Deprecated SetAudioPacketSize from RTPSender and removed calls to it.

webrtc/voice_engine/channel.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 1305 matching lines @@
   if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) {
     _rtpRtcpModule->DeRegisterSendPayload(codec.pltype);
     if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) {
       WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
                    "SetSendCodec() failed to register codec to"
                    " RTP/RTCP module");
       return -1;
     }
   }
 
-  if (_rtpRtcpModule->SetAudioPacketSize(codec.pacsize) != 0) {
-    WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
-                 "SetSendCodec() failed to set audio packet size");
-    return -1;
-  }
-
   return 0;
 }
 
 void Channel::SetBitRate(int bitrate_bps, int64_t probing_interval_ms) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetBitRate(bitrate_bps=%d)", bitrate_bps);
   audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
     if (*encoder)
       (*encoder)->OnReceivedTargetAudioBitrate(bitrate_bps);
   });
@@ skipping 1943 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc