Bug 693095 - Fix audio stream position estimation for remoted streams. Also resurrect audio thread wait removed in bug 669556 when using remoted audio streams. r=cpearce

content/media/VideoUtils.h

@@ -166,4 +166,10 @@ void ScaleDisplayByAspectRatio(nsIntSize& aDisplay, float aAspectRatio);
 #define MEDIA_THREAD_STACK_SIZE nsIThreadManager::DEFAULT_STACK_SIZE
 #endif
 
+// Android's audio backend is not available in content processes, so audio must
+// be remoted to the parent chrome process.
+#if defined(ANDROID)
+#define REMOTE_AUDIO 1
+#endif
+
 #endif

content/media/nsAudioStream.cpp

@@ -66,12 +66,6 @@ using namespace mozilla;
 #define SA_PER_STREAM_VOLUME 1
 #endif
 
-// Android's audio backend is not available in content processes, so audio must
-// be remoted to the parent chrome process.
-#if defined(ANDROID)
-#define REMOTE_AUDIO 1
-#endif
-
 using mozilla::TimeStamp;
 
 #ifdef PR_LOGGING
@@ -80,6 +74,9 @@ PRLogModuleInfo* gAudioStreamLog = nsnull;
 
 static const PRUint32 FAKE_BUFFER_SIZE = 176400;
 
+// Number of milliseconds per second.
+static const PRInt64 MS_PER_S = 1000;
+
 class nsNativeAudioStream : public nsAudioStream
 {
  public:
@@ -752,9 +749,9 @@ nsRemotedAudioStream::GetPositionInFrames()
     return 0;
 
   PRInt64 time = mAudioChild->GetLastKnownPositionTimestamp();
-  PRInt64 result = position + (mRate * (PR_IntervalNow() - time) / USECS_PER_S);
+  PRInt64 dt = PR_IntervalToMilliseconds(PR_IntervalNow() - time);
 
-  return result;
+  return position + (mRate * dt / MS_PER_S);
 }
 
 bool

content/media/nsBuiltinDecoderStateMachine.cpp

@@ -635,6 +635,29 @@ void nsBuiltinDecoderStateMachine::AudioLoop()
         NS_WARNING("Int overflow calculating audio end time");
         break;
       }
+
+// The remoted audio stream does not block writes when the other end's buffers
+// are full, so this sleep is necessary to stop the audio thread spinning its
+// wheels.  When bug 695612 is fixed, this block of code can be removed.
+#if defined(REMOTE_AUDIO)
+      PRInt64 audioAhead = mAudioEndTime - GetMediaTime();
+      if (audioAhead > AMPLE_AUDIO_USECS &&
+          framesWritten > minWriteFrames)
+      {
+        // We've pushed enough audio onto the hardware that we've queued up a
+        // significant amount ahead of the playback position. The decode
+        // thread will be going to sleep, so we won't get any new audio
+        // anyway, so sleep until we need to push to the hardware again.
+        Wait(AMPLE_AUDIO_USECS / 2);
+        // Kick the decode thread; since above we only do a NotifyAll when
+        // we pop an audio chunk of the queue, the decoder won't wake up if
+        // we've got no more decoded chunks to push to the hardware. We can
+        // hit this condition if the last frame in the stream doesn't have
+        // it's EOS flag set, and the decode thread sleeps just after decoding
+        // that packet, but before realising there's no more packets.
+        mon.NotifyAll();
+      }
+#endif
     }
   }
   if (mReader->mAudioQueue.AtEndOfStream() &&