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

webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.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 16 matching lines @@
 RtpReceiver* RtpReceiver::CreateVideoReceiver(
     Clock* clock,
     RtpData* incoming_payload_callback,
     RtpFeedback* incoming_messages_callback,
     RTPPayloadRegistry* rtp_payload_registry) {
   if (!incoming_payload_callback)
     incoming_payload_callback = NullObjectRtpData();
   if (!incoming_messages_callback)
     incoming_messages_callback = NullObjectRtpFeedback();
   return new RtpReceiverImpl(
-      clock, incoming_messages_callback, rtp_payload_registry,
+      clock, NullObjectRtpAudioFeedback(), incoming_messages_callback,
+      rtp_payload_registry,
       RTPReceiverStrategy::CreateVideoStrategy(incoming_payload_callback));
 }
 
 RtpReceiver* RtpReceiver::CreateAudioReceiver(
     Clock* clock,
+    RtpAudioFeedback* incoming_audio_feedback,
     RtpData* incoming_payload_callback,
     RtpFeedback* incoming_messages_callback,
     RTPPayloadRegistry* rtp_payload_registry) {
+  if (!incoming_audio_feedback)
+    incoming_audio_feedback = NullObjectRtpAudioFeedback();
   if (!incoming_payload_callback)
     incoming_payload_callback = NullObjectRtpData();
   if (!incoming_messages_callback)
     incoming_messages_callback = NullObjectRtpFeedback();
   return new RtpReceiverImpl(
-      clock, incoming_messages_callback, rtp_payload_registry,
-      RTPReceiverStrategy::CreateAudioStrategy(incoming_payload_callback));
+      clock, incoming_audio_feedback, incoming_messages_callback,
+      rtp_payload_registry,
+      RTPReceiverStrategy::CreateAudioStrategy(incoming_payload_callback,
+                                               incoming_audio_feedback));
 }
 
 RtpReceiverImpl::RtpReceiverImpl(
     Clock* clock,
+    RtpAudioFeedback* incoming_audio_messages_callback,
     RtpFeedback* incoming_messages_callback,
     RTPPayloadRegistry* rtp_payload_registry,
     RTPReceiverStrategy* rtp_media_receiver)
     : clock_(clock),
       rtp_payload_registry_(rtp_payload_registry),
       rtp_media_receiver_(rtp_media_receiver),
       cb_rtp_feedback_(incoming_messages_callback),
       critical_section_rtp_receiver_(
           CriticalSectionWrapper::CreateCriticalSection()),
       last_receive_time_(0),
       last_received_payload_length_(0),
       ssrc_(0),
       num_csrcs_(0),
       current_remote_csrc_(),
       last_received_timestamp_(0),
       last_received_frame_time_ms_(-1),
       last_received_sequence_number_(0),
       nack_method_(kNackOff) {
+  assert(incoming_audio_messages_callback);
   assert(incoming_messages_callback);
 
   memset(current_remote_csrc_, 0, sizeof(current_remote_csrc_));
 }
 
 RtpReceiverImpl::~RtpReceiverImpl() {
   for (int i = 0; i < num_csrcs_; ++i) {
     cb_rtp_feedback_->OnIncomingCSRCChanged(current_remote_csrc_[i], false);
   }
 }
@@ skipping 385 matching lines @@
     // implementations might have CSRC 0 as a valid value.
     if (num_csrcs_diff > 0) {
       cb_rtp_feedback_->OnIncomingCSRCChanged(0, true);
     } else if (num_csrcs_diff < 0) {
       cb_rtp_feedback_->OnIncomingCSRCChanged(0, false);
     }
   }
 }
 
 }  // namespace webrtc