Philipp Hancke 1f80451932 Fix stats inheritance and rename RTP to Rtp
making RTCOutboundRtpStreamStats inherit from RTCSentRtpStreamStats
as defined in
  https://www.w3.org/TR/webrtc-stats/#outboundrtpstats-dict*

This removes the duplicated definitions of packetsSent and bytesSent.

BUG=webrtc:14948

Change-Id: I184998b65d59dbd0d1288733d55d8a884e6de970
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295660
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39481}
2023-03-06 13:43:27 +00:00
2023-01-31 10:14:47 +00:00
2023-02-13 10:30:38 +00:00
.gn
2022-09-14 08:49:56 +00:00
2022-02-20 14:22:13 +00:00
2021-12-08 08:53:00 +00:00
2022-12-02 09:21:47 +00:00
2022-12-02 09:21:47 +00:00
2022-05-13 09:01:34 +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
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