Markus Handell cb838e2c4e Move packets into RtpRtcpInterface and RtpSenderEgress.
This CL prepares for send packet batching support in later CLs.

Bug: chromium:1439830
Change-Id: I0bbecfa895aa6d4317ef8049b3789272a440d032
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304282
Auto-Submit: Markus Handell <handellm@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40009}
2023-05-08 12:39:54 +00:00
2023-05-04 08:03:29 +00:00
2023-05-03 12:56:39 +00:00
2023-04-27 09:53:00 +00:00
2023-05-03 12:56:39 +00:00
2023-05-03 12:56:39 +00:00
.gn
2023-03-13 12:37:57 +00:00
2023-04-12 07:09:41 +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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