Adding a MediaStream parameter to createSender.

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 85 matching lines @@
   rtc::scoped_refptr<StreamCollectionInterface> remote_streams() override;
   bool AddStream(MediaStreamInterface* local_stream) override;
   void RemoveStream(MediaStreamInterface* local_stream) override;
 
   virtual WebRtcSession* session() { return session_.get(); }
 
   rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
       AudioTrackInterface* track) override;
 
   rtc::scoped_refptr<RtpSenderInterface> CreateSender(
-      const std::string& kind) override;
+      const std::string& kind,
+      const std::string& stream_id) override;
 
   std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
       const override;
   std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
       const override;
 
   rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
       const std::string& label,
       const DataChannelInit* config) override;
   bool GetStats(StatsObserver* observer,
@@ skipping 282 matching lines @@
   // because its destruction fires signals (such as VoiceChannelDestroyed)
   // which will trigger some final actions in PeerConnection...
   rtc::scoped_ptr<WebRtcSession> session_;
   // ... But stats_ depends on session_ so it should be destroyed even earlier.
   rtc::scoped_ptr<StatsCollector> stats_;
 };
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_PEERCONNECTION_H_