Propagate packet pacing information to SenTimeHistory

webrtc/modules/congestion_controller/transport_feedback_adapter.cc

 /*
  *  Copyright (c) 2015 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 60 matching lines @@
 void TransportFeedbackAdapter::SetMinBitrate(int min_bitrate_bps) {
   rtc::CritScope cs(&bwe_lock_);
   delay_based_bwe_->SetMinBitrate(min_bitrate_bps);
 }
 
 int64_t TransportFeedbackAdapter::GetProbingIntervalMs() const {
   rtc::CritScope cs(&bwe_lock_);
   return delay_based_bwe_->GetProbingIntervalMs();
 }
 
+void TransportFeedbackAdapter::OnProbingClusterCreated(int cluster_id,
+                                                       int min_bytes,
+                                                       int min_probes) {
+  rtc::CritScope cs(&bwe_lock_);
+  delay_based_bwe_->OnProbingClusterCreated(cluster_id, min_bytes, min_probes);
+}
+
 std::vector<PacketInfo> TransportFeedbackAdapter::GetPacketFeedbackVector(
     const rtcp::TransportFeedback& feedback) {
   int64_t timestamp_us = feedback.GetBaseTimeUs();
   // Add timestamp deltas to a local time base selected on first packet arrival.
   // This won't be the true time base, but makes it easier to manually inspect
   // time stamps.
   if (last_timestamp_us_ == kNoTimestamp) {
     current_offset_ms_ = clock_->TimeInMilliseconds();
   } else {
     int64_t delta = timestamp_us - last_timestamp_us_;
@@ skipping 63 matching lines @@
   return last_packet_feedback_vector_;
 }
 
 void TransportFeedbackAdapter::OnRttUpdate(int64_t avg_rtt_ms,
                                            int64_t max_rtt_ms) {
   rtc::CritScope cs(&bwe_lock_);
   delay_based_bwe_->OnRttUpdate(avg_rtt_ms, max_rtt_ms);
 }
 
 }  // namespace webrtc