:
+ LOG_ALWAYS_FATAL("invalid mixtype %d", MIXTYPE);
+ break;
+ }
+ } while (--frameCount);
+ }
+}
+
+};
+
+#endif /* ANDROID_AUDIO_MIXER_OPS_H */
diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index 562c4ea..38c9061 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -40,8 +40,8 @@
class AudioResamplerOrder1 : public AudioResampler {
public:
- AudioResamplerOrder1(int bitDepth, int inChannelCount, int32_t sampleRate) :
- AudioResampler(bitDepth, inChannelCount, sampleRate, LOW_QUALITY), mX0L(0), mX0R(0) {
+ AudioResamplerOrder1(int inChannelCount, int32_t sampleRate) :
+ AudioResampler(inChannelCount, sampleRate, LOW_QUALITY), mX0L(0), mX0R(0) {
}
virtual void resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
@@ -145,7 +145,7 @@
static pthread_mutex_t mutex = PTHREAD_MUTEX_INITIALIZER;
static uint32_t currentMHz = 0;
-AudioResampler* AudioResampler::create(int bitDepth, int inChannelCount,
+AudioResampler* AudioResampler::create(audio_format_t format, int inChannelCount,
int32_t sampleRate, src_quality quality) {
bool atFinalQuality;
@@ -216,33 +216,40 @@
default:
case LOW_QUALITY:
ALOGV("Create linear Resampler");
- resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate);
+ LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT);
+ resampler = new AudioResamplerOrder1(inChannelCount, sampleRate);
break;
case MED_QUALITY:
ALOGV("Create cubic Resampler");
- resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate);
+ LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT);
+ resampler = new AudioResamplerCubic(inChannelCount, sampleRate);
break;
case HIGH_QUALITY:
ALOGV("Create HIGH_QUALITY sinc Resampler");
- resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate);
+ LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT);
+ resampler = new AudioResamplerSinc(inChannelCount, sampleRate);
break;
case VERY_HIGH_QUALITY:
ALOGV("Create VERY_HIGH_QUALITY sinc Resampler = %d", quality);
- resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate, quality);
+ LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT);
+ resampler = new AudioResamplerSinc(inChannelCount, sampleRate, quality);
break;
case DYN_LOW_QUALITY:
case DYN_MED_QUALITY:
case DYN_HIGH_QUALITY:
ALOGV("Create dynamic Resampler = %d", quality);
- if (bitDepth == 32) { /* bitDepth == 32 signals float precision */
- resampler = new AudioResamplerDyn<float, float, float>(bitDepth, inChannelCount,
- sampleRate, quality);
- } else if (quality == DYN_HIGH_QUALITY) {
- resampler = new AudioResamplerDyn<int32_t, int16_t, int32_t>(bitDepth, inChannelCount,
+ if (format == AUDIO_FORMAT_PCM_FLOAT) {
+ resampler = new AudioResamplerDyn<float, float, float>(inChannelCount,
sampleRate, quality);
} else {
- resampler = new AudioResamplerDyn<int16_t, int16_t, int32_t>(bitDepth, inChannelCount,
- sampleRate, quality);
+ LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT);
+ if (quality == DYN_HIGH_QUALITY) {
+ resampler = new AudioResamplerDyn<int32_t, int16_t, int32_t>(inChannelCount,
+ sampleRate, quality);
+ } else {
+ resampler = new AudioResamplerDyn<int16_t, int16_t, int32_t>(inChannelCount,
+ sampleRate, quality);
+ }
}
break;
}
@@ -252,26 +259,25 @@
return resampler;
}
-AudioResampler::AudioResampler(int bitDepth, int inChannelCount,
+AudioResampler::AudioResampler(int inChannelCount,
int32_t sampleRate, src_quality quality) :
- mBitDepth(bitDepth), mChannelCount(inChannelCount),
- mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputIndex(0),
- mPhaseFraction(0), mLocalTimeFreq(0),
- mPTS(AudioBufferProvider::kInvalidPTS), mQuality(quality) {
- // sanity check on format
- if ((bitDepth != 16) ||(inChannelCount < 1) || (inChannelCount > 2)) {
- ALOGE("Unsupported sample format, %d bits, %d channels", bitDepth,
- inChannelCount);
- // ALOG_ASSERT(0);
+ mChannelCount(inChannelCount),
+ mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputIndex(0),
+ mPhaseFraction(0), mLocalTimeFreq(0),
+ mPTS(AudioBufferProvider::kInvalidPTS), mQuality(quality) {
+
+ if (inChannelCount < 1
+ || inChannelCount > (quality < DYN_LOW_QUALITY ? 2 : 8)) {
+ LOG_ALWAYS_FATAL("Unsupported sample format %d quality %d channels",
+ quality, inChannelCount);
}
if (sampleRate <= 0) {
- ALOGE("Unsupported sample rate %d Hz", sampleRate);
+ LOG_ALWAYS_FATAL("Unsupported sample rate %d Hz", sampleRate);
}
// initialize common members
mVolume[0] = mVolume[1] = 0;
mBuffer.frameCount = 0;
-
}
AudioResampler::~AudioResampler() {
diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h
index b84567e..be747f6 100644
--- a/services/audioflinger/AudioResampler.h
+++ b/services/audioflinger/AudioResampler.h
@@ -22,6 +22,7 @@
#include <cutils/compiler.h>
#include <media/AudioBufferProvider.h>
+#include <system/audio.h>
namespace android {
// ----------------------------------------------------------------------------
@@ -46,7 +47,7 @@
DYN_HIGH_QUALITY=7,
};
- static AudioResampler* create(int bitDepth, int inChannelCount,
+ static AudioResampler* create(audio_format_t format, int inChannelCount,
int32_t sampleRate, src_quality quality=DEFAULT_QUALITY);
virtual ~AudioResampler();
@@ -86,7 +87,7 @@
// multiplier to calculate fixed point phase increment
static const double kPhaseMultiplier;
- AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate, src_quality quality);
+ AudioResampler(int inChannelCount, int32_t sampleRate, src_quality quality);
// prevent copying
AudioResampler(const AudioResampler&);
@@ -94,7 +95,6 @@
int64_t calculateOutputPTS(int outputFrameIndex);
- const int32_t mBitDepth;
const int32_t mChannelCount;
const int32_t mSampleRate;
int32_t mInSampleRate;
diff --git a/services/audioflinger/AudioResamplerCubic.h b/services/audioflinger/AudioResamplerCubic.h
index 203b933..b315da5 100644
--- a/services/audioflinger/AudioResamplerCubic.h
+++ b/services/audioflinger/AudioResamplerCubic.h
@@ -28,8 +28,8 @@
class AudioResamplerCubic : public AudioResampler {
public:
- AudioResamplerCubic(int bitDepth, int inChannelCount, int32_t sampleRate) :
- AudioResampler(bitDepth, inChannelCount, sampleRate, MED_QUALITY) {
+ AudioResamplerCubic(int inChannelCount, int32_t sampleRate) :
+ AudioResampler(inChannelCount, sampleRate, MED_QUALITY) {
}
virtual void resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
diff --git a/services/audioflinger/AudioResamplerDyn.cpp b/services/audioflinger/AudioResamplerDyn.cpp
index 318eb57..043c803 100644
--- a/services/audioflinger/AudioResamplerDyn.cpp
+++ b/services/audioflinger/AudioResamplerDyn.cpp
@@ -38,11 +38,6 @@
namespace android {
-// generate a unique resample type compile-time constant (constexpr)
-#define RESAMPLETYPE(CHANNELS, LOCKED, STRIDE) \
- ((((CHANNELS)-1)&1) | !!(LOCKED)<<1 \
- | ((STRIDE)==8 ? 1 : (STRIDE)==16 ? 2 : 0)<<2)
-
/*
* InBuffer is a type agnostic input buffer.
*
@@ -167,9 +162,9 @@
}
template<typename TC, typename TI, typename TO>
-AudioResamplerDyn<TC, TI, TO>::AudioResamplerDyn(int bitDepth,
+AudioResamplerDyn<TC, TI, TO>::AudioResamplerDyn(
int inChannelCount, int32_t sampleRate, src_quality quality)
- : AudioResampler(bitDepth, inChannelCount, sampleRate, quality),
+ : AudioResampler(inChannelCount, sampleRate, quality),
mResampleFunc(0), mFilterSampleRate(0), mFilterQuality(DEFAULT_QUALITY),
mCoefBuffer(NULL)
{
@@ -403,12 +398,76 @@
// determine which resampler to use
// check if locked phase (works only if mPhaseIncrement has no "fractional phase bits")
int locked = (mPhaseIncrement << (sizeof(mPhaseIncrement)*8 - c.mShift)) == 0;
- int stride = (c.mHalfNumCoefs&7)==0 ? 16 : (c.mHalfNumCoefs&3)==0 ? 8 : 2;
if (locked) {
mPhaseFraction = mPhaseFraction >> c.mShift << c.mShift; // remove fractional phase
}
- setResampler(RESAMPLETYPE(mChannelCount, locked, stride));
+ // stride is the minimum number of filter coefficients processed per loop iteration.
+ // We currently only allow a stride of 16 to match with SIMD processing.
+ // This means that the filter length must be a multiple of 16,
+ // or half the filter length (mHalfNumCoefs) must be a multiple of 8.
+ //
+ // Note: A stride of 2 is achieved with non-SIMD processing.
+ int stride = ((c.mHalfNumCoefs & 7) == 0) ? 16 : 2;
+ LOG_ALWAYS_FATAL_IF(stride < 16, "Resampler stride must be 16 or more");
+ LOG_ALWAYS_FATAL_IF(mChannelCount > 8 || mChannelCount < 1,
+ "Resampler channels(%d) must be between 1 to 8", mChannelCount);
+ // stride 16 (falls back to stride 2 for machines that do not support NEON)
+ if (locked) {
+ switch (mChannelCount) {
+ case 1:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<1, true, 16>;
+ break;
+ case 2:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<2, true, 16>;
+ break;
+ case 3:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<3, true, 16>;
+ break;
+ case 4:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<4, true, 16>;
+ break;
+ case 5:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<5, true, 16>;
+ break;
+ case 6:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<6, true, 16>;
+ break;
+ case 7:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<7, true, 16>;
+ break;
+ case 8:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<8, true, 16>;
+ break;
+ }
+ } else {
+ switch (mChannelCount) {
+ case 1:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<1, false, 16>;
+ break;
+ case 2:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<2, false, 16>;
+ break;
+ case 3:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<3, false, 16>;
+ break;
+ case 4:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<4, false, 16>;
+ break;
+ case 5:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<5, false, 16>;
+ break;
+ case 6:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<6, false, 16>;
+ break;
+ case 7:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<7, false, 16>;
+ break;
+ case 8:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<8, false, 16>;
+ break;
+ }
+ }
#ifdef DEBUG_RESAMPLER
printf("channels:%d %s stride:%d %s coef:%d shift:%d\n",
mChannelCount, locked ? "locked" : "interpolated",
@@ -424,34 +483,12 @@
}
template<typename TC, typename TI, typename TO>
-void AudioResamplerDyn<TC, TI, TO>::setResampler(unsigned resampleType)
-{
- // stride 16 (falls back to stride 2 for machines that do not support NEON)
- switch (resampleType) {
- case RESAMPLETYPE(1, true, 16):
- mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<1, true, 16>;
- return;
- case RESAMPLETYPE(2, true, 16):
- mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<2, true, 16>;
- return;
- case RESAMPLETYPE(1, false, 16):
- mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<1, false, 16>;
- return;
- case RESAMPLETYPE(2, false, 16):
- mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<2, false, 16>;
- return;
- default:
- LOG_ALWAYS_FATAL("Invalid resampler type: %u", resampleType);
- mResampleFunc = NULL;
- return;
- }
-}
-
-template<typename TC, typename TI, typename TO>
template<int CHANNELS, bool LOCKED, int STRIDE>
void AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount,
AudioBufferProvider* provider)
{
+ // TODO Mono -> Mono is not supported. OUTPUT_CHANNELS reflects minimum of stereo out.
+ const int OUTPUT_CHANNELS = (CHANNELS < 2) ? 2 : CHANNELS;
const Constants& c(mConstants);
const TC* const coefs = mConstants.mFirCoefs;
TI* impulse = mInBuffer.getImpulse();
@@ -459,7 +496,7 @@
uint32_t phaseFraction = mPhaseFraction;
const uint32_t phaseIncrement = mPhaseIncrement;
size_t outputIndex = 0;
- size_t outputSampleCount = outFrameCount * 2; // stereo output
+ size_t outputSampleCount = outFrameCount * OUTPUT_CHANNELS;
const uint32_t phaseWrapLimit = c.mL << c.mShift;
size_t inFrameCount = (phaseIncrement * (uint64_t)outFrameCount + phaseFraction)
/ phaseWrapLimit;
@@ -490,7 +527,7 @@
while (mBuffer.frameCount == 0 && inFrameCount > 0) {
mBuffer.frameCount = inFrameCount;
provider->getNextBuffer(&mBuffer,
- calculateOutputPTS(outputIndex / 2));
+ calculateOutputPTS(outputIndex / OUTPUT_CHANNELS));
if (mBuffer.raw == NULL) {
goto resample_exit;
}
@@ -538,7 +575,8 @@
phaseFraction, phaseWrapLimit,
coefShift, halfNumCoefs, coefs,
impulse, volumeSimd);
- outputIndex += 2;
+
+ outputIndex += OUTPUT_CHANNELS;
phaseFraction += phaseIncrement;
while (phaseFraction >= phaseWrapLimit) {
diff --git a/services/audioflinger/AudioResamplerDyn.h b/services/audioflinger/AudioResamplerDyn.h
index 8c56319..3044bfb 100644
--- a/services/audioflinger/AudioResamplerDyn.h
+++ b/services/audioflinger/AudioResamplerDyn.h
@@ -41,7 +41,7 @@
template<typename TC, typename TI, typename TO>
class AudioResamplerDyn: public AudioResampler {
public:
- AudioResamplerDyn(int bitDepth, int inChannelCount,
+ AudioResamplerDyn(int inChannelCount,
int32_t sampleRate, src_quality quality);
virtual ~AudioResamplerDyn();
@@ -110,12 +110,10 @@
void createKaiserFir(Constants &c, double stopBandAtten,
int inSampleRate, int outSampleRate, double tbwCheat);
- void setResampler(unsigned resampleType);
-
template<int CHANNELS, bool LOCKED, int STRIDE>
void resample(TO* out, size_t outFrameCount, AudioBufferProvider* provider);
- // declare a pointer to member function for resample
+ // define a pointer to member function type for resample
typedef void (AudioResamplerDyn<TC, TI, TO>::*resample_ABP_t)(TO* out,
size_t outFrameCount, AudioBufferProvider* provider);
diff --git a/services/audioflinger/AudioResamplerFirProcess.h b/services/audioflinger/AudioResamplerFirProcess.h
index 76d2d66..bb0f1c9 100644
--- a/services/audioflinger/AudioResamplerFirProcess.h
+++ b/services/audioflinger/AudioResamplerFirProcess.h
@@ -44,14 +44,14 @@
void mac(float& l, float& r, TC coef, const float* samples)
{
l += *samples++ * coef;
- r += *samples++ * coef;
+ r += *samples * coef;
}
template<typename TC>
static inline
void mac(float& l, TC coef, const float* samples)
{
- l += *samples++ * coef;
+ l += *samples * coef;
}
/* variant for output type TO = int32_t output samples */
@@ -69,62 +69,48 @@
}
/*
- * Calculates a single output frame (two samples).
+ * Helper template functions for loop unrolling accumulator operations.
*
- * This function computes both the positive half FIR dot product and
- * the negative half FIR dot product, accumulates, and then applies the volume.
- *
- * This is a locked phase filter (it does not compute the interpolation).
- *
- * Use fir() to compute the proper coefficient pointers for a polyphase
- * filter bank.
+ * Unrolling the loops achieves about 2x gain.
+ * Using a recursive template rather than an array of TO[] for the accumulator
+ * values is an additional 10-20% gain.
*/
-template <int CHANNELS, int STRIDE, typename TC, typename TI, typename TO>
-static inline
-void ProcessL(TO* const out,
- int count,
- const TC* coefsP,
- const TC* coefsN,
- const TI* sP,
- const TI* sN,
- const TO* const volumeLR)
+template<int CHANNELS, typename TO>
+class Accumulator : public Accumulator<CHANNELS-1, TO> // recursive
{
- COMPILE_TIME_ASSERT_FUNCTION_SCOPE(CHANNELS >= 1 && CHANNELS <= 2)
- if (CHANNELS == 2) {
- TO l = 0;
- TO r = 0;
- do {
- mac(l, r, *coefsP++, sP);
- sP -= CHANNELS;
- mac(l, r, *coefsN++, sN);
- sN += CHANNELS;
- } while (--count > 0);
- out[0] += volumeAdjust(l, volumeLR[0]);
- out[1] += volumeAdjust(r, volumeLR[1]);
- } else { /* CHANNELS == 1 */
- TO l = 0;
- do {
- mac(l, *coefsP++, sP);
- sP -= CHANNELS;
- mac(l, *coefsN++, sN);
- sN += CHANNELS;
- } while (--count > 0);
- out[0] += volumeAdjust(l, volumeLR[0]);
- out[1] += volumeAdjust(l, volumeLR[1]);
+public:
+ inline void clear() {
+ value = 0;
+ Accumulator<CHANNELS-1, TO>::clear();
}
-}
+ template<typename TC, typename TI>
+ inline void acc(TC coef, const TI*& data) {
+ mac(value, coef, data++);
+ Accumulator<CHANNELS-1, TO>::acc(coef, data);
+ }
+ inline void volume(TO*& out, TO gain) {
+ *out++ = volumeAdjust(value, gain);
+ Accumulator<CHANNELS-1, TO>::volume(out, gain);
+ }
+
+ TO value; // one per recursive inherited base class
+};
+
+template<typename TO>
+class Accumulator<0, TO> {
+public:
+ inline void clear() {
+ }
+ template<typename TC, typename TI>
+ inline void acc(TC coef __unused, const TI*& data __unused) {
+ }
+ inline void volume(TO*& out __unused, TO gain __unused) {
+ }
+};
/*
- * Calculates a single output frame (two samples) interpolating phase.
- *
- * This function computes both the positive half FIR dot product and
- * the negative half FIR dot product, accumulates, and then applies the volume.
- *
- * This is an interpolated phase filter.
- *
- * Use fir() to compute the proper coefficient pointers for a polyphase
- * filter bank.
+ * Helper template functions for interpolating filter coefficients.
*/
template<typename TC, typename T>
@@ -159,6 +145,131 @@
return mulAdd(static_cast<int16_t>(lerp), (coef_1-coef_0)<<1, coef_0);
}
+/* class scope for passing in functions into templates */
+struct InterpCompute {
+ template<typename TC, typename TINTERP>
+ static inline
+ TC interpolatep(TC coef_0, TC coef_1, TINTERP lerp) {
+ return interpolate(coef_0, coef_1, lerp);
+ }
+
+ template<typename TC, typename TINTERP>
+ static inline
+ TC interpolaten(TC coef_0, TC coef_1, TINTERP lerp) {
+ return interpolate(coef_0, coef_1, lerp);
+ }
+};
+
+struct InterpNull {
+ template<typename TC, typename TINTERP>
+ static inline
+ TC interpolatep(TC coef_0, TC coef_1 __unused, TINTERP lerp __unused) {
+ return coef_0;
+ }
+
+ template<typename TC, typename TINTERP>
+ static inline
+ TC interpolaten(TC coef_0 __unused, TC coef_1, TINTERP lerp __unused) {
+ return coef_1;
+ }
+};
+
+/*
+ * Calculates a single output frame (two samples).
+ *
+ * The Process*() functions compute both the positive half FIR dot product and
+ * the negative half FIR dot product, accumulates, and then applies the volume.
+ *
+ * Use fir() to compute the proper coefficient pointers for a polyphase
+ * filter bank.
+ *
+ * ProcessBase() is the fundamental processing template function.
+ *
+ * ProcessL() calls ProcessBase() with TFUNC = InterpNull, for fixed/locked phase.
+ * Process() calls ProcessBase() with TFUNC = InterpCompute, for interpolated phase.
+ */
+
+template <int CHANNELS, int STRIDE, typename TFUNC, typename TC, typename TI, typename TO, typename TINTERP>
+static inline
+void ProcessBase(TO* const out,
+ int count,
+ const TC* coefsP,
+ const TC* coefsN,
+ const TI* sP,
+ const TI* sN,
+ TINTERP lerpP,
+ const TO* const volumeLR)
+{
+ COMPILE_TIME_ASSERT_FUNCTION_SCOPE(CHANNELS > 0)
+
+ if (CHANNELS > 2) {
+ // TO accum[CHANNELS];
+ Accumulator<CHANNELS, TO> accum;
+
+ // for (int j = 0; j < CHANNELS; ++j) accum[j] = 0;
+ accum.clear();
+ for (size_t i = 0; i < count; ++i) {
+ TC c = TFUNC::interpolatep(coefsP[0], coefsP[count], lerpP);
+
+ // for (int j = 0; j < CHANNELS; ++j) mac(accum[j], c, sP + j);
+ const TI *tmp_data = sP; // tmp_ptr seems to work better
+ accum.acc(c, tmp_data);
+
+ coefsP++;
+ sP -= CHANNELS;
+ c = TFUNC::interpolaten(coefsN[count], coefsN[0], lerpP);
+
+ // for (int j = 0; j < CHANNELS; ++j) mac(accum[j], c, sN + j);
+ tmp_data = sN; // tmp_ptr seems faster than directly using sN
+ accum.acc(c, tmp_data);
+
+ coefsN++;
+ sN += CHANNELS;
+ }
+ // for (int j = 0; j < CHANNELS; ++j) out[j] += volumeAdjust(accum[j], volumeLR[0]);
+ TO *tmp_out = out; // may remove if const out definition changes.
+ accum.volume(tmp_out, volumeLR[0]);
+ } else if (CHANNELS == 2) {
+ TO l = 0;
+ TO r = 0;
+ for (size_t i = 0; i < count; ++i) {
+ mac(l, r, TFUNC::interpolatep(coefsP[0], coefsP[count], lerpP), sP);
+ coefsP++;
+ sP -= CHANNELS;
+ mac(l, r, TFUNC::interpolaten(coefsN[count], coefsN[0], lerpP), sN);
+ coefsN++;
+ sN += CHANNELS;
+ }
+ out[0] += volumeAdjust(l, volumeLR[0]);
+ out[1] += volumeAdjust(r, volumeLR[1]);
+ } else { /* CHANNELS == 1 */
+ TO l = 0;
+ for (size_t i = 0; i < count; ++i) {
+ mac(l, TFUNC::interpolatep(coefsP[0], coefsP[count], lerpP), sP);
+ coefsP++;
+ sP -= CHANNELS;
+ mac(l, TFUNC::interpolaten(coefsN[count], coefsN[0], lerpP), sN);
+ coefsN++;
+ sN += CHANNELS;
+ }
+ out[0] += volumeAdjust(l, volumeLR[0]);
+ out[1] += volumeAdjust(l, volumeLR[1]);
+ }
+}
+
+template <int CHANNELS, int STRIDE, typename TC, typename TI, typename TO>
+static inline
+void ProcessL(TO* const out,
+ int count,
+ const TC* coefsP,
+ const TC* coefsN,
+ const TI* sP,
+ const TI* sN,
+ const TO* const volumeLR)
+{
+ ProcessBase<CHANNELS, STRIDE, InterpNull>(out, count, coefsP, coefsN, sP, sN, 0, volumeLR);
+}
+
template <int CHANNELS, int STRIDE, typename TC, typename TI, typename TO, typename TINTERP>
static inline
void Process(TO* const out,
@@ -172,35 +283,8 @@
TINTERP lerpP,
const TO* const volumeLR)
{
- COMPILE_TIME_ASSERT_FUNCTION_SCOPE(CHANNELS >= 1 && CHANNELS <= 2)
- adjustLerp<TC, TINTERP>(lerpP); // coefficient type adjustment for interpolation
-
- if (CHANNELS == 2) {
- TO l = 0;
- TO r = 0;
- for (size_t i = 0; i < count; ++i) {
- mac(l, r, interpolate(coefsP[0], coefsP[count], lerpP), sP);
- coefsP++;
- sP -= CHANNELS;
- mac(l, r, interpolate(coefsN[count], coefsN[0], lerpP), sN);
- coefsN++;
- sN += CHANNELS;
- }
- out[0] += volumeAdjust(l, volumeLR[0]);
- out[1] += volumeAdjust(r, volumeLR[1]);
- } else { /* CHANNELS == 1 */
- TO l = 0;
- for (size_t i = 0; i < count; ++i) {
- mac(l, interpolate(coefsP[0], coefsP[count], lerpP), sP);
- coefsP++;
- sP -= CHANNELS;
- mac(l, interpolate(coefsN[count], coefsN[0], lerpP), sN);
- coefsN++;
- sN += CHANNELS;
- }
- out[0] += volumeAdjust(l, volumeLR[0]);
- out[1] += volumeAdjust(l, volumeLR[1]);
- }
+ adjustLerp<TC, TINTERP>(lerpP); // coefficient type adjustment for interpolations
+ ProcessBase<CHANNELS, STRIDE, InterpCompute>(out, count, coefsP, coefsN, sP, sN, lerpP, volumeLR);
}
/*
diff --git a/services/audioflinger/AudioResamplerSinc.cpp b/services/audioflinger/AudioResamplerSinc.cpp
index 35553ef..60ff88e 100644
--- a/services/audioflinger/AudioResamplerSinc.cpp
+++ b/services/audioflinger/AudioResamplerSinc.cpp
@@ -452,9 +452,9 @@
// ----------------------------------------------------------------------------
-AudioResamplerSinc::AudioResamplerSinc(int bitDepth,
+AudioResamplerSinc::AudioResamplerSinc(
int inChannelCount, int32_t sampleRate, src_quality quality)
- : AudioResampler(bitDepth, inChannelCount, sampleRate, quality),
+ : AudioResampler(inChannelCount, sampleRate, quality),
mState(0), mImpulse(0), mRingFull(0), mFirCoefs(0)
{
/*
diff --git a/services/audioflinger/AudioResamplerSinc.h b/services/audioflinger/AudioResamplerSinc.h
index 1ea4474..97ae3d0 100644
--- a/services/audioflinger/AudioResamplerSinc.h
+++ b/services/audioflinger/AudioResamplerSinc.h
@@ -34,7 +34,7 @@
class AudioResamplerSinc : public AudioResampler {
public:
- AudioResamplerSinc(int bitDepth, int inChannelCount, int32_t sampleRate,
+ AudioResamplerSinc(int inChannelCount, int32_t sampleRate,
src_quality quality = HIGH_QUALITY);
virtual ~AudioResamplerSinc();
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 6130084..79bdfe8 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -54,6 +54,7 @@
return mStreamType;
}
bool isOffloaded() const { return (mFlags & IAudioFlinger::TRACK_OFFLOAD) != 0; }
+ bool isDirect() const { return (mFlags & IAudioFlinger::TRACK_DIRECT) != 0; }
status_t setParameters(const String8& keyValuePairs);
status_t attachAuxEffect(int EffectId);
void setAuxBuffer(int EffectId, int32_t *buffer);
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index d6390b1..11d13a8 100755
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -1157,12 +1157,12 @@
type_t type)
: ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type),
mNormalFrameCount(0), mSinkBuffer(NULL),
- mMixerBufferEnabled(false),
+ mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision),
mMixerBuffer(NULL),
mMixerBufferSize(0),
mMixerBufferFormat(AUDIO_FORMAT_INVALID),
mMixerBufferValid(false),
- mEffectBufferEnabled(false),
+ mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision),
mEffectBuffer(NULL),
mEffectBufferSize(0),
mEffectBufferFormat(AUDIO_FORMAT_INVALID),
@@ -1401,9 +1401,10 @@
frameCount, mFrameCount);
} else {
ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d "
- "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u "
+ "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x "
+ "sampleRate=%u mSampleRate=%u "
"hasFastMixer=%d tid=%d fastTrackAvailMask=%#x",
- isTimed, sharedBuffer.get(), frameCount, mFrameCount, format,
+ isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat,
audio_is_linear_pcm(format),
channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask);
*flags &= ~IAudioFlinger::TRACK_FAST;
@@ -1660,7 +1661,7 @@
track->mState = TrackBase::STOPPED;
if (!trackActive) {
removeTrack_l(track);
- } else if (track->isFastTrack() || track->isOffloaded()) {
+ } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) {
track->mState = TrackBase::STOPPING_1;
}
@@ -1809,11 +1810,12 @@
if (!audio_is_valid_format(mFormat)) {
LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat);
}
- if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) {
- LOG_ALWAYS_FATAL("HAL format %#x not supported for mixed output; "
- "must be AUDIO_FORMAT_PCM_16_BIT", mFormat);
+ if ((mType == MIXER || mType == DUPLICATING)
+ && !isValidPcmSinkFormat(mFormat)) {
+ LOG_FATAL("HAL format %#x not supported for mixed output",
+ mFormat);
}
- mFrameSize = audio_stream_frame_size(&mOutput->stream->common);
+ mFrameSize = audio_stream_out_frame_size(mOutput->stream);
mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common);
mFrameCount = mBufferSize / mFrameSize;
if (mFrameCount & 15) {
@@ -1868,7 +1870,9 @@
}
mNormalFrameCount = multiplier * mFrameCount;
// round up to nearest 16 frames to satisfy AudioMixer
- mNormalFrameCount = (mNormalFrameCount + 15) & ~15;
+ if (mType == MIXER || mType == DUPLICATING) {
+ mNormalFrameCount = (mNormalFrameCount + 15) & ~15;
+ }
ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount,
mNormalFrameCount);
@@ -2656,7 +2660,7 @@
if (mNormalSink != 0) {
return mNormalSink->getTimestamp(timestamp);
}
- if (mType == OFFLOAD && mOutput->stream->get_presentation_position) {
+ if ((mType == OFFLOAD || mType == DIRECT) && mOutput->stream->get_presentation_position) {
uint64_t position64;
int ret = mOutput->stream->get_presentation_position(
mOutput->stream, &position64, ×tamp.mTime);
@@ -2860,8 +2864,6 @@
}
#endif
- } else {
- mFastMixer = NULL;
}
switch (kUseFastMixer) {
@@ -2880,7 +2882,7 @@
AudioFlinger::MixerThread::~MixerThread()
{
- if (mFastMixer != NULL) {
+ if (mFastMixer != 0) {
FastMixerStateQueue *sq = mFastMixer->sq();
FastMixerState *state = sq->begin();
if (state->mCommand == FastMixerState::COLD_IDLE) {
@@ -2902,7 +2904,7 @@
ALOG_ASSERT(fastTrack->mBufferProvider != NULL);
delete fastTrack->mBufferProvider;
sq->end(false /*didModify*/);
- delete mFastMixer;
+ mFastMixer.clear();
#ifdef AUDIO_WATCHDOG
if (mAudioWatchdog != 0) {
mAudioWatchdog->requestExit();
@@ -2918,7 +2920,7 @@
uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const
{
- if (mFastMixer != NULL) {
+ if (mFastMixer != 0) {
MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
latency += (pipe->getAvgFrames() * 1000) / mSampleRate;
}
@@ -2935,7 +2937,7 @@
{
// FIXME we should only do one push per cycle; confirm this is true
// Start the fast mixer if it's not already running
- if (mFastMixer != NULL) {
+ if (mFastMixer != 0) {
FastMixerStateQueue *sq = mFastMixer->sq();
FastMixerState *state = sq->begin();
if (state->mCommand != FastMixerState::MIX_WRITE &&
@@ -2969,7 +2971,7 @@
void AudioFlinger::MixerThread::threadLoop_standby()
{
// Idle the fast mixer if it's currently running
- if (mFastMixer != NULL) {
+ if (mFastMixer != 0) {
FastMixerStateQueue *sq = mFastMixer->sq();
FastMixerState *state = sq->begin();
if (!(state->mCommand & FastMixerState::IDLE)) {
@@ -3132,7 +3134,7 @@
FastMixerState *state = NULL;
bool didModify = false;
FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED;
- if (mFastMixer != NULL) {
+ if (mFastMixer != 0) {
sq = mFastMixer->sq();
state = sq->begin();
}
@@ -3675,7 +3677,7 @@
// if !&IDLE, holds the FastMixer state to restore after new parameters processed
FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE;
- if (mFastMixer != NULL) {
+ if (mFastMixer != 0) {
FastMixerStateQueue *sq = mFastMixer->sq();
FastMixerState *state = sq->begin();
if (!(state->mCommand & FastMixerState::IDLE)) {
@@ -3780,7 +3782,7 @@
}
if (!(previousCommand & FastMixerState::IDLE)) {
- ALOG_ASSERT(mFastMixer != NULL);
+ ALOG_ASSERT(mFastMixer != 0);
FastMixerStateQueue *sq = mFastMixer->sq();
FastMixerState *state = sq->begin();
ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE);
@@ -3947,14 +3949,16 @@
// The first time a track is added we wait
// for all its buffers to be filled before processing it
uint32_t minFrames;
- if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) {
+ if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing()) {
minFrames = mNormalFrameCount;
} else {
minFrames = 1;
}
- if ((track->framesReady() >= minFrames) && track->isReady() &&
- !track->isPaused() && !track->isTerminated())
+ ALOGI("prepareTracks_l minFrames %d state %d frames ready %d, ",
+ minFrames, track->mState, track->framesReady());
+ if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() &&
+ !track->isStopping_2() && !track->isStopped())
{
ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer);
@@ -3981,17 +3985,26 @@
if (!mEffectChains.isEmpty() && last) {
mEffectChains[0]->clearInputBuffer();
}
-
- ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer);
- if ((track->sharedBuffer() != 0) || track->isTerminated() ||
- track->isStopped() || track->isPaused()) {
+ if (track->isStopping_1()) {
+ track->mState = TrackBase::STOPPING_2;
+ }
+ if ((track->sharedBuffer() != 0) || track->isStopped() ||
+ track->isStopping_2() || track->isPaused()) {
// We have consumed all the buffers of this track.
