| commit | 62aee9379c03bf0573d2549bc75571784522e736 | [log] [tgz] |
|---|---|---|
| author | Sebastian Jansson <srte@webrtc.org> | Wed Oct 02 10:27:06 2019 |
| committer | Commit Bot <commit-bot@chromium.org> | Wed Oct 02 13:42:15 2019 |
| tree | c3d67103abf65473ec10b6da935874fc9555859d | |
| parent | f1e97b9ebd23c12d12ffd6b18bdf3eb4951153b4 [diff] |
Adds trial to calculate audio overhead based on available data. This adds the ability to disable legacy overhead calculation so we'll use the available data on per packet over head and frame length range to set the min and max total allocatable bitrate. Bug: webrtc:11001 Change-Id: I2a94499433e15bad11a08f81fe7f1dfc27982cdf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155175 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29368}
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.