commit | c2a028887f71d92b81d7744fbc9618a126368604 | [log] [tgz] |
---|---|---|
author | Christoffer Rodbro <crodbro@webrtc.org> | Tue Aug 07 14:10:56 2018 +0200 |
committer | Commit Bot <commit-bot@chromium.org> | Tue Aug 07 13:49:04 2018 +0000 |
tree | fed3d8c193a619f5e054f390b3c54ac9ee9e202f | |
parent | b3e2c8eb1b16f9a804c6525bac1056200f09d8da [diff] |
Enable audio in video_quality_test. Allows enabling audio for RunWithAnalyzer method, and prints out audio jitterbuffer performance stats. Also fixes for RunWithRenderer when enabling audio (seg-faulted). Bug: b/112299470 Change-Id: Ic7c0de1c455891f38cca317001c6c216e82f6ec3 Reviewed-on: https://webrtc-review.googlesource.com/92800 Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24208}
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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.