// Remove it from the list of active tracks.
- // TODO: implement behavior for compressed audio
- size_t audioHALFrames = (latency_l() * mSampleRate) / 1000;
+ size_t audioHALFrames;
+ if (audio_is_linear_pcm(mFormat)) {
+ audioHALFrames = (latency_l() * mSampleRate) / 1000;
+ } else {
+ audioHALFrames = 0;
+ }
+
size_t framesWritten = mBytesWritten / mFrameSize;
if (mStandby || !last ||
track->presentationComplete(framesWritten, audioHALFrames)) {
+ if (track->isStopping_2()) {
+ track->mState = TrackBase::STOPPED;
+ }
if (track->isStopped()) {
track->reset();
}
@@ -4453,7 +4466,7 @@
size_t audioHALFrames =
(mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000;
size_t framesWritten =
- mBytesWritten / audio_stream_frame_size(&mOutput->stream->common);
+ mBytesWritten / audio_stream_out_frame_size(mOutput->stream);
track->presentationComplete(framesWritten, audioHALFrames);
track->reset();
tracksToRemove->add(track);
@@ -5430,6 +5443,7 @@
audio_channel_mask_t channelMask,
size_t *pFrameCount,
int sessionId,
+ size_t *notificationFrames,
int uid,
IAudioFlinger::track_flags_t *flags,
pid_t tid,
@@ -5499,6 +5513,7 @@
}
}
*pFrameCount = frameCount;
+ *notificationFrames = 0; // FIXME implement
lStatus = initCheck();
if (lStatus != NO_ERROR) {
@@ -5998,7 +6013,7 @@
if (mFormat != AUDIO_FORMAT_PCM_16_BIT) {
ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat);
}
- mFrameSize = audio_stream_frame_size(&mInput->stream->common);
+ mFrameSize = audio_stream_in_frame_size(mInput->stream);
mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common);
mFrameCount = mBufferSize / mFrameSize;
// This is the formula for calculating the temporary buffer size.
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 07887fb..c265833 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -233,7 +233,6 @@
// dynamic externally-visible
uint32_t sampleRate() const { return mSampleRate; }
- uint32_t channelCount() const { return mChannelCount; }
audio_channel_mask_t channelMask() const { return mChannelMask; }
audio_format_t format() const { return mFormat; }
// Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects,
@@ -852,7 +851,7 @@
AudioMixer* mAudioMixer; // normal mixer
private:
// one-time initialization, no locks required
- FastMixer* mFastMixer; // non-NULL if there is also a fast mixer
+ sp<FastMixer> mFastMixer; // non-0 if there is also a fast mixer
sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread
// contents are not guaranteed to be consistent, no locks required
@@ -868,7 +867,7 @@
int32_t mFastMixerFutex; // for cold idle
public:
- virtual bool hasFastMixer() const { return mFastMixer != NULL; }
+ virtual bool hasFastMixer() const { return mFastMixer != 0; }
virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const {
ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks);
return mFastMixerDumpState.mTracks[fastIndex].mUnderruns;
@@ -1073,6 +1072,7 @@
audio_channel_mask_t channelMask,
size_t *pFrameCount,
int sessionId,
+ size_t *notificationFrames,
int uid,
IAudioFlinger::track_flags_t *flags,
pid_t tid,
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 1e906ad..0ef9fe5 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -706,7 +706,7 @@
if (playbackThread->mActiveTracks.indexOf(this) < 0) {
reset();
mState = STOPPED;
- } else if (!isFastTrack() && !isOffloaded()) {
+ } else if (!isFastTrack() && !isOffloaded() && !isDirect()) {
mState = STOPPED;
} else {
// For fast tracks prepareTracks_l() will set state to STOPPING_2
@@ -860,7 +860,7 @@
}
Mutex::Autolock _l(thread->mLock);
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
- if (!isOffloaded()) {
+ if (!isOffloaded() && !isDirect()) {
if (!playbackThread->mLatchQValid) {
mPreviousValid = false;
return INVALID_OPERATION;
@@ -980,8 +980,6 @@
}
if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) {
- ALOGV("presentationComplete() session %d complete: framesWritten %d",
- mSessionId, framesWritten);
triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
mAudioTrackServerProxy->setStreamEndDone();
return true;
@@ -1897,7 +1895,8 @@
if (thread->mSampleRate != sampleRate && thread->mChannelCount <= FCC_2 &&
channelCount <= FCC_2) {
// sink SR
- mResampler = AudioResampler::create(16, thread->mChannelCount, sampleRate);
+ mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_16_BIT,
+ thread->mChannelCount, sampleRate);
// source SR
mResampler->setSampleRate(thread->mSampleRate);
mResampler->setVolume(AudioMixer::UNITY_GAIN_INT, AudioMixer::UNITY_GAIN_INT);
diff --git a/services/audioflinger/test-resample.cpp b/services/audioflinger/test-resample.cpp
index e14b4ae..78c9927 100644
--- a/services/audioflinger/test-resample.cpp
+++ b/services/audioflinger/test-resample.cpp
@@ -14,8 +14,6 @@
* limitations under the License.
*/
-#include "AudioResampler.h"
-#include <media/AudioBufferProvider.h>
#include <unistd.h>
#include <stdio.h>
#include <stdlib.h>
@@ -30,6 +28,8 @@
#include <audio_utils/primitives.h>
#include <audio_utils/sndfile.h>
#include <utils/Vector.h>
+#include <media/AudioBufferProvider.h>
+#include "AudioResampler.h"
using namespace android;
@@ -329,7 +329,7 @@
printf("%zu input frames\n", input_frames);
}
- int bit_depth = useFloat ? 32 : 16;
+ audio_format_t format = useFloat ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
int output_channels = channels > 2 ? channels : 2; // output is at least stereo samples
size_t output_framesize = output_channels * (useFloat ? sizeof(float) : sizeof(int32_t));
size_t output_frames = ((int64_t) input_frames * output_freq) / input_freq;
@@ -342,7 +342,7 @@
//
// On fast devices, filters should be generated between 0.1ms - 1ms.
// (single threaded).
- AudioResampler* resampler = AudioResampler::create(bit_depth, channels,
+ AudioResampler* resampler = AudioResampler::create(format, channels,
8000, quality);
int looplimit = 100;
timespec start, end;
@@ -380,7 +380,7 @@
}
void* output_vaddr = malloc(output_size);
- AudioResampler* resampler = AudioResampler::create(bit_depth, channels,
+ AudioResampler* resampler = AudioResampler::create(format, channels,
output_freq, quality);
diff --git a/services/audioflinger/tests/Android.mk b/services/audioflinger/tests/Android.mk
index 874f18f..7bba05b 100644
--- a/services/audioflinger/tests/Android.mk
+++ b/services/audioflinger/tests/Android.mk
@@ -1,5 +1,8 @@
# Build the unit tests for audioflinger
+#
+# resampler unit test
+#
LOCAL_PATH:= $(call my-dir)
include $(CLEAR_VARS)
@@ -20,6 +23,7 @@
bionic/libstdc++/include \
external/gtest/include \
external/stlport/stlport \
+ $(call include-path-for, audio-utils) \
frameworks/av/services/audioflinger
LOCAL_SRC_FILES := \
@@ -29,3 +33,41 @@
LOCAL_MODULE_TAGS := tests
include $(BUILD_EXECUTABLE)
+
+#
+# audio mixer test tool
+#
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES:= \
+ test-mixer.cpp \
+ ../AudioMixer.cpp.arm \
+
+LOCAL_C_INCLUDES := \
+ bionic \
+ bionic/libstdc++/include \
+ external/stlport/stlport \
+ $(call include-path-for, audio-effects) \
+ $(call include-path-for, audio-utils) \
+ frameworks/av/services/audioflinger
+
+LOCAL_STATIC_LIBRARIES := \
+ libsndfile
+
+LOCAL_SHARED_LIBRARIES := \
+ libstlport \
+ libeffects \
+ libnbaio \
+ libcommon_time_client \
+ libaudioresampler \
+ libaudioutils \
+ libdl \
+ libcutils \
+ libutils \
+ liblog
+
+LOCAL_MODULE:= test-mixer
+
+LOCAL_MODULE_TAGS := optional
+
+include $(BUILD_EXECUTABLE)
diff --git a/services/audioflinger/tests/mixer_to_wav_tests.sh b/services/audioflinger/tests/mixer_to_wav_tests.sh
new file mode 100755
index 0000000..93bff47
--- /dev/null
+++ b/services/audioflinger/tests/mixer_to_wav_tests.sh
@@ -0,0 +1,134 @@
+#!/bin/bash
+#
+# This script uses test-mixer to generate WAV files
+# for evaluation of the AudioMixer component.
+#
+# Sine and chirp signals are used for input because they
+# show up as clear lines, either horizontal or diagonal,
+# on a spectrogram. This means easy verification of multiple
+# track mixing.
+#
+# After execution, look for created subdirectories like
+# mixer_i_i
+# mixer_i_f
+# mixer_f_f
+#
+# Recommend using a program such as audacity to evaluate
+# the output WAV files, e.g.
+#
+# cd testdir
+# audacity *.wav
+#
+# Using Audacity:
+#
+# Under "Waveform" view mode you can zoom into the
+# start of the WAV file to verify proper ramping.
+#
+# Select "Spectrogram" to see verify the lines
+# (sine = horizontal, chirp = diagonal) which should
+# be clear (except for around the start as the volume
+# ramping causes spectral distortion).
+
+if [ -z "$ANDROID_BUILD_TOP" ]; then
+ echo "Android build environment not set"
+ exit -1
+fi
+
+# ensure we have mm
+. $ANDROID_BUILD_TOP/build/envsetup.sh
+
+pushd $ANDROID_BUILD_TOP/frameworks/av/services/audioflinger/
+
+# build
+pwd
+mm
+
+# send to device
+echo "waiting for device"
+adb root && adb wait-for-device remount
+adb push $OUT/system/lib/libaudioresampler.so /system/lib
+adb push $OUT/system/bin/test-mixer /system/bin
+
+# createwav creates a series of WAV files testing various
+# mixer settings
+# $1 = flags
+# $2 = directory
+function createwav() {
+# create directory if it doesn't exist
+ if [ ! -d $2 ]; then
+ mkdir $2
+ fi
+
+# Test:
+# process__genericResampling
+# track__Resample / track__genericResample
+ adb shell test-mixer $1 -s 48000 \
+ -o /sdcard/tm48000gr.wav \
+ sine:2,4000,7520 chirp:2,9200 sine:1,3000,18000
+ adb pull /sdcard/tm48000gr.wav $2
+
+# Test:
+# process__genericResample
+# track__Resample / track__genericResample
+# track__NoResample / track__16BitsStereo / track__16BitsMono
+# Aux buffer
+ adb shell test-mixer $1 -s 9307 \
+ -a /sdcard/aux9307gra.wav -o /sdcard/tm9307gra.wav \
+ sine:2,1000,3000 sine:1,2000,9307 chirp:2,9307
+ adb pull /sdcard/tm9307gra.wav $2
+ adb pull /sdcard/aux9307gra.wav $2
+
+# Test:
+# process__genericNoResampling
+# track__NoResample / track__16BitsStereo / track__16BitsMono
+ adb shell test-mixer $1 -s 32000 \
+ -o /sdcard/tm32000gnr.wav \
+ sine:2,1000,32000 chirp:2,32000 sine:1,3000,32000
+ adb pull /sdcard/tm32000gnr.wav $2
+
+# Test:
+# process__genericNoResampling
+# track__NoResample / track__16BitsStereo / track__16BitsMono
+# Aux buffer
+ adb shell test-mixer $1 -s 32000 \
+ -a /sdcard/aux32000gnra.wav -o /sdcard/tm32000gnra.wav \
+ sine:2,1000,32000 chirp:2,32000 sine:1,3000,32000
+ adb pull /sdcard/tm32000gnra.wav $2
+ adb pull /sdcard/aux32000gnra.wav $2
+
+# Test:
+# process__NoResampleOneTrack / process__OneTrack16BitsStereoNoResampling
+# Downmixer
+ adb shell test-mixer $1 -s 32000 \
+ -o /sdcard/tm32000nrot.wav \
+ sine:6,1000,32000
+ adb pull /sdcard/tm32000nrot.wav $2
+
+# Test:
+# process__NoResampleOneTrack / OneTrack16BitsStereoNoResampling
+# Aux buffer
+ adb shell test-mixer $1 -s 44100 \
+ -a /sdcard/aux44100nrota.wav -o /sdcard/tm44100nrota.wav \
+ sine:2,2000,44100
+ adb pull /sdcard/tm44100nrota.wav $2
+ adb pull /sdcard/aux44100nrota.wav $2
+}
+
+#
+# Call createwav to generate WAV files in various combinations
+#
+# i_i = integer input track, integer mixer output
+# f_f = float input track, float mixer output
+# i_f = integer input track, float_mixer output
+#
+# If the mixer output is float, then the output WAV file is pcm float.
+#
+# TODO: create a "snr" like "diff" to automatically
+# compare files in these directories together.
+#
+
+createwav "" "tests/mixer_i_i"
+createwav "-f -m" "tests/mixer_f_f"
+createwav "-m" "tests/mixer_i_f"
+
+popd
diff --git a/services/audioflinger/tests/resampler_tests.cpp b/services/audioflinger/tests/resampler_tests.cpp
index 8f9c270..987162c 100644
--- a/services/audioflinger/tests/resampler_tests.cpp
+++ b/services/audioflinger/tests/resampler_tests.cpp
@@ -33,202 +33,10 @@
#include <gtest/gtest.h>
#include <media/AudioBufferProvider.h>
#include "AudioResampler.h"
+#include "test_utils.h"
-#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
-
-template<typename T, typename U>
-struct is_same
-{
- static const bool value = false;
-};
-
-template<typename T>
-struct is_same<T, T> // partial specialization
-{
- static const bool value = true;
-};
-
-template<typename T>
-static inline T convertValue(double val)
-{
- if (is_same<T, int16_t>::value) {
- return floor(val * 32767.0 + 0.5);
- } else if (is_same<T, int32_t>::value) {
- return floor(val * (1UL<<31) + 0.5);
- }
- return val; // assume float or double
-}
-
-/* Creates a type-independent audio buffer provider from
- * a buffer base address, size, framesize, and input increment array.
- *
- * No allocation or deallocation of the provided buffer is done.
- */
-class TestProvider : public android::AudioBufferProvider {
-public:
- TestProvider(const void* addr, size_t frames, size_t frameSize,
- const std::vector<size_t>& inputIncr)
- : mAddr(addr),
- mNumFrames(frames),
- mFrameSize(frameSize),
- mNextFrame(0), mUnrel(0), mInputIncr(inputIncr), mNextIdx(0)
- {
- }
-
- virtual android::status_t getNextBuffer(Buffer* buffer, int64_t pts __unused = kInvalidPTS )
- {
- size_t requestedFrames = buffer->frameCount;
- if (requestedFrames > mNumFrames - mNextFrame) {
- buffer->frameCount = mNumFrames - mNextFrame;
- }
- if (!mInputIncr.empty()) {
- size_t provided = mInputIncr[mNextIdx++];
- ALOGV("getNextBuffer() mValue[%d]=%u not %u",
- mNextIdx-1, provided, buffer->frameCount);
- if (provided < buffer->frameCount) {
- buffer->frameCount = provided;
- }
- if (mNextIdx >= mInputIncr.size()) {
- mNextIdx = 0;
- }
- }
- ALOGV("getNextBuffer() requested %u frames out of %u frames available"
- " and returned %u frames\n",
- requestedFrames, mNumFrames - mNextFrame, buffer->frameCount);
- mUnrel = buffer->frameCount;
- if (buffer->frameCount > 0) {
- buffer->raw = (char *)mAddr + mFrameSize * mNextFrame;
- return android::NO_ERROR;
- } else {
- buffer->raw = NULL;
- return android::NOT_ENOUGH_DATA;
- }
- }
-
- virtual void releaseBuffer(Buffer* buffer)
- {
- if (buffer->frameCount > mUnrel) {
- ALOGE("releaseBuffer() released %u frames but only %u available "
- "to release\n", buffer->frameCount, mUnrel);
- mNextFrame += mUnrel;
- mUnrel = 0;
- } else {
-
- ALOGV("releaseBuffer() released %u frames out of %u frames available "
- "to release\n", buffer->frameCount, mUnrel);
- mNextFrame += buffer->frameCount;
- mUnrel -= buffer->frameCount;
- }
- buffer->frameCount = 0;
- buffer->raw = NULL;
- }
-
- void reset()
- {
- mNextFrame = 0;
- }
-
- size_t getNumFrames()
- {
- return mNumFrames;
- }
-
- void setIncr(const std::vector<size_t> inputIncr)
- {
- mNextIdx = 0;
- mInputIncr = inputIncr;
- }
-
-protected:
- const void* mAddr; // base address
- size_t mNumFrames; // total frames
- int mFrameSize; // frame size (# channels * bytes per sample)
- size_t mNextFrame; // index of next frame to provide
- size_t mUnrel; // number of frames not yet released
- std::vector<size_t> mInputIncr; // number of frames provided per call
- size_t mNextIdx; // index of next entry in mInputIncr to use
-};
-
-/* Creates a buffer filled with a sine wave.
- *
- * Returns a pair consisting of the sine signal buffer and the number of frames.
- * The caller must delete[] the buffer when no longer needed (no shared_ptr<>).
- */
-template<typename T>
-static std::pair<T*, size_t> createSine(size_t channels,
- double freq, double samplingRate, double time)
-{
- double tscale = 1. / samplingRate;
- size_t frames = static_cast<size_t>(samplingRate * time);
- T* buffer = new T[frames * channels];
- for (size_t i = 0; i < frames; ++i) {
- double t = i * tscale;
- double y = sin(2. * M_PI * freq * t);
- T yt = convertValue<T>(y);
-
- for (size_t j = 0; j < channels; ++j) {
- buffer[i*channels + j] = yt / (j + 1);
- }
- }
- return std::make_pair(buffer, frames);
-}
-
-/* Creates a buffer filled with a chirp signal (a sine wave sweep).
- *
- * Returns a pair consisting of the chirp signal buffer and the number of frames.
- * The caller must delete[] the buffer when no longer needed (no shared_ptr<>).
- *
- * When creating the Chirp, note that the frequency is the true sinusoidal
- * frequency not the sampling rate.
- *
- * http://en.wikipedia.org/wiki/Chirp
- */
-template<typename T>
-static std::pair<T*, size_t> createChirp(size_t channels,
- double minfreq, double maxfreq, double samplingRate, double time)
-{
- double tscale = 1. / samplingRate;
- size_t frames = static_cast<size_t>(samplingRate * time);
- T *buffer = new T[frames * channels];
- // note the chirp constant k has a divide-by-two.
- double k = (maxfreq - minfreq) / (2. * time);
- for (size_t i = 0; i < frames; ++i) {
- double t = i * tscale;
- double y = sin(2. * M_PI * (k * t + minfreq) * t);
- T yt = convertValue<T>(y);
-
- for (size_t j = 0; j < channels; ++j) {
- buffer[i*channels + j] = yt / (j + 1);
- }
- }
- return std::make_pair(buffer, frames);
-}
-
-/* This derived class creates a buffer provider of datatype T,
- * consisting of an input signal, e.g. from createChirp().
- * The number of frames can be obtained from the base class
- * TestProvider::getNumFrames().
- */
-template <typename T>
-class SignalProvider : public TestProvider {
-public:
- SignalProvider(const std::pair<T*, size_t>& bufferInfo, size_t channels,
- const std::vector<size_t>& values)
- : TestProvider(bufferInfo.first, bufferInfo.second, channels * sizeof(T), values),
- mManagedPtr(bufferInfo.first)
- {
- }
-
- virtual ~SignalProvider()
- {
- delete[] mManagedPtr;
- }
-
-protected:
- T* mManagedPtr;
-};
-
-void resample(void *output, size_t outputFrames, const std::vector<size_t> &outputIncr,
+void resample(int channels, void *output,
+ size_t outputFrames, const std::vector<size_t> &outputIncr,
android::AudioBufferProvider *provider, android::AudioResampler *resampler)
{
for (size_t i = 0, j = 0; i < outputFrames; ) {
@@ -239,7 +47,7 @@
if (thisFrames == 0 || thisFrames > outputFrames - i) {
thisFrames = outputFrames - i;
}
- resampler->resample((int32_t*) output + 2*i, thisFrames, provider);
+ resampler->resample((int32_t*) output + channels*i, thisFrames, provider);
i += thisFrames;
}
}
@@ -257,18 +65,26 @@
}
}
-void testBufferIncrement(size_t channels, unsigned inputFreq, unsigned outputFreq,
+void testBufferIncrement(size_t channels, bool useFloat,
+ unsigned inputFreq, unsigned outputFreq,
enum android::AudioResampler::src_quality quality)
{
+ const audio_format_t format = useFloat ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
// create the provider
- std::vector<size_t> inputIncr;
- SignalProvider<int16_t> provider(createChirp<int16_t>(channels,
- 0., outputFreq/2., outputFreq, outputFreq/2000.),
- channels, inputIncr);
+ std::vector<int> inputIncr;
+ SignalProvider provider;
+ if (useFloat) {
+ provider.setChirp<float>(channels,
+ 0., outputFreq/2., outputFreq, outputFreq/2000.);
+ } else {
+ provider.setChirp<int16_t>(channels,
+ 0., outputFreq/2., outputFreq, outputFreq/2000.);
+ }
+ provider.setIncr(inputIncr);
// calculate the output size
size_t outputFrames = ((int64_t) provider.getNumFrames() * outputFreq) / inputFreq;
- size_t outputFrameSize = 2 * sizeof(int32_t);
+ size_t outputFrameSize = channels * (useFloat ? sizeof(float) : sizeof(int32_t));
size_t outputSize = outputFrameSize * outputFrames;
outputSize &= ~7;
@@ -276,7 +92,7 @@
const int volumePrecision = 12; /* typical unity gain */
android::AudioResampler* resampler;
- resampler = android::AudioResampler::create(16, channels, outputFreq, quality);
+ resampler = android::AudioResampler::create(format, channels, outputFreq, quality);
resampler->setSampleRate(inputFreq);
resampler->setVolume(1 << volumePrecision, 1 << volumePrecision);
@@ -284,7 +100,7 @@
std::vector<size_t> refIncr;
refIncr.push_back(outputFrames);
void* reference = malloc(outputSize);
- resample(reference, outputFrames, refIncr, &provider, resampler);
+ resample(channels, reference, outputFrames, refIncr, &provider, resampler);
provider.reset();
@@ -293,7 +109,7 @@
resampler->reset();
#else
delete resampler;
- resampler = android::AudioResampler::create(16, channels, outputFreq, quality);
+ resampler = android::AudioResampler::create(format, channels, outputFreq, quality);
resampler->setSampleRate(inputFreq);
resampler->setVolume(1 << volumePrecision, 1 << volumePrecision);
#endif
@@ -304,7 +120,10 @@
outIncr.push_back(2);
outIncr.push_back(3);
void* test = malloc(outputSize);
- resample(test, outputFrames, outIncr, &provider, resampler);
+ inputIncr.push_back(1);
+ inputIncr.push_back(3);
+ provider.setIncr(inputIncr);
+ resample(channels, test, outputFrames, outIncr, &provider, resampler);
// check
buffercmp(reference, test, outputFrameSize, outputFrames);
@@ -339,14 +158,15 @@
enum android::AudioResampler::src_quality quality)
{
// create the provider
- std::vector<size_t> inputIncr;
- SignalProvider<int16_t> provider(createChirp<int16_t>(channels,
- 0., inputFreq/2., inputFreq, inputFreq/2000.),
- channels, inputIncr);
+ std::vector<int> inputIncr;
+ SignalProvider provider;
+ provider.setChirp<int16_t>(channels,
+ 0., inputFreq/2., inputFreq, inputFreq/2000.);
+ provider.setIncr(inputIncr);
// calculate the output size
size_t outputFrames = ((int64_t) provider.getNumFrames() * outputFreq) / inputFreq;
- size_t outputFrameSize = 2 * sizeof(int32_t);
+ size_t outputFrameSize = channels * sizeof(int32_t);
size_t outputSize = outputFrameSize * outputFrames;
outputSize &= ~7;
@@ -354,7 +174,8 @@
const int volumePrecision = 12; /* typical unity gain */
android::AudioResampler* resampler;
- resampler = android::AudioResampler::create(16, channels, outputFreq, quality);
+ resampler = android::AudioResampler::create(AUDIO_FORMAT_PCM_16_BIT,
+ channels, outputFreq, quality);
resampler->setSampleRate(inputFreq);
resampler->setVolume(1 << volumePrecision, 1 << volumePrecision);
@@ -362,7 +183,7 @@
std::vector<size_t> refIncr;
refIncr.push_back(outputFrames);
void* reference = malloc(outputSize);
- resample(reference, outputFrames, refIncr, &provider, resampler);
+ resample(channels, reference, outputFrames, refIncr, &provider, resampler);
int32_t *out = reinterpret_cast<int32_t *>(reference);
@@ -417,7 +238,7 @@
};
for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
- testBufferIncrement(2, 48000, 32000, kQualityArray[i]);
+ testBufferIncrement(2, false, 48000, 32000, kQualityArray[i]);
}
}
@@ -434,7 +255,33 @@
};
for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
- testBufferIncrement(2, 22050, 48000, kQualityArray[i]);
+ testBufferIncrement(2, false, 22050, 48000, kQualityArray[i]);
+ }
+}
+
+TEST(audioflinger_resampler, bufferincrement_fixedphase_multi) {
+ // only dynamic quality
+ static const enum android::AudioResampler::src_quality kQualityArray[] = {
+ android::AudioResampler::DYN_LOW_QUALITY,
+ android::AudioResampler::DYN_MED_QUALITY,
+ android::AudioResampler::DYN_HIGH_QUALITY,
+ };
+
+ for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
+ testBufferIncrement(4, false, 48000, 32000, kQualityArray[i]);
+ }
+}
+
+TEST(audioflinger_resampler, bufferincrement_interpolatedphase_multi_float) {
+ // only dynamic quality
+ static const enum android::AudioResampler::src_quality kQualityArray[] = {
+ android::AudioResampler::DYN_LOW_QUALITY,
+ android::AudioResampler::DYN_MED_QUALITY,
+ android::AudioResampler::DYN_HIGH_QUALITY,
+ };
+
+ for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
+ testBufferIncrement(8, true, 22050, 48000, kQualityArray[i]);
}
}
diff --git a/services/audioflinger/tests/test-mixer.cpp b/services/audioflinger/tests/test-mixer.cpp
new file mode 100644
index 0000000..3940702
--- /dev/null
+++ b/services/audioflinger/tests/test-mixer.cpp
@@ -0,0 +1,286 @@
+/*
+ * Copyright (C) 2014 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 <stdio.h>
+#include <inttypes.h>
+#include <math.h>
+#include <vector>
+#include <audio_utils/primitives.h>
+#include <audio_utils/sndfile.h>
+#include <media/AudioBufferProvider.h>
+#include "AudioMixer.h"
+#include "test_utils.h"
+
+/* Testing is typically through creation of an output WAV file from several
+ * source inputs, to be later analyzed by an audio program such as Audacity.
