Move responsibility for RTP header extensions on video receive.

call/call.h

 /*
  *  Copyright (c) 2013 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.
  */
 #ifndef CALL_CALL_H_
@@ skipping 113 matching lines @@
     int64_t rtt_ms = -1;
   };
 
   static Call* Create(const Call::Config& config);
 
   // Allows mocking |transport_send| for testing.
   static Call* Create(
       const Call::Config& config,
       std::unique_ptr<RtpTransportControllerSendInterface> transport_send);
 
+  // TODO(nisse): Should move to RtpTransportController.
+  // Rtp header extensions can be renegotiated mid-call.
+  virtual void SetVideoReceiveRtpHeaderExtensions(
+      const std::vector<RtpExtension>& extensions) = 0;
+
   virtual AudioSendStream* CreateAudioSendStream(
       const AudioSendStream::Config& config) = 0;
   virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0;
 
   virtual AudioReceiveStream* CreateAudioReceiveStream(
       const AudioReceiveStream::Config& config) = 0;
   virtual void DestroyAudioReceiveStream(
       AudioReceiveStream* receive_stream) = 0;
 
   virtual VideoSendStream* CreateVideoSendStream(
@@ skipping 54 matching lines @@
       const rtc::NetworkRoute& network_route) = 0;
 
   virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
 
   virtual ~Call() {}
 };
 
 }  // namespace webrtc
 
 #endif  // CALL_CALL_H_