Erik Språng d621b419a3 Make WebRTC-Video-AV1EvenPayloadSizes default-on.
Experimentation shows only a slight increase in bitrate due to improved
BWE. There's no negative side-effects we have been able to see so far.

This CL flips the experiment to default-on but is kept around as a
kill-switch until the next milestone just in case something unexpected
is discovered.

Bug: webrtc:42226301
Change-Id: I4a0b1c85e912b909d7bff58d78966cf161857f7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374182
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43773}
2025-01-20 07:23:58 -08:00
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2024-11-12 10:04:10 +00:00
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2025-01-08 08:39:49 -08:00
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2022-12-02 09:21:47 +00:00
2023-09-25 15:56:09 +00:00
2024-05-27 12:46:57 +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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