This code base aims at adding more support to to WebRTC. It is forked from libwebrtc, with sample commits to implement:

hardware-acceleration
system audio capture
The code only shows the concept and "just work" on windows, I do not have enough time to refactor it. :)

The following is readme.md from libwebrtc:

WebRTC-SDK

This repository contains a fork of WebRTC from Google with various improvements.

Main changes

All

Dynamically acquire decoder to mitigate decoder limitations #25
Support for video simulcast with hardware & software encoders patch
Frame cryptor support (for end-to-end encryption) patch

Android

WrappedVideoDecoderFactory #74

iOS / Mac

Sane audio handling patch
- Do not acquire microphone/permissions unless actively publishing audio
- Abililty to bypass voice processing on iOS
- Remove hardcoded limitation of outputting only to right speaker on MacBook Pro
Desktop capture for Mac patch

Windows

Fixed unable to acquire Mic when built-in AEC is enabled #29

LICENSE

Google WebRTC, is licensed under BSD license.
Contains patches from shiguredo-webrtc-build, licensed under Apache 2.0.
Contains changes from LiveKit, licensed under Apache 2.0.

Who is using this project

Are you using WebRTC SDK in your framework or app? Feel free to open a PR and add yourself!