Propagate packet pacing information to SendTimeHistory.

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 376 matching lines @@
       frame_type, payload_type, time_stamp, capture_time_ms, payload_data,
       payload_size, fragmentation, rtp_video_header, transport_frame_id_out);
 }
 
 bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc,
                                          uint16_t sequence_number,
                                          int64_t capture_time_ms,
                                          bool retransmission,
                                          const PacedPacketInfo& pacing_info) {
   return rtp_sender_.TimeToSendPacket(ssrc, sequence_number, capture_time_ms,
-                                      retransmission,
-                                      pacing_info.probe_cluster_id);
+                                      retransmission, pacing_info);

[nisse-webrtc, 2017/02/22 12:37:47]

And this is where pacing_info gets passed further on.

Is it populated at this point? Are values used yet at the end of the line?

[philipel, 2017/02/22 13:11:26]

Yes, |pacing_info| is populated by the pacer and is not used until we actually
receive feedback. The |pacing_info| is just passed around everywhere just to end
up in SendTimeHistory.

 }
 
 size_t ModuleRtpRtcpImpl::TimeToSendPadding(
     size_t bytes,
     const PacedPacketInfo& pacing_info) {
-  return rtp_sender_.TimeToSendPadding(bytes, pacing_info.probe_cluster_id);
+  return rtp_sender_.TimeToSendPadding(bytes, pacing_info);
 }
 
 size_t ModuleRtpRtcpImpl::MaxPayloadSize() const {
   return rtp_sender_.MaxPayloadSize();
 }
 
 size_t ModuleRtpRtcpImpl::MaxRtpPacketSize() const {
   return rtp_sender_.MaxRtpPacketSize();
 }
 
@@ skipping 464 matching lines @@
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_.GetRtpStatisticsCallback();
 }
 
 void ModuleRtpRtcpImpl::SetVideoBitrateAllocation(
     const BitrateAllocation& bitrate) {
   rtcp_sender_.SetVideoBitrateAllocation(bitrate);
 }
 }  // namespace webrtc