bd9031bf22
This is similar to Change-Id: I12a16f44f775da3711f3aa52a68a0bf24f70d2f8 but with the entire send buffer as scope, not a single stream. This can be used by clients to take alternate action (such as delaying transmission or using other buffering) if the send buffer ever becomes full, as they can now be notified when the send buffer is no longer full. Bug: webrtc:12794 Change-Id: Icf3be3b118888ffb5ced955fd7ba4826a37140f9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220360 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34143}
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