Revert of MediaStream audio object graph untangling and clean-ups.

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 "content/renderer/media/media_stream_audio_level_calculator.h"
-#include "content/renderer/media/media_stream_audio_processor.h"
 #include "third_party/webrtc/api/mediastreamtrack.h"
 #include "third_party/webrtc/media/base/audiorenderer.h"
 
 namespace cricket {
 class AudioRenderer;
 }
 
 namespace webrtc {
 class AudioSourceInterface;
 class AudioProcessorInterface;
 }
 
 namespace content {
 
 class MediaStreamAudioProcessor;
 class WebRtcAudioSinkAdapter;
 class WebRtcLocalAudioTrack;
 
-// Provides an implementation of the webrtc::AudioTrackInterface that can be
-// bound/unbound to/from a MediaStreamAudioTrack.  In other words, this is an
-// adapter that sits between the media stream object graph and WebRtc's object
-// graph and proxies between the two.
 class CONTENT_EXPORT WebRtcLocalAudioTrackAdapter
     : 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,
-      scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner);
+      const scoped_refptr<base::SingleThreadTaskRunner>& signaling_thread);
 
   ~WebRtcLocalAudioTrackAdapter() override;
 
   void Initialize(WebRtcLocalAudioTrack* owner);
 
-  // Set the object that provides shared access to the current audio signal
-  // level.  This method may only be called once, before the audio data flow
-  // starts, and before any calls to GetSignalLevel() might be made.
-  void SetLevel(scoped_refptr<MediaStreamAudioLevelCalculator::Level> level);
+  // 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.
-  // This method may only be called once, before the audio data flow starts, and
-  // before any calls to GetAudioProcessor() might be made.
-  void SetAudioProcessor(scoped_refptr<MediaStreamAudioProcessor> processor);
+  void SetAudioProcessor(
+      const scoped_refptr<MediaStreamAudioProcessor>& processor);
 
   // webrtc::MediaStreamTrack implementation.
   std::string kind() const override;
   bool set_enabled(bool enable) override;
 
  private:
   // 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;
   webrtc::AudioSourceInterface* GetSource() const 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_task_runner_;
+  const scoped_refptr<base::SingleThreadTaskRunner> signaling_thread_;
 
   // The audio processsor that applies audio processing on the data of audio
-  // track.  This must be set before calls to GetAudioProcessor() are made.
+  // 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_;
 
-  // Thread-safe accessor to current audio signal level.  This must be set
-  // before calls to GetSignalLevel() are made.
-  scoped_refptr<MediaStreamAudioLevelCalculator::Level> level_;
+  // The amplitude of the signal.
+  int signal_level_;
+
+  // Thread checker for libjingle's 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_