Split libjingle's signaling thread from the UI thread

content/renderer/media/webrtc_local_audio_track.h

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_
 
 #include <list>
 #include <string>
 
@@ skipping 42 matching lines @@
   // Add/remove PeerConnection sink to/from the track.
   // TODO(xians): Remove these two methods after PeerConnection can use the
   // same sink interface as MediaStreamAudioSink.
   void AddSink(PeerConnectionAudioSink* sink);
   void RemoveSink(PeerConnectionAudioSink* sink);
 
   // Starts the local audio track. Called on the main render thread and
   // should be called only once when audio track is created.
   void Start();
 
+  // Overrides for MediaStreamTrack.
+
+  void SetEnabled(bool enabled) override;
+
   // Stops the local audio track. Called on the main render thread and
   // should be called only once when audio track going away.
   void Stop() override;
 
+  webrtc::AudioTrackInterface* GetAudioAdapter() override;
+
   // Method called by the capturer to deliver the capture data.
   // Called on the capture audio thread.
   void Capture(const int16* audio_data,
                base::TimeDelta delay,
                int volume,
                bool key_pressed,
                bool need_audio_processing,
                bool force_report_nonzero_energy);
 
   // Method called by the capturer to set the audio parameters used by source
   // of the capture data..
   // Called on the capture audio thread.
   void OnSetFormat(const media::AudioParameters& params);
 
   // Method called by the capturer to set the processor that applies signal
   // processing on the data of the track.
   // Called on the capture audio thread.
   void SetAudioProcessor(
       const scoped_refptr<MediaStreamAudioProcessor>& processor);
 
  private:
   typedef TaggedList<MediaStreamAudioTrackSink> SinkList;
 
   // All usage of libjingle is through this adapter. The adapter holds
-  // a reference on this object, but not vice versa.
-  WebRtcLocalAudioTrackAdapter* adapter_;
+  // a pointer to this object, but no reference.
+  const scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_;
 
   // The provider of captured data to render.
   scoped_refptr<WebRtcAudioCapturer> capturer_;
 
   // The source of the audio track which is used by WebAudio, which provides
   // data to the audio track when hooking up with WebAudio.
   scoped_refptr<WebAudioCapturerSource> webaudio_source_;
 
   // A tagged list of sinks that the audio data is fed to. Tags
   // indicate tracks that need to be notified that the audio format
   // has changed.
   SinkList sinks_;
 
   // Used to DCHECK that some methods are called on the main render thread.
   base::ThreadChecker main_render_thread_checker_;
+  // Tests that methods are called on libjingle's signaling thread.
+  base::ThreadChecker signal_thread_checker_;
 
   // Used to DCHECK that some methods are called on the capture audio thread.
   base::ThreadChecker capture_thread_checker_;
 
   // Protects |params_| and |sinks_|.
   mutable base::Lock lock_;
 
   // Audio parameters of the audio capture stream.
   // Accessed on only the audio capture thread.
   media::AudioParameters audio_parameters_;
 
   // Used to calculate the signal level that shows in the UI.
   // Accessed on only the audio thread.
   scoped_ptr<MediaStreamAudioLevelCalculator> level_calculator_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_