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Added initial fec configuration for rtp module.

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Review URL: https://webrtc-codereview.appspot.com/833004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2808 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
asapersson@webrtc.org 2012-09-24 11:33:49 +00:00
parent 69d46b4821
commit ce42ace6ed
1 changed files with 9 additions and 0 deletions
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9
src/video_engine/vie_channel.cc
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@ -232,6 +232,11 @@ WebRtc_Word32 ViEChannel::SetSendCodec(const VideoCodec& video_codec,
NACKMethod nack_method = rtp_rtcp_->NACK();
bool transmission_smoothening = rtp_rtcp_->TransmissionSmoothingStatus();
bool fec_enabled = false;
WebRtc_UWord8 payload_type_red;
WebRtc_UWord8 payload_type_fec;
rtp_rtcp_->GenericFECStatus(fec_enabled, payload_type_red, payload_type_fec);
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
if (video_codec.numberOfSimulcastStreams > 0) {
@ -261,6 +266,10 @@ WebRtc_Word32 ViEChannel::SetSendCodec(const VideoCodec& video_codec,
rtp_rtcp->SetStorePacketsStatus(true, kNackHistorySize);
rtp_rtcp->SetNACKStatus(nack_method);
}
if (fec_enabled) {
rtp_rtcp->SetGenericFECStatus(fec_enabled, payload_type_red,
payload_type_fec);
}
rtp_rtcp->SetSendingMediaStatus(rtp_rtcp_->SendingMedia());
simulcast_rtp_rtcp_.push_back(rtp_rtcp);
}