Drop return value from RtpRtcp::IncomingRtcpPacket.

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 252 matching lines @@
                                               int associated_payload_type) {
   rtp_sender_->SetRtxPayloadType(payload_type, associated_payload_type);
 }
 
 rtc::Optional<uint32_t> ModuleRtpRtcpImpl::FlexfecSsrc() const {
   if (rtp_sender_)
     return rtp_sender_->FlexfecSsrc();
   return rtc::Optional<uint32_t>();
 }
 
-int32_t ModuleRtpRtcpImpl::IncomingRtcpPacket(
-    const uint8_t* rtcp_packet,
-    const size_t length) {
-  return rtcp_receiver_.IncomingPacket(rtcp_packet, length) ? 0 : -1;
+void ModuleRtpRtcpImpl::IncomingRtcpPacket(const uint8_t* rtcp_packet,
+                                           const size_t length) {
+  rtcp_receiver_.IncomingPacket(rtcp_packet, length);
 }
 
 int32_t ModuleRtpRtcpImpl::RegisterSendPayload(
     const CodecInst& voice_codec) {
   return rtp_sender_->RegisterPayload(
       voice_codec.plname, voice_codec.pltype, voice_codec.plfreq,
       voice_codec.channels, (voice_codec.rate < 0) ? 0 : voice_codec.rate);
 }
 
 int32_t ModuleRtpRtcpImpl::RegisterSendPayload(const VideoCodec& video_codec) {
@@ skipping 636 matching lines @@
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_->GetRtpStatisticsCallback();
 }
 
 void ModuleRtpRtcpImpl::SetVideoBitrateAllocation(
     const BitrateAllocation& bitrate) {
   rtcp_sender_.SetVideoBitrateAllocation(bitrate);
 }
 }  // namespace webrtc