Philipp Hancke 0322493aed Refactor MediaSession to unify audio/video codec handling
since the offer/answer rules do not depend on the media type for
the most part. Also make use of recently introduced Codec types.

BUG=webrtc:15214

Change-Id: Ieae27247a8910c3fcaa9609dca0297985907f86a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327740
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41221}
2023-11-23 14:27:54 +00:00
2023-11-08 15:49:37 +00:00
2023-11-17 14:36:35 +00:00
2023-10-30 14:56:36 +00:00
2021-01-20 15:01:07 +00:00
2022-02-20 14:22:13 +00:00
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2022-12-02 09:21:47 +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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