Harald Alvestrand ecc38d8d29 Take out callback that modifies voice receive codec based on send codec
No functionality that depends on this information has been identified; no tests break when it is taken out.

Bug: webrtc:15224
Change-Id: I37298479c6b8a4acb82f59d32130c105371936b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324760
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41037}
2023-10-30 12:36:29 +00:00
2023-10-27 16:04:31 +00:00
2023-06-02 07:49:24 +00:00
2023-10-26 12:03:02 +00:00
2023-10-27 22:00:50 +00:00
2023-09-25 15:56:09 +00:00
2023-05-16 08:24:54 +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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