Today, behaviour is decided based on if transport sequence number v2 is in the SDP answer. But it might be better to decide based on received packets since it is valid to negotiate both extensions. Another bonus With this solution is that Call does not need to know about receive header exensions. This is an alternative to https://webrtc-review.googlesource.com/c/src/+/291337 Bug: webrtc:7135 Change-Id: Ib75474127d6e2e2029557b8bb2528eaac66979f8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291525 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Johannes Kron <kron@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39226}
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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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