| commit | 39adce149810089db056e35adeaa8a4d0d3e8a35 | [log] [tgz] |
|---|---|---|
| author | Jakob Ivarsson <jakobi@webrtc.org> | Thu Jun 25 14:09:58 2020 +0200 |
| committer | Commit Bot <commit-bot@chromium.org> | Thu Jun 25 14:51:13 2020 +0000 |
| tree | 7cb7bbb69e12d22a87ee05405b1251b173ef89ed | |
| parent | 118d01ac35adcf4d1efb5d225f9d85d857115c0b [diff] |
Add RtpEncodingParameters.adaptive_ptime. When enabled: - Creates an audio network adapter config that is passed to audio send stream. - Configures a lower default min bitrate. All parameters can be configured via a field trial that can also force enable the audio network adaptor (this is mainly intended for testing). Bug: chromium:1086942 Change-Id: I48dfcca1ee2948084199352abed6212a6c78eb6c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177840 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31565}
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 here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.