Exposing RtpSenders and RtpReceivers from PeerConnection.

talk/app/webrtc/mediastreamsignaling.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 269 matching lines @@
       return !msid_supported && (default_audio_track_needed ||
           default_video_track_needed);
     }
   };
 
   struct TrackInfo {
     TrackInfo() : ssrc(0) {}
     TrackInfo(const std::string& stream_label,
               const std::string track_id,
               uint32 ssrc)
-        : stream_label(stream_label),
-          track_id(track_id),
-          ssrc(ssrc) {
-    }
+        : stream_label(stream_label), track_id(track_id), ssrc(ssrc) {}
     std::string stream_label;
     std::string track_id;
     uint32 ssrc;
   };
   typedef std::vector<TrackInfo> TrackInfos;
 
   // Makes sure a MediaStream Track is created for each StreamParam in
   // |streams|. |media_type| is the type of the |streams| and can be either
   // audio or video.
   // If a new MediaStream is created it is added to |new_streams|.
@@ skipping 93 matching lines @@
   int last_allocated_sctp_even_sid_;
   int last_allocated_sctp_odd_sid_;
 
   RtpDataChannels rtp_data_channels_;
   SctpDataChannels sctp_data_channels_;
 };
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_MEDIASTREAMSIGNALING_H_