When a TCP TURN port is destroyed, a TURN refresh request with zero lifetime is first sent to release the TURN allocation at the server, and the underlying TCP connection is closed afterwards. The closing of the TCP connection is handled first by the VirtualSocketServer in our test infrastructure, and the corresponding server socket is asynchronously destroyed at the TURN server. The refresh request is however still passed to this server socket and further signaled to the TURN server, which fails a DCHECK. The server implementation should disable any firing of signals from a server socket to be destroyed. The bug id is set to None since this is a one-liner CL. TBR=pthatcher@webrtc.org Bug: None Change-Id: Ib457b3800511a322ef69d67c71f2de05f3d67967 Reviewed-on: https://webrtc-review.googlesource.com/86501 Reviewed-by: Qingsi Wang <qingsi@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Commit-Queue: Qingsi Wang <qingsi@google.com> Cr-Commit-Position: refs/heads/master@{#23809}
Allows audio bitrate allocation in video calls without enabling TWCC (Transport Wide Congestion Control as defined at https://tools.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01.html) for audio stream.
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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 http://www.webrtc.org/native-code/development 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: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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