Remove contention between RTCP packets and encoding.

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 178 matching lines @@
   congestion_controller_->SetChannelRembStatus(true, false,
                                                vie_channel_->rtp_rtcp());
 
   // Enable NACK, FEC or both.
   const bool enable_protection_nack = config_.rtp.nack.rtp_history_ms > 0;
   const bool enable_protection_fec = config_.rtp.fec.red_payload_type != -1;
   // TODO(changbin): Should set RTX for RED mapping in RTP sender in future.
   vie_channel_->SetProtectionMode(enable_protection_nack, enable_protection_fec,
                                   config_.rtp.fec.red_payload_type,
                                   config_.rtp.fec.ulpfec_payload_type);
-  vie_encoder_->UpdateProtectionMethod(enable_protection_nack,
-                                       enable_protection_fec);
+  vie_encoder_->SetProtectionMethod(enable_protection_nack,
+                                    enable_protection_fec);
 
   ConfigureSsrcs();
 
   vie_channel_->SetRTCPCName(config_.rtp.c_name.c_str());
 
   input_.reset(new internal::VideoCaptureInput(
       module_process_thread_, vie_encoder_.get(), config_.local_renderer,
       &stats_proxy_, this, config_.encoding_time_observer));
 
   // 28 to match packet overhead in ModuleRtpRtcpImpl.
@@ skipping 345 matching lines @@
 
   if (vie_channel_->SetSendCodec(video_codec, false) != 0) {
     LOG(LS_ERROR) << "Failed to set send codec.";
     return false;
   }
 
   // Not all configured SSRCs have to be utilized (simulcast senders don't have
   // to send on all SSRCs at once etc.)
   std::vector<uint32_t> used_ssrcs = config_.rtp.ssrcs;
   used_ssrcs.resize(static_cast<size_t>(video_codec.numberOfSimulcastStreams));
-
-  // Update used SSRCs.
   vie_encoder_->SetSsrcs(used_ssrcs);
 
-  // Update the protection mode, we might be switching NACK/FEC.
-  vie_encoder_->UpdateProtectionMethod(vie_encoder_->nack_enabled(),
-                                       vie_channel_->IsSendingFecEnabled());
-
   // Restart the media flow
   vie_encoder_->Restart();
 
   return true;
 }
 
 }  // namespace internal
 }  // namespace webrtc