Adding REMB to receive stream configuration, the send side will always
react to incoming REMB for now.

Adding a test to verify the receive side is generating RTCP REMB and
will follow up with a send side test as soon as the bitrate stats are
wired up for the new API.

TEST=See above.
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5286 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/video_receive_stream.h b/video_receive_stream.h
index e5a6829..e548728 100644
--- a/video_receive_stream.h
+++ b/video_receive_stream.h
@@ -108,7 +108,8 @@
       Rtp()
           : remote_ssrc(0),
             local_ssrc(0),
-            rtcp_mode(newapi::kRtcpReducedSize) {}
+            rtcp_mode(newapi::kRtcpReducedSize),
+            remb(false) {}
 
       // Synchronization source (stream identifier) to be received.
       uint32_t remote_ssrc;
@@ -118,6 +119,9 @@
       // See RtcpMode for description.
       newapi::RtcpMode rtcp_mode;
 
+      // See draft-alvestrand-rmcat-remb for information.
+      bool remb;
+
       // See NackConfig for description.
       NackConfig nack;