dbf13e32ec
This allows mixing different number of streams depending on the client's capabilities. This CL adds `WebRTC.Audio.AudioMixer.NumIncomingActiveStreams2`, which is defined in [1], since the histogram is not logged anymore as enum. [1] https://chromium-review.googlesource.com/c/chromium/src/+/2883627 Bug: webrtc:12746 Change-Id: I0d9b3888f0f95269806539e33b56619b757a5c68 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218160 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Johannes Kron <kron@webrtc.org> Commit-Queue: Doudou Kisabaka <doudouk@google.com> Cr-Commit-Position: refs/heads/master@{#34024}
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