Log the DTLS handshake error code in OpenSSLStreamAdapter. Forward the error code to WebRTCSession with the Signals. This part is only for the WebRTC native code. To make it work, need another CL for Chromium. BUG=webrtc:5959 Review-Url: https://codereview.webrtc.org/2167363002 Cr-Commit-Position: refs/heads/master@{#13940}
Revert of CQ: Remove android_arm64_rel trybot (patchset #1 id:1 of https://codereview.webrtc.org/2270943003/ )
Revert of Delete method cricket::VideoFrame::Copy. (patchset #3 id:210001 of https://codereview.webrtc.org/2275243002/ )
Revert of Add field_trial_default dependency to libjingle_peerconnection (patchset #3 id:40001 of https://codereview.webrtc.org/2120673004/ )
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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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