There is no need to redefine SSL_CTX. base.h/ossl_typ.h defines it already. Additionally, switch the base.h includes to the OpenSSL-compatible ossl_typ.h spelling. That just got landed in https://webrtc-review.googlesource.com/c/104120, so I'm guessing OpenSSL consumers just didn't notice yet. While getting the current BoringSSL name mangling scheme working with WebRTC is a ways off, one of the requirements will almost certainly be that WebRTC never forward-declare any BoringSSL types itself, instead leaving it to openssl/base.h (or openssl/ossl_typ.h, the OpenSSL-compatible alias). This is because we'd need to rename the struct names themselves where they participate in C++ name mangling. E.g. std::pair<RSA*, int> would mangle as rsa_st. Bug: webrtc:5664 Change-Id: Ib9695d4ae4bc07d2bc54c9fdfb8600f44b5ec7bb Reviewed-on: https://webrtc-review.googlesource.com/c/106675 Commit-Queue: David Benjamin <davidben@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25257}
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.
Authoritative list of directories that contain the native API header files.
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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