Per 83d0910694 Move Ownership of RtpModules to VideoSendStream from VieChannel and remove use of vie_channel and vie_receiver from video_send_stream.
The purpose of this refactoring is a first step of separating the encoder parts from the RTP transport.

BUG=webrtc:5687
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1864313003 .

Cr-Commit-Position: refs/heads/master@{#12377}
2016-04-15 12:59:21 +00:00
2014-06-17 08:54:03 +00:00
2015-09-11 09:04:09 +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. 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

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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