Merge "Camera: delete preview callback when preview size is changed"
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index 45134c4..3d839fc 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -486,12 +486,11 @@
int mSessionId;
transfer_type mTransfer;
- audio_io_handle_t mInput; // returned by AudioSystem::getInput()
-
- // Next 3 fields may be changed if IAudioRecord is re-created, but always != 0
+ // Next 4 fields may be changed if IAudioRecord is re-created, but always != 0
sp<IAudioRecord> mAudioRecord;
sp<IMemory> mCblkMemory;
audio_track_cblk_t* mCblk; // re-load after mLock.unlock()
+ audio_io_handle_t mInput; // returned by AudioSystem::getInput()
int mPreviousPriority; // before start()
SchedPolicy mPreviousSchedulingGroup;
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 644e55c..3a60217 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -454,7 +454,7 @@
* Returned value:
* handle on audio hardware output
*/
- audio_io_handle_t getOutput();
+ audio_io_handle_t getOutput() const;
/* Returns the unique session ID associated with this track.
*
@@ -634,20 +634,12 @@
// caller must hold lock on mLock for all _l methods
- status_t createTrack_l(audio_stream_type_t streamType,
- uint32_t sampleRate,
- audio_format_t format,
- size_t frameCount,
- audio_output_flags_t flags,
- const sp<IMemory>& sharedBuffer,
- audio_io_handle_t output,
- size_t epoch);
+ status_t createTrack_l(size_t epoch);
// can only be called when mState != STATE_ACTIVE
void flush_l();
void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount);
- audio_io_handle_t getOutput_l();
// FIXME enum is faster than strcmp() for parameter 'from'
status_t restoreTrack_l(const char *from);
@@ -655,10 +647,11 @@
bool isOffloaded_l() const
{ return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; }
- // Next 3 fields may be changed if IAudioTrack is re-created, but always != 0
+ // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0
sp<IAudioTrack> mAudioTrack;
sp<IMemory> mCblkMemory;
audio_track_cblk_t* mCblk; // re-load after mLock.unlock()
+ audio_io_handle_t mOutput; // returned by AudioSystem::getOutput()
sp<AudioTrackThread> mAudioTrackThread;
@@ -763,7 +756,6 @@
sp<DeathNotifier> mDeathNotifier;
uint32_t mSequence; // incremented for each new IAudioTrack attempt
- audio_io_handle_t mOutput; // cached output io handle
int mClientUid;
};
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index a794e87..4bd111a 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -64,7 +64,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
track_flags_t *flags,
const sp<IMemory>& sharedBuffer,
// On successful return, AudioFlinger takes over the handle
@@ -88,7 +88,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
track_flags_t *flags,
pid_t tid, // -1 means unused, otherwise must be valid non-0
int *sessionId,
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 85862a8..b5a4c0b 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -96,11 +96,7 @@
// The value should be used "for entertainment purposes only",
// which means don't make important decisions based on it.
- size_t frameCount_; // used during creation to pass actual track buffer size
- // from AudioFlinger to client, and not referenced again
- // FIXME remove here and replace by createTrack() in/out
- // parameter
- // renamed to "_" to detect incorrect use
+ uint32_t mPad1; // unused
volatile int32_t mFutex; // event flag: down (P) by client,
// up (V) by server or binderDied() or interrupt()
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 6565a16..a999e7e 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -244,7 +244,7 @@
// create the IAudioRecord
status = openRecord_l(0 /*epoch*/);
- if (status) {
+ if (status != NO_ERROR) {
return status;
}
@@ -255,9 +255,6 @@
mStatus = NO_ERROR;
- // Update buffer size in case it has been limited by AudioFlinger during track creation
- mFrameCount = mCblk->frameCount_;
-
mActive = false;
mCbf = cbf;
mRefreshRemaining = true;
@@ -466,12 +463,17 @@
ALOGE("Could not get audio input for record source %d", mInputSource);
return BAD_VALUE;
}
+ {
+ // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger,
+ // we must release it ourselves if anything goes wrong.
+ size_t temp = mFrameCount; // temp may be replaced by a revised value of frameCount,
+ // but we will still need the original value also
int originalSessionId = mSessionId;
sp<IAudioRecord> record = audioFlinger->openRecord(input,
mSampleRate, mFormat,
mChannelMask,
- mFrameCount,
+ &temp,
&trackFlags,
tid,
&mSessionId,
@@ -481,9 +483,11 @@
if (record == 0 || status != NO_ERROR) {
ALOGE("AudioFlinger could not create record track, status: %d", status);
- AudioSystem::releaseInput(input);
- return status;
+ goto release;
}
+ // AudioFlinger now owns the reference to the I/O handle,
+ // so we are no longer responsible for releasing it.
