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

webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.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.
  */
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h"
 
 #include <assert.h>  // assert
 #include <math.h>   // pow()
 #include <string.h>  // memcpy()
 
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 RTPReceiverStrategy* RTPReceiverStrategy::CreateAudioStrategy(
-    RtpData* data_callback,
-    RtpAudioFeedback* incoming_messages_callback) {
-  return new RTPReceiverAudio(data_callback, incoming_messages_callback);
+    RtpData* data_callback) {
+  return new RTPReceiverAudio(data_callback);
 }
 
-RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback,
-                                   RtpAudioFeedback* incoming_messages_callback)
+RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback)
     : RTPReceiverStrategy(data_callback),
       TelephoneEventHandler(),
       last_received_frequency_(8000),
       telephone_event_forward_to_decoder_(false),
       telephone_event_payload_type_(-1),
       cng_nb_payload_type_(-1),
       cng_wb_payload_type_(-1),
       cng_swb_payload_type_(-1),
       cng_fb_payload_type_(-1),
       cng_payload_type_(-1),
       g722_payload_type_(-1),
       last_received_g722_(false),
       num_energy_(0),
-      current_remote_energy_(),
-      cb_audio_feedback_(incoming_messages_callback) {
+      current_remote_energy_() {
   last_payload_.Audio.channels = 1;
   memset(current_remote_energy_, 0, sizeof(current_remote_energy_));
 }
 
 // Outband TelephoneEvent(DTMF) detection
 void RTPReceiverAudio::SetTelephoneEventForwardToDecoder(
     bool forward_to_decoder) {
   CriticalSectionScoped lock(crit_sect_.get());
   telephone_event_forward_to_decoder_ = forward_to_decoder;
 }
@@ skipping 321 matching lines @@
     // only one frame in the RED strip the one byte to help NetEq
     return data_callback_->OnReceivedPayloadData(
         payload_data + 1, payload_length - 1, rtp_header);
   }
 
   rtp_header->type.Audio.channel = audio_specific.channels;
   return data_callback_->OnReceivedPayloadData(
       payload_data, payload_length, rtp_header);
 }
 }  // namespace webrtc