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Commit e151216d authored by Eric Laurent's avatar Eric Laurent
Browse files

AudioFlinger: rename variables to clarify reference to track channel count or channel mask 

Some variables and structure members should be renamed to reflect the fact that they contain the
number of channels in a track (channel count) or the actual channels used by a track (channel mask).
Especially member "channels" of track control block (struct audio_track_cblk_t) is actually the
number of channels (channels count).

Change-Id: I220c8dede9fc00c8a5693389e790073b6ed307b8
parent 73e1599b

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;