Remove deprecated RtpReceiver::CreateAudioReceiver() function.

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.
  */
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h"
 
 #include <assert.h>
 #include <math.h>
 #include <stdlib.h>
 #include <string.h>
 
 #include "webrtc/base/logging.h"

[hlundin-webrtc, 2016/03/31 18:22:47]

Can you remove any of these includes now that RtpAudioFeedback is no longer
used?

 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 
 namespace webrtc {
 
 using RtpUtility::Payload;
 
 RtpReceiver* RtpReceiver::CreateVideoReceiver(
     Clock* clock,
@@ skipping 16 matching lines @@
     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,
       RTPReceiverStrategy::CreateAudioStrategy(incoming_payload_callback));
 }
 
-// TODO(solenberg): Remove, after updating downstream code.
-RtpReceiver* RtpReceiver::CreateAudioReceiver(
-    Clock* clock,
-    RtpAudioFeedback* incoming_audio_feedback,
-    RtpData* incoming_payload_callback,
-    RtpFeedback* incoming_messages_callback,
-    RTPPayloadRegistry* rtp_payload_registry) {
-  return CreateAudioReceiver(clock,
-                             incoming_payload_callback,
-                             incoming_messages_callback,
-                             rtp_payload_registry);
-}
-
 RtpReceiverImpl::RtpReceiverImpl(
     Clock* clock,
     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_(
@@ skipping 405 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