+ *
+ * Sine or chirp functions are typically more useful as input to the mixer
+ * as they show up as straight lines on a spectrogram if successfully mixed.
+ *
+ * A sample shell script is provided: mixer_to_wave_tests.sh
+ */
+
+using namespace android;
+
+static void usage(const char* name) {
+ fprintf(stderr, "Usage: %s [-f] [-m]"
+ " [-s sample-rate] [-o <output-file>] [-a <aux-buffer-file>] [-P csv]"
+ " (<input-file> | <command>)+\n", name);
+ fprintf(stderr, " -f enable floating point input track\n");
+ fprintf(stderr, " -m enable floating point mixer output\n");
+ fprintf(stderr, " -s mixer sample-rate\n");
+ fprintf(stderr, " -o <output-file> WAV file, pcm16 (or float if -m specified)\n");
+ fprintf(stderr, " -a <aux-buffer-file>\n");
+ fprintf(stderr, " -P # frames provided per call to resample() in CSV format\n");
+ fprintf(stderr, " <input-file> is a WAV file\n");
+ fprintf(stderr, " <command> can be 'sine:<channels>,<frequency>,<samplerate>'\n");
+ fprintf(stderr, " 'chirp:<channels>,<samplerate>'\n");
+}
+
+static int writeFile(const char *filename, const void *buffer,
+ uint32_t sampleRate, uint32_t channels, size_t frames, bool isBufferFloat) {
+ if (filename == NULL) {
+ return 0; // ok to pass in NULL filename
+ }
+ // write output to file.
+ SF_INFO info;
+ info.frames = 0;
+ info.samplerate = sampleRate;
+ info.channels = channels;
+ info.format = SF_FORMAT_WAV | (isBufferFloat ? SF_FORMAT_FLOAT : SF_FORMAT_PCM_16);
+ printf("saving file:%s channels:%d samplerate:%d frames:%d\n",
+ filename, info.channels, info.samplerate, frames);
+ SNDFILE *sf = sf_open(filename, SFM_WRITE, &info);
+ if (sf == NULL) {
+ perror(filename);
+ return EXIT_FAILURE;
+ }
+ if (isBufferFloat) {
+ (void) sf_writef_float(sf, (float*)buffer, frames);
+ } else {
+ (void) sf_writef_short(sf, (short*)buffer, frames);
+ }
+ sf_close(sf);
+ return EXIT_SUCCESS;
+}
+
+int main(int argc, char* argv[]) {
+ const char* const progname = argv[0];
+ bool useInputFloat = false;
+ bool useMixerFloat = false;
+ bool useRamp = true;
+ uint32_t outputSampleRate = 48000;
+ uint32_t outputChannels = 2; // stereo for now
+ std::vector<int> Pvalues;
+ const char* outputFilename = NULL;
+ const char* auxFilename = NULL;
+ std::vector<int32_t> Names;
+ std::vector<SignalProvider> Providers;
+
+ for (int ch; (ch = getopt(argc, argv, "fms:o:a:P:")) != -1;) {
+ switch (ch) {
+ case 'f':
+ useInputFloat = true;
+ break;
+ case 'm':
+ useMixerFloat = true;
+ break;
+ case 's':
+ outputSampleRate = atoi(optarg);
+ break;
+ case 'o':
+ outputFilename = optarg;
+ break;
+ case 'a':
+ auxFilename = optarg;
+ break;
+ case 'P':
+ if (parseCSV(optarg, Pvalues) < 0) {
+ fprintf(stderr, "incorrect syntax for -P option\n");
+ return EXIT_FAILURE;
+ }
+ break;
+ case '?':
+ default:
+ usage(progname);
+ return EXIT_FAILURE;
+ }
+ }
+ argc -= optind;
+ argv += optind;
+
+ if (argc == 0) {
+ usage(progname);
+ return EXIT_FAILURE;
+ }
+ if ((unsigned)argc > AudioMixer::MAX_NUM_TRACKS) {
+ fprintf(stderr, "too many tracks: %d > %u", argc, AudioMixer::MAX_NUM_TRACKS);
+ return EXIT_FAILURE;
+ }
+
+ size_t outputFrames = 0;
+
+ // create providers for each track
+ Providers.resize(argc);
+ for (int i = 0; i < argc; ++i) {
+ static const char chirp[] = "chirp:";
+ static const char sine[] = "sine:";
+ static const double kSeconds = 1;
+
+ if (!strncmp(argv[i], chirp, strlen(chirp))) {
+ std::vector<int> v;
+
+ parseCSV(argv[i] + strlen(chirp), v);
+ if (v.size() == 2) {
+ printf("creating chirp(%d %d)\n", v[0], v[1]);
+ if (useInputFloat) {
+ Providers[i].setChirp<float>(v[0], 0, v[1]/2, v[1], kSeconds);
+ } else {
+ Providers[i].setChirp<int16_t>(v[0], 0, v[1]/2, v[1], kSeconds);
+ }
+ Providers[i].setIncr(Pvalues);
+ } else {
+ fprintf(stderr, "malformed input '%s'\n", argv[i]);
+ }
+ } else if (!strncmp(argv[i], sine, strlen(sine))) {
+ std::vector<int> v;
+
+ parseCSV(argv[i] + strlen(sine), v);
+ if (v.size() == 3) {
+ printf("creating sine(%d %d)\n", v[0], v[1]);
+ if (useInputFloat) {
+ Providers[i].setSine<float>(v[0], v[1], v[2], kSeconds);
+ } else {
+ Providers[i].setSine<int16_t>(v[0], v[1], v[2], kSeconds);
+ }
+ Providers[i].setIncr(Pvalues);
+ } else {
+ fprintf(stderr, "malformed input '%s'\n", argv[i]);
+ }
+ } else {
+ printf("creating filename(%s)\n", argv[i]);
+ if (useInputFloat) {
+ Providers[i].setFile<float>(argv[i]);
+ } else {
+ Providers[i].setFile<short>(argv[i]);
+ }
+ Providers[i].setIncr(Pvalues);
+ }
+ // calculate the number of output frames
+ size_t nframes = (int64_t) Providers[i].getNumFrames() * outputSampleRate
+ / Providers[i].getSampleRate();
+ if (i == 0 || outputFrames > nframes) { // choose minimum for outputFrames
+ outputFrames = nframes;
+ }
+ }
+
+ // create the output buffer.
+ const size_t outputFrameSize = outputChannels
+ * (useMixerFloat ? sizeof(float) : sizeof(int16_t));
+ const size_t outputSize = outputFrames * outputFrameSize;
+ void *outputAddr = NULL;
+ (void) posix_memalign(&outputAddr, 32, outputSize);
+ memset(outputAddr, 0, outputSize);
+
+ // create the aux buffer, if needed.
+ const size_t auxFrameSize = sizeof(int32_t); // Q4.27 always
+ const size_t auxSize = outputFrames * auxFrameSize;
+ void *auxAddr = NULL;
+ if (auxFilename) {
+ (void) posix_memalign(&auxAddr, 32, auxSize);
+ memset(auxAddr, 0, auxSize);
+ }
+
+ // create the mixer.
+ const size_t mixerFrameCount = 320; // typical numbers may range from 240 or 960
+ AudioMixer *mixer = new AudioMixer(mixerFrameCount, outputSampleRate);
+ audio_format_t inputFormat = useInputFloat
+ ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
+ audio_format_t mixerFormat = useMixerFloat
+ ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
+ float f = AudioMixer::UNITY_GAIN_FLOAT / Providers.size(); // normalize volume by # tracks
+ static float f0; // zero
+
+ // set up the tracks.
+ for (size_t i = 0; i < Providers.size(); ++i) {
+ //printf("track %d out of %d\n", i, Providers.size());
+ uint32_t channelMask = audio_channel_out_mask_from_count(Providers[i].getNumChannels());
+ int32_t name = mixer->getTrackName(channelMask,
+ inputFormat, AUDIO_SESSION_OUTPUT_MIX);
+ ALOG_ASSERT(name >= 0);
+ Names.push_back(name);
+ mixer->setBufferProvider(name, &Providers[i]);
+ mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
+ (void *) outputAddr);
+ mixer->setParameter(
+ name,
+ AudioMixer::TRACK,
+ AudioMixer::MIXER_FORMAT, (void *)mixerFormat);
+ mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::FORMAT,
+ (void *)(uintptr_t)inputFormat);
+ mixer->setParameter(
+ name,
+ AudioMixer::RESAMPLE,
+ AudioMixer::SAMPLE_RATE,
+ (void *)(uintptr_t)Providers[i].getSampleRate());
+ if (useRamp) {
+ mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, &f0);
+ mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, &f0);
+ mixer->setParameter(name, AudioMixer::RAMP_VOLUME, AudioMixer::VOLUME0, &f);
+ mixer->setParameter(name, AudioMixer::RAMP_VOLUME, AudioMixer::VOLUME1, &f);
+ } else {
+ mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, &f);
+ mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, &f);
+ }
+ if (auxFilename) {
+ mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::AUX_BUFFER,
+ (void *) auxAddr);
+ mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::AUXLEVEL, &f0);
+ mixer->setParameter(name, AudioMixer::RAMP_VOLUME, AudioMixer::AUXLEVEL, &f);
+ }
+ mixer->enable(name);
+ }
+
+ // pump the mixer to process data.
+ size_t i;
+ for (i = 0; i < outputFrames - mixerFrameCount; i += mixerFrameCount) {
+ for (size_t j = 0; j < Names.size(); ++j) {
+ mixer->setParameter(Names[j], AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
+ (char *) outputAddr + i * outputFrameSize);
+ if (auxFilename) {
+ mixer->setParameter(Names[j], AudioMixer::TRACK, AudioMixer::AUX_BUFFER,
+ (char *) auxAddr + i * auxFrameSize);
+ }
+ }
+ mixer->process(AudioBufferProvider::kInvalidPTS);
+ }
+ outputFrames = i; // reset output frames to the data actually produced.
+
+ // write to files
+ writeFile(outputFilename, outputAddr,
+ outputSampleRate, outputChannels, outputFrames, useMixerFloat);
+ if (auxFilename) {
+ // Aux buffer is always in q4_27 format for now.
+ // memcpy_to_i16_from_q4_27(), but with stereo frame count (not sample count)
+ ditherAndClamp((int32_t*)auxAddr, (int32_t*)auxAddr, outputFrames >> 1);
+ writeFile(auxFilename, auxAddr, outputSampleRate, 1, outputFrames, false);
+ }
+
+ delete mixer;
+ free(outputAddr);
+ free(auxAddr);
+ return EXIT_SUCCESS;
+}
diff --git a/services/audioflinger/tests/test_utils.h b/services/audioflinger/tests/test_utils.h
new file mode 100644
index 0000000..f954292
--- /dev/null
+++ b/services/audioflinger/tests/test_utils.h
@@ -0,0 +1,307 @@
+/*
+ * Copyright (C) 2014 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_AUDIO_TEST_UTILS_H
+#define ANDROID_AUDIO_TEST_UTILS_H
+
+#include <audio_utils/sndfile.h>
+
+#ifndef ARRAY_SIZE
+#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
+#endif
+
+template<typename T, typename U>
+struct is_same
+{
+ static const bool value = false;
+};
+
+template<typename T>
+struct is_same<T, T> // partial specialization
+{
+ static const bool value = true;
+};
+
+template<typename T>
+static inline T convertValue(double val)
+{
+ if (is_same<T, int16_t>::value) {
+ return floor(val * 32767.0 + 0.5);
+ } else if (is_same<T, int32_t>::value) {
+ return floor(val * (1UL<<31) + 0.5);
+ }
+ return val; // assume float or double
+}
+
+// Convert a list of integers in CSV format to a Vector of those values.
+// Returns the number of elements in the list, or -1 on error.
+static inline int parseCSV(const char *string, std::vector<int>& values)
+{
+ // pass 1: count the number of values and do syntax check
+ size_t numValues = 0;
+ bool hadDigit = false;
+ for (const char *p = string; ; ) {
+ switch (*p++) {
+ case '0': case '1': case '2': case '3': case '4':
+ case '5': case '6': case '7': case '8': case '9':
+ hadDigit = true;
+ break;
+ case '\0':
+ if (hadDigit) {
+ // pass 2: allocate and initialize vector of values
+ values.resize(++numValues);
+ values[0] = atoi(p = string);
+ for (size_t i = 1; i < numValues; ) {
+ if (*p++ == ',') {
+ values[i++] = atoi(p);
+ }
+ }
+ return numValues;
+ }
+ // fall through
+ case ',':
+ if (hadDigit) {
+ hadDigit = false;
+ numValues++;
+ break;
+ }
+ // fall through
+ default:
+ return -1;
+ }
+ }
+}
+
+/* Creates a type-independent audio buffer provider from
+ * a buffer base address, size, framesize, and input increment array.
+ *
+ * No allocation or deallocation of the provided buffer is done.
+ */
+class TestProvider : public android::AudioBufferProvider {
+public:
+ TestProvider(void* addr, size_t frames, size_t frameSize,
+ const std::vector<int>& inputIncr)
+ : mAddr(addr),
+ mNumFrames(frames),
+ mFrameSize(frameSize),
+ mNextFrame(0), mUnrel(0), mInputIncr(inputIncr), mNextIdx(0)
+ {
+ }
+
+ TestProvider()
+ : mAddr(NULL), mNumFrames(0), mFrameSize(0),
+ mNextFrame(0), mUnrel(0), mNextIdx(0)
+ {
+ }
+
+ void setIncr(const std::vector<int>& inputIncr) {
+ mInputIncr = inputIncr;
+ mNextIdx = 0;
+ }
+
+ virtual android::status_t getNextBuffer(Buffer* buffer, int64_t pts __unused = kInvalidPTS)
+ {
+ size_t requestedFrames = buffer->frameCount;
+ if (requestedFrames > mNumFrames - mNextFrame) {
+ buffer->frameCount = mNumFrames - mNextFrame;
+ }
+ if (!mInputIncr.empty()) {
+ size_t provided = mInputIncr[mNextIdx++];
+ ALOGV("getNextBuffer() mValue[%d]=%u not %u",
+ mNextIdx-1, provided, buffer->frameCount);
+ if (provided < buffer->frameCount) {
+ buffer->frameCount = provided;
+ }
+ if (mNextIdx >= mInputIncr.size()) {
+ mNextIdx = 0;
+ }
+ }
+ ALOGV("getNextBuffer() requested %u frames out of %u frames available"
+ " and returned %u frames\n",
+ requestedFrames, mNumFrames - mNextFrame, buffer->frameCount);
+ mUnrel = buffer->frameCount;
+ if (buffer->frameCount > 0) {
+ buffer->raw = (char *)mAddr + mFrameSize * mNextFrame;
+ return android::NO_ERROR;
+ } else {
+ buffer->raw = NULL;
+ return android::NOT_ENOUGH_DATA;
+ }
+ }
+
+ virtual void releaseBuffer(Buffer* buffer)
+ {
+ if (buffer->frameCount > mUnrel) {
+ ALOGE("releaseBuffer() released %u frames but only %u available "
+ "to release\n", buffer->frameCount, mUnrel);
+ mNextFrame += mUnrel;
+ mUnrel = 0;
+ } else {
+
+ ALOGV("releaseBuffer() released %u frames out of %u frames available "
+ "to release\n", buffer->frameCount, mUnrel);
+ mNextFrame += buffer->frameCount;
+ mUnrel -= buffer->frameCount;
+ }
+ buffer->frameCount = 0;
+ buffer->raw = NULL;
+ }
+
+ void reset()
+ {
+ mNextFrame = 0;
+ }
+
+ size_t getNumFrames()
+ {
+ return mNumFrames;
+ }
+
+
+protected:
+ void* mAddr; // base address
+ size_t mNumFrames; // total frames
+ int mFrameSize; // frame size (# channels * bytes per sample)
+ size_t mNextFrame; // index of next frame to provide
+ size_t mUnrel; // number of frames not yet released
+ std::vector<int> mInputIncr; // number of frames provided per call
+ size_t mNextIdx; // index of next entry in mInputIncr to use
+};
+
+/* Creates a buffer filled with a sine wave.
+ */
+template<typename T>
+static void createSine(void *vbuffer, size_t frames,
+ size_t channels, double sampleRate, double freq)
+{
+ double tscale = 1. / sampleRate;
+ T* buffer = reinterpret_cast<T*>(vbuffer);
+ for (size_t i = 0; i < frames; ++i) {
+ double t = i * tscale;
+ double y = sin(2. * M_PI * freq * t);
+ T yt = convertValue<T>(y);
+
+ for (size_t j = 0; j < channels; ++j) {
+ buffer[i*channels + j] = yt / (j + 1);
+ }
+ }
+}
+
+/* Creates a buffer filled with a chirp signal (a sine wave sweep).
+ *
+ * When creating the Chirp, note that the frequency is the true sinusoidal
+ * frequency not the sampling rate.
+ *
+ * http://en.wikipedia.org/wiki/Chirp
+ */
+template<typename T>
+static void createChirp(void *vbuffer, size_t frames,
+ size_t channels, double sampleRate, double minfreq, double maxfreq)
+{
+ double tscale = 1. / sampleRate;
+ T *buffer = reinterpret_cast<T*>(vbuffer);
+ // note the chirp constant k has a divide-by-two.
+ double k = (maxfreq - minfreq) / (2. * tscale * frames);
+ for (size_t i = 0; i < frames; ++i) {
+ double t = i * tscale;
+ double y = sin(2. * M_PI * (k * t + minfreq) * t);
+ T yt = convertValue<T>(y);
+
+ for (size_t j = 0; j < channels; ++j) {
+ buffer[i*channels + j] = yt / (j + 1);
+ }
+ }
+}
+
+/* This derived class creates a buffer provider of datatype T,
+ * consisting of an input signal, e.g. from createChirp().
+ * The number of frames can be obtained from the base class
+ * TestProvider::getNumFrames().
+ */
+
+class SignalProvider : public TestProvider {
+public:
+ SignalProvider()
+ : mSampleRate(0),
+ mChannels(0)
+ {
+ }
+
+ virtual ~SignalProvider()
+ {
+ free(mAddr);
+ mAddr = NULL;
+ }
+
+ template <typename T>
+ void setChirp(size_t channels, double minfreq, double maxfreq, double sampleRate, double time)
+ {
+ createBufferByFrames<T>(channels, sampleRate, sampleRate*time);
+ createChirp<T>(mAddr, mNumFrames, mChannels, mSampleRate, minfreq, maxfreq);
+ }
+
+ template <typename T>
+ void setSine(size_t channels,
+ double freq, double sampleRate, double time)
+ {
+ createBufferByFrames<T>(channels, sampleRate, sampleRate*time);
+ createSine<T>(mAddr, mNumFrames, mChannels, mSampleRate, freq);
+ }
+
+ template <typename T>
+ void setFile(const char *file_in)
+ {
+ SF_INFO info;
+ info.format = 0;
+ SNDFILE *sf = sf_open(file_in, SFM_READ, &info);
+ if (sf == NULL) {
+ perror(file_in);
+ return;
+ }
+ createBufferByFrames<T>(info.channels, info.samplerate, info.frames);
+ if (is_same<T, float>::value) {
+ (void) sf_readf_float(sf, (float *) mAddr, mNumFrames);
+ } else if (is_same<T, short>::value) {
+ (void) sf_readf_short(sf, (short *) mAddr, mNumFrames);
+ }
+ sf_close(sf);
+ }
+
+ template <typename T>
+ void createBufferByFrames(size_t channels, uint32_t sampleRate, size_t frames)
+ {
+ mNumFrames = frames;
+ mChannels = channels;
+ mFrameSize = mChannels * sizeof(T);
+ free(mAddr);
+ mAddr = malloc(mFrameSize * mNumFrames);
+ mSampleRate = sampleRate;
+ }
+
+ uint32_t getSampleRate() const {
+ return mSampleRate;
+ }
+
+ uint32_t getNumChannels() const {
+ return mChannels;
+ }
+
+protected:
+ uint32_t mSampleRate;
+ uint32_t mChannels;
+};
+
+#endif // ANDROID_AUDIO_TEST_UTILS_H
diff --git a/services/audiopolicy/Android.mk b/services/audiopolicy/Android.mk
old mode 100644
new mode 100755
index cddc503..f3be42d
--- a/services/audiopolicy/Android.mk
+++ b/services/audiopolicy/Android.mk
@@ -3,7 +3,8 @@
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
- AudioPolicyService.cpp
+ AudioPolicyService.cpp \
+ AudioPolicyEffects.cpp
ifeq ($(USE_LEGACY_AUDIO_POLICY), 1)
LOCAL_SRC_FILES += \
diff --git a/services/audiopolicy/AudioPolicyEffects.cpp b/services/audiopolicy/AudioPolicyEffects.cpp
new file mode 100755
index 0000000..185e1cc
--- /dev/null
+++ b/services/audiopolicy/AudioPolicyEffects.cpp
@@ -0,0 +1,638 @@
+/*
+ * Copyright (C) 2014 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 "AudioPolicyEffects"
+#define LOG_NDEBUG 0
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <cutils/misc.h>
+#include <media/AudioEffect.h>
+#include <system/audio.h>
+#include <hardware/audio_effect.h>
+#include <audio_effects/audio_effects_conf.h>
+#include <utils/Vector.h>
+#include <utils/SortedVector.h>
+#include <cutils/config_utils.h>
+#include "AudioPolicyEffects.h"
+#include "ServiceUtilities.h"
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+// AudioPolicyEffects Implementation
+// ----------------------------------------------------------------------------
+
+AudioPolicyEffects::AudioPolicyEffects()
+{
+ // load automatic audio effect modules
+ if (access(AUDIO_EFFECT_VENDOR_CONFIG_FILE, R_OK) == 0) {
+ loadAudioEffectConfig(AUDIO_EFFECT_VENDOR_CONFIG_FILE);
+ } else if (access(AUDIO_EFFECT_DEFAULT_CONFIG_FILE, R_OK) == 0) {
+ loadAudioEffectConfig(AUDIO_EFFECT_DEFAULT_CONFIG_FILE);
+ }
+}
+
+
+AudioPolicyEffects::~AudioPolicyEffects()
+{
+ size_t i = 0;
+ // release audio input processing resources
+ for (i = 0; i < mInputSources.size(); i++) {
+ delete mInputSources.valueAt(i);
+ }
+ mInputSources.clear();
+
+ for (i = 0; i < mInputs.size(); i++) {
+ mInputs.valueAt(i)->mEffects.clear();
+ delete mInputs.valueAt(i);
+ }
+ mInputs.clear();
+
+ // release audio output processing resources
+ for (i = 0; i < mOutputStreams.size(); i++) {
+ delete mOutputStreams.valueAt(i);
+ }
+ mOutputStreams.clear();
+
+ for (i = 0; i < mOutputSessions.size(); i++) {
+ mOutputSessions.valueAt(i)->mEffects.clear();
+ delete mOutputSessions.valueAt(i);
+ }
+ mOutputSessions.clear();
+}
+
+
+status_t AudioPolicyEffects::addInputEffects(audio_io_handle_t input,
+ audio_source_t inputSource,
+ int audioSession)
+{
+ status_t status = NO_ERROR;
+
+ // create audio pre processors according to input source
+ audio_source_t aliasSource = (inputSource == AUDIO_SOURCE_HOTWORD) ?
