Deprecated SetAudioPacketSize from RTPSender and removed calls to it.

webrtc/modules/rtp_rtcp/source/rtp_sender.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 1135 matching lines @@
   if (!audio_configured_) {
     return -1;
   }
   return audio_->SendTelephoneEvent(key, time_ms, level);
 }
 
 int32_t RTPSender::SetAudioPacketSize(uint16_t packet_size_samples) {
   if (!audio_configured_) {
     return -1;
   }
-  return audio_->SetAudioPacketSize(packet_size_samples);
+  return 0;
 }
 
 int32_t RTPSender::SetAudioLevel(uint8_t level_d_bov) {
   return audio_->SetAudioLevel(level_d_bov);
 }
 
 RtpVideoCodecTypes RTPSender::VideoCodecType() const {
   assert(!audio_configured_ && "Sender is an audio stream!");
   return video_->VideoCodecType();
 }
@@ skipping 150 matching lines @@
       return;
     }
     rtp_overhead_bytes_per_packet_ = packet.headers_size();
     overhead_bytes_per_packet =
         rtp_overhead_bytes_per_packet_ + transport_overhead_bytes_per_packet_;
   }
   overhead_observer_->OnOverheadChanged(overhead_bytes_per_packet);
 }
 
 }  // namespace webrtc