Jakob Ivarsson e54914a79e Implement nack_count metric for inbound audio rtp streams.
Bug: webrtc:12925
Change-Id: I4542ca0f14a7dd7485ad5a2b6f2bd7051076f71f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224085
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34401}
2021-07-01 10:38:44 +00:00
2021-06-29 10:19:11 +00:00
2021-06-21 22:21:04 +00:00
2021-07-01 07:38:43 +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

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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