This information is now readily available. Let's expose it. In practise we don't pace audio by default and the delay is ~0, however we can tell that this metric is working as intended by setting PacingController's pace_audio_ to true via the "WebRTC-Pacer-BlockAudio" field trial. In this case chrome://webrtc-internals/ plots neats graphs for audio send delay. Bug: webrtc:10635 Change-Id: Iecfd93bb84ec61e5d54232769a9e7a500601b199 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280523 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38483}
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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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