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min_pacing:8ms, to avoid the situation where bursts of frames are sent to the decoder at once due to network jitter. The bursts of frames caused the queues further down in the processing to be full and therefore drop all frames. max_decode_queue_size:8, in the event that too many frames have piled up, do as before and send all frames to the decoder to avoid building up any latency. These setting only affect the low-latency video pipeline that is enabled by setting the playout RTP header extension to min=0ms, max>0ms. Bug: chromium:1138888 Change-Id: I8154bf3efe7450b770da8387f8fb6b23f6be26bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233220 Commit-Queue: Johannes Kron <kron@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35119}
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