Rename mMixBuffer to mSinkBuffer in AudioFlinger
Change-Id: I1f55630425b8ce36e9df2aaf64e06a8960b69a47
Signed-off-by: Andy Hung <hunga@google.com>
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 3e8c133..d278008 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -1066,7 +1066,7 @@
audio_devices_t device,
type_t type)
: ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type),
- mNormalFrameCount(0), mMixBuffer(NULL),
+ mNormalFrameCount(0), mSinkBuffer(NULL),
mSuspended(0), mBytesWritten(0),
mActiveTracksGeneration(0),
// mStreamTypes[] initialized in constructor body
@@ -1125,7 +1125,7 @@
AudioFlinger::PlaybackThread::~PlaybackThread()
{
mAudioFlinger->unregisterWriter(mNBLogWriter);
- delete[] mMixBuffer;
+ delete[] mSinkBuffer;
}
void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
@@ -1210,7 +1210,7 @@
fdprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites);
fdprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no");
fdprintf(fd, " Suspend count: %d\n", mSuspended);
- fdprintf(fd, " Mix buffer : %p\n", mMixBuffer);
+ fdprintf(fd, " Sink buffer : %p\n", mSinkBuffer);
fdprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask);
dumpBase(fd, args);
@@ -1758,11 +1758,11 @@
ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount,
mNormalFrameCount);
- delete[] mMixBuffer;
+ delete[] mSinkBuffer;
size_t normalBufferSize = mNormalFrameCount * mFrameSize;
- // For historical reasons mMixBuffer is int16_t[], but mFrameSize can be odd (such as 1)
- mMixBuffer = new int16_t[(normalBufferSize + 1) >> 1];
- memset(mMixBuffer, 0, normalBufferSize);
+ // For historical reasons mSinkBuffer is int16_t[], but mFrameSize can be odd (such as 1)
+ mSinkBuffer = new int16_t[(normalBufferSize + 1) >> 1];
+ memset(mSinkBuffer, 0, normalBufferSize);
// force reconfiguration of effect chains and engines to take new buffer size and audio
// parameters into account
@@ -1958,7 +1958,7 @@
(pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
}
}
- ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count);
+ ssize_t framesWritten = mNormalSink->write(mSinkBuffer + offset, count);
ATRACE_END();
if (framesWritten > 0) {
bytesWritten = framesWritten << mBitShift;
@@ -1987,7 +1987,7 @@
// FIXME We should have an implementation of timestamps for direct output threads.
// They are used e.g for multichannel PCM playback over HDMI.
bytesWritten = mOutput->stream->write(mOutput->stream,
- (char *)mMixBuffer + offset, mBytesRemaining);
+ (char *)mSinkBuffer + offset, mBytesRemaining);
if (mUseAsyncWrite &&
((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) {
// do not wait for async callback in case of error of full write
@@ -2068,13 +2068,13 @@
status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
{
int session = chain->sessionId();
- int16_t *buffer = mMixBuffer;
+ int16_t *buffer = mSinkBuffer;
bool ownsBuffer = false;
ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
if (session > 0) {
// Only one effect chain can be present in direct output thread and it uses
- // the mix buffer as input
+ // the sink buffer as input
if (mType != DIRECT) {
size_t numSamples = mNormalFrameCount * mChannelCount;
buffer = new int16_t[numSamples];
@@ -2108,7 +2108,7 @@
}
chain->setInBuffer(buffer, ownsBuffer);
- chain->setOutBuffer(mMixBuffer);
+ chain->setOutBuffer(mSinkBuffer);
// Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
// chains list in order to be processed last as it contains output stage effects
// Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
@@ -2158,7 +2158,7 @@
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> track = mTracks[i];
if (session == track->sessionId()) {
- track->setMainBuffer(mMixBuffer);
+ track->setMainBuffer(mSinkBuffer);
chain->decTrackCnt();
}
}
@@ -2861,7 +2861,7 @@
sleepTime = idleSleepTime;
}
} else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) {
- memset(mMixBuffer, 0, mixBufferSize);
+ memset(mSinkBuffer, 0, mixBufferSize);
sleepTime = 0;
ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED),
"anticipated start");
@@ -3109,10 +3109,10 @@
mixedTracks++;
- // track->mainBuffer() != mMixBuffer means there is an effect chain
+ // track->mainBuffer() != mSinkBuffer means there is an effect chain
// connected to the track
chain.clear();
- if (track->mainBuffer() != mMixBuffer) {
+ if (track->mainBuffer() != mSinkBuffer) {
chain = getEffectChain_l(track->sessionId());
// Delegate volume control to effect in track effect chain if needed
if (chain != 0) {
@@ -3355,13 +3355,13 @@
// remove all the tracks that need to be...
