Philipp Hancke ae566cd831 audio/red: provide default fmtp line
otherwise the generated codec won't match the preassigned codec
and red will use 96 as payload type, increasing the payload type
congestion in the upper range.

BUG=webrtc:11640

Change-Id: I466ed6d4e025ef116f3099e85855e10493408ab1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233560
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35130}
2021-10-01 13:02:30 +00:00
2021-08-23 19:52:17 +00:00
2021-09-27 11:14:35 +00:00
2021-01-20 15:01:07 +00:00
2021-07-22 16:41:26 +00:00
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2021-09-15 16:56:30 +00:00
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2020-07-13 11:42:07 +00:00
2021-08-23 13:37:55 +00:00
2021-09-24 20:09:34 +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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