Move responsibility for RTP header extensions on video receive.

webrtc/call/call.h

Index: webrtc/call/call.h
diff --git a/webrtc/call/call.h b/webrtc/call/call.h
index ec73bf7714926b2a6b04210f3c66a881f9acbedf..8a57eb43aa92912f49d2f39de2d4f9777a8538d8 100644
--- a/webrtc/call/call.h
+++ b/webrtc/call/call.h
@@ -100,6 +100,11 @@ class Call {
 
   static Call* Create(const Call::Config& config);
 
+  // TODO(nisse): Should move to RtpTransportController.
+  // Rtp header extensions can be renegotiated mid-call.
+  virtual void SetVideoReceiveRtpHeaderExtensions(
+      const std::vector<RtpExtension>& extensions) = 0;

[pthatcher1, 2017/04/29 00:25:18]

It makes sense for this to be per-RtpTransport, but not per Call or
RtpTransportController.  The non-bundled case, the extensions can be different. 
It's only in bundled case (one RtpTransport) that you can assume the extensions
will be the same across the bundled m-lines.  In the non-bundled case, you can't
assume that.

+
   virtual AudioSendStream* CreateAudioSendStream(
       const AudioSendStream::Config& config) = 0;
   virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0;