+
sp<IMemory> iMem = record->getCblk();
if (iMem == 0) {
ALOGE("Could not get control block");
@@ -498,11 +502,19 @@
mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this);
mDeathNotifier.clear();
}
+
+ // We retain a copy of the I/O handle, but don't own the reference
mInput = input;
mAudioRecord = record;
mCblkMemory = iMem;
audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
mCblk = cblk;
+ // note that temp is the (possibly revised) value of mFrameCount
+ if (temp < mFrameCount || (mFrameCount == 0 && temp == 0)) {
+ ALOGW("Requested frameCount %u but received frameCount %u", mFrameCount, temp);
+ }
+ mFrameCount = temp;
+
// FIXME missing fast track frameCount logic
mAwaitBoost = false;
if (mFlags & AUDIO_INPUT_FLAG_FAST) {
@@ -535,6 +547,14 @@
mAudioRecord->asBinder()->linkToDeath(mDeathNotifier, this);
return NO_ERROR;
+ }
+
+release:
+ AudioSystem::releaseInput(input);
+ if (status == NO_ERROR) {
+ status = NO_INIT;
+ }
+ return status;
}
status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 72be5ca..f61a265 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -245,8 +245,6 @@
return INVALID_OPERATION;
}
- mOutput = 0;
-
// handle default values first.
if (streamType == AUDIO_STREAM_DEFAULT) {
streamType = AUDIO_STREAM_MUSIC;
@@ -319,17 +317,6 @@
mFrameSizeAF = sizeof(uint8_t);
}
- audio_io_handle_t output = AudioSystem::getOutput(
- streamType,
- sampleRate, format, channelMask,
- flags,
- offloadInfo);
-
- if (output == 0) {
- ALOGE("Could not get audio output for stream type %d", streamType);
- return BAD_VALUE;
- }
-
// Make copy of input parameter offloadInfo so that in the future:
// (a) createTrack_l doesn't need it as an input parameter
// (b) we can support re-creation of offloaded tracks
@@ -363,14 +350,7 @@
}
// create the IAudioTrack
- status = createTrack_l(streamType,
- sampleRate,
- format,
- frameCount,
- flags,
- sharedBuffer,
- output,
- 0 /*epoch*/);
+ status = createTrack_l(0 /*epoch*/);
if (status != NO_ERROR) {
if (mAudioTrackThread != 0) {
@@ -379,9 +359,15 @@
mAudioTrackThread.clear();
}
// Use of direct and offloaded output streams is ref counted by audio policy manager.
+#if 0 // FIXME This should no longer be needed
+ //Use of direct and offloaded output streams is ref counted by audio policy manager.
// As getOutput was called above and resulted in an output stream to be opened,
// we need to release it.
- AudioSystem::releaseOutput(output);
+ if (mOutput != 0) {
+ AudioSystem::releaseOutput(mOutput);
+ mOutput = 0;
+ }
+#endif
return status;
}
@@ -397,7 +383,6 @@
mSequence = 1;
mObservedSequence = mSequence;
mInUnderrun = false;
- mOutput = output;
return NO_ERROR;
}
@@ -821,23 +806,12 @@
return NO_ERROR;
}
-audio_io_handle_t AudioTrack::getOutput()
+audio_io_handle_t AudioTrack::getOutput() const
{
AutoMutex lock(mLock);
return mOutput;
}
-// must be called with mLock held
-audio_io_handle_t AudioTrack::getOutput_l()
-{
- if (mOutput) {
- return mOutput;
- } else {
- return AudioSystem::getOutput(mStreamType,
- mSampleRate, mFormat, mChannelMask, mFlags);
- }
-}
-
status_t AudioTrack::attachAuxEffect(int effectId)
{
AutoMutex lock(mLock);
@@ -851,15 +825,7 @@
// -------------------------------------------------------------------------
// must be called with mLock held
-status_t AudioTrack::createTrack_l(
- audio_stream_type_t streamType,
- uint32_t sampleRate,
- audio_format_t format,
- size_t frameCount,
- audio_output_flags_t flags,
- const sp<IMemory>& sharedBuffer,
- audio_io_handle_t output,
- size_t epoch)
+status_t AudioTrack::createTrack_l(size_t epoch)
{
status_t status;
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
@@ -868,41 +834,52 @@
return NO_INIT;
}
+ audio_io_handle_t output = AudioSystem::getOutput(mStreamType, mSampleRate, mFormat,
+ mChannelMask, mFlags, mOffloadInfo);
+ if (output == 0) {
+ 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);
+ return BAD_VALUE;
+ }
+ {
+ // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger,
+ // we must release it ourselves if anything goes wrong.
