Reason for revert: We will make it possible to link to BoringSSL for WebRTC's usages of the crypto APIs and OpenSSL for other usages in the same binary. Once that is completed, we will reland this. Original issue's description: > Remove code interfacing legacy openssl. > > BUG=webrtc:5664 > > Committed: https://crrev.com/4cd331beade6de16c073dcdaf89c4e038bdbf73f > Cr-Commit-Position: refs/heads/master@{#12041} TBR=tommi@webrtc.org,davidben@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:5664 Review URL: https://codereview.webrtc.org/1828773003 . Cr-Commit-Position: refs/heads/master@{#12117}
Revert of Added webrtc/base/safe_conversions.h as a pseudonym (patchset #1 id:20001 of https://codereview.webrtc.org/1774933003/ )
Revert of Revert opus memcheck suppression (patchset #1 id:1 of https://codereview.webrtc.org/1801233002/ )
Revert of Remove code interfacing legacy openssl. (patchset #3 id:40001 of https://codereview.webrtc.org/1808763002/ )
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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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