Drop return value from RtpRtcp::IncomingRtcpPacket.

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 1766 matching lines @@
   return !in_order && statistician->IsRetransmitOfOldPacket(header, min_rtt);
 }
 
 int32_t Channel::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
   WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::ReceivedRTCPPacket()");
   // Store playout timestamp for the received RTCP packet
   UpdatePlayoutTimestamp(true);
 
   // Deliver RTCP packet to RTP/RTCP module for parsing
-  if (_rtpRtcpModule->IncomingRtcpPacket(data, length) == -1) {
-    _engineStatisticsPtr->SetLastError(
-        VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceWarning,
-        "Channel::IncomingRTPPacket() RTCP packet is invalid");
-  }
+  _rtpRtcpModule->IncomingRtcpPacket(data, length);
 
   int64_t rtt = GetRTT(true);
   if (rtt == 0) {
     // Waiting for valid RTT.
     return 0;
   }
 
   int64_t nack_window_ms = rtt;
   if (nack_window_ms < kMinRetransmissionWindowMs) {
     nack_window_ms = kMinRetransmissionWindowMs;
@@ skipping 1330 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