Bug 463627 - Audio playback stops after some seconds on Linux - r=kinetik sr=roc

content/media/video/public/nsAudioStream.h

@@ -40,6 +40,7 @@
 
 #include "nscore.h"
 #include "prlog.h"
+#include "nsTArray.h"
 
 extern PRLogModuleInfo* gAudioStreamLog;
 
@@ -117,6 +118,12 @@ class nsAudioStream
 
   SampleFormat mFormat;
 
+  // When a Write() request is made, and the number of samples
+  // requested to be written exceeds the buffer size of the audio
+  // backend, the remaining samples are stored in this variable. They
+  // will be written on the next Write() request.
+  nsTArray<short> mBufferOverflow;
+
   PRPackedBool mPaused;
 };
 #endif

content/media/video/src/nsAudioStream.cpp

@@ -122,49 +122,63 @@ void nsAudioStream::Write(const void* aBuf, PRUint32 aCount)
                     "Buffer size must be divisible by channel count");
 
   mSamplesBuffered += aCount;
+  PRUint32 offset = mBufferOverflow.Length();
+  PRInt32 count = aCount + offset;
 
   if (!mAudioHandle)
     return;
 
-  nsAutoArrayPtr<short> s_data(new short[aCount]);
+  nsAutoArrayPtr<short> s_data(new short[count]);
 
   if (s_data) {
+    for (PRUint32 i=0; i < offset; ++i) {
+      s_data[i] = mBufferOverflow.ElementAt(i);
+    }
+    mBufferOverflow.Clear();
+
     switch (mFormat) {
-    case FORMAT_U8: {
+      case FORMAT_U8: {
-      const PRUint8* buf = static_cast<const PRUint8*>(aBuf);
+        const PRUint8* buf = static_cast<const PRUint8*>(aBuf);
-      PRInt32 volume = PRInt32((1 << 16) * mVolume);
+        PRInt32 volume = PRInt32((1 << 16) * mVolume);
-      for (PRUint32 i = 0; i < aCount; ++i) {
+        for (PRUint32 i = 0; i < aCount; ++i) {
-        s_data[i] = short(((PRInt32(buf[i]) - 128) * volume) >> 8);
+          s_data[i + offset] = short(((PRInt32(buf[i]) - 128) * volume) >> 8);
-      }
-      break;
-    }
-    case FORMAT_S16_LE: {
-      const short* buf = static_cast<const short*>(aBuf);
-      PRInt32 volume = PRInt32((1 << 16) * mVolume);
-      for (PRUint32 i = 0; i < aCount; ++i) {
-        s_data[i] = short((PRInt32(buf[i]) * volume) >> 16);
-      }
-      break;
-    }
-    case FORMAT_FLOAT32_LE: {
-      const float* buf = static_cast<const float*>(aBuf);
-      for (PRUint32 i= 0; i <  aCount; ++i) {
-        float scaled_value = floorf(0.5 + 32768 * buf[i] * mVolume);
-        if (buf[i] < 0.0) {
-          s_data[i] = (scaled_value < -32768.0) ?
-            -32768 :
-            short(scaled_value);
-        } else {
-          s_data[i] = (scaled_value > 32767.0) ?
-            32767 :
-            short(scaled_value);
         }
+        break;
+      }
+      case FORMAT_S16_LE: {
+        const short* buf = static_cast<const short*>(aBuf);
+        PRInt32 volume = PRInt32((1 << 16) * mVolume);
+        for (PRUint32 i = 0; i < aCount; ++i) {
+          s_data[i + offset] = short((PRInt32(buf[i]) * volume) >> 16);
+        }
+        break;
+      }
+      case FORMAT_FLOAT32_LE: {
+        const float* buf = static_cast<const float*>(aBuf);
+        for (PRUint32 i = 0; i <  aCount; ++i) {
+          float scaled_value = floorf(0.5 + 32768 * buf[i] * mVolume);
+          if (buf[i] < 0.0) {
+            s_data[i + offset] = (scaled_value < -32768.0) ?
+              -32768 :
+              short(scaled_value);
+          } else {
+            s_data[i+offset] = (scaled_value > 32767.0) ?
+              32767 :
+              short(scaled_value);
+          }
+        }
+        break;
       }
-      break;
-    }
     }
 
-    if (sa_stream_write(static_cast<sa_stream_t*>(mAudioHandle), s_data.get(), aCount * sizeof(short)) != SA_SUCCESS) {
+    PRInt32 available = Available();
+    if (available < count) {
+      mBufferOverflow.AppendElements(s_data.get() + available, (count - available));
+      count = available;
+    }
+
+    if (sa_stream_write(static_cast<sa_stream_t*>(mAudioHandle),
+       s_data.get(), count * sizeof(short)) != SA_SUCCESS) {
       PR_LOG(gAudioStreamLog, PR_LOG_ERROR, ("nsAudioStream: sa_stream_write error"));
       Shutdown();
     }
@@ -197,11 +211,21 @@ void nsAudioStream::SetVolume(float aVolume)
 void nsAudioStream::Drain()
 {
   if (!mAudioHandle) {
+    // mSamplesBuffered already accounts for the data in the
+    // mBufferOverflow array.
     PRUint32 drainTime = (float(mSamplesBuffered) / mRate / mChannels - GetTime()) * 1000.0;
     PR_Sleep(PR_MillisecondsToInterval(drainTime));
     return;
   }
 
