Jakob Ivarsson 8da15c43dd Avoid depending on codec info for audio jitter stat.
The clock rate is already known by the RTP statistician.

Also included some minor code cleanup.

Bug: b/331602608
Change-Id: I335fa2a1cfd7dcceb286706d295a175a92f6797c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368920
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43436}
2024-11-21 18:47:29 +00:00
2024-11-12 10:04:10 +00:00
2024-10-31 15:31:38 +00:00
2024-11-14 14:36:58 +00:00
2021-01-20 15:01:07 +00:00
2022-02-20 14:22:13 +00:00
2022-12-02 09:21:47 +00:00
2023-09-25 15:56:09 +00:00
2024-05-27 12:46:57 +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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