Remove audio/video distinction for probe packets.

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 41 matching lines @@
 
 void TransportFeedbackAdapter::SetBitrateEstimator(
     RemoteBitrateEstimator* rbe) {
   if (bitrate_estimator_.get() != rbe) {
     bitrate_estimator_.reset(rbe);
   }
 }
 
 void TransportFeedbackAdapter::AddPacket(uint16_t sequence_number,
                                          size_t length,
-                                         bool was_paced,
                                          int probe_cluster_id) {
   rtc::CritScope cs(&lock_);
-  send_time_history_.AddAndRemoveOld(sequence_number, length, was_paced,
-                                     probe_cluster_id);
+  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);
 }
 
 void TransportFeedbackAdapter::OnTransportFeedback(
     const rtcp::TransportFeedback& feedback) {
@@ skipping 61 matching lines @@
   bitrate_controller_->UpdateDelayBasedEstimate(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