This CL adds all the scaffolding needed to support xctest target in sdk/objc/Framework and adds the target to the bot configuration. The benefits of this are two-fold. 1. We'll separate framework unittests from their current target, `rtc_unittests`, that has many many other tests. This way framework unit tests will have nice, compact, selfcontained target. 2. We'll harvest the power of XCTest (native testing framework) that should hopefully make adding and writing objc tests easier. This CL migrates only one test to prove the setup works. More tests will be migrated in follow up cls. Bug: webrtc:8382 Change-Id: I0b5b9596c2a6d91683d13632323441de1aa461e0 Reviewed-on: https://webrtc-review.googlesource.com/8501 Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Reviewed-by: Daniela Jovanoska Petrenko <denicija@webrtc.org> Commit-Queue: Daniela Jovanoska Petrenko <denicija@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20289}
Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ )
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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