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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: Rebased and fixed CongestionController unittests. Created 4 years, 7 months ago
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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();
}
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