summaryrefslogtreecommitdiffstats
path: root/services/audioflinger/Threads.cpp
diff options
context:
space:
mode:
Diffstat (limited to 'services/audioflinger/Threads.cpp')
-rw-r--r--services/audioflinger/Threads.cpp86
1 files changed, 55 insertions, 31 deletions
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 82c516c..12d453e 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -1145,7 +1145,7 @@ AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinge
AudioFlinger::PlaybackThread::~PlaybackThread()
{
mAudioFlinger->unregisterWriter(mNBLogWriter);
- delete[] mSinkBuffer;
+ free(mSinkBuffer);
free(mMixerBuffer);
free(mEffectBuffer);
}
@@ -1340,7 +1340,9 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac
}
*pFrameCount = frameCount;
- if (mType == DIRECT) {
+ switch (mType) {
+
+ case DIRECT:
if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x "
@@ -1350,7 +1352,9 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac
goto Exit;
}
}
- } else if (mType == OFFLOAD) {
+ break;
+
+ case OFFLOAD:
if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \""
"for output %p with format %#x",
@@ -1358,7 +1362,9 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac
lStatus = BAD_VALUE;
goto Exit;
}
- } else {
+ break;
+
+ default:
if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) {
ALOGE("createTrack_l() Bad parameter: format %#x \""
"for output %p with format %#x",
@@ -1372,11 +1378,13 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac
lStatus = BAD_VALUE;
goto Exit;
}
+ break;
+
}
lStatus = initCheck();
if (lStatus != NO_ERROR) {
- ALOGE("Audio driver not initialized.");
+ ALOGE("createTrack_l() audio driver not initialized");
goto Exit;
}
@@ -1416,7 +1424,6 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac
// track must be cleared from the caller as the caller has the AF lock
goto Exit;
}
-
mTracks.add(track);
sp<EffectChain> chain = getEffectChain_l(sessionId);
@@ -1782,11 +1789,14 @@ void AudioFlinger::PlaybackThread::readOutputParameters_l()
ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount,
mNormalFrameCount);
- delete[] mSinkBuffer;
- size_t normalBufferSize = mNormalFrameCount * mFrameSize;
- // For historical reasons mSinkBuffer is int16_t[], but mFrameSize can be odd (such as 1)
- mSinkBuffer = new int16_t[(normalBufferSize + 1) >> 1];
- memset(mSinkBuffer, 0, normalBufferSize);
+ // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames.
+ // Originally this was int16_t[] array, need to remove legacy implications.
+ free(mSinkBuffer);
+ mSinkBuffer = NULL;
+ // For sink buffer size, we use the frame size from the downstream sink to avoid problems
+ // with non PCM formats for compressed music, e.g. AAC, and Offload threads.
+ const size_t sinkBufferSize = mNormalFrameCount * mFrameSize;
+ (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize);
// We resize the mMixerBuffer according to the requirements of the sink buffer which
// drives the output.
