19c51ea537
This is a minor change for places where we use AudioFrame::kMaxDataSizeSamples sized intermediary buffers. The change uses `std::array<>` instead of C style arrays which allows for use of utility templates that incorporate type based buffer size checking. Also adding `ClearSamples()` method, which complements CopySamples. Bug: chromium:335805780 Change-Id: I813feb32937e020ceb9ca4b00632dc90907c93fb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351681 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42400}
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