2237eb07c3
This is a reland of commit 38fcd58429b29c9474f1647efed7ebeb543c0637 Original change's description: > Change default NetEq sample rate to 48k. > > This should avoid some resampling before any packets have been received given that the vast majority of devices use 48k sample rate and the most common codec is Opus (which we always decode in 48k). > > Bug: none > Change-Id: Ie7baea57c3eb1b763a6460c3b06b56d67b2b258e > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280662 > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> > Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38536} Bug: none Change-Id: Id634799286f6d1f1eaf315ebe8e70de669d589db Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281900 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38572}
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.
Development
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.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation