Propagate packet pacing information to SenTimeHistory

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 604 matching lines @@
     paced_sender_->InsertPacket(RtpPacketSender::kNormalPriority,
                                 packet->Ssrc(), packet->SequenceNumber(),
                                 corrected_capture_tims_ms,
                                 packet->payload_size(), true);
 
     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,
-                            PacketInfo::kNotAProbe))
+                            PacedPacketInfo::kNotAProbe))
     return -1;
   return packet_size;
 }
 
 bool RTPSender::SendPacketToNetwork(const RtpPacketToSend& packet,
                                     const PacketOptions& options) {
   int bytes_sent = -1;
   if (transport_) {
     UpdateRtpOverhead(packet);
     bytes_sent = transport_->SendRtp(packet.data(), packet.size(), options)
@@ skipping 237 matching lines @@
       TRACE_EVENT_ASYNC_BEGIN1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                                "PacedSend", corrected_time_ms,
                                "capture_time_ms", corrected_time_ms);
     }
     return true;
   }
 
   PacketOptions options;
   if (UpdateTransportSequenceNumber(packet.get(), &options.packet_id)) {
     AddPacketToTransportFeedback(options.packet_id, *packet.get(),
-                                 PacketInfo::kNotAProbe);
+                                 PacedPacketInfo::kNotAProbe);
   }
 
   UpdateDelayStatistics(packet->capture_time_ms(), now_ms);
   UpdateOnSendPacket(options.packet_id, packet->capture_time_ms(),
                      packet->Ssrc());
 
   bool sent = SendPacketToNetwork(*packet, options);
 
   if (sent) {
     {
@@ skipping 400 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