| commit | a7382f787911314066c0a0c9b3e7e8d2aa0a7932 | [log] [tgz] |
|---|---|---|
| author | Alessio Bazzica <alessiob@webrtc.org> | Thu Feb 27 12:46:51 2020 |
| committer | Commit Bot <commit-bot@chromium.org> | Thu Feb 27 14:27:23 2020 |
| tree | df1b0da076d28f20684868e30ca78498b36c2af9 | |
| parent | 4a6f81829b68fcf43e46a5cdead4c75e8b009eef [diff] |
iSAC API wrapper unit test fix Use speech content instead of white noise and enable target vs measured bitrate tests. Bug: webrtc:11360 Change-Id: If8c8e73f943eda14efeb22ba406c7a1bed7d32b4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168660 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30630}
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.