The new method returns the current total delay (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. This CL also pipes the new metric through ACM and uses it in VoiceEngine. It replaces the previous method of estimating the buffer delay (where an inserted packet's RTP timestamp was compared with the last played timestamp from NetEq). The new method works better under periods of DTX/CNG. Review-Url: https://codereview.webrtc.org/2262203002 Cr-Commit-Position: refs/heads/master@{#13855}
Name: WebRTC URL: http://www.webrtc.org Version: 90 License: BSD License File: LICENSE Description: WebRTC provides real time voice and video processing functionality to enable the implementation of PeerConnection/MediaStream. Third party code used in this project is described in the file LICENSE_THIRD_PARTY.