Remove ViEEncoder::SetNetworkStatus

webrtc/video/video_send_stream.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 473 matching lines @@
   rtp_rtcp_modules_[0]->SetREMBStatus(false);
   remb_->RemoveRembSender(rtp_rtcp_modules_[0]);
 
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
     congestion_controller_->packet_router()->RemoveRtpModule(rtp_rtcp);
     module_process_thread_->DeRegisterModule(rtp_rtcp);
     delete rtp_rtcp;
   }
 }
 
-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);
   return true;
 }
 
 void VideoSendStream::Start() {
   if (payload_router_.active())
     return;
   TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Start");
@@ skipping 253 matching lines @@
   }
 
   for (size_t i = 0; i < config_.rtp.rtx.ssrcs.size(); ++i) {
     uint32_t ssrc = config_.rtp.rtx.ssrcs[i];
     rtp_states[ssrc] = rtp_rtcp_modules_[i]->GetRtxState();
   }
 
   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();
 }
 
 void VideoSendStream::OnBitrateUpdated(uint32_t bitrate_bps,
                                        uint8_t fraction_loss,
                                        int64_t rtt) {
   payload_router_.SetTargetSendBitrate(bitrate_bps);
   vie_encoder_.OnBitrateUpdated(bitrate_bps, fraction_loss, rtt);
 }
@@ skipping 16 matching lines @@
                           &module_nack_rate);
     *sent_video_rate_bps += module_video_rate;
     *sent_nack_rate_bps += module_nack_rate;
     *sent_fec_rate_bps += module_fec_rate;
   }
   return 0;
 }
 
 }  // namespace internal
 }  // namespace webrtc