To properly test the functionality, following changes were needed - Make RTCMTLVideoView compiliable for all cpu architectures not just arm64. This is needed so that the test can run on any device and on simulator as well. - Refactor RTCMTLVideoView to have mockable class methods. The unittest class, RTCMTLVideoViewTests was designed to provide easy transition to XCTest when the time comes for that. To transition to XCTest it would suffice to inherit from XCTestCase and remove the gtest methods. BUG=webrtc:7079 Review-Url: https://codereview.webrtc.org/2723903003 Cr-Commit-Position: refs/heads/master@{#17014}
Revert of Enable GN check for webrtc/examples (patchset #7 id:120001 of https://codereview.webrtc.org/2714343002/ )
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
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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. This page is maintained by the Google Chrome team.
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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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