eb9af84a55
- Filter out very old packets (to ensure that the estimate doesn't drop to zero if sending is paused and later resumed). - Discard packets older than previously discarded packets (to avoid the estimate dropping after deep reordering.) - Add tests cases for high loss, deep reordering and paused/resumed streams to unittest. - Remove some field trial settings that have very minor effect and rename some of the others. - Change analyzer.cc to only draw data points if the estimators have valid estimates. Bug: webrtc:13402 Change-Id: I47ead8aa4454cced5134d10895ca061d2c3e32f4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236347 Commit-Queue: Björn Terelius <terelius@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36849}
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