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
diff --git a/libs/audioflinger/AudioFlinger.cpp b/libs/audioflinger/AudioFlinger.cpp
index 06443ef..58eb590 100644
--- a/libs/audioflinger/AudioFlinger.cpp
+++ b/libs/audioflinger/AudioFlinger.cpp
@@ -783,7 +783,7 @@
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 @@
int AudioFlinger::ThreadBase::channelCount() const
{
- return mChannelCount;
+ return (int)mChannelCount;
}
int AudioFlinger::ThreadBase::format() const
@@ -1064,7 +1064,7 @@
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 @@
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::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 @@
// 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 @@
// 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::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 @@
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 @@
(mClient == NULL) ? getpid() : mClient->pid(),
mStreamType,
mFormat,
- mCblk->channels,
+ mCblk->channelCount,
mFrameCount,
mState,
mMute,
@@ -2827,7 +2827,7 @@
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 @@
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 @@
{
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 @@
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 @@
}
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 @@
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 @@
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 @@
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 @@
}
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 @@
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 @@
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];