Propagate packet pacing information to SenTimeHistory

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 390 matching lines @@
   }
   return rtp_sender_.SendOutgoingData(
       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,
-                                         int probe_cluster_id) {
+                                         const PacedPacketInfo& pacing_info) {
   return rtp_sender_.TimeToSendPacket(ssrc, sequence_number, capture_time_ms,
-                                      retransmission, probe_cluster_id);
+                                      retransmission,
+                                      pacing_info.probe_cluster_id);
 }
 
-size_t ModuleRtpRtcpImpl::TimeToSendPadding(size_t bytes,
-                                            int probe_cluster_id) {
-  return rtp_sender_.TimeToSendPadding(bytes, probe_cluster_id);
+size_t ModuleRtpRtcpImpl::TimeToSendPadding(
+    size_t bytes,
+    const PacedPacketInfo& pacing_info) {
+  return rtp_sender_.TimeToSendPadding(bytes, pacing_info.probe_cluster_id);
 }
 
 size_t ModuleRtpRtcpImpl::MaxPayloadSize() const {
   return rtp_sender_.MaxPayloadSize();
 }
 
 size_t ModuleRtpRtcpImpl::MaxRtpPacketSize() const {
   return rtp_sender_.MaxRtpPacketSize();
 }
 
@@ skipping 482 matching lines @@
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_.GetRtpStatisticsCallback();
 }
 
 void ModuleRtpRtcpImpl::SetVideoBitrateAllocation(
     const BitrateAllocation& bitrate) {
   rtcp_sender_.SetVideoBitrateAllocation(bitrate);
 }
 }  // namespace webrtc