instead use the standard API to get the rollover counter and determine the extended sequence number which is the basis for the packet index. See https://github.com/cisco/libsrtp/issues/738 and https://github.com/cisco/libsrtp/issues/721 BUG=webrtc:357776213 Change-Id: I90c5a4a538f56132158aa48db8700187fcdb47d2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371960 Commit-Queue: Philipp Hancke <phancke@meta.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43802}
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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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