Bug 1397793 - Allow switching processing on/off dynamically. r=pehrsons

MozReview-Commit-ID: G0NJRkKEVeM

--HG--
extra : rebase_source : 0574254dfbfa0965ed4d1c8a07c9f535c390d8a1
extra : source : d12894d7777055463b68b8d9ea7e3d6c0940b4aa

dom/media/webrtc/MediaEngineWebRTC.h

@@ -508,12 +508,12 @@ private:
 
   RefPtr<mozilla::AudioInput> mAudioInput;
   RefPtr<WebRTCAudioDataListener> mListener;
-  RefPtr<AudioOutputObserver> mAudioOutputObserver;
 
   // Note: shared across all microphone sources
   static int sChannelsOpen;
 
   const UniquePtr<webrtc::AudioProcessing> mAudioProcessing;
+  const RefPtr<AudioOutputObserver> mAudioOutputObserver;
 
   // accessed from the GraphDriver thread except for deletion
   nsAutoPtr<AudioPacketizer<AudioDataValue, float>> mPacketizer;
@@ -543,7 +543,7 @@ private:
   // mSkipProcessing is true if none of the processing passes are enabled,
   // because of prefs or constraints. This allows simply copying the audio into
   // the MSG, skipping resampling and the whole webrtc.org code.
-  bool mSkipProcessing;
+  std::atomic_bool mSkipProcessing;
 
   // To only update microphone when needed, we keep track of previous settings.
   MediaEnginePrefs mLastPrefs;

dom/media/webrtc/MediaEngineWebRTCAudio.cpp

@@ -588,7 +588,7 @@ MediaEngineWebRTCMicrophoneSource::Start(SourceMediaStream *aStream,
   if (mSampleFrequency == MediaEngine::USE_GRAPH_RATE) {
     mSampleFrequency = aStream->GraphRate();
   }
-  aStream->AddAudioTrack(aID, mSampleFrequency, 0, segment, SourceMediaStream::ADDTRACK_QUEUED);
+  aStream->AddAudioTrack(aID, aStream->GraphRate(), 0, segment, SourceMediaStream::ADDTRACK_QUEUED);
 
   // XXX Make this based on the pref.
   aStream->RegisterForAudioMixing();
@@ -609,9 +609,7 @@ MediaEngineWebRTCMicrophoneSource::Start(SourceMediaStream *aStream,
   // Make sure logger starts before capture
   AsyncLatencyLogger::Get(true);
 
-  if (mAudioOutputObserver) {
-    mAudioOutputObserver->Clear();
-  }
+  mAudioOutputObserver->Clear();
 
   mAudioInput->StartRecording(aStream, mListener);
 
@@ -675,7 +673,7 @@ MediaEngineWebRTCMicrophoneSource::NotifyOutputData(MediaStreamGraph* aGraph,
                                                     TrackRate aRate,
                                                     uint32_t aChannels)
 {
-  if (mAudioOutputObserver) {
+  if (!PassThrough()) {
     mAudioOutputObserver->InsertFarEnd(aBuffer, aFrames, false,
                                   aRate, aChannels);
   }
@@ -714,7 +712,7 @@ MediaEngineWebRTCMicrophoneSource::PacketizeAndProcess(MediaStreamGraph* aGraph,
   while (mAudioOutputObserver->Size() > 0) {
     // Bug 1414837: This will call `free()`, and we should remove it.
     // Pop gives ownership.
-    UniquePtr<FarEndAudioChunk> buffer(mAudioOutputObserver->Pop()); // only call if size() > 0
+    nsAutoPtr<FarEndAudioChunk> buffer(mAudioOutputObserver->Pop()); // only call if size() > 0
     if (!buffer) {
       continue;
     }