| commit | 83bd37cda43249e69de07d40bcba5539bcbe7f9e | [log] [tgz] |
|---|---|---|
| author | Jakob Ivarsson <jakobi@webrtc.org> | Mon Oct 15 07:54:46 2018 |
| committer | Commit Bot <commit-bot@chromium.org> | Mon Oct 15 08:59:43 2018 |
| tree | fc1ffa443b8b096f0ec2e833987000854e1cca0d | |
| parent | fcebe0e1ca5c4c32211337a287af81fe0a8b9e3f [diff] |
Add field trials for configuring Opus encoder packet loss rate. Add options to: 1. Bypass optimization (use reported packet loss). 2. Set a maximum value. 3. Set a coefficient. Bug: webrtc:9866 Change-Id: I3fef43e5186a4f0f50fda3506e445860518cfbd7 Reviewed-on: https://webrtc-review.googlesource.com/c/105304 Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25161}
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.