Per K 7d9cf9a537 Add PeerScenario test RampUpWithUndemuxableRtpPackets
Tests that caller BWE can rampup even if callee can not demux incoming RTP packets.

Bug: webrtc:14928
Change-Id: I3c89a14e67c6d781a26439980b5a99570a430dc7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299482
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39749}
2023-04-03 13:29:47 +00:00
2023-03-13 13:16:22 +00:00
2023-04-03 07:12:11 +00:00
2023-02-13 10:30:38 +00:00
.gn
2023-03-13 12:37:57 +00:00
2022-12-02 09:21:47 +00:00
2022-12-02 09:21:47 +00:00
2022-11-17 21:29:53 +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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