Revert of Move delay_based_bwe_ into CongestionController

webrtc/audio/audio_send_stream.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.
  */
 
@@ skipping 68 matching lines @@
     } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
       channel_proxy_->EnableSendTransportSequenceNumber(extension.id);
       congestion_controller->EnablePeriodicAlrProbing(true);
       bandwidth_observer_.reset(congestion_controller->GetBitrateController()
                                     ->CreateRtcpBandwidthObserver());
     } else {
       RTC_NOTREACHED() << "Registering unsupported RTP extension.";
     }
   }
   channel_proxy_->RegisterSenderCongestionControlObjects(
-      congestion_controller->pacer(), congestion_controller, packet_router,
+      congestion_controller->pacer(),
+      congestion_controller->GetTransportFeedbackObserver(), packet_router,
       bandwidth_observer_.get());
   if (!SetupSendCodec()) {
     LOG(LS_ERROR) << "Failed to set up send codec state.";
   }
 }
 
 AudioSendStream::~AudioSendStream() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
   channel_proxy_->DeRegisterExternalTransport();
@@ skipping 273 matching lines @@
         LOG(LS_WARNING) << "SetVADStatus() failed.";
         return false;
       }
     }
   }
   return true;
 }
 
 }  // namespace internal
 }  // namespace webrtc