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

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This commit is contained in:
Eric Laurent
2010-05-14 09:39:12 -07:00
committed by Android (Google) Code Review
parent ead68d5cd6 b0a0147f34
commit 9f0810e8c7
3 changed files with 36 additions and 34 deletions
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2
include/private/media/AudioTrackShared.h
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@ -59,7 +59,7 @@ struct audio_track_cblk_t
// 8 bit PCM data: in this case, mCblk->frameSize is based on a sample size of
// 16 bit because data is converted to 16 bit before being stored in buffer
uint32_t frameSize;
uint8_t channels;
uint8_t channelCount;
uint8_t flowControlFlag; // underrun (out) or overrrun (in) indication
uint8_t out; // out equals 1 for AudioTrack and 0 for AudioRecord
uint8_t forceReady;

63
libs/audioflinger/AudioFlinger.cpp
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@ -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];

5
libs/audioflinger/AudioFlinger.h
View File

@ -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;