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Merge "TimedAudio: Track of the number of pending frames." into ics-aah

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This commit is contained in:
John Grossman
2012-04-10 15:43:17 -07:00
committed by Android (Google) Code Review
parent 82321d1fbf e843f378c9
commit 5f48909604
2 changed files with 92 additions and 34 deletions
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101
services/audioflinger/AudioFlinger.cpp
View File

@@ -63,6 +63,18 @@
// ----------------------------------------------------------------------------
// Note: the following macro is used for extremely verbose logging message. In
// order to run with LOG_ASSERT turned on, we need to have LOG_NDEBUG set to 0;
// but one side effect of this is to turn all LOGV's as well. Some messages are
// so verbose that we want to suppress them even when we have LOG_ASSERT turned
// on. Do not uncomment the #def below unless you really know what you are
// doing and want to see all of the extremely verbose messages.
//#define VERY_VERY_VERBOSE_LOGGING
#ifdef VERY_VERY_VERBOSE_LOGGING
#define LOGVV LOGV
#else
#define LOGVV(a...) do { } while(0)
#endif
namespace android {
@@ -3843,6 +3855,7 @@ AudioFlinger::PlaybackThread::TimedTrack::TimedTrack(
frameCount, sharedBuffer, sessionId),
mQueueHeadInFlight(false),
mTrimQueueHeadOnRelease(false),
mFramesPendingInQueue(0),
mTimedSilenceBuffer(NULL),
mTimedSilenceBufferSize(0),
mTimedAudioOutputOnTime(false),
@@ -3921,9 +3934,9 @@ void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() {
}
}
size_t trimIndex;
for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) {
int64_t frameCount = mTimedBufferQueue[trimIndex].buffer()->size()
size_t trimEnd;
for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) {
int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size()
/ mCblk->frameSize;
int64_t bufEnd;
@@ -3936,7 +3949,7 @@ void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() {
__PRETTY_FUNCTION__);
break;
}
bufEnd += mTimedBufferQueue[trimIndex].pts();
bufEnd += mTimedBufferQueue[trimEnd].pts();
if (bufEnd > mediaTimeNow)
break;
@@ -3944,17 +3957,55 @@ void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() {
// Is the buffer we want to use in the middle of a mix operation right
// now? If so, don't actually trim it. Just wait for the releaseBuffer
// from the mixer which should be coming back shortly.
if (!trimIndex && mQueueHeadInFlight) {
if (!trimEnd && mQueueHeadInFlight) {
mTrimQueueHeadOnRelease = true;
}
}
size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0;
if (trimStart < trimIndex) {
mTimedBufferQueue.removeItemsAt(trimStart, trimIndex);
if (trimStart < trimEnd) {
// Update the bookkeeping for framesReady()
for (size_t i = trimStart; i < trimEnd; ++i) {
updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim");
}
// Now actually remove the buffers from the queue.
mTimedBufferQueue.removeItemsAt(trimStart, trimEnd);
}
}
void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l(
const char* logTag) {
LOG_ASSERT(mTimedBufferQueue.size() > 0,
"%s called (reason \"%s\"), but timed buffer queue has no"
" elements to trim.", __FUNCTION__, logTag);
updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag);
mTimedBufferQueue.removeAt(0);
}
void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l(
const TimedBuffer& buf,
const char* logTag) {
uint32_t bufBytes = buf.buffer()->size();
uint32_t consumedAlready = buf.position();
LOG_ASSERT(consumedAlready <= bufFrames,
"Bad bookkeeping while updating frames pending. Timed buffer is"
" only %u bytes long, but claims to have consumed %u"
" bytes. (update reason: \"%s\")",
bufFrames, consumedAlready, logTag);
uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize;
LOG_ASSERT(mFramesPendingInQueue >= bufFrames,
"Bad bookkeeping while updating frames pending. Should have at"
" least %u queued frames, but we think we have only %u. (update"
" reason: \"%s\")",
bufFrames, mFramesPendingInQueue, logTag);
mFramesPendingInQueue -= bufFrames;
}
status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer(
const sp<IMemory>& buffer, int64_t pts) {
@@ -3966,6 +4017,8 @@ status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer(
Mutex::Autolock _l(mTimedBufferQueueLock);
uint32_t bufFrames = buffer->size() / mCblk->frameSize;
mFramesPendingInQueue += bufFrames;
mTimedBufferQueue.add(TimedBuffer(buffer, pts));
return NO_ERROR;
@@ -3974,7 +4027,7 @@ status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer(
status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform(
const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) {
LOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__,
LOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d",
xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom,
target);
@@ -4050,7 +4103,7 @@ status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
LOGW("timedGetNextBuffer transform failed");
buffer->raw = 0;
buffer->frameCount = 0;
mTimedBufferQueue.removeAt(0);
trimTimedBufferQueueHead_l("getNextBuffer; no transform");
return NO_ERROR;
}
@@ -4079,18 +4132,18 @@ status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
int64_t sampleDelta;
if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) {
LOGV("*** head buffer is too far from PTS: dropped buffer");
mTimedBufferQueue.removeAt(0);
trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from mix");
continue;
}
if (!mLocalTimeToSampleTransform.doForwardTransform(
(effectivePTS - pts) << 32, &sampleDelta)) {
LOGV("*** too late during sample rate transform: dropped buffer");
mTimedBufferQueue.removeAt(0);
trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample");
continue;
}
LOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]",
__PRETTY_FUNCTION__, head.pts(), head.position(), pts,
LOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]",
head.pts(), head.position(), pts,
static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)),
static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF));
@@ -4110,7 +4163,7 @@ status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
// with the last output
timedYieldSamples_l(buffer);
LOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
LOGVV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
return NO_ERROR;
} else if (sampleDelta > 0) {
// the gap between the current output position and the proper start of
@@ -4130,7 +4183,7 @@ status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
// all the remaining samples in the head are too late, so
// drop it and move on
LOGV("*** too late: dropped buffer");
mTimedBufferQueue.removeAt(0);
trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer");
continue;
} else {
// skip over the late samples
@@ -4226,9 +4279,16 @@ void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer(
(buffer->frameCount * mCblk->frameSize));
mQueueHeadInFlight = false;
LOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount,
"Bad bookkeeping during releaseBuffer! Should have at"
" least %u queued frames, but we think we have only %u",
buffer->frameCount, mFramesPendingInQueue);
mFramesPendingInQueue -= buffer->frameCount;
if ((static_cast<size_t>(head.position()) >= head.buffer()->size())
|| mTrimQueueHeadOnRelease) {
mTimedBufferQueue.removeAt(0);
trimTimedBufferQueueHead_l("releaseBuffer");
mTrimQueueHeadOnRelease = false;
}
} else {
@@ -4243,14 +4303,7 @@ done:
uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const {
Mutex::Autolock _l(mTimedBufferQueueLock);
uint32_t frames = 0;
for (size_t i = 0; i < mTimedBufferQueue.size(); i++) {
const TimedBuffer& tb = mTimedBufferQueue[i];
frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize;
}
return frames;
return mFramesPendingInQueue;
}
AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer()

