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

webrtc/modules/rtp_rtcp/source/rtp_sender.h

 /*
  *  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 72 matching lines @@
                                    VideoRotation rotation) const = 0;
   virtual bool IsRtpHeaderExtensionRegistered(RTPExtensionType type) = 0;
   virtual CVOMode ActivateCVORtpHeaderExtension() = 0;
 };
 
 class RTPSender : public RTPSenderInterface {
  public:
   RTPSender(bool audio,
             Clock* clock,
             Transport* transport,
-            RtpAudioFeedback* audio_feedback,
             RtpPacketSender* paced_sender,
             TransportSequenceNumberAllocator* sequence_number_allocator,
             TransportFeedbackObserver* transport_feedback_callback,
             BitrateStatisticsObserver* bitrate_callback,
             FrameCountObserver* frame_count_observer,
             SendSideDelayObserver* send_side_delay_observer,
             RtcEventLog* event_log);
   virtual ~RTPSender();
 
   void ProcessBitrate();
@@ skipping 391 matching lines @@
   // that the target bitrate is still valid.
   rtc::scoped_ptr<CriticalSectionWrapper> target_bitrate_critsect_;
   uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_);
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_