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/include/media/AudioRecord.h b/include/media/AudioRecord.h
index bfb5d3a..2672db1 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -185,7 +185,7 @@
audio_format_t format() const;
int channelCount() const;
uint32_t frameCount() const;
- size_t frameSize() const;
+ size_t frameSize() const { return mFrameSize; }
audio_source_t inputSource() const;
@@ -378,6 +378,7 @@
uint32_t mFrameCount;
audio_format_t mFormat;
uint8_t mChannelCount;
+ size_t mFrameSize; // app-level frame size == AudioFlinger frame size
audio_source_t mInputSource;
status_t mStatus;
uint32_t mLatency;
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 639b567..3504f1f 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -229,7 +229,7 @@
/* Return channelCount * (bit depth per channel / 8).
* channelCount is determined from channelMask, and bit depth comes from format.
*/
- size_t frameSize() const;
+ size_t frameSize() const { return mFrameSize; }
sp<IMemory>& sharedBuffer();
@@ -517,6 +517,13 @@
uint8_t mMuted;
uint8_t mReserved;
audio_channel_mask_t mChannelMask;
+
+ // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data.
+ // For 8-bit PCM data, mFrameSizeAF is
+ // twice as large because data is expanded to 16-bit before being stored in buffer.
+ size_t mFrameSize; // app-level frame size
+ size_t mFrameSizeAF; // AudioFlinger frame size
+
status_t mStatus;
uint32_t mLatency;
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 8ef90c7..ac5372f 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -77,12 +77,9 @@
uint32_t sampleRate;
- // NOTE: audio_track_cblk_t::frameSize is not equal to AudioTrack::frameSize() for
- // 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
+ uint8_t mPad2; // unused
// read-only for client, server writes once at initialization and is then read-only
- uint8_t frameSize; // would normally be size_t, but 8 bits is plenty
uint8_t mName; // normal tracks: track name, fast tracks: track index
// used by client only
@@ -111,7 +108,7 @@
// if there is a shared buffer, "buffers" is the value of pointer() for the shared
// buffer, otherwise "buffers" points immediately after the control block
- void* buffer(void *buffers, uint32_t offset) const;
+ void* buffer(void *buffers, uint32_t frameSize, uint32_t offset) const;
uint32_t framesAvailableIn() { return framesAvailable(false); }
uint32_t framesAvailableOut() { return framesAvailable(true); }
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 263a7c7..b40aaf5 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -213,6 +213,13 @@
mFrameCount = mCblk->frameCount;
mChannelCount = (uint8_t)channelCount;
mChannelMask = channelMask;
+
+ if (audio_is_linear_pcm(mFormat)) {
+ mFrameSize = channelCount * audio_bytes_per_sample(format);
+ } else {
+ mFrameSize = sizeof(uint8_t);
+ }
+
mActive = false;
mCbf = cbf;
mNotificationFrames = notificationFrames;
@@ -258,15 +265,6 @@
return mFrameCount;
}
-size_t AudioRecord::frameSize() const
-{
- if (audio_is_linear_pcm(mFormat)) {
- return channelCount()*audio_bytes_per_sample(mFormat);
- } else {
- return sizeof(uint8_t);
- }
-}
-
audio_source_t AudioRecord::inputSource() const
{
return mInputSource;
@@ -560,8 +558,8 @@
}
audioBuffer->frameCount = framesReq;
- audioBuffer->size = framesReq*cblk->frameSize;
- audioBuffer->raw = cblk->buffer(mBuffers, u);
+ audioBuffer->size = framesReq * mFrameSize;
+ audioBuffer->raw = cblk->buffer(mBuffers, mFrameSize, u);
active = mActive;
return active ? status_t(NO_ERROR) : status_t(STOPPED);
}
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 26cf877..4a4759e 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -286,6 +286,15 @@
mFormat = format;
mChannelMask = channelMask;
mChannelCount = channelCount;
+
+ if (audio_is_linear_pcm(format)) {
+ mFrameSize = channelCount * audio_bytes_per_sample(format);
+ mFrameSizeAF = channelCount * sizeof(int16_t);
+ } else {
+ mFrameSize = sizeof(uint8_t);
+ mFrameSizeAF = sizeof(uint8_t);
+ }
+
mSharedBuffer = sharedBuffer;
mMuted = false;
mActive = false;
@@ -332,15 +341,6 @@
return mCblk->frameCount;
}
-size_t AudioTrack::frameSize() const
-{
- if (audio_is_linear_pcm(mFormat)) {
- return channelCount()*audio_bytes_per_sample(mFormat);
- } else {
- return sizeof(uint8_t);
- }
-}
-
sp<IMemory>& AudioTrack::sharedBuffer()
{
return mSharedBuffer;
@@ -1026,8 +1026,8 @@
}
audioBuffer->frameCount = framesReq;
- audioBuffer->size = framesReq * cblk->frameSize;
- audioBuffer->raw = cblk->buffer(mBuffers, u);
+ audioBuffer->size = framesReq * mFrameSizeAF;
+ audioBuffer->raw = cblk->buffer(mBuffers, mFrameSizeAF, u);
active = mActive;
return active ? status_t(NO_ERROR) : status_t(STOPPED);
}
@@ -1302,7 +1302,7 @@
// NOTE: cblk->frameSize is not equal to AudioTrack::frameSize() for
// 8 bit PCM data: in this case, cblk->frameSize is based on a sample size of
// 16 bit.
- audioBuffer.frameCount = writtenSize/cblk->frameSize;
+ audioBuffer.frameCount = writtenSize / mFrameSizeAF;
frames -= audioBuffer.frameCount;
@@ -1373,7 +1373,7 @@
if (user > server) {
frames = ((user - server) > newCblk->frameCount) ?
newCblk->frameCount : (user - server);
- memset(mBuffers, 0, frames * newCblk->frameSize);
+ memset(mBuffers, 0, frames * mFrameSizeAF);
}
// restart playback even if buffer is not completely filled.
android_atomic_or(CBLK_FORCEREADY, &newCblk->flags);
@@ -1588,7 +1588,7 @@
return true;
}
-void* audio_track_cblk_t::buffer(void *buffers, uint32_t offset) const
+void* audio_track_cblk_t::buffer(void *buffers, size_t frameSize, uint32_t offset) const
{
return (int8_t *)buffers + (offset - userBase) * frameSize;
}
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;
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index fc24bed..38744d0 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -472,6 +472,10 @@
const uint32_t mSampleRate; // initial sample rate only; for tracks which
// support dynamic rates, the current value is in control block
const audio_format_t mFormat;
+ size_t mFrameSize; // AudioFlinger's view of frame size in shared memory,
+ // where for AudioTrack (but not AudioRecord),
+ // 8-bit PCM samples are stored as 16-bit
+ // FIXME should be const
bool mStepServerFailed;
const int mSessionId;
uint8_t mChannelCount;