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;