Set actual transport overhead in rtp_rtcp

webrtc/video/video_send_stream.cc

 /*
  *  Copyright (c) 2013 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.
  */
 #include "webrtc/video/video_send_stream.h"
@@ skipping 256 matching lines @@
   void SignalNetworkState(NetworkState state);
   bool DeliverRtcp(const uint8_t* packet, size_t length);
   void Start();
   void Stop();
 
   VideoSendStream::RtpStateMap GetRtpStates() const;
 
   void EnableEncodedFrameRecording(const std::vector<rtc::PlatformFile>& files,
                                    size_t byte_limit);
 
+  void SetTransportOverhead(int transport_overhead_per_packet);
+
  private:
   class CheckEncoderActivityTask;
   class EncoderReconfiguredTask;
 
   // Implements BitrateAllocatorObserver.
   uint32_t OnBitrateUpdated(uint32_t bitrate_bps,
                             uint8_t fraction_loss,
                             int64_t rtt) override;
 
   // Implements webrtc::VCMProtectionCallback.
@@ skipping 323 matching lines @@
   vie_encoder_->DeRegisterProcessThread();
   VideoSendStream::RtpStateMap state_map;
   send_stream_->DeRegisterProcessThread();
   worker_queue_->PostTask(
       std::unique_ptr<rtc::QueuedTask>(new DestructAndGetRtpStateTask(
           &state_map, std::move(send_stream_), &thread_sync_event_)));
   thread_sync_event_.Wait(rtc::Event::kForever);
   return state_map;
 }
 
+void VideoSendStream::SetTransportOverhead(int transport_overhead_per_packet) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  VideoSendStreamImpl* send_stream = send_stream_.get();
+  worker_queue_->PostTask([send_stream, transport_overhead_per_packet] {
+    send_stream->SetTransportOverhead(transport_overhead_per_packet);
+  });
+}
+
 bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
   // Called on a network thread.
   return send_stream_->DeliverRtcp(packet, length);
 }
 
 void VideoSendStream::EnableEncodedFrameRecording(
     const std::vector<rtc::PlatformFile>& files,
     size_t byte_limit) {
   send_stream_->EnableEncodedFrameRecording(files, byte_limit);
 }
@@ skipping 485 matching lines @@
     rtp_rtcp->SetFecParameters(*delta_params, *key_params);
     rtp_rtcp->BitrateSent(&not_used, &module_video_rate, &module_fec_rate,
                           &module_nack_rate);
     *sent_video_rate_bps += module_video_rate;
     *sent_nack_rate_bps += module_nack_rate;
     *sent_fec_rate_bps += module_fec_rate;
   }
   return 0;
 }
 
+void VideoSendStreamImpl::SetTransportOverhead(
+    int transport_overhead_per_packet) {
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
+    rtp_rtcp->SetTransportOverhead(transport_overhead_per_packet);
+}
+
 }  // namespace internal
 }  // namespace webrtc