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

webrtc/modules/rtp_rtcp/source/rtp_sender.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 90 matching lines @@
 
 const BitrateStatistics&
 RTPSender::BitrateAggregator::BitrateObserver::statistics() const {
   return statistics_;
 }
 
 RTPSender::RTPSender(
     bool audio,
     Clock* clock,
     Transport* transport,
-    RtpAudioFeedback* audio_feedback,
     RtpPacketSender* paced_sender,
     TransportSequenceNumberAllocator* sequence_number_allocator,
     TransportFeedbackObserver* transport_feedback_observer,
     BitrateStatisticsObserver* bitrate_callback,
     FrameCountObserver* frame_count_observer,
     SendSideDelayObserver* send_side_delay_observer,
     RtcEventLog* event_log)
     : clock_(clock),
       // TODO(holmer): Remove this conversion when we remove the use of
       // TickTime.
       clock_delta_ms_(clock_->TimeInMilliseconds() -
                       TickTime::MillisecondTimestamp()),
       random_(clock_->TimeInMicroseconds()),
       bitrates_(bitrate_callback),
       total_bitrate_sent_(clock, bitrates_.total_bitrate_observer()),
       audio_configured_(audio),
-      audio_(audio ? new RTPSenderAudio(clock, this, audio_feedback) : nullptr),
+      audio_(audio ? new RTPSenderAudio(clock, this) : nullptr),
       video_(audio ? nullptr : new RTPSenderVideo(clock, this)),
       paced_sender_(paced_sender),
       transport_sequence_number_allocator_(sequence_number_allocator),
       transport_feedback_observer_(transport_feedback_observer),
       last_capture_time_ms_sent_(0),
       transport_(transport),
       sending_media_(true),                      // Default to sending media.
       max_payload_length_(IP_PACKET_SIZE - 28),  // Default is IP-v4/UDP.
       packet_over_head_(28),
       payload_type_(-1),
@@ skipping 1772 matching lines @@
   rtc::CritScope lock(&send_critsect_);
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc