The same as https://webrtc-review.googlesource.com/c/src/+/331340, but for interleaved messages. This avoids inserting into maps where possible, and also fixes a bug when the payload was accidentally copied unintentionally - crbug.com/365594101. Bug: chromium:365594101 Change-Id: Iaeaa97b0cf3a26ada9afc61f2545760b7ab4c731 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363960 Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43099}
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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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