Remove deprecated RtpRtcp::SetAudioPacketSize

webrtc/modules/rtp_rtcp/source/rtp_rtcp_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 759 matching lines @@
 }
 
 // Send a TelephoneEvent tone using RFC 2833 (4733).
 int32_t ModuleRtpRtcpImpl::SendTelephoneEventOutband(
     const uint8_t key,
     const uint16_t time_ms,
     const uint8_t level) {
   return rtp_sender_->SendTelephoneEvent(key, time_ms, level);
 }
 
-int32_t ModuleRtpRtcpImpl::SetAudioPacketSize(
-    const uint16_t packet_size_samples) {
-  return audio_ ? 0 : -1;
-}
-
 int32_t ModuleRtpRtcpImpl::SetAudioLevel(
     const uint8_t level_d_bov) {
   return rtp_sender_->SetAudioLevel(level_d_bov);
 }
 
 int32_t ModuleRtpRtcpImpl::SetKeyFrameRequestMethod(
     const KeyFrameRequestMethod method) {
   key_frame_req_method_ = method;
   return 0;
 }
@@ skipping 126 matching lines @@
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_->GetRtpStatisticsCallback();
 }
 
 void ModuleRtpRtcpImpl::SetVideoBitrateAllocation(
     const BitrateAllocation& bitrate) {
   rtcp_sender_.SetVideoBitrateAllocation(bitrate);
 }
 }  // namespace webrtc