Steve Anton 75737c0c6a Merge WebRtcSession into PeerConnection
This literally copies & pastes the code from WebRtcSession into
PeerConnection as private methods. The only other changes were to
inline the WebRtcSession construction/initialization/destruction
into PeerConnection and fix issues using rtc::Bind on the
reference-counted PeerConnection.

Bug: webrtc:8323
Change-Id: Ib3f071ac10d18566a21a3b04813b1d4ec691ef3c
Reviewed-on: https://webrtc-review.googlesource.com/15160
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20574}
2017-11-06 21:15:20 +00:00
2017-10-31 21:08:43 +00:00
.gn
2017-09-25 15:34:41 +00:00
2017-09-15 04:25:06 +00:00
2017-09-15 04:25:06 +00:00
2017-10-31 17:46:42 +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.

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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