+
// Not all of these values are needed under all conditions, but it is easier to get them all
uint32_t afLatency;
- status = AudioSystem::getLatency(output, streamType, &afLatency);
+ status = AudioSystem::getLatency(output, mStreamType, &afLatency);
if (status != NO_ERROR) {
ALOGE("getLatency(%d) failed status %d", output, status);
- return NO_INIT;
+ goto release;
}
size_t afFrameCount;
- status = AudioSystem::getFrameCount(output, streamType, &afFrameCount);
+ status = AudioSystem::getFrameCount(output, mStreamType, &afFrameCount);
if (status != NO_ERROR) {
- ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status);
- return NO_INIT;
+ ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, mStreamType, status);
+ goto release;
}
uint32_t afSampleRate;
- status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate);
+ status = AudioSystem::getSamplingRate(output, mStreamType, &afSampleRate);
if (status != NO_ERROR) {
- ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType, status);
- return NO_INIT;
+ ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, mStreamType, status);
+ goto release;
}
// Client decides whether the track is TIMED (see below), but can only express a preference
// for FAST. Server will perform additional tests.
- if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !(
+ if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) && !(
// either of these use cases:
// use case 1: shared buffer
- (sharedBuffer != 0) ||
+ (mSharedBuffer != 0) ||
// use case 2: callback handler
(mCbf != NULL))) {
ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client");
// once denied, do not request again if IAudioTrack is re-created
- flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
- mFlags = flags;
+ mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
}
ALOGV("createTrack_l() output %d afLatency %d", output, afLatency);
@@ -913,43 +890,45 @@
// n = 3 normal track, with sample rate conversion
// (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering)
// n > 3 very high latency or very small notification interval; nBuffering is ignored
- const uint32_t nBuffering = (sampleRate == afSampleRate) ? 2 : 3;
+ const uint32_t nBuffering = (mSampleRate == afSampleRate) ? 2 : 3;
mNotificationFramesAct = mNotificationFramesReq;
- if (!audio_is_linear_pcm(format)) {
+ size_t frameCount = mReqFrameCount;
+ if (!audio_is_linear_pcm(mFormat)) {
- if (sharedBuffer != 0) {
+ if (mSharedBuffer != 0) {
// Same comment as below about ignoring frameCount parameter for set()
- frameCount = sharedBuffer->size();
+ frameCount = mSharedBuffer->size();
} else if (frameCount == 0) {
frameCount = afFrameCount;
}
if (mNotificationFramesAct != frameCount) {
mNotificationFramesAct = frameCount;
}
- } else if (sharedBuffer != 0) {
+ } else if (mSharedBuffer != 0) {
// Ensure that buffer alignment matches channel count
// 8-bit data in shared memory is not currently supported by AudioFlinger
- size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2;
+ size_t alignment = /* mFormat == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2;
if (mChannelCount > 1) {
// More than 2 channels does not require stronger alignment than stereo
alignment <<= 1;
}
- if (((size_t)sharedBuffer->pointer() & (alignment - 1)) != 0) {
+ if (((size_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) {
ALOGE("Invalid buffer alignment: address %p, channel count %u",
- sharedBuffer->pointer(), mChannelCount);
- return BAD_VALUE;
+ mSharedBuffer->pointer(), mChannelCount);
+ status = BAD_VALUE;
+ goto release;
}
// When initializing a shared buffer AudioTrack via constructors,
// there's no frameCount parameter.
// But when initializing a shared buffer AudioTrack via set(),
// there _is_ a frameCount parameter. We silently ignore it.
