This commit replaces the internal use of DurationMs, with millisecond precision, to webrtc::TimeDelta, which uses microsecond precision. This is just a refactoring. The only change to the public API is convenience methods to convert between DurationMs and webrtc::TimeDelta. Bug: webrtc:15593 Change-Id: Ida861bf585c716be5f898d0e7ef98da2c15268b7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325402 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41062}
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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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