Issue 2265163: Audio still reported routed through earpiece on sholes
This is a second attempt to fix the audio routed to earpiece syndrom.
The root cause identified this time is the crash of an application having an active AudioTrack playing on the VOICE_CALL stream type.
When this happens, the AudioTrack destructor is not called and the audio policy manager is not notified of the track stop.
Results a situation where the VOICE_CALL stream is considered as always in use by audio policy manager which makes that audio is routed to earpiece.
The fix consists in moving the track start/stop/close notification to audio policiy manager from AudioTrack to AudioFlinger Track objet.
The net result is that in the case of a client application crash, the AudioFlinger TrackHandle object (which implements the remote side of the IAudioTrack binder interface) destructor is called which in turn destroys the Track object and we can notify the audio policy manager of the track stop and removal.
The same modification is made for AudioRecord although no bug related to record has been reported yet.
Also fixed a potential problem if record stop is called while the record thread is exiting.
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index e63c0d2..c3828f0 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -46,7 +46,7 @@
// ---------------------------------------------------------------------------
AudioRecord::AudioRecord()
- : mStatus(NO_INIT), mInput(0)
+ : mStatus(NO_INIT)
{
}
@@ -60,7 +60,7 @@
callback_t cbf,
void* user,
int notificationFrames)
- : mStatus(NO_INIT), mInput(0)
+ : mStatus(NO_INIT)
{
mStatus = set(inputSource, sampleRate, format, channels,
frameCount, flags, cbf, user, notificationFrames);
@@ -79,7 +79,6 @@
}
mAudioRecord.clear();
IPCThreadState::self()->flushCommands();
- AudioSystem::releaseInput(mInput);
}
}
@@ -123,9 +122,9 @@
}
int channelCount = AudioSystem::popCount(channels);
- mInput = AudioSystem::getInput(inputSource,
+ audio_io_handle_t input = AudioSystem::getInput(inputSource,
sampleRate, format, channels, (AudioSystem::audio_in_acoustics)flags);
- if (mInput == 0) {
+ if (input == 0) {
LOGE("Could not get audio output for stream type %d", inputSource);
return BAD_VALUE;
}
@@ -168,7 +167,7 @@
// create the IAudioRecord
status_t status = openRecord(sampleRate, format, channelCount,
- frameCount, flags);
+ frameCount, flags, input);
if (status != NO_ERROR) {
return status;
@@ -187,6 +186,7 @@
// Update buffer size in case it has been limited by AudioFlinger during track creation
mFrameCount = mCblk->frameCount;
mChannelCount = (uint8_t)channelCount;
+ mChannels = channels;
mActive = 0;
mCbf = cbf;
mNotificationFrames = notificationFrames;
@@ -265,29 +265,28 @@
}
if (android_atomic_or(1, &mActive) == 0) {
- ret = AudioSystem::startInput(mInput);
- if (ret == NO_ERROR) {
- ret = mAudioRecord->start();
- if (ret == DEAD_OBJECT) {
- LOGV("start() dead IAudioRecord: creating a new one");
- ret = openRecord(mCblk->sampleRate, mFormat, mChannelCount,
- mFrameCount, mFlags);
- }
+ ret = mAudioRecord->start();
+ if (ret == DEAD_OBJECT) {
+ LOGV("start() dead IAudioRecord: creating a new one");
+ ret = openRecord(mCblk->sampleRate, mFormat, mChannelCount,
+ mFrameCount, mFlags, getInput());
if (ret == NO_ERROR) {
- mNewPosition = mCblk->user + mUpdatePeriod;
- mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
- mCblk->waitTimeMs = 0;
- if (t != 0) {
- t->run("ClientRecordThread", THREAD_PRIORITY_AUDIO_CLIENT);
- } else {
- setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT);
- }
- } else {
- LOGV("start() failed");
- AudioSystem::stopInput(mInput);
- android_atomic_and(~1, &mActive);
+ ret = mAudioRecord->start();
}
}
+ if (ret == NO_ERROR) {
+ mNewPosition = mCblk->user + mUpdatePeriod;
+ mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
+ mCblk->waitTimeMs = 0;
+ if (t != 0) {
