Dan Tan 43c0cf9cf8 Support borrowing of underused audio bitrate.
Controlled via added field trial WebRTC-ElasticBitrateAllocation.

Bug: webrtc:350555527
Change-Id: If57552144bd4a50421d618fd8bdab31d7c4afc35
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359506
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Dan Tan <dwtan@google.com>
Cr-Commit-Position: refs/heads/main@{#42834}
2024-08-23 07:44:45 +00:00
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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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