Many of the tests follow a pattern of "wait for N candidates to be gathered, then (without waiting) assert that gathering is complete". But this only works if the "gathering complete" signal happens in the same task as the last candidate being gathered, which isn't an API guarantee. So the tests will be less fragile if they do the reverse: "wait for gathering to be complete, then (without waiting) assert that N candidates were gathered". Also fixing some somewhat unrelated issues elsewhere. Like a test that was supposed to be waiting for some period of time and ensuring no additional candidates were gathered, but wasn't actually waiting at all. BUG=None Review-Url: https://codereview.webrtc.org/3018493002 Cr-Commit-Position: refs/heads/master@{#19872}
Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ )
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.
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
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