+ t->run("ClientRecordThread", THREAD_PRIORITY_AUDIO_CLIENT);
+ } else {
+ setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT);
+ }
+ } else {
+ LOGV("start() failed");
+ android_atomic_and(~1, &mActive);
+ }
}
if (t != 0) {
@@ -318,7 +317,6 @@
} else {
setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
}
- AudioSystem::stopInput(mInput);
}
if (t != 0) {
@@ -395,7 +393,8 @@
int format,
int channelCount,
int frameCount,
- uint32_t flags)
+ uint32_t flags,
+ audio_io_handle_t input)
{
status_t status;
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
@@ -403,7 +402,7 @@
return NO_INIT;
}
- sp<IAudioRecord> record = audioFlinger->openRecord(getpid(), mInput,
+ sp<IAudioRecord> record = audioFlinger->openRecord(getpid(), input,
sampleRate, format,
channelCount,
frameCount,
@@ -425,7 +424,8 @@
mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
mCblk->out = 0;
-
+ mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
+ mCblk->waitTimeMs = 0;
return NO_ERROR;
}
@@ -466,10 +466,10 @@
if (result == DEAD_OBJECT) {
LOGW("obtainBuffer() dead IAudioRecord: creating a new one");
result = openRecord(cblk->sampleRate, mFormat, mChannelCount,
- mFrameCount, mFlags);
+ mFrameCount, mFlags, getInput());
if (result == NO_ERROR) {
cblk = mCblk;
- cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
+ mAudioRecord->start();
}
}
cblk->lock.lock();
@@ -516,6 +516,14 @@
cblk->stepUser(audioBuffer->frameCount);
}
+audio_io_handle_t AudioRecord::getInput()
+{
+ return AudioSystem::getInput(mInputSource,
+ mCblk->sampleRate,
+ mFormat, mChannels,
+ (AudioSystem::audio_in_acoustics)mFlags);
+}
+
// -------------------------------------------------------------------------
ssize_t AudioRecord::read(void* buffer, size_t userSize)
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index cedd79d..ad0f42e 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -97,7 +97,6 @@
}
mAudioTrack.clear();
IPCThreadState::self()->flushCommands();
- AudioSystem::releaseOutput(getOutput());
}
}
@@ -318,8 +317,6 @@
}
if (android_atomic_or(1, &mActive) == 0) {
- audio_io_handle_t output = getOutput();
- AudioSystem::startOutput(output, (AudioSystem::stream_type)mStreamType);
mNewPosition = mCblk->server + mUpdatePeriod;
mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
mCblk->waitTimeMs = 0;
@@ -333,10 +330,13 @@
if (status == DEAD_OBJECT) {
LOGV("start() dead IAudioTrack: creating a new one");
status = createTrack(mStreamType, mCblk->sampleRate, mFormat, mChannelCount,
- mFrameCount, mFlags, mSharedBuffer, output);
- mNewPosition = mCblk->server + mUpdatePeriod;
- mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
- mCblk->waitTimeMs = 0;
+ mFrameCount, mFlags, mSharedBuffer, getOutput());
+ if (status == NO_ERROR) {
+ status = mAudioTrack->start();
+ if (status == NO_ERROR) {
+ mNewPosition = mCblk->server + mUpdatePeriod;
+ }
+ }
}
if (status != NO_ERROR) {
LOGV("start() failed");
@@ -346,7 +346,6 @@
} else {
setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
}
- AudioSystem::stopOutput(output, (AudioSystem::stream_type)mStreamType);
}
}
@@ -383,7 +382,6 @@
} else {
setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
}
- AudioSystem::stopOutput(getOutput(), (AudioSystem::stream_type)mStreamType);
}
if (t != 0) {
@@ -418,9 +416,7 @@
{
LOGV("pause");
if (android_atomic_and(~1, &mActive) == 1) {
- mActive = 0;
mAudioTrack->pause();
- AudioSystem::stopOutput(getOutput(), (AudioSystem::stream_type)mStreamType);
}
}
@@ -658,7 +654,8 @@
}
mCblk->volumeLR = (int32_t(int16_t(mVolume[LEFT] * 0x1000)) << 16) | int16_t(mVolume[RIGHT] * 0x1000);
-
+ mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
+ mCblk->waitTimeMs = 0;
return NO_ERROR;
}
@@ -709,6 +706,7 @@
if (result == NO_ERROR) {
cblk = mCblk;
cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
+ mAudioTrack->start();
}
}
cblk->lock.lock();