- frameCount = sharedBuffer->size()/mChannelCount/sizeof(int16_t);
+ frameCount = mSharedBuffer->size()/mChannelCount/sizeof(int16_t);
- } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
+ } else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) {
// FIXME move these calculations and associated checks to server
@@ -961,10 +940,10 @@
minBufCount = nBuffering;
}
- size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
+ size_t minFrameCount = (afFrameCount*mSampleRate*minBufCount)/afSampleRate;
ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u"
", afLatency=%d",
- minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency);
+ minFrameCount, afFrameCount, minBufCount, mSampleRate, afSampleRate, afLatency);
if (frameCount == 0) {
frameCount = minFrameCount;
@@ -989,26 +968,28 @@
}
pid_t tid = -1;
- if (flags & AUDIO_OUTPUT_FLAG_FAST) {
+ if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
trackFlags |= IAudioFlinger::TRACK_FAST;
if (mAudioTrackThread != 0) {
tid = mAudioTrackThread->getTid();
}
}
- if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+ if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
trackFlags |= IAudioFlinger::TRACK_OFFLOAD;
}
- sp<IAudioTrack> track = audioFlinger->createTrack(streamType,
- sampleRate,
+ 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,
+ mSampleRate,
// AudioFlinger only sees 16-bit PCM
- format == AUDIO_FORMAT_PCM_8_BIT ?
- AUDIO_FORMAT_PCM_16_BIT : format,
+ mFormat == AUDIO_FORMAT_PCM_8_BIT ?
+ AUDIO_FORMAT_PCM_16_BIT : mFormat,
mChannelMask,
- frameCount,
+ &temp,
&trackFlags,
- sharedBuffer,
+ mSharedBuffer,
output,
tid,
&mSessionId,
@@ -1018,8 +999,11 @@
if (track == 0) {
ALOGE("AudioFlinger could not create track, status: %d", status);
- return status;
+ goto release;
}
+ // AudioFlinger now owns the reference to the I/O handle,
+ // so we are no longer responsible for releasing it.
+
sp<IMemory> iMem = track->getCblk();
if (iMem == 0) {
ALOGE("Could not get control block");
@@ -1039,7 +1023,7 @@
mCblkMemory = iMem;
audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
mCblk = cblk;
- size_t temp = cblk->frameCount_;
+ // note that temp is the (possibly revised) value of frameCount
if (temp < frameCount || (frameCount == 0 && temp == 0)) {
// 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
@@ -1048,11 +1032,11 @@
}
frameCount = temp;
mAwaitBoost = false;
- if (flags & AUDIO_OUTPUT_FLAG_FAST) {
+ if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
if (trackFlags & IAudioFlinger::TRACK_FAST) {
ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount);
mAwaitBoost = true;
- if (sharedBuffer == 0) {
+ if (mSharedBuffer == 0) {
// Theoretically double-buffering is not required for fast tracks,
// due to tighter scheduling. But in practice, to accommodate kernels with
// scheduling jitter, and apps with computation jitter, we use double-buffering.
@@ -1063,26 +1047,27 @@
} else {
ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount);
// once denied, do not request again if IAudioTrack is re-created
- flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
- mFlags = flags;
- if (sharedBuffer == 0) {
+ mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
+ if (mSharedBuffer == 0) {
if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) {
mNotificationFramesAct = frameCount/nBuffering;
}
}
}
}
- if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+ if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) {
ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful");
} else {
ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server");
- flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
- mFlags = flags;
- return NO_INIT;
+ mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
+ // 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;
mRefreshRemaining = true;
// Starting address of buffers in shared memory. If there is a shared buffer, buffers
@@ -1090,15 +1075,15 @@
// immediately after the control block. This address is for the mapping within client
// address space. AudioFlinger::TrackBase::mBuffer is for the server address space.
void* buffers;
- if (sharedBuffer == 0) {
+ if (mSharedBuffer == 0) {
buffers = (char*)cblk + sizeof(audio_track_cblk_t);
} else {
- buffers = sharedBuffer->pointer();
+ buffers = mSharedBuffer->pointer();
}
mAudioTrack->attachAuxEffect(mAuxEffectId);
// FIXME don't believe this lie
- mLatency = afLatency + (1000*frameCount) / sampleRate;
+ mLatency = afLatency + (1000*frameCount) / mSampleRate;
mFrameCount = frameCount;
// If IAudioTrack is re-created, don't let the requested frameCount
// decrease. This can confuse clients that cache frameCount().
@@ -1107,7 +1092,7 @@
}
// update proxy
- if (sharedBuffer == 0) {
+ if (mSharedBuffer == 0) {
mStaticProxy.clear();
mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF);
} else {
@@ -1125,6 +1110,14 @@
mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this);
return NO_ERROR;
+ }
+
+release:
+ AudioSystem::releaseOutput(output);
+ if (status == NO_ERROR) {
+ status = NO_INIT;
+ }
+ return status;
}
status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
@@ -1706,7 +1699,7 @@
status_t result;
// refresh the audio configuration cache in this process to make sure we get new
- // output parameters in getOutput_l() and createTrack_l()
+ // output parameters in createTrack_l()
AudioSystem::clearAudioConfigCache();
if (isOffloaded_l()) {
@@ -1714,10 +1707,6 @@
return DEAD_OBJECT;
}
- // force new output query from audio policy manager;
- mOutput = 0;
- audio_io_handle_t output = getOutput_l();
-
// if the new IAudioTrack is created, createTrack_l() will modify the
// following member variables: mAudioTrack, mCblkMemory and mCblk.
