NetEq/ACM: Refactor how packet waiting times are calculated

webrtc/modules/audio_coding/neteq/neteq_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 300 matching lines @@
                                          decoder_frame_length_) +
       sync_buffer_->FutureLength();
   assert(delay_manager_.get());
   assert(decision_logic_.get());
   stats_.GetNetworkStatistics(fs_hz_, total_samples_in_buffers,
                               decoder_frame_length_, *delay_manager_.get(),
                               *decision_logic_.get(), stats);
   return 0;
 }
 
-void NetEqImpl::WaitingTimes(std::vector<int>* waiting_times) {
-  CriticalSectionScoped lock(crit_sect_.get());
-  stats_.WaitingTimes(waiting_times);
-}
-
 void NetEqImpl::GetRtcpStatistics(RtcpStatistics* stats) {
   CriticalSectionScoped lock(crit_sect_.get());
   if (stats) {
     rtcp_.GetStatistics(false, stats);
   }
 }
 
 void NetEqImpl::GetRtcpStatisticsNoReset(RtcpStatistics* stats) {
   CriticalSectionScoped lock(crit_sect_.get());
   if (stats) {
@@ skipping 1639 matching lines @@
 
 void NetEqImpl::CreateDecisionLogic() {
   decision_logic_.reset(DecisionLogic::Create(fs_hz_, output_size_samples_,
                                               playout_mode_,
                                               decoder_database_.get(),
                                               *packet_buffer_.get(),
                                               delay_manager_.get(),
                                               buffer_level_filter_.get()));
 }
 }  // namespace webrtc