daef292e03
* git merge 04cb763 * See all upstream changes since the previous merge in branch aosp/upstream-master: git diff cb3f9bd..04cb763 * Modify webrtc/.gitignore to keep *.mk files. * Removed old files from *.mk files: - thread.cc - thread_posix.cc * Add new files to *.mk files: - event_tracer.cc * Android relevant upstream changes: - Make Beamforming dynamically settable for Android platform builds - Remove additional channel constraints when Beamforming is enabled in AudioProcessing - Use an explicit identifier in Config Change-Id: I384a4e8f6982c31c5bc70eef521bb2d90800b9a4
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. This page is maintained by the Google Chrome team.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- 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