Exposing RtpSenders and RtpReceivers from PeerConnection.

talk/app/webrtc/peerconnection.h

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 22 matching lines @@
 #include "talk/app/webrtc/dtlsidentitystore.h"
 #include "talk/app/webrtc/mediastreamsignaling.h"
 #include "talk/app/webrtc/peerconnectionfactory.h"
 #include "talk/app/webrtc/peerconnectioninterface.h"
 #include "talk/app/webrtc/statscollector.h"
 #include "talk/app/webrtc/streamcollection.h"
 #include "talk/app/webrtc/webrtcsession.h"
 #include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
-class MediaStreamHandlerContainer;
 
 typedef std::vector<PortAllocatorFactoryInterface::StunConfiguration>
     StunConfigurations;
 typedef std::vector<PortAllocatorFactoryInterface::TurnConfiguration>
     TurnConfigurations;
 
 // PeerConnectionImpl implements the PeerConnection interface.
 // It uses MediaStreamSignaling and WebRtcSession to implement
 // the PeerConnection functionality.
 class PeerConnection : public PeerConnectionInterface,
@@ skipping 11 matching lines @@
       rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
       PeerConnectionObserver* observer);
   virtual rtc::scoped_refptr<StreamCollectionInterface> local_streams();
   virtual rtc::scoped_refptr<StreamCollectionInterface> remote_streams();
   virtual bool AddStream(MediaStreamInterface* local_stream);
   virtual void RemoveStream(MediaStreamInterface* local_stream);
 
   virtual rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
       AudioTrackInterface* track);
 
+  PeerConnectionInterface::RtpSenders GetSenders() override {
+    return rtp_senders_;
+  }
+
+  PeerConnectionInterface::RtpReceivers GetReceivers() override {
+    return rtp_receivers_;
+  }
+
   virtual rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
       const std::string& label,
       const DataChannelInit* config);
   virtual bool GetStats(StatsObserver* observer,
                         webrtc::MediaStreamTrackInterface* track,
                         StatsOutputLevel level);
 
   virtual SignalingState signaling_state();
 
   // TODO(bemasc): Remove ice_state() when callers are removed.
@@ skipping 32 matching lines @@
  private:
   // Implements MessageHandler.
   virtual void OnMessage(rtc::Message* msg);
 
   // Implements MediaStreamSignalingObserver.
   void OnAddRemoteStream(MediaStreamInterface* stream) override;
   void OnRemoveRemoteStream(MediaStreamInterface* stream) override;
   void OnAddDataChannel(DataChannelInterface* data_channel) override;
   void OnAddRemoteAudioTrack(MediaStreamInterface* stream,
                              AudioTrackInterface* audio_track,
-                             uint32 ssrc) override;
+                             uint32 ssrc,
+                             const std::string& mid) override;
   void OnAddRemoteVideoTrack(MediaStreamInterface* stream,
                              VideoTrackInterface* video_track,
-                             uint32 ssrc) override;
+                             uint32 ssrc,
+                             const std::string& mid) override;
   void OnRemoveRemoteAudioTrack(MediaStreamInterface* stream,
                                 AudioTrackInterface* audio_track) override;
   void OnRemoveRemoteVideoTrack(MediaStreamInterface* stream,
                                 VideoTrackInterface* video_track) override;
   void OnAddLocalAudioTrack(MediaStreamInterface* stream,
                             AudioTrackInterface* audio_track,
-                            uint32 ssrc) override;
+                            uint32 ssrc,
+                            const std::string& mid) override;
   void OnAddLocalVideoTrack(MediaStreamInterface* stream,
                             VideoTrackInterface* video_track,
-                            uint32 ssrc) override;
+                            uint32 ssrc,
+                            const std::string& mid) override;
   void OnRemoveLocalAudioTrack(MediaStreamInterface* stream,
                                AudioTrackInterface* audio_track,
                                uint32 ssrc) override;
   void OnRemoveLocalVideoTrack(MediaStreamInterface* stream,
                                VideoTrackInterface* video_track) override;
   void OnRemoveLocalStream(MediaStreamInterface* stream) override;
 
   // Implements IceObserver
   void OnIceConnectionChange(IceConnectionState new_state) override;
   void OnIceGatheringChange(IceGatheringState new_state) override;
@@ skipping 10 matching lines @@
     return factory_->signaling_thread();
   }
 
   void PostSetSessionDescriptionFailure(SetSessionDescriptionObserver* observer,
                                         const std::string& error);
 
   bool IsClosed() const {
     return signaling_state_ == PeerConnectionInterface::kClosed;
   }
 
+  PeerConnectionInterface::RtpSenders::iterator FindRtpSenderForTrack(
+      MediaStreamTrackInterface* track);
+  PeerConnectionInterface::RtpReceivers::iterator FindRtpReceiverForTrack(
+      MediaStreamTrackInterface* track);
+
   // Storing the factory as a scoped reference pointer ensures that the memory
   // in the PeerConnectionFactoryImpl remains available as long as the
   // PeerConnection is running. It is passed to PeerConnection as a raw pointer.
   // However, since the reference counting is done in the
   // PeerConnectionFactoryInteface all instances created using the raw pointer
   // will refer to the same reference count.
   rtc::scoped_refptr<PeerConnectionFactory> factory_;
   PeerConnectionObserver* observer_;
   UMAObserver* uma_observer_;
   SignalingState signaling_state_;
   // TODO(bemasc): Remove ice_state_.
   IceState ice_state_;
   IceConnectionState ice_connection_state_;
   IceGatheringState ice_gathering_state_;
 
   rtc::scoped_ptr<cricket::PortAllocator> port_allocator_;
   rtc::scoped_ptr<WebRtcSession> session_;
   rtc::scoped_ptr<MediaStreamSignaling> mediastream_signaling_;
-  rtc::scoped_ptr<MediaStreamHandlerContainer> stream_handler_container_;
   rtc::scoped_ptr<StatsCollector> stats_;
+
+  PeerConnectionInterface::RtpSenders rtp_senders_;
+  PeerConnectionInterface::RtpReceivers rtp_receivers_;
 };
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_PEERCONNECTION_H_