3354157d36
The existing restriction of max 48k seems old and outdated. I am unable to see any issues by simply extending the support to 96 and utilize the existing resampler in WebRTC. There are no memory limitations involved either. It is a rather common case today in Chrome that users need 96k/192k input; hence this simple change will have a positive impact for many WebRTC clients using gUM. Bug: webrtc:10958 Test: https://webrtc.github.io/samples/src/content/peerconnection/audio/ using mic @96k Change-Id: I8123da886ef7d48cbec9482795ec837ec1f61d81 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152162 Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29135}
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