Zhi Huang 2a4d70cc93 Make the DtlsSrtpTransport cache the RtpAbsSendTimeHeaderExtension.
Add a method to DtlsSrtpTransport to cache the RTP Absolute Send Time
extension id. The method would be called when using DTLS-SRTP with
external authentication.

TBR=pthatcher@webrtc.org

Bug: webrtc:7013
Change-Id: Ie9edb9382cbb4cf43eea5da3030991a0d20293a5
Reviewed-on: https://webrtc-review.googlesource.com/27260
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20947}
2017-11-30 02:17:09 +00:00
2017-11-22 09:12:18 +00:00
2017-09-15 04:25:06 +00:00
2017-09-15 04:25:06 +00:00
2017-09-15 04:25:06 +00:00
2017-09-15 04:25:06 +00:00

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

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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