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_level_calculator.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 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.

[o1ka, 2016/02/29 14:28:05]

Can we indeed make MediaStreamLocalAudioTrack and WebRtcLocalAudioTrack to have
some matching names?

[miu, 2016/03/01 09:43:55]

I'll be addressing this is my next CL.  (BTW, for reference, see:
https://codereview.chromium.org/1647773002/ for the approximate "end state" I
want to achieve with this series of CLs.).

tl;dr: I'll be moving WebRtcLocalAudioTrack into the webrtc/ subdir, and moving
its non-WebRTC-specific functionality into MediaStreamAudioTrack (parent class).

[o1ka, 2016/03/01 14:18:59]

Acknowledged.

+// 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
     : NON_EXPORTED_BASE(public MediaStreamAudioTrack) {
  public:
-  WebRtcLocalAudioTrack(WebRtcLocalAudioTrackAdapter* adapter,
-                        const scoped_refptr<WebRtcAudioCapturer>& capturer,
-                        WebAudioCapturerSource* webaudio_source);
+  explicit WebRtcLocalAudioTrack(
+      scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter);
 
   ~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,
-               bool force_report_nonzero_energy);
+               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);
 
-  // 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.
+  // 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) {
+    adapter_->SetLevel(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(
-      const scoped_refptr<MediaStreamAudioProcessor>& processor);
+      const scoped_refptr<MediaStreamAudioProcessor>& processor) {
+    adapter_->SetAudioProcessor(processor);
+  }
 
  private:
   typedef TaggedList<MediaStreamAudioTrackSink> SinkList;
 
+  // Called when this MediaStreamAudioTrack is stopped to notify all sinks the
+  // stream has ended, and to remove all references to the sinks.
+  void RemoveAllSinks();
+
   // 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_;

[o1ka, 2016/02/29 14:28:05]

I know it's not your change and probably I am paranoid but I would not use one
lock to protect two unrelated entities, even if the scope of locking is small:
you never know what you end up with in a year or two.

[miu, 2016/03/01 09:43:55]

Totally agree.  But, this is yet another thing that will be fixed w/ the next
CL; so I'd prefer not to do any redundant work here.  :)

[o1ka, 2016/03/01 14:18:59]

Acknowledged.

 
   // 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_