// It will also delete the strong references on previous IAudioTrack and IMemory
@@ -1725,14 +1714,7 @@
// take the frames that will be lost by track recreation into account in saved position
size_t position = mProxy->getPosition() + mProxy->getFramesFilled();
size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0;
- result = createTrack_l(mStreamType,
- mSampleRate,
- mFormat,
- mReqFrameCount, // so that frame count never goes down
- mFlags,
- mSharedBuffer,
- output,
- position /*epoch*/);
+ result = createTrack_l(position /*epoch*/);
if (result == NO_ERROR) {
// continue playback from last known position, but
@@ -1761,9 +1743,15 @@
}
if (result != NO_ERROR) {
// Use of direct and offloaded output streams is ref counted by audio policy manager.
+#if 0 // FIXME This should no longer be needed
+ //Use of direct and offloaded output streams is ref counted by audio policy manager.
// As getOutput was called above and resulted in an output stream to be opened,
// we need to release it.
- AudioSystem::releaseOutput(output);
+ if (mOutput != 0) {
+ AudioSystem::releaseOutput(mOutput);
+ mOutput = 0;
+ }
+#endif
ALOGW("restoreTrack_l() failed status %d", result);
mState = STATE_STOPPED;
}
diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp
index 7a1e207..21018a0 100644
--- a/media/libmedia/AudioTrackShared.cpp
+++ b/media/libmedia/AudioTrackShared.cpp
@@ -26,7 +26,7 @@
namespace android {
audio_track_cblk_t::audio_track_cblk_t()
- : mServer(0), frameCount_(0), mFutex(0), mMinimum(0),
+ : mServer(0), mFutex(0), mMinimum(0),
mVolumeLR(0x10001000), mSampleRate(0), mSendLevel(0), mFlags(0)
{
memset(&u, 0, sizeof(u));
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index 86a4d74..f3f3e15 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -89,7 +89,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
track_flags_t *flags,
const sp<IMemory>& sharedBuffer,
audio_io_handle_t output,
@@ -106,6 +106,7 @@
data.writeInt32(sampleRate);
data.writeInt32(format);
data.writeInt32(channelMask);
+ size_t frameCount = pFrameCount != NULL ? *pFrameCount : 0;
data.writeInt32(frameCount);
track_flags_t lFlags = flags != NULL ? *flags : (track_flags_t) TRACK_DEFAULT;
data.writeInt32(lFlags);
@@ -127,6 +128,10 @@
if (lStatus != NO_ERROR) {
ALOGE("createTrack error: %s", strerror(-lStatus));
} else {
+ frameCount = reply.readInt32();
+ if (pFrameCount != NULL) {
+ *pFrameCount = frameCount;
+ }
lFlags = reply.readInt32();
if (flags != NULL) {
*flags = lFlags;
@@ -161,7 +166,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
track_flags_t *flags,
pid_t tid,
int *sessionId,
@@ -174,6 +179,7 @@
data.writeInt32(sampleRate);
data.writeInt32(format);
data.writeInt32(channelMask);
+ size_t frameCount = pFrameCount != NULL ? *pFrameCount : 0;
data.writeInt32(frameCount);
track_flags_t lFlags = flags != NULL ? *flags : (track_flags_t) TRACK_DEFAULT;
data.writeInt32(lFlags);
@@ -187,6 +193,10 @@
if (lStatus != NO_ERROR) {
ALOGE("openRecord error: %s", strerror(-lStatus));
} else {
+ frameCount = reply.readInt32();
+ if (pFrameCount != NULL) {
+ *pFrameCount = frameCount;
+ }
lFlags = reply.readInt32();
if (flags != NULL) {
*flags = lFlags;
@@ -807,10 +817,11 @@
} else {
track = createTrack(
(audio_stream_type_t) streamType, sampleRate, format,
- channelMask, frameCount, &flags, buffer, output, tid,
+ channelMask, &frameCount, &flags, buffer, output, tid,
&sessionId, name, clientUid, &status);
LOG_ALWAYS_FATAL_IF((track != 0) != (status == NO_ERROR));
}
+ reply->writeInt32(frameCount);
reply->writeInt32(flags);
reply->writeInt32(sessionId);
reply->writeString8(name);
@@ -830,8 +841,9 @@
int sessionId = data.readInt32();
status_t status;
sp<IAudioRecord> record = openRecord(input,
- sampleRate, format, channelMask, frameCount, &flags, tid, &sessionId, &status);
+ sampleRate, format, channelMask, &frameCount, &flags, tid, &sessionId, &status);
LOG_ALWAYS_FATAL_IF((record != 0) != (status == NO_ERROR));
+ reply->writeInt32(frameCount);
reply->writeInt32(flags);
