for consistency with usrsctp_dumppacket which prefixes its output with a newline. This makes the packets easier to grep and process with text2pcap. BUG=webrtc:12614 Change-Id: I67bc2e0026250b21b030daf967ebc697640f2d7e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220102 Reviewed-by: Victor Boivie <boivie@webrtc.org> Commit-Queue: Philipp Hancke <phancke@nvidia.com> Cr-Commit-Position: refs/heads/master@{#34114}
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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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