Per K 98db63cfb6 Introduce RtpTransportConfig:allow_bandwidht_estimation_probe_without_media
If allow_bandwidht_estimation_probe_without_media is true and a writable
video rtp stream with RTX exist, a probe can be sent immediately without
waiting for a large media packet.

Bug: webrtc:14928
Change-Id: Ie2204734f9fe3e6bff9aed4a1f7f8995956d35cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336000
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41626}
2024-01-29 07:41:32 +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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