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

webrtc/modules/audio_coding/include/audio_coding_module.h

 /*
  *  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 661 matching lines @@
   ///////////////////////////////////////////////////////////////////////////
   // int32_t PlayoutTimestamp()
   // The send timestamp of an RTP packet is associated with the decoded
   // audio of the packet in question. This function returns the timestamp of
   // the latest audio obtained by calling PlayoutData10ms(), or empty if no
   // valid timestamp is available.
   //
   virtual rtc::Optional<uint32_t> PlayoutTimestamp() = 0;
 
   ///////////////////////////////////////////////////////////////////////////
+  // int FilteredCurrentDelayMs()
+  // Returns the current total delay from NetEq (packet buffer and sync buffer)
+  // in ms, with smoothing applied to even out short-time fluctuations due to
+  // jitter. The packet buffer part of the delay is not updated during DTX/CNG
+  // periods.
+  //
+  virtual int FilteredCurrentDelayMs() const = 0;
+
+  ///////////////////////////////////////////////////////////////////////////
   // int32_t PlayoutData10Ms(
   // Get 10 milliseconds of raw audio data for playout, at the given sampling
   // frequency. ACM will perform a resampling if required.
   //
   // Input:
   //   -desired_freq_hz    : the desired sampling frequency, in Hertz, of the
   //                         output audio. If set to -1, the function returns
   //                         the audio at the current sampling frequency.
   //
   // Output:
@@ skipping 124 matching lines @@
   virtual std::vector<uint16_t> GetNackList(
       int64_t round_trip_time_ms) const = 0;
 
   virtual void GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_