reply->writeInt32(sessionId);
reply->writeInt32(status);
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 1257161..f9cc17b 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -481,7 +481,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *frameCount,
IAudioFlinger::track_flags_t *flags,
const sp<IMemory>& sharedBuffer,
audio_io_handle_t output,
@@ -1277,7 +1277,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *frameCount,
IAudioFlinger::track_flags_t *flags,
pid_t tid,
int *sessionId,
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 0ab43e0..e0d1404 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -102,7 +102,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
IAudioFlinger::track_flags_t *flags,
const sp<IMemory>& sharedBuffer,
audio_io_handle_t output,
@@ -117,7 +117,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
IAudioFlinger::track_flags_t *flags,
pid_t tid,
int *sessionId,
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index d5a0e21..2b37761 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -1190,7 +1190,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
IAudioFlinger::track_flags_t *flags,
@@ -1198,6 +1198,7 @@
int uid,
status_t *status)
{
+ size_t frameCount = *pFrameCount;
sp<Track> track;
status_t lStatus;
@@ -1266,6 +1267,7 @@
}
}
}
+ *pFrameCount = frameCount;
if (mType == DIRECT) {
if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
@@ -3038,10 +3040,12 @@
// add frames already consumed but not yet released by the resampler
// because mAudioTrackServerProxy->framesReady() will include these frames
desiredFrames += mAudioMixer->getUnreleasedFrames(track->name());
+#if 0
// the minimum track buffer size is normally twice the number of frames necessary
// to fill one buffer and the resampler should not leave more than one buffer worth
// of unreleased frames after each pass, but just in case...
ALOG_ASSERT(desiredFrames <= cblk->frameCount_);
+#endif
}
uint32_t minFrames = 1;
if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() &&
@@ -4772,13 +4776,14 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
int sessionId,
int uid,
IAudioFlinger::track_flags_t *flags,
pid_t tid,
status_t *status)
{
+ size_t frameCount = *pFrameCount;
sp<RecordTrack> track;
status_t lStatus;
@@ -4837,6 +4842,7 @@
}
}
}
+ *pFrameCount = frameCount;
// FIXME use flags and tid similar to createTrack_l()
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 2b749fa..8df6f94 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -422,7 +422,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
IAudioFlinger::track_flags_t *flags,
@@ -880,7 +880,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
int sessionId,
int uid,
IAudioFlinger::track_flags_t *flags,
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index a5b9ac5..d8d7790 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -133,7 +133,6 @@
if (mCblk != NULL) {
new(mCblk) audio_track_cblk_t();
// clear all buffers
- mCblk->frameCount_ = frameCount;
if (sharedBuffer == 0) {
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
memset(mBuffer, 0, bufferSize);
@@ -1516,9 +1515,9 @@
mOutBuffer.frameCount = 0;
playbackThread->mTracks.add(this);
ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, "
- "mCblk->frameCount_ %u, mChannelMask 0x%08x",
+ "frameCount %u, mChannelMask 0x%08x",
mCblk, mBuffer,
- mCblk->frameCount_, mChannelMask);
+ frameCount, mChannelMask);
// since client and server are in the same process,
// the buffer has the same virtual address on both sides
mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize);
diff --git a/services/camera/libcameraservice/api1/Camera2Client.cpp b/services/camera/libcameraservice/api1/Camera2Client.cpp
index 0244aec..2bb3ff8 100644
--- a/services/camera/libcameraservice/api1/Camera2Client.cpp
+++ b/services/camera/libcameraservice/api1/Camera2Client.cpp
@@ -813,6 +813,8 @@
return res;
}
outputStreams.push(getZslStreamId());
+ } else {
+ mZslProcessor->deleteStream();
}
outputStreams.push(getPreviewStreamId());