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

webrtc/modules/audio_coding/acm2/audio_coding_module.cc

Index: webrtc/modules/audio_coding/acm2/audio_coding_module.cc
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
index 3f404f736f31364f2b120c874efd3e8695b5a15d..f719b6c59cb68dd8bba3ea18e18fa06e5c7c9893 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
@@ -159,6 +159,8 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
 
   rtc::Optional<uint32_t> PlayoutTimestamp() override;
 
+  int FilteredCurrentDelayMs() const override;
+
   // Get 10 milliseconds of raw audio data to play out, and
   // automatic resample to the requested frequency if > 0.
   int PlayoutData10Ms(int desired_freq_hz,
@@ -1225,6 +1227,10 @@ rtc::Optional<uint32_t> AudioCodingModuleImpl::PlayoutTimestamp() {
   return receiver_.GetPlayoutTimestamp();
 }
 
+int AudioCodingModuleImpl::FilteredCurrentDelayMs() const {
+  return receiver_.FilteredCurrentDelayMs();
+}
+
 bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const {
   if (!encoder_stack_) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,