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

webrtc/modules/audio_coding/include/audio_coding_module.h

Index: webrtc/modules/audio_coding/include/audio_coding_module.h
diff --git a/webrtc/modules/audio_coding/include/audio_coding_module.h b/webrtc/modules/audio_coding/include/audio_coding_module.h
index 5b35bcf7a5443317343a7d003ad1ec28458f874c..b7f8e291beb3ee38bdac0108ad8ed5acd806cb49 100644
--- a/webrtc/modules/audio_coding/include/audio_coding_module.h
+++ b/webrtc/modules/audio_coding/include/audio_coding_module.h
@@ -675,6 +675,15 @@ class AudioCodingModule {
   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.