Set actual transport overhead in rtp_rtcp

webrtc/audio/audio_send_stream.cc

Index: webrtc/audio/audio_send_stream.cc
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index ee84b96aaa3700a8e7d937e0cf3d4ec1dfd0990d..9376dd9c90f965992eb5dc12c4550ec053826776 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -278,6 +278,12 @@ const webrtc::AudioSendStream::Config& AudioSendStream::config() const {
   return config_;
 }
 
+void AudioSendStream::SetTransportOverhead(
+    int transport_overhead_per_packet_byte) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  channel_proxy_->SetTransportOverhead(transport_overhead_per_packet_byte);
+}
+
 VoiceEngine* AudioSendStream::voice_engine() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());