Jerome Jiang 3cc1a6509b Set av1 speed from resolution.
Use speed 6 for better quality for low resolution, speed 8 for HD for better speed.
This will better balance speed and quality.

Change-Id: I3d8dbd45533471ce58d53c1ac26f92c7b1106259
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175281
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jerome Jiang <jianj@google.com>
Cr-Commit-Position: refs/heads/master@{#31336}
2020-05-20 20:06:46 +00:00
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2020-05-20 20:06:46 +00:00
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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
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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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