Index: webrtc/video/video_send_stream.cc |
diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc |
index 840991b4baceadf16fe026bd02915d816aa84da8..f673a98a3e1797ffc297bc904d9bea47ccd12dd4 100644 |
--- a/webrtc/video/video_send_stream.cc |
+++ b/webrtc/video/video_send_stream.cc |
@@ -491,21 +491,6 @@ VideoSendStream::~VideoSendStream() { |
} |
} |
-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); |
- } |
-} |
- |
bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) { |
for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
rtp_rtcp->IncomingRtcpPacket(packet, length); |
@@ -779,6 +764,13 @@ std::map<uint32_t, RtpState> VideoSendStream::GetRtpStates() const { |
return rtp_states; |
} |
+void VideoSendStream::SignalNetworkState(NetworkState state) { |
+ for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
+ rtp_rtcp->SetRTCPStatus(state == kNetworkUp ? config_.rtp.rtcp_mode |
+ : RtcpMode::kOff); |
+ } |
+} |
+ |
int VideoSendStream::GetPaddingNeededBps() const { |
return vie_encoder_.GetPaddingNeededBps(); |
} |