+  // Write any remaining unwritten sound data in the overflow buffer
+  if (!mBufferOverflow.IsEmpty()) {
+    if (sa_stream_write(static_cast<sa_stream_t*>(mAudioHandle),
+                        mBufferOverflow.Elements(),
+                        mBufferOverflow.Length() * sizeof(short)) != SA_SUCCESS)
+      return;
+  }
+
   if (sa_stream_drain(static_cast<sa_stream_t*>(mAudioHandle)) != SA_SUCCESS) {
         PR_LOG(gAudioStreamLog, PR_LOG_ERROR, ("nsAudioStream: sa_stream_drain error"));
         Shutdown();

content/media/video/src/nsOggDecoder.cpp

@@ -141,6 +141,15 @@ public:
       MOZ_COUNT_DTOR(FrameData);
     }
 
+    // Write the audio data from the frame to the Audio stream.
+    void Write(nsAudioStream* aStream)
+    {
+      PRUint32 length = mAudioData.Length();
+      if (length == 0)
+        return;
+
+      aStream->Write(mAudioData.Elements(), length);
+    }
 
     nsAutoArrayPtr<unsigned char> mVideoData;
     nsTArray<float> mAudioData;
@@ -260,9 +269,8 @@ public:
   void PlayVideo(FrameData* aFrame);
 
   // Play the audio data from the given frame. The decode monitor must
-  // be locked when calling this method. Returns PR_FALSE if unable to
+  // be locked when calling this method.
-  // write to the audio device without blocking.
+  void PlayAudio(FrameData* aFrame);
-  PRBool PlayAudio(FrameData* aFrame);
 
   // Called from the main thread to get the current frame time. The decoder
   // monitor must be obtained before calling this.
@@ -629,13 +637,13 @@ void nsOggDecodeStateMachine::PlayFrame() {
 
       double time = (PR_IntervalToMilliseconds(PR_IntervalNow()-mPlayStartTime-mPauseDuration)/1000.0);
       if (time >= frame->mTime) {
-        mDecodedFrames.Pop();
         // Audio for the current frame is played, but video for the next frame
         // is displayed, to account for lag from the time the audio is written
         // to when it is played. This will go away when we move to a/v sync
         // using the audio hardware clock.
-        PlayVideo(mDecodedFrames.IsEmpty() ? frame : mDecodedFrames.Peek());
         PlayAudio(frame);
+        mDecodedFrames.Pop();
+        PlayVideo(mDecodedFrames.IsEmpty() ? frame : mDecodedFrames.Peek());
         UpdatePlaybackPosition(frame->mDecodedFrameTime);
         delete frame;
       }
@@ -677,17 +685,13 @@ void nsOggDecodeStateMachine::PlayVideo(FrameData* aFrame)
   }
 }
 
-PRBool nsOggDecodeStateMachine::PlayAudio(FrameData* aFrame)
+void nsOggDecodeStateMachine::PlayAudio(FrameData* aFrame)
 {
   //  NS_ASSERTION(PR_InMonitor(mDecoder->GetMonitor()), "PlayAudio() called without acquiring decoder monitor");
-  if (mAudioStream && aFrame && !aFrame->mAudioData.IsEmpty()) {
+  if (!mAudioStream)
-    if (PRUint32(mAudioStream->Available()) < aFrame->mAudioData.Length())
+    return;
-      return PR_FALSE;
 
-    mAudioStream->Write(aFrame->mAudioData.Elements(), aFrame->mAudioData.Length());
+  aFrame->Write(mAudioStream);
-  }
-
-  return PR_TRUE;
 }
 
 void nsOggDecodeStateMachine::OpenAudioStream()
@@ -1054,6 +1058,14 @@ nsresult nsOggDecodeStateMachine::Run()
         if (mState != DECODER_STATE_COMPLETED)
           continue;
 
+        if (mAudioStream) {
+          mon.Exit();
+          mAudioStream->Drain();
+          mon.Enter();
+          if (mState != DECODER_STATE_COMPLETED)
+            continue;
+        }
+
         nsCOMPtr<nsIRunnable> event =
           NS_NEW_RUNNABLE_METHOD(nsOggDecoder, mDecoder, PlaybackEnded);
         NS_DispatchToMainThread(event, NS_DISPATCH_NORMAL);