a656b9d781
This ensure the absolute capture timestamp from the first audio sample encoded in the payload is used for the corresponding rtp header. Bug: webrtc:42226041 Change-Id: Ib8f2e3a5df5c82c5806171bd5b36a26d92fbea72 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349265 Commit-Queue: Lionel Koenig <lionelk@webrtc.org> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42281}
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