Add overhead per packet observer to the rtp_sender.

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 64 matching lines @@
                   configuration.outgoing_transport,
                   configuration.paced_sender,
                   configuration.flexfec_sender,
                   configuration.transport_sequence_number_allocator,
                   configuration.transport_feedback_callback,
                   configuration.send_bitrate_observer,
                   configuration.send_frame_count_observer,
                   configuration.send_side_delay_observer,
                   configuration.event_log,
                   configuration.send_packet_observer,
-                  configuration.retransmission_rate_limiter),
+                  configuration.retransmission_rate_limiter,
+                  configuration.overhead_observer),
       rtcp_sender_(configuration.audio,
                    configuration.clock,
                    configuration.receive_statistics,
                    configuration.rtcp_packet_type_counter_observer,
                    configuration.event_log,
                    configuration.outgoing_transport),
       rtcp_receiver_(configuration.clock,
                      configuration.receiver_only,
                      configuration.rtcp_packet_type_counter_observer,
                      configuration.bandwidth_callback,
@@ skipping 357 matching lines @@
   return 0;
 }
 
 void ModuleRtpRtcpImpl::SetTransportOverhead(
     int transport_overhead_per_packet) {
   RTC_DCHECK_GT(transport_overhead_per_packet, 0);
   int mtu = rtp_sender_.MaxPayloadLength() + packet_overhead_;
   RTC_DCHECK_LT(transport_overhead_per_packet, mtu);
   size_t max_payload_length = mtu - transport_overhead_per_packet;
   packet_overhead_ = transport_overhead_per_packet;
+  rtp_sender_.SetTransportOverhead(packet_overhead_);
   rtcp_sender_.SetMaxPayloadLength(max_payload_length);
   rtp_sender_.SetMaxPayloadLength(max_payload_length);
 }
 
 int32_t ModuleRtpRtcpImpl::SetMaxTransferUnit(uint16_t mtu) {
   RTC_DCHECK_LE(mtu, IP_PACKET_SIZE) << "MTU too large: " << mtu;
   RTC_DCHECK_GT(mtu, packet_overhead_) << "MTU too small: " << mtu;
   size_t max_payload_length = mtu - packet_overhead_;
   rtcp_sender_.SetMaxPayloadLength(max_payload_length);
   rtp_sender_.SetMaxPayloadLength(max_payload_length);
@@ skipping 472 matching lines @@
 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
     StreamDataCountersCallback* callback) {
   rtp_sender_.RegisterRtpStatisticsCallback(callback);
 }
 
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_.GetRtpStatisticsCallback();
 }
 }  // namespace webrtc