Move responsibility for RTP header extensions on video receive.

webrtc/media/engine/fakewebrtccall.cc

Index: webrtc/media/engine/fakewebrtccall.cc
diff --git a/webrtc/media/engine/fakewebrtccall.cc b/webrtc/media/engine/fakewebrtccall.cc
index b53966586f8302723eb3b00da13eab77c9610d86..f04c9a7750d3b24c59d8f869bfebf94d084cb1cc 100644
--- a/webrtc/media/engine/fakewebrtccall.cc
+++ b/webrtc/media/engine/fakewebrtccall.cc
@@ -421,6 +421,9 @@ webrtc::NetworkState FakeCall::GetNetworkState(webrtc::MediaType media) const {
   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_++,