Merge "Explicit define getFrameAtIndex as pure virtual function."
diff --git a/media/libaudioclient/IAudioFlinger.cpp b/media/libaudioclient/IAudioFlinger.cpp
index 5db60f3..56ddd4f 100644
--- a/media/libaudioclient/IAudioFlinger.cpp
+++ b/media/libaudioclient/IAudioFlinger.cpp
@@ -22,6 +22,7 @@
#include <stdint.h>
#include <sys/types.h>
+#include <binder/IPCThreadState.h>
#include <binder/Parcel.h>
#include "IAudioFlinger.h"
@@ -836,6 +837,35 @@
status_t BnAudioFlinger::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
+ // make sure transactions reserved to AudioPolicyManager do not come from other processes
+ switch (code) {
+ case SET_STREAM_VOLUME:
+ case SET_STREAM_MUTE:
+ case SET_MODE:
+ case OPEN_OUTPUT:
+ case OPEN_DUPLICATE_OUTPUT:
+ case CLOSE_OUTPUT:
+ case SUSPEND_OUTPUT:
+ case RESTORE_OUTPUT:
+ case OPEN_INPUT:
+ case CLOSE_INPUT:
+ case INVALIDATE_STREAM:
+ case SET_VOICE_VOLUME:
+ case MOVE_EFFECTS:
+ case LOAD_HW_MODULE:
+ case LIST_AUDIO_PORTS:
+ case GET_AUDIO_PORT:
+ case CREATE_AUDIO_PATCH:
+ case RELEASE_AUDIO_PATCH:
+ case LIST_AUDIO_PATCHES:
+ case SET_AUDIO_PORT_CONFIG:
+ ALOGW("%s: transaction %d received from PID %d",
+ __func__, code, IPCThreadState::self()->getCallingPid());
+ return INVALID_OPERATION;
+ default:
+ break;
+ }
+
// Whitelist of relevant events to trigger log merging.
// Log merging should activate during audio activity of any kind. This are considered the
// most relevant events.
@@ -845,12 +875,8 @@
case CREATE_RECORD:
case SET_MASTER_VOLUME:
case SET_MASTER_MUTE:
- case SET_STREAM_VOLUME:
- case SET_STREAM_MUTE:
case SET_MIC_MUTE:
case SET_PARAMETERS:
- case OPEN_INPUT:
- case SET_VOICE_VOLUME:
case CREATE_EFFECT:
case SYSTEM_READY: {
requestLogMerge();
@@ -859,6 +885,7 @@
default:
break;
}
+
switch (code) {
case CREATE_TRACK: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
diff --git a/media/libaudioclient/IAudioPolicyService.cpp b/media/libaudioclient/IAudioPolicyService.cpp
index 970ae90..53bc1b7 100644
--- a/media/libaudioclient/IAudioPolicyService.cpp
+++ b/media/libaudioclient/IAudioPolicyService.cpp
@@ -22,6 +22,7 @@
#include <math.h>
#include <sys/types.h>
+#include <binder/IPCThreadState.h>
#include <binder/Parcel.h>
#include <media/AudioEffect.h>
@@ -831,10 +832,33 @@
// ----------------------------------------------------------------------
-
status_t BnAudioPolicyService::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
+ // make sure transactions reserved to AudioFlinger do not come from other processes
+ switch (code) {
+ case START_OUTPUT:
+ case STOP_OUTPUT:
+ case RELEASE_OUTPUT:
+ case GET_INPUT_FOR_ATTR:
+ case START_INPUT:
+ case STOP_INPUT:
+ case RELEASE_INPUT:
+ case GET_STRATEGY_FOR_STREAM:
+ case GET_OUTPUT_FOR_EFFECT:
+ case REGISTER_EFFECT:
+ case UNREGISTER_EFFECT:
+ case SET_EFFECT_ENABLED:
+ case GET_OUTPUT_FOR_ATTR:
+ case ACQUIRE_SOUNDTRIGGER_SESSION:
+ case RELEASE_SOUNDTRIGGER_SESSION:
+ ALOGW("%s: transaction %d received from PID %d",
+ __func__, code, IPCThreadState::self()->getCallingPid());
+ return INVALID_OPERATION;
+ default:
+ break;
+ }
+
switch (code) {
case SET_DEVICE_CONNECTION_STATE: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
diff --git a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
index 146e9e8..8ebae11 100644
--- a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
+++ b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
@@ -3330,14 +3330,19 @@
//ALOGV("\tEffect_process Not Calling process with %d effects enabled, %d called: Effect %d",
//pContext->pBundledContext->NumberEffectsEnabled,
//pContext->pBundledContext->NumberEffectsCalled, pContext->EffectType);
- // 2 is for stereo input
+
if (pContext->config.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
- for (size_t i=0; i < outBuffer->frameCount*2; i++){
- outBuffer->s16[i] =
- clamp16((LVM_INT32)outBuffer->s16[i] + (LVM_INT32)inBuffer->s16[i]);
+ for (size_t i = 0; i < outBuffer->frameCount * FCC_2; ++i){
+#ifdef NATIVE_FLOAT_BUFFER
+ outBuffer->f32[i] += inBuffer->f32[i];
+#else
+ outBuffer->s16[i] = clamp16((LVM_INT32)outBuffer->s16[i] + inBuffer->s16[i]);
+#endif
}
} else if (outBuffer->raw != inBuffer->raw) {
- memcpy(outBuffer->raw, inBuffer->raw, outBuffer->frameCount*sizeof(LVM_INT16)*2);
+ memcpy(outBuffer->raw,
+ inBuffer->raw,
+ outBuffer->frameCount * sizeof(effect_buffer_t) * FCC_2);
}
}
diff --git a/media/libeffects/visualizer/EffectVisualizer.cpp b/media/libeffects/visualizer/EffectVisualizer.cpp
index 0e82339..c33f9f5 100644
--- a/media/libeffects/visualizer/EffectVisualizer.cpp
+++ b/media/libeffects/visualizer/EffectVisualizer.cpp
@@ -594,7 +594,7 @@
deltaSmpl = CAPTURE_BUF_SIZE;
}
- int32_t capturePoint = pContext->mCaptureIdx - deltaSmpl;
+ int32_t capturePoint = (int32_t)pContext->mCaptureIdx - deltaSmpl;
// a negative capturePoint means we wrap the buffer.
if (capturePoint < 0) {
uint32_t size = -capturePoint;
diff --git a/media/libmedia/IMediaHTTPService.cpp b/media/libmedia/IMediaHTTPService.cpp
index 062a07a..74d8ee8 100644
--- a/media/libmedia/IMediaHTTPService.cpp
+++ b/media/libmedia/IMediaHTTPService.cpp
@@ -34,7 +34,7 @@
: BpInterface<IMediaHTTPService>(impl) {
}
- virtual sp<IMediaHTTPConnection> makeHTTPConnection() {
+ virtual sp<MediaHTTPConnection> makeHTTPConnection() {
Parcel data, reply;
data.writeInterfaceToken(
IMediaHTTPService::getInterfaceDescriptor());
diff --git a/media/libmedia/include/media/IMediaHTTPConnection.h b/media/libmedia/include/media/IMediaHTTPConnection.h
index 2a63eb7..0fb6bb1 100644
--- a/media/libmedia/include/media/IMediaHTTPConnection.h
+++ b/media/libmedia/include/media/IMediaHTTPConnection.h
@@ -19,16 +19,15 @@
#define I_MEDIA_HTTP_CONNECTION_H_
#include <binder/IInterface.h>
+#include <media/MediaHTTPConnection.h>
#include <media/stagefright/foundation/ABase.h>
#include <utils/KeyedVector.h>
namespace android {
-struct IMediaHTTPConnection;
-
/** MUST stay in sync with IMediaHTTPConnection.aidl */
-struct IMediaHTTPConnection : public IInterface {
+struct IMediaHTTPConnection : public MediaHTTPConnection, public IInterface {
DECLARE_META_INTERFACE(MediaHTTPConnection);
virtual bool connect(
diff --git a/media/libmedia/include/media/IMediaHTTPService.h b/media/libmedia/include/media/IMediaHTTPService.h
index f66d6c8..e948b78 100644
--- a/media/libmedia/include/media/IMediaHTTPService.h
+++ b/media/libmedia/include/media/IMediaHTTPService.h
@@ -19,18 +19,19 @@
#define I_MEDIA_HTTP_SERVICE_H_
#include <binder/IInterface.h>
+#include <media/MediaHTTPService.h>
#include <media/stagefright/foundation/ABase.h>
namespace android {
-struct IMediaHTTPConnection;
+struct MediaHTTPConnection;
/** MUST stay in sync with IMediaHTTPService.aidl */
-struct IMediaHTTPService : public IInterface {
+struct IMediaHTTPService : public MediaHTTPService, public IInterface {
DECLARE_META_INTERFACE(MediaHTTPService);
- virtual sp<IMediaHTTPConnection> makeHTTPConnection() = 0;
+ virtual sp<MediaHTTPConnection> makeHTTPConnection() = 0;
private:
DISALLOW_EVIL_CONSTRUCTORS(IMediaHTTPService);
diff --git a/media/libmedia/include/media/MediaHTTPConnection.h b/media/libmedia/include/media/MediaHTTPConnection.h
new file mode 100644
index 0000000..82a79e5
--- /dev/null
+++ b/media/libmedia/include/media/MediaHTTPConnection.h
@@ -0,0 +1,44 @@
+/*
+ * Copyright 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef MEDIA_HTTP_CONNECTION_H_
+
+#define MEDIA_HTTP_CONNECTION_H_
+
+#include <media/stagefright/foundation/ABase.h>
+#include <utils/KeyedVector.h>
+
+namespace android {
+
+struct MediaHTTPConnection : public virtual RefBase {
+ MediaHTTPConnection() {}
+
+ virtual bool connect(
+ const char *uri, const KeyedVector<String8, String8> *headers) = 0;
+
+ virtual void disconnect() = 0;
+ virtual ssize_t readAt(off64_t offset, void *data, size_t size) = 0;
+ virtual off64_t getSize() = 0;
+ virtual status_t getMIMEType(String8 *mimeType) = 0;
+ virtual status_t getUri(String8 *uri) = 0;
+
+private:
+ DISALLOW_EVIL_CONSTRUCTORS(MediaHTTPConnection);
+};
+
+} // namespace android
+
+#endif // MEDIA_HTTP_CONNECTION_H_
diff --git a/media/libmedia/include/media/MediaHTTPService.h b/media/libmedia/include/media/MediaHTTPService.h
new file mode 100644
index 0000000..6e9f125
--- /dev/null
+++ b/media/libmedia/include/media/MediaHTTPService.h
@@ -0,0 +1,38 @@
+/*
+ * Copyright 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef MEDIA_HTTP_SERVICE_H_
+
+#define MEDIA_HTTP_SERVICE_H_
+
+#include <media/stagefright/foundation/ABase.h>
+
+namespace android {
+
+struct MediaHTTPConnection;
+
+struct MediaHTTPService : public virtual RefBase {
+ MediaHTTPService() {}
+
+ virtual sp<MediaHTTPConnection> makeHTTPConnection() = 0;
+
+private:
+ DISALLOW_EVIL_CONSTRUCTORS(MediaHTTPService);
+};
+
+} // namespace android
+
+#endif // MEDIA_HTTP_SERVICE_H_
diff --git a/media/libstagefright/DataSourceFactory.cpp b/media/libstagefright/DataSourceFactory.cpp
index aee858c..54bf0cc 100644
--- a/media/libstagefright/DataSourceFactory.cpp
+++ b/media/libstagefright/DataSourceFactory.cpp
@@ -19,8 +19,8 @@
#include "include/HTTPBase.h"
#include "include/NuCachedSource2.h"
-#include <media/IMediaHTTPConnection.h>
-#include <media/IMediaHTTPService.h>
+#include <media/MediaHTTPConnection.h>
+#include <media/MediaHTTPService.h>
#include <media/stagefright/DataSourceFactory.h>
#include <media/stagefright/DataURISource.h>
#include <media/stagefright/FileSource.h>
@@ -31,7 +31,7 @@
// static
sp<DataSource> DataSourceFactory::CreateFromURI(
- const sp<IMediaHTTPService> &httpService,
+ const sp<MediaHTTPService> &httpService,
const char *uri,
