b1a15d7711ffd419acd7dea3d307bf22f33ecba3
Previously the helper methods (like "CreateLocalAudioTrack") were using hard-coded IDs. This means if you try to add two tracks to the same PeerConnection you'll get an ID conflict. Tests were getting around this by using APIs to explicitly add tracks with different IDs. But this leaks an implementation detail of the helper methods, and is a hassle. So instead, just use random UUIDs from the helper methods. This is how IDs are created in the real world, so it's pretty reasonable and allows the tests to be made more readable. BUG=None Review-Url: https://codereview.webrtc.org/3011713002 Cr-Commit-Position: refs/heads/master@{#19736}
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://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
- Coding style guide
- Code of conduct
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