Steve Anton 8c0f7a7a70 Add GetRemoteAudioSSLCertificate() to PeerConnection
This method allows the client to get details about the SSL
certificate sent by the remote side in the DTLS handshake.

This functionality in this new method has been standardized in the
RTCDtlsTransport, but until we have that implemented we wish to
expose this functionality so clients do not need to depend on
WebRtcSession.

Bug: webrtc:8323
Change-Id: Ic964266dd7e734cec07289a147fd8d090d74ce6b
Reviewed-on: https://webrtc-review.googlesource.com/5641
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20129}
2017-10-03 23:26:28 +00:00
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2017-10-03 13:39:55 +00:00
.gn
2017-09-25 15:34:41 +00:00
2017-10-03 13:39:55 +00:00
2017-09-15 04:25:06 +00:00
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2017-09-15 04:25:06 +00:00
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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.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

More info

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
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