Remove deprecated RtpRtcp::SetAudioPacketSize

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 1114 matching lines @@
 // Audio.
 int32_t RTPSender::SendTelephoneEvent(uint8_t key,
                                       uint16_t time_ms,
                                       uint8_t level) {
   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 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();
 }
 
 void RTPSender::SetUlpfecConfig(int red_payload_type, int ulpfec_payload_type) {
@@ skipping 151 matching lines @@
     rtc::CritScope lock(&send_critsect_);
     packet->SetTimestamp(last_rtp_timestamp_);
     packet->set_capture_time_ms(capture_time_ms_);
   }
   AssignSequenceNumber(packet.get());
   SendToNetwork(std::move(packet), StorageType::kDontRetransmit,
                 RtpPacketSender::Priority::kLowPriority);
 }
 
 }  // namespace webrtc