generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

Update SSRC in RtpRtcp for audio channel so that it can have NTP values for further AV sync.

Browse Source 
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
This commit is contained in:
dwkang@webrtc.org 2013-08-29 07:34:12 +00:00
parent d7301775f5
commit b295a3f592
1 changed files with 3 additions and 0 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

3
webrtc/voice_engine/channel.cc
View File

@ -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);