MediaStream audio: Refactor 3 separate "glue" implementations into one.

content/renderer/media/webrtc/processed_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>
+#ifndef CONTENT_RENDERER_MEDIA_WEBRTC_PROCESSED_LOCAL_AUDIO_TRACK_H_
+#define CONTENT_RENDERER_MEDIA_WEBRTC_PROCESSED_LOCAL_AUDIO_TRACK_H_
 
 #include "base/macros.h"
 #include "base/memory/ref_counted.h"
-#include "base/synchronization/lock.h"
-#include "base/threading/thread_checker.h"
+#include "content/renderer/media/media_stream_audio_level_calculator.h"
+#include "content/renderer/media/media_stream_audio_processor.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 WebRtcLocalAudioTrackAdapter;
 
-// A WebRtcLocalAudioTrack manages thread-safe connects/disconnects to sinks,
-// and the delivery of audio data from the source to the sinks.
-class CONTENT_EXPORT WebRtcLocalAudioTrack
+// A MediaStreamAudioTrack that also holds a WebRtcLocalAudioTrackAdapter and
+// provides an "adapter" bridge between the media stream object graph in the
+// content namespace and the separate object graph in the webrtc namespace.
+//
+// This class is instantiated by ProcessLocalAudioSource.
+class CONTENT_EXPORT ProcessedLocalAudioTrack final
     : NON_EXPORTED_BASE(public MediaStreamAudioTrack) {
  public:
-  explicit WebRtcLocalAudioTrack(
+  explicit ProcessedLocalAudioTrack(
       scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter);
 
-  ~WebRtcLocalAudioTrack() override;
+  ~ProcessedLocalAudioTrack() final;
 
-  // 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;
+  // If |track| is an instance of ProcessedLocalAudioTrack, return a type-casted
+  // pointer to it.  Otherwise, return null.
+  static ProcessedLocalAudioTrack* From(MediaStreamAudioTrack* track);
 
-  // Remove a sink from the track.
-  // Called on the main render thread.
-  void RemoveSink(MediaStreamAudioSink* sink) override;
+  const scoped_refptr<WebRtcLocalAudioTrackAdapter>& adapter() const {

[perkj_chrome, 2016/04/08 14:05:42]

Why isn't WebRtcLocalAudioTrackAdapter just a sink to a MediaStreamAudioTrack
that is created when needed in rtc_peerconnection_handler? Too big of a change? 

[miu, 2016/04/19 00:40:22]

It is now!  :)  As of Patch Set 4, WebRtcAudioSink was introduced to get rid of
this weirdness.  In fact, I was also able to delete ProcessedLocalAudioTrack as
well (it's no longer needed).

+    return adapter_;
+  }
 
-  // Overrides for MediaStreamTrack.
-  void SetEnabled(bool enabled) override;
-  webrtc::AudioTrackInterface* GetAudioAdapter() override;
-  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);
-
-  // 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);
+  // Override for MediaStreamTrack.
+  void SetEnabled(bool enabled) final;
 
   // 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);
 
  private:
-  typedef TaggedList<MediaStreamAudioTrackSink> SinkList;
-
   // MediaStreamAudioTrack override.
-  void OnStop() final;
+  void* GetClassIdentifier() const 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_;
 
-  // 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_;
+  // In debug builds, check that all methods that could cause object graph
+  // or data flow changes are being called on the main thread.
+  base::ThreadChecker 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.
-  media::AudioParameters audio_parameters_;
-
-  DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack);
+  DISALLOW_COPY_AND_ASSIGN(ProcessedLocalAudioTrack);
 };
 
 }  // namespace content
 
-#endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_
+#endif  // CONTENT_RENDERER_MEDIA_WEBRTC_PROCESSED_LOCAL_AUDIO_TRACK_H_