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

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>
 
 #include "base/macros.h"
 #include "base/memory/ref_counted.h"
+#include "base/memory/scoped_ptr.h"
 #include "base/synchronization/lock.h"
 #include "base/threading/thread_checker.h"
 #include "content/renderer/media/media_stream_audio_track.h"
 #include "content/renderer/media/tagged_list.h"
-#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
 #include "media/audio/audio_parameters.h"
 
 namespace media {
 class AudioBus;
 }
 
 namespace content {
 
 class MediaStreamAudioLevelCalculator;
 class MediaStreamAudioProcessor;
 class MediaStreamAudioSink;
 class MediaStreamAudioSinkOwner;
 class MediaStreamAudioTrackSink;
+class WebAudioCapturerSource;
+class WebRtcAudioCapturer;
+class WebRtcLocalAudioTrackAdapter;
 
-// A WebRtcLocalAudioTrack manages thread-safe connects/disconnects to sinks,
-// and the delivery of audio data from the source to the sinks.
+// A WebRtcLocalAudioTrack instance contains the implementations of
+// MediaStreamTrackExtraData.
+// When an instance is created, it will register itself as a track to the
+// WebRtcAudioCapturer to get the captured data, and forward the data to
+// its |sinks_|. The data flow can be stopped by disabling the audio track.
+// TODO(tommi): Rename to MediaStreamLocalAudioTrack.
 class CONTENT_EXPORT WebRtcLocalAudioTrack
     : NON_EXPORTED_BASE(public MediaStreamAudioTrack) {
  public:
-  explicit WebRtcLocalAudioTrack(
-      scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter);
+  WebRtcLocalAudioTrack(WebRtcLocalAudioTrackAdapter* adapter,
+                        const scoped_refptr<WebRtcAudioCapturer>& capturer,
+                        WebAudioCapturerSource* webaudio_source);
 
   ~WebRtcLocalAudioTrack() override;
 
   // Add a sink to the track. This function will trigger a OnSetFormat()
   // call on the |sink|.
   // Called on the main render thread.
   void AddSink(MediaStreamAudioSink* sink) override;
 
   // Remove a sink from the track.
   // Called on the main render thread.
   void RemoveSink(MediaStreamAudioSink* sink) override;
 
+  // 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;
+
+  // Returns the output format of the capture source. May return an invalid
+  // AudioParameters if the format is not yet available.
+  // Called on the main render thread.
   media::AudioParameters GetOutputFormat() const override;
 
   // Method called by the capturer to deliver the capture data.
   // Called on the capture audio thread.
   void Capture(const media::AudioBus& audio_bus,
-               base::TimeTicks estimated_capture_time);
+               base::TimeTicks estimated_capture_time,
+               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);
 
-  // Called by the capturer before the audio data flow begins to set the object
-  // that provides shared access to the current audio signal level.
-  void SetLevel(scoped_refptr<MediaStreamAudioLevelCalculator::Level> level);
-
-  // Called by the capturer before the audio data flow begins to provide a
-  // reference to the audio processor so that the track can query stats from it.
-  void SetAudioProcessor(scoped_refptr<MediaStreamAudioProcessor> processor);
+  // 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;
 
-  // MediaStreamAudioTrack override.
-  void OnStop() final;
-
   // All usage of libjingle is through this adapter. The adapter holds
   // 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_;
 
   // 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_