const KeyedVector<String8, String8> *headers,
String8 *contentType,
@@ -50,7 +50,7 @@
}
if (httpSource == NULL) {
- sp<IMediaHTTPConnection> conn = httpService->makeHTTPConnection();
+ sp<MediaHTTPConnection> conn = httpService->makeHTTPConnection();
if (conn == NULL) {
ALOGE("Failed to make http connection from http service!");
return NULL;
@@ -101,12 +101,12 @@
return source->initCheck() != OK ? nullptr : source;
}
-sp<DataSource> DataSourceFactory::CreateMediaHTTP(const sp<IMediaHTTPService> &httpService) {
+sp<DataSource> DataSourceFactory::CreateMediaHTTP(const sp<MediaHTTPService> &httpService) {
if (httpService == NULL) {
return NULL;
}
- sp<IMediaHTTPConnection> conn = httpService->makeHTTPConnection();
+ sp<MediaHTTPConnection> conn = httpService->makeHTTPConnection();
if (conn == NULL) {
return NULL;
} else {
diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp
index 1fe5f60..8db00f0 100644
--- a/media/libstagefright/MPEG4Writer.cpp
+++ b/media/libstagefright/MPEG4Writer.cpp
@@ -2951,212 +2951,215 @@
mGotStartKeyFrame = true;
}
////////////////////////////////////////////////////////////////////////////////
- if (mStszTableEntries->count() == 0) {
- mFirstSampleTimeRealUs = systemTime() / 1000;
- mStartTimestampUs = timestampUs;
- mOwner->setStartTimestampUs(mStartTimestampUs);
- previousPausedDurationUs = mStartTimestampUs;
- }
- if (mResumed) {
- int64_t durExcludingEarlierPausesUs = timestampUs - previousPausedDurationUs;
- if (WARN_UNLESS(durExcludingEarlierPausesUs >= 0ll, "for %s track", trackName)) {
- copy->release();
- mSource->stop();
- mIsMalformed = true;
- break;
- }
-
- int64_t pausedDurationUs = durExcludingEarlierPausesUs - mTrackDurationUs;
- if (WARN_UNLESS(pausedDurationUs >= lastDurationUs, "for %s track", trackName)) {
- copy->release();
- mSource->stop();
- mIsMalformed = true;
- break;
- }
-
- previousPausedDurationUs += pausedDurationUs - lastDurationUs;
- mResumed = false;
- }
- TimestampDebugHelperEntry timestampDebugEntry;
- timestampUs -= previousPausedDurationUs;
- timestampDebugEntry.pts = timestampUs;
- if (WARN_UNLESS(timestampUs >= 0ll, "for %s track", trackName)) {
- copy->release();
- mSource->stop();
- mIsMalformed = true;
- break;
- }
-
- if (mIsVideo) {
- /*
- * Composition time: timestampUs
- * Decoding time: decodingTimeUs
- * Composition time offset = composition time - decoding time
- */
- int64_t decodingTimeUs;
- CHECK(meta_data->findInt64(kKeyDecodingTime, &decodingTimeUs));
- decodingTimeUs -= previousPausedDurationUs;
-
- // ensure non-negative, monotonic decoding time
- if (mLastDecodingTimeUs < 0) {
- decodingTimeUs = std::max((int64_t)0, decodingTimeUs);
- } else {
- // increase decoding time by at least the larger vaule of 1 tick and
- // 0.1 milliseconds. This needs to take into account the possible
- // delta adjustment in DurationTicks in below.
- decodingTimeUs = std::max(mLastDecodingTimeUs +
- std::max(100, divUp(1000000, mTimeScale)), decodingTimeUs);
- }
-
- mLastDecodingTimeUs = decodingTimeUs;
- timestampDebugEntry.dts = decodingTimeUs;
- timestampDebugEntry.frameType = isSync ? "Key frame" : "Non-Key frame";
- // Insert the timestamp into the mTimestampDebugHelper
- if (mTimestampDebugHelper.size() >= kTimestampDebugCount) {
- mTimestampDebugHelper.pop_front();
- }
- mTimestampDebugHelper.push_back(timestampDebugEntry);
-
- cttsOffsetTimeUs =
- timestampUs + kMaxCttsOffsetTimeUs - decodingTimeUs;
- if (WARN_UNLESS(cttsOffsetTimeUs >= 0ll, "for %s track", trackName)) {
- copy->release();
- mSource->stop();
- mIsMalformed = true;
- break;
- }
-
- timestampUs = decodingTimeUs;
- ALOGV("decoding time: %" PRId64 " and ctts offset time: %" PRId64,
- timestampUs, cttsOffsetTimeUs);
-
- // Update ctts box table if necessary
- currCttsOffsetTimeTicks =
- (cttsOffsetTimeUs * mTimeScale + 500000LL) / 1000000LL;
- if (WARN_UNLESS(currCttsOffsetTimeTicks <= 0x0FFFFFFFFLL, "for %s track", trackName)) {
- copy->release();
- mSource->stop();
- mIsMalformed = true;
- break;
- }
-
+ if (!mIsHeic) {
if (mStszTableEntries->count() == 0) {
- // Force the first ctts table entry to have one single entry
- // so that we can do adjustment for the initial track start
- // time offset easily in writeCttsBox().
- lastCttsOffsetTimeTicks = currCttsOffsetTimeTicks;
- addOneCttsTableEntry(1, currCttsOffsetTimeTicks);
- cttsSampleCount = 0; // No sample in ctts box is pending
- } else {
- if (currCttsOffsetTimeTicks != lastCttsOffsetTimeTicks) {
- addOneCttsTableEntry(cttsSampleCount, lastCttsOffsetTimeTicks);
- lastCttsOffsetTimeTicks = currCttsOffsetTimeTicks;
- cttsSampleCount = 1; // One sample in ctts box is pending
+ mFirstSampleTimeRealUs = systemTime() / 1000;
+ mStartTimestampUs = timestampUs;
+ mOwner->setStartTimestampUs(mStartTimestampUs);
+ previousPausedDurationUs = mStartTimestampUs;
+ }
+
+ if (mResumed) {
+ int64_t durExcludingEarlierPausesUs = timestampUs - previousPausedDurationUs;
+ if (WARN_UNLESS(durExcludingEarlierPausesUs >= 0ll, "for %s track", trackName)) {
+ copy->release();
+ mSource->stop();
+ mIsMalformed = true;
+ break;
+ }
+
+ int64_t pausedDurationUs = durExcludingEarlierPausesUs - mTrackDurationUs;
+ if (WARN_UNLESS(pausedDurationUs >= lastDurationUs, "for %s track", trackName)) {
+ copy->release();
+ mSource->stop();
+ mIsMalformed = true;
+ break;
+ }
+
+ previousPausedDurationUs += pausedDurationUs - lastDurationUs;
+ mResumed = false;
+ }
+ TimestampDebugHelperEntry timestampDebugEntry;
+ timestampUs -= previousPausedDurationUs;
+ timestampDebugEntry.pts = timestampUs;
+ if (WARN_UNLESS(timestampUs >= 0ll, "for %s track", trackName)) {
+ copy->release();
+ mSource->stop();
+ mIsMalformed = true;
+ break;
+ }
+
+ if (mIsVideo) {
+ /*
+ * Composition time: timestampUs
+ * Decoding time: decodingTimeUs
+ * Composition time offset = composition time - decoding time
+ */
+ int64_t decodingTimeUs;
+ CHECK(meta_data->findInt64(kKeyDecodingTime, &decodingTimeUs));
+ decodingTimeUs -= previousPausedDurationUs;
+
+ // ensure non-negative, monotonic decoding time
+ if (mLastDecodingTimeUs < 0) {
+ decodingTimeUs = std::max((int64_t)0, decodingTimeUs);
} else {
- ++cttsSampleCount;
+ // increase decoding time by at least the larger vaule of 1 tick and
+ // 0.1 milliseconds. This needs to take into account the possible
+ // delta adjustment in DurationTicks in below.
+ decodingTimeUs = std::max(mLastDecodingTimeUs +
+ std::max(100, divUp(1000000, mTimeScale)), decodingTimeUs);
}
- }
- // Update ctts time offset range
- if (mStszTableEntries->count() == 0) {
- mMinCttsOffsetTicks = currCttsOffsetTimeTicks;
- mMaxCttsOffsetTicks = currCttsOffsetTimeTicks;
- } else {
- if (currCttsOffsetTimeTicks > mMaxCttsOffsetTicks) {
- mMaxCttsOffsetTicks = currCttsOffsetTimeTicks;
- } else if (currCttsOffsetTimeTicks < mMinCttsOffsetTicks) {
+ mLastDecodingTimeUs = decodingTimeUs;
+ timestampDebugEntry.dts = decodingTimeUs;
+ timestampDebugEntry.frameType = isSync ? "Key frame" : "Non-Key frame";
+ // Insert the timestamp into the mTimestampDebugHelper
+ if (mTimestampDebugHelper.size() >= kTimestampDebugCount) {
+ mTimestampDebugHelper.pop_front();
+ }
+ mTimestampDebugHelper.push_back(timestampDebugEntry);
+
+ cttsOffsetTimeUs =
+ timestampUs + kMaxCttsOffsetTimeUs - decodingTimeUs;
+ if (WARN_UNLESS(cttsOffsetTimeUs >= 0ll, "for %s track", trackName)) {
+ copy->release();
+ mSource->stop();
+ mIsMalformed = true;
+ break;
+ }
+
+ timestampUs = decodingTimeUs;
+ ALOGV("decoding time: %" PRId64 " and ctts offset time: %" PRId64,
+ timestampUs, cttsOffsetTimeUs);
+
+ // Update ctts box table if necessary
+ currCttsOffsetTimeTicks =
+ (cttsOffsetTimeUs * mTimeScale + 500000LL) / 1000000LL;
+ if (WARN_UNLESS(currCttsOffsetTimeTicks <= 0x0FFFFFFFFLL, "for %s track", trackName)) {
+ copy->release();
+ mSource->stop();
+ mIsMalformed = true;
+ break;
+ }
+
+ if (mStszTableEntries->count() == 0) {
+ // Force the first ctts table entry to have one single entry
+ // so that we can do adjustment for the initial track start
+ // time offset easily in writeCttsBox().
+ lastCttsOffsetTimeTicks = currCttsOffsetTimeTicks;
+ addOneCttsTableEntry(1, currCttsOffsetTimeTicks);
+ cttsSampleCount = 0; // No sample in ctts box is pending
+ } else {
+ if (currCttsOffsetTimeTicks != lastCttsOffsetTimeTicks) {
+ addOneCttsTableEntry(cttsSampleCount, lastCttsOffsetTimeTicks);
+ lastCttsOffsetTimeTicks = currCttsOffsetTimeTicks;
+ cttsSampleCount = 1; // One sample in ctts box is pending
+ } else {
+ ++cttsSampleCount;
+ }
+ }
+
+ // Update ctts time offset range
+ if (mStszTableEntries->count() == 0) {
mMinCttsOffsetTicks = currCttsOffsetTimeTicks;
- mMinCttsOffsetTimeUs = cttsOffsetTimeUs;
+ mMaxCttsOffsetTicks = currCttsOffsetTimeTicks;
+ } else {
+ if (currCttsOffsetTimeTicks > mMaxCttsOffsetTicks) {
+ mMaxCttsOffsetTicks = currCttsOffsetTimeTicks;
+ } else if (currCttsOffsetTimeTicks < mMinCttsOffsetTicks) {
+ mMinCttsOffsetTicks = currCttsOffsetTimeTicks;
+ mMinCttsOffsetTimeUs = cttsOffsetTimeUs;
+ }
}
}
- }
- if (mOwner->isRealTimeRecording()) {
- if (mIsAudio) {
- updateDriftTime(meta_data);
- }
- }
-
- if (WARN_UNLESS(timestampUs >= 0ll, "for %s track", trackName)) {
- copy->release();
- mSource->stop();
- mIsMalformed = true;
- break;
- }
-
- ALOGV("%s media time stamp: %" PRId64 " and previous paused duration %" PRId64,
- trackName, timestampUs, previousPausedDurationUs);
- if (timestampUs > mTrackDurationUs) {
- mTrackDurationUs = timestampUs;
- }
-
- // We need to use the time scale based ticks, rather than the
- // timestamp itself to determine whether we have to use a new
- // stts entry, since we may have rounding errors.