@@ -1984,12 +1994,12 @@ ssize_t AudioFlinger::PlaybackThread::threadLoop_write()
mLastWriteTime = systemTime();
mInWrite = true;
ssize_t bytesWritten;
+ const size_t offset = mCurrentWriteLength - mBytesRemaining;
// If an NBAIO sink is present, use it to write the normal mixer's submix
if (mNormalSink != 0) {
-#define mBitShift 2 // FIXME
- size_t count = mBytesRemaining >> mBitShift;
- size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1;
+ const size_t count = mBytesRemaining / mFrameSize;
+
ATRACE_BEGIN("write");
// update the setpoint when AudioFlinger::mScreenState changes
uint32_t screenState = AudioFlinger::mScreenState;
@@ -2001,10 +2011,10 @@ ssize_t AudioFlinger::PlaybackThread::threadLoop_write()
(pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
}
}
- ssize_t framesWritten = mNormalSink->write(mSinkBuffer + offset, count);
+ ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count);
ATRACE_END();
if (framesWritten > 0) {
- bytesWritten = framesWritten << mBitShift;
+ bytesWritten = framesWritten * mFrameSize;
} else {
bytesWritten = framesWritten;
}
@@ -2019,7 +2029,7 @@ ssize_t AudioFlinger::PlaybackThread::threadLoop_write()
// otherwise use the HAL / AudioStreamOut directly
} else {
// Direct output and offload threads
- size_t offset = (mCurrentWriteLength - mBytesRemaining);
+
if (mUseAsyncWrite) {
ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request");
mWriteAckSequence += 2;
@@ -2111,8 +2121,8 @@ void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamTy
status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
{
int session = chain->sessionId();
- int16_t *buffer = mEffectBufferEnabled
- ? reinterpret_cast<int16_t*>(mEffectBuffer) : mSinkBuffer;
+ int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled
+ ? mEffectBuffer : mSinkBuffer);
bool ownsBuffer = false;
ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
@@ -2152,8 +2162,8 @@ status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& c
}
chain->setInBuffer(buffer, ownsBuffer);
- chain->setOutBuffer(mEffectBufferEnabled
- ? reinterpret_cast<int16_t*>(mEffectBuffer) : mSinkBuffer);
+ chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled
+ ? mEffectBuffer : mSinkBuffer));
// Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
// chains list in order to be processed last as it contains output stage effects
// Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
@@ -2203,7 +2213,7 @@ size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>&
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> track = mTracks[i];
if (session == track->sessionId()) {
- track->setMainBuffer(mSinkBuffer);
+ track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer));
chain->decTrackCnt();
}
}
@@ -4471,7 +4481,15 @@ void AudioFlinger::DuplicatingThread::threadLoop_sleepTime()
ssize_t AudioFlinger::DuplicatingThread::threadLoop_write()
{
for (size_t i = 0; i < outputTracks.size(); i++) {
- outputTracks[i]->write(mSinkBuffer, writeFrames);
+ // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT
+ // for delivery downstream as needed. This in-place conversion is safe as
+ // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format
+ // (AUDIO_FORMAT_PCM_8_BIT is not allowed here).
+ if (mFormat != AUDIO_FORMAT_PCM_16_BIT) {
+ memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT,
+ mSinkBuffer, mFormat, writeFrames * mChannelCount);
+ }
+ outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames);
}
mStandby = false;
return (ssize_t)mSinkBufferSize;
@@ -4500,10 +4518,16 @@ void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
Mutex::Autolock _l(mLock);
// FIXME explain this formula
size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate();
+ // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat
+ // due to current usage case and restrictions on the AudioBufferProvider.
+ // Actual buffer conversion is done in threadLoop_write().
+ //
+ // TODO: This may change in the future, depending on multichannel
+ // (and non int16_t*) support on AF::PlaybackThread::OutputTrack
OutputTrack *outputTrack = new OutputTrack(thread,
this,
mSampleRate,
- mFormat,
+ AUDIO_FORMAT_PCM_16_BIT,
mChannelMask,
frameCount,
IPCThreadState::self()->getCallingUid());
@@ -5036,6 +5060,7 @@ void AudioFlinger::RecordThread::inputStandBy()
mInput->stream->common.standby(&mInput->stream->common);
}
+// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held
sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l(
const sp<AudioFlinger::Client>& client,
uint32_t sampleRate,
@@ -5052,12 +5077,6 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRe
sp<RecordTrack> track;
status_t lStatus;
- lStatus = initCheck();
- if (lStatus != NO_ERROR) {
- ALOGE("createRecordTrack_l() audio driver not initialized");
- goto Exit;
- }
-
// client expresses a preference for FAST, but we get the final say
if (*flags & IAudioFlinger::TRACK_FAST) {
if (
@@ -5110,7 +5129,11 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRe
}
*pFrameCount = frameCount;
- // FIXME use flags and tid similar to createTrack_l()
+ lStatus = initCheck();
+ if (lStatus != NO_ERROR) {
+ ALOGE("createRecordTrack_l() audio driver not initialized");
+ goto Exit;
+ }
{ // scope for mLock
Mutex::Autolock _l(mLock);
@@ -5139,6 +5162,7 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRe
sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp);
}
}
+
lStatus = NO_ERROR;
Exit:
generated by cgit v1.1 at 2024-06-01 15:56:47 +0000