Add probe logging to RtcEventLog.

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 560 matching lines @@
     PacketOptions options;
     bool has_transport_seq_num =
         UpdateTransportSequenceNumber(&padding_packet, &options.packet_id);
     padding_packet.SetPadding(padding_bytes_in_packet, &random_);
 
     if (has_transport_seq_num) {
       AddPacketToTransportFeedback(options.packet_id, padding_packet,
                                    pacing_info);
     }
 
-    if (!SendPacketToNetwork(padding_packet, options))
+    if (!SendPacketToNetwork(padding_packet, options, pacing_info))
       break;
 
     bytes_sent += padding_bytes_in_packet;
     UpdateRtpStats(padding_packet, over_rtx, false);
   }
 
   return bytes_sent;
 }
 
 void RTPSender::SetStorePacketsStatus(bool enable, uint16_t number_to_store) {
@@ skipping 31 matching lines @@
     return packet->size();
   }
   bool rtx = (RtxStatus() & kRtxRetransmitted) > 0;
   int32_t packet_size = static_cast<int32_t>(packet->size());
   if (!PrepareAndSendPacket(std::move(packet), rtx, true, PacedPacketInfo()))
     return -1;
   return packet_size;
 }
 
 bool RTPSender::SendPacketToNetwork(const RtpPacketToSend& packet,
-                                    const PacketOptions& options) {
+                                    const PacketOptions& options,
+                                    const PacedPacketInfo& pacing_info) {
   int bytes_sent = -1;
   if (transport_) {
     UpdateRtpOverhead(packet);
     bytes_sent = transport_->SendRtp(packet.data(), packet.size(), options)
                      ? static_cast<int>(packet.size())
                      : -1;
     if (event_log_ && bytes_sent > 0) {
       event_log_->LogRtpHeader(kOutgoingPacket, MediaType::ANY, packet.data(),
-                               packet.size());
+                               packet.size(), pacing_info.probe_cluster_id);
     }
   }
   TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                        "RTPSender::SendPacketToNetwork", "size", packet.size(),
                        "sent", bytes_sent);
   // TODO(pwestin): Add a separate bitrate for sent bitrate after pacer.
   if (bytes_sent <= 0) {
     LOG(LS_WARNING) << "Transport failed to send packet.";
     return false;
   }
@@ skipping 100 matching lines @@
     AddPacketToTransportFeedback(options.packet_id, *packet_to_send,
                                  pacing_info);
   }
 
   if (!is_retransmit && !send_over_rtx) {
     UpdateDelayStatistics(packet->capture_time_ms(), now_ms);
     UpdateOnSendPacket(options.packet_id, packet->capture_time_ms(),
                        packet->Ssrc());
   }
 
-  if (!SendPacketToNetwork(*packet_to_send, options))
+  if (!SendPacketToNetwork(*packet_to_send, options, pacing_info))
     return false;
 
   {
     rtc::CritScope lock(&send_critsect_);
     media_has_been_sent_ = true;
   }
   UpdateRtpStats(*packet_to_send, send_over_rtx, is_retransmit);
   return true;
 }
 
@@ skipping 109 matching lines @@
   PacketOptions options;
   if (UpdateTransportSequenceNumber(packet.get(), &options.packet_id)) {
     AddPacketToTransportFeedback(options.packet_id, *packet.get(),
                                  PacedPacketInfo());
   }
 
   UpdateDelayStatistics(packet->capture_time_ms(), now_ms);
   UpdateOnSendPacket(options.packet_id, packet->capture_time_ms(),
                      packet->Ssrc());
 
-  bool sent = SendPacketToNetwork(*packet, options);
+  bool sent = SendPacketToNetwork(*packet, options, PacedPacketInfo());
 
   if (sent) {
     {
       rtc::CritScope lock(&send_critsect_);
       media_has_been_sent_ = true;
     }
     UpdateRtpStats(*packet, false, false);
   }
 
   // To support retransmissions, we store the media packet as sent in the
@@ skipping 370 matching lines @@
     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_;
   }
   overhead_observer_->OnOverheadChanged(overhead_bytes_per_packet);
 }
 
 }  // namespace webrtc