b3f1c5d2fe
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}
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others. This page is maintained by the Google Chrome team.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc