Move responsibility for RTP header extensions on video receive.

webrtc/media/engine/fakewebrtccall.cc

 /*
  *  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 403 matching lines @@
       ADD_FAILURE() << "GetNetworkState called with unknown parameter.";
       return webrtc::kNetworkDown;
   }
   // Even though all the values for the enum class are listed above,the compiler
   // will emit a warning as the method may be called with a value outside of the
   // valid enum range, unless this case is also handled.
   ADD_FAILURE() << "GetNetworkState called with unknown parameter.";
   return webrtc::kNetworkDown;
 }
 
+void FakeCall::SetVideoReceiveRtpHeaderExtensions(
+    const std::vector<webrtc::RtpExtension>& extensions) {}
+
 webrtc::AudioSendStream* FakeCall::CreateAudioSendStream(
     const webrtc::AudioSendStream::Config& config) {
   FakeAudioSendStream* fake_stream = new FakeAudioSendStream(next_stream_id_++,
                                                              config);
   audio_send_streams_.push_back(fake_stream);
   ++num_created_send_streams_;
   return fake_stream;
 }
 
 void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
@@ skipping 182 matching lines @@
 }
 
 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
   last_sent_packet_ = sent_packet;
   if (sent_packet.packet_id >= 0) {
     last_sent_nonnegative_packet_id_ = sent_packet.packet_id;
   }
 }
 
 }  // namespace cricket