Tony Herre 9c6874607a Consolidate encoded transform mocks into api/test/
Includes removing the duplicate MockTransformableAudioFrame definition
in test/ in favour of the existing one in api/test/

Bug: webrtc:15802
Change-Id: Ib5f86b8b2095dd4e580cd9ff0038134f8a43cd93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336340
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41622}
2024-01-26 12:46:34 +00:00
2024-01-12 16:39:54 +00:00
2023-10-30 14:56:36 +00:00
2022-02-20 14:22:13 +00:00
2023-11-27 11:44:50 +00:00
2022-12-02 09:21:47 +00:00
2023-09-25 15:56:09 +00:00
2023-11-27 11:44:50 +00:00
2024-01-09 13:32:42 +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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