Also renames "peerconnection_unittests" to "peerconnection_integrationtests", and moves the ICE URL parsing code to separate files. The main problem previously was that the test assertions occurred in various places in the main test class, and this shared test code was overly complex and stateful. As a result, it was difficult to tell what a test even does, let alone what assertions it's meant to be making. And writing a new test that does what you want can be a frustrating ordeal. The new code still uses helper methods, but they have intuitive names and a smaller role; all of the important parts of the test's logic are in the test case itself. We're planning on merging PeerConnection and WebRtcSession at some point soon, so it seemed valuable to do this, so that the WebRtcSession tests can be rewritten as PeerConnection tests using better patterns. BUG=None Review-Url: https://codereview.webrtc.org/2738353003 Cr-Commit-Position: refs/heads/master@{#17458}
Revert of CQ: Remove ARM64 trybots until toolchain issues are resolved. (patchset #1 id:1 of https://codereview.webrtc.org/2776393005/ )
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
Revert of PyLint fixes for tools-webrtc and webrtc/tools (patchset #3 id:40001 of https://codereview.webrtc.org/2736233003/ )
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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.
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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