To properly handle SSRC collisions in non-BUNDLE we need to change how RTP stats IDs are generated, but that is a riskier change to be dealt with in a separate CL. For now, we just make sure that crashing is not a possibility during SSRC collisions as a mitigation for https://crbug.com/1361612. This is achieved by adding a TryAddStats() method to RTCStatsReport returning whether successful. Bug: chromium:1361612 Change-Id: I8577ae4c84a7c1eb3c7527e9efd8d1b0254269a3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275766 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38197}
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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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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