Move responsibility for RTP header extensions on video receive.

webrtc/media/base/mediachannel.h

 /*
  *  Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 881 matching lines @@
   virtual std::string ToString() const {
     std::ostringstream ost;
     ost << "{";
     ost << "codecs: " << VectorToString(codecs) << ", ";
     ost << "extensions: " << VectorToString(extensions);
     ost << "}";
     return ost.str();
   }
 
   std::vector<Codec> codecs;
+  // TODO(nisse): Delete, RTP extensions are really per-transport, not
+  // per stream. Now unused for video receive streams.

[pthatcher1, 2017/04/29 00:25:18]

These are not per-stream, these are per-m-line.  With Unified Plan SDP, that
will mean per-stream.  Currently they are not per-stream.

   std::vector<webrtc::RtpExtension> extensions;
   // TODO(pthatcher): Add streams.
   RtcpParameters rtcp;
   virtual ~RtpParameters() = default;
 };
 
 // TODO(deadbeef): Rename to RtpSenderParameters, since they're intended to
 // encapsulate all the parameters needed for an RtpSender.
 template <class Codec>
 struct RtpSendParameters : RtpParameters<Codec> {
@@ skipping 293 matching lines @@
                    const char*,
                    size_t> SignalDataReceived;
   // Signal when the media channel is ready to send the stream. Arguments are:
   //     writable(bool)
   sigslot::signal1<bool> SignalReadyToSend;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_