- // The calculation is intended to reduce the accumulated
- // rounding errors.
- currDurationTicks =
- ((timestampUs * mTimeScale + 500000LL) / 1000000LL -
- (lastTimestampUs * mTimeScale + 500000LL) / 1000000LL);
- if (currDurationTicks < 0ll) {
- ALOGE("do not support out of order frames (timestamp: %lld < last: %lld for %s track",
- (long long)timestampUs, (long long)lastTimestampUs, trackName);
- copy->release();
- mSource->stop();
- mIsMalformed = true;
- break;
- }
-
- // if the duration is different for this sample, see if it is close enough to the previous
- // duration that we can fudge it and use the same value, to avoid filling the stts table
- // with lots of near-identical entries.
- // "close enough" here means that the current duration needs to be adjusted by less
- // than 0.1 milliseconds
- if (lastDurationTicks && (currDurationTicks != lastDurationTicks)) {
- int64_t deltaUs = ((lastDurationTicks - currDurationTicks) * 1000000LL
- + (mTimeScale / 2)) / mTimeScale;
- if (deltaUs > -100 && deltaUs < 100) {
- // use previous ticks, and adjust timestamp as if it was actually that number
- // of ticks
- currDurationTicks = lastDurationTicks;
- timestampUs += deltaUs;
- }
- }
- mStszTableEntries->add(htonl(sampleSize));
- if (mStszTableEntries->count() > 2) {
-
- // Force the first sample to have its own stts entry so that
- // we can adjust its value later to maintain the A/V sync.
- if (mStszTableEntries->count() == 3 || currDurationTicks != lastDurationTicks) {
- addOneSttsTableEntry(sampleCount, lastDurationTicks);
- sampleCount = 1;
- } else {
- ++sampleCount;
+ if (mOwner->isRealTimeRecording()) {
+ if (mIsAudio) {
+ updateDriftTime(meta_data);
+ }
}
- }
- if (mSamplesHaveSameSize) {
- if (mStszTableEntries->count() >= 2 && previousSampleSize != sampleSize) {
- mSamplesHaveSameSize = false;
+ if (WARN_UNLESS(timestampUs >= 0ll, "for %s track", trackName)) {
+ copy->release();
+ mSource->stop();
+ mIsMalformed = true;
+ break;
}
- previousSampleSize = sampleSize;
- }
- ALOGV("%s timestampUs/lastTimestampUs: %" PRId64 "/%" PRId64,
- trackName, timestampUs, lastTimestampUs);
- lastDurationUs = timestampUs - lastTimestampUs;
- lastDurationTicks = currDurationTicks;
- lastTimestampUs = timestampUs;
- if (isSync != 0) {
- addOneStssTableEntry(mStszTableEntries->count());
- }
-
- if (mTrackingProgressStatus) {
- if (mPreviousTrackTimeUs <= 0) {
- mPreviousTrackTimeUs = mStartTimestampUs;
+ ALOGV("%s media time stamp: %" PRId64 " and previous paused duration %" PRId64,
+ trackName, timestampUs, previousPausedDurationUs);
+ if (timestampUs > mTrackDurationUs) {
+ mTrackDurationUs = timestampUs;
}
- trackProgressStatus(timestampUs);
+
+ // We need to use the time scale based ticks, rather than the
+ // timestamp itself to determine whether we have to use a new
+ // stts entry, since we may have rounding errors.
+ // The calculation is intended to reduce the accumulated
+ // rounding errors.
+ currDurationTicks =
+ ((timestampUs * mTimeScale + 500000LL) / 1000000LL -
+ (lastTimestampUs * mTimeScale + 500000LL) / 1000000LL);
+ if (currDurationTicks < 0ll) {
+ ALOGE("do not support out of order frames (timestamp: %lld < last: %lld for %s track",
+ (long long)timestampUs, (long long)lastTimestampUs, trackName);
+ copy->release();
+ mSource->stop();
+ mIsMalformed = true;
+ break;
+ }
+
+ // if the duration is different for this sample, see if it is close enough to the previous
+ // duration that we can fudge it and use the same value, to avoid filling the stts table
+ // with lots of near-identical entries.
+ // "close enough" here means that the current duration needs to be adjusted by less
+ // than 0.1 milliseconds
+ if (lastDurationTicks && (currDurationTicks != lastDurationTicks)) {
+ int64_t deltaUs = ((lastDurationTicks - currDurationTicks) * 1000000LL
+ + (mTimeScale / 2)) / mTimeScale;
+ if (deltaUs > -100 && deltaUs < 100) {
+ // use previous ticks, and adjust timestamp as if it was actually that number
+ // of ticks
+ currDurationTicks = lastDurationTicks;
+ timestampUs += deltaUs;
+ }
+ }
+ mStszTableEntries->add(htonl(sampleSize));
+ if (mStszTableEntries->count() > 2) {
+
+ // Force the first sample to have its own stts entry so that
+ // we can adjust its value later to maintain the A/V sync.
+ if (mStszTableEntries->count() == 3 || currDurationTicks != lastDurationTicks) {
+ addOneSttsTableEntry(sampleCount, lastDurationTicks);
+ sampleCount = 1;
+ } else {
+ ++sampleCount;
+ }
+
+ }
+ if (mSamplesHaveSameSize) {
+ if (mStszTableEntries->count() >= 2 && previousSampleSize != sampleSize) {
+ mSamplesHaveSameSize = false;
+ }
+ previousSampleSize = sampleSize;
+ }
+ ALOGV("%s timestampUs/lastTimestampUs: %" PRId64 "/%" PRId64,
+ trackName, timestampUs, lastTimestampUs);
+ lastDurationUs = timestampUs - lastTimestampUs;
+ lastDurationTicks = currDurationTicks;
+ lastTimestampUs = timestampUs;
+
+ if (isSync != 0) {
+ addOneStssTableEntry(mStszTableEntries->count());
+ }
+
+ if (mTrackingProgressStatus) {
+ if (mPreviousTrackTimeUs <= 0) {
+ mPreviousTrackTimeUs = mStartTimestampUs;
+ }
+ trackProgressStatus(timestampUs);
+ }
}
if (!hasMultipleTracks) {
size_t bytesWritten;
@@ -4331,9 +4334,12 @@
}
// patch up the mPrimaryItemId and count items with prop associations
+ uint16_t firstVisibleItemId = 0;
for (size_t index = 0; index < mItems.size(); index++) {
if (mItems[index].isPrimary) {
mPrimaryItemId = mItems[index].itemId;
+ } else if (!firstVisibleItemId && !mItems[index].isHidden) {
+ firstVisibleItemId = mItems[index].itemId;
}
if (!mItems[index].properties.empty()) {
@@ -4342,8 +4348,13 @@
}
if (mPrimaryItemId == 0) {
- ALOGW("didn't find primary, using first item");
- mPrimaryItemId = mItems[0].itemId;
+ if (firstVisibleItemId > 0) {
+ ALOGW("didn't find primary, using first visible item");
+ mPrimaryItemId = firstVisibleItemId;
+ } else {
+ ALOGW("no primary and no visible item, using first item");
+ mPrimaryItemId = mItems[0].itemId;
+ }
}
beginBox("meta");
diff --git a/media/libstagefright/NuMediaExtractor.cpp b/media/libstagefright/NuMediaExtractor.cpp
index a176382..17c9648 100644
--- a/media/libstagefright/NuMediaExtractor.cpp
+++ b/media/libstagefright/NuMediaExtractor.cpp
@@ -72,7 +72,7 @@
}
status_t NuMediaExtractor::setDataSource(
- const sp<IMediaHTTPService> &httpService,
+ const sp<MediaHTTPService> &httpService,
const char *path,
const KeyedVector<String8, String8> *headers) {
Mutex::Autolock autoLock(mLock);
diff --git a/media/libstagefright/codecs/avcenc/SoftAVCEnc.cpp b/media/libstagefright/codecs/avcenc/SoftAVCEnc.cpp
index 358c743..32fdbd3 100644
--- a/media/libstagefright/codecs/avcenc/SoftAVCEnc.cpp
+++ b/media/libstagefright/codecs/avcenc/SoftAVCEnc.cpp
@@ -1170,6 +1170,12 @@
ps_inp_raw_buf->e_color_fmt = mIvVideoColorFormat;
source = NULL;
if ((inputBufferHeader != NULL) && inputBufferHeader->nFilledLen) {
+ OMX_ERRORTYPE error = validateInputBuffer(inputBufferHeader);
+ if (error != OMX_ErrorNone) {
+ ALOGE("b/69065651");
+ android_errorWriteLog(0x534e4554, "69065651");
+ return error;
+ }
source = inputBufferHeader->pBuffer + inputBufferHeader->nOffset;
if (mInputDataIsMeta) {
diff --git a/media/libstagefright/codecs/m4v_h263/enc/SoftMPEG4Encoder.cpp b/media/libstagefright/codecs/m4v_h263/enc/SoftMPEG4Encoder.cpp
index 7b90a01..f6a7b0e 100644
--- a/media/libstagefright/codecs/m4v_h263/enc/SoftMPEG4Encoder.cpp
+++ b/media/libstagefright/codecs/m4v_h263/enc/SoftMPEG4Encoder.cpp
@@ -434,6 +434,14 @@
}
if (inHeader->nFilledLen > 0) {
+ OMX_ERRORTYPE error = validateInputBuffer(inHeader);
+ if (error != OMX_ErrorNone) {
+ ALOGE("b/69065651");
+ android_errorWriteLog(0x534e4554, "69065651");
+ mSignalledError = true;
+ notify(OMX_EventError, error, 0, 0);
+ return;
+ }
const uint8_t *inputData = NULL;
if (mInputDataIsMeta) {
inputData =
diff --git a/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp b/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp
index a5666da..f6257b1 100644
--- a/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp
+++ b/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp
@@ -653,6 +653,13 @@
return;
}
+ OMX_ERRORTYPE error = validateInputBuffer(inputBufferHeader);
+ if (error != OMX_ErrorNone) {
+ ALOGE("b/27569635");
+ android_errorWriteLog(0x534e4554, "27569635");
+ notify(OMX_EventError, error, 0, 0);
+ return;
+ }
const uint8_t *source =
inputBufferHeader->pBuffer + inputBufferHeader->nOffset;
@@ -668,14 +675,6 @@
return;
}
} else {
- if (inputBufferHeader->nFilledLen < frameSize) {
- android_errorWriteLog(0x534e4554, "27569635");
- notify(OMX_EventError, OMX_ErrorUndefined, 0, 0);
- return;
- } else if (inputBufferHeader->nFilledLen > frameSize) {
- ALOGW("Input buffer contains too many pixels");
- }
-
if (mColorFormat == OMX_COLOR_FormatYUV420SemiPlanar) {
ConvertYUV420SemiPlanarToYUV420Planar(
source, mConversionBuffer, mWidth, mHeight);
diff --git a/media/libstagefright/http/MediaHTTP.cpp b/media/libstagefright/http/MediaHTTP.cpp
index 5b18814..84837e8 100644
--- a/media/libstagefright/http/MediaHTTP.cpp
+++ b/media/libstagefright/http/MediaHTTP.cpp
@@ -25,11 +25,11 @@
#include <media/stagefright/foundation/ALooper.h>
