Split libjingle's signaling thread from the UI thread

content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h

 // Copyright 2014 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_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_
 
 #include <vector>
 
 #include "base/memory/ref_counted.h"
 #include "base/memory/scoped_vector.h"
+#include "base/single_thread_task_runner.h"
 #include "base/synchronization/lock.h"
 #include "base/threading/thread_checker.h"
 #include "content/common/content_export.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediastreamtrack.h"
 #include "third_party/libjingle/source/talk/media/base/audiorenderer.h"
 
 namespace cricket {
 class AudioRenderer;
 }
 
@@ skipping 12 matching lines @@
     : NON_EXPORTED_BASE(public cricket::AudioRenderer),
       NON_EXPORTED_BASE(
           public webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>) {
  public:
   static scoped_refptr<WebRtcLocalAudioTrackAdapter> Create(
       const std::string& label,
       webrtc::AudioSourceInterface* track_source);
 
   WebRtcLocalAudioTrackAdapter(
       const std::string& label,
-      webrtc::AudioSourceInterface* track_source);
+      webrtc::AudioSourceInterface* track_source,
+      const scoped_refptr<base::SingleThreadTaskRunner>& signaling_thread);
 
   ~WebRtcLocalAudioTrackAdapter() override;
 
   void Initialize(WebRtcLocalAudioTrack* owner);
 
   std::vector<int> VoeChannels() const;
 
   // Called on the audio thread by the WebRtcLocalAudioTrack to set the signal
   // level of the audio data.
   void SetSignalLevel(int signal_level);
 
   // Method called by the WebRtcLocalAudioTrack to set the processor that
   // applies signal processing on the data of the track.
   // This class will keep a reference of the |processor|.
   // Called on the main render thread.
   void SetAudioProcessor(
       const scoped_refptr<MediaStreamAudioProcessor>& processor);
 
  private:
   // webrtc::MediaStreamTrack implementation.
   std::string kind() const override;
+  bool set_enabled(bool enable) override;
 
   // webrtc::AudioTrackInterface implementation.
   void AddSink(webrtc::AudioTrackSinkInterface* sink) override;
   void RemoveSink(webrtc::AudioTrackSinkInterface* sink) override;
   bool GetSignalLevel(int* level) override;
   rtc::scoped_refptr<webrtc::AudioProcessorInterface> GetAudioProcessor()
       override;
 
   // cricket::AudioCapturer implementation.
   void AddChannel(int channel_id) override;
   void RemoveChannel(int channel_id) override;
 
   // webrtc::AudioTrackInterface implementation.
   webrtc::AudioSourceInterface* GetSource() const override;
   cricket::AudioRenderer* GetRenderer() override;
 
   // Weak reference.
   WebRtcLocalAudioTrack* owner_;
 
   // The source of the audio track which handles the audio constraints.
   // TODO(xians): merge |track_source_| to |capturer_| in WebRtcLocalAudioTrack.
   rtc::scoped_refptr<webrtc::AudioSourceInterface> track_source_;
 
+  // Libjingle's signaling thread.
+  const scoped_refptr<base::SingleThreadTaskRunner> signaling_thread_;
+
   // The audio processsor that applies audio processing on the data of audio
   // track.
   scoped_refptr<MediaStreamAudioProcessor> audio_processor_;
 
   // A vector of WebRtc VoE channels that the capturer sends data to.
   std::vector<int> voe_channels_;
 
   // A vector of the peer connection sink adapters which receive the audio data
   // from the audio track.
   ScopedVector<WebRtcAudioSinkAdapter> sink_adapters_;
 
   // The amplitude of the signal.
   int signal_level_;
 
   // Thread checker for libjingle's signaling thread.
-  base::ThreadChecker signaling_thread_;
+  base::ThreadChecker signaling_thread_checker_;
   base::ThreadChecker capture_thread_;
 
   // Protects |voe_channels_|, |audio_processor_| and |signal_level_|.
   mutable base::Lock lock_;
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_