Taylor Brandstetter 4770fd935a Move JsepTransport from p2p/base to pc/.
The JsepTransport class is moved to pc/ and the utility methods and
enums are moved to where they are used.

With JsepTransport moved to pc/, JsepTransport can depend on objects in
pc/ including RtpTranport, SrtpTransport etc.

Forked from https://webrtc-review.googlesource.com/c/src/+/31762/7

Bug: webrtc:8636
Change-Id: I4e8569fe3012946e87deb280f6139f0fd98de34d
Reviewed-on: https://webrtc-review.googlesource.com/33701
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21333}
2017-12-18 18:59:43 +00:00
2017-12-15 16:48:57 +00:00
2017-11-22 09:12:18 +00:00
2017-12-18 18:08:43 +00:00
2017-12-18 15:20:59 +00:00
2017-12-12 14:32:56 +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
Languages
C++ 90.3%
Java 2.9%
C 2.2%
Objective-C++ 2%
Python 1.3%
Other 1%