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

2 files changed, 35 insertions, 33 deletions

diff --git a/libs/audioflinger/AudioFlinger.cpp b/libs/audioflinger/AudioFlinger.cpp
index 06443ef671..58eb590bac 100644
--- a/libs/audioflinger/AudioFlinger.cpp
+++ b/libs/audioflinger/AudioFlinger.cpp
@@ -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];
diff --git a/libs/audioflinger/AudioFlinger.h b/libs/audioflinger/AudioFlinger.h
index 13aac8b96f..c4a53055f5 100644
--- a/libs/audioflinger/AudioFlinger.h
+++ b/libs/audioflinger/AudioFlinger.h
@@ -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;