Add time line for acked bitrate.

webrtc/modules/remote_bitrate_estimator/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 52 matching lines @@
   rtc::CritScope cs(&lock_);
   send_time_history_.AddAndRemoveOld(sequence_number, length, probe_cluster_id);
 }
 
 void TransportFeedbackAdapter::OnSentPacket(uint16_t sequence_number,
                                             int64_t send_time_ms) {
   rtc::CritScope cs(&lock_);
   send_time_history_.OnSentPacket(sequence_number, send_time_ms);
 }
 
-const std::vector<PacketInfo> TransportFeedbackAdapter::GetPacketFeedbackVector(
+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 39 matching lines @@
       LOG(LS_WARNING) << "Failed to lookup send time for " << failed_lookups
                       << " packet" << (failed_lookups > 1 ? "s" : "")
                       << ". Send time history too small?";
     }
   }
   return packet_feedback_vector;
 }
 
 void TransportFeedbackAdapter::OnTransportFeedback(
     const rtcp::TransportFeedback& feedback) {
-  const std::vector<PacketInfo> packet_feedback_vector =
-      GetPacketFeedbackVector(feedback);
-  RTC_DCHECK(bitrate_estimator_.get() != nullptr);
-  bitrate_estimator_->IncomingPacketFeedbackVector(packet_feedback_vector);
+  last_packet_feedback_vector_ = GetPacketFeedbackVector(feedback);
+  if (bitrate_estimator_.get())
+    bitrate_estimator_->IncomingPacketFeedbackVector(
+        last_packet_feedback_vector_);
+}
+
+std::vector<PacketInfo> TransportFeedbackAdapter::GetTransportFeedbackVector()
+    const {
+  return last_packet_feedback_vector_;
 }
 
 void TransportFeedbackAdapter::OnReceiveBitrateChanged(
     const std::vector<uint32_t>& ssrcs,
     uint32_t bitrate) {
   bitrate_controller_->UpdateDelayBasedEstimate(bitrate);
 }
 
 void TransportFeedbackAdapter::OnProbeBitrate(uint32_t bitrate) {
   bitrate_controller_->UpdateProbeBitrate(bitrate);
 }
 
 void TransportFeedbackAdapter::OnRttUpdate(int64_t avg_rtt_ms,
                                            int64_t max_rtt_ms) {
   RTC_DCHECK(bitrate_estimator_.get() != nullptr);
   bitrate_estimator_->OnRttUpdate(avg_rtt_ms, max_rtt_ms);
 }
 
 }  // namespace webrtc