| commit | 637bed5f8d90c6990687b4de03350609cf907783 | [log] [tgz] |
|---|---|---|
| author | Joachim Reiersen <joachimr@fb.com> | Thu Apr 25 10:41:35 2019 -0700 |
| committer | Commit Bot <commit-bot@chromium.org> | Thu Apr 25 21:12:57 2019 +0000 |
| tree | ad2256cce9e18c73a12c5433fc8ee25865a9c221 | |
| parent | e9145d65c106e3493c5f7b00c6e58c73536087e0 [diff] |
Add missing BoringSSL ifdef to OpenSSLStreamAdapter Compiling without BoringSSL fails since g_use_time_callback_for_testing is defined inside a OPENSSL_IS_BORINGSSL block. Bug: webrtc:10160 Change-Id: I25c27fa8ed128a50aa855db2012026c97954b91b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134226 Reviewed-by: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Qingsi Wang <qingsi@webrtc.org> Commit-Queue: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27779}
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.
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.