#include <media/stagefright/Utils.h>
-#include <media/IMediaHTTPConnection.h>
+#include <media/MediaHTTPConnection.h>
namespace android {
-MediaHTTP::MediaHTTP(const sp<IMediaHTTPConnection> &conn)
+MediaHTTP::MediaHTTP(const sp<MediaHTTPConnection> &conn)
: mInitCheck((conn != NULL) ? OK : NO_INIT),
mHTTPConnection(conn),
mCachedSizeValid(false),
diff --git a/media/libstagefright/httplive/HTTPDownloader.cpp b/media/libstagefright/httplive/HTTPDownloader.cpp
index 3fef764..72604e3 100644
--- a/media/libstagefright/httplive/HTTPDownloader.cpp
+++ b/media/libstagefright/httplive/HTTPDownloader.cpp
@@ -22,8 +22,8 @@
#include "M3UParser.h"
#include <media/DataSource.h>
-#include <media/IMediaHTTPConnection.h>
-#include <media/IMediaHTTPService.h>
+#include <media/MediaHTTPConnection.h>
+#include <media/MediaHTTPService.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaHTTP.h>
@@ -36,7 +36,7 @@
namespace android {
HTTPDownloader::HTTPDownloader(
- const sp<IMediaHTTPService> &httpService,
+ const sp<MediaHTTPService> &httpService,
const KeyedVector<String8, String8> &headers) :
mHTTPDataSource(new MediaHTTP(httpService->makeHTTPConnection())),
mExtraHeaders(headers),
diff --git a/media/libstagefright/httplive/HTTPDownloader.h b/media/libstagefright/httplive/HTTPDownloader.h
index 1db4a48..0d4bd31 100644
--- a/media/libstagefright/httplive/HTTPDownloader.h
+++ b/media/libstagefright/httplive/HTTPDownloader.h
@@ -28,12 +28,12 @@
struct ABuffer;
class DataSource;
struct HTTPBase;
-struct IMediaHTTPService;
+struct MediaHTTPService;
struct M3UParser;
struct HTTPDownloader : public RefBase {
HTTPDownloader(
- const sp<IMediaHTTPService> &httpService,
+ const sp<MediaHTTPService> &httpService,
const KeyedVector<String8, String8> &headers);
void reconnect();
diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp
index 4c2e0d4..1e2e684 100644
--- a/media/libstagefright/httplive/LiveSession.cpp
+++ b/media/libstagefright/httplive/LiveSession.cpp
@@ -26,7 +26,7 @@
#include "mpeg2ts/AnotherPacketSource.h"
#include <cutils/properties.h>
-#include <media/IMediaHTTPService.h>
+#include <media/MediaHTTPService.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/AMessage.h>
@@ -274,7 +274,7 @@
LiveSession::LiveSession(
const sp<AMessage> ¬ify, uint32_t flags,
- const sp<IMediaHTTPService> &httpService)
+ const sp<MediaHTTPService> &httpService)
: mNotify(notify),
mFlags(flags),
mHTTPService(httpService),
diff --git a/media/libstagefright/httplive/LiveSession.h b/media/libstagefright/httplive/LiveSession.h
index abf8cf0..7a6d487 100644
--- a/media/libstagefright/httplive/LiveSession.h
+++ b/media/libstagefright/httplive/LiveSession.h
@@ -33,7 +33,7 @@
struct AnotherPacketSource;
class DataSource;
struct HTTPBase;
-struct IMediaHTTPService;
+struct MediaHTTPService;
struct LiveDataSource;
struct M3UParser;
struct PlaylistFetcher;
@@ -71,7 +71,7 @@
LiveSession(
const sp<AMessage> ¬ify,
uint32_t flags,
- const sp<IMediaHTTPService> &httpService);
+ const sp<MediaHTTPService> &httpService);
void setBufferingSettings(const BufferingSettings &buffering);
@@ -187,7 +187,7 @@
sp<AMessage> mNotify;
uint32_t mFlags;
- sp<IMediaHTTPService> mHTTPService;
+ sp<MediaHTTPService> mHTTPService;
bool mBuffering;
bool mInPreparationPhase;
diff --git a/media/libstagefright/include/SDPLoader.h b/media/libstagefright/include/SDPLoader.h
index 2c4f543..b901c97 100644
--- a/media/libstagefright/include/SDPLoader.h
+++ b/media/libstagefright/include/SDPLoader.h
@@ -25,7 +25,7 @@
namespace android {
struct HTTPBase;
-struct IMediaHTTPService;
+struct MediaHTTPService;
struct SDPLoader : public AHandler {
enum Flags {
@@ -38,7 +38,7 @@
SDPLoader(
const sp<AMessage> ¬ify,
uint32_t flags,
- const sp<IMediaHTTPService> &httpService);
+ const sp<MediaHTTPService> &httpService);
void load(const char* url, const KeyedVector<String8, String8> *headers);
diff --git a/media/libstagefright/include/media/stagefright/DataSourceFactory.h b/media/libstagefright/include/media/stagefright/DataSourceFactory.h
index 89add13..2a1d491 100644
--- a/media/libstagefright/include/media/stagefright/DataSourceFactory.h
+++ b/media/libstagefright/include/media/stagefright/DataSourceFactory.h
@@ -23,20 +23,20 @@
namespace android {
-struct IMediaHTTPService;
+struct MediaHTTPService;
class String8;
struct HTTPBase;
class DataSourceFactory {
public:
static sp<DataSource> CreateFromURI(
- const sp<IMediaHTTPService> &httpService,
+ const sp<MediaHTTPService> &httpService,
const char *uri,
const KeyedVector<String8, String8> *headers = NULL,
String8 *contentType = NULL,
HTTPBase *httpSource = NULL);
- static sp<DataSource> CreateMediaHTTP(const sp<IMediaHTTPService> &httpService);
+ static sp<DataSource> CreateMediaHTTP(const sp<MediaHTTPService> &httpService);
static sp<DataSource> CreateFromFd(int fd, int64_t offset, int64_t length);
};
diff --git a/media/libstagefright/include/media/stagefright/MediaHTTP.h b/media/libstagefright/include/media/stagefright/MediaHTTP.h
index 006d8d8..94a2ecd 100644
--- a/media/libstagefright/include/media/stagefright/MediaHTTP.h
+++ b/media/libstagefright/include/media/stagefright/MediaHTTP.h
@@ -24,10 +24,10 @@
namespace android {
-struct IMediaHTTPConnection;
+struct MediaHTTPConnection;
struct MediaHTTP : public HTTPBase {
- MediaHTTP(const sp<IMediaHTTPConnection> &conn);
+ MediaHTTP(const sp<MediaHTTPConnection> &conn);
virtual status_t connect(
const char *uri,
@@ -56,7 +56,7 @@
private:
status_t mInitCheck;
- sp<IMediaHTTPConnection> mHTTPConnection;
+ sp<MediaHTTPConnection> mHTTPConnection;
KeyedVector<String8, String8> mLastHeaders;
AString mLastURI;
diff --git a/media/libstagefright/include/media/stagefright/NuMediaExtractor.h b/media/libstagefright/include/media/stagefright/NuMediaExtractor.h
index 5af0745..eed0f05 100644
--- a/media/libstagefright/include/media/stagefright/NuMediaExtractor.h
+++ b/media/libstagefright/include/media/stagefright/NuMediaExtractor.h
@@ -34,7 +34,7 @@
struct ABuffer;
struct AMessage;
class DataSource;
-struct IMediaHTTPService;
+struct MediaHTTPService;
class MediaBuffer;
class MediaExtractor;
struct MediaSource;
@@ -54,7 +54,7 @@
NuMediaExtractor();
status_t setDataSource(
- const sp<IMediaHTTPService> &httpService,
+ const sp<MediaHTTPService> &httpService,
const char *path,
const KeyedVector<String8, String8> *headers = NULL);
diff --git a/media/libstagefright/omx/SoftVideoEncoderOMXComponent.cpp b/media/libstagefright/omx/SoftVideoEncoderOMXComponent.cpp
index fa15ab3..2fbbb44 100644
--- a/media/libstagefright/omx/SoftVideoEncoderOMXComponent.cpp
+++ b/media/libstagefright/omx/SoftVideoEncoderOMXComponent.cpp
@@ -664,4 +664,17 @@
return SimpleSoftOMXComponent::getExtensionIndex(name, index);
}
+OMX_ERRORTYPE SoftVideoEncoderOMXComponent::validateInputBuffer(
+ const OMX_BUFFERHEADERTYPE *inputBufferHeader) {
+ size_t frameSize = mInputDataIsMeta ?
+ max(sizeof(VideoNativeMetadata), sizeof(VideoGrallocMetadata))
+ : mWidth * mHeight * 3 / 2;
+ if (inputBufferHeader->nFilledLen < frameSize) {
+ return OMX_ErrorUndefined;
+ } else if (inputBufferHeader->nFilledLen > frameSize) {
+ ALOGW("Input buffer contains more data than expected.");
+ }
+ return OMX_ErrorNone;
+}
+
} // namespace android
diff --git a/media/libstagefright/omx/include/media/stagefright/omx/SoftVideoEncoderOMXComponent.h b/media/libstagefright/omx/include/media/stagefright/omx/SoftVideoEncoderOMXComponent.h
index db5496a..2d6f31b 100644
--- a/media/libstagefright/omx/include/media/stagefright/omx/SoftVideoEncoderOMXComponent.h
+++ b/media/libstagefright/omx/include/media/stagefright/omx/SoftVideoEncoderOMXComponent.h
@@ -67,6 +67,8 @@
virtual OMX_ERRORTYPE getExtensionIndex(const char *name, OMX_INDEXTYPE *index);
+ OMX_ERRORTYPE validateInputBuffer(const OMX_BUFFERHEADERTYPE *inputBufferHeader);
+
enum {
kInputPortIndex = 0,
kOutputPortIndex = 1,
diff --git a/media/libstagefright/rtsp/SDPLoader.cpp b/media/libstagefright/rtsp/SDPLoader.cpp
index 0f46c83..d459cbd 100644
--- a/media/libstagefright/rtsp/SDPLoader.cpp
+++ b/media/libstagefright/rtsp/SDPLoader.cpp
@@ -22,8 +22,8 @@
#include "ASessionDescription.h"
-#include <media/IMediaHTTPConnection.h>
-#include <media/IMediaHTTPService.h>
+#include <media/MediaHTTPConnection.h>
+#include <media/MediaHTTPService.h>
#include <media/stagefright/MediaHTTP.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
@@ -36,7 +36,7 @@
SDPLoader::SDPLoader(
const sp<AMessage> ¬ify,
uint32_t flags,
- const sp<IMediaHTTPService> &httpService)
+ const sp<MediaHTTPService> &httpService)
: mNotify(notify),
mFlags(flags),
mNetLooper(new ALooper),
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index e0d0d7b..bfb0fe2 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -296,6 +296,43 @@
const bool auxType =
(mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY;
+ // safeInputOutputSampleCount is 0 if the channel count between input and output
+ // buffers do not match. This prevents automatic accumulation or copying between the
+ // input and output effect buffers without an intermediary effect process.
+ // TODO: consider implementing channel conversion.
