Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(98)

Unified Diff: webrtc/video/video_send_stream.cc

Issue 1932683002: Remove ViEEncoder::SetNetworkStatus (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@move_pacer
Patch Set: Created 4 years, 8 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
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 {

Powered by Google App Engine
This is Rietveld 408576698