Henrik Boström 31c373b865 Only expose RTCCodecStats that are referenced by an RTP stream.
According to a pprof, creating RTCCodecStats is one of the places where
we spend the most CPU time in the event of creating hundreds of them:
https://screenshot.googleplex.com/B6QNDvvoX8dK5vk

The lifetime was recently updated so that we no longer have to risk
creating hundreds of them, here is the relevant section:
https://w3c.github.io/webrtc-stats/#codec-dict*

This allows code simplifications and the deletion of
ProduceCodecStats_n since we can now do a lazy instantiation of codec
stats at the point of being referenced.

Bug: webrtc:14444
Change-Id: I342c5bfebe6a4be0359da3ea106692c7a217779e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275763
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38209}
2022-09-26 15:03:20 +00:00
2022-08-29 21:04:32 +00:00
2022-09-15 14:08:22 +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
2022-09-26 10:59:50 +00:00
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2022-08-11 14:44:52 +00:00
2022-09-23 07:49:47 +00:00
2022-08-12 11:03:03 +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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