Remove audio/video distinction for probe packets.

webrtc/modules/remote_bitrate_estimator/send_time_history.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.
  */
 
 #include <assert.h>
 
 #include "webrtc/modules/remote_bitrate_estimator/include/send_time_history.h"
 
 namespace webrtc {
 
 SendTimeHistory::SendTimeHistory(Clock* clock, int64_t packet_age_limit)
     : clock_(clock),
       packet_age_limit_(packet_age_limit),
       oldest_sequence_number_(0) {}
 
 SendTimeHistory::~SendTimeHistory() {
 }
 
 void SendTimeHistory::Clear() {
   history_.clear();
 }
 
-void SendTimeHistory::AddAndRemoveOld(uint16_t sequence_number,
-                                      size_t length,
-                                      bool was_paced) {
+void SendTimeHistory::AddAndRemoveOld(uint16_t sequence_number, size_t length) {
   EraseOld();
 
   if (history_.empty())
     oldest_sequence_number_ = sequence_number;
 
   history_.insert(std::pair<uint16_t, PacketInfo>(
       sequence_number, PacketInfo(clock_->TimeInMilliseconds(), 0, -1,
-                                  sequence_number, length, was_paced)));
+                                  sequence_number, length)));
 }
 
 bool SendTimeHistory::OnSentPacket(uint16_t sequence_number,
                                    int64_t send_time_ms) {
   auto it = history_.find(sequence_number);
   if (it == history_.end())
     return false;
   it->second.send_time_ms = send_time_ms;
   return true;
 }
@@ skipping 42 matching lines @@
   packet->arrival_time_ms = receive_time;
   if (remove) {
     history_.erase(it);
     if (packet->sequence_number == oldest_sequence_number_)
       UpdateOldestSequenceNumber();
   }
   return true;
 }
 
 }  // namespace webrtc