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

webrtc/modules/audio_coding/acm2/acm_receiver.cc

 /*
  *  Copyright (c) 2013 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 291 matching lines @@
     last_packet_sample_rate_hz_ = rtc::Optional<int>();
   }
   decoders_.erase(it);
   return 0;
 }
 
 rtc::Optional<uint32_t> AcmReceiver::GetPlayoutTimestamp() {
   return neteq_->GetPlayoutTimestamp();
 }
 
+int AcmReceiver::FilteredCurrentDelayMs() const {
+  return neteq_->FilteredCurrentDelayMs();
+}
+
 int AcmReceiver::LastAudioCodec(CodecInst* codec) const {
   rtc::CritScope lock(&crit_sect_);
   if (!last_audio_decoder_) {
     return -1;
   }
   *codec = *RentACodec::CodecInstById(
       *RentACodec::CodecIdFromIndex(last_audio_decoder_->acm_codec_id));
   codec->pltype = last_audio_decoder_->payload_type;
   codec->channels = last_audio_decoder_->channels;
   codec->plfreq = last_audio_decoder_->sample_rate_hz;
@@ skipping 89 matching lines @@
 
 void AcmReceiver::GetDecodingCallStatistics(
     AudioDecodingCallStats* stats) const {
   rtc::CritScope lock(&crit_sect_);
   *stats = call_stats_.GetDecodingStatistics();
 }
 
 }  // namespace acm2
 
 }  // namespace webrtc