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

webrtc/voice_engine/channel.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.
  */
 
@@ skipping 3155 matching lines @@
   return audio_coding_->GetNetworkStatistics(&stats);
 }
 
 void Channel::GetDecodingCallStatistics(AudioDecodingCallStats* stats) const {
   audio_coding_->GetDecodingCallStatistics(stats);
 }
 
 bool Channel::GetDelayEstimate(int* jitter_buffer_delay_ms,
                                int* playout_buffer_delay_ms) const {
   rtc::CritScope lock(&video_sync_lock_);
-  if (_average_jitter_buffer_delay_us == 0) {
-    return false;
-  }
-  *jitter_buffer_delay_ms =
-      (_average_jitter_buffer_delay_us + 500) / 1000 + _recPacketDelayMs;
+  *jitter_buffer_delay_ms = audio_coding_->FilteredCurrentDelayMs();
   *playout_buffer_delay_ms = playout_delay_ms_;
   return true;
 }
 
 uint32_t Channel::GetDelayEstimate() const {
   int jitter_buffer_delay_ms = 0;
   int playout_buffer_delay_ms = 0;
   GetDelayEstimate(&jitter_buffer_delay_ms, &playout_buffer_delay_ms);
   return jitter_buffer_delay_ms + playout_buffer_delay_ms;
 }
@@ skipping 192 matching lines @@
     if (rtcp) {
       playout_timestamp_rtcp_ = playout_timestamp;
     } else {
       playout_timestamp_rtp_ = playout_timestamp;
     }
     playout_delay_ms_ = delay_ms;
   }
 }
 
 // Called for incoming RTP packets after successful RTP header parsing.
+// TODO(henrik.lundin): Clean out this method. With the introduction of
+// AudioCoding::FilteredCurrentDelayMs() most (if not all) of this method can
+// be deleted, along with a few member variables. (WebRTC issue 6237.)

[the sun, 2016/08/23 06:45:07]

Oh, that would be so nice!

 void Channel::UpdatePacketDelay(uint32_t rtp_timestamp,
                                 uint16_t sequence_number) {
   WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::UpdatePacketDelay(timestamp=%lu, sequenceNumber=%u)",
                rtp_timestamp, sequence_number);
 
   // Get frequency of last received payload
   int rtp_receive_frequency = GetPlayoutFrequency();
 
   // |jitter_buffer_playout_timestamp_| updated in UpdatePlayoutTimestamp for
@@ skipping 154 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc