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Commit 2098f274 authored by Andy Hung's avatar Andy Hung
Browse files

Rename mMixBuffer to mSinkBuffer in AudioFlinger 



Change-Id: I1f55630425b8ce36e9df2aaf64e06a8960b69a47
Signed-off-by: default avatarAndy Hung <hunga@google.com>
parent 4d3c5bde

services/audioflinger/Threads.cpp

+25 −25
Original line number Diff line number Diff line
@@ -1066,7 +1066,7 @@ AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinge 
                                             audio_devices_t device,

                                             type_t type)

    :   ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type),

        mNormalFrameCount(0), mMixBuffer(NULL),

        mNormalFrameCount(0), mSinkBuffer(NULL),

        mSuspended(0), mBytesWritten(0),

        mActiveTracksGeneration(0),

        // mStreamTypes[] initialized in constructor body
@@ -1125,7 +1125,7 @@ AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinge 
AudioFlinger::PlaybackThread::~PlaybackThread()

{

    mAudioFlinger->unregisterWriter(mNBLogWriter);

    delete[] mMixBuffer;

    delete[] mSinkBuffer;

}



void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
@@ -1210,7 +1210,7 @@ void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& 
    fdprintf(fd, "  Delayed writes: %d\n", mNumDelayedWrites);

    fdprintf(fd, "  Blocked in write: %s\n", mInWrite ? "yes" : "no");

    fdprintf(fd, "  Suspend count: %d\n", mSuspended);

    fdprintf(fd, "  Mix buffer : %p\n", mMixBuffer);

    fdprintf(fd, "  Sink buffer : %p\n", mSinkBuffer);

    fdprintf(fd, "  Fast track availMask=%#x\n", mFastTrackAvailMask);



    dumpBase(fd, args);
@@ -1758,11 +1758,11 @@ void AudioFlinger::PlaybackThread::readOutputParameters_l() 
    ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount,

            mNormalFrameCount);



    delete[] mMixBuffer;

    delete[] mSinkBuffer;

    size_t normalBufferSize = mNormalFrameCount * mFrameSize;

    // For historical reasons mMixBuffer is int16_t[], but mFrameSize can be odd (such as 1)

    mMixBuffer = new int16_t[(normalBufferSize + 1) >> 1];

    memset(mMixBuffer, 0, normalBufferSize);

    // 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);



    // force reconfiguration of effect chains and engines to take new buffer size and audio

    // parameters into account
@@ -1958,7 +1958,7 @@ ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 
                        (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);

            }

        }

        ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count);

        ssize_t framesWritten = mNormalSink->write(mSinkBuffer + offset, count);

        ATRACE_END();

        if (framesWritten > 0) {

            bytesWritten = framesWritten << mBitShift;
@@ -1987,7 +1987,7 @@ ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 
        // FIXME We should have an implementation of timestamps for direct output threads.

        // They are used e.g for multichannel PCM playback over HDMI.

        bytesWritten = mOutput->stream->write(mOutput->stream,

                                                   (char *)mMixBuffer + offset, mBytesRemaining);

                                                   (char *)mSinkBuffer + offset, mBytesRemaining);

        if (mUseAsyncWrite &&

                ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) {

            // do not wait for async callback in case of error of full write
@@ -2068,13 +2068,13 @@ 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 = mMixBuffer;

    int16_t *buffer = mSinkBuffer;

    bool ownsBuffer = false;



    ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);

    if (session > 0) {

        // Only one effect chain can be present in direct output thread and it uses

        // the mix buffer as input

        // the sink buffer as input

        if (mType != DIRECT) {

            size_t numSamples = mNormalFrameCount * mChannelCount;

            buffer = new int16_t[numSamples];
@@ -2108,7 +2108,7 @@ status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& c 
    }



    chain->setInBuffer(buffer, ownsBuffer);

    chain->setOutBuffer(mMixBuffer);

    chain->setOutBuffer(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
@@ -2158,7 +2158,7 @@ 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(mMixBuffer);

                    track->setMainBuffer(mSinkBuffer);

                    chain->decTrackCnt();

                }

            }
@@ -2861,7 +2861,7 @@ void AudioFlinger::MixerThread::threadLoop_sleepTime() 
            sleepTime = idleSleepTime;

        }

    } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) {

        memset(mMixBuffer, 0, mixBufferSize);

        memset(mSinkBuffer, 0, mixBufferSize);

        sleepTime = 0;

        ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED),

                "anticipated start");
@@ -3109,10 +3109,10 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac 


            mixedTracks++;



            // track->mainBuffer() != mMixBuffer means there is an effect chain

            // track->mainBuffer() != mSinkBuffer means there is an effect chain

            // connected to the track

            chain.clear();

            if (track->mainBuffer() != mMixBuffer) {

            if (track->mainBuffer() != mSinkBuffer) {

                chain = getEffectChain_l(track->sessionId());

                // Delegate volume control to effect in track effect chain if needed

                if (chain != 0) {
@@ -3355,13 +3355,13 @@ track_is_ready: ; 
    // remove all the tracks that need to be...

    removeTracks_l(*tracksToRemove);



    // mix buffer must be cleared if all tracks are connected to an

    // sink buffer must be cleared if all tracks are connected to an

    // effect chain as in this case the mixer will not write to

    // mix buffer and track effects will accumulate into it

    // sink buffer and track effects will accumulate into it

    if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) ||

            (mixedTracks == 0 && fastTracks > 0))) {

        // FIXME as a performance optimization, should remember previous zero status

        memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t));

        memset(mSinkBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t));

    }



    // if any fast tracks, then status is ready
@@ -3749,7 +3749,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prep 
void AudioFlinger::DirectOutputThread::threadLoop_mix()

{

    size_t frameCount = mFrameCount;

    int8_t *curBuf = (int8_t *)mMixBuffer;

    int8_t *curBuf = (int8_t *)mSinkBuffer;

    // output audio to hardware

    while (frameCount) {

        AudioBufferProvider::Buffer buffer;
@@ -3764,7 +3764,7 @@ void AudioFlinger::DirectOutputThread::threadLoop_mix() 
        curBuf += buffer.frameCount * mFrameSize;

        mActiveTrack->releaseBuffer(&buffer);

    }

    mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer;

    mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer;

    sleepTime = 0;

    standbyTime = systemTime() + standbyDelay;

    mActiveTrack.clear();
@@ -3779,7 +3779,7 @@ void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 
            sleepTime = idleSleepTime;

        }

    } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) {

        memset(mMixBuffer, 0, mFrameCount * mFrameSize);

        memset(mSinkBuffer, 0, mFrameCount * mFrameSize);

        sleepTime = 0;

    }

}
@@ -4306,7 +4306,7 @@ void AudioFlinger::DuplicatingThread::threadLoop_mix() 
    if (outputsReady(outputTracks)) {

        mAudioMixer->process(AudioBufferProvider::kInvalidPTS);

    } else {

        memset(mMixBuffer, 0, mixBufferSize);

        memset(mSinkBuffer, 0, mixBufferSize);

    }

    sleepTime = 0;

    writeFrames = mNormalFrameCount;
@@ -4325,7 +4325,7 @@ void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 
    } else if (mBytesWritten != 0) {

        if (mMixerStatus == MIXER_TRACKS_ENABLED) {

            writeFrames = mNormalFrameCount;

            memset(mMixBuffer, 0, mixBufferSize);

            memset(mSinkBuffer, 0, mixBufferSize);

        } else {

            // flush remaining overflow buffers in output tracks

            writeFrames = 0;
@@ -4337,7 +4337,7 @@ void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 
ssize_t AudioFlinger::DuplicatingThread::threadLoop_write()

{

    for (size_t i = 0; i < outputTracks.size(); i++) {

        outputTracks[i]->write(mMixBuffer, writeFrames);

        outputTracks[i]->write(mSinkBuffer, writeFrames);

    }

    mStandby = false;

    return (ssize_t)mixBufferSize;

services/audioflinger/Threads.h

+3 −2
Original line number Diff line number Diff line
@@ -450,7 +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);

                int16_t     *mixBuffer() const { return mMixBuffer; };

                // 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<AudioFlinger::PlaybackThread::Track> track,
@@ -481,7 +482,7 @@ protected: 
    // updated by readOutputParameters_l()

    size_t                          mNormalFrameCount;  // normal mixer and effects



    int16_t*                        mMixBuffer;         // frame size aligned mix buffer

    int16_t*                        mSinkBuffer;         // frame size aligned sink buffer



    // suspend count, > 0 means suspended.  While suspended, the thread continues to pull from

    // tracks and mix, but doesn't write to HAL.  A2DP and SCO HAL implementations can't handle