This removes use of the MockPeerConnection and replaces it with a fake implementation of PeerConnection tailored to the needs of StatsCollector and (soon) RTCStatsCollector. The stats collector tests really care about the PeerConnection only so much as to set up scenarios to test the StatsCollector with. Since each scenario (e.g., adding a track) affects the results of multiple methods (e.g., voice_channel and SessionStats), the tests were needing to manually configure these dependent operations which was tedious, error prone and difficult to change. The new fake lets the tests express the scenario more concisely (e.g., AddVoiceChannel) while filling in all the affected methods on the PeerConnection automatically. Furthermore, this can be expanded to use with the RTCStatsCollector and to cover more scenarios in the future. Bug: webrtc:8764 Change-Id: I195074174684256543f7cdc27c3834e5ff0b4702 Reviewed-on: https://webrtc-review.googlesource.com/43521 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21778}
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%