ae65b0e0d9
This CL adds a fuzzer for the audio processing module that exercises the handling of all user input sample rates logged by the UMA histogram WebAudio.AudioContext.HardwareSampleRate. The fuzzer inherits a lot of structure from the audio_processing_configs_fuzzer, but is greatly simplified and therefore the only shared code is test::FuzzDataHelper. Tested: Modified the build to explicitly trigger resampling issue and verified it exercises the code, then let an unmodified fuzzer run locally over the weekend without finding issues. Bug: webrtc:14263 Change-Id: Id3f19adee53c8842e92b6bf31cd2f360e19244d5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268192 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37540}
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