Evan Shrubsole 2e2f67443e Add VideoStreamEncoder tests for DegredationPreference switching.
This test ensures that when changing degradation preference,
a resource that was previously downgraded in a different degradation
preference can not adapt up.

Bug: webrtc:11522, webrtc:11523
Change-Id: Id362530408db4c49b0d0b2516be9a11ccc7c8f37
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175012
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#31250}
2020-05-14 09:21:32 +00:00
2020-04-21 09:06:37 +00:00
2020-05-14 08:05:37 +00:00
2020-05-12 14:43:43 +00:00
2020-02-27 14:27:23 +00:00
2019-10-28 12:27:50 +00:00
.gn
2020-03-18 18:04:41 +00:00
2020-05-14 08:05:37 +00:00
2020-03-30 12:15:56 +00:00
2020-04-16 11:08:43 +00:00
2020-05-11 05:38:59 +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 here 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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