+ AUDIO_SOURCE_VOICE_RECOGNITION : inputSource;
+
+ ssize_t index = mInputSources.indexOfKey(aliasSource);
+ if (index < 0) {
+ ALOGV("addInputEffects(): no processing needs to be attached to this source");
+ return status;
+ }
+ ssize_t idx = mInputs.indexOfKey(input);
+ EffectVector *inputDesc;
+ if (idx < 0) {
+ inputDesc = new EffectVector(audioSession);
+ mInputs.add(input, inputDesc);
+ } else {
+ inputDesc = mInputs.valueAt(idx);
+ }
+
+ Vector <EffectDesc *> effects = mInputSources.valueAt(index)->mEffects;
+ for (size_t i = 0; i < effects.size(); i++) {
+ EffectDesc *effect = effects[i];
+ sp<AudioEffect> fx = new AudioEffect(NULL, &effect->mUuid, -1, 0, 0, audioSession, input);
+ status_t status = fx->initCheck();
+ if (status != NO_ERROR && status != ALREADY_EXISTS) {
+ ALOGW("addInputEffects(): failed to create Fx %s on source %d",
+ effect->mName, (int32_t)aliasSource);
+ // fx goes out of scope and strong ref on AudioEffect is released
+ continue;
+ }
+ for (size_t j = 0; j < effect->mParams.size(); j++) {
+ fx->setParameter(effect->mParams[j]);
+ }
+ ALOGV("addInputEffects(): added Fx %s on source: %d", effect->mName, (int32_t)aliasSource);
+ inputDesc->mEffects.add(fx);
+ }
+ setProcessorEnabled(inputDesc, true);
+
+ return status;
+}
+
+
+status_t AudioPolicyEffects::releaseInputEffects(audio_io_handle_t input)
+{
+ status_t status = NO_ERROR;
+
+ ssize_t index = mInputs.indexOfKey(input);
+ if (index < 0) {
+ return status;
+ }
+ EffectVector *inputDesc = mInputs.valueAt(index);
+ setProcessorEnabled(inputDesc, false);
+ delete inputDesc;
+ mInputs.removeItemsAt(index);
+ ALOGV("releaseInputEffects(): all effects released");
+ return status;
+}
+
+status_t AudioPolicyEffects::queryDefaultInputEffects(int audioSession,
+ effect_descriptor_t *descriptors,
+ uint32_t *count)
+{
+ status_t status = NO_ERROR;
+
+ size_t index;
+ for (index = 0; index < mInputs.size(); index++) {
+ if (mInputs.valueAt(index)->mSessionId == audioSession) {
+ break;
+ }
+ }
+ if (index == mInputs.size()) {
+ *count = 0;
+ return BAD_VALUE;
+ }
+ Vector< sp<AudioEffect> > effects = mInputs.valueAt(index)->mEffects;
+
+ for (size_t i = 0; i < effects.size(); i++) {
+ effect_descriptor_t desc = effects[i]->descriptor();
+ if (i < *count) {
+ descriptors[i] = desc;
+ }
+ }
+ if (effects.size() > *count) {
+ status = NO_MEMORY;
+ }
+ *count = effects.size();
+ return status;
+}
+
+
+status_t AudioPolicyEffects::queryDefaultOutputSessionEffects(int audioSession,
+ effect_descriptor_t *descriptors,
+ uint32_t *count)
+{
+ status_t status = NO_ERROR;
+
+ size_t index;
+ for (index = 0; index < mOutputSessions.size(); index++) {
+ if (mOutputSessions.valueAt(index)->mSessionId == audioSession) {
+ break;
+ }
+ }
+ if (index == mOutputSessions.size()) {
+ *count = 0;
+ return BAD_VALUE;
+ }
+ Vector< sp<AudioEffect> > effects = mOutputSessions.valueAt(index)->mEffects;
+
+ for (size_t i = 0; i < effects.size(); i++) {
+ effect_descriptor_t desc = effects[i]->descriptor();
+ if (i < *count) {
+ descriptors[i] = desc;
+ }
+ }
+ if (effects.size() > *count) {
+ status = NO_MEMORY;
+ }
+ *count = effects.size();
+ return status;
+}
+
+
+status_t AudioPolicyEffects::addOutputSessionEffects(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ int audioSession)
+{
+ status_t status = NO_ERROR;
+
+ // create audio processors according to stream
+ ssize_t index = mOutputStreams.indexOfKey(stream);
+ if (index < 0) {
+ ALOGV("addOutputSessionEffects(): no output processing needed for this stream");
+ return NO_ERROR;
+ }
+
+ ssize_t idx = mOutputSessions.indexOfKey(audioSession);
+ EffectVector *procDesc;
+ if (idx < 0) {
+ procDesc = new EffectVector(audioSession);
+ mOutputSessions.add(audioSession, procDesc);
+ } else {
+ procDesc = mOutputSessions.valueAt(idx);
+ }
+
+ Vector <EffectDesc *> effects = mOutputStreams.valueAt(index)->mEffects;
+ for (size_t i = 0; i < effects.size(); i++) {
+ EffectDesc *effect = effects[i];
+ sp<AudioEffect> fx = new AudioEffect(NULL, &effect->mUuid, 0, 0, 0, audioSession, output);
+ status_t status = fx->initCheck();
+ if (status != NO_ERROR && status != ALREADY_EXISTS) {
+ ALOGE("addOutputSessionEffects(): failed to create Fx %s on session %d",
+ effect->mName, audioSession);
+ // fx goes out of scope and strong ref on AudioEffect is released
+ continue;
+ }
+ ALOGV("addOutputSessionEffects(): added Fx %s on session: %d for stream: %d",
+ effect->mName, audioSession, (int32_t)stream);
+ procDesc->mEffects.add(fx);
+ }
+
+ setProcessorEnabled(procDesc, true);
+
+ return status;
+}
+
+status_t AudioPolicyEffects::releaseOutputSessionEffects(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ int audioSession)
+{
+ status_t status = NO_ERROR;
+ (void) output; // argument not used for now
+ (void) stream; // argument not used for now
+
+ ssize_t index = mOutputSessions.indexOfKey(audioSession);
+ if (index < 0) {
+ ALOGV("releaseOutputSessionEffects: no output processing was attached to this stream");
+ return NO_ERROR;
+ }
+
+ EffectVector *procDesc = mOutputSessions.valueAt(index);
+ setProcessorEnabled(procDesc, false);
+ procDesc->mEffects.clear();
+ delete procDesc;
+ mOutputSessions.removeItemsAt(index);
+ ALOGV("releaseOutputSessionEffects(): output processing released from session: %d",
+ audioSession);
+ return status;
+}
+
+
+void AudioPolicyEffects::setProcessorEnabled(const EffectVector *effectVector, bool enabled)
+{
+ const Vector<sp<AudioEffect> > &fxVector = effectVector->mEffects;
+ for (size_t i = 0; i < fxVector.size(); i++) {
+ fxVector.itemAt(i)->setEnabled(enabled);
+ }
+}
+
+
+// ----------------------------------------------------------------------------
+// Audio processing configuration
+// ----------------------------------------------------------------------------
+
+/*static*/ const char * const AudioPolicyEffects::kInputSourceNames[AUDIO_SOURCE_CNT -1] = {
+ MIC_SRC_TAG,
+ VOICE_UL_SRC_TAG,
+ VOICE_DL_SRC_TAG,
+ VOICE_CALL_SRC_TAG,
+ CAMCORDER_SRC_TAG,
+ VOICE_REC_SRC_TAG,
+ VOICE_COMM_SRC_TAG
+};
+
+// returns the audio_source_t enum corresponding to the input source name or
+// AUDIO_SOURCE_CNT is no match found
+audio_source_t AudioPolicyEffects::inputSourceNameToEnum(const char *name)
+{
+ int i;
+ for (i = AUDIO_SOURCE_MIC; i < AUDIO_SOURCE_CNT; i++) {
+ if (strcmp(name, kInputSourceNames[i - AUDIO_SOURCE_MIC]) == 0) {
+ ALOGV("inputSourceNameToEnum found source %s %d", name, i);
+ break;
+ }
+ }
+ return (audio_source_t)i;
+}
+
+const char *AudioPolicyEffects::kStreamNames[AUDIO_STREAM_CNT+1] = {
+ AUDIO_STREAM_DEFAULT_TAG,
+ AUDIO_STREAM_VOICE_CALL_TAG,
+ AUDIO_STREAM_SYSTEM_TAG,
+ AUDIO_STREAM_RING_TAG,
+ AUDIO_STREAM_MUSIC_TAG,
+ AUDIO_STREAM_ALARM_TAG,
+ AUDIO_STREAM_NOTIFICATION_TAG,
+ AUDIO_STREAM_BLUETOOTH_SCO_TAG,
+ AUDIO_STREAM_ENFORCED_AUDIBLE_TAG,
+ AUDIO_STREAM_DTMF_TAG,
+ AUDIO_STREAM_TTS_TAG
+};
+
+// returns the audio_stream_t enum corresponding to the output stream name or
+// AUDIO_STREAM_CNT is no match found
+audio_stream_type_t AudioPolicyEffects::streamNameToEnum(const char *name)
+{
+ int i;
+ for (i = AUDIO_STREAM_DEFAULT; i < AUDIO_STREAM_CNT; i++) {
+ if (strcmp(name, kStreamNames[i - AUDIO_STREAM_DEFAULT]) == 0) {
+ ALOGV("streamNameToEnum found stream %s %d", name, i);
+ break;
+ }
+ }
+ return (audio_stream_type_t)i;
+}
+
+// ----------------------------------------------------------------------------
+// Audio Effect Config parser
+// ----------------------------------------------------------------------------
+
+size_t AudioPolicyEffects::growParamSize(char *param,
+ size_t size,
+ size_t *curSize,
+ size_t *totSize)
+{
+ // *curSize is at least sizeof(effect_param_t) + 2 * sizeof(int)
+ size_t pos = ((*curSize - 1 ) / size + 1) * size;
+
+ if (pos + size > *totSize) {
+ while (pos + size > *totSize) {
+ *totSize += ((*totSize + 7) / 8) * 4;
+ }
+ param = (char *)realloc(param, *totSize);
+ }
+ *curSize = pos + size;
+ return pos;
+}
+
+size_t AudioPolicyEffects::readParamValue(cnode *node,
+ char *param,
+ size_t *curSize,
+ size_t *totSize)
+{
+ if (strncmp(node->name, SHORT_TAG, sizeof(SHORT_TAG) + 1) == 0) {
+ size_t pos = growParamSize(param, sizeof(short), curSize, totSize);
+ *(short *)((char *)param + pos) = (short)atoi(node->value);
+ ALOGV("readParamValue() reading short %d", *(short *)((char *)param + pos));
+ return sizeof(short);
+ } else if (strncmp(node->name, INT_TAG, sizeof(INT_TAG) + 1) == 0) {
+ size_t pos = growParamSize(param, sizeof(int), curSize, totSize);
+ *(int *)((char *)param + pos) = atoi(node->value);
+ ALOGV("readParamValue() reading int %d", *(int *)((char *)param + pos));
+ return sizeof(int);
+ } else if (strncmp(node->name, FLOAT_TAG, sizeof(FLOAT_TAG) + 1) == 0) {
+ size_t pos = growParamSize(param, sizeof(float), curSize, totSize);
+ *(float *)((char *)param + pos) = (float)atof(node->value);
+ ALOGV("readParamValue() reading float %f",*(float *)((char *)param + pos));
+ return sizeof(float);
+ } else if (strncmp(node->name, BOOL_TAG, sizeof(BOOL_TAG) + 1) == 0) {
+ size_t pos = growParamSize(param, sizeof(bool), curSize, totSize);
+ if (strncmp(node->value, "false", strlen("false") + 1) == 0) {
+ *(bool *)((char *)param + pos) = false;
+ } else {
+ *(bool *)((char *)param + pos) = true;
+ }
+ ALOGV("readParamValue() reading bool %s",*(bool *)((char *)param + pos) ? "true" : "false");
+ return sizeof(bool);
+ } else if (strncmp(node->name, STRING_TAG, sizeof(STRING_TAG) + 1) == 0) {
+ size_t len = strnlen(node->value, EFFECT_STRING_LEN_MAX);
+ if (*curSize + len + 1 > *totSize) {
+ *totSize = *curSize + len + 1;
+ param = (char *)realloc(param, *totSize);
+ }
+ strncpy(param + *curSize, node->value, len);
+ *curSize += len;
+ param[*curSize] = '\0';
+ ALOGV("readParamValue() reading string %s", param + *curSize - len);
+ return len;
+ }
+ ALOGW("readParamValue() unknown param type %s", node->name);
+ return 0;
+}
+
+effect_param_t *AudioPolicyEffects::loadEffectParameter(cnode *root)
+{
+ cnode *param;
+ cnode *value;
+ size_t curSize = sizeof(effect_param_t);
+ size_t totSize = sizeof(effect_param_t) + 2 * sizeof(int);
+ effect_param_t *fx_param = (effect_param_t *)malloc(totSize);
+
+ param = config_find(root, PARAM_TAG);
+ value = config_find(root, VALUE_TAG);
+ if (param == NULL && value == NULL) {
+ // try to parse simple parameter form {int int}
+ param = root->first_child;
+ 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));
+ ALOGW("loadEffectParameter() ptr %p ptr2 %p", ptr, ptr2);
+ *ptr++ = atoi(param->name);
+ *ptr = atoi(param->value);
+ fx_param->psize = sizeof(int);
+ fx_param->vsize = sizeof(int);
+ return fx_param;
+ }
+ }
+ if (param == NULL || value == NULL) {
+ ALOGW("loadEffectParameter() invalid parameter description %s", root->name);
+ goto error;
+ }
+
+ fx_param->psize = 0;
+ param = param->first_child;
+ while (param) {
+ ALOGV("loadEffectParameter() reading param of type %s", param->name);
+ size_t size = readParamValue(param, (char *)fx_param, &curSize, &totSize);
+ if (size == 0) {
+ goto error;
+ }
+ fx_param->psize += size;
+ param = param->next;
+ }
+
+ // align start of value field on 32 bit boundary
+ curSize = ((curSize - 1 ) / sizeof(int) + 1) * sizeof(int);
+
+ fx_param->vsize = 0;
+ value = value->first_child;
+ while (value) {
+ ALOGV("loadEffectParameter() reading value of type %s", value->name);
+ size_t size = readParamValue(value, (char *)fx_param, &curSize, &totSize);
+ if (size == 0) {
+ goto error;
+ }
+ fx_param->vsize += size;
+ value = value->next;
+ }
+
+ return fx_param;
+
+error:
+ delete fx_param;
+ return NULL;
+}
+
+void AudioPolicyEffects::loadEffectParameters(cnode *root, Vector <effect_param_t *>& params)
+{
+ cnode *node = root->first_child;
+ while (node) {
+ ALOGV("loadEffectParameters() loading param %s", node->name);
+ effect_param_t *param = loadEffectParameter(node);
+ if (param == NULL) {
+ node = node->next;
+ continue;
+ }
+ params.add(param);
+ node = node->next;
+ }
+}
+
+
+AudioPolicyEffects::EffectDescVector *AudioPolicyEffects::loadEffectConfig(
+ cnode *root,
+ const Vector <EffectDesc *>& effects)
+{
+ cnode *node = root->first_child;
+ if (node == NULL) {
+ ALOGW("loadInputSource() empty element %s", root->name);
+ return NULL;
+ }
+ EffectDescVector *desc = new EffectDescVector();
+ while (node) {
+ size_t i;
+ for (i = 0; i < effects.size(); i++) {
+ if (strncmp(effects[i]->mName, node->name, EFFECT_STRING_LEN_MAX) == 0) {
+ ALOGV("loadEffectConfig() found effect %s in list", node->name);
+ break;
+ }
+ }
+ if (i == effects.size()) {
+ ALOGV("loadEffectConfig() effect %s not in list", node->name);
+ node = node->next;
+ continue;
+ }
+ EffectDesc *effect = new EffectDesc(*effects[i]); // deep copy
+ loadEffectParameters(node, effect->mParams);
+ ALOGV("loadEffectConfig() adding effect %s uuid %08x",
+ effect->mName, effect->mUuid.timeLow);
+ desc->mEffects.add(effect);
+ node = node->next;
+ }
+ if (desc->mEffects.size() == 0) {
+ ALOGW("loadEffectConfig() no valid effects found in config %s", root->name);
+ delete desc;
+ return NULL;
+ }
+ return desc;
+}
+
+status_t AudioPolicyEffects::loadInputEffectConfigurations(cnode *root,
+ const Vector <EffectDesc *>& effects)
+{
+ cnode *node = config_find(root, PREPROCESSING_TAG);
+ if (node == NULL) {
+ return -ENOENT;
+ }
+ node = node->first_child;
+ while (node) {
+ audio_source_t source = inputSourceNameToEnum(node->name);
+ if (source == AUDIO_SOURCE_CNT) {
+ ALOGW("loadInputSources() invalid input source %s", node->name);
+ node = node->next;
+ continue;
+ }
+ ALOGV("loadInputSources() loading input source %s", node->name);
+ EffectDescVector *desc = loadEffectConfig(node, effects);
+ if (desc == NULL) {
+ node = node->next;
+ continue;
+ }
+ mInputSources.add(source, desc);
+ node = node->next;
+ }
+ return NO_ERROR;
+}
+
+status_t AudioPolicyEffects::loadStreamEffectConfigurations(cnode *root,
+ const Vector <EffectDesc *>& effects)
+{
+ cnode *node = config_find(root, OUTPUT_SESSION_PROCESSING_TAG);
+ if (node == NULL) {
+ return -ENOENT;
+ }
+ node = node->first_child;
+ while (node) {
+ audio_stream_type_t stream = streamNameToEnum(node->name);
+ if (stream == AUDIO_STREAM_CNT) {
+ ALOGW("loadStreamEffectConfigurations() invalid output stream %s", node->name);
+ node = node->next;
+ continue;
+ }
+ ALOGV("loadStreamEffectConfigurations() loading output stream %s", node->name);
+ EffectDescVector *desc = loadEffectConfig(node, effects);
+ if (desc == NULL) {
+ node = node->next;
+ continue;
+ }
+ mOutputStreams.add(stream, desc);
+ node = node->next;
+ }
+ return NO_ERROR;
+}
+
+AudioPolicyEffects::EffectDesc *AudioPolicyEffects::loadEffect(cnode *root)
+{
+ cnode *node = config_find(root, UUID_TAG);
+ if (node == NULL) {
+ return NULL;
+ }
+ effect_uuid_t uuid;
+ if (AudioEffect::stringToGuid(node->value, &uuid) != NO_ERROR) {
+ ALOGW("loadEffect() invalid uuid %s", node->value);
+ return NULL;
+ }
+ return new EffectDesc(root->name, uuid);
+}
+
+status_t AudioPolicyEffects::loadEffects(cnode *root, Vector <EffectDesc *>& effects)
+{
+ cnode *node = config_find(root, EFFECTS_TAG);
+ if (node == NULL) {
+ return -ENOENT;
+ }
+ node = node->first_child;
+ while (node) {
+ ALOGV("loadEffects() loading effect %s", node->name);
+ EffectDesc *effect = loadEffect(node);
+ if (effect == NULL) {
+ node = node->next;
+ continue;
+ }
+ effects.add(effect);
+ node = node->next;
+ }
+ return NO_ERROR;
+}
+
+status_t AudioPolicyEffects::loadAudioEffectConfig(const char *path)
+{
+ cnode *root;
+ char *data;
+
+ data = (char *)load_file(path, NULL);
+ if (data == NULL) {
+ return -ENODEV;
+ }
+ root = config_node("", "");
+ config_load(root, data);
+
+ Vector <EffectDesc *> effects;
+ loadEffects(root, effects);
+ loadInputEffectConfigurations(root, effects);
+ loadStreamEffectConfigurations(root, effects);
+
+ config_free(root);
+ free(root);
+ free(data);
+
+ return NO_ERROR;
+}
+
+
+}; // namespace android
diff --git a/services/audiopolicy/AudioPolicyEffects.h b/services/audiopolicy/AudioPolicyEffects.h
new file mode 100755
index 0000000..351cb1a
--- /dev/null
+++ b/services/audiopolicy/AudioPolicyEffects.h
@@ -0,0 +1,187 @@
+/*
+ * Copyright (C) 2014 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_AUDIOPOLICYEFFECTS_H
+#define ANDROID_AUDIOPOLICYEFFECTS_H
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <cutils/misc.h>
+#include <media/AudioEffect.h>
+#include <system/audio.h>
+#include <hardware/audio_effect.h>
+#include <utils/Vector.h>
+#include <utils/SortedVector.h>
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+// AudioPolicyEffects class
+// This class will manage all effects attached to input and output streams in
+// AudioPolicyService as configured in audio_effects.conf.
+class AudioPolicyEffects : public RefBase
+{
+
+public:
+
+ // The constructor will parse audio_effects.conf
+ // First it will look whether vendor specific file exists,
+ // otherwise it will parse the system default file.
+ AudioPolicyEffects();
+ virtual ~AudioPolicyEffects();
+
+ // Return a list of effect descriptors for default input effects
+ // associated with audioSession
+ status_t queryDefaultInputEffects(int audioSession,
+ effect_descriptor_t *descriptors,
+ uint32_t *count);
+
+ // Add all input effects associated with this input
+ // Effects are attached depending on the audio_source_t
+ status_t addInputEffects(audio_io_handle_t input,
+ audio_source_t inputSource,
+ int audioSession);
+
+ // Add all input effects associated to this input
+ status_t releaseInputEffects(audio_io_handle_t input);
+
+
+ // Return a list of effect descriptors for default output effects
+ // associated with audioSession
+ status_t queryDefaultOutputSessionEffects(int audioSession,
+ effect_descriptor_t *descriptors,
+ uint32_t *count);
+
+ // Add all output effects associated to this output
+ // Effects are attached depending on the audio_stream_type_t
+ status_t addOutputSessionEffects(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ int audioSession);
+
+ // release all output effects associated with this output stream and audiosession
+ status_t releaseOutputSessionEffects(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ int audioSession);
+
+private:
+
+ // class to store the description of an effects and its parameters
+ // as defined in audio_effects.conf
+ class EffectDesc {
+ public:
+ 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;
+ };
+
+ // class to store voctor of EffectDesc
+ class EffectDescVector {
+ public:
+ EffectDescVector() {}
+ /*virtual*/ ~EffectDescVector() {
+ for (size_t j = 0; j < mEffects.size(); j++) {
+ delete mEffects[j];
+ }
+ }
+ Vector <EffectDesc *> mEffects;
+ };
+
+ // class to store voctor of AudioEffects
+ class EffectVector {
+ public:
+ EffectVector(int session) : mSessionId(session) {}
+ /*virtual*/ ~EffectVector() {}
+ const int mSessionId;
+ Vector< sp<AudioEffect> >mEffects;
+ };
+
+
+ static const char * const kInputSourceNames[AUDIO_SOURCE_CNT -1];
+ audio_source_t inputSourceNameToEnum(const char *name);
+
+ static const char *kStreamNames[AUDIO_STREAM_CNT+1]; //+1 required as streams start from -1
+ audio_stream_type_t streamNameToEnum(const char *name);
+
+ // Enable or disable all effects in effect vector
+ void setProcessorEnabled(const EffectVector *effectVector, bool enabled);
+
+ // Parse audio_effects.conf
+ status_t loadAudioEffectConfig(const char *path);
+
+ // Load all effects descriptors in configuration file
+ status_t loadEffects(cnode *root, Vector <EffectDesc *>& effects);
+ EffectDesc *loadEffect(cnode *root);
+
+ // Load all automatic effect configurations
+ status_t loadInputEffectConfigurations(cnode *root, const Vector <EffectDesc *>& effects);
+ status_t loadStreamEffectConfigurations(cnode *root, const Vector <EffectDesc *>& effects);
+ EffectDescVector *loadEffectConfig(cnode *root, const Vector <EffectDesc *>& effects);
+
+ // Load all automatic effect parameters
+ void loadEffectParameters(cnode *root, Vector <effect_param_t *>& params);
+ effect_param_t *loadEffectParameter(cnode *root);
+ size_t readParamValue(cnode *node,
+ char *param,
+ size_t *curSize,
+ size_t *totSize);
+ size_t growParamSize(char *param,
+ size_t size,
+ size_t *curSize,
+ size_t *totSize);
+
+ // Automatic input effects are configured per audio_source_t
+ KeyedVector< audio_source_t, EffectDescVector* > mInputSources;
+ // Automatic input effects are unique for audio_io_handle_t
+ KeyedVector< audio_io_handle_t, EffectVector* > mInputs;
+
+ // Automatic output effects are organized per audio_stream_type_t
+ KeyedVector< audio_stream_type_t, EffectDescVector* > mOutputStreams;
+ // Automatic output effects are unique for audiosession ID
+ KeyedVector< int32_t, EffectVector* > mOutputSessions;
+};
+
+}; // namespace android
+
+#endif // ANDROID_AUDIOPOLICYEFFECTS_H
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index c025a45..33e4397 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -90,6 +90,12 @@
audio_channel_mask_t channelMask,
audio_output_flags_t flags,
const audio_offload_info_t *offloadInfo) = 0;
+ virtual audio_io_handle_t getOutputForAttr(const audio_attributes_t *attr,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo) = 0;
// indicates to the audio policy manager that the output starts being used by corresponding stream.
virtual status_t startOutput(audio_io_handle_t output,
audio_stream_type_t stream,
diff --git a/services/audiopolicy/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/AudioPolicyInterfaceImpl.cpp
old mode 100644
new mode 100755
index 8cc386a..5a13ac2
--- a/services/audiopolicy/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/AudioPolicyInterfaceImpl.cpp
@@ -131,6 +131,22 @@
format, channelMask, flags, offloadInfo);
}
+audio_io_handle_t AudioPolicyService::getOutputForAttr(const audio_attributes_t *attr,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo)
+{
+ if (mAudioPolicyManager == NULL) {
+ return 0;
+ }
+ ALOGV("getOutput()");
+ Mutex::Autolock _l(mLock);
+ return mAudioPolicyManager->getOutputForAttr(attr, samplingRate,
+ format, channelMask, flags, offloadInfo);
+}
+
status_t AudioPolicyService::startOutput(audio_io_handle_t output,
audio_stream_type_t stream,
int session)
@@ -140,6 +156,13 @@
}
ALOGV("startOutput()");
Mutex::Autolock _l(mLock);
+
+ // create audio processors according to stream
+ status_t status = mAudioPolicyEffects->addOutputSessionEffects(output, stream, session);
+ if (status != NO_ERROR && status != ALREADY_EXISTS) {
+ ALOGW("Failed to add effects on session %d", session);
+ }
+
return mAudioPolicyManager->startOutput(output, stream, session);
}
@@ -161,6 +184,13 @@
{
ALOGV("doStopOutput from tid %d", gettid());
Mutex::Autolock _l(mLock);
+
+ // release audio processors from the stream
+ status_t status = mAudioPolicyEffects->releaseOutputSessionEffects(output, stream, session);
+ if (status != NO_ERROR && status != ALREADY_EXISTS) {
+ ALOGW("Failed to release effects on session %d", session);
+ }
+
return mAudioPolicyManager->stopOutput(output, stream, session);
}
@@ -206,39 +236,13 @@
if (input == 0) {
return input;
}
+
// create audio pre processors according to input source
- audio_source_t aliasSource = (inputSource == AUDIO_SOURCE_HOTWORD) ?
- AUDIO_SOURCE_VOICE_RECOGNITION : inputSource;
-
- ssize_t index = mInputSources.indexOfKey(aliasSource);
- if (index < 0) {
- return input;
- }
- ssize_t idx = mInputs.indexOfKey(input);
- InputDesc *inputDesc;
- if (idx < 0) {
- inputDesc = new InputDesc(audioSession);
- mInputs.add(input, inputDesc);
- } else {
- inputDesc = mInputs.valueAt(idx);
+ status_t status = mAudioPolicyEffects->addInputEffects(input, inputSource, audioSession);
+ if (status != NO_ERROR && status != ALREADY_EXISTS) {
+ ALOGW("Failed to add effects on input %d", input);
}
- Vector <EffectDesc *> effects = mInputSources.valueAt(index)->mEffects;
- for (size_t i = 0; i < effects.size(); i++) {
- EffectDesc *effect = effects[i];
- sp<AudioEffect> fx = new AudioEffect(NULL, &effect->mUuid, -1, 0, 0, audioSession, input);
- status_t status = fx->initCheck();
- if (status != NO_ERROR && status != ALREADY_EXISTS) {
- ALOGW("Failed to create Fx %s on input %d", effect->mName, input);
- // fx goes out of scope and strong ref on AudioEffect is released
- continue;
- }
- for (size_t j = 0; j < effect->mParams.size(); j++) {
- fx->setParameter(effect->mParams[j]);
- }
- inputDesc->mEffects.add(fx);
- }
- setPreProcessorEnabled(inputDesc, true);
return input;
}
@@ -270,14 +274,11 @@
Mutex::Autolock _l(mLock);
mAudioPolicyManager->releaseInput(input);
- ssize_t index = mInputs.indexOfKey(input);
- if (index < 0) {
- return;
+ // release audio processors from the input
+ status_t status = mAudioPolicyEffects->releaseInputEffects(input);
+ if(status != NO_ERROR) {
+ ALOGW("Failed to release effects on input %d", input);
}
- InputDesc *inputDesc = mInputs.valueAt(index);
- setPreProcessorEnabled(inputDesc, false);
- delete inputDesc;
- mInputs.removeItemsAt(index);
}
status_t AudioPolicyService::initStreamVolume(audio_stream_type_t stream,
@@ -420,37 +421,13 @@
effect_descriptor_t *descriptors,
uint32_t *count)
{
-
if (mAudioPolicyManager == NULL) {
*count = 0;
return NO_INIT;
}
Mutex::Autolock _l(mLock);
- status_t status = NO_ERROR;
- size_t index;
- for (index = 0; index < mInputs.size(); index++) {
- if (mInputs.valueAt(index)->mSessionId == audioSession) {
- break;
- }
- }
- if (index == mInputs.size()) {
- *count = 0;
- return BAD_VALUE;
- }
- Vector< sp<AudioEffect> > effects = mInputs.valueAt(index)->mEffects;
-
- for (size_t i = 0; i < effects.size(); i++) {
- effect_descriptor_t desc = effects[i]->descriptor();
- if (i < *count) {
- descriptors[i] = desc;
- }
- }
- if (effects.size() > *count) {
- status = NO_MEMORY;
- }
- *count = effects.size();
- return status;
+ return mAudioPolicyEffects->queryDefaultInputEffects(audioSession, descriptors, count);
}
bool AudioPolicyService::isOffloadSupported(const audio_offload_info_t& info)
diff --git a/services/audiopolicy/AudioPolicyInterfaceImplLegacy.cpp b/services/audiopolicy/AudioPolicyInterfaceImplLegacy.cpp
old mode 100644
new mode 100755
index 0bf4982..406988c
--- a/services/audiopolicy/AudioPolicyInterfaceImplLegacy.cpp
+++ b/services/audiopolicy/AudioPolicyInterfaceImplLegacy.cpp
@@ -144,6 +144,13 @@
}
ALOGV("startOutput()");
Mutex::Autolock _l(mLock);
+
+ // create audio processors according to stream
+ status_t status = mAudioPolicyEffects->addOutputSessionEffects(output, stream, session);
+ if (status != NO_ERROR && status != ALREADY_EXISTS) {
+ ALOGW("Failed to add effects on session %d", session);
+ }
+
return mpAudioPolicy->start_output(mpAudioPolicy, output, stream, session);
}
@@ -165,6 +172,13 @@
{
ALOGV("doStopOutput from tid %d", gettid());
Mutex::Autolock _l(mLock);
+
+ // release audio processors from the stream
+ status_t status = mAudioPolicyEffects->releaseOutputSessionEffects(output, stream, session);
+ if (status != NO_ERROR && status != ALREADY_EXISTS) {
+ ALOGW("Failed to release effects on session %d", session);
+ }
+
return mpAudioPolicy->stop_output(mpAudioPolicy, output, stream, session);
}
@@ -210,39 +224,13 @@
if (input == 0) {
return input;
}
+
// create audio pre processors according to input source
- audio_source_t aliasSource = (inputSource == AUDIO_SOURCE_HOTWORD) ?
