commit 2bb1bdab8d11f5445693c028335fb3ace631f636
author pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Mon Jul 07 13:06:48 2014
committer pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Mon Jul 07 13:06:48 2014
tree 2bb71f0cacfe8e8136f229004bc81e9fdb439dce
parent 73823cafa42a32dfa443df8a880e868bcdc6b211

Preserve RTP states for restarted VideoSendStreams.

A restarted VideoSendStream would previously be completely reset,
causing gaps in sequence numbers and potentially RTP timestamps as well.
This broke SRTP which requires fairly sequential sequence numbers.
Presumably, were this sent without SRTP, we'd still have problems on the
receiving end as the corresponding receiver is unaware of this reset.

Also adding annotation to RTPSender and addressing some unlocked
access to ssrc_, ssrc_rtx_ and rtx_.

BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6612 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/test/call_test.h

diff --git a/webrtc/test/call_test.h b/webrtc/test/call_test.h
index 37a883d..e24d835 100644
--- a/webrtc/test/call_test.h
+++ b/webrtc/test/call_test.h
@@ -35,7 +35,7 @@
   static const uint8_t kSendPayloadType;
   static const uint8_t kSendRtxPayloadType;
   static const uint8_t kFakeSendPayloadType;
-  static const uint32_t kSendRtxSsrc;
+  static const uint32_t kSendRtxSsrcs[kNumSsrcs];
   static const uint32_t kSendSsrcs[kNumSsrcs];
   static const uint32_t kReceiverLocalSsrc;
   static const int kNackRtpHistoryMs;
@@ -58,6 +58,8 @@
   void Stop();
   void DestroyStreams();
 
+  Clock* const clock_;
+
   scoped_ptr<Call> sender_call_;
   VideoSendStream::Config send_config_;
   std::vector<VideoStream> video_streams_;