Move responsibility for RTP header extensions on video receive.

media/engine/fakewebrtccall.cc

Index: media/engine/fakewebrtccall.cc
diff --git a/media/engine/fakewebrtccall.cc b/media/engine/fakewebrtccall.cc
index 8556cc7db000e0f3909d0c9425d731007591dc53..b1414d85c6e3be5966f7f9345cc6906ae8fc3f4a 100644
--- a/media/engine/fakewebrtccall.cc
+++ b/media/engine/fakewebrtccall.cc
@@ -428,6 +428,9 @@ 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_++,