Zach Stein 3ca452be48 Create an RtpEncodingParameters struct for each simulcast stream
The additional structs are not used anywhere yet.

Bug: webrtc:8653
Change-Id: I8b3891e7f8d92286ffd43ea6010258a5828fa3b8
Reviewed-on: https://webrtc-review.googlesource.com/35007
Commit-Queue: Zach Stein <zstein@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21682}
2018-01-18 19:02:43 +00:00
2018-01-18 08:03:27 +00:00
2018-01-18 14:24:48 +00:00
.gn
2018-01-18 16:55:58 +00:00
2017-09-15 04:25:06 +00:00
2018-01-12 11:31:52 +00:00
2017-09-15 04:25:06 +00:00
2018-01-12 11:31:52 +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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