- AUDIO_SOURCE_VOICE_RECOGNITION : inputSource;
-
- ssize_t index = mInputSources.indexOfKey(aliasSource);
- if (index < 0) {
- return input;
- }
- ssize_t idx = mInputs.indexOfKey(input);
- InputDesc *inputDesc;
- if (idx < 0) {
- inputDesc = new InputDesc(audioSession);
- mInputs.add(input, inputDesc);
- } else {
- inputDesc = mInputs.valueAt(idx);
+ status_t status = mAudioPolicyEffects->addInputEffects(input, inputSource, audioSession);
+ if (status != NO_ERROR && status != ALREADY_EXISTS) {
+ ALOGW("Failed to add effects on input %d", input);
}
- Vector <EffectDesc *> effects = mInputSources.valueAt(index)->mEffects;
- for (size_t i = 0; i < effects.size(); i++) {
- EffectDesc *effect = effects[i];
- sp<AudioEffect> fx = new AudioEffect(NULL, &effect->mUuid, -1, 0, 0, audioSession, input);
- status_t status = fx->initCheck();
- if (status != NO_ERROR && status != ALREADY_EXISTS) {
- ALOGW("Failed to create Fx %s on input %d", effect->mName, input);
- // fx goes out of scope and strong ref on AudioEffect is released
- continue;
- }
- for (size_t j = 0; j < effect->mParams.size(); j++) {
- fx->setParameter(effect->mParams[j]);
- }
- inputDesc->mEffects.add(fx);
- }
- setPreProcessorEnabled(inputDesc, true);
return input;
}
@@ -274,14 +262,11 @@
Mutex::Autolock _l(mLock);
mpAudioPolicy->release_input(mpAudioPolicy, input);
- ssize_t index = mInputs.indexOfKey(input);
- if (index < 0) {
- return;
+ // release audio processors from the input
+ status_t status = mAudioPolicyEffects->releaseInputEffects(input);
+ if(status != NO_ERROR) {
+ ALOGW("Failed to release effects on input %d", input);
}
- InputDesc *inputDesc = mInputs.valueAt(index);
- setPreProcessorEnabled(inputDesc, false);
- delete inputDesc;
- mInputs.removeItemsAt(index);
}
status_t AudioPolicyService::initStreamVolume(audio_stream_type_t stream,
@@ -437,37 +422,13 @@
effect_descriptor_t *descriptors,
uint32_t *count)
{
-
if (mpAudioPolicy == NULL) {
*count = 0;
return NO_INIT;
}
Mutex::Autolock _l(mLock);
- status_t status = NO_ERROR;
- size_t index;
- for (index = 0; index < mInputs.size(); index++) {
- if (mInputs.valueAt(index)->mSessionId == audioSession) {
- break;
- }
- }
- if (index == mInputs.size()) {
- *count = 0;
- return BAD_VALUE;
- }
- Vector< sp<AudioEffect> > effects = mInputs.valueAt(index)->mEffects;
-
- for (size_t i = 0; i < effects.size(); i++) {
- effect_descriptor_t desc = effects[i]->descriptor();
- if (i < *count) {
- descriptors[i] = desc;
- }
- }
- if (effects.size() > *count) {
- status = NO_MEMORY;
- }
- *count = effects.size();
- return status;
+ return mAudioPolicyEffects->queryDefaultInputEffects(audioSession, descriptors, count);
}
bool AudioPolicyService::isOffloadSupported(const audio_offload_info_t& info)
diff --git a/services/audiopolicy/AudioPolicyManager.cpp b/services/audiopolicy/AudioPolicyManager.cpp
index 9708d0f..98f367e 100644
--- a/services/audiopolicy/AudioPolicyManager.cpp
+++ b/services/audiopolicy/AudioPolicyManager.cpp
@@ -123,6 +123,11 @@
STRING_TO_ENUM(AUDIO_FORMAT_MP3),
STRING_TO_ENUM(AUDIO_FORMAT_AAC),
STRING_TO_ENUM(AUDIO_FORMAT_VORBIS),
+ STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V1),
+ STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V2),
+ STRING_TO_ENUM(AUDIO_FORMAT_OPUS),
+ STRING_TO_ENUM(AUDIO_FORMAT_AC3),
+ STRING_TO_ENUM(AUDIO_FORMAT_E_AC3),
};
const StringToEnum sOutChannelsNameToEnumTable[] = {
@@ -512,7 +517,10 @@
config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY &&
config != AUDIO_POLICY_FORCE_ANALOG_DOCK &&
config != AUDIO_POLICY_FORCE_DIGITAL_DOCK && config != AUDIO_POLICY_FORCE_NONE &&
- config != AUDIO_POLICY_FORCE_NO_BT_A2DP) {
+ config != AUDIO_POLICY_FORCE_NO_BT_A2DP &&
+ config != AUDIO_POLICY_FORCE_SYSTEM_AUDIO_HDMI_ARC &&
+ config != AUDIO_POLICY_FORCE_SYSTEM_AUDIO_SPDIF &&
+ config != AUDIO_POLICY_FORCE_SYSTEM_AUDIO_LINE) {
ALOGW("setForceUse() invalid config %d for FOR_MEDIA", config);
return;
}
@@ -624,13 +632,53 @@
audio_output_flags_t flags,
const audio_offload_info_t *offloadInfo)
{
- audio_io_handle_t output = 0;
- uint32_t latency = 0;
+
routing_strategy strategy = getStrategy(stream);
audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
ALOGV("getOutput() device %d, stream %d, samplingRate %d, format %x, channelMask %x, flags %x",
device, stream, samplingRate, format, channelMask, flags);
+ return getOutputForDevice(device, stream, samplingRate,format, channelMask, flags,
+ offloadInfo);
+}
+
+audio_io_handle_t AudioPolicyManager::getOutputForAttr(const audio_attributes_t *attr,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo)
+{
+ if (attr == NULL) {
+ ALOGE("getOutputForAttr() called with NULL audio attributes");
+ return 0;
+ }
+ ALOGV("getOutputForAttr() usage=%d, content=%d, tag=%s",
+ attr->usage, attr->content_type, attr->tags);
+
+ // TODO this is where filtering for custom policies (rerouting, dynamic sources) will go
+ routing_strategy strategy = (routing_strategy) getStrategyForAttr(attr);
+ audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
+ ALOGV("getOutputForAttr() device %d, samplingRate %d, format %x, channelMask %x, flags %x",
+ device, samplingRate, format, channelMask, flags);
+
+ audio_stream_type_t stream = streamTypefromAttributesInt(attr);
+ return getOutputForDevice(device, stream, samplingRate, format, channelMask, flags,
+ offloadInfo);
+}
+
+audio_io_handle_t AudioPolicyManager::getOutputForDevice(
+ audio_devices_t device,
+ audio_stream_type_t stream,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo)
+{
+ audio_io_handle_t output = 0;
+ uint32_t latency = 0;
+
#ifdef AUDIO_POLICY_TEST
if (mCurOutput != 0) {
ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d",
@@ -1516,7 +1564,7 @@
snprintf(buffer, SIZE,
" Stream Can be muted Index Min Index Max Index Cur [device : index]...\n");
write(fd, buffer, strlen(buffer));
- for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
+ for (size_t i = 0; i < AUDIO_STREAM_CNT; i++) {
snprintf(buffer, SIZE, " %02zu ", i);
write(fd, buffer, strlen(buffer));
mStreams[i].dump(fd);
@@ -1647,14 +1695,20 @@
*num_ports += mInputs.size();
}
if (role == AUDIO_PORT_ROLE_SOURCE || role == AUDIO_PORT_ROLE_NONE) {
- for (size_t i = 0; i < mOutputs.size() && portsWritten < portsMax; i++) {
- mOutputs[i]->toAudioPort(&ports[portsWritten++]);
+ size_t numOutputs = 0;
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ if (!mOutputs[i]->isDuplicated()) {
+ numOutputs++;
+ if (portsWritten < portsMax) {
+ mOutputs[i]->toAudioPort(&ports[portsWritten++]);
+ }
+ }
}
- *num_ports += mOutputs.size();
+ *num_ports += numOutputs;
}
}
*generation = curAudioPortGeneration();
- ALOGV("listAudioPorts() got %d ports needed %d", portsWritten, *num_ports);
+ ALOGV("listAudioPorts() got %zu ports needed %d", portsWritten, *num_ports);
return NO_ERROR;
}
@@ -1782,6 +1836,8 @@
ALOGV("createAudioPatch() output not found for id %d", patch->sources[0].id);
return BAD_VALUE;
}
+ ALOG_ASSERT(!outputDesc->isDuplicated(),"duplicated output %d in source in ports",
+ outputDesc->mIoHandle);
if (patchDesc != 0) {
if (patchDesc->mPatch.sources[0].id != patch->sources[0].id) {
ALOGV("createAudioPatch() source id differs for patch current id %d new id %d",
@@ -1796,7 +1852,7 @@
return BAD_VALUE;
}
- if (!outputDesc->mProfile->isCompatibleProfile(devDesc->mType,
+ if (!outputDesc->mProfile->isCompatibleProfile(devDesc->mDeviceType,
patch->sources[0].sample_rate,
patch->sources[0].format,
patch->sources[0].channel_mask,
@@ -1805,9 +1861,9 @@
}
// TODO: reconfigure output format and channels here
ALOGV("createAudioPatch() setting device %08x on output %d",
- devDesc->mType, outputDesc->mIoHandle);
+ devDesc->mDeviceType, outputDesc->mIoHandle);
setOutputDevice(outputDesc->mIoHandle,
- devDesc->mType,
+ devDesc->mDeviceType,
true,
0,
handle);
@@ -1841,7 +1897,7 @@
return BAD_VALUE;
}
- if (!inputDesc->mProfile->isCompatibleProfile(devDesc->mType,
+ if (!inputDesc->mProfile->isCompatibleProfile(devDesc->mDeviceType,
patch->sinks[0].sample_rate,
patch->sinks[0].format,
patch->sinks[0].channel_mask,
@@ -1850,9 +1906,9 @@
}
// TODO: reconfigure output format and channels here
ALOGV("createAudioPatch() setting device %08x on output %d",
- devDesc->mType, inputDesc->mIoHandle);
+ devDesc->mDeviceType, inputDesc->mIoHandle);
setInputDevice(inputDesc->mIoHandle,
- devDesc->mType,
+ devDesc->mDeviceType,
true,
handle);
index = mAudioPatches.indexOfKey(*handle);
@@ -1998,7 +2054,7 @@
generation == NULL) {
return BAD_VALUE;
}
- ALOGV("listAudioPatches() num_patches %d patches %p available patches %d",
+ ALOGV("listAudioPatches() num_patches %d patches %p available patches %zu",
*num_patches, patches, mAudioPatches.size());
if (patches == NULL) {
*num_patches = 0;
@@ -2010,13 +2066,13 @@
i < mAudioPatches.size() && patchesWritten < patchesMax; i++) {
patches[patchesWritten] = mAudioPatches[i]->mPatch;
patches[patchesWritten++].id = mAudioPatches[i]->mHandle;
- ALOGV("listAudioPatches() patch %d num_sources %d num_sinks %d",
+ ALOGV("listAudioPatches() patch %zu num_sources %d num_sinks %d",
i, mAudioPatches[i]->mPatch.num_sources, mAudioPatches[i]->mPatch.num_sinks);
}
*num_patches = mAudioPatches.size();
*generation = curAudioPortGeneration();
- ALOGV("listAudioPatches() got %d patches needed %d", patchesWritten, *num_patches);
+ ALOGV("listAudioPatches() got %zu patches needed %d", patchesWritten, *num_patches);
return NO_ERROR;
}
@@ -2040,6 +2096,9 @@
if (outputDesc == NULL) {
return BAD_VALUE;
}
+ ALOG_ASSERT(!outputDesc->isDuplicated(),
+ "setAudioPortConfig() called on duplicated output %d",
+ outputDesc->mIoHandle);
audioPortConfig = outputDesc;
} else if (config->role == AUDIO_PORT_ROLE_SINK) {
sp<AudioInputDescriptor> inputDesc = getInputFromId(config->id);
@@ -2780,7 +2839,7 @@
{
if (mHwModules[module_idx]->mInputProfiles[profile_index]->mSupportedDevices.types()
& (device & ~AUDIO_DEVICE_BIT_IN)) {
- ALOGV("checkInputsForDevice(): adding profile %d from module %d",
+ ALOGV("checkInputsForDevice(): adding profile %zu from module %zu",
profile_index, module_idx);
profiles.add(mHwModules[module_idx]->mInputProfiles[profile_index]);
}
@@ -2904,7 +2963,7 @@
profile_index++) {
sp<IOProfile> profile = mHwModules[module_index]->mInputProfiles[profile_index];
if (profile->mSupportedDevices.types() & device) {
- ALOGV("checkInputsForDevice(): clearing direct input profile %d on module %d",
+ ALOGV("checkInputsForDevice(): clearing direct input profile %zu on module %zu",
profile_index, module_index);
if (profile->mSamplingRates[0] == 0) {
profile->mSamplingRates.clear();
@@ -3240,6 +3299,44 @@
}
}
+uint32_t AudioPolicyManager::getStrategyForAttr(const audio_attributes_t *attr) {
+ // flags to strategy mapping
+ if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) {
+ return (uint32_t) STRATEGY_ENFORCED_AUDIBLE;
+ }
+
+ // usage to strategy mapping
+ switch (attr->usage) {
+ case AUDIO_USAGE_MEDIA:
+ case AUDIO_USAGE_GAME:
+ case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
+ case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
+ case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
+ return (uint32_t) STRATEGY_MEDIA;
+
+ case AUDIO_USAGE_VOICE_COMMUNICATION:
+ return (uint32_t) STRATEGY_PHONE;
+
+ case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
+ return (uint32_t) STRATEGY_DTMF;
+
+ case AUDIO_USAGE_ALARM:
+ case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
+ return (uint32_t) STRATEGY_SONIFICATION;
+
+ case AUDIO_USAGE_NOTIFICATION:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
+ case AUDIO_USAGE_NOTIFICATION_EVENT:
+ return (uint32_t) STRATEGY_SONIFICATION_RESPECTFUL;
+
+ case AUDIO_USAGE_UNKNOWN:
+ default:
+ return (uint32_t) STRATEGY_MEDIA;
+ }
+}
+
void AudioPolicyManager::handleNotificationRoutingForStream(audio_stream_type_t stream) {
switch(stream) {
case AUDIO_STREAM_MUSIC:
@@ -3443,10 +3540,32 @@
if (device2 == AUDIO_DEVICE_NONE) {
device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER;
}
+ int device3 = AUDIO_DEVICE_NONE;
+ if (strategy == STRATEGY_MEDIA) {
+ // Hdmi system audio should use manually configured device type.
+ if (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA]
+ == AUDIO_POLICY_FORCE_SYSTEM_AUDIO_HDMI_ARC) {
+ device3 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_HDMI_ARC;
+ } else if (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA]
+ == AUDIO_POLICY_FORCE_SYSTEM_AUDIO_SPDIF) {
+ device3 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPDIF;
+ } else if (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA]
+ == AUDIO_POLICY_FORCE_SYSTEM_AUDIO_LINE) {
+ device3 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_LINE;
+ }
+ }
+ // Merge hdmi cec system audio and existing device for media. If system audio is on,
+ // internal speaker will be muted but others are not.
+ device2 |= device3;
// device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or
// STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise
device |= device2;
+
+ // If system audio mode is on and proper audio out is set, remove speaker from device.
+ if (device3 != AUDIO_DEVICE_NONE) {
+ device &= ~AUDIO_DEVICE_OUT_SPEAKER;
+ }
if (device) break;
device = mDefaultOutputDevice->mDeviceType;
if (device == AUDIO_DEVICE_NONE) {
@@ -4125,9 +4244,7 @@
if (stream == AUDIO_STREAM_MUSIC &&
index != mStreams[stream].mIndexMin &&
(device == AUDIO_DEVICE_OUT_AUX_DIGITAL ||
- device == AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET ||
- device == AUDIO_DEVICE_OUT_USB_ACCESSORY ||
- device == AUDIO_DEVICE_OUT_USB_DEVICE)) {
+ device == AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET)) {
return 1.0;
}
@@ -4501,10 +4618,12 @@
struct audio_port_config *dstConfig,
const struct audio_port_config *srcConfig) const
{
+ ALOG_ASSERT(!isDuplicated(), "toAudioPortConfig() called on duplicated output %d", mIoHandle);
+
dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
if (srcConfig != NULL) {
- dstConfig->config_mask &= srcConfig->config_mask;
+ dstConfig->config_mask |= srcConfig->config_mask;
}
AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
@@ -4519,6 +4638,7 @@
void AudioPolicyManager::AudioOutputDescriptor::toAudioPort(
struct audio_port *port) const
{
+ ALOG_ASSERT(!isDuplicated(), "toAudioPort() called on duplicated output %d", mIoHandle);
mProfile->toAudioPort(port);
port->id = mId;
toAudioPortConfig(&port->active_config);
@@ -4584,10 +4704,12 @@
struct audio_port_config *dstConfig,
const struct audio_port_config *srcConfig) const
{
+ ALOG_ASSERT(mProfile != 0,
+ "toAudioPortConfig() called on input with null profile %d", mIoHandle);
dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
if (srcConfig != NULL) {
- dstConfig->config_mask &= srcConfig->config_mask;
+ dstConfig->config_mask |= srcConfig->config_mask;
}
AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
@@ -4603,6 +4725,8 @@
void AudioPolicyManager::AudioInputDescriptor::toAudioPort(
struct audio_port *port) const
{
+ ALOG_ASSERT(mProfile != 0, "toAudioPort() called on input with null profile %d", mIoHandle);
+
mProfile->toAudioPort(port);
port->id = mId;
toAudioPortConfig(&port->active_config);
@@ -4912,7 +5036,7 @@
}
port->num_formats = i;
- ALOGV("AudioPort::toAudioPort() num gains %d", mGains.size());
+ ALOGV("AudioPort::toAudioPort() num gains %zu", mGains.size());
for (i = 0; i < mGains.size() && i < AUDIO_PORT_MAX_GAINS; i++) {
port->gains[i] = mGains[i]->mGain;
@@ -5610,7 +5734,7 @@
{
sp<DeviceDescriptor> device;
for (size_t i = 0; i < size(); i++) {
- ALOGV("DeviceVector::getDeviceFromId(%d) itemAt(%d)->mId %d", id, i, itemAt(i)->mId);
+ ALOGV("DeviceVector::getDeviceFromId(%d) itemAt(%zu)->mId %d", id, i, itemAt(i)->mId);
if (itemAt(i)->mId == id) {
device = itemAt(i);
break;
@@ -5653,7 +5777,7 @@
{
dstConfig->config_mask = AUDIO_PORT_CONFIG_CHANNEL_MASK|AUDIO_PORT_CONFIG_GAIN;
if (srcConfig != NULL) {
- dstConfig->config_mask &= srcConfig->config_mask;
+ dstConfig->config_mask |= srcConfig->config_mask;
}
AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
@@ -5917,4 +6041,46 @@
mHwModules.add(module);
}
+audio_stream_type_t AudioPolicyManager::streamTypefromAttributesInt(const audio_attributes_t *attr)
+{
+ // flags to stream type mapping
+ if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) {
+ return AUDIO_STREAM_ENFORCED_AUDIBLE;
+ }
+ if ((attr->flags & AUDIO_FLAG_SCO) == AUDIO_FLAG_SCO) {
+ return AUDIO_STREAM_BLUETOOTH_SCO;
+ }
+
+ // usage to stream type mapping
+ switch (attr->usage) {
+ case AUDIO_USAGE_MEDIA:
+ case AUDIO_USAGE_GAME:
+ case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
+ case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
+ return AUDIO_STREAM_MUSIC;
+ case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
+ return AUDIO_STREAM_SYSTEM;
+ case AUDIO_USAGE_VOICE_COMMUNICATION:
+ return AUDIO_STREAM_VOICE_CALL;
+
+ case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
+ return AUDIO_STREAM_DTMF;
+
+ case AUDIO_USAGE_ALARM:
+ return AUDIO_STREAM_ALARM;
+ case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
+ return AUDIO_STREAM_RING;
+
+ case AUDIO_USAGE_NOTIFICATION:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
+ case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
+ case AUDIO_USAGE_NOTIFICATION_EVENT:
+ return AUDIO_STREAM_NOTIFICATION;
+
+ case AUDIO_USAGE_UNKNOWN:
+ default:
+ return AUDIO_STREAM_MUSIC;
+ }
+}
}; // namespace android
diff --git a/services/audiopolicy/AudioPolicyManager.h b/services/audiopolicy/AudioPolicyManager.h
index 1abeb6a..c23d994 100644
--- a/services/audiopolicy/AudioPolicyManager.h
+++ b/services/audiopolicy/AudioPolicyManager.h
@@ -84,6 +84,12 @@
audio_channel_mask_t channelMask,
audio_output_flags_t flags,
const audio_offload_info_t *offloadInfo);
+ virtual audio_io_handle_t getOutputForAttr(const audio_attributes_t *attr,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo);
virtual status_t startOutput(audio_io_handle_t output,
audio_stream_type_t stream,
int session = 0);
@@ -116,6 +122,8 @@
// return the strategy corresponding to a given stream type
virtual uint32_t getStrategyForStream(audio_stream_type_t stream);
+ // return the strategy corresponding to the given audio attributes
+ virtual uint32_t getStrategyForAttr(const audio_attributes_t *attr);
// return the enabled output devices for the given stream type
virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream);
@@ -755,6 +763,17 @@
uint32_t curAudioPortGeneration() const { return mAudioPortGeneration; }
// converts device address to string sent to audio HAL via setParameters
static String8 addressToParameter(audio_devices_t device, const String8 address);
+ // internal method to return the output handle for the given device and format
+ audio_io_handle_t getOutputForDevice(
+ audio_devices_t device,
+ audio_stream_type_t stream,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo);
+ // internal function to derive a stream type value from audio attributes
+ audio_stream_type_t streamTypefromAttributesInt(const audio_attributes_t *attr);
};
};
diff --git a/services/audiopolicy/AudioPolicyService.cpp b/services/audiopolicy/AudioPolicyService.cpp
old mode 100644
new mode 100755
index 93fab11..9435797
--- a/services/audiopolicy/AudioPolicyService.cpp
+++ b/services/audiopolicy/AudioPolicyService.cpp
@@ -40,8 +40,6 @@
#include <system/audio.h>
#include <system/audio_policy.h>
#include <hardware/audio_policy.h>
-#include <audio_effects/audio_effects_conf.h>
-#include <media/AudioParameter.h>
namespace android {
@@ -111,12 +109,8 @@
mAudioPolicyManager = createAudioPolicyManager(mAudioPolicyClient);
#endif
- // load audio pre processing modules
- if (access(AUDIO_EFFECT_VENDOR_CONFIG_FILE, R_OK) == 0) {
- loadPreProcessorConfig(AUDIO_EFFECT_VENDOR_CONFIG_FILE);
- } else if (access(AUDIO_EFFECT_DEFAULT_CONFIG_FILE, R_OK) == 0) {
- loadPreProcessorConfig(AUDIO_EFFECT_DEFAULT_CONFIG_FILE);
- }
+ // load audio processing modules
+ mAudioPolicyEffects = new AudioPolicyEffects();
}
AudioPolicyService::~AudioPolicyService()
@@ -125,18 +119,6 @@
mAudioCommandThread->exit();
mOutputCommandThread->exit();
- // release audio pre processing resources
- for (size_t i = 0; i < mInputSources.size(); i++) {
- delete mInputSources.valueAt(i);
- }
- mInputSources.clear();
-
- for (size_t i = 0; i < mInputs.size(); i++) {
- mInputs.valueAt(i)->mEffects.clear();
- delete mInputs.valueAt(i);
- }
- mInputs.clear();
-
#ifdef USE_LEGACY_AUDIO_POLICY
if (mpAudioPolicy != NULL && mpAudioPolicyDev != NULL) {
mpAudioPolicyDev->destroy_audio_policy(mpAudioPolicyDev, mpAudioPolicy);
@@ -150,6 +132,7 @@
#endif
mNotificationClients.clear();
+ mAudioPolicyEffects.clear();
}
// A notification client is always registered by AudioSystem when the client process
@@ -353,14 +336,6 @@
return NO_ERROR;
}
-void AudioPolicyService::setPreProcessorEnabled(const InputDesc *inputDesc, bool enabled)
-{
- const Vector<sp<AudioEffect> > &fxVector = inputDesc->mEffects;
- for (size_t i = 0; i < fxVector.size(); i++) {
- fxVector.itemAt(i)->setEnabled(enabled);
- }
-}
-
status_t AudioPolicyService::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
@@ -936,304 +911,6 @@
return (int)mAudioCommandThread->voiceVolumeCommand(volume, delayMs);
}
-// ----------------------------------------------------------------------------
-// Audio pre-processing configuration
-// ----------------------------------------------------------------------------
-
-/*static*/ const char * const AudioPolicyService::kInputSourceNames[AUDIO_SOURCE_CNT -1] = {
- MIC_SRC_TAG,
- VOICE_UL_SRC_TAG,
- VOICE_DL_SRC_TAG,
- VOICE_CALL_SRC_TAG,
- CAMCORDER_SRC_TAG,
- VOICE_REC_SRC_TAG,
- VOICE_COMM_SRC_TAG
-};
-
-// returns the audio_source_t enum corresponding to the input source name or
-// AUDIO_SOURCE_CNT is no match found
-audio_source_t AudioPolicyService::inputSourceNameToEnum(const char *name)
-{
- int i;
- for (i = AUDIO_SOURCE_MIC; i < AUDIO_SOURCE_CNT; i++) {
- if (strcmp(name, kInputSourceNames[i - AUDIO_SOURCE_MIC]) == 0) {
- ALOGV("inputSourceNameToEnum found source %s %d", name, i);
- break;
- }
- }
- return (audio_source_t)i;
-}
-
-size_t AudioPolicyService::growParamSize(char *param,
- size_t size,
- size_t *curSize,
- size_t *totSize)
-{
- // *curSize is at least sizeof(effect_param_t) + 2 * sizeof(int)
- size_t pos = ((*curSize - 1 ) / size + 1) * size;
-
- if (pos + size > *totSize) {
- while (pos + size > *totSize) {
- *totSize += ((*totSize + 7) / 8) * 4;
- }
- param = (char *)realloc(param, *totSize);
- }
- *curSize = pos + size;
- return pos;
-}
-
-size_t AudioPolicyService::readParamValue(cnode *node,
- char *param,
- size_t *curSize,
- size_t *totSize)
-{
- if (strncmp(node->name, SHORT_TAG, sizeof(SHORT_TAG) + 1) == 0) {
- size_t pos = growParamSize(param, sizeof(short), curSize, totSize);
- *(short *)((char *)param + pos) = (short)atoi(node->value);
- ALOGV("readParamValue() reading short %d", *(short *)((char *)param + pos));
- return sizeof(short);
- } else if (strncmp(node->name, INT_TAG, sizeof(INT_TAG) + 1) == 0) {
- size_t pos = growParamSize(param, sizeof(int), curSize, totSize);
- *(int *)((char *)param + pos) = atoi(node->value);
- ALOGV("readParamValue() reading int %d", *(int *)((char *)param + pos));
- return sizeof(int);
- } else if (strncmp(node->name, FLOAT_TAG, sizeof(FLOAT_TAG) + 1) == 0) {
- size_t pos = growParamSize(param, sizeof(float), curSize, totSize);
- *(float *)((char *)param + pos) = (float)atof(node->value);
- ALOGV("readParamValue() reading float %f",*(float *)((char *)param + pos));
- return sizeof(float);
- } else if (strncmp(node->name, BOOL_TAG, sizeof(BOOL_TAG) + 1) == 0) {
- size_t pos = growParamSize(param, sizeof(bool), curSize, totSize);
- if (strncmp(node->value, "false", strlen("false") + 1) == 0) {
- *(bool *)((char *)param + pos) = false;
- } else {
- *(bool *)((char *)param + pos) = true;
- }
- ALOGV("readParamValue() reading bool %s",*(bool *)((char *)param + pos) ? "true" : "false");
- return sizeof(bool);
- } else if (strncmp(node->name, STRING_TAG, sizeof(STRING_TAG) + 1) == 0) {
- size_t len = strnlen(node->value, EFFECT_STRING_LEN_MAX);
- if (*curSize + len + 1 > *totSize) {
- *totSize = *curSize + len + 1;
- param = (char *)realloc(param, *totSize);
- }
- strncpy(param + *curSize, node->value, len);
- *curSize += len;
- param[*curSize] = '\0';
- ALOGV("readParamValue() reading string %s", param + *curSize - len);
- return len;
- }
- ALOGW("readParamValue() unknown param type %s", node->name);
- return 0;
-}
-
-effect_param_t *AudioPolicyService::loadEffectParameter(cnode *root)
-{
- cnode *param;
- cnode *value;
- size_t curSize = sizeof(effect_param_t);
- size_t totSize = sizeof(effect_param_t) + 2 * sizeof(int);
- effect_param_t *fx_param = (effect_param_t *)malloc(totSize);
-
- param = config_find(root, PARAM_TAG);
- value = config_find(root, VALUE_TAG);
- if (param == NULL && value == NULL) {
- // try to parse simple parameter form {int int}
- param = root->first_child;
- 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));
- ALOGW("loadEffectParameter() ptr %p ptr2 %p", ptr, ptr2);
- *ptr++ = atoi(param->name);
- *ptr = atoi(param->value);
- fx_param->psize = sizeof(int);
- fx_param->vsize = sizeof(int);
- return fx_param;
- }
- }
- if (param == NULL || value == NULL) {
- ALOGW("loadEffectParameter() invalid parameter description %s", root->name);
- goto error;
- }
-
- fx_param->psize = 0;
- param = param->first_child;
- while (param) {
- ALOGV("loadEffectParameter() reading param of type %s", param->name);
- size_t size = readParamValue(param, (char *)fx_param, &curSize, &totSize);
- if (size == 0) {
- goto error;
- }
- fx_param->psize += size;
- param = param->next;
- }
-
- // align start of value field on 32 bit boundary
- curSize = ((curSize - 1 ) / sizeof(int) + 1) * sizeof(int);
-
- fx_param->vsize = 0;
- value = value->first_child;
- while (value) {
- ALOGV("loadEffectParameter() reading value of type %s", value->name);
- size_t size = readParamValue(value, (char *)fx_param, &curSize, &totSize);
- if (size == 0) {
- goto error;
- }
- fx_param->vsize += size;
- value = value->next;
- }
-
- return fx_param;
-
-error:
- free(fx_param);
- return NULL;
-}
-
-void AudioPolicyService::loadEffectParameters(cnode *root, Vector <effect_param_t *>& params)
-{
- cnode *node = root->first_child;
- while (node) {
- ALOGV("loadEffectParameters() loading param %s", node->name);
- effect_param_t *param = loadEffectParameter(node);
- if (param == NULL) {
- node = node->next;
- continue;
- }
- params.add(param);
- node = node->next;
- }
-}
-
-AudioPolicyService::InputSourceDesc *AudioPolicyService::loadInputSource(
- cnode *root,
- const Vector <EffectDesc *>& effects)
-{
- cnode *node = root->first_child;
- if (node == NULL) {
- ALOGW("loadInputSource() empty element %s", root->name);
- return NULL;
- }
- InputSourceDesc *source = new InputSourceDesc();
- while (node) {
- size_t i;
- for (i = 0; i < effects.size(); i++) {
- if (strncmp(effects[i]->mName, node->name, EFFECT_STRING_LEN_MAX) == 0) {
- ALOGV("loadInputSource() found effect %s in list", node->name);
- break;
- }
- }
- if (i == effects.size()) {
- ALOGV("loadInputSource() effect %s not in list", node->name);
- node = node->next;
- continue;
- }
- 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);
- node = node->next;
- }
- if (source->mEffects.size() == 0) {
- ALOGW("loadInputSource() no valid effects found in source %s", root->name);
- delete source;
- return NULL;
- }
- return source;
-}
-
-status_t AudioPolicyService::loadInputSources(cnode *root, const Vector <EffectDesc *>& effects)
-{
- cnode *node = config_find(root, PREPROCESSING_TAG);
- if (node == NULL) {
- return -ENOENT;
- }
- node = node->first_child;
- while (node) {
- audio_source_t source = inputSourceNameToEnum(node->name);
- if (source == AUDIO_SOURCE_CNT) {
- ALOGW("loadInputSources() invalid input source %s", node->name);
- node = node->next;
- continue;
- }
- ALOGV("loadInputSources() loading input source %s", node->name);
- InputSourceDesc *desc = loadInputSource(node, effects);
- if (desc == NULL) {
- node = node->next;
- continue;
- }
- mInputSources.add(source, desc);
- node = node->next;
- }
- return NO_ERROR;
-}
-
-AudioPolicyService::EffectDesc *AudioPolicyService::loadEffect(cnode *root)
-{
- cnode *node = config_find(root, UUID_TAG);
- if (node == NULL) {
- return NULL;
- }
- effect_uuid_t uuid;
- if (AudioEffect::stringToGuid(node->value, &uuid) != NO_ERROR) {
- ALOGW("loadEffect() invalid uuid %s", node->value);
- return NULL;
- }
- return new EffectDesc(root->name, uuid);
-}
-
-status_t AudioPolicyService::loadEffects(cnode *root, Vector <EffectDesc *>& effects)
-{
- cnode *node = config_find(root, EFFECTS_TAG);
- if (node == NULL) {
- return -ENOENT;
- }
- node = node->first_child;
- while (node) {
- ALOGV("loadEffects() loading effect %s", node->name);
- EffectDesc *effect = loadEffect(node);
- if (effect == NULL) {
- node = node->next;
- continue;
- }
- effects.add(effect);
- node = node->next;
- }
- return NO_ERROR;
-}
-
-status_t AudioPolicyService::loadPreProcessorConfig(const char *path)
-{
- cnode *root;
- char *data;
-
- data = (char *)load_file(path, NULL);
- if (data == NULL) {
- return -ENODEV;
- }
- root = config_node("", "");
- config_load(root, data);
-
- Vector <EffectDesc *> effects;
- loadEffects(root, effects);
- loadInputSources(root, effects);
-
- // delete effects to fix memory leak.
