Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.

webrtc/api/rtpreceiver.h

Index: webrtc/api/rtpreceiver.h
diff --git a/webrtc/api/rtpreceiver.h b/webrtc/api/rtpreceiver.h
index 001264d7a761181f76380a6fc1b27e09569f9f60..c255400df50ff6e7c1794520bb562e6ddb7d88ed 100644
--- a/webrtc/api/rtpreceiver.h
+++ b/webrtc/api/rtpreceiver.h
@@ -10,19 +10,20 @@
 
 // This file contains classes that implement RtpReceiverInterface.
 // An RtpReceiver associates a MediaStreamTrackInterface with an underlying
-// transport (provided by AudioProviderInterface/VideoProviderInterface)
+// transport (provided by cricket::VoiceChannel/cricket::VideoChannel)
 
 #ifndef WEBRTC_API_RTPRECEIVER_H_
 #define WEBRTC_API_RTPRECEIVER_H_
 
 #include <string>
 
-#include "webrtc/api/mediastreamprovider.h"
+#include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/rtpreceiverinterface.h"
 #include "webrtc/api/remoteaudiosource.h"
 #include "webrtc/api/videotracksource.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/media/base/videobroadcaster.h"
+#include "webrtc/pc/channel.h"
 
 namespace webrtc {
 
@@ -39,7 +40,7 @@ class AudioRtpReceiver : public ObserverInterface,
   AudioRtpReceiver(MediaStreamInterface* stream,
                    const std::string& track_id,
                    uint32_t ssrc,
-                   AudioProviderInterface* provider);
+                   cricket::VoiceChannel* channel);
 
   virtual ~AudioRtpReceiver();
 
@@ -58,6 +59,10 @@ class AudioRtpReceiver : public ObserverInterface,
     return track_.get();
   }
 
+  cricket::MediaType media_type() const override {
+    return cricket::MEDIA_TYPE_AUDIO;
+  }
+
   std::string id() const override { return id_; }
 
   RtpParameters GetParameters() const override;
@@ -66,14 +71,20 @@ class AudioRtpReceiver : public ObserverInterface,
   // RtpReceiverInternal implementation.
   void Stop() override;
 
+  // Does not take ownership.
+  // Should call SetChannel(nullptr) before |channel| is destroyed.
+  void SetChannel(cricket::VoiceChannel* channel) { channel_ = channel; }
+
  private:
   void Reconfigure();
 
   const std::string id_;
   const uint32_t ssrc_;
-  AudioProviderInterface* provider_;  // Set to null in Stop().
+  cricket::VoiceChannel* channel_;
   const rtc::scoped_refptr<AudioTrackInterface> track_;
   bool cached_track_enabled_;
+  double cached_volume_ = 1;
+  bool stopped_ = false;
 };
 
 class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInternal> {
@@ -82,7 +93,7 @@ class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInternal> {
                    const std::string& track_id,
                    rtc::Thread* worker_thread,
                    uint32_t ssrc,
-                   VideoProviderInterface* provider);
+                   cricket::VideoChannel* channel);
 
   virtual ~VideoRtpReceiver();
 
@@ -95,6 +106,10 @@ class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInternal> {
     return track_.get();
   }
 
+  cricket::MediaType media_type() const override {
+    return cricket::MEDIA_TYPE_VIDEO;
+  }
+
   std::string id() const override { return id_; }
 
   RtpParameters GetParameters() const override;
@@ -103,10 +118,14 @@ class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInternal> {
   // RtpReceiverInternal implementation.
   void Stop() override;
 
+  // Does not take ownership.
+  // Should call SetChannel(nullptr) before |channel| is destroyed.
+  void SetChannel(cricket::VideoChannel* channel) { channel_ = channel; }
+
  private:
   std::string id_;
   uint32_t ssrc_;
-  VideoProviderInterface* provider_;
+  cricket::VideoChannel* channel_;
   // |broadcaster_| is needed since the decoder can only handle one sink.
   // It might be better if the decoder can handle multiple sinks and consider
   // the VideoSinkWants.
@@ -115,6 +134,7 @@ class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInternal> {
   // the VideoRtpReceiver is stopped.
   rtc::scoped_refptr<VideoTrackSource> source_;
   rtc::scoped_refptr<VideoTrackInterface> track_;
+  bool stopped_ = false;
 };
 
 }  // namespace webrtc