Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.

webrtc/api/remoteaudiosource.h

 /*
  *  Copyright 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #ifndef WEBRTC_API_REMOTEAUDIOSOURCE_H_
 #define WEBRTC_API_REMOTEAUDIOSOURCE_H_
 
 #include <list>
 #include <string>
 
-#include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/notifier.h"
 #include "webrtc/audio_sink.h"
 #include "webrtc/base/criticalsection.h"
+#include "webrtc/pc/channel.h"
 
 namespace rtc {
 struct Message;
 class Thread;
 }  // namespace rtc
 
 namespace webrtc {
 
-class AudioProviderInterface;
-
 // This class implements the audio source used by the remote audio track.
 class RemoteAudioSource : public Notifier<AudioSourceInterface> {

[pthatcher1, 2016/06/21 07:45:41]

It sounds like this should be in pc/, not api/.

[Taylor Brandstetter, 2016/06/22 00:50:17]

Yep. Same with most things currently in /api/.

  public:
   // Creates an instance of RemoteAudioSource.
   static rtc::scoped_refptr<RemoteAudioSource> Create(
       uint32_t ssrc,
-      AudioProviderInterface* provider);
+      cricket::VoiceChannel* channel);

[pthatcher1, 2016/06/21 07:45:41]

Could this take an RtpReceiver instead?

[Taylor Brandstetter, 2016/06/22 00:50:17]

I agree some refactoring is in order here, but I don't know if that's the best
solution. We would then have a circular dependency between RemoteAudioSource and
AudioRtpSender. I think it would be preferable if RemoteAudioSource just had a
method that returned a Sink object and RtpReceiverInternal could hook up the
wires.

But again, this is orthogonal to this CL.

 
   // MediaSourceInterface implementation.
   MediaSourceInterface::SourceState state() const override;
   bool remote() const override;
 
   void AddSink(AudioTrackSinkInterface* sink) override;
   void RemoveSink(AudioTrackSinkInterface* sink) override;
 
  protected:
   RemoteAudioSource();
   ~RemoteAudioSource() override;
 
   // Post construction initialize where we can do things like save a reference
   // to ourselves (need to be fully constructed).
-  void Initialize(uint32_t ssrc, AudioProviderInterface* provider);
+  void Initialize(uint32_t ssrc, cricket::VoiceChannel* channel);
 
  private:
   typedef std::list<AudioObserver*> AudioObserverList;
 
   // AudioSourceInterface implementation.
   void SetVolume(double volume) override;
   void RegisterAudioObserver(AudioObserver* observer) override;
   void UnregisterAudioObserver(AudioObserver* observer) override;
 
   class Sink;
   void OnData(const AudioSinkInterface::Data& audio);
-  void OnAudioProviderGone();
+  void OnAudioChannelGone();
 
   class MessageHandler;
   void OnMessage(rtc::Message* msg);
 
   AudioObserverList audio_observers_;
   rtc::CriticalSection sink_lock_;
   std::list<AudioTrackSinkInterface*> sinks_;
   rtc::Thread* const main_thread_;
   SourceState state_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_REMOTEAUDIOSOURCE_H_