commit 4b50db1f8ce21cc7faf4f7c80b6a492dffb2a1d5
author pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Tue Dec 03 10:13:04 2013 +0000
committer pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Tue Dec 03 10:13:04 2013 +0000
tree 210078ccbc0623366abc6d5ef6aa0d933b6a8c31
parent cc19dce72f11c225416b88392ef017d760700160

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

call.cc

diff --git a/call.cc b/call.cc
index 9e23b24..5736947 100644
--- a/call.cc
+++ b/call.cc
@@ -260,8 +260,8 @@
       video_engine_, config, config_.send_transport, config_.voice_engine);
 
   WriteLockScoped write_lock(*receive_lock_);
-  assert(receive_ssrcs_.find(config.rtp.ssrc) == receive_ssrcs_.end());
-  receive_ssrcs_[config.rtp.ssrc] = receive_stream;
+  assert(receive_ssrcs_.find(config.rtp.remote_ssrc) == receive_ssrcs_.end());
+  receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
   return receive_stream;
 }