Revert "Convert AudioFlinger mSinkBuffer to flexible format"
This reverts commit e7e676fd2866fa4898712c4effa9e624e969c182.
Bug: 13450717.
Change-Id: Ib80b0d14428fecce33c62003a1fcf83f71cee03b
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 8aee194..82c516c 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -1145,7 +1145,7 @@
AudioFlinger::PlaybackThread::~PlaybackThread()
{
mAudioFlinger->unregisterWriter(mNBLogWriter);
- free(mSinkBuffer);
+ delete[] mSinkBuffer;
free(mMixerBuffer);
free(mEffectBuffer);
}
@@ -1782,13 +1782,11 @@
ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount,
mNormalFrameCount);
- // 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;
- const size_t sinkBufferSize = mNormalFrameCount * mChannelCount
- * audio_bytes_per_sample(mFormat);
- (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize);
+ 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);
// We resize the mMixerBuffer according to the requirements of the sink buffer which
// drives the output.
@@ -1986,12 +1984,12 @@
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) {
- const size_t count = mBytesRemaining / mFrameSize;
-
+#define mBitShift 2 // FIXME
+ size_t count = mBytesRemaining >> mBitShift;
+ size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1;
ATRACE_BEGIN("write");
// update the setpoint when AudioFlinger::mScreenState changes
uint32_t screenState = AudioFlinger::mScreenState;
@@ -2003,10 +2001,10 @@
(pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
}
}
- ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count);
+ ssize_t framesWritten = mNormalSink->write(mSinkBuffer + offset, count);
ATRACE_END();
if (framesWritten > 0) {
- bytesWritten = framesWritten * mFrameSize;
+ bytesWritten = framesWritten << mBitShift;
} else {
bytesWritten = framesWritten;
}
@@ -2021,7 +2019,7 @@
// 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;
@@ -2113,8 +2111,8 @@
status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
{
int session = chain->sessionId();
- int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled
- ? mEffectBuffer : mSinkBuffer);
+ int16_t *buffer = mEffectBufferEnabled
+ ? reinterpret_cast<int16_t*>(mEffectBuffer) : mSinkBuffer;
bool ownsBuffer = false;
ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
@@ -2154,8 +2152,8 @@
}
chain->setInBuffer(buffer, ownsBuffer);
- chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled
- ? mEffectBuffer : mSinkBuffer));
+ chain->setOutBuffer(mEffectBufferEnabled
+ ? reinterpret_cast<int16_t*>(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
@@ -2205,7 +2203,7 @@
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> track = mTracks[i];
if (session == track->sessionId()) {
- track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer));
+ track->setMainBuffer(mSinkBuffer);
chain->decTrackCnt();
}
}
@@ -4473,15 +4471,7 @@
ssize_t AudioFlinger::DuplicatingThread::threadLoop_write()
{
for (size_t i = 0; i < outputTracks.size(); i++) {
- // 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);
+ outputTracks[i]->write(mSinkBuffer, writeFrames);
}
mStandby = false;
return (ssize_t)mSinkBufferSize;
@@ -4510,16 +4500,10 @@
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,
- AUDIO_FORMAT_PCM_16_BIT,
+ mFormat,
mChannelMask,
frameCount,
IPCThreadState::self()->getCallingUid());