Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code.

webrtc/voice_engine/channel.cc

 /*
  *  Copyright (c) 2012 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 704 matching lines @@
     : _instanceId(instanceId),
       _channelId(channelId),
       event_log_(event_log),
       rtp_header_parser_(RtpHeaderParser::Create()),
       rtp_payload_registry_(
           new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(true))),
       rtp_receive_statistics_(
           ReceiveStatistics::Create(Clock::GetRealTimeClock())),
       rtp_receiver_(
           RtpReceiver::CreateAudioReceiver(Clock::GetRealTimeClock(),
-                                           nullptr,
                                            this,
                                            this,
                                            rtp_payload_registry_.get())),
       telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
       _outputAudioLevel(),
       _externalTransport(false),
       _inputFilePlayerPtr(NULL),
       _outputFilePlayerPtr(NULL),
       _outputFileRecorderPtr(NULL),
       // Avoid conflict with other channels by adding 1024 - 1026,
@@ skipping 2793 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc