| Index: webrtc/video/video_send_stream.cc
|
| diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc
|
| index e87f56f383edbb3bddb2c4bb4e5a3860fdeff2a5..78193b0b60a66b1b3b3285924dc82a382fdd5d20 100644
|
| --- a/webrtc/video/video_send_stream.cc
|
| +++ b/webrtc/video/video_send_stream.cc
|
| @@ -486,21 +486,6 @@ 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);
|
| @@ -742,6 +727,13 @@ 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();
|
| }
|
|
|