Chromium Code Reviews| Index: webrtc/video/video_send_stream.cc |
| diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc |
| index ad3fbadd14f73bfac3e637866516ee3852646123..a6e97b7ab3bd9c4d64c558409da4ed786190c2a3 100644 |
| --- a/webrtc/video/video_send_stream.cc |
| +++ b/webrtc/video/video_send_stream.cc |
| @@ -719,18 +719,9 @@ std::map<uint32_t, RtpState> VideoSendStream::GetRtpStates() const { |
| } |
| void VideoSendStream::SignalNetworkState(NetworkState state) { |
| - // When network goes up, enable RTCP status before setting transmission state. |
| - // When it goes down, disable RTCP afterwards. This ensures that any packets |
| - // sent due to the network state changed will not be dropped. |
| - if (state == kNetworkUp) { |
| - for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
| - rtp_rtcp->SetRTCPStatus(config_.rtp.rtcp_mode); |
| - } |
| - vie_encoder_.SetNetworkTransmissionState(state == kNetworkUp); |
| - if (state == kNetworkDown) { |
| - for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
| - rtp_rtcp->SetRTCPStatus(RtcpMode::kOff); |
| - } |
| + for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
|
stefan-webrtc
2016/05/02 10:56:31
{}
|
| + rtp_rtcp->SetRTCPStatus(state == kNetworkUp ? config_.rtp.rtcp_mode |
| + : RtcpMode::kOff); |
| } |
| int VideoSendStream::GetPaddingNeededBps() const { |