Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.

webrtc/api/webrtcsession.h

Index: webrtc/api/webrtcsession.h
diff --git a/webrtc/api/webrtcsession.h b/webrtc/api/webrtcsession.h
index 76af6c7c3d2669830665c77731d3dfd0e661651d..d69abc0dedb67097ba6c56c0f1f019e950e53504 100644
--- a/webrtc/api/webrtcsession.h
+++ b/webrtc/api/webrtcsession.h
@@ -19,7 +19,6 @@
 #include "webrtc/api/datachannel.h"
 #include "webrtc/api/dtmfsender.h"
 #include "webrtc/api/mediacontroller.h"
-#include "webrtc/api/mediastreamprovider.h"
 #include "webrtc/api/peerconnectioninterface.h"
 #include "webrtc/api/statstypes.h"
 #include "webrtc/base/constructormagic.h"
@@ -115,11 +114,11 @@ struct SessionStats {
 // participates in the network-level negotiation.  The individual streams of
 // packets are represented by TransportChannels.  The application-level protocol
 // is represented by SessionDecription objects.
-class WebRtcSession : public AudioProviderInterface,
-                      public VideoProviderInterface,
-                      public DtmfProviderInterface,
-                      public DataChannelProviderInterface,
-                      public sigslot::has_slots<> {
+class WebRtcSession :
+
+    public DtmfProviderInterface,
+    public DataChannelProviderInterface,
+    public sigslot::has_slots<> {
  public:
   enum State {
     STATE_INIT = 0,
@@ -234,41 +233,6 @@ class WebRtcSession : public AudioProviderInterface,
   virtual bool GetLocalTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
   virtual bool GetRemoteTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
 
-  // AudioMediaProviderInterface implementation.
-  void SetAudioPlayout(uint32_t ssrc, bool enable) override;
-  void SetAudioSend(uint32_t ssrc,
-                    bool enable,
-                    const cricket::AudioOptions& options,
-                    cricket::AudioSource* source) override;
-  void SetAudioPlayoutVolume(uint32_t ssrc, double volume) override;
-  void SetRawAudioSink(uint32_t ssrc,
-                       std::unique_ptr<AudioSinkInterface> sink) override;
-
-  RtpParameters GetAudioRtpSendParameters(uint32_t ssrc) const override;
-  bool SetAudioRtpSendParameters(uint32_t ssrc,
-                                 const RtpParameters& parameters) override;
-  RtpParameters GetAudioRtpReceiveParameters(uint32_t ssrc) const override;
-  bool SetAudioRtpReceiveParameters(uint32_t ssrc,
-                                    const RtpParameters& parameters) override;
-
-  // Implements VideoMediaProviderInterface.
-  void SetVideoPlayout(
-      uint32_t ssrc,
-      bool enable,
-      rtc::VideoSinkInterface<cricket::VideoFrame>* sink) override;
-  void SetVideoSend(
-      uint32_t ssrc,
-      bool enable,
-      const cricket::VideoOptions* options,
-      rtc::VideoSourceInterface<cricket::VideoFrame>* source) override;
-
-  RtpParameters GetVideoRtpSendParameters(uint32_t ssrc) const override;
-  bool SetVideoRtpSendParameters(uint32_t ssrc,
-                                 const RtpParameters& parameters) override;
-  RtpParameters GetVideoRtpReceiveParameters(uint32_t ssrc) const override;
-  bool SetVideoRtpReceiveParameters(uint32_t ssrc,
-                                    const RtpParameters& parameters) override;
-
   // Implements DtmfProviderInterface.
   bool CanInsertDtmf(const std::string& track_id) override;
   bool InsertDtmf(const std::string& track_id,
@@ -310,8 +274,6 @@ class WebRtcSession : public AudioProviderInterface,
   void OnCertificateReady(
       const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
   void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp);
-  // Called when the channel received the first packet.
-  void OnChannelFirstPacketReceived(cricket::BaseChannel*);
 
   // For unit test.
   bool waiting_for_certificate_for_testing() const;