Yves Gerey 3caf50dbb8 Make ChangeBitrateVP9 unittest a bit more lenient.
This CL slightly increases rc_threshold.max_avg_bitrate_mismatch_percent
to account for libvpx change [1].
This will allow to unblock chromium roll [2].

[1] df7039cf9a
[2] https://webrtc-review.googlesource.com/c/src/+/126301/

Bug: webrtc:10349
Change-Id: Id9d8947a4f67e384f716dc73abb8c1cbbf2c6040
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126462
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27029}
2019-03-08 09:31:44 +00:00
2019-02-20 16:02:59 +00:00
2018-10-05 14:40:21 +00:00
.gn
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2017-09-15 04:25:06 +00:00
2018-12-18 12:30:58 +00:00
2018-11-09 14:23:59 +00:00
2017-09-15 04:25:06 +00:00
2018-07-23 15:28:48 +00:00
2018-07-23 15:28:48 +00:00
2017-09-15 04:25:06 +00:00
2019-01-31 22:09:16 +00:00
2018-02-23 10:34:16 +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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