TimedAudio: Track of the number of pending frames.
This is a manual merge from ics-aah
> TimedAudio: Track of the number of pending frames.
>
> Keep track of the number of frames pending in the timed audio queue so
> we can implement framesReady in O(1) time instead of O(N). This
> change partially addresses bug 6020970; the bug will be completely
> addressed once this change has been up-integrated into master.
>
> Change-Id: I599eb15ea1f6d715b97b30e65214fb6fadd169df
> Signed-off-by: John Grossman <johngro@google.com>
Change-Id: I6cbbbc3afc8efd066fe94865326ede0c6b3db2bd
Signed-off-by: John Grossman <johngro@google.com>
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index ecc6147..70fbf02 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -74,6 +74,18 @@
// ----------------------------------------------------------------------------
+// Note: the following macro is used for extremely verbose logging message. In
+// order to run with ALOG_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 ALOG_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 ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
namespace android {
@@ -3898,6 +3910,7 @@
frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED),
mQueueHeadInFlight(false),
mTrimQueueHeadOnRelease(false),
+ mFramesPendingInQueue(0),
mTimedSilenceBuffer(NULL),
mTimedSilenceBufferSize(0),
mTimedAudioOutputOnTime(false),
@@ -3976,9 +3989,9 @@
}
}
- 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;
@@ -3991,7 +4004,7 @@
__PRETTY_FUNCTION__);
break;
}
- bufEnd += mTimedBufferQueue[trimIndex].pts();
+ bufEnd += mTimedBufferQueue[trimEnd].pts();
if (bufEnd > mediaTimeNow)
break;
@@ -3999,17 +4012,55 @@
// 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) {
@@ -4021,6 +4072,8 @@
Mutex::Autolock _l(mTimedBufferQueueLock);
+ uint32_t bufFrames = buffer->size() / mCblk->frameSize;
+ mFramesPendingInQueue += bufFrames;
mTimedBufferQueue.add(TimedBuffer(buffer, pts));
return NO_ERROR;
@@ -4029,9 +4082,9 @@
status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform(
const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) {
- ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__,
- xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom,
- target);
+ ALOGVV("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);
if (!(target == TimedAudioTrack::LOCAL_TIME ||
target == TimedAudioTrack::COMMON_TIME)) {
@@ -4096,7 +4149,7 @@
ALOGW("timedGetNextBuffer transform failed");
buffer->raw = 0;
buffer->frameCount = 0;
- mTimedBufferQueue.removeAt(0);
+ trimTimedBufferQueueHead_l("getNextBuffer; no transform");
return NO_ERROR;
}
@@ -4125,20 +4178,23 @@
int64_t sampleDelta;
if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) {
ALOGV("*** 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)) {
- ALOGV("*** too late during sample rate transform: dropped buffer");
- mTimedBufferQueue.removeAt(0);
+ LOGV("*** too late during sample rate transform: dropped buffer");
+ trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample");
continue;
}
- ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]",
- __PRETTY_FUNCTION__, head.pts(), head.position(), pts,
- static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)),
- static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF));
+ ALOGVV("*** 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));
// if the delta between the ideal placement for the next input sample and
// the current output position is within this threshold, then we will
@@ -4156,7 +4212,8 @@
// with the last output
timedYieldSamples_l(buffer);
- ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
+ ALOGVV("*** 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
@@ -4176,7 +4233,7 @@
// all the remaining samples in the head are too late, so
// drop it and move on
ALOGV("*** too late: dropped buffer");
- mTimedBufferQueue.removeAt(0);
+ trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer");
continue;
} else {
// skip over the late samples
@@ -4273,9 +4330,16 @@
(buffer->frameCount * mCblk->frameSize));
mQueueHeadInFlight = false;
+ ALOG_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 {
@@ -4290,14 +4354,7 @@
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()
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 2812e27..9e6201a 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -771,32 +771,30 @@
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;
// AudioBufferProvider interface
- virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts);
+ 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(PlaybackThread *thread,
@@ -809,6 +807,14 @@
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;
@@ -817,6 +823,7 @@
Vector<TimedBuffer> mTimedBufferQueue;
bool mQueueHeadInFlight;
bool mTrimQueueHeadOnRelease;
+ uint32_t mFramesPendingInQueue;
uint8_t* mTimedSilenceBuffer;
uint32_t mTimedSilenceBufferSize;