This also deletes unused features of the video_capturer interface, the classes VideoCaptureFeedBack, VideoCaptureEncodeInterface and related methods, and the module id which used to be passed as an argument to the VideoCaptureDataCallback. In theory the module id could have been used to let a single VideoCaptureDataCallback serve several capturers, and demultiplex on the id, but in practice, it was unused. With this change, it is required to use a separate VideoSinkInterface for each capturer. BUG=webrtc:6789 Review-Url: https://codereview.webrtc.org/2534553002 Cr-Commit-Position: refs/heads/master@{#15540}
Revert of CQ: Disable android_more_configs trybot (patchset #1 id:1 of https://codereview.webrtc.org/2522953003/ )
Set OPENSSL_EC_NAMED_CURVE explicitly on EC key so that certificate has ASN1 OID and NIST curve info. Without this openSSL handshake negotiation fails throwing NO_SHARED_CIPHER error. the change made is along the lines of openssl behavior documented here: https://wiki.openssl.org/index.php/Elliptic_Curve_Diffie_Hellman#ECDH_and_Named_Curves
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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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