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Correctly set SSRCs for extra send RTP modules.

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Fixes a regression introduced in r4096.

BUG=1845
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1585004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4134 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org 2013-05-29 14:07:54 +00:00
parent 7bdfff3503
commit eea2622350
1 changed files with 3 additions and 6 deletions
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9
webrtc/video_engine/vie_channel.cc
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@ -956,9 +956,6 @@ int32_t ViEChannel::SetSSRC(const uint32_t SSRC,
return rtp_rtcp_->SetSSRC(SSRC); return rtp_rtcp_->SetSSRC(SSRC);
} }
CriticalSectionScoped cs(rtp_rtcp_cs_.get()); CriticalSectionScoped cs(rtp_rtcp_cs_.get());
if (rtp_rtcp_->SetSSRC(SSRC) != 0) {
return -1;
}
if (simulcast_idx > simulcast_rtp_rtcp_.size()) { if (simulcast_idx > simulcast_rtp_rtcp_.size()) {
return -1; return -1;
} }
@ -968,11 +965,11 @@ int32_t ViEChannel::SetSSRC(const uint32_t SSRC,
return -1; return -1;
} }
} }
RtpRtcp* rtp_rtcp = *it; RtpRtcp* rtp_rtcp_module = *it;
if (usage == kViEStreamTypeRtx) { if (usage == kViEStreamTypeRtx) {
return rtp_rtcp->SetRTXSendStatus(kRtxRetransmitted, true, SSRC); return rtp_rtcp_module->SetRTXSendStatus(kRtxRetransmitted, true, SSRC);
} }
return 0; return rtp_rtcp_module->SetSSRC(SSRC);
} }
int32_t ViEChannel::SetRemoteSSRCType(const StreamType usage, int32_t ViEChannel::SetRemoteSSRCType(const StreamType usage,