Add NetEq::FilteredCurrentDelayMs() and use it in VoiceEngine

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.
  */
 
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   const size_t delay_samples =
       packet_buffer_->NumSamplesInBuffer(decoder_database_.get(),
                                          decoder_frame_length_) +
       sync_buffer_->FutureLength();
   // The division below will truncate.
   const int delay_ms =
       static_cast<int>(delay_samples) / rtc::CheckedDivExact(fs_hz_, 1000);
   return delay_ms;
 }
 
+int NetEqImpl::FilteredCurrentDelayMs() const {
+  rtc::CritScope lock(&crit_sect_);
+  // Calculate the filtered packet buffer level in samples. The value from
+  // |buffer_level_filter_| is in number of packets, represented in Q8.
+  const size_t packet_buffer_samples =
+      (buffer_level_filter_->filtered_current_level() *
+       decoder_frame_length_) >>
+      8;
+  // Sum up the filtered packet buffer level with the future length of the sync
+  // buffer, and divide the sum by the sample rate.
+  const size_t delay_samples =
+      packet_buffer_samples + sync_buffer_->FutureLength();
+  // The division below will truncate. The return value is in ms.
+  return static_cast<int>(delay_samples) / rtc::CheckedDivExact(fs_hz_, 1000);
+}
+
 // Deprecated.
 // TODO(henrik.lundin) Delete.
 void NetEqImpl::SetPlayoutMode(NetEqPlayoutMode mode) {
   rtc::CritScope lock(&crit_sect_);
   if (mode != playout_mode_) {
     playout_mode_ = mode;
     CreateDecisionLogic();
   }
 }
 
@@ skipping 1757 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(),
       tick_timer_.get()));
 }
 }  // namespace webrtc