Jakob Ivarsson e91c992fa1 Implement nack_count metric for outbound audio rtp streams.
Bug: webrtc:12510
Change-Id: Ia035885bced3c3d202bb9ffeb88c2556d4830e92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225021
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34444}
2021-07-09 13:29:10 +00:00
2018-10-05 14:40:21 +00:00
2021-06-21 22:21:04 +00:00
2021-07-09 07:49:43 +00:00
2021-07-09 07:49:43 +00:00
2021-01-20 15:01:07 +00:00
2021-04-26 16:39:07 +00:00
2020-07-13 11:42:07 +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

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