TimedAudioTrack: Optimize the queue trim operation.

Hand merge from ics-aah

> TimedAudioTrack: Optimize the queue trim operation.
>
> Don't perform the end PTS calculation for each buffer during trimming.
> Instead, only calculate the ending PTS of a buffer if there is no next
> buffer in the queue.  This optimization assumes that the buffers being
> queued are in monotonic media time order (a fair assumption for now)
> and that the timestamps in the audio are contiguous (not a requirement
> for this API, but a reality of how it is being used right now).
>
> In the case where the audio is discontinuous on purpose, it is
> that this optimization will cause the system hold one extra buffer
> which it could have safely trimmed.  It should not be much of an issue
> since in real life the audio is almost always contiguous, and as long
> as the media clock is running and the mixer is mixing, the buffer will
> be used up and discard as part of the normal flow anyway.
>
> Change-Id: I00061e85ee7d5651fcf80751646c7d7415894a14
> Signed-off-by: John Grossman <johngro@google.com>

Change-Id: I0054b58e1389fa005aa990cb5710caf4af7b706a
Signed-off-by: John Grossman <johngro@google.com>

1 files changed, 29 insertions, 11 deletions

diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 3a6e476..bce30d7 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -3994,20 +3994,38 @@ void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() {
 
     size_t trimEnd;
     for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) {
-        int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size()
-                           / mCblk->frameSize;
         int64_t bufEnd;
 
-        if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount,
-                                                            &bufEnd)) {
-            ALOGE("Failed to convert frame count of %lld to media time duration"
-                  " (scale factor %d/%u) in %s", frameCount,
-                  mMediaTimeToSampleTransform.a_to_b_numer,
-                  mMediaTimeToSampleTransform.a_to_b_denom,
-                  __PRETTY_FUNCTION__);
-            break;
+        if ((trimEnd + 1) < mTimedBufferQueue.size()) {
+            // We have a next buffer.  Just use its PTS as the PTS of the frame
+            // following the last frame in this buffer.  If the stream is sparse
+            // (ie, there are deliberate gaps left in the stream which should be
+            // filled with silence by the TimedAudioTrack), then this can result
+            // in one extra buffer being left un-trimmed when it could have
+            // been.  In general, this is not typical, and we would rather
+            // optimized away the TS calculation below for the more common case
+            // where PTSes are contiguous.
+            bufEnd = mTimedBufferQueue[trimEnd + 1].pts();
+        } else {
+            // We have no next buffer.  Compute the PTS of the frame following
+            // the last frame in this buffer by computing the duration of of
+            // this frame in media time units and adding it to the PTS of the
+            // buffer.
+            int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size()
+                               / mCblk->frameSize;
+
+            if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount,
+                                                                &bufEnd)) {
+                ALOGE("Failed to convert frame count of %lld to media time"
+                      " duration" " (scale factor %d/%u) in %s",
+                      frameCount,
+                      mMediaTimeToSampleTransform.a_to_b_numer,
+                      mMediaTimeToSampleTransform.a_to_b_denom,
+                      __PRETTY_FUNCTION__);
+                break;
+            }
+            bufEnd += mTimedBufferQueue[trimEnd].pts();
         }
-        bufEnd += mTimedBufferQueue[trimEnd].pts();
 
         if (bufEnd > mediaTimeNow)
             break;