From b295a3f5920e2706d42c960c5180c7cc6e1f435e Mon Sep 17 00:00:00 2001 From: "dwkang@webrtc.org" Date: Thu, 29 Aug 2013 07:34:12 +0000 Subject: [PATCH] Update SSRC in RtpRtcp for audio channel so that it can have NTP values for further AV sync. Background: Since we had http://review.webrtc.org/2048004, the SSRC value in RtpRtcp for audio hasn't been updated. Because this prevents NTP update in RtpRtcp, the sync logic in ViESyncModule::Process() does not work. BUG=b/10484087 TEST= pass 'git try' except tests already broken in http://build.chromium.org/p/tryserver.webrtc/console R=henrika@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2131004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4638 4adac7df-926f-26a2-2b94-8c16560cd09d --- webrtc/voice_engine/channel.cc | 3 +++ 1 file changed, 3 insertions(+) diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc index 4671b6fd12..d857776a6e 100644 --- a/webrtc/voice_engine/channel.cc +++ b/webrtc/voice_engine/channel.cc @@ -369,6 +369,9 @@ Channel::OnIncomingSSRCChanged(int32_t id, uint32_t ssrc) int32_t channel = VoEChannelId(id); assert(channel == _channelId); + // Update ssrc so that NTP for AV sync can be updated. + _rtpRtcpModule->SetRemoteSSRC(ssrc); + if (_rtpObserver) { CriticalSectionScoped cs(&_callbackCritSect);