Make OverheadObserver::OnOverheadChanged count RTP headers only

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 896 matching lines @@
       _externalMixing(false),
       _mixFileWithMicrophone(false),
       input_mute_(false),
       previous_frame_muted_(false),
       _panLeft(1.0f),
       _panRight(1.0f),
       _outputGain(1.0f),
       _lastLocalTimeStamp(0),
       _lastPayloadType(0),
       _includeAudioLevelIndication(false),
+      transport_overhead_per_packet_(0),
+      rtp_overhead_per_packet_(0),
       _outputSpeechType(AudioFrame::kNormalSpeech),
       restored_packet_in_use_(false),
       rtcp_observer_(new VoERtcpObserver(this)),
       associate_send_channel_(ChannelOwner(nullptr)),
       pacing_enabled_(config.enable_voice_pacing),
       feedback_observer_proxy_(new TransportFeedbackProxy()),
       seq_num_allocator_proxy_(new TransportSequenceNumberProxy()),
       rtp_packet_sender_proxy_(new RtpPacketSenderProxy()),
       retransmission_rate_limiter_(new RateLimiter(Clock::GetRealTimeClock(),
                                                    kMaxRetransmissionWindowMs)),
@@ skipping 1944 matching lines @@
 }
 
 void Channel::SetRtcEventLog(RtcEventLog* event_log) {
   event_log_proxy_->SetEventLog(event_log);
 }
 
 void Channel::SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats) {
   rtcp_rtt_stats_proxy_->SetRtcpRttStats(rtcp_rtt_stats);
 }
 
-void Channel::SetTransportOverhead(int transport_overhead_per_packet) {
-  _rtpRtcpModule->SetTransportOverhead(transport_overhead_per_packet);
+void Channel::UpdateOverheadForEncoder() {
+  audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
+    if (*encoder) {
+      (*encoder)->OnReceivedOverhead(transport_overhead_per_packet_ +
+                                     rtp_overhead_per_packet_);
+    }
+  });
+}
+
+void Channel::SetTransportOverhead(size_t transport_overhead_per_packet) {
+  transport_overhead_per_packet_ = transport_overhead_per_packet;
+  UpdateOverheadForEncoder();
 }
 
 void Channel::OnOverheadChanged(size_t overhead_bytes_per_packet) {
-  audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
-    if (*encoder) {
-      (*encoder)->OnReceivedOverhead(overhead_bytes_per_packet);
-    }
-  });
+  rtp_overhead_per_packet_ = overhead_bytes_per_packet;
+  UpdateOverheadForEncoder();
 }
 
 int Channel::RegisterExternalMediaProcessing(ProcessingTypes type,
                                              VoEMediaProcess& processObject) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::RegisterExternalMediaProcessing()");
 
   rtc::CritScope cs(&_callbackCritSect);
 
   if (kPlaybackPerChannel == type) {
@@ skipping 387 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