This will be used by third_party/blink/renderer/platform/peerconnection/webrtc_video_track_source.cc to provide capture_time_identifier_ms_ from media::VideoFrame. This identifier would then be passed to webrtc::EncodedFrame and webrtc::TransformableVideoSenderFrame (in the future CLs) to be used as an identifier for encoded frames. Bug: webrtc:14878 Change-Id: I1d8a27891323d86fdc2f014988a8da572df84119 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291805 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Palak Agarwal <agpalak@google.com> Cr-Commit-Position: refs/heads/main@{#39270}
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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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