Set local SSRC for VideoReceiveStream.
As a bonus, also removes GenerateRandomSsrc, which only worked on sender
configs. There's no point to generate random SSRCs in tests.
BUG=2691
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5201 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/video_receive_stream.h b/video_receive_stream.h
index 8b4f643..e5a6829 100644
--- a/video_receive_stream.h
+++ b/video_receive_stream.h
@@ -105,11 +105,15 @@
// Receive-stream specific RTP settings.
struct Rtp {
- Rtp() : ssrc(0), rtcp_mode(newapi::kRtcpReducedSize) {}
+ Rtp()
+ : remote_ssrc(0),
+ local_ssrc(0),
+ rtcp_mode(newapi::kRtcpReducedSize) {}
- // TODO(mflodman) Do we require a set ssrc? What happens if the ssrc
- // changes?
- uint32_t ssrc;
+ // Synchronization source (stream identifier) to be received.
+ uint32_t remote_ssrc;
+ // Sender SSRC used for sending RTCP (such as receiver reports).
+ uint32_t local_ssrc;
// See RtcpMode for description.
newapi::RtcpMode rtcp_mode;