Move frame size out of the control block
This is part of a series to clean up the control block.
Change-Id: Ifab1c42ac0f8be704e571b292713cd2250d12a3f
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index c0f5c7b..eb20019 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -4190,6 +4190,7 @@
mState(IDLE),
mSampleRate(sampleRate),
mFormat(format),
+ mFrameSize(0), // will be set to correct value in constructor
mStepServerFailed(false),
mSessionId(sessionId)
// mChannelCount
@@ -4311,7 +4312,7 @@
void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
audio_track_cblk_t* cblk = this->cblk();
- size_t frameSize = cblk->frameSize;
+ size_t frameSize = mFrameSize;
int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize;
int8_t *bufferEnd = bufferStart + frames * frameSize;
@@ -4363,11 +4364,11 @@
mUnderrunCount(0),
mCachedVolume(1.0)
{
+ // NOTE: frame size for 8 bit PCM data is based on a sample size of
+ // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
+ mFrameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) :
+ sizeof(uint8_t);
if (mCblk != NULL) {
- // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
- // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
- mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) :
- sizeof(uint8_t);
// to avoid leaking a track name, do not allocate one unless there is an mCblk
mName = thread->getTrackName_l(channelMask, sessionId);
mCblk->mName = mName;
@@ -5014,7 +5015,7 @@
// this frame in media time units and adding it to the PTS of the
// buffer.
int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size()
- / mCblk->frameSize;
+ / mFrameSize;
if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount,
&bufEnd)) {
@@ -5074,7 +5075,7 @@
" bytes. (update reason: \"%s\")",
bufBytes, consumedAlready, logTag);
- uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize;
+ uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize;
ALOG_ASSERT(mFramesPendingInQueue >= bufFrames,
"Bad bookkeeping while updating frames pending. Should have at"
" least %u queued frames, but we think we have only %u. (update"
@@ -5095,7 +5096,7 @@
Mutex::Autolock _l(mTimedBufferQueueLock);
- uint32_t bufFrames = buffer->size() / mCblk->frameSize;
+ uint32_t bufFrames = buffer->size() / mFrameSize;
mFramesPendingInQueue += bufFrames;
mTimedBufferQueue.add(TimedBuffer(buffer, pts));
@@ -5192,7 +5193,7 @@
// adjust the head buffer's PTS to reflect the portion of the head buffer
// that has already been consumed
int64_t effectivePTS = headLocalPTS +
- ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate());
+ ((head.position() / mFrameSize) * mLocalTimeFreq / sampleRate());
// Calculate the delta in samples between the head of the input buffer
// queue and the start of the next output buffer that will be written.
@@ -5257,7 +5258,7 @@
// the next input sample is late
uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32));
size_t onTimeSamplePosition =
- head.position() + lateFrames * mCblk->frameSize;
+ head.position() + lateFrames * mFrameSize;
if (onTimeSamplePosition > head.buffer()->size()) {
// all the remaining samples in the head are too late, so
@@ -5292,7 +5293,7 @@
head.position());
uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) /
- mCblk->frameSize);
+ mFrameSize);
size_t framesRequested = buffer->frameCount;
buffer->frameCount = min(framesLeftInHead, framesRequested);
@@ -5307,9 +5308,9 @@
uint32_t numFrames, AudioBufferProvider::Buffer* buffer) {
// lazily allocate a buffer filled with silence
- if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) {
+ if (mTimedSilenceBufferSize < numFrames * mFrameSize) {
delete [] mTimedSilenceBuffer;
- mTimedSilenceBufferSize = numFrames * mCblk->frameSize;
+ mTimedSilenceBufferSize = numFrames * mFrameSize;
mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize];
memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize);
}
@@ -5357,7 +5358,7 @@
start, end, buffer->raw);
head.setPosition(head.position() +
- (buffer->frameCount * mCblk->frameSize));
+ (buffer->frameCount * mFrameSize));
mQueueHeadInFlight = false;
ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount,
@@ -5409,15 +5410,13 @@
channelMask, frameCount, 0 /*sharedBuffer*/, sessionId),
mOverflow(false)
{
- if (mCblk != NULL) {
- ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
- if (format == AUDIO_FORMAT_PCM_16_BIT) {
- mCblk->frameSize = mChannelCount * sizeof(int16_t);
- } else if (format == AUDIO_FORMAT_PCM_8_BIT) {
- mCblk->frameSize = mChannelCount * sizeof(int8_t);
- } else {
- mCblk->frameSize = sizeof(int8_t);
- }
+ ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
+ if (format == AUDIO_FORMAT_PCM_16_BIT) {
+ mFrameSize = mChannelCount * sizeof(int16_t);
+ } else if (format == AUDIO_FORMAT_PCM_8_BIT) {
+ mFrameSize = mChannelCount * sizeof(int8_t);
+ } else {
+ mFrameSize = sizeof(int8_t);
}
}
@@ -5749,7 +5748,7 @@
}
buffer->frameCount = framesReq;
- buffer->raw = cblk->buffer(mBuffers, u);
+ buffer->raw = cblk->buffer(mBuffers, mFrameSize, u);
return NO_ERROR;
}
@@ -6157,7 +6156,7 @@
if (framesIn) {
int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) *
- mActiveTrack->mCblk->frameSize;
+ mActiveTrack->mFrameSize;
if (framesIn > framesOut)
framesIn = framesOut;
mRsmpInIndex += framesIn;