Åsa Persson e87cfe2315 Remove unused method PacketLossModeToStr.
Add method FrameType for frame to TestConfig.

Bug: none
Change-Id: Icfeb12fcb961559c9b36a3aedb081a840b9d8556
Reviewed-on: https://webrtc-review.googlesource.com/16120
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20458}
2017-10-27 08:51:27 +00:00
2017-10-25 11:51:36 +00:00
2017-10-27 08:24:37 +00:00
2017-10-26 15:11:36 +00:00
.gn
2017-09-25 15:34:41 +00:00
2017-10-27 08:32:47 +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
2017-10-24 16:14:18 +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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