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

webrtc/modules/rtp_rtcp/include/rtp_rtcp.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 37 matching lines @@
     *                         data. May not be NULL; default callback will do
     *                         nothing.
     *  incoming_messages    - Callback object that will receive the incoming
     *                         RTP messages. May not be NULL; default callback
     *                         will do nothing.
     *  outgoing_transport   - Transport object that will be called when packets
     *                         are ready to be sent out on the network
     *  intra_frame_callback - Called when the receiver request a intra frame.
     *  bandwidth_callback   - Called when we receive a changed estimate from
     *                         the receiver of out stream.
-    *  audio_messages       - Telephone events. May not be NULL; default
-    *                         callback will do nothing.
     *  remote_bitrate_estimator - Estimates the bandwidth available for a set of
     *                             streams from the same client.
     *  paced_sender             - Spread any bursts of packets into smaller
     *                             bursts to minimize packet loss.
     */
     bool audio;
     bool receiver_only;
     Clock* clock;
     ReceiveStatistics* receive_statistics;
     Transport* outgoing_transport;
     RtcpIntraFrameObserver* intra_frame_callback;
     RtcpBandwidthObserver* bandwidth_callback;
     TransportFeedbackObserver* transport_feedback_callback;
     RtcpRttStats* rtt_stats;
     RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer;
-    RtpAudioFeedback* audio_messages;
     RemoteBitrateEstimator* remote_bitrate_estimator;
     RtpPacketSender* paced_sender;
     TransportSequenceNumberAllocator* transport_sequence_number_allocator;
     BitrateStatisticsObserver* send_bitrate_observer;
     FrameCountObserver* send_frame_count_observer;
     SendSideDelayObserver* send_side_delay_observer;
     RtcEventLog* event_log;
 
     RTC_DISALLOW_COPY_AND_ASSIGN(Configuration);
   };
@@ skipping 574 matching lines @@
 
     /*
     *   send a request for a keyframe
     *
     *   return -1 on failure else 0
     */
     virtual int32_t RequestKeyFrame() = 0;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_