- // as effects is local var and valgrind would treat this as memory leak
- // and although it only did in mediaserver init, but free it in case mediaserver reboot
- size_t i;
- for (i = 0; i < effects.size(); i++) {
- delete effects[i];
- }
-
- config_free(root);
- free(root);
- free(data);
-
- return NO_ERROR;
-}
-
extern "C" {
audio_module_handle_t aps_load_hw_module(void *service __unused,
const char *name);
diff --git a/services/audiopolicy/AudioPolicyService.h b/services/audiopolicy/AudioPolicyService.h
old mode 100644
new mode 100755
index 66d9cad..380fd5e
--- a/services/audiopolicy/AudioPolicyService.h
+++ b/services/audiopolicy/AudioPolicyService.h
@@ -31,8 +31,10 @@
#include <media/ToneGenerator.h>
#include <media/AudioEffect.h>
#include <hardware_legacy/AudioPolicyInterface.h>
+#include "AudioPolicyEffects.h"
#include "AudioPolicyManager.h"
+
namespace android {
// ----------------------------------------------------------------------------
@@ -70,6 +72,12 @@
audio_output_flags_t flags =
AUDIO_OUTPUT_FLAG_NONE,
const audio_offload_info_t *offloadInfo = NULL);
+ virtual audio_io_handle_t getOutputForAttr(const audio_attributes_t *attr,
+ uint32_t samplingRate = 0,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
+ audio_channel_mask_t channelMask = 0,
+ audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+ const audio_offload_info_t *offloadInfo = NULL);
virtual status_t startOutput(audio_io_handle_t output,
audio_stream_type_t stream,
int session = 0);
@@ -331,60 +339,6 @@
wp<AudioPolicyService> mService;
};
- class EffectDesc {
- public:
- 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;
- };
-
- class InputSourceDesc {
- public:
- InputSourceDesc() {}
- /*virtual*/ ~InputSourceDesc() {
- for (size_t j = 0; j < mEffects.size(); j++) {
- delete mEffects[j];
- }
- }
- Vector <EffectDesc *> mEffects;
- };
-
-
- class InputDesc {
- public:
- InputDesc(int session) : mSessionId(session) {}
- /*virtual*/ ~InputDesc() {}
- const int mSessionId;
- Vector< sp<AudioEffect> >mEffects;
- };
-
class AudioPolicyClient : public AudioPolicyClientInterface
{
public:
@@ -508,26 +462,6 @@
const sp<IAudioPolicyServiceClient> mAudioPolicyServiceClient;
};
- static const char * const kInputSourceNames[AUDIO_SOURCE_CNT -1];
-
- 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);
- status_t loadInputSources(cnode *root, const Vector <EffectDesc *>& effects);
- audio_source_t inputSourceNameToEnum(const char *name);
- InputSourceDesc *loadInputSource(cnode *root, const Vector <EffectDesc *>& effects);
- void loadEffectParameters(cnode *root, Vector <effect_param_t *>& params);
- effect_param_t *loadEffectParameter(cnode *root);
- size_t readParamValue(cnode *node,
- char *param,
- size_t *curSize,
- size_t *totSize);
- size_t growParamSize(char *param,
- size_t size,
- size_t *curSize,
- size_t *totSize);
-
// Internal dump utilities.
status_t dumpPermissionDenial(int fd);
@@ -542,10 +476,10 @@
AudioPolicyInterface *mAudioPolicyManager;
AudioPolicyClient *mAudioPolicyClient;
- KeyedVector< audio_source_t, InputSourceDesc* > mInputSources;
- KeyedVector< audio_io_handle_t, InputDesc* > mInputs;
-
DefaultKeyedVector< uid_t, sp<NotificationClient> > mNotificationClients;
+
+ // Manage all effects configured in audio_effects.conf
+ sp<AudioPolicyEffects> mAudioPolicyEffects;
};
}; // namespace android
diff --git a/services/camera/libcameraservice/CameraService.cpp b/services/camera/libcameraservice/CameraService.cpp
index 73eccbf..648e82c 100644
--- a/services/camera/libcameraservice/CameraService.cpp
+++ b/services/camera/libcameraservice/CameraService.cpp
@@ -659,7 +659,8 @@
int clientUid,
int callingPid,
/*out*/
- sp<Client>& client) {
+ sp<Client>& client,
+ int halVersion) {
int facing = -1;
int deviceVersion = getDeviceVersion(cameraId, &facing);
@@ -672,28 +673,47 @@
cameraId);
}
- switch(deviceVersion) {
- case CAMERA_DEVICE_API_VERSION_1_0:
- client = new CameraClient(this, cameraClient,
- clientPackageName, cameraId,
- facing, callingPid, clientUid, getpid());
- break;
- case CAMERA_DEVICE_API_VERSION_2_0:
- case CAMERA_DEVICE_API_VERSION_2_1:
- case CAMERA_DEVICE_API_VERSION_3_0:
- case CAMERA_DEVICE_API_VERSION_3_1:
- case CAMERA_DEVICE_API_VERSION_3_2:
- client = new Camera2Client(this, cameraClient,
- clientPackageName, cameraId,
- facing, callingPid, clientUid, getpid(),
- deviceVersion);
- break;
- case -1:
- ALOGE("Invalid camera id %d", cameraId);
- return BAD_VALUE;
- default:
- ALOGE("Unknown camera device HAL version: %d", deviceVersion);
- return INVALID_OPERATION;
+ if (halVersion < 0 || halVersion == deviceVersion) {
+ // Default path: HAL version is unspecified by caller, create CameraClient
+ // based on device version reported by the HAL.
+ switch(deviceVersion) {
+ case CAMERA_DEVICE_API_VERSION_1_0:
+ client = new CameraClient(this, cameraClient,
+ clientPackageName, cameraId,
+ facing, callingPid, clientUid, getpid());
+ break;
+ case CAMERA_DEVICE_API_VERSION_2_0:
+ case CAMERA_DEVICE_API_VERSION_2_1:
+ case CAMERA_DEVICE_API_VERSION_3_0:
+ case CAMERA_DEVICE_API_VERSION_3_1:
+ case CAMERA_DEVICE_API_VERSION_3_2:
+ client = new Camera2Client(this, cameraClient,
+ clientPackageName, cameraId,
+ facing, callingPid, clientUid, getpid());
+ break;
+ case -1:
+ ALOGE("Invalid camera id %d", cameraId);
+ return BAD_VALUE;
+ default:
+ ALOGE("Unknown camera device HAL version: %d", deviceVersion);
+ return INVALID_OPERATION;
+ }
+ } else {
+ // A particular HAL version is requested by caller. Create CameraClient
+ // based on the requested HAL version.
+ if (deviceVersion > CAMERA_DEVICE_API_VERSION_1_0 &&
+ halVersion == CAMERA_DEVICE_API_VERSION_1_0) {
+ // Only support higher HAL version device opened as HAL1.0 device.
+ client = new CameraClient(this, cameraClient,
+ clientPackageName, cameraId,
+ facing, callingPid, clientUid, getpid());
+ } else {
+ // Other combinations (e.g. HAL3.x open as HAL2.x) are not supported yet.
+ ALOGE("Invalid camera HAL version %x: HAL %x device can only be"
+ " opened as HAL %x device", halVersion, deviceVersion,
+ CAMERA_DEVICE_API_VERSION_1_0);
+ return INVALID_OPERATION;
+ }
}
status_t status = connectFinishUnsafe(client, client->getRemote());
@@ -762,6 +782,70 @@
return OK;
}
+status_t CameraService::connectLegacy(
+ const sp<ICameraClient>& cameraClient,
+ int cameraId, int halVersion,
+ const String16& clientPackageName,
+ int clientUid,
+ /*out*/
+ sp<ICamera>& device) {
+
+ if (halVersion != CAMERA_HAL_API_VERSION_UNSPECIFIED &&
+ mModule->common.module_api_version < CAMERA_MODULE_API_VERSION_2_3) {
+ /*
+ * Either the HAL version is unspecified in which case this just creates
+ * a camera client selected by the latest device version, or
+ * it's a particular version in which case the HAL must supported
+ * the open_legacy call
+ */
+ ALOGE("%s: camera HAL module version %x doesn't support connecting to legacy HAL devices!",
+ __FUNCTION__, mModule->common.module_api_version);
+ return INVALID_OPERATION;
+ }
+
+ String8 clientName8(clientPackageName);
+ int callingPid = getCallingPid();
+
+ LOG1("CameraService::connect legacy E (pid %d \"%s\", id %d)", callingPid,
+ clientName8.string(), cameraId);
+
+ status_t status = validateConnect(cameraId, /*inout*/clientUid);
+ if (status != OK) {
+ return status;
+ }
+
+ sp<Client> client;
+ {
+ Mutex::Autolock lock(mServiceLock);
+ sp<BasicClient> clientTmp;
+ if (!canConnectUnsafe(cameraId, clientPackageName,
+ cameraClient->asBinder(),
+ /*out*/clientTmp)) {
+ return -EBUSY;
+ } else if (client.get() != NULL) {
+ device = static_cast<Client*>(clientTmp.get());
+ return OK;
+ }
+
+ status = connectHelperLocked(cameraClient,
+ cameraId,
+ clientPackageName,
+ clientUid,
+ callingPid,
+ client,
+ halVersion);
+ if (status != OK) {
+ return status;
+ }
+
+ }
+ // important: release the mutex here so the client can call back
+ // into the service from its destructor (can be at the end of the call)
+
+ device = client;
+ return OK;
+}
+
status_t CameraService::connectFinishUnsafe(const sp<BasicClient>& client,
const sp<IBinder>& remoteCallback) {
status_t status = client->initialize(mModule);
@@ -1196,6 +1280,7 @@
case BnCameraService::CONNECT:
case BnCameraService::CONNECT_PRO:
case BnCameraService::CONNECT_DEVICE:
+ case BnCameraService::CONNECT_LEGACY:
const int pid = getCallingPid();
const int self_pid = getpid();
if (pid != self_pid) {
diff --git a/services/camera/libcameraservice/CameraService.h b/services/camera/libcameraservice/CameraService.h
index b2b65b8..28590eb 100644
--- a/services/camera/libcameraservice/CameraService.h
+++ b/services/camera/libcameraservice/CameraService.h
@@ -83,6 +83,11 @@
/*out*/
sp<ICamera>& device);
+ virtual status_t connectLegacy(const sp<ICameraClient>& cameraClient, int cameraId,
+ int halVersion, const String16& clientPackageName, int clientUid,
+ /*out*/
+ sp<ICamera>& device);
+
virtual status_t connectPro(const sp<IProCameraCallbacks>& cameraCb,
int cameraId, const String16& clientPackageName, int clientUid,
/*out*/
@@ -450,7 +455,8 @@
int clientUid,
int callingPid,
/*out*/
- sp<Client>& client);
+ sp<Client>& client,
+ int halVersion = CAMERA_HAL_API_VERSION_UNSPECIFIED);
};
} // namespace android
diff --git a/services/camera/libcameraservice/api1/Camera2Client.cpp b/services/camera/libcameraservice/api1/Camera2Client.cpp
index 0447979..1642896 100644
--- a/services/camera/libcameraservice/api1/Camera2Client.cpp
+++ b/services/camera/libcameraservice/api1/Camera2Client.cpp
@@ -53,12 +53,10 @@
int cameraFacing,
int clientPid,
uid_t clientUid,
- int servicePid,
- int deviceVersion):
+ int servicePid):
Camera2ClientBase(cameraService, cameraClient, clientPackageName,
cameraId, cameraFacing, clientPid, clientUid, servicePid),
- mParameters(cameraId, cameraFacing),
- mDeviceVersion(deviceVersion)
+ mParameters(cameraId, cameraFacing)
{
ATRACE_CALL();
@@ -80,7 +78,7 @@
{
SharedParameters::Lock l(mParameters);
- res = l.mParameters.initialize(&(mDevice->info()));
+ res = l.mParameters.initialize(&(mDevice->info()), mDeviceVersion);
if (res != OK) {
ALOGE("%s: Camera %d: unable to build defaults: %s (%d)",
__FUNCTION__, mCameraId, strerror(-res), res);
@@ -755,6 +753,7 @@
// ever take a picture.
// TODO: Find a better compromise, though this likely would involve HAL
// changes.
+ int lastJpegStreamId = mJpegProcessor->getStreamId();
res = updateProcessorStream(mJpegProcessor, params);
if (res != OK) {
ALOGE("%s: Camera %d: Can't pre-configure still image "
@@ -762,6 +761,7 @@
__FUNCTION__, mCameraId, strerror(-res), res);
return res;
}
+ bool jpegStreamChanged = mJpegProcessor->getStreamId() != lastJpegStreamId;
Vector<int32_t> outputStreams;
bool callbacksEnabled = (params.previewCallbackFlags &
@@ -817,6 +817,12 @@
__FUNCTION__, mCameraId, strerror(-res), res);
return res;
}
+
+ if (jpegStreamChanged) {
+ ALOGV("%s: Camera %d: Clear ZSL buffer queue when Jpeg size is changed",
+ __FUNCTION__, mCameraId);
+ mZslProcessor->clearZslQueue();
+ }
outputStreams.push(getZslStreamId());
} else {
mZslProcessor->deleteStream();
@@ -1270,6 +1276,7 @@
ALOGV("%s: Camera %d: Starting picture capture", __FUNCTION__, mCameraId);
+ int lastJpegStreamId = mJpegProcessor->getStreamId();
res = updateProcessorStream(mJpegProcessor, l.mParameters);
if (res != OK) {
ALOGE("%s: Camera %d: Can't set up still image stream: %s (%d)",
@@ -1277,6 +1284,14 @@
return res;
}
takePictureCounter = ++l.mParameters.takePictureCounter;
+
+ // Clear ZSL buffer queue when Jpeg size is changed.
+ bool jpegStreamChanged = mJpegProcessor->getStreamId() != lastJpegStreamId;
+ if (l.mParameters.zslMode && jpegStreamChanged) {
+ ALOGV("%s: Camera %d: Clear ZSL buffer queue when Jpeg size is changed",
+ __FUNCTION__, mCameraId);
+ mZslProcessor->clearZslQueue();
+ }
}
ATRACE_ASYNC_BEGIN(kTakepictureLabel, takePictureCounter);
@@ -1659,8 +1674,8 @@
}
status_t Camera2Client::registerFrameListener(int32_t minId, int32_t maxId,
- wp<camera2::FrameProcessor::FilteredListener> listener) {
- return mFrameProcessor->registerListener(minId, maxId, listener);
+ wp<camera2::FrameProcessor::FilteredListener> listener, bool sendPartials) {
+ return mFrameProcessor->registerListener(minId, maxId, listener, sendPartials);
}
status_t Camera2Client::removeFrameListener(int32_t minId, int32_t maxId,
diff --git a/services/camera/libcameraservice/api1/Camera2Client.h b/services/camera/libcameraservice/api1/Camera2Client.h
index fe0bf74..5ce757a 100644
--- a/services/camera/libcameraservice/api1/Camera2Client.h
+++ b/services/camera/libcameraservice/api1/Camera2Client.h
@@ -89,8 +89,7 @@
int cameraFacing,
int clientPid,
uid_t clientUid,
- int servicePid,
- int deviceVersion);
+ int servicePid);
virtual ~Camera2Client();
@@ -118,7 +117,8 @@
int getZslStreamId() const;
status_t registerFrameListener(int32_t minId, int32_t maxId,
- wp<camera2::FrameProcessor::FilteredListener> listener);
+ wp<camera2::FrameProcessor::FilteredListener> listener,
+ bool sendPartials = true);
status_t removeFrameListener(int32_t minId, int32_t maxId,
wp<camera2::FrameProcessor::FilteredListener> listener);
@@ -170,7 +170,6 @@
void setPreviewCallbackFlagL(Parameters ¶ms, int flag);
status_t updateRequests(Parameters ¶ms);
- int mDeviceVersion;
// Used with stream IDs
static const int NO_STREAM = -1;
diff --git a/services/camera/libcameraservice/api1/CameraClient.cpp b/services/camera/libcameraservice/api1/CameraClient.cpp
index 30b7bb8..517226d 100644
--- a/services/camera/libcameraservice/api1/CameraClient.cpp
+++ b/services/camera/libcameraservice/api1/CameraClient.cpp
@@ -79,7 +79,7 @@
ALOGE("%s: Camera %d: unable to initialize device: %s (%d)",
__FUNCTION__, mCameraId, strerror(-res), res);
mHardware.clear();
- return NO_INIT;
+ return res;
}
mHardware->setCallbacks(notifyCallback,
diff --git a/services/camera/libcameraservice/api1/client2/CaptureSequencer.cpp b/services/camera/libcameraservice/api1/client2/CaptureSequencer.cpp
index 8268f65..cb9aca6 100644
--- a/services/camera/libcameraservice/api1/client2/CaptureSequencer.cpp
+++ b/services/camera/libcameraservice/api1/client2/CaptureSequencer.cpp
@@ -350,8 +350,10 @@
return DONE;
}
+ // We don't want to get partial results for ZSL capture.
client->registerFrameListener(mCaptureId, mCaptureId + 1,
- this);
+ this,
+ /*sendPartials*/false);
// TODO: Actually select the right thing here.
res = processor->pushToReprocess(mCaptureId);
@@ -393,8 +395,14 @@
bool isAeConverged = false;
// Get the onFrameAvailable callback when the requestID == mCaptureId
+ // We don't want to get partial results for normal capture, as we need
+ // Get ANDROID_SENSOR_TIMESTAMP from the capture result, but partial
+ // result doesn't have to have this metadata available.
+ // TODO: Update to use the HALv3 shutter notification for remove the
+ // need for this listener and make it faster. see bug 12530628.
client->registerFrameListener(mCaptureId, mCaptureId + 1,
- this);
+ this,
+ /*sendPartials*/false);
{
Mutex::Autolock l(mInputMutex);
diff --git a/services/camera/libcameraservice/api1/client2/FrameProcessor.cpp b/services/camera/libcameraservice/api1/client2/FrameProcessor.cpp
index 69bea24..3de5d90 100644
--- a/services/camera/libcameraservice/api1/client2/FrameProcessor.cpp
+++ b/services/camera/libcameraservice/api1/client2/FrameProcessor.cpp
@@ -78,7 +78,7 @@
}
if (mSynthesize3ANotify) {
- process3aState(frame.mMetadata, client);
+ process3aState(frame, client);
}
return FrameProcessorBase::processSingleFrame(frame, device);
@@ -212,14 +212,15 @@
return OK;
}
-status_t FrameProcessor::process3aState(const CameraMetadata &frame,
+status_t FrameProcessor::process3aState(const CaptureResult &frame,
const sp<Camera2Client> &client) {
ATRACE_CALL();
+ const CameraMetadata &metadata = frame.mMetadata;
camera_metadata_ro_entry_t entry;
int cameraId = client->getCameraId();
- entry = frame.find(ANDROID_REQUEST_FRAME_COUNT);
+ entry = metadata.find(ANDROID_REQUEST_FRAME_COUNT);
int32_t frameNumber = entry.data.i32[0];
// Don't send 3A notifications for the same frame number twice
@@ -238,26 +239,31 @@
// TODO: Also use AE mode, AE trigger ID
- gotAllStates &= get3aResult<uint8_t>(frame, ANDROID_CONTROL_AF_MODE,
+ gotAllStates &= get3aResult<uint8_t>(metadata, ANDROID_CONTROL_AF_MODE,
&new3aState.afMode, frameNumber, cameraId);
- gotAllStates &= get3aResult<uint8_t>(frame, ANDROID_CONTROL_AWB_MODE,
+ gotAllStates &= get3aResult<uint8_t>(metadata, ANDROID_CONTROL_AWB_MODE,
&new3aState.awbMode, frameNumber, cameraId);
- gotAllStates &= get3aResult<uint8_t>(frame, ANDROID_CONTROL_AE_STATE,
+ gotAllStates &= get3aResult<uint8_t>(metadata, ANDROID_CONTROL_AE_STATE,
&new3aState.aeState, frameNumber, cameraId);
- gotAllStates &= get3aResult<uint8_t>(frame, ANDROID_CONTROL_AF_STATE,
+ gotAllStates &= get3aResult<uint8_t>(metadata, ANDROID_CONTROL_AF_STATE,
&new3aState.afState, frameNumber, cameraId);
- gotAllStates &= get3aResult<uint8_t>(frame, ANDROID_CONTROL_AWB_STATE,
+ gotAllStates &= get3aResult<uint8_t>(metadata, ANDROID_CONTROL_AWB_STATE,
&new3aState.awbState, frameNumber, cameraId);
- gotAllStates &= get3aResult<int32_t>(frame, ANDROID_CONTROL_AF_TRIGGER_ID,
- &new3aState.afTriggerId, frameNumber, cameraId);
+ if (client->getCameraDeviceVersion() >= CAMERA_DEVICE_API_VERSION_3_2) {
+ new3aState.afTriggerId = frame.mResultExtras.afTriggerId;
+ new3aState.aeTriggerId = frame.mResultExtras.precaptureTriggerId;
+ } else {
+ gotAllStates &= get3aResult<int32_t>(metadata, ANDROID_CONTROL_AF_TRIGGER_ID,
+ &new3aState.afTriggerId, frameNumber, cameraId);
- gotAllStates &= get3aResult<int32_t>(frame, ANDROID_CONTROL_AE_PRECAPTURE_ID,
- &new3aState.aeTriggerId, frameNumber, cameraId);
+ gotAllStates &= get3aResult<int32_t>(metadata, ANDROID_CONTROL_AE_PRECAPTURE_ID,
+ &new3aState.aeTriggerId, frameNumber, cameraId);
+ }
if (!gotAllStates) return BAD_VALUE;
diff --git a/services/camera/libcameraservice/api1/client2/FrameProcessor.h b/services/camera/libcameraservice/api1/client2/FrameProcessor.h
index 514bd1a..4afca50 100644
--- a/services/camera/libcameraservice/api1/client2/FrameProcessor.h
+++ b/services/camera/libcameraservice/api1/client2/FrameProcessor.h
@@ -58,7 +58,7 @@
const sp<Camera2Client> &client);
// Send 3A state change notifications to client based on frame metadata
- status_t process3aState(const CameraMetadata &frame,
+ status_t process3aState(const CaptureResult &frame,
const sp<Camera2Client> &client);
// Helper for process3aState
diff --git a/services/camera/libcameraservice/api1/client2/Parameters.cpp b/services/camera/libcameraservice/api1/client2/Parameters.cpp
index 51b1980..6459300 100644
--- a/services/camera/libcameraservice/api1/client2/Parameters.cpp
+++ b/services/camera/libcameraservice/api1/client2/Parameters.cpp
@@ -29,6 +29,7 @@
#include "Parameters.h"
#include "system/camera.h"
+#include "hardware/camera_common.h"
#include <media/MediaProfiles.h>
#include <media/mediarecorder.h>
@@ -45,7 +46,7 @@
Parameters::~Parameters() {
}
-status_t Parameters::initialize(const CameraMetadata *info) {
+status_t Parameters::initialize(const CameraMetadata *info, int deviceVersion) {
status_t res;
if (info->entryCount() == 0) {
@@ -53,6 +54,7 @@
return BAD_VALUE;
}
Parameters::info = info;
+ mDeviceVersion = deviceVersion;
res = buildFastInfo();
if (res != OK) return res;
@@ -140,16 +142,14 @@
previewTransform = degToTransform(0,
cameraFacing == CAMERA_FACING_FRONT);
- camera_metadata_ro_entry_t availableFormats =
- staticInfo(ANDROID_SCALER_AVAILABLE_FORMATS);
-
{
String8 supportedPreviewFormats;
+ SortedVector<int32_t> outputFormats = getAvailableOutputFormats();
bool addComma = false;
- for (size_t i=0; i < availableFormats.count; i++) {
+ for (size_t i=0; i < outputFormats.size(); i++) {
if (addComma) supportedPreviewFormats += ",";
addComma = true;
- switch (availableFormats.data.i32[i]) {
+ switch (outputFormats[i]) {
case HAL_PIXEL_FORMAT_YCbCr_422_SP:
supportedPreviewFormats +=
CameraParameters::PIXEL_FORMAT_YUV422SP;
@@ -191,7 +191,7 @@
default:
ALOGW("%s: Camera %d: Unknown preview format: %x",
- __FUNCTION__, cameraId, availableFormats.data.i32[i]);
+ __FUNCTION__, cameraId, outputFormats[i]);
addComma = false;
break;
}
@@ -239,24 +239,23 @@
supportedPreviewFrameRates);
}
- camera_metadata_ro_entry_t availableJpegSizes =
- staticInfo(ANDROID_SCALER_AVAILABLE_JPEG_SIZES, 2);
- if (!availableJpegSizes.count) return NO_INIT;
+ Vector<Size> availableJpegSizes = getAvailableJpegSizes();
+ if (!availableJpegSizes.size()) return NO_INIT;
// TODO: Pick maximum
- pictureWidth = availableJpegSizes.data.i32[0];
- pictureHeight = availableJpegSizes.data.i32[1];
+ pictureWidth = availableJpegSizes[0].width;
+ pictureHeight = availableJpegSizes[0].height;
params.setPictureSize(pictureWidth,
pictureHeight);
{
String8 supportedPictureSizes;
- for (size_t i=0; i < availableJpegSizes.count; i += 2) {
+ for (size_t i=0; i < availableJpegSizes.size(); i++) {
if (i != 0) supportedPictureSizes += ",";
supportedPictureSizes += String8::format("%dx%d",
- availableJpegSizes.data.i32[i],
- availableJpegSizes.data.i32[i+1]);
+ availableJpegSizes[i].width,
+ availableJpegSizes[i].height);
}
params.set(CameraParameters::KEY_SUPPORTED_PICTURE_SIZES,
supportedPictureSizes);
@@ -952,9 +951,8 @@
staticInfo(ANDROID_LENS_INFO_AVAILABLE_FOCAL_LENGTHS);
if (!availableFocalLengths.count) return NO_INIT;
- camera_metadata_ro_entry_t availableFormats =
- staticInfo(ANDROID_SCALER_AVAILABLE_FORMATS);
- if (!availableFormats.count) return NO_INIT;
+ SortedVector<int32_t> availableFormats = getAvailableOutputFormats();
+ if (!availableFormats.size()) return NO_INIT;
if (sceneModeOverrides.count > 0) {
@@ -1038,8 +1036,8 @@
// Check if the HAL supports HAL_PIXEL_FORMAT_YCbCr_420_888
fastInfo.useFlexibleYuv = false;
- for (size_t i = 0; i < availableFormats.count; i++) {
- if (availableFormats.data.i32[i] == HAL_PIXEL_FORMAT_YCbCr_420_888) {
+ for (size_t i = 0; i < availableFormats.size(); i++) {
+ if (availableFormats[i] == HAL_PIXEL_FORMAT_YCbCr_420_888) {
fastInfo.useFlexibleYuv = true;
break;
}
@@ -1198,8 +1196,7 @@
"is active!", __FUNCTION__);
return BAD_VALUE;
}
- camera_metadata_ro_entry_t availableFormats =
- staticInfo(ANDROID_SCALER_AVAILABLE_FORMATS);
+ SortedVector<int32_t> availableFormats = getAvailableOutputFormats();
// If using flexible YUV, always support NV21/YV12. Otherwise, check
// HAL's list.
if (! (fastInfo.useFlexibleYuv &&
@@ -1208,11 +1205,10 @@
validatedParams.previewFormat ==
HAL_PIXEL_FORMAT_YV12) ) ) {
// Not using flexible YUV format, so check explicitly
- for (i = 0; i < availableFormats.count; i++) {
- if (availableFormats.data.i32[i] ==
- validatedParams.previewFormat) break;
+ for (i = 0; i < availableFormats.size(); i++) {
+ if (availableFormats[i] == validatedParams.previewFormat) break;
}
- if (i == availableFormats.count) {
+ if (i == availableFormats.size()) {
ALOGE("%s: Requested preview format %s (0x%x) is not supported",
__FUNCTION__, newParams.getPreviewFormat(),
validatedParams.previewFormat);
@@ -1302,15 +1298,14 @@
&validatedParams.pictureHeight);
if (validatedParams.pictureWidth == pictureWidth ||
validatedParams.pictureHeight == pictureHeight) {
- camera_metadata_ro_entry_t availablePictureSizes =
- staticInfo(ANDROID_SCALER_AVAILABLE_JPEG_SIZES);
- for (i = 0; i < availablePictureSizes.count; i+=2) {
- if ((availablePictureSizes.data.i32[i] ==
+ Vector<Size> availablePictureSizes = getAvailableJpegSizes();
+ for (i = 0; i < availablePictureSizes.size(); i++) {
+ if ((availablePictureSizes[i].width ==
validatedParams.pictureWidth) &&
- (availablePictureSizes.data.i32[i+1] ==
+ (availablePictureSizes[i].height ==
validatedParams.pictureHeight)) break;
}
- if (i == availablePictureSizes.count) {
+ if (i == availablePictureSizes.size()) {
ALOGE("%s: Requested picture size %d x %d is not supported",
__FUNCTION__, validatedParams.pictureWidth,
validatedParams.pictureHeight);
@@ -2527,22 +2522,37 @@
ALOGE("%s: Input size is null", __FUNCTION__);
return BAD_VALUE;
}
+ sizes->clear();
- const size_t SIZE_COUNT = sizeof(Size) / sizeof(int);
- camera_metadata_ro_entry_t availableProcessedSizes =
- staticInfo(ANDROID_SCALER_AVAILABLE_PROCESSED_SIZES, SIZE_COUNT);
- if (availableProcessedSizes.count < SIZE_COUNT) return BAD_VALUE;
-
- Size filteredSize;
- for (size_t i = 0; i < availableProcessedSizes.count; i += SIZE_COUNT) {
- filteredSize.width = availableProcessedSizes.data.i32[i];
- filteredSize.height = availableProcessedSizes.data.i32[i+1];
- // Need skip the preview sizes that are too large.
- if (filteredSize.width <= limit.width &&
- filteredSize.height <= limit.height) {
- sizes->push(filteredSize);
+ if (mDeviceVersion >= CAMERA_DEVICE_API_VERSION_3_2) {
+ Vector<StreamConfiguration> scs = getStreamConfigurations();
+ for (size_t i=0; i < scs.size(); i++) {
+ const StreamConfiguration &sc = scs[i];
+ if (sc.isInput == ANDROID_SCALER_AVAILABLE_STREAM_CONFIGURATIONS_OUTPUT &&
+ sc.format == HAL_PIXEL_FORMAT_IMPLEMENTATION_DEFINED &&
+ sc.width <= limit.width && sc.height <= limit.height) {
+ Size sz = {sc.width, sc.height};
+ sizes->push(sz);
}
+ }
+ } else {
+ const size_t SIZE_COUNT = sizeof(Size) / sizeof(int);
+ camera_metadata_ro_entry_t availableProcessedSizes =
+ staticInfo(ANDROID_SCALER_AVAILABLE_PROCESSED_SIZES, SIZE_COUNT);
+ if (availableProcessedSizes.count < SIZE_COUNT) return BAD_VALUE;
+
+ Size filteredSize;
+ for (size_t i = 0; i < availableProcessedSizes.count; i += SIZE_COUNT) {
+ filteredSize.width = availableProcessedSizes.data.i32[i];
+ filteredSize.height = availableProcessedSizes.data.i32[i+1];
+ // Need skip the preview sizes that are too large.
+ if (filteredSize.width <= limit.width &&
+ filteredSize.height <= limit.height) {
+ sizes->push(filteredSize);
+ }
+ }
}
+
if (sizes->isEmpty()) {
ALOGE("generated preview size list is empty!!");
return BAD_VALUE;
@@ -2576,6 +2586,78 @@
return maxSize;
}
+Vector<Parameters::StreamConfiguration> Parameters::getStreamConfigurations() {
+ const int STREAM_CONFIGURATION_SIZE = 4;
+ const int STREAM_FORMAT_OFFSET = 0;
+ const int STREAM_WIDTH_OFFSET = 1;
+ const int STREAM_HEIGHT_OFFSET = 2;
+ const int STREAM_IS_INPUT_OFFSET = 3;
+ Vector<StreamConfiguration> scs;
+ if (mDeviceVersion < CAMERA_DEVICE_API_VERSION_3_2) {
+ ALOGE("StreamConfiguration is only valid after device HAL 3.2!");
+ return scs;
+ }
+
+ camera_metadata_ro_entry_t availableStreamConfigs =
+ staticInfo(ANDROID_SCALER_AVAILABLE_STREAM_CONFIGURATIONS);
+ for (size_t i=0; i < availableStreamConfigs.count; i+= STREAM_CONFIGURATION_SIZE) {
+ int32_t format = availableStreamConfigs.data.i32[i + STREAM_FORMAT_OFFSET];
+ int32_t width = availableStreamConfigs.data.i32[i + STREAM_WIDTH_OFFSET];
+ int32_t height = availableStreamConfigs.data.i32[i + STREAM_HEIGHT_OFFSET];
+ int32_t isInput = availableStreamConfigs.data.i32[i + STREAM_IS_INPUT_OFFSET];
+ StreamConfiguration sc = {format, width, height, isInput};
+ scs.add(sc);
+ }
+ return scs;
+}
+
+SortedVector<int32_t> Parameters::getAvailableOutputFormats() {
+ SortedVector<int32_t> outputFormats; // Non-duplicated output formats
+ if (mDeviceVersion >= CAMERA_DEVICE_API_VERSION_3_2) {
+ Vector<StreamConfiguration> scs = getStreamConfigurations();
+ for (size_t i=0; i < scs.size(); i++) {
+ const StreamConfiguration &sc = scs[i];
+ if (sc.isInput == ANDROID_SCALER_AVAILABLE_STREAM_CONFIGURATIONS_OUTPUT) {
+ outputFormats.add(sc.format);
+ }
+ }
+ } else {
+ camera_metadata_ro_entry_t availableFormats = staticInfo(ANDROID_SCALER_AVAILABLE_FORMATS);
+ for (size_t i=0; i < availableFormats.count; i++) {
+ outputFormats.add(availableFormats.data.i32[i]);
+ }
+ }
+ return outputFormats;
+}
+
+Vector<Parameters::Size> Parameters::getAvailableJpegSizes() {
+ Vector<Parameters::Size> jpegSizes;
+ if (mDeviceVersion >= CAMERA_DEVICE_API_VERSION_3_2) {
+ Vector<StreamConfiguration> scs = getStreamConfigurations();
+ for (size_t i=0; i < scs.size(); i++) {
+ const StreamConfiguration &sc = scs[i];
+ if (sc.isInput == ANDROID_SCALER_AVAILABLE_STREAM_CONFIGURATIONS_OUTPUT &&
+ sc.format == HAL_PIXEL_FORMAT_BLOB) {
+ Size sz = {sc.width, sc.height};
+ jpegSizes.add(sz);
+ }
+ }
+ } else {
+ const int JPEG_SIZE_ENTRY_COUNT = 2;
+ const int WIDTH_OFFSET = 0;
+ const int HEIGHT_OFFSET = 1;
+ camera_metadata_ro_entry_t availableJpegSizes =
+ staticInfo(ANDROID_SCALER_AVAILABLE_JPEG_SIZES);
+ for (size_t i=0; i < availableJpegSizes.count; i+= JPEG_SIZE_ENTRY_COUNT) {
+ int width = availableJpegSizes.data.i32[i + WIDTH_OFFSET];
+ int height = availableJpegSizes.data.i32[i + HEIGHT_OFFSET];
+ Size sz = {width, height};
+ jpegSizes.add(sz);
+ }
+ }
+ return jpegSizes;
+}
+
Parameters::CropRegion Parameters::calculateCropRegion(
Parameters::CropRegion::Outputs outputs) const {
diff --git a/services/camera/libcameraservice/api1/client2/Parameters.h b/services/camera/libcameraservice/api1/client2/Parameters.h
index 28dd788..f95c69a 100644
--- a/services/camera/libcameraservice/api1/client2/Parameters.h
+++ b/services/camera/libcameraservice/api1/client2/Parameters.h
@@ -226,7 +226,7 @@
~Parameters();
// Sets up default parameters
- status_t initialize(const CameraMetadata *info);
+ status_t initialize(const CameraMetadata *info, int deviceVersion);
// Build fast-access device static info from static info
status_t buildFastInfo();
@@ -346,6 +346,24 @@
status_t getFilteredSizes(Size limit, Vector<Size> *sizes);
// Get max size (from the size array) that matches the given aspect ratio.
Size getMaxSizeForRatio(float ratio, const int32_t* sizeArray, size_t count);
+
+ struct StreamConfiguration {
+ int32_t format;
+ int32_t width;
+ int32_t height;
+ int32_t isInput;
+ };
+ // Helper function extract available stream configuration
+ // Only valid since device HAL version 3.2
+ // returns an empty Vector if device HAL version does support it
+ Vector<StreamConfiguration> getStreamConfigurations();
+
+ // Helper function to get non-duplicated available output formats
+ SortedVector<int32_t> getAvailableOutputFormats();
+ // Helper function to get available output jpeg sizes
+ Vector<Size> getAvailableJpegSizes();
+
+ int mDeviceVersion;
};
// This class encapsulates the Parameters class so that it can only be accessed
diff --git a/services/camera/libcameraservice/api1/client2/StreamingProcessor.cpp b/services/camera/libcameraservice/api1/client2/StreamingProcessor.cpp
index 2064e2c..911f55a 100644
--- a/services/camera/libcameraservice/api1/client2/StreamingProcessor.cpp
+++ b/services/camera/libcameraservice/api1/client2/StreamingProcessor.cpp
@@ -89,8 +89,26 @@
Mutex::Autolock m(mMutex);
if (mPreviewRequest.entryCount() == 0) {
- res = device->createDefaultRequest(CAMERA2_TEMPLATE_PREVIEW,
- &mPreviewRequest);
+ sp<Camera2Client> client = mClient.promote();
+ if (client == 0) {
+ ALOGE("%s: Camera %d: Client does not exist", __FUNCTION__, mId);
+ return INVALID_OPERATION;
+ }
+
+ // Use CAMERA3_TEMPLATE_ZERO_SHUTTER_LAG for ZSL streaming case.
+ if (client->getCameraDeviceVersion() >= CAMERA_DEVICE_API_VERSION_3_0) {
+ if (params.zslMode && !params.recordingHint) {
+ res = device->createDefaultRequest(CAMERA3_TEMPLATE_ZERO_SHUTTER_LAG,
+ &mPreviewRequest);
+ } else {
+ res = device->createDefaultRequest(CAMERA3_TEMPLATE_PREVIEW,
+ &mPreviewRequest);
+ }
+ } else {
+ res = device->createDefaultRequest(CAMERA2_TEMPLATE_PREVIEW,
+ &mPreviewRequest);
+ }
+
if (res != OK) {
ALOGE("%s: Camera %d: Unable to create default preview request: "
"%s (%d)", __FUNCTION__, mId, strerror(-res), res);
@@ -430,10 +448,13 @@
Mutex::Autolock m(mMutex);
- // If a recording stream is being started up, free up any
- // outstanding buffers left from the previous recording session.
- // There should never be any, so if there are, warn about it.
- if (isStreamActive(outputStreams, mRecordingStreamId)) {
+ // If a recording stream is being started up and no recording
+ // stream is active yet, free up any outstanding buffers left
+ // from the previous recording session. There should never be
+ // any, so if there are, warn about it.
+ bool isRecordingStreamIdle = !isStreamActive(mActiveStreamIds, mRecordingStreamId);
+ bool startRecordingStream = isStreamActive(outputStreams, mRecordingStreamId);
+ if (startRecordingStream && isRecordingStreamIdle) {
releaseAllRecordingFramesLocked();
}
diff --git a/services/camera/libcameraservice/api1/client2/ZslProcessor.cpp b/services/camera/libcameraservice/api1/client2/ZslProcessor.cpp
index 2a2a5af..10463c1 100644
--- a/services/camera/libcameraservice/api1/client2/ZslProcessor.cpp
+++ b/services/camera/libcameraservice/api1/client2/ZslProcessor.cpp
@@ -202,7 +202,8 @@
}
client->registerFrameListener(Camera2Client::kPreviewRequestIdStart,
Camera2Client::kPreviewRequestIdEnd,
- this);
+ this,
+ /*sendPartials*/false);
return OK;
}
diff --git a/services/camera/libcameraservice/api1/client2/ZslProcessor3.cpp b/services/camera/libcameraservice/api1/client2/ZslProcessor3.cpp
index 1dcb718..79f75a5 100644
--- a/services/camera/libcameraservice/api1/client2/ZslProcessor3.cpp
+++ b/services/camera/libcameraservice/api1/client2/ZslProcessor3.cpp
@@ -52,8 +52,31 @@
mFrameListHead(0),
mZslQueueHead(0),
mZslQueueTail(0) {
- mZslQueue.insertAt(0, kZslBufferDepth);
- mFrameList.insertAt(0, kFrameListDepth);
+ // Initialize buffer queue and frame list based on pipeline max depth.
+ size_t pipelineMaxDepth = kDefaultMaxPipelineDepth;
+ if (client != 0) {
+ sp<Camera3Device> device =
+ static_cast<Camera3Device*>(client->getCameraDevice().get());
+ if (device != 0) {
+ camera_metadata_ro_entry_t entry =
+ device->info().find(ANDROID_REQUEST_PIPELINE_MAX_DEPTH);
+ if (entry.count == 1) {
+ pipelineMaxDepth = entry.data.u8[0];
+ } else {
+ ALOGW("%s: Unable to find the android.request.pipelineMaxDepth,"
+ " use default pipeline max depth %zu", __FUNCTION__,
+ kDefaultMaxPipelineDepth);
+ }
+ }
+ }
+
+ ALOGV("%s: Initialize buffer queue and frame list depth based on max pipeline depth (%d)",
+ __FUNCTION__, pipelineMaxDepth);
+ mBufferQueueDepth = pipelineMaxDepth + 1;
+ mFrameListDepth = pipelineMaxDepth + 1;
+
+ mZslQueue.insertAt(0, mBufferQueueDepth);
+ mFrameList.insertAt(0, mFrameListDepth);
sp<CaptureSequencer> captureSequencer = mSequencer.promote();
if (captureSequencer != 0) captureSequencer->setZslProcessor(this);
}
@@ -70,13 +93,25 @@
camera_metadata_ro_entry_t entry;
entry = result.mMetadata.find(ANDROID_SENSOR_TIMESTAMP);
nsecs_t timestamp = entry.data.i64[0];
+ if (entry.count == 0) {
+ ALOGE("%s: metadata doesn't have timestamp, skip this result");
+ return;
+ }
(void)timestamp;
- ALOGVV("Got preview metadata for timestamp %" PRId64, timestamp);
+
+ entry = result.mMetadata.find(ANDROID_REQUEST_FRAME_COUNT);
+ if (entry.count == 0) {
+ ALOGE("%s: metadata doesn't have frame number, skip this result");
+ return;
+ }
+ int32_t frameNumber = entry.data.i32[0];
+
+ ALOGVV("Got preview metadata for frame %d with timestamp %" PRId64, frameNumber, timestamp);
if (mState != RUNNING) return;
mFrameList.editItemAt(mFrameListHead) = result.mMetadata;
- mFrameListHead = (mFrameListHead + 1) % kFrameListDepth;
+ mFrameListHead = (mFrameListHead + 1) % mFrameListDepth;
}
status_t ZslProcessor3::updateStream(const Parameters ¶ms) {
@@ -136,7 +171,7 @@
// Note that format specified internally in Camera3ZslStream
res = device->createZslStream(
params.fastInfo.arrayWidth, params.fastInfo.arrayHeight,
- kZslBufferDepth,
+ mBufferQueueDepth,
&mZslStreamId,
&mZslStream);
if (res != OK) {
@@ -145,10 +180,15 @@
strerror(-res), res);
return res;
}
+
+ // Only add the camera3 buffer listener when the stream is created.
+ mZslStream->addBufferListener(this);
}
+
client->registerFrameListener(Camera2Client::kPreviewRequestIdStart,
Camera2Client::kPreviewRequestIdEnd,
- this);
+ this,
+ /*sendPartials*/false);
return OK;
}
@@ -250,18 +290,45 @@
uint8_t requestType = ANDROID_REQUEST_TYPE_REPROCESS;
res = request.update(ANDROID_REQUEST_TYPE,
&requestType, 1);
+ if (res != OK) {
+ ALOGE("%s: Unable to update request type",
+ __FUNCTION__);
+ return INVALID_OPERATION;
+ }
+
int32_t inputStreams[1] =
{ mZslStreamId };
- if (res == OK) request.update(ANDROID_REQUEST_INPUT_STREAMS,
+ res = request.update(ANDROID_REQUEST_INPUT_STREAMS,
inputStreams, 1);
+ if (res != OK) {
+ ALOGE("%s: Unable to update request input streams",
+ __FUNCTION__);
+ return INVALID_OPERATION;
+ }
+
+ uint8_t captureIntent =
+ static_cast<uint8_t>(ANDROID_CONTROL_CAPTURE_INTENT_STILL_CAPTURE);
+ res = request.update(ANDROID_CONTROL_CAPTURE_INTENT,
+ &captureIntent, 1);
+ if (res != OK ) {
+ ALOGE("%s: Unable to update request capture intent",
+ __FUNCTION__);
+ return INVALID_OPERATION;
+ }
+
// TODO: Shouldn't we also update the latest preview frame?
int32_t outputStreams[1] =
{ client->getCaptureStreamId() };
- if (res == OK) request.update(ANDROID_REQUEST_OUTPUT_STREAMS,
+ res = request.update(ANDROID_REQUEST_OUTPUT_STREAMS,
outputStreams, 1);
+ if (res != OK) {
+ ALOGE("%s: Unable to update request output streams",
+ __FUNCTION__);
+ return INVALID_OPERATION;
+ }
+
res = request.update(ANDROID_REQUEST_ID,
&requestId, 1);
-
if (res != OK ) {
ALOGE("%s: Unable to update frame to a reprocess request",
__FUNCTION__);
@@ -276,15 +343,6 @@
return INVALID_OPERATION;
}
- // Flush device to clear out all in-flight requests pending in HAL.
- res = client->getCameraDevice()->flush();
- if (res != OK) {
- ALOGE("%s: Camera %d: Failed to flush device: "
- "%s (%d)",
- __FUNCTION__, client->getCameraId(), strerror(-res), res);
- return res;
- }
-
// Update JPEG settings
{
SharedParameters::Lock l(client->getParameters());
@@ -322,11 +380,19 @@
status_t ZslProcessor3::clearZslQueueLocked() {
if (mZslStream != 0) {
+ // clear result metadata list first.
+ clearZslResultQueueLocked();
return mZslStream->clearInputRingBuffer();
}
return OK;
}
+void ZslProcessor3::clearZslResultQueueLocked() {
+ mFrameList.clear();
+ mFrameListHead = 0;
+ mFrameList.insertAt(0, mFrameListDepth);
+}
+
void ZslProcessor3::dump(int fd, const Vector<String16>& /*args*/) const {
Mutex::Autolock l(mInputMutex);
if (!mLatestCapturedRequest.isEmpty()) {
@@ -480,11 +546,17 @@
// We need to guarantee that if we do two back-to-back captures,
// the second won't use a buffer that's older/the same as the first, which
// is theoretically possible if we don't clear out the queue and the
- // selection criteria is something like 'newest'. Clearing out the queue
- // on a completed capture ensures we'll only use new data.
+ // selection criteria is something like 'newest'. Clearing out the result
+ // metadata queue on a completed capture ensures we'll only use new timestamp.
+ // Calling clearZslQueueLocked is a guaranteed deadlock because this callback
+ // holds the Camera3Stream internal lock (mLock), and clearZslQueueLocked requires
+ // to hold the same lock.
+ // TODO: need figure out a way to clear the Zsl buffer queue properly. Right now
+ // it is safe not to do so, as back to back ZSL capture requires stop and start
+ // preview, which will flush ZSL queue automatically.
ALOGV("%s: Memory optimization, clearing ZSL queue",
__FUNCTION__);
- clearZslQueueLocked();
+ clearZslResultQueueLocked();
// Required so we accept more ZSL requests
mState = RUNNING;
diff --git a/services/camera/libcameraservice/api1/client2/ZslProcessor3.h b/services/camera/libcameraservice/api1/client2/ZslProcessor3.h
index 4c52a64..dfb1457 100644
--- a/services/camera/libcameraservice/api1/client2/ZslProcessor3.h
+++ b/services/camera/libcameraservice/api1/client2/ZslProcessor3.h
@@ -107,8 +107,9 @@
CameraMetadata frame;
};
- static const size_t kZslBufferDepth = 4;
- static const size_t kFrameListDepth = kZslBufferDepth * 2;
+ static const int32_t kDefaultMaxPipelineDepth = 4;
+ size_t mBufferQueueDepth;
+ size_t mFrameListDepth;
Vector<CameraMetadata> mFrameList;
size_t mFrameListHead;
@@ -124,6 +125,8 @@
status_t clearZslQueueLocked();
+ void clearZslResultQueueLocked();
+
void dumpZslQueue(int id) const;
nsecs_t getCandidateTimestampLocked(size_t* metadataIdx) const;
diff --git a/services/camera/libcameraservice/api2/CameraDeviceClient.cpp b/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
index 8154e59..de42cee 100644
--- a/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
+++ b/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
@@ -82,7 +82,7 @@
mFrameProcessor->registerListener(FRAME_PROCESSOR_LISTENER_MIN_ID,
FRAME_PROCESSOR_LISTENER_MAX_ID,
/*listener*/this,
- /*quirkSendPartials*/true);
+ /*sendPartials*/true);
return OK;
}
@@ -102,7 +102,7 @@
status_t CameraDeviceClient::submitRequestList(List<sp<CaptureRequest> > requests,
bool streaming, int64_t* lastFrameNumber) {
ATRACE_CALL();
- ALOGV("%s-start of function. Request list size %d", __FUNCTION__, requests.size());
+ ALOGV("%s-start of function. Request list size %zu", __FUNCTION__, requests.size());
status_t res;
if ( (res = checkPid(__FUNCTION__) ) != OK) return res;
@@ -177,7 +177,7 @@
metadata.update(ANDROID_REQUEST_ID, &requestId, /*size*/1);
loopCounter++; // loopCounter starts from 1
- ALOGV("%s: Camera %d: Creating request with ID %d (%d of %d)",
+ ALOGV("%s: Camera %d: Creating request with ID %d (%d of %zu)",
__FUNCTION__, mCameraId, requestId, loopCounter, requests.size());
metadataRequestList.push_back(metadata);
diff --git a/services/camera/libcameraservice/common/Camera2ClientBase.cpp b/services/camera/libcameraservice/common/Camera2ClientBase.cpp
index 19efd30..13c9f48 100644
--- a/services/camera/libcameraservice/common/Camera2ClientBase.cpp
+++ b/services/camera/libcameraservice/common/Camera2ClientBase.cpp
@@ -54,7 +54,8 @@
int servicePid):
TClientBase(cameraService, remoteCallback, clientPackageName,
cameraId, cameraFacing, clientPid, clientUid, servicePid),
- mSharedCameraCallbacks(remoteCallback)
+ mSharedCameraCallbacks(remoteCallback),
+ mDeviceVersion(cameraService->getDeviceVersion(cameraId))
{
ALOGI("Camera %d: Opened", cameraId);
@@ -280,6 +281,11 @@
}
template <typename TClientBase>
+int Camera2ClientBase<TClientBase>::getCameraDeviceVersion() const {
+ return mDeviceVersion;
+}
+
+template <typename TClientBase>
const sp<CameraDeviceBase>& Camera2ClientBase<TClientBase>::getCameraDevice() {
return mDevice;
}
diff --git a/services/camera/libcameraservice/common/Camera2ClientBase.h b/services/camera/libcameraservice/common/Camera2ClientBase.h
index 9feca93..f57d204 100644
--- a/services/camera/libcameraservice/common/Camera2ClientBase.h
+++ b/services/camera/libcameraservice/common/Camera2ClientBase.h
@@ -76,6 +76,7 @@
int getCameraId() const;
const sp<CameraDeviceBase>&
getCameraDevice();
+ int getCameraDeviceVersion() const;
const sp<CameraService>&
getCameraService();
@@ -122,6 +123,7 @@
/** CameraDeviceBase instance wrapping HAL2+ entry */
+ const int mDeviceVersion;
sp<CameraDeviceBase> mDevice;
/** Utility members */
diff --git a/services/camera/libcameraservice/common/CameraDeviceBase.h b/services/camera/libcameraservice/common/CameraDeviceBase.h
index 7597b10..c7bd886 100644
--- a/services/camera/libcameraservice/common/CameraDeviceBase.h
+++ b/services/camera/libcameraservice/common/CameraDeviceBase.h
@@ -26,6 +26,7 @@
#include <camera/camera2/ICameraDeviceCallbacks.h>
#include "hardware/camera2.h"
+#include "hardware/camera3.h"
#include "camera/CameraMetadata.h"
#include "camera/CaptureResult.h"
diff --git a/services/camera/libcameraservice/common/FrameProcessorBase.cpp b/services/camera/libcameraservice/common/FrameProcessorBase.cpp
index f6a971a..482f687 100644
--- a/services/camera/libcameraservice/common/FrameProcessorBase.cpp
+++ b/services/camera/libcameraservice/common/FrameProcessorBase.cpp
@@ -37,11 +37,23 @@
}
status_t FrameProcessorBase::registerListener(int32_t minId,
- int32_t maxId, wp<FilteredListener> listener, bool quirkSendPartials) {
+ int32_t maxId, wp<FilteredListener> listener, bool sendPartials) {
Mutex::Autolock l(mInputMutex);
+ List<RangeListener>::iterator item = mRangeListeners.begin();
+ while (item != mRangeListeners.end()) {
+ if (item->minId == minId &&
+ item->maxId == maxId &&
+ item->listener == listener) {
+ // already registered, just return
+ ALOGV("%s: Attempt to register the same client twice, ignoring",
+ __FUNCTION__);
+ return OK;
+ }
+ item++;
+ }
ALOGV("%s: Registering listener for frame id range %d - %d",
__FUNCTION__, minId, maxId);
- RangeListener rListener = { minId, maxId, listener, quirkSendPartials };
+ RangeListener rListener = { minId, maxId, listener, sendPartials };
mRangeListeners.push_back(rListener);
return OK;
}
@@ -176,7 +188,7 @@
List<RangeListener>::iterator item = mRangeListeners.begin();
while (item != mRangeListeners.end()) {
if (requestId >= item->minId && requestId < item->maxId &&
- (!quirkIsPartial || item->quirkSendPartials)) {
+ (!quirkIsPartial || item->sendPartials)) {
sp<FilteredListener> listener = item->listener.promote();
if (listener == 0) {
item = mRangeListeners.erase(item);
diff --git a/services/camera/libcameraservice/common/FrameProcessorBase.h b/services/camera/libcameraservice/common/FrameProcessorBase.h
index 15a014e..3649c45 100644
--- a/services/camera/libcameraservice/common/FrameProcessorBase.h
+++ b/services/camera/libcameraservice/common/FrameProcessorBase.h
@@ -44,11 +44,12 @@
};
// Register a listener for a range of IDs [minId, maxId). Multiple listeners
- // can be listening to the same range.
- // QUIRK: sendPartials controls whether partial results will be sent.
+ // can be listening to the same range. Registering the same listener with
+ // the same range of IDs has no effect.
+ // sendPartials controls whether partial results will be sent.
status_t registerListener(int32_t minId, int32_t maxId,
wp<FilteredListener> listener,
- bool quirkSendPartials = true);
+ bool sendPartials = true);
status_t removeListener(int32_t minId, int32_t maxId,
wp<FilteredListener> listener);
@@ -66,7 +67,7 @@
int32_t minId;
int32_t maxId;
wp<FilteredListener> listener;
- bool quirkSendPartials;
+ bool sendPartials;
};
List<RangeListener> mRangeListeners;
diff --git a/services/camera/libcameraservice/device1/CameraHardwareInterface.h b/services/camera/libcameraservice/device1/CameraHardwareInterface.h
index 87b2807..2746f6f 100644
--- a/services/camera/libcameraservice/device1/CameraHardwareInterface.h
+++ b/services/camera/libcameraservice/device1/CameraHardwareInterface.h
@@ -92,8 +92,22 @@
status_t initialize(hw_module_t *module)
{
ALOGI("Opening camera %s", mName.string());
- int rc = module->methods->open(module, mName.string(),
- (hw_device_t **)&mDevice);
+ camera_module_t *cameraModule = reinterpret_cast<camera_module_t *>(module);
+ camera_info info;
+ status_t res = cameraModule->get_camera_info(atoi(mName.string()), &info);
+ if (res != OK) return res;
+
+ int rc = OK;
+ if (module->module_api_version >= CAMERA_MODULE_API_VERSION_2_3 &&
+ info.device_version > CAMERA_DEVICE_API_VERSION_1_0) {
+ // Open higher version camera device as HAL1.0 device.
+ rc = cameraModule->open_legacy(module, mName.string(),
+ CAMERA_DEVICE_API_VERSION_1_0,
+ (hw_device_t **)&mDevice);
+ } else {
+ rc = module->methods->open(module, mName.string(),
+ (hw_device_t **)&mDevice);
+ }
if (rc != OK) {
ALOGE("Could not open camera %s: %d", mName.string(), rc);
return rc;
@@ -611,9 +625,14 @@
static int __set_buffers_geometry(struct preview_stream_ops* w,
int width, int height, int format)
{
+ int rc;
ANativeWindow *a = anw(w);
- return native_window_set_buffers_geometry(a,
- width, height, format);
+
+ rc = native_window_set_buffers_dimensions(a, width, height);
+ if (!rc) {
+ rc = native_window_set_buffers_format(a, format);
+ }
+ return rc;
}
static int __set_crop(struct preview_stream_ops *w,
diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index 16d6f42..6ceb9d4 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -113,7 +113,6 @@
}
/** Cross-check device version */
-
if (device->common.version < CAMERA_DEVICE_API_VERSION_3_0) {
SET_ERR_L("Could not open camera: "
"Camera device should be at least %x, reports %x instead",
@@ -173,6 +172,7 @@
/** Everything is good to go */
+ mDeviceVersion = device->common.version;
mDeviceInfo = info.static_camera_characteristics;
mHal3Device = device;
mStatus = STATUS_UNCONFIGURED;
@@ -284,42 +284,74 @@
return gotLock;
}
+Camera3Device::Size Camera3Device::getMaxJpegResolution() const {
+ int32_t maxJpegWidth = 0, maxJpegHeight = 0;
+ if (mDeviceVersion >= CAMERA_DEVICE_API_VERSION_3_2) {
+ const int STREAM_CONFIGURATION_SIZE = 4;
+ const int STREAM_FORMAT_OFFSET = 0;
+ const int STREAM_WIDTH_OFFSET = 1;
+ const int STREAM_HEIGHT_OFFSET = 2;
+ const int STREAM_IS_INPUT_OFFSET = 3;
+ camera_metadata_ro_entry_t availableStreamConfigs =
+ mDeviceInfo.find(ANDROID_SCALER_AVAILABLE_STREAM_CONFIGURATIONS);
+ if (availableStreamConfigs.count == 0 ||
+ availableStreamConfigs.count % STREAM_CONFIGURATION_SIZE != 0) {
+ return Size(0, 0);
+ }
+
+ // Get max jpeg size (area-wise).
+ for (size_t i=0; i < availableStreamConfigs.count; i+= STREAM_CONFIGURATION_SIZE) {
+ int32_t format = availableStreamConfigs.data.i32[i + STREAM_FORMAT_OFFSET];
+ int32_t width = availableStreamConfigs.data.i32[i + STREAM_WIDTH_OFFSET];
+ int32_t height = availableStreamConfigs.data.i32[i + STREAM_HEIGHT_OFFSET];
+ int32_t isInput = availableStreamConfigs.data.i32[i + STREAM_IS_INPUT_OFFSET];
+ if (isInput == ANDROID_SCALER_AVAILABLE_STREAM_CONFIGURATIONS_OUTPUT
+ && format == HAL_PIXEL_FORMAT_BLOB &&
+ (width * height > maxJpegWidth * maxJpegHeight)) {
+ maxJpegWidth = width;
+ maxJpegHeight = height;
+ }
+ }
+ } else {
+ camera_metadata_ro_entry availableJpegSizes =
+ mDeviceInfo.find(ANDROID_SCALER_AVAILABLE_JPEG_SIZES);
+ if (availableJpegSizes.count == 0 || availableJpegSizes.count % 2 != 0) {
+ return Size(0, 0);
+ }
+
+ // Get max jpeg size (area-wise).
+ for (size_t i = 0; i < availableJpegSizes.count; i += 2) {
+ if ((availableJpegSizes.data.i32[i] * availableJpegSizes.data.i32[i + 1])
+ > (maxJpegWidth * maxJpegHeight)) {
+ maxJpegWidth = availableJpegSizes.data.i32[i];
+ maxJpegHeight = availableJpegSizes.data.i32[i + 1];
+ }
+ }
+ }
+ return Size(maxJpegWidth, maxJpegHeight);
+}
+
ssize_t Camera3Device::getJpegBufferSize(uint32_t width, uint32_t height) const {
- // TODO: replace below with availableStreamConfiguration for HAL3.2+.
- camera_metadata_ro_entry availableJpegSizes =
- mDeviceInfo.find(ANDROID_SCALER_AVAILABLE_JPEG_SIZES);
- if (availableJpegSizes.count == 0 || availableJpegSizes.count % 2 != 0) {
+ // Get max jpeg size (area-wise).
+ Size maxJpegResolution = getMaxJpegResolution();
+ if (maxJpegResolution.width == 0) {
ALOGE("%s: Camera %d: Can't find find valid available jpeg sizes in static metadata!",
__FUNCTION__, mId);
return BAD_VALUE;
}
- // Get max jpeg size (area-wise).
- int32_t maxJpegWidth = 0, maxJpegHeight = 0;
- bool foundMax = false;
- for (size_t i = 0; i < availableJpegSizes.count; i += 2) {
- if ((availableJpegSizes.data.i32[i] * availableJpegSizes.data.i32[i + 1])
- > (maxJpegWidth * maxJpegHeight)) {
- maxJpegWidth = availableJpegSizes.data.i32[i];
- maxJpegHeight = availableJpegSizes.data.i32[i + 1];
- foundMax = true;
- }
- }
- if (!foundMax) {
- return BAD_VALUE;
- }
-
// Get max jpeg buffer size
ssize_t maxJpegBufferSize = 0;
- camera_metadata_ro_entry jpegMaxSize = mDeviceInfo.find(ANDROID_JPEG_MAX_SIZE);
- if (jpegMaxSize.count == 0) {
+ camera_metadata_ro_entry jpegBufMaxSize = mDeviceInfo.find(ANDROID_JPEG_MAX_SIZE);
+ if (jpegBufMaxSize.count == 0) {
ALOGE("%s: Camera %d: Can't find maximum JPEG size in static metadata!", __FUNCTION__, mId);
return BAD_VALUE;
}
- maxJpegBufferSize = jpegMaxSize.data.i32[0];
+ maxJpegBufferSize = jpegBufMaxSize.data.i32[0];
// Calculate final jpeg buffer size for the given resolution.
- float scaleFactor = ((float) (width * height)) / (maxJpegWidth * maxJpegHeight);
+ float scaleFactor = ((float) (width * height)) /
+ (maxJpegResolution.width * maxJpegResolution.height);
ssize_t jpegBufferSize = scaleFactor * maxJpegBufferSize;
// Bound the buffer size to [MIN_JPEG_BUFFER_SIZE, maxJpegBufferSize].
if (jpegBufferSize > maxJpegBufferSize) {
@@ -1156,7 +1188,7 @@
{
ANDROID_CONTROL_AF_TRIGGER_ID,
static_cast<int32_t>(id)
- },
+ }
};
return mRequestThread->queueTrigger(trigger,
@@ -1177,7 +1209,7 @@
{
ANDROID_CONTROL_AF_TRIGGER_ID,
static_cast<int32_t>(id)
- },
+ }
};
return mRequestThread->queueTrigger(trigger,
@@ -1198,7 +1230,7 @@
{
ANDROID_CONTROL_AE_PRECAPTURE_ID,
static_cast<int32_t>(id)
- },
+ }
};
return mRequestThread->queueTrigger(trigger,
@@ -1501,12 +1533,12 @@
*/
status_t Camera3Device::registerInFlight(uint32_t frameNumber,
- int32_t numBuffers, CaptureResultExtras resultExtras) {
+ int32_t numBuffers, CaptureResultExtras resultExtras, bool hasInput) {
ATRACE_CALL();
Mutex::Autolock l(mInFlightLock);
ssize_t res;
- res = mInFlightMap.add(frameNumber, InFlightRequest(numBuffers, resultExtras));
+ res = mInFlightMap.add(frameNumber, InFlightRequest(numBuffers, resultExtras, hasInput));
if (res < 0) return res;
return OK;
@@ -1539,8 +1571,6 @@
uint8_t aeState;
uint8_t afState;
uint8_t awbState;
- int32_t afTriggerId;
- int32_t aeTriggerId;
gotAllStates &= get3AResult(partial, ANDROID_CONTROL_AF_MODE,
&afMode, frameNumber);
@@ -1557,12 +1587,6 @@
gotAllStates &= get3AResult(partial, ANDROID_CONTROL_AWB_STATE,
&awbState, frameNumber);
- gotAllStates &= get3AResult(partial, ANDROID_CONTROL_AF_TRIGGER_ID,
- &afTriggerId, frameNumber);
-
- gotAllStates &= get3AResult(partial, ANDROID_CONTROL_AE_PRECAPTURE_ID,
- &aeTriggerId, frameNumber);
-
if (!gotAllStates) return false;
ALOGVV("%s: Camera %d: Frame %d, Request ID %d: AF mode %d, AWB mode %d, "
@@ -1571,7 +1595,7 @@
__FUNCTION__, mId, frameNumber, resultExtras.requestId,
afMode, awbMode,
afState, aeState, awbState,
- afTriggerId, aeTriggerId);
+ resultExtras.afTriggerId, resultExtras.precaptureTriggerId);
// Got all states, so construct a minimal result to send
// In addition to the above fields, this means adding in
@@ -1635,12 +1659,12 @@
}
if (!insert3AResult(min3AResult.mMetadata, ANDROID_CONTROL_AF_TRIGGER_ID,
- &afTriggerId, frameNumber)) {
+ &resultExtras.afTriggerId, frameNumber)) {
return false;
}
if (!insert3AResult(min3AResult.mMetadata, ANDROID_CONTROL_AE_PRECAPTURE_ID,
- &aeTriggerId, frameNumber)) {
+ &resultExtras.precaptureTriggerId, frameNumber)) {
return false;
}
@@ -1696,7 +1720,8 @@
status_t res;
uint32_t frameNumber = result->frame_number;
- if (result->result == NULL && result->num_output_buffers == 0) {
+ if (result->result == NULL && result->num_output_buffers == 0 &&
+ result->input_buffer == NULL) {
SET_ERR("No result data provided by HAL for frame %d",
frameNumber);
return;
@@ -1704,6 +1729,7 @@
bool partialResultQuirk = false;
CameraMetadata collectedQuirkResult;
CaptureResultExtras resultExtras;
+ bool hasInputBufferInRequest = false;
// Get capture timestamp and resultExtras from list of in-flight requests,
// where it was added by the shutter notification for this frame.
@@ -1752,6 +1778,7 @@
timestamp = request.captureTimestamp;
resultExtras = request.resultExtras;
+ hasInputBufferInRequest = request.hasInputBuffer;
/**
* One of the following must happen before it's legal to call process_capture_result,
@@ -1780,8 +1807,17 @@
request.haveResultMetadata = true;
}
- request.numBuffersLeft -= result->num_output_buffers;
-
+ uint32_t numBuffersReturned = result->num_output_buffers;
+ if (result->input_buffer != NULL) {
+ if (hasInputBufferInRequest) {
+ numBuffersReturned += 1;
+ } else {
+ ALOGW("%s: Input buffer should be NULL if there is no input"
+ " buffer sent in the request",
+ __FUNCTION__);
+ }
+ }
+ request.numBuffersLeft -= numBuffersReturned;
if (request.numBuffersLeft < 0) {
SET_ERR("Too many buffers returned for frame %d",
frameNumber);
@@ -1882,6 +1918,25 @@
}
}
+ if (result->input_buffer != NULL) {
+ if (hasInputBufferInRequest) {
+ Camera3Stream *stream =
+ Camera3Stream::cast(result->input_buffer->stream);
+ res = stream->returnInputBuffer(*(result->input_buffer));
+ // Note: stream may be deallocated at this point, if this buffer was the
+ // last reference to it.
+ if (res != OK) {
+ ALOGE("%s: RequestThread: Can't return input buffer for frame %d to"
+ " its stream:%s (%d)", __FUNCTION__,
+ frameNumber, strerror(-res), res);
+ }
+ } else {
+ ALOGW("%s: Input buffer should be NULL if there is no input"
+ " buffer sent in the request, skipping input buffer return.",
+ __FUNCTION__);
+ }
+ }
+
// Finally, signal any waiters for new frames
if (gotResult) {
@@ -2126,6 +2181,17 @@
return OK;
}
+bool Camera3Device::RequestThread::isRepeatingRequestLocked(const sp<CaptureRequest> requestIn) {
+ if (mRepeatingRequests.empty()) {
+ return false;
+ }
+ int32_t requestId = requestIn->mResultExtras.requestId;
+ const RequestList &repeatRequests = mRepeatingRequests;
+ // All repeating requests are guaranteed to have same id so only check first quest
+ const sp<CaptureRequest> firstRequest = *repeatRequests.begin();
+ return (firstRequest->mResultExtras.requestId == requestId);
+}
+
status_t Camera3Device::RequestThread::clearRepeatingRequests(/*out*/int64_t *lastFrameNumber) {
Mutex::Autolock l(mRequestLock);
mRepeatingRequests.clear();
@@ -2140,6 +2206,18 @@
Mutex::Autolock l(mRequestLock);
ALOGV("RequestThread::%s:", __FUNCTION__);
mRepeatingRequests.clear();
+
+ // Decrement repeating frame count for those requests never sent to device
+ // TODO: Remove this after we have proper error handling so these requests
+ // will generate an error callback. This might be the only place calling
+ // isRepeatingRequestLocked. If so, isRepeatingRequestLocked should also be removed.
+ const RequestList &requests = mRequestQueue;
+ for (RequestList::const_iterator it = requests.begin();
+ it != requests.end(); ++it) {
+ if (isRepeatingRequestLocked(*it)) {
+ mRepeatingLastFrameNumber--;
+ }
+ }
mRequestQueue.clear();
mTriggerMap.clear();
if (lastFrameNumber != NULL) {
@@ -2271,6 +2349,7 @@
}
camera3_stream_buffer_t inputBuffer;
+ uint32_t totalNumBuffers = 0;
// Fill in buffers
@@ -2283,6 +2362,7 @@
cleanUpFailedRequest(request, nextRequest, outputBuffers);
return true;
}
+ totalNumBuffers += 1;
} else {
request.input_buffer = NULL;
}
@@ -2301,6 +2381,7 @@
}
request.num_output_buffers++;
}
+ totalNumBuffers += request.num_output_buffers;
// Log request in the in-flight queue
sp<Camera3Device> parent = mParent.promote();
@@ -2311,7 +2392,8 @@
}
res = parent->registerInFlight(request.frame_number,
- request.num_output_buffers, nextRequest->mResultExtras);
+ totalNumBuffers, nextRequest->mResultExtras,
+ /*hasInput*/request.input_buffer != NULL);
ALOGVV("%s: registered in flight requestId = %" PRId32 ", frameNumber = %" PRId64
", burstId = %" PRId32 ".",
__FUNCTION__,
@@ -2367,21 +2449,6 @@
}
mPrevTriggers = triggerCount;
- // Return input buffer back to framework
- if (request.input_buffer != NULL) {
- Camera3Stream *stream =
- Camera3Stream::cast(request.input_buffer->stream);
- res = stream->returnInputBuffer(*(request.input_buffer));
- // Note: stream may be deallocated at this point, if this buffer was the
- // last reference to it.
- if (res != OK) {
- ALOGE("%s: RequestThread: Can't return input buffer for frame %d to"
- " its stream:%s (%d)", __FUNCTION__,
- request.frame_number, strerror(-res), res);
- // TODO: Report error upstream
- }
- }
-
return true;
}
@@ -2554,13 +2621,29 @@
Mutex::Autolock al(mTriggerMutex);
+ sp<Camera3Device> parent = mParent.promote();
+ if (parent == NULL) {
+ CLOGE("RequestThread: Parent is gone");
+ return DEAD_OBJECT;
+ }
+
CameraMetadata &metadata = request->mSettings;
size_t count = mTriggerMap.size();
for (size_t i = 0; i < count; ++i) {
RequestTrigger trigger = mTriggerMap.valueAt(i);
-
uint32_t tag = trigger.metadataTag;
+
+ if (tag == ANDROID_CONTROL_AF_TRIGGER_ID || tag == ANDROID_CONTROL_AE_PRECAPTURE_ID) {
+ bool isAeTrigger = (trigger.metadataTag == ANDROID_CONTROL_AE_PRECAPTURE_ID);
+ uint32_t triggerId = static_cast<uint32_t>(trigger.entryValue);
+ isAeTrigger ? request->mResultExtras.precaptureTriggerId = triggerId :
+ request->mResultExtras.afTriggerId = triggerId;
+ if (parent->mDeviceVersion >= CAMERA_DEVICE_API_VERSION_3_2) {
+ continue; // Trigger ID tag is deprecated since device HAL 3.2
+ }
+ }
+
camera_metadata_entry entry = metadata.find(tag);
if (entry.count > 0) {
diff --git a/services/camera/libcameraservice/device3/Camera3Device.h b/services/camera/libcameraservice/device3/Camera3Device.h
index 00ae771..ea958b7 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.h
+++ b/services/camera/libcameraservice/device3/Camera3Device.h
@@ -168,6 +168,8 @@
CameraMetadata mDeviceInfo;
+ int mDeviceVersion;
+
enum Status {
STATUS_ERROR,
STATUS_UNINITIALIZED,
@@ -297,6 +299,18 @@
*/
bool tryLockSpinRightRound(Mutex& lock);
+ struct Size {
+ int width;
+ int height;
+ Size(int w, int h) : width(w), height(h){}
+ };
+
+ /**
+ * Helper function to get the largest Jpeg resolution (in area)
+ * Return Size(0, 0) if static metatdata is invalid
+ */
+ Size getMaxJpegResolution() const;
+
/**
* Get Jpeg buffer size for a given jpeg resolution.
* Negative values are error codes.
@@ -430,6 +444,9 @@
// Relay error to parent device object setErrorState
void setErrorState(const char *fmt, ...);
+ // If the input request is in mRepeatingRequests. Must be called with mRequestLock hold
+ bool isRepeatingRequestLocked(const sp<CaptureRequest>);
+
wp<Camera3Device> mParent;
wp<camera3::StatusTracker> mStatusTracker;
camera3_device_t *mHal3Device;
@@ -484,9 +501,11 @@
// Set by process_capture_result call with valid metadata
bool haveResultMetadata;
// Decremented by calls to process_capture_result with valid output
- // buffers
+ // and input buffers
int numBuffersLeft;
CaptureResultExtras resultExtras;
+ // If this request has any input buffer
+ bool hasInputBuffer;
// Fields used by the partial result quirk only
struct PartialResultQuirkInFlight {
@@ -505,14 +524,16 @@
captureTimestamp(0),
requestStatus(OK),
haveResultMetadata(false),
- numBuffersLeft(0) {
+ numBuffersLeft(0),
+ hasInputBuffer(false){
}
InFlightRequest(int numBuffers) :
captureTimestamp(0),
requestStatus(OK),
haveResultMetadata(false),
- numBuffersLeft(numBuffers) {
+ numBuffersLeft(numBuffers),
+ hasInputBuffer(false){
}
InFlightRequest(int numBuffers, CaptureResultExtras extras) :
@@ -520,9 +541,19 @@
requestStatus(OK),
haveResultMetadata(false),
numBuffersLeft(numBuffers),
- resultExtras(extras) {
+ resultExtras(extras),
+ hasInputBuffer(false){
}
- };
+
+ InFlightRequest(int numBuffers, CaptureResultExtras extras, bool hasInput) :
+ captureTimestamp(0),
+ requestStatus(OK),
+ haveResultMetadata(false),
+ numBuffersLeft(numBuffers),
+ resultExtras(extras),
+ hasInputBuffer(hasInput){
+ }
+};
// Map from frame number to the in-flight request state
typedef KeyedVector<uint32_t, InFlightRequest> InFlightMap;
@@ -530,7 +561,7 @@
InFlightMap mInFlightMap;
status_t registerInFlight(uint32_t frameNumber,
- int32_t numBuffers, CaptureResultExtras resultExtras);
+ int32_t numBuffers, CaptureResultExtras resultExtras, bool hasInput);
/**
* For the partial result quirk, check if all 3A state fields are available
diff --git a/services/camera/libcameraservice/device3/Camera3Stream.cpp b/services/camera/libcameraservice/device3/Camera3Stream.cpp
index 7645a2a..d7b1871 100644
--- a/services/camera/libcameraservice/device3/Camera3Stream.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Stream.cpp
@@ -485,6 +485,18 @@
void Camera3Stream::addBufferListener(
wp<Camera3StreamBufferListener> listener) {
Mutex::Autolock l(mLock);
+
+ List<wp<Camera3StreamBufferListener> >::iterator it, end;
+ for (it = mBufferListenerList.begin(), end = mBufferListenerList.end();
+ it != end;
+ ) {
+ if (*it == listener) {
+ ALOGE("%s: Try to add the same listener twice, ignoring...", __FUNCTION__);
+ return;
+ }
+ it++;
+ }
+
mBufferListenerList.push_back(listener);
}
diff --git a/services/camera/libcameraservice/device3/Camera3Stream.h b/services/camera/libcameraservice/device3/Camera3Stream.h
index 14f5387..a77f27c 100644
--- a/services/camera/libcameraservice/device3/Camera3Stream.h
+++ b/services/camera/libcameraservice/device3/Camera3Stream.h
@@ -226,8 +226,17 @@
*/
virtual void dump(int fd, const Vector<String16> &args) const = 0;
+ /**
+ * Add a camera3 buffer listener. Adding the same listener twice has
+ * no effect.
+ */
void addBufferListener(
wp<Camera3StreamBufferListener> listener);
+
+ /**
+ * Remove a camera3 buffer listener. Removing the same listener twice
+ * or the listener that was never added has no effect.
+ */
void removeBufferListener(
const sp<Camera3StreamBufferListener>& listener);
diff --git a/services/camera/libcameraservice/device3/Camera3ZslStream.cpp b/services/camera/libcameraservice/device3/Camera3ZslStream.cpp
index 05b3d1f..6c298f9 100644
--- a/services/camera/libcameraservice/device3/Camera3ZslStream.cpp
+++ b/services/camera/libcameraservice/device3/Camera3ZslStream.cpp
@@ -300,6 +300,7 @@
nsecs_t actual = pinnedBuffer->getBufferItem().mTimestamp;
if (actual != timestamp) {
+ // TODO: this is problematic, we'll end up with using wrong result for this pinned buffer.
ALOGW("%s: ZSL buffer candidate search didn't find an exact match --"
" requested timestamp = %" PRId64 ", actual timestamp = %" PRId64,
__FUNCTION__, timestamp, actual);
diff --git a/services/soundtrigger/SoundTriggerHwService.cpp b/services/soundtrigger/SoundTriggerHwService.cpp
index fa59388..747af79 100644
--- a/services/soundtrigger/SoundTriggerHwService.cpp
+++ b/services/soundtrigger/SoundTriggerHwService.cpp
@@ -377,7 +377,7 @@
}
status_t SoundTriggerHwService::Module::startRecognition(sound_model_handle_t handle,
- const sp<IMemory>& dataMemory)
+ const sp<IMemory>& dataMemory)
{
ALOGV("startRecognition() model handle %d", handle);
@@ -391,28 +391,25 @@
if (model == 0) {
return BAD_VALUE;
}
+ if ((dataMemory == 0) ||
+ (dataMemory->size() < sizeof(struct sound_trigger_recognition_config))) {
+ return BAD_VALUE;
+ }
if (model->mState == Model::STATE_ACTIVE) {
return INVALID_OPERATION;
}
model->mState = Model::STATE_ACTIVE;
- char *data = NULL;
- unsigned int data_size = 0;
- if (dataMemory != 0 && dataMemory->size() != 0) {
- data_size = (unsigned int)dataMemory->size();
- data = (char *)dataMemory->pointer();
- ALOGV("startRecognition() data size %d data %d - %d",
- data_size, data[0], data[data_size - 1]);
- }
+ struct sound_trigger_recognition_config *config =
+ (struct sound_trigger_recognition_config *)dataMemory->pointer();
//TODO: get capture handle and device from audio policy service
- audio_io_handle_t capture_handle = 0;
- return mHwDevice->start_recognition(mHwDevice, handle, capture_handle, AUDIO_DEVICE_NONE,
+ config->capture_handle = AUDIO_IO_HANDLE_NONE;
+ config->capture_device = AUDIO_DEVICE_NONE;
+ return mHwDevice->start_recognition(mHwDevice, handle, config,
SoundTriggerHwService::recognitionCallback,
- this,
- data_size,
- data);
+ this);
}
status_t SoundTriggerHwService::Module::stopRecognition(sound_model_handle_t handle)