This code base aims at adding more support to to WebRTC. It is forked from libwebrtc, with sample commits to implement:
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hardware-acceleration
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system audio capture
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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:
This repository contains a fork of WebRTC from Google with various improvements.
- 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
- WrappedVideoDecoderFactory #74
- 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
- Fixed unable to acquire Mic when built-in AEC is enabled #29
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Google WebRTC, is licensed under BSD license.
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Contains patches from shiguredo-webrtc-build, licensed under Apache 2.0.
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Contains changes from LiveKit, licensed under Apache 2.0.
Are you using WebRTC SDK in your framework or app? Feel free to open a PR and add yourself!