removeTracks_l(*tracksToRemove);
- // mix buffer must be cleared if all tracks are connected to an
+ // sink buffer must be cleared if all tracks are connected to an
// effect chain as in this case the mixer will not write to
- // mix buffer and track effects will accumulate into it
+ // sink buffer and track effects will accumulate into it
if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) ||
(mixedTracks == 0 && fastTracks > 0))) {
// FIXME as a performance optimization, should remember previous zero status
- memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t));
+ memset(mSinkBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t));
}
// if any fast tracks, then status is ready
@@ -3749,7 +3749,7 @@
void AudioFlinger::DirectOutputThread::threadLoop_mix()
{
size_t frameCount = mFrameCount;
- int8_t *curBuf = (int8_t *)mMixBuffer;
+ int8_t *curBuf = (int8_t *)mSinkBuffer;
// output audio to hardware
while (frameCount) {
AudioBufferProvider::Buffer buffer;
@@ -3764,7 +3764,7 @@
curBuf += buffer.frameCount * mFrameSize;
mActiveTrack->releaseBuffer(&buffer);
}
- mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer;
+ mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer;
sleepTime = 0;
standbyTime = systemTime() + standbyDelay;
mActiveTrack.clear();
@@ -3779,7 +3779,7 @@
sleepTime = idleSleepTime;
}
} else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) {
- memset(mMixBuffer, 0, mFrameCount * mFrameSize);
+ memset(mSinkBuffer, 0, mFrameCount * mFrameSize);
sleepTime = 0;
}
}
@@ -4306,7 +4306,7 @@
if (outputsReady(outputTracks)) {
mAudioMixer->process(AudioBufferProvider::kInvalidPTS);
} else {
- memset(mMixBuffer, 0, mixBufferSize);
+ memset(mSinkBuffer, 0, mixBufferSize);
}
sleepTime = 0;
writeFrames = mNormalFrameCount;
@@ -4325,7 +4325,7 @@
} else if (mBytesWritten != 0) {
if (mMixerStatus == MIXER_TRACKS_ENABLED) {
writeFrames = mNormalFrameCount;
- memset(mMixBuffer, 0, mixBufferSize);
+ memset(mSinkBuffer, 0, mixBufferSize);
} else {
// flush remaining overflow buffers in output tracks
writeFrames = 0;
@@ -4337,7 +4337,7 @@
ssize_t AudioFlinger::DuplicatingThread::threadLoop_write()
{
for (size_t i = 0; i < outputTracks.size(); i++) {
- outputTracks[i]->write(mMixBuffer, writeFrames);
+ outputTracks[i]->write(mSinkBuffer, writeFrames);
}
mStandby = false;
return (ssize_t)mixBufferSize;
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index fa3563c..5efa746 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -450,7 +450,8 @@
virtual String8 getParameters(const String8& keys);
virtual void audioConfigChanged_l(int event, int param = 0);
status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames);
- int16_t *mixBuffer() const { return mMixBuffer; };
+ // TODO: rename mixBuffer() to sinkBuffer() or try to remove external use.
+ int16_t *mixBuffer() const { return mSinkBuffer; };
virtual void detachAuxEffect_l(int effectId);
status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track,
@@ -481,7 +482,7 @@
// updated by readOutputParameters_l()
size_t mNormalFrameCount; // normal mixer and effects
- int16_t* mMixBuffer; // frame size aligned mix buffer
+ int16_t* mSinkBuffer; // frame size aligned sink buffer
// suspend count, > 0 means suspended. While suspended, the thread continues to pull from
// tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle