Require negotiation to send transport cc feedback over RTCP.

webrtc/video/video_quality_test.cc

 /*
  *  Copyright (c) 2015 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.
  */
 #include <stdio.h>
@@ skipping 810 matching lines @@
   encoder_config_.streams = params_.ss.streams;
   encoder_config_.spatial_layers = params_.ss.spatial_layers;
 
   CreateMatchingReceiveConfigs(recv_transport);
 
   for (size_t i = 0; i < num_streams; ++i) {
     receive_configs_[i].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
     receive_configs_[i].rtp.rtx[kSendRtxPayloadType].ssrc = kSendRtxSsrcs[i];
     receive_configs_[i].rtp.rtx[kSendRtxPayloadType].payload_type =
         kSendRtxPayloadType;
+    receive_configs_[i].rtp.transport_cc = params_.common.send_side_bwe;
   }
 }
 
 void VideoQualityTest::SetupScreenshare() {
   RTC_CHECK(params_.screenshare.enabled);
 
   // Fill out codec settings.
   encoder_config_.content_type = VideoEncoderConfig::ContentType::kScreen;
   if (params_.common.codec == "VP8") {
     codec_settings_.VP8 = VideoEncoder::GetDefaultVp8Settings();
@@ skipping 222 matching lines @@
   send_stream_->Stop();
   receive_stream->Stop();
 
   call->DestroyVideoReceiveStream(receive_stream);
   call->DestroyVideoSendStream(send_stream_);
 
   transport.StopSending();
 }
 
 }  // namespace webrtc