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Commit 9f6d082b authored by Eric Laurent's avatar Eric Laurent Committed by Android (Google) Code Review
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Merge "AudioFlinger: rename variables to clarify reference to track channel... 

Merge "AudioFlinger: rename variables to clarify reference to track channel count or channel mask" into kraken
parents a6022122 e151216d

libs/audioflinger/AudioFlinger.cpp

+32 −31
Original line number Diff line number Diff line
@@ -783,7 +783,7 @@ void AudioFlinger::removeClient_l(pid_t pid) 
AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id)

    :   Thread(false),

        mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0),

        mFormat(0), mFrameSize(1), mStandby(false), mId(id), mExiting(false)

        mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false)

{

}
@@ -816,7 +816,7 @@ uint32_t AudioFlinger::ThreadBase::sampleRate() const 


int AudioFlinger::ThreadBase::channelCount() const

{

    return mChannelCount;

    return (int)mChannelCount;

}



int AudioFlinger::ThreadBase::format() const
@@ -1064,7 +1064,7 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTra 
    status_t lStatus;



    if (mType == DIRECT) {

        if (sampleRate != mSampleRate || format != mFormat || channelCount != mChannelCount) {

        if (sampleRate != mSampleRate || format != mFormat || channelCount != (int)mChannelCount) {

            LOGE("createTrack_l() Bad parameter:  sampleRate %d format %d, channelCount %d for output %p",

                 sampleRate, format, channelCount, mOutput);

            lStatus = BAD_VALUE;
@@ -1243,7 +1243,7 @@ void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 
    switch (event) {

    case AudioSystem::OUTPUT_OPENED:

    case AudioSystem::OUTPUT_CONFIG_CHANGED:

        desc.channels = mChannelCount;

        desc.channels = mChannels;

        desc.samplingRate = mSampleRate;

        desc.format = mFormat;

        desc.frameCount = mFrameCount;
@@ -1264,10 +1264,10 @@ void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 
void AudioFlinger::PlaybackThread::readOutputParameters()

{

    mSampleRate = mOutput->sampleRate();

    mChannelCount = AudioSystem::popCount(mOutput->channels());



    mChannels = mOutput->channels();

    mChannelCount = (uint16_t)AudioSystem::popCount(mChannels);

    mFormat = mOutput->format();

    mFrameSize = mOutput->frameSize();

    mFrameSize = (uint16_t)mOutput->frameSize();

    mFrameCount = mOutput->bufferSize() / mFrameSize;



    // FIXME - Current mixer implementation only supports stereo output: Always
@@ -2342,7 +2342,7 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase( 
                // clear all buffers

                mCblk->frameCount = frameCount;

                mCblk->sampleRate = sampleRate;

                mCblk->channels = (uint8_t)channelCount;

                mCblk->channelCount = (uint8_t)channelCount;

                if (sharedBuffer == 0) {

                    mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);

                    memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
@@ -2366,7 +2366,7 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase( 
           // clear all buffers

           mCblk->frameCount = frameCount;

           mCblk->sampleRate = sampleRate;

           mCblk->channels = (uint8_t)channelCount;

           mCblk->channelCount = (uint8_t)channelCount;

           mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);

           memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));

           // Force underrun condition to avoid false underrun callback until first data is
@@ -2433,7 +2433,7 @@ int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 
}



int AudioFlinger::ThreadBase::TrackBase::channelCount() const {

    return (int)mCblk->channels;

    return (int)mCblk->channelCount;

}



void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
@@ -2445,9 +2445,9 @@ void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t f 
    if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||

        ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) {

        LOGE("TrackBase::getBuffer buffer out of range:\n    start: %p, end %p , mBuffer %p mBufferEnd %p\n    \

                server %d, serverBase %d, user %d, userBase %d, channels %d",

                server %d, serverBase %d, user %d, userBase %d, channelCount %d",

                bufferStart, bufferEnd, mBuffer, mBufferEnd,

                cblk->server, cblk->serverBase, cblk->user, cblk->userBase, cblk->channels);

                cblk->server, cblk->serverBase, cblk->user, cblk->userBase, cblk->channelCount);

        return 0;

    }
@@ -2532,7 +2532,7 @@ void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 
            (mClient == NULL) ? getpid() : mClient->pid(),

            mStreamType,

            mFormat,

            mCblk->channels,

            mCblk->channelCount,

            mFrameCount,

            mState,

            mMute,
@@ -2827,7 +2827,7 @@ void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 
    snprintf(buffer, size, "   %05d %03u %03u %04u %01d %05u  %08x %08x\n",

            (mClient == NULL) ? getpid() : mClient->pid(),

            mFormat,

            mCblk->channels,

            mCblk->channelCount,

            mFrameCount,

            mState,

            mCblk->sampleRate,
@@ -2856,8 +2856,8 @@ AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 
        mCblk->volume[0] = mCblk->volume[1] = 0x1000;

        mOutBuffer.frameCount = 0;

        playbackThread->mTracks.add(this);

        LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channels %d mBufferEnd %p",

                mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channels, mBufferEnd);

        LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channelCount %d mBufferEnd %p",

                mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channelCount, mBufferEnd);

    } else {

        LOGW("Error creating output track on thread %p", playbackThread);

    }
@@ -2892,7 +2892,7 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr 
{

    Buffer *pInBuffer;

