Previously, it would be possible for the RTCStatsCollector to start an async network task to gather stats that would be run after the PeerConnection was closed when the transport controller was set to null. Bug: chromium:829238 Change-Id: I22fb4ce603caf2614814780b95b62127cee28284 Reviewed-on: https://webrtc-review.googlesource.com/72525 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23046}
…
…
…
…
…
…
…
…
…
…
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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Languages
C++
90.3%
Java
2.9%
C
2.2%
Objective-C++
2%
Python
1.3%
Other
1%