Add overhead to transport feedback observer.

webrtc/modules/rtp_rtcp/source/rtp_sender.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 11 matching lines @@
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/playout_delay_oracle.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
 #include "webrtc/modules/rtp_rtcp/source/time_util.h"
+#include "webrtc/system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 
 namespace {
 // Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
 constexpr size_t kMaxPaddingLength = 224;
 constexpr int kSendSideDelayWindowMs = 1000;
 constexpr size_t kRtpHeaderLength = 12;
 constexpr uint16_t kMaxInitRtpSeqNumber = 32767;  // 2^15 -1.
 constexpr uint32_t kTimestampTicksPerMs = 90;
@@ skipping 1249 matching lines @@
     transport_overhead_bytes_per_packet_ = transport_overhead;
     overhead_bytes_per_packet =
         rtp_overhead_bytes_per_packet_ + transport_overhead_bytes_per_packet_;
   }
   overhead_observer_->OnOverheadChanged(overhead_bytes_per_packet);
 }
 
 void RTPSender::AddPacketToTransportFeedback(uint16_t packet_id,
                                              const RtpPacketToSend& packet,
                                              int probe_cluster_id) {
+  size_t packet_size = packet.payload_size() + packet.padding_size();
+  if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe-WithOverhead") ==
+      "Enabled") {
+    rtc::CritScope lock(&send_critsect_);
+    packet_size = packet.size() + transport_overhead_bytes_per_packet_;
+  }
+
   if (transport_feedback_observer_) {
-    transport_feedback_observer_->AddPacket(
-        packet_id, packet.payload_size() + packet.padding_size(),
-        probe_cluster_id);
+    transport_feedback_observer_->AddPacket(packet_id, packet_size,
+                                            probe_cluster_id);
   }
 }
 
 void RTPSender::UpdateRtpOverhead(const RtpPacketToSend& packet) {
   if (!overhead_observer_)
     return;
   size_t overhead_bytes_per_packet = 0;
   {
     rtc::CritScope lock(&send_critsect_);
     if (rtp_overhead_bytes_per_packet_ == packet.headers_size()) {
       return;
     }
     rtp_overhead_bytes_per_packet_ = packet.headers_size();
     overhead_bytes_per_packet =
         rtp_overhead_bytes_per_packet_ + transport_overhead_bytes_per_packet_;
   }
   overhead_observer_->OnOverheadChanged(overhead_bytes_per_packet);
 }
 
 }  // namespace webrtc