+ const size_t safeInputOutputSampleCount =
+ inChannelCount != outChannelCount ? 0
+ : outChannelCount * std::min(
+ mConfig.inputCfg.buffer.frameCount,
+ mConfig.outputCfg.buffer.frameCount);
+ const auto accumulateInputToOutput = [this, safeInputOutputSampleCount]() {
+#ifdef FLOAT_EFFECT_CHAIN
+ accumulate_float(
+ mConfig.outputCfg.buffer.f32,
+ mConfig.inputCfg.buffer.f32,
+ safeInputOutputSampleCount);
+#else
+ accumulate_i16(
+ mConfig.outputCfg.buffer.s16,
+ mConfig.inputCfg.buffer.s16,
+ safeInputOutputSampleCount);
+#endif
+ };
+ const auto copyInputToOutput = [this, safeInputOutputSampleCount]() {
+#ifdef FLOAT_EFFECT_CHAIN
+ memcpy(
+ mConfig.outputCfg.buffer.f32,
+ mConfig.inputCfg.buffer.f32,
+ safeInputOutputSampleCount * sizeof(*mConfig.outputCfg.buffer.f32));
+
+#else
+ memcpy(
+ mConfig.outputCfg.buffer.s16,
+ mConfig.inputCfg.buffer.s16,
+ safeInputOutputSampleCount * sizeof(*mConfig.outputCfg.buffer.s16));
+#endif
+ };
+
if (isProcessEnabled()) {
int ret;
if (isProcessImplemented()) {
@@ -308,97 +345,69 @@
static_assert(sizeof(float) <= sizeof(int32_t),
"in-place conversion requires sizeof(float) <= sizeof(int32_t)");
- const int32_t * const p32 = mConfig.inputCfg.buffer.s32;
- float * const pFloat = mConfig.inputCfg.buffer.f32;
- memcpy_to_float_from_q4_27(pFloat, p32, mConfig.inputCfg.buffer.frameCount);
- } else {
- memcpy_to_i16_from_q4_27(mConfig.inputCfg.buffer.s16,
+ memcpy_to_float_from_q4_27(
+ mConfig.inputCfg.buffer.f32,
+ mConfig.inputCfg.buffer.s32,
+ mConfig.inputCfg.buffer.frameCount);
+ } else
+#endif
+ {
+ memcpy_to_i16_from_q4_27(
+ mConfig.inputCfg.buffer.s16,
mConfig.inputCfg.buffer.s32,
mConfig.inputCfg.buffer.frameCount);
}
-#else
- memcpy_to_i16_from_q4_27(mConfig.inputCfg.buffer.s16,
- mConfig.inputCfg.buffer.s32,
- mConfig.inputCfg.buffer.frameCount);
-#endif
}
#ifdef FLOAT_EFFECT_CHAIN
- if (mSupportsFloat) {
- ret = mEffectInterface->process();
- } else {
- { // convert input to int16_t as effect doesn't support float.
- if (!auxType) {
- if (mInBuffer16.get() == nullptr) {
- ALOGW("%s: mInBuffer16 is null, bypassing", __func__);
- goto data_bypass;
- }
- const float * const pIn = mInBuffer->audioBuffer()->f32;
- int16_t * const pIn16 = mInBuffer16->audioBuffer()->s16;
- memcpy_to_i16_from_float(
- pIn16, pIn, inChannelCount * mConfig.inputCfg.buffer.frameCount);
+ if (!mSupportsFloat) { // convert input to int16_t as effect doesn't support float.
+ if (!auxType) {
+ if (mInConversionBuffer.get() == nullptr) {
+ ALOGW("%s: mInConversionBuffer is null, bypassing", __func__);
+ goto data_bypass;
}
- if (mConfig.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
- if (mOutBuffer16.get() == nullptr) {
- ALOGW("%s: mOutBuffer16 is null, bypassing", __func__);
- goto data_bypass;
- }
- int16_t * const pOut16 = mOutBuffer16->audioBuffer()->s16;
- const float * const pOut = mOutBuffer->audioBuffer()->f32;
- memcpy_to_i16_from_float(
- pOut16,
- pOut,
- outChannelCount * mConfig.outputCfg.buffer.frameCount);
- }
+ memcpy_to_i16_from_float(
+ mInConversionBuffer->audioBuffer()->s16,
+ mInBuffer->audioBuffer()->f32,
+ inChannelCount * mConfig.inputCfg.buffer.frameCount);
}
-
- ret = mEffectInterface->process();
-
- { // convert output back to float.
- const int16_t * const pOut16 = mOutBuffer16->audioBuffer()->s16;
- float * const pOut = mOutBuffer->audioBuffer()->f32;
- memcpy_to_float_from_i16(
- pOut, pOut16, outChannelCount * mConfig.outputCfg.buffer.frameCount);
+ if (mConfig.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
+ if (mOutConversionBuffer.get() == nullptr) {
+ ALOGW("%s: mOutConversionBuffer is null, bypassing", __func__);
+ goto data_bypass;
+ }
+ memcpy_to_i16_from_float(
+ mOutConversionBuffer->audioBuffer()->s16,
+ mOutBuffer->audioBuffer()->f32,
+ outChannelCount * mConfig.outputCfg.buffer.frameCount);
}
}
-#else
+#endif
+
ret = mEffectInterface->process();
+
+#ifdef FLOAT_EFFECT_CHAIN
+ if (!mSupportsFloat) { // convert output int16_t back to float.
+ memcpy_to_float_from_i16(
+ mOutBuffer->audioBuffer()->f32,
+ mOutConversionBuffer->audioBuffer()->s16,
+ outChannelCount * mConfig.outputCfg.buffer.frameCount);
+ }
#endif
} else {
#ifdef FLOAT_EFFECT_CHAIN
data_bypass:
#endif
if (!auxType /* aux effects do not require data bypass */
- && mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw
- && inChannelCount == outChannelCount) {
- const size_t sampleCount = std::min(
- mConfig.inputCfg.buffer.frameCount,
- mConfig.outputCfg.buffer.frameCount) * outChannelCount;
-
-#ifdef FLOAT_EFFECT_CHAIN
- const float * const in = mConfig.inputCfg.buffer.f32;
- float * const out = mConfig.outputCfg.buffer.f32;
-
+ && mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
if (mConfig.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
- accumulate_float(out, in, sampleCount);
+ accumulateInputToOutput();
} else {
- memcpy(mConfig.outputCfg.buffer.f32, mConfig.inputCfg.buffer.f32,
- sampleCount * sizeof(*mConfig.outputCfg.buffer.f32));
+ copyInputToOutput();
}
-
-#else
- const int16_t * const in = mConfig.inputCfg.buffer.s16;
- int16_t * const out = mConfig.outputCfg.buffer.s16;
-
- if (mConfig.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
- accumulate_i16(out, in, sampleCount);
- } else {
- memcpy(mConfig.outputCfg.buffer.s16, mConfig.inputCfg.buffer.s16,
- sampleCount * sizeof(*mConfig.outputCfg.buffer.s16));
- }
-#endif
}
ret = -ENODATA;
}
+
// force transition to IDLE state when engine is ready
if (mState == STOPPED && ret == -ENODATA) {
mDisableWaitCnt = 1;
@@ -417,21 +426,8 @@
// If an insert effect is idle and input buffer is different from output buffer,
// accumulate input onto output
sp<EffectChain> chain = mChain.promote();
- if (chain != 0
- && chain->activeTrackCnt() != 0
- && inChannelCount == outChannelCount) {
- const size_t sampleCount = std::min(
- mConfig.inputCfg.buffer.frameCount,
- mConfig.outputCfg.buffer.frameCount) * outChannelCount;
-#ifdef FLOAT_EFFECT_CHAIN
- const float * const in = mConfig.inputCfg.buffer.f32;
- float * const out = mConfig.outputCfg.buffer.f32;
- accumulate_float(out, in, sampleCount);
-#else
- const int16_t * const in = mConfig.inputCfg.buffer.s16;
- int16_t * const out = mConfig.outputCfg.buffer.s16;
- accumulate_i16(out, in, sampleCount);
-#endif
+ if (chain.get() != nullptr && chain->activeTrackCnt() != 0) {
+ accumulateInputToOutput();
}
}
}
@@ -906,7 +902,7 @@
mEffectInterface->setInBuffer(buffer);
#ifdef FLOAT_EFFECT_CHAIN
- // aux effects do in place conversion to float - we don't allocate mInBuffer16 for them.
+ // aux effects do in place conversion to float - we don't allocate mInConversionBuffer.
// Theoretically insert effects can also do in-place conversions (destroying
// the original buffer) when the output buffer is identical to the input buffer,
// but we don't optimize for it here.
@@ -920,17 +916,17 @@
ALOGV("%s: setInBuffer updating for inChannels:%d inFrameCount:%zu total size:%zu",
__func__, inChannels, inFrameCount, size);
- if (size > 0 && (mInBuffer16.get() == nullptr || size > mInBuffer16->getSize())) {
- mInBuffer16.clear();
- ALOGV("%s: allocating mInBuffer16 %zu", __func__, size);
- (void)EffectBufferHalInterface::allocate(size, &mInBuffer16);
+ if (size > 0 && (mInConversionBuffer.get() == nullptr
+ || size > mInConversionBuffer->getSize())) {
+ mInConversionBuffer.clear();
+ ALOGV("%s: allocating mInConversionBuffer %zu", __func__, size);
+ (void)EffectBufferHalInterface::allocate(size, &mInConversionBuffer);
}
- if (mInBuffer16.get() != nullptr) {
- // FIXME: confirm buffer has enough size.
- mInBuffer16->setFrameCount(inFrameCount);
- mEffectInterface->setInBuffer(mInBuffer16);
+ if (mInConversionBuffer.get() != nullptr) {
+ mInConversionBuffer->setFrameCount(inFrameCount);
+ mEffectInterface->setInBuffer(mInConversionBuffer);
} else if (size > 0) {
- ALOGE("%s cannot create mInBuffer16", __func__);
+ ALOGE("%s cannot create mInConversionBuffer", __func__);
}
}
#endif
@@ -948,7 +944,7 @@
mEffectInterface->setOutBuffer(buffer);
#ifdef FLOAT_EFFECT_CHAIN
- // Note: Any effect that does not accumulate does not need mOutBuffer16 and
+ // Note: Any effect that does not accumulate does not need mOutConversionBuffer and
// can do in-place conversion from int16_t to float. We don't optimize here.
