commit | 63489787a0d0b7e613e8be3d9d06959ebc2abc45 | [log] [tgz] |
---|---|---|
author | henrik.lundin <henrik.lundin@webrtc.org> | Tue Sep 20 08:47:12 2016 |
committer | Commit bot <commit-bot@chromium.org> | Tue Sep 20 08:47:19 2016 |
tree | c322f5248fceb49d4f863de71c576fd1dc02540e | |
parent | 1b1863a11aa795ceb5286a73b0d391b1be86f556 [diff] |
Add new decoding statistics for muted output This change adds a new statistic for logging how many calls to NetEq::GetAudio resulted in a "muted output". A muted output happens if the packet stream has been dead for some time (and the last decoded packet was not comfort noise). BUG=webrtc:5606 BUG=b/31256483 Review-Url: https://codereview.webrtc.org/2341293002 Cr-Commit-Position: refs/heads/master@{#14302}
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.