Move delay_based_bwe_ into CongestionController

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.
  */
 #include "webrtc/video/video_send_stream.h"
@@ skipping 771 matching lines @@
       encoder_feedback_(Clock::GetRealTimeClock(),
                         config_->rtp.ssrcs,
                         vie_encoder),
       protection_bitrate_calculator_(Clock::GetRealTimeClock(), this),
       bandwidth_observer_(congestion_controller_->GetBitrateController()
                               ->CreateRtcpBandwidthObserver()),
       rtp_rtcp_modules_(CreateRtpRtcpModules(
           config_->send_transport,
           &encoder_feedback_,
           bandwidth_observer_.get(),
-          congestion_controller_->GetTransportFeedbackObserver(),
+          congestion_controller_,
           call_stats_->rtcp_rtt_stats(),
           congestion_controller_->pacer(),
           packet_router_,
           flexfec_sender_.get(),
           stats_proxy_,
           send_delay_stats,
           event_log,
           congestion_controller_->GetRetransmissionRateLimiter(),
           this,
           config_->rtp.ssrcs.size())),
@@ skipping 535 matching lines @@
       std::min(config_->rtp.max_packet_size,
                kPathMTU - transport_overhead_bytes_per_packet_);
 
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
     rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size);
   }
 }
 
 }  // namespace internal
 }  // namespace webrtc