25
services/audioflinger/AudioFlinger.h
View File

@@ -697,32 +697,28 @@ private:
TimedBuffer(const sp<IMemory>& buffer, int64_t pts);
const sp<IMemory>& buffer() const { return mBuffer; }
int64_t pts() const { return mPTS; }
int position() const { return mPosition; }
void setPosition(int pos) { mPosition = pos; }
uint32_t position() const { return mPosition; }
void setPosition(uint32_t pos) { mPosition = pos; }
private:
sp<IMemory> mBuffer;
int64_t mPTS;
int mPosition;
int64_t mPTS;
uint32_t mPosition;
};
// Mixer facing methods.
virtual bool isTimedTrack() const { return true; }
virtual uint32_t framesReady() const;
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer,
int64_t pts);
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer);
void timedYieldSilence_l(uint32_t numFrames,
AudioBufferProvider::Buffer* buffer);
// Client/App facing methods.
status_t allocateTimedBuffer(size_t size,
sp<IMemory>* buffer);
status_t queueTimedBuffer(const sp<IMemory>& buffer,
int64_t pts);
status_t setMediaTimeTransform(const LinearTransform& xform,
TimedAudioTrack::TargetTimeline target);
void trimTimedBufferQueue_l();
private:
TimedTrack(const wp<ThreadBase>& thread,
@@ -735,6 +731,14 @@ private:
const sp<IMemory>& sharedBuffer,
int sessionId);
void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer);
void timedYieldSilence_l(uint32_t numFrames,
AudioBufferProvider::Buffer* buffer);
void trimTimedBufferQueue_l();
void trimTimedBufferQueueHead_l(const char* logTag);
void updateFramesPendingAfterTrim_l(const TimedBuffer& buf,
const char* logTag);
uint64_t mLocalTimeFreq;
LinearTransform mLocalTimeToSampleTransform;
LinearTransform mMediaTimeToSampleTransform;
@@ -743,6 +747,7 @@ private:
Vector<TimedBuffer> mTimedBufferQueue;
bool mQueueHeadInFlight;
bool mTrimQueueHeadOnRelease;
uint32_t mFramesPendingInQueue;
uint8_t* mTimedSilenceBuffer;
uint32_t mTimedSilenceBufferSize;