if (!mSupportsFloat && mOutBuffer.get() != nullptr) {
const size_t outFrameCount = mConfig.outputCfg.buffer.frameCount;
@@ -958,16 +954,17 @@
ALOGV("%s: setOutBuffer updating for outChannels:%d outFrameCount:%zu total size:%zu",
__func__, outChannels, outFrameCount, size);
- if (size > 0 && (mOutBuffer16.get() == nullptr || size > mOutBuffer16->getSize())) {
- mOutBuffer16.clear();
- ALOGV("%s: allocating mOutBuffer16 %zu", __func__, size);
- (void)EffectBufferHalInterface::allocate(size, &mOutBuffer16);
+ if (size > 0 && (mOutConversionBuffer.get() == nullptr
+ || size > mOutConversionBuffer->getSize())) {
+ mOutConversionBuffer.clear();
+ ALOGV("%s: allocating mOutConversionBuffer %zu", __func__, size);
+ (void)EffectBufferHalInterface::allocate(size, &mOutConversionBuffer);
}
- if (mOutBuffer16.get() != nullptr) {
- mOutBuffer16->setFrameCount(outFrameCount);
- mEffectInterface->setOutBuffer(mOutBuffer16);
+ if (mOutConversionBuffer.get() != nullptr) {
+ mOutConversionBuffer->setFrameCount(outFrameCount);
+ mEffectInterface->setOutBuffer(mOutConversionBuffer);
} else if (size > 0) {
- ALOGE("%s cannot create mOutBuffer16", __func__);
+ ALOGE("%s cannot create mOutConversionBuffer", __func__);
}
}
#endif
@@ -1241,6 +1238,20 @@
return s;
}
+static std::string dumpInOutBuffer(bool isInput, const sp<EffectBufferHalInterface> &buffer) {
+ std::stringstream ss;
+
+ if (buffer.get() == nullptr) {
+ return "nullptr"; // make different than below
+ } else if (buffer->externalData() != nullptr) {
+ ss << (isInput ? buffer->externalData() : buffer->audioBuffer()->raw)
+ << " -> "
+ << (isInput ? buffer->audioBuffer()->raw : buffer->externalData());
+ } else {
+ ss << buffer->audioBuffer()->raw;
+ }
+ return ss.str();
+}
void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args __unused)
{
@@ -1305,19 +1316,13 @@
result.append(buffer);
#ifdef FLOAT_EFFECT_CHAIN
- if (!mSupportsFloat) {
- int16_t* pIn16 = mInBuffer16 != 0 ? mInBuffer16->audioBuffer()->s16 : NULL;
- int16_t* pOut16 = mOutBuffer16 != 0 ? mOutBuffer16->audioBuffer()->s16 : NULL;
- result.append("\t\t- Float and int16 buffers\n");
- result.append("\t\t\tIn_float In_int16 Out_float Out_int16\n");
- snprintf(buffer, SIZE,"\t\t\t%p %p %p %p\n",
- mConfig.inputCfg.buffer.raw,
- pIn16,
- pOut16,
- mConfig.outputCfg.buffer.raw);
- result.append(buffer);
- }
+ result.appendFormat("\t\t- HAL buffers:\n"
+ "\t\t\tIn(%s) InConversion(%s) Out(%s) OutConversion(%s)\n",
+ dumpInOutBuffer(true /* isInput */, mInBuffer).c_str(),
+ dumpInOutBuffer(true /* isInput */, mInConversionBuffer).c_str(),
+ dumpInOutBuffer(false /* isInput */, mOutBuffer).c_str(),
+ dumpInOutBuffer(false /* isInput */, mOutConversionBuffer).c_str());
#endif
snprintf(buffer, SIZE, "\t\t%zu Clients:\n", mHandles.size());
@@ -2161,19 +2166,6 @@
}
}
-static void dumpInOutBuffer(
- char *dump, size_t dumpSize, bool isInput, EffectBufferHalInterface *buffer) {
- if (buffer == nullptr) {
- snprintf(dump, dumpSize, "%p", buffer);
- } else if (buffer->externalData() != nullptr) {
- snprintf(dump, dumpSize, "%p -> %p",
- isInput ? buffer->externalData() : buffer->audioBuffer()->raw,
- isInput ? buffer->audioBuffer()->raw : buffer->externalData());
- } else {
- snprintf(dump, dumpSize, "%p", buffer->audioBuffer()->raw);
- }
-}
-
void AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
@@ -2191,15 +2183,13 @@
result.append("\tCould not lock mutex:\n");
}
- char inBufferStr[64], outBufferStr[64];
- dumpInOutBuffer(inBufferStr, sizeof(inBufferStr), true, mInBuffer.get());
- dumpInOutBuffer(outBufferStr, sizeof(outBufferStr), false, mOutBuffer.get());
- snprintf(buffer, SIZE, "\t%-*s%-*s Active tracks:\n",
- (int)strlen(inBufferStr), "In buffer ",
- (int)strlen(outBufferStr), "Out buffer ");
- result.append(buffer);
- snprintf(buffer, SIZE, "\t%s %s %d\n", inBufferStr, outBufferStr, mActiveTrackCnt);
- result.append(buffer);
+ const std::string inBufferStr = dumpInOutBuffer(true /* isInput */, mInBuffer);
+ const std::string outBufferStr = dumpInOutBuffer(false /* isInput */, mOutBuffer);
+ result.appendFormat("\t%-*s%-*s Active tracks:\n",
+ (int)inBufferStr.size(), "In buffer ",
+ (int)outBufferStr.size(), "Out buffer ");
+ result.appendFormat("\t%s %s %d\n",
+ inBufferStr.c_str(), outBufferStr.c_str(), mActiveTrackCnt);
write(fd, result.string(), result.size());
for (size_t i = 0; i < numEffects; ++i) {
diff --git a/services/audioflinger/Effects.h b/services/audioflinger/Effects.h
index 1864e0f..eea3208 100644
--- a/services/audioflinger/Effects.h
+++ b/services/audioflinger/Effects.h
@@ -171,8 +171,8 @@
#ifdef FLOAT_EFFECT_CHAIN
bool mSupportsFloat; // effect supports float processing
- sp<EffectBufferHalInterface> mInBuffer16; // Buffers for interacting with HAL at 16 bits
- sp<EffectBufferHalInterface> mOutBuffer16;
+ sp<EffectBufferHalInterface> mInConversionBuffer; // Buffers for HAL conversion if needed.
+ sp<EffectBufferHalInterface> mOutConversionBuffer;
#endif
};
diff --git a/services/audioflinger/FastMixerDumpState.cpp b/services/audioflinger/FastMixerDumpState.cpp
index 6475f22..2e4fb8c 100644
--- a/services/audioflinger/FastMixerDumpState.cpp
+++ b/services/audioflinger/FastMixerDumpState.cpp
@@ -78,7 +78,12 @@
uint32_t bounds = mBounds;
uint32_t newestOpen = bounds & 0xFFFF;
uint32_t oldestClosed = bounds >> 16;
- uint32_t n = (newestOpen - oldestClosed) & 0xFFFF;
+
+ //uint32_t n = (newestOpen - oldestClosed) & 0xFFFF;
+ uint32_t n;
+ __builtin_sub_overflow(newestOpen, oldestClosed, &n);
+ n = n & 0xFFFF;
+
if (n > mSamplingN) {
ALOGE("too many samples %u", n);
n = mSamplingN;
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 1445572..cdd8ca0 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -1102,11 +1102,12 @@
void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type)
{
- for (size_t i = 0; i < mSyncEvents.size(); i++) {
+ for (size_t i = 0; i < mSyncEvents.size();) {
if (mSyncEvents[i]->type() == type) {
mSyncEvents[i]->trigger();
mSyncEvents.removeAt(i);
- i--;
+ } else {
+ ++i;
}
}
}
diff --git a/services/audiopolicy/common/managerdefinitions/include/IOProfile.h b/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
index 118f0d2..ec04ef7 100644
--- a/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
+++ b/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
@@ -34,11 +34,7 @@
{
public:
IOProfile(const String8 &name, audio_port_role_t role)
- : AudioPort(name, AUDIO_PORT_TYPE_MIX, role),
- maxOpenCount((role == AUDIO_PORT_ROLE_SOURCE) ? 1 : 0),
- curOpenCount(0),
- maxActiveCount(1),
- curActiveCount(0) {}
+ : AudioPort(name, AUDIO_PORT_TYPE_MIX, role) {}
// For a Profile aka MixPort, tag name and name are equivalent.
virtual const String8 getTagName() const { return getName(); }
@@ -107,34 +103,6 @@
const DeviceVector &getSupportedDevices() const { return mSupportedDevices; }
- bool canOpenNewIo() {
- if (maxOpenCount == 0 || curOpenCount < maxOpenCount) {
- return true;
- }
- return false;
- }
-
- bool canStartNewIo() {
- if (maxActiveCount == 0 || curActiveCount < maxActiveCount) {
- return true;
- }
- return false;
- }
-
- // Maximum number of input or output streams that can be simultaneously opened for this profile.
- // By convention 0 means no limit. To respect legacy behavior, initialized to 1 for output
- // profiles and 0 for input profiles
- uint32_t maxOpenCount;
- // Number of streams currently opened for this profile.
- uint32_t curOpenCount;
- // Maximum number of input or output streams that can be simultaneously active for this profile.
- // By convention 0 means no limit. To respect legacy behavior, initialized to 0 for output
- // profiles and 1 for input profiles
- uint32_t maxActiveCount;
- // Number of streams currently active for this profile. This is not the number of active clients
- // (AudioTrack or AudioRecord) but the number of active HAL streams.
- uint32_t curActiveCount;
-
private:
DeviceVector mSupportedDevices; // supported devices: this input/output can be routed from/to
};
diff --git a/services/audiopolicy/common/managerdefinitions/include/Serializer.h b/services/audiopolicy/common/managerdefinitions/include/Serializer.h
index 3b0e209..078b582 100644
--- a/services/audiopolicy/common/managerdefinitions/include/Serializer.h
+++ b/services/audiopolicy/common/managerdefinitions/include/Serializer.h
@@ -92,8 +92,6 @@
static const char name[];
static const char role[];
static const char flags[];
- static const char maxOpenCount[];
- static const char maxActiveCount[];
};
typedef IOProfile Element;
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
index 737872d..624e688 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
@@ -236,7 +236,6 @@
mFormat = lConfig.format;
mId = AudioPort::getNextUniqueId();
mIoHandle = *input;
- mProfile->curOpenCount++;
}
return status;
@@ -247,10 +246,6 @@
{
if (mIoHandle != AUDIO_IO_HANDLE_NONE) {
mClientInterface->closeInput(mIoHandle);
- LOG_ALWAYS_FATAL_IF(mProfile->curOpenCount < 1, "%s profile open count %u",
- __FUNCTION__, mProfile->curOpenCount);
- mProfile->curOpenCount--;
- mIoHandle = AUDIO_IO_HANDLE_NONE;
}
}
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index be5a1c1..f96c5bc 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -444,7 +444,6 @@
mFormat = lConfig.format;
mId = AudioPort::getNextUniqueId();
mIoHandle = *output;
- mProfile->curOpenCount++;
}
return status;
@@ -459,11 +458,6 @@
mClientInterface->setParameters(mIoHandle, param.toString());
mClientInterface->closeOutput(mIoHandle);
-
- LOG_ALWAYS_FATAL_IF(mProfile->curOpenCount < 1, "%s profile open count %u",
- __FUNCTION__, mProfile->curOpenCount);
- mProfile->curOpenCount--;
- mIoHandle = AUDIO_IO_HANDLE_NONE;
}
}
diff --git a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
index cc56fb8..32a46f3 100644
--- a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
@@ -252,60 +252,40 @@
sp <HwModule> HwModuleCollection::getModuleFromName(const char *name) const
{
- sp <HwModule> module;
-
- for (size_t i = 0; i < size(); i++)
- {
- if (strcmp(itemAt(i)->getName(), name) == 0) {
- return itemAt(i);
+ for (const auto& module : *this) {
+ if (strcmp(module->getName(), name) == 0) {
+ return module;
}
}
- return module;
+ return nullptr;
}
-
sp <HwModule> HwModuleCollection::getModuleForDevice(audio_devices_t device) const
{
- sp <HwModule> module;
-
- for (size_t i = 0; i < size(); i++) {
- if (itemAt(i)->getHandle() == 0) {
- continue;
- }
- if (audio_is_output_device(device)) {
- for (size_t j = 0; j < itemAt(i)->mOutputProfiles.size(); j++)
- {
- if (itemAt(i)->mOutputProfiles[j]->supportDevice(device)) {
- return itemAt(i);
- }
- }
- } else {
- for (size_t j = 0; j < itemAt(i)->mInputProfiles.size(); j++) {
- if (itemAt(i)->mInputProfiles[j]->supportDevice(device)) {
- return itemAt(i);
- }
+ for (const auto& module : *this) {
+ IOProfileCollection& profiles = audio_is_output_device(device) ?