    Buffer inBuffer;

    uint32_t channels = mCblk->channels;

    uint32_t channelCount = mCblk->channelCount;

    bool outputBufferFull = false;

    inBuffer.frameCount = frames;

    inBuffer.i16 = data;
@@ -2908,10 +2908,10 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr 
                if (mBufferQueue.size() < kMaxOverFlowBuffers) {

                    uint32_t startFrames = (mCblk->frameCount - frames);

                    pInBuffer = new Buffer;

                    pInBuffer->mBuffer = new int16_t[startFrames * channels];

                    pInBuffer->mBuffer = new int16_t[startFrames * channelCount];

                    pInBuffer->frameCount = startFrames;

                    pInBuffer->i16 = pInBuffer->mBuffer;

                    memset(pInBuffer->raw, 0, startFrames * channels * sizeof(int16_t));

                    memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));

                    mBufferQueue.add(pInBuffer);

                } else {

                    LOGW ("OutputTrack::write() %p no more buffers in queue", this);
@@ -2949,12 +2949,12 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr 
        }



        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount;

        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channels * sizeof(int16_t));

        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));

        mCblk->stepUser(outFrames);

        pInBuffer->frameCount -= outFrames;

        pInBuffer->i16 += outFrames * channels;

        pInBuffer->i16 += outFrames * channelCount;

        mOutBuffer.frameCount -= outFrames;

        mOutBuffer.i16 += outFrames * channels;

        mOutBuffer.i16 += outFrames * channelCount;



        if (pInBuffer->frameCount == 0) {

            if (mBufferQueue.size()) {
@@ -2974,10 +2974,10 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr 
        if (thread != 0 && !thread->standby()) {

            if (mBufferQueue.size() < kMaxOverFlowBuffers) {

                pInBuffer = new Buffer;

                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channels];

                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];

                pInBuffer->frameCount = inBuffer.frameCount;

                pInBuffer->i16 = pInBuffer->mBuffer;

                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channels * sizeof(int16_t));

                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t));

                mBufferQueue.add(pInBuffer);

                LOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());

            } else {
@@ -2993,10 +2993,10 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr 
        if (mCblk->user < mCblk->frameCount) {

            frames = mCblk->frameCount - mCblk->user;

            pInBuffer = new Buffer;

            pInBuffer->mBuffer = new int16_t[frames * channels];

            pInBuffer->mBuffer = new int16_t[frames * channelCount];

            pInBuffer->frameCount = frames;

            pInBuffer->i16 = pInBuffer->mBuffer;

            memset(pInBuffer->raw, 0, frames * channels * sizeof(int16_t));

            memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));

            mBufferQueue.add(pInBuffer);

        } else if (mActive) {

            stop();
@@ -3371,7 +3371,7 @@ bool AudioFlinger::RecordThread::threadLoop() 
                                framesIn = framesOut;

                            mRsmpInIndex += framesIn;

                            framesOut -= framesIn;

                            if (mChannelCount == mReqChannelCount ||

                            if ((int)mChannelCount == mReqChannelCount ||

                                mFormat != AudioSystem::PCM_16_BIT) {

                                memcpy(dst, src, framesIn * mFrameSize);

                            } else {
@@ -3392,7 +3392,7 @@ bool AudioFlinger::RecordThread::threadLoop() 
                        }

                        if (framesOut && mFrameCount == mRsmpInIndex) {

                            if (framesOut == mFrameCount &&

                                (mChannelCount == mReqChannelCount || mFormat != AudioSystem::PCM_16_BIT)) {

                                ((int)mChannelCount == mReqChannelCount || mFormat != AudioSystem::PCM_16_BIT)) {

                                mBytesRead = mInput->read(buffer.raw, mInputBytes);

                                framesOut = 0;

                            } else {
@@ -3696,7 +3696,7 @@ void AudioFlinger::RecordThread::audioConfigChanged(int event, int param) { 
    switch (event) {

    case AudioSystem::INPUT_OPENED:

    case AudioSystem::INPUT_CONFIG_CHANGED:

        desc.channels = mChannelCount;

        desc.channels = mChannels;

        desc.samplingRate = mSampleRate;

        desc.format = mFormat;

        desc.frameCount = mFrameCount;
@@ -3720,9 +3720,10 @@ void AudioFlinger::RecordThread::readInputParameters() 
    mResampler = 0;



    mSampleRate = mInput->sampleRate();

    mChannelCount = AudioSystem::popCount(mInput->channels());

    mChannels = mInput->channels();

    mChannelCount = (uint16_t)AudioSystem::popCount(mChannels);

    mFormat = mInput->format();

    mFrameSize = mInput->frameSize();

    mFrameSize = (uint16_t)mInput->frameSize();

    mInputBytes = mInput->bufferSize();

    mFrameCount = mInputBytes / mFrameSize;

    mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];

libs/audioflinger/AudioFlinger.h

+3 −2
Original line number Diff line number Diff line
@@ -366,9 +366,10 @@ private: 
                    sp<AudioFlinger>        mAudioFlinger;

                    uint32_t                mSampleRate;

                    size_t                  mFrameCount;

                    int                     mChannelCount;

                    uint32_t                mChannels;

                    uint16_t                mChannelCount;

                    uint16_t                mFrameSize;

                    int                     mFormat;

                    uint32_t                mFrameSize;

                    Condition               mParamCond;

                    Vector<String8>         mNewParameters;

                    status_t                mParamStatus;