|
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| -**WebRTC is a free, open software project** that provides browsers and mobile |
2 |
| -applications with Real-Time Communications (RTC) capabilities via simple APIs. |
3 |
| -The WebRTC components have been optimized to best serve this purpose. |
| 1 | +# WebRTC-SDK |
4 | 2 |
|
5 |
| -**Our mission:** To enable rich, high-quality RTC applications to be |
6 |
| -developed for the browser, mobile platforms, and IoT devices, and allow them |
7 |
| -all to communicate via a common set of protocols. |
| 3 | +This repository contains a fork of WebRTC from Google with various improvements. |
8 | 4 |
|
9 |
| -The WebRTC initiative is a project supported by Google, Mozilla and Opera, |
10 |
| -amongst others. |
| 5 | +## Main changes |
11 | 6 |
|
12 |
| -### Development |
| 7 | +### All |
13 | 8 |
|
14 |
| -See [here][native-dev] for instructions on how to get started |
15 |
| -developing with the native code. |
| 9 | +- Dynamically acquire decoder to mitigate decoder limitations [#25](https://github.com/webrtc-sdk/webrtc/pull/25) |
| 10 | +- Support for video simulcast with hardware & software encoders [patch](https://github.com/webrtc-sdk/webrtc/commit/ee030264e2274a2c90548a99b448782049e48fb4) |
| 11 | +- Frame cryptor support (for end-to-end encryption) [patch](https://github.com/webrtc-sdk/webrtc/commit/3a2c008529a15fecde5f979a6ebb75c05463d45e) |
16 | 12 |
|
17 |
| -[Authoritative list](native-api.md) of directories that contain the |
18 |
| -native API header files. |
| 13 | +### Android |
19 | 14 |
|
20 |
| -### More info |
| 15 | +- WrappedVideoDecoderFactory [#74](https://github.com/webrtc-sdk/webrtc/pull/74) |
21 | 16 |
|
22 |
| - * Official web site: http://www.webrtc.org |
23 |
| - * Master source code repo: https://webrtc.googlesource.com/src |
24 |
| - * Samples and reference apps: https://github.com/webrtc |
25 |
| - * Mailing list: http://groups.google.com/group/discuss-webrtc |
26 |
| - * Continuous build: https://ci.chromium.org/p/webrtc/g/ci/console |
27 |
| - * [Coding style guide](g3doc/style-guide.md) |
28 |
| - * [Code of conduct](CODE_OF_CONDUCT.md) |
29 |
| - * [Reporting bugs](docs/bug-reporting.md) |
30 |
| - * [Documentation](g3doc/sitemap.md) |
| 17 | +### iOS / Mac |
31 | 18 |
|
32 |
| -[native-dev]: https://webrtc.googlesource.com/src/+/main/docs/native-code/ |
| 19 | +- Sane audio handling [patch](https://github.com/webrtc-sdk/webrtc/commit/272127d457ab48e36241e82549870405864851f6) |
| 20 | + - Do not acquire microphone/permissions unless actively publishing audio |
| 21 | + - Abililty to bypass voice processing on iOS |
| 22 | + - Remove hardcoded limitation of outputting only to right speaker on MacBook Pro |
| 23 | +- Desktop capture for Mac [patch](https://github.com/webrtc-sdk/webrtc/commit/8e832d1163644ab504412c9b8f3ba8510d9890d6) |
| 24 | + |
| 25 | +### Windows |
| 26 | + |
| 27 | +- Fixed unable to acquire Mic when built-in AEC is enabled [#29](https://github.com/webrtc-sdk/webrtc/pull/29) |
| 28 | + |
| 29 | +## LICENSE |
| 30 | + |
| 31 | +- [Google WebRTC](https://chromium.googlesource.com/external/webrtc.git), is licensed under [BSD license](/LICENSE). |
| 32 | + |
| 33 | +- Contains patches from [shiguredo-webrtc-build](https://github.com/shiguredo-webrtc-build), licensed under [Apache 2.0](/NOTICE). |
| 34 | + |
| 35 | +- Contains changes from LiveKit, licensed under Apache 2.0. |
| 36 | + |
| 37 | +## Who is using this project |
| 38 | + |
| 39 | +- [flutter-webrtc](https://github.com/flutter-webrtc/flutter-webrtc) |
| 40 | + |
| 41 | +- [LiveKit](https://github.com/livekit) |
| 42 | + |
| 43 | +- [Membrane Framework](https://github.com/membraneframework/membrane_rtc_engine) |
| 44 | + |
| 45 | +- [Louper](https://louper.io) |
| 46 | + |
| 47 | +Are you using WebRTC SDK in your framework or app? Feel free to open a PR and add yourself! |
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