+ module->mOutputProfiles : module->mInputProfiles;
+ for (const auto& profile : profiles) {
+ if (profile->supportDevice(device)) {
+ return module;
}
}
}
- return module;
+ return nullptr;
}
-sp<DeviceDescriptor> HwModuleCollection::getDeviceDescriptor(const audio_devices_t device,
- const char *device_address,
- const char *device_name,
- bool matchAdress) const
+sp<DeviceDescriptor> HwModuleCollection::getDeviceDescriptor(const audio_devices_t device,
+ const char *device_address,
+ const char *device_name,
+ bool matchAdress) const
{
- String8 address = (device_address == NULL) ? String8("") : String8(device_address);
+ String8 address = (device_address == nullptr) ? String8("") : String8(device_address);
// handle legacy remote submix case where the address was not always specified
if (device_distinguishes_on_address(device) && (address.length() == 0)) {
address = String8("0");
}
- for (size_t i = 0; i < size(); i++) {
- const sp<HwModule> hwModule = itemAt(i);
- if (hwModule->mHandle == 0) {
- continue;
- }
+ for (const auto& hwModule : *this) {
DeviceVector declaredDevices = hwModule->getDeclaredDevices();
DeviceVector deviceList = declaredDevices.getDevicesFromTypeAddr(device, address);
if (!deviceList.isEmpty()) {
@@ -340,4 +320,5 @@
return NO_ERROR;
}
+
} //namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp b/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
index fc89672..74ef4ec 100644
--- a/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
@@ -122,16 +122,6 @@
result.append("\n");
write(fd, result.string(), result.size());
mSupportedDevices.dump(fd, String8("Supported"), 4, false);
-
- result.clear();
- snprintf(buffer, SIZE, "\n - maxOpenCount: %u - curOpenCount: %u\n",
- maxOpenCount, curOpenCount);
- result.append(buffer);
- snprintf(buffer, SIZE, " - maxActiveCount: %u - curActiveCount: %u\n",
- maxActiveCount, curActiveCount);
- result.append(buffer);
-
- write(fd, result.string(), result.size());
}
void IOProfile::log()
diff --git a/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp b/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
index aa589f4..0908ffc 100644
--- a/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
@@ -217,8 +217,6 @@
const char MixPortTraits::Attributes::name[] = "name";
const char MixPortTraits::Attributes::role[] = "role";
const char MixPortTraits::Attributes::flags[] = "flags";
-const char MixPortTraits::Attributes::maxOpenCount[] = "maxOpenCount";
-const char MixPortTraits::Attributes::maxActiveCount[] = "maxActiveCount";
status_t MixPortTraits::deserialize(_xmlDoc *doc, const _xmlNode *child, PtrElement &mixPort,
PtrSerializingCtx /*serializingContext*/)
@@ -261,14 +259,6 @@
mixPort->setFlags(InputFlagConverter::maskFromString(flags));
}
}
- string maxOpenCount = getXmlAttribute(child, Attributes::maxOpenCount);
- if (!maxOpenCount.empty()) {
- convertTo(maxOpenCount, mixPort->maxOpenCount);
- }
- string maxActiveCount = getXmlAttribute(child, Attributes::maxActiveCount);
- if (!maxActiveCount.empty()) {
- convertTo(maxActiveCount, mixPort->maxActiveCount);
- }
// Deserialize children
AudioGainTraits::Collection gains;
deserializeCollection<AudioGainTraits>(doc, child, gains, NULL);
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index b363779..bfefbf8 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -721,12 +721,9 @@
sp<IOProfile> profile;
- for (size_t i = 0; i < mHwModules.size(); i++) {
- if (mHwModules[i]->mHandle == 0) {
- continue;
- }
- for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) {
- sp<IOProfile> curProfile = mHwModules[i]->mOutputProfiles[j];
+ for (const auto& hwModule : mHwModules) {
+ for (size_t j = 0; j < hwModule->mOutputProfiles.size(); j++) {
+ sp<IOProfile> curProfile = hwModule->mOutputProfiles[j];
if (!curProfile->isCompatibleProfile(device, String8(""),
samplingRate, NULL /*updatedSamplingRate*/,
format, NULL /*updatedFormat*/,
@@ -926,29 +923,37 @@
}
if (profile != 0) {
+ sp<SwAudioOutputDescriptor> outputDesc = NULL;
+
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
if (!desc->isDuplicated() && (profile == desc->mProfile)) {
+ outputDesc = desc;
// reuse direct output if currently open by the same client
// and configured with same parameters
- if ((config->sample_rate == desc->mSamplingRate) &&
- audio_formats_match(config->format, desc->mFormat) &&
- (config->channel_mask == desc->mChannelMask) &&
- (session == desc->mDirectClientSession)) {
- desc->mDirectOpenCount++;
- ALOGV("getOutputForDevice() reusing direct output %d for session %d",
- mOutputs.keyAt(i), session);
- return mOutputs.keyAt(i);
+ if ((config->sample_rate == outputDesc->mSamplingRate) &&
+ audio_formats_match(config->format, outputDesc->mFormat) &&
+ (config->channel_mask == outputDesc->mChannelMask)) {
+ if (session == outputDesc->mDirectClientSession) {
+ outputDesc->mDirectOpenCount++;
+ ALOGV("getOutputForDevice() reusing direct output %d for session %d",
+ mOutputs.keyAt(i), session);
+ return mOutputs.keyAt(i);
+ } else {
+ ALOGV("getOutputForDevice() do not reuse direct output because"
+ "current client (%d) is not the same as requesting client (%d)",
+ outputDesc->mDirectClientSession, session);
+ goto non_direct_output;
+ }
}
}
}
-
- if (!profile->canOpenNewIo()) {
- goto non_direct_output;
+ // close direct output if currently open and configured with different parameters
+ if (outputDesc != NULL) {
+ closeOutput(outputDesc->mIoHandle);
}
- sp<SwAudioOutputDescriptor> outputDesc =
- new SwAudioOutputDescriptor(profile, mpClientInterface);
+ outputDesc = new SwAudioOutputDescriptor(profile, mpClientInterface);
status = outputDesc->open(config, device, String8(""), stream, flags, &output);
// only accept an output with the requested parameters
@@ -1106,13 +1111,6 @@
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
- if (!outputDesc->isActive()) {
- if (!outputDesc->mProfile->canStartNewIo()) {
- return INVALID_OPERATION;
- }
- outputDesc->mProfile->curActiveCount++;
- }
-
// Routing?
mOutputRoutes.incRouteActivity(session);
@@ -1140,12 +1138,6 @@
if (status != NO_ERROR) {
mOutputRoutes.decRouteActivity(session);
- if (!outputDesc->isActive()) {
- LOG_ALWAYS_FATAL_IF(outputDesc->mProfile->curActiveCount < 1,
- "%s invalid profile active count %u",
- __FUNCTION__, outputDesc->mProfile->curActiveCount);
- outputDesc->mProfile->curActiveCount--;
- }
return status;
}
// Automatically enable the remote submix input when output is started on a re routing mix
@@ -1334,15 +1326,7 @@
}
}
- status_t status = stopSource(outputDesc, stream, forceDeviceUpdate);
-
- if (status == NO_ERROR && !outputDesc->isActive()) {
- LOG_ALWAYS_FATAL_IF(outputDesc->mProfile->curActiveCount < 1,
- "%s invalid profile active count %u",
- __FUNCTION__, outputDesc->mProfile->curActiveCount);
- outputDesc->mProfile->curActiveCount--;
- }
- return status;
+ return stopSource(outputDesc, stream, forceDeviceUpdate);
}
status_t AudioPolicyManager::stopSource(const sp<AudioOutputDescriptor>& outputDesc,
@@ -1728,10 +1712,6 @@
}
#endif
- if (!profile->canOpenNewIo()) {
- return AUDIO_IO_HANDLE_NONE;
- }
-
sp<AudioInputDescriptor> inputDesc = new AudioInputDescriptor(profile, mpClientInterface);
audio_config_t lConfig = AUDIO_CONFIG_INITIALIZER;
@@ -1969,13 +1949,6 @@
setInputDevice(input, device, true /* force */);
if (inputDesc->getAudioSessionCount(true/*activeOnly*/) == 1) {
- if (!inputDesc->mProfile->canStartNewIo()) {
- mInputRoutes.decRouteActivity(session);
- audioSession->changeActiveCount(-1);
- return INVALID_OPERATION;
- }
- inputDesc->mProfile->curActiveCount++;
-
// if input maps to a dynamic policy with an activity listener, notify of state change
if ((inputDesc->mPolicyMix != NULL)
&& ((inputDesc->mPolicyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0)) {
@@ -2045,11 +2018,6 @@
if (inputDesc->isActive()) {
setInputDevice(input, getNewInputDevice(inputDesc), false /* force */);
} else {
- LOG_ALWAYS_FATAL_IF(inputDesc->mProfile->curActiveCount < 1,
- "%s invalid profile active count %u",
- __FUNCTION__, inputDesc->mProfile->curActiveCount);
- inputDesc->mProfile->curActiveCount--;
-
// if input maps to a dynamic policy with an activity listener, notify of state change
if ((inputDesc->mPolicyMix != NULL)
&& ((inputDesc->mPolicyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0)) {
@@ -2424,23 +2392,16 @@
break;
}
if ((mixes[i].mRouteFlags & MIX_ROUTE_FLAG_LOOP_BACK) == MIX_ROUTE_FLAG_LOOP_BACK) {
- // Loop back through "remote submix"
- if (rSubmixModule == 0) {
- for (size_t j = 0; i < mHwModules.size(); j++) {
- if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[j]->mName) == 0
- && mHwModules[j]->mHandle != 0) {
- rSubmixModule = mHwModules[j];
- break;
- }
- }
- }
-
ALOGV("registerPolicyMixes() mix %zu of %zu is LOOP_BACK", i, mixes.size());
-
if (rSubmixModule == 0) {
- ALOGE(" Unable to find audio module for submix, aborting mix %zu registration", i);
- res = INVALID_OPERATION;
- break;
+ rSubmixModule = mHwModules.getModuleFromName(
+ AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX);
+ if (rSubmixModule == 0) {
+ ALOGE(" Unable to find audio module for submix, aborting mix %zu registration",
+ i);
+ res = INVALID_OPERATION;
+ break;
+ }
}
String8 address = mixes[i].mDeviceAddress;
@@ -2523,18 +2484,13 @@
if ((mixes[i].mRouteFlags & MIX_ROUTE_FLAG_LOOP_BACK) == MIX_ROUTE_FLAG_LOOP_BACK) {
if (rSubmixModule == 0) {
- for (size_t j = 0; i < mHwModules.size(); j++) {
- if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[j]->mName) == 0
- && mHwModules[j]->mHandle != 0) {
- rSubmixModule = mHwModules[j];
- break;
- }
+ rSubmixModule = mHwModules.getModuleFromName(
+ AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX);
+ if (rSubmixModule == 0) {
+ res = INVALID_OPERATION;
+ continue;
}
}
- if (rSubmixModule == 0) {
- res = INVALID_OPERATION;
- continue;
- }
String8 address = mixes[i].mDeviceAddress;
@@ -2611,7 +2567,7 @@
mAvailableOutputDevices.dump(fd, String8("Available output"));
mAvailableInputDevices.dump(fd, String8("Available input"));
- mHwModules.dump(fd);
+ mHwModulesAll.dump(fd);
mOutputs.dump(fd);
mInputs.dump(fd);
mVolumeCurves->dump(fd);
@@ -3555,13 +3511,13 @@
#ifdef USE_XML_AUDIO_POLICY_CONF
mVolumeCurves = new VolumeCurvesCollection();
- AudioPolicyConfig config(mHwModules, mAvailableOutputDevices, mAvailableInputDevices,
+ AudioPolicyConfig config(mHwModulesAll, mAvailableOutputDevices, mAvailableInputDevices,
mDefaultOutputDevice, speakerDrcEnabled,
static_cast<VolumeCurvesCollection *>(mVolumeCurves));
if (deserializeAudioPolicyXmlConfig(config) != NO_ERROR) {
#else
mVolumeCurves = new StreamDescriptorCollection();
- AudioPolicyConfig config(mHwModules, mAvailableOutputDevices, mAvailableInputDevices,
+ AudioPolicyConfig config(mHwModulesAll, mAvailableOutputDevices, mAvailableInputDevices,
mDefaultOutputDevice, speakerDrcEnabled);
if ((ConfigParsingUtils::loadConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE, config) != NO_ERROR) &&
(ConfigParsingUtils::loadConfig(AUDIO_POLICY_CONFIG_FILE, config) != NO_ERROR)) {
@@ -3593,28 +3549,23 @@
// open all output streams needed to access attached devices
audio_devices_t outputDeviceTypes = mAvailableOutputDevices.types();
audio_devices_t inputDeviceTypes = mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN;
- for (size_t i = 0; i < mHwModules.size(); i++) {
- mHwModules[i]->mHandle = mpClientInterface->loadHwModule(mHwModules[i]->getName());
- if (mHwModules[i]->mHandle == 0) {
- ALOGW("could not open HW module %s", mHwModules[i]->getName());
+ for (const auto& hwModule : mHwModulesAll) {
+ hwModule->mHandle = mpClientInterface->loadHwModule(hwModule->getName());
+ if (hwModule->getHandle() == AUDIO_MODULE_HANDLE_NONE) {
+ ALOGW("could not open HW module %s", hwModule->getName());
continue;
}
+ mHwModules.push_back(hwModule);
// open all output streams needed to access attached devices
// except for direct output streams that are only opened when they are actually
// required by an app.
