When a large queue of frames builds up due to a lost frame, the decode delay can sometimes become quite large. In this case the stream may signal as timed out when in fact it is not. Instead, the delay should be capped at the timeout limit. Bug: webrtc:14168 Change-Id: I5b4e8851b2c6d7d27a698627dc1633931d7fc00e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265404 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37199}
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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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