Move responsibility for RTP header extensions on video receive.

webrtc/media/engine/fakewebrtccall.h

 /*
  *  Copyright (c) 2015 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 238 matching lines @@
   int last_sent_nonnegative_packet_id() const {
     return last_sent_nonnegative_packet_id_;
   }
 
   webrtc::NetworkState GetNetworkState(webrtc::MediaType media) const;
   int GetNumCreatedSendStreams() const;
   int GetNumCreatedReceiveStreams() const;
   void SetStats(const webrtc::Call::Stats& stats);
 
  private:
+  void SetVideoReceiveRtpHeaderExtensions(
+      const std::vector<webrtc::RtpExtension>& extensions) override;
   webrtc::AudioSendStream* CreateAudioSendStream(
       const webrtc::AudioSendStream::Config& config) override;
   void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override;
 
   webrtc::AudioReceiveStream* CreateAudioReceiveStream(
       const webrtc::AudioReceiveStream::Config& config) override;
   void DestroyAudioReceiveStream(
       webrtc::AudioReceiveStream* receive_stream) override;
 
   webrtc::VideoSendStream* CreateVideoSendStream(
@@ skipping 45 matching lines @@
 
   int num_created_send_streams_;
   int num_created_receive_streams_;
 
   int audio_transport_overhead_;
   int video_transport_overhead_;
 };
 
 }  // namespace cricket
 #endif  // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCCALL_H_