From e2a9c29f35e0c09782558542fc4cf9823779590e Mon Sep 17 00:00:00 2001 From: Eric Laurent Date: Thu, 13 Mar 2014 10:44:14 -0700 Subject: Revert "Convert AudioFlinger mSinkBuffer to flexible format" This reverts commit e7e676fd2866fa4898712c4effa9e624e969c182. Bug: 13450717. Change-Id: Ib80b0d14428fecce33c62003a1fcf83f71cee03b --- services/audioflinger/Threads.cpp | 54 ++++++++++++++------------------------- services/audioflinger/Threads.h | 9 +++---- 2 files changed, 22 insertions(+), 41 deletions(-) (limited to 'services/audioflinger') 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(const sp& audioFlinge AudioFlinger::PlaybackThread::~PlaybackThread() { mAudioFlinger->unregisterWriter(mNBLogWriter); - free(mSinkBuffer); + delete[] mSinkBuffer; free(mMixerBuffer); free(mEffectBuffer); } @@ -1782,13 +1782,11 @@ void AudioFlinger::PlaybackThread::readOutputParameters_l() 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 @@ 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) { - 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 @@ ssize_t AudioFlinger::PlaybackThread::threadLoop_write() (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 @@ 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; @@ -2113,8 +2111,8 @@ void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamTy status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp& chain) { int session = chain->sessionId(); - int16_t* buffer = reinterpret_cast(mEffectBufferEnabled - ? mEffectBuffer : mSinkBuffer); + int16_t *buffer = mEffectBufferEnabled + ? reinterpret_cast(mEffectBuffer) : mSinkBuffer; bool ownsBuffer = false; ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); @@ -2154,8 +2152,8 @@ status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp& c } chain->setInBuffer(buffer, ownsBuffer); - chain->setOutBuffer(reinterpret_cast(mEffectBufferEnabled - ? mEffectBuffer : mSinkBuffer)); + chain->setOutBuffer(mEffectBufferEnabled + ? reinterpret_cast(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 @@ size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp& for (size_t i = 0; i < mTracks.size(); ++i) { sp track = mTracks[i]; if (session == track->sessionId()) { - track->setMainBuffer(reinterpret_cast(mSinkBuffer)); + track->setMainBuffer(mSinkBuffer); chain->decTrackCnt(); } } @@ -4473,15 +4471,7 @@ void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 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(mSinkBuffer), writeFrames); + outputTracks[i]->write(mSinkBuffer, writeFrames); } mStandby = false; return (ssize_t)mSinkBufferSize; @@ -4510,16 +4500,10 @@ 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, - AUDIO_FORMAT_PCM_16_BIT, + mFormat, mChannelMask, frameCount, IPCThreadState::self()->getCallingUid()); diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h index 59d5c66..3af4874 100644 --- a/services/audioflinger/Threads.h +++ b/services/audioflinger/Threads.h @@ -450,11 +450,8 @@ public: virtual String8 getParameters(const String8& keys); virtual void audioConfigChanged_l(int event, int param = 0); status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); - // FIXME rename mixBuffer() to sinkBuffer() and remove int16_t* dependency. - // Consider also removing and passing an explicit mMainBuffer initialization - // parameter to AF::PlaybackThread::Track::Track(). - int16_t *mixBuffer() const { - return reinterpret_cast(mSinkBuffer); }; + // TODO: rename mixBuffer() to sinkBuffer() or try to remove external use. + int16_t *mixBuffer() const { return mSinkBuffer; }; virtual void detachAuxEffect_l(int effectId); status_t attachAuxEffect(const sp track, @@ -485,7 +482,7 @@ protected: // updated by readOutputParameters_l() size_t mNormalFrameCount; // normal mixer and effects - void* mSinkBuffer; // frame size aligned sink buffer + int16_t* mSinkBuffer; // frame size aligned sink buffer // TODO: // Rearrange the buffer info into a struct/class with -- cgit v1.1