commit | b3e30010de6a03def6c17ffd5cde759d618e68ff | [log] [tgz] |
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author | henrik.lundin <henrik.lundin@webrtc.org> | Wed Aug 31 21:09:51 2016 |
committer | Commit bot <commit-bot@chromium.org> | Wed Aug 31 21:09:55 2016 |
tree | 2e170e181736104fe3539c6acadfa82c8b01c549 | |
parent | 5a601d909fa4231d498a920ca74eec96a3c593b0 [diff] |
Remove Channel::UpdatePacketDelay and some member variables The method is no longer used, since the jitter buffer delay is obtained directly from AudioCodingModule instead of being calculated and smoothed in VoiceEngine. Deleting a few obsolete member variables as well. BUG=webrtc:6237 Review-Url: https://codereview.webrtc.org/2290253002 Cr-Commit-Position: refs/heads/master@{#14007}
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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.