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

content/renderer/media/webrtc/peer_connection_remote_audio_source.h

 // Copyright 2015 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_MEDIA_STREAM_REMOTE_AUDIO_TRACK_H_
-#define CONTENT_RENDERER_MEDIA_WEBRTC_MEDIA_STREAM_REMOTE_AUDIO_TRACK_H_
+#ifndef CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_REMOTE_AUDIO_SOURCE_H_
+#define CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_REMOTE_AUDIO_SOURCE_H_
 
 #include "base/memory/ref_counted.h"
-#include "base/threading/thread_checker.h"
+#include "content/renderer/media/media_stream_audio_source.h"
 #include "content/renderer/media/media_stream_audio_track.h"
-#include "third_party/WebKit/public/platform/WebMediaStreamSource.h"
+#include "third_party/webrtc/api/mediastreaminterface.h"
+
+namespace media {
+class AudioBus;
+}
 
 namespace content {
 
-class MediaStreamRemoteAudioSource;
+// PeerConnectionRemoteAudioTrack is a WebRTC specific implementation of an
+// audio track whose data is sourced from a PeerConnection.
+class PeerConnectionRemoteAudioTrack final
+    : NON_EXPORTED_BASE(public MediaStreamAudioTrack) {
+ public:
+  explicit PeerConnectionRemoteAudioTrack(
+      scoped_refptr<webrtc::AudioTrackInterface> track_interface);
+  ~PeerConnectionRemoteAudioTrack() final;
 
-// MediaStreamRemoteAudioTrack is a WebRTC specific implementation of an
-// audio track received from a PeerConnection.
-// TODO(tommi): Chrome shouldn't have to care about remote vs local so
-// we should have a single track implementation that delegates to the
-// sources that do different things depending on the type of source.
-class MediaStreamRemoteAudioTrack : public MediaStreamAudioTrack {
- public:
-  explicit MediaStreamRemoteAudioTrack(
-      const blink::WebMediaStreamSource& source, bool enabled);
-  ~MediaStreamRemoteAudioTrack() override;
+  // If |track| is an instance of PeerConnectionRemoteAudioTrack, return a
+  // type-casted pointer to it.  Otherwise, return null.
+  static PeerConnectionRemoteAudioTrack* From(MediaStreamAudioTrack* track);
 
-  // MediaStreamTrack override.
+  webrtc::AudioTrackInterface* track_interface() const {
+    return track_interface_.get();
+  }
+
+  // MediaStreamAudioTrack override.
   void SetEnabled(bool enabled) override;
 
+ private:
   // MediaStreamAudioTrack overrides.
-  void AddSink(MediaStreamAudioSink* sink) override;
-  void RemoveSink(MediaStreamAudioSink* sink) override;
-  media::AudioParameters GetOutputFormat() const override;
+  void* GetClassIdentifier() const final;
+  void OnStop() final;
 
-  webrtc::AudioTrackInterface* GetAudioAdapter() override;
+  const scoped_refptr<webrtc::AudioTrackInterface> track_interface_;
+
+  // 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_;
+};
+
+// Represents the audio provided by the receiving end of a PeerConnection.
+class PeerConnectionRemoteAudioSource final

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

Suggest name MediaStreamRemoteAudioSource to match with
MediaStreamRemoteVideoSource .

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

I renamed it from that because this source is PeerConnection/WebRTC-specific. 
There are other non-WebRTC remote audio sources in Chromium nowadays (e.g., the
Cast Streaming Receiver), so I wanted to make the distinction to avoid
confusion.  Sound good?

[perkj_chrome, 2016/04/20 13:34:53]

Acknowledged.

+    : NON_EXPORTED_BASE(public MediaStreamAudioSource),
+      NON_EXPORTED_BASE(protected webrtc::AudioTrackSinkInterface) {
+ public:
+  explicit PeerConnectionRemoteAudioSource(
+      scoped_refptr<webrtc::AudioTrackInterface> track_interface);
+  ~PeerConnectionRemoteAudioSource() final;
+
+ protected:
+  // MediaStreamAudioSource implementation.
+  scoped_ptr<MediaStreamAudioTrack> CreateMediaStreamAudioTrack(
+      const std::string& id) final;
+  bool EnsureSourceIsStarted() final;
+  void EnsureSourceIsStopped() final;
+
+  // webrtc::AudioTrackSinkInterface implementation.
+  void OnData(const void* audio_data, int bits_per_sample, int sample_rate,
+              size_t number_of_channels, size_t number_of_frames) final;
 
  private:
-  // MediaStreamAudioTrack override.
-  void OnStop() final;
+  // Interface to the implementation that calls OnData().
+  const scoped_refptr<webrtc::AudioTrackInterface> track_interface_;
 
-  MediaStreamRemoteAudioSource* source() const;
+  // 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_;
 
-  blink::WebMediaStreamSource source_;
-  bool enabled_;
-};
+  // True if |this| is receiving an audio flow as a sink of the remote
+  // PeerConnection via |track_interface_|.
+  bool is_sink_of_peer_connection_;
 
-// Inheriting from ExtraData directly since MediaStreamAudioSource has
-// too much unrelated bloat.
-// TODO(tommi): MediaStreamAudioSource needs refactoring.
-// TODO(miu): On it!  ;-)
-class MediaStreamRemoteAudioSource
-    : public blink::WebMediaStreamSource::ExtraData {
- public:
-  explicit MediaStreamRemoteAudioSource(
-      const scoped_refptr<webrtc::AudioTrackInterface>& track);
-  ~MediaStreamRemoteAudioSource() override;
-
-  // Controls whether or not the source is included in the main, mixed, audio
-  // output from WebRTC as rendered by WebRtcAudioRenderer (media players).
-  void SetEnabledForMixing(bool enabled);
-
-  // Adds an audio sink for a track belonging to this source.
-  // |enabled| is the enabled state of the track and can be updated via
-  // a call to SetSinksEnabled.
-  void AddSink(MediaStreamAudioSink* sink, MediaStreamAudioTrack* track,
-               bool enabled);
-
-  // Removes an audio sink for a track belonging to this source.
-  void RemoveSink(MediaStreamAudioSink* sink, MediaStreamAudioTrack* track);
-
-  // Turns audio callbacks on/off for all sinks belonging to a track.
-  void SetSinksEnabled(MediaStreamAudioTrack* track, bool enabled);
-
-  // Removes all sinks belonging to a track.
-  void RemoveAll(MediaStreamAudioTrack* track);
-
-  webrtc::AudioTrackInterface* GetAudioAdapter();
-
- private:
-  class AudioSink;
-  scoped_ptr<AudioSink> sink_;
-  const scoped_refptr<webrtc::AudioTrackInterface> track_;
-  base::ThreadChecker thread_checker_;
+  // Buffer for converting from interleaved signed-integer PCM samples to the
+  // planar float format.  Only used on the thread that calls OnData().
+  scoped_ptr<media::AudioBus> audio_bus_;
 };
 
 }  // namespace content
 
-#endif  // CONTENT_RENDERER_MEDIA_WEBRTC_MEDIA_STREAM_REMOTE_AUDIO_TRACK_H_
+#endif  // CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_REMOTE_AUDIO_SOURCE_H_