There can be error log each frame when maximum playout delay sent with the frame exceed delay derived from the av-sync. In such scenario prefer to limit the playout delay by the one attached to the received frame. Bug: b/390043766 Change-Id: Ia57969df72f7a649e5a9280d5bb29986f5ea14b7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374284 Reviewed-by: Johannes Kron <kron@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43814}
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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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