This refactoring takes a careful approach to avoid rushing the change: * stub headers are left in all the old locations of webrtc/base * existing GN targets are kept and now just forward to the moved ones using public_deps. The only exception to the above is the base_java target and its .java files, which were moved to webrtc/rtc_base right away since it's not possible to use public_deps for android_library. To avoid breaking builds, a temporary Dummy.java file was added to the new intermediate target in webrtc/rtc_base:base_java as well to avoid hitting a GN assert in the android_library template. The above approach should make the transition smooth without breaking downstream. A helper script was created (https://codereview.webrtc.org/2879203002/) and was run like this: stub-headers.py -s webrtc/base -d webrtc/rtc_base -i 7634 stub-headers.py -s webrtc/base/numerics -d webrtc/rtc_base/numerics -i 7634 Fixed invalid header guards in the following files: webrtc/base/base64.h webrtc/base/cryptstring.h webrtc/base/event.h webrtc/base/flags.h webrtc/base/httpbase.h webrtc/base/httpcommon-inl.h webrtc/base/httpcommon.h webrtc/base/httpserver.h webrtc/base/logsinks.h webrtc/base/macutils.h webrtc/base/nattypes.h webrtc/base/openssladapter.h webrtc/base/opensslstreamadapter.h webrtc/base/pathutils.h webrtc/base/physicalsocketserver.h webrtc/base/proxyinfo.h webrtc/base/sigslot.h webrtc/base/sigslotrepeater.h webrtc/base/socket.h webrtc/base/socketaddresspair.h webrtc/base/socketfactory.h webrtc/base/stringutils.h webrtc/base/testbase64.h webrtc/base/testutils.h webrtc/base/transformadapter.h webrtc/base/win32filesystem.h Added new header guards to: sslroots.h testbase64.h BUG=webrtc:7634 NOTRY=True NOPRESUBMIT=True R=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2877023002 . Cr-Commit-Position: refs/heads/master@{#18816}
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
Reland of PyLint fixes for tools-webrtc and webrtc/tools (patchset #1 id:1 of https://codereview.webrtc.org/2737233003/ )
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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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