// This also validates mAvailableOutputDevices list
- for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
+ for (size_t j = 0; j < hwModule->mOutputProfiles.size(); j++)
{
- const sp<IOProfile> outProfile = mHwModules[i]->mOutputProfiles[j];
-
- if (!outProfile->canOpenNewIo()) {
- ALOGE("Invalid Output profile max open count %u for profile %s",
- outProfile->maxOpenCount, outProfile->getTagName().c_str());
- continue;
- }
+ const sp<IOProfile> outProfile = hwModule->mOutputProfiles[j];
if (!outProfile->hasSupportedDevices()) {
- ALOGW("Output profile contains no device on module %s", mHwModules[i]->getName());
+ ALOGW("Output profile contains no device on module %s", hwModule->getName());
continue;
}
if ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_TTS) != 0) {
@@ -3648,13 +3599,13 @@
if (status != NO_ERROR) {
ALOGW("Cannot open output stream for device %08x on hw module %s",
outputDesc->mDevice,
- mHwModules[i]->getName());
+ hwModule->getName());
} else {
for (size_t k = 0; k < supportedDevices.size(); k++) {
ssize_t index = mAvailableOutputDevices.indexOf(supportedDevices[k]);
// give a valid ID to an attached device once confirmed it is reachable
if (index >= 0 && !mAvailableOutputDevices[index]->isAttached()) {
- mAvailableOutputDevices[index]->attach(mHwModules[i]);
+ mAvailableOutputDevices[index]->attach(hwModule);
}
}
if (mPrimaryOutput == 0 &&
@@ -3672,18 +3623,12 @@
}
// open input streams needed to access attached devices to validate
// mAvailableInputDevices list
- for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++)
+ for (size_t j = 0; j < hwModule->mInputProfiles.size(); j++)
{
- const sp<IOProfile> inProfile = mHwModules[i]->mInputProfiles[j];
-
- if (!inProfile->canOpenNewIo()) {
- ALOGE("Invalid Input profile max open count %u for profile %s",
- inProfile->maxOpenCount, inProfile->getTagName().c_str());
- continue;
- }
+ const sp<IOProfile> inProfile = hwModule->mInputProfiles[j];
if (!inProfile->hasSupportedDevices()) {
- ALOGW("Input profile contains no device on module %s", mHwModules[i]->getName());
+ ALOGW("Input profile contains no device on module %s", hwModule->getName());
continue;
}
// chose first device present in profile's SupportedDevices also part of
@@ -3712,7 +3657,7 @@
if (index >= 0) {
sp<DeviceDescriptor> devDesc = mAvailableInputDevices[index];
if (!devDesc->isAttached()) {
- devDesc->attach(mHwModules[i]);
+ devDesc->attach(hwModule);
devDesc->importAudioPort(inProfile, true);
}
}
@@ -3721,7 +3666,7 @@
} else {
ALOGW("Cannot open input stream for device %08x on hw module %s",
profileType,
- mHwModules[i]->getName());
+ hwModule->getName());
}
}
}
@@ -3771,6 +3716,7 @@
mOutputs.clear();
mInputs.clear();
mHwModules.clear();
+ mHwModulesAll.clear();
}
status_t AudioPolicyManager::initCheck()
@@ -3842,19 +3788,17 @@
}
// then look for output profiles that can be routed to this device
SortedVector< sp<IOProfile> > profiles;
- for (size_t i = 0; i < mHwModules.size(); i++)
+ for (const auto& hwModule : mHwModules)
{
- if (mHwModules[i]->mHandle == 0) {
- continue;
- }
- for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
+ for (size_t j = 0; j < hwModule->mOutputProfiles.size(); j++)
{
- sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j];
+ sp<IOProfile> profile = hwModule->mOutputProfiles[j];
if (profile->supportDevice(device)) {
if (!device_distinguishes_on_address(device) ||
profile->supportDeviceAddress(address)) {
profiles.add(profile);
- ALOGV("checkOutputsForDevice(): adding profile %zu from module %zu", j, i);
+ ALOGV("checkOutputsForDevice(): adding profile %zu from module %s",
+ j, hwModule->getName());
}
}
}
@@ -3889,12 +3833,6 @@
continue;
}
- if (!profile->canOpenNewIo()) {
- ALOGW("Max Output number %u already opened for this profile %s",
- profile->maxOpenCount, profile->getTagName().c_str());
- continue;
- }
-
ALOGV("opening output for device %08x with params %s profile %p name %s",
device, address.string(), profile.get(), profile->getName().string());
desc = new SwAudioOutputDescriptor(profile, mpClientInterface);
@@ -4024,17 +3962,15 @@
}
}
// Clear any profiles associated with the disconnected device.
- for (size_t i = 0; i < mHwModules.size(); i++)
+ for (const auto& hwModule : mHwModules)
{
- if (mHwModules[i]->mHandle == 0) {
- continue;
- }
- for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
+ for (size_t j = 0; j < hwModule->mOutputProfiles.size(); j++)
{
- sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j];
+ sp<IOProfile> profile = hwModule->mOutputProfiles[j];
if (profile->supportDevice(device)) {
ALOGV("checkOutputsForDevice(): "
- "clearing direct output profile %zu on module %zu", j, i);
+ "clearing direct output profile %zu on module %s",
+ j, hwModule->getName());
profile->clearAudioProfiles();
}
}
@@ -4068,23 +4004,20 @@
// then look for input profiles that can be routed to this device
SortedVector< sp<IOProfile> > profiles;
- for (size_t module_idx = 0; module_idx < mHwModules.size(); module_idx++)
+ for (const auto& hwModule : mHwModules)
{
- if (mHwModules[module_idx]->mHandle == 0) {
- continue;
- }
for (size_t profile_index = 0;
- profile_index < mHwModules[module_idx]->mInputProfiles.size();
+ profile_index < hwModule->mInputProfiles.size();
profile_index++)
{
- sp<IOProfile> profile = mHwModules[module_idx]->mInputProfiles[profile_index];
+ sp<IOProfile> profile = hwModule->mInputProfiles[profile_index];
if (profile->supportDevice(device)) {
if (!device_distinguishes_on_address(device) ||
profile->supportDeviceAddress(address)) {
profiles.add(profile);
- ALOGV("checkInputsForDevice(): adding profile %zu from module %zu",
- profile_index, module_idx);
+ ALOGV("checkInputsForDevice(): adding profile %zu from module %s",
+ profile_index, hwModule->getName());
}
}
}
@@ -4100,7 +4033,6 @@
for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) {
sp<IOProfile> profile = profiles[profile_index];
-
// nothing to do if one input is already opened for this profile
size_t input_index;
for (input_index = 0; input_index < mInputs.size(); input_index++) {
@@ -4116,12 +4048,6 @@
continue;
}
- if (!profile->canOpenNewIo()) {
- ALOGW("Max Input number %u already opened for this profile %s",
- profile->maxOpenCount, profile->getTagName().c_str());
- continue;
- }
-
desc = new AudioInputDescriptor(profile, mpClientInterface);
audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
status_t status = desc->open(nullptr,
@@ -4178,17 +4104,14 @@
}
}
// Clear any profiles associated with the disconnected device.
- for (size_t module_index = 0; module_index < mHwModules.size(); module_index++) {
- if (mHwModules[module_index]->mHandle == 0) {
- continue;
- }
+ for (const auto& hwModule : mHwModules) {
for (size_t profile_index = 0;
- profile_index < mHwModules[module_index]->mInputProfiles.size();
+ profile_index < hwModule->mInputProfiles.size();
profile_index++) {
- sp<IOProfile> profile = mHwModules[module_index]->mInputProfiles[profile_index];
+ sp<IOProfile> profile = hwModule->mInputProfiles[profile_index];
if (profile->supportDevice(device)) {
- ALOGV("checkInputsForDevice(): clearing direct input profile %zu on module %zu",
- profile_index, module_index);
+ ALOGV("checkInputsForDevice(): clearing direct input profile %zu on module %s",
+ profile_index, hwModule->getName());
profile->clearAudioProfiles();
}
}
@@ -5027,14 +4950,11 @@
// TODO: perhaps isCompatibleProfile should return a "matching" score so we can return
// the best matching profile, not the first one.
- for (size_t i = 0; i < mHwModules.size(); i++)
+ for (const auto& hwModule : mHwModules)
{
- if (mHwModules[i]->mHandle == 0) {
- continue;
- }
- for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++)
+ for (size_t j = 0; j < hwModule->mInputProfiles.size(); j++)
{
- sp<IOProfile> profile = mHwModules[i]->mInputProfiles[j];
+ sp<IOProfile> profile = hwModule->mInputProfiles[j];
// profile->log();
if (profile->isCompatibleProfile(device, address, samplingRate,
&samplingRate /*updatedSamplingRate*/,
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index 2d41bd1..b61bc2d 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -266,7 +266,7 @@
{
return mDefaultOutputDevice;
}
-protected:
+
void addOutput(audio_io_handle_t output, const sp<SwAudioOutputDescriptor>& outputDesc);
void removeOutput(audio_io_handle_t output);
void addInput(audio_io_handle_t input, const sp<AudioInputDescriptor>& inputDesc);
@@ -530,7 +530,9 @@
EffectDescriptorCollection mEffects; // list of registered audio effects
bool mA2dpSuspended; // true if A2DP output is suspended
sp<DeviceDescriptor> mDefaultOutputDevice; // output device selected by default at boot time
- HwModuleCollection mHwModules;
+ HwModuleCollection mHwModules; // contains only modules that have been loaded successfully
+ HwModuleCollection mHwModulesAll; // normally not needed, used during construction and for
+ // dumps
volatile int32_t mAudioPortGeneration;
diff --git a/services/mediacodec/Android.mk b/services/mediacodec/Android.mk
index 9348ecd..caa0703 100644
--- a/services/mediacodec/Android.mk
+++ b/services/mediacodec/Android.mk
@@ -26,10 +26,6 @@
LOCAL_32_BIT_ONLY := true
LOCAL_INIT_RC := android.hardware.media.omx@1.0-service.rc
-ifeq ($(PRODUCT_FULL_TREBLE),true)
-LOCAL_CFLAGS += -DUSE_VNDBINDER
-endif
-
include $(BUILD_EXECUTABLE)
# service seccomp policy
diff --git a/services/mediacodec/main_codecservice.cpp b/services/mediacodec/main_codecservice.cpp
index 6f14a42..701ca6e 100644
--- a/services/mediacodec/main_codecservice.cpp
+++ b/services/mediacodec/main_codecservice.cpp
@@ -40,10 +40,8 @@
signal(SIGPIPE, SIG_IGN);
SetUpMinijail(kSystemSeccompPolicyPath, kVendorSeccompPolicyPath);
-#ifdef USE_VNDBINDER
android::ProcessState::initWithDriver("/dev/vndbinder");
android::ProcessState::self()->startThreadPool();
-#endif // USE_VNDBINDER
::android::hardware::configureRpcThreadpool(64, false);