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https://github.com/orottier/web-audio-api-rs

A Rust implementation of the Web Audio API, for use in non-browser contexts
https://github.com/orottier/web-audio-api-rs

audio dsp rust web-audio-api webaudio

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A Rust implementation of the Web Audio API, for use in non-browser contexts

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# Rust Web Audio API



[![crates.io](https://img.shields.io/crates/v/web-audio-api.svg)](https://crates.io/crates/web-audio-api)

[![docs.rs](https://img.shields.io/docsrs/web-audio-api)](https://docs.rs/web-audio-api)



A pure Rust implementation of the Web Audio API, for use in non-browser contexts



## About the Web Audio API



The [Web Audio API](https://www.w3.org/TR/webaudio/)

([MDN docs](https://developer.mozilla.org/en-US/docs/Web/API/Web_Audio_API))

provides a powerful and versatile system for controlling audio on the Web,

allowing developers to choose audio sources, add effects to audio, create audio

visualizations, apply spatial effects (such as panning) and much more.



Our Rust implementation decouples the Web Audio API from the Web. You can now

use it in desktop apps, command line utilities, headless execution, etc.



## Example usage



```rust,no_run

use web_audio_api::context::{AudioContext, BaseAudioContext};

use web_audio_api::node::{AudioNode, AudioScheduledSourceNode};



// set up the audio context with optimized settings for your hardware

let context = AudioContext::default();



// for background music, read from local file

let file = std::fs::File::open("samples/major-scale.ogg").unwrap();

let buffer = context.decode_audio_data_sync(file).unwrap();



// setup an AudioBufferSourceNode

let mut src = context.create_buffer_source();

src.set_buffer(buffer);

src.set_loop(true);



// create a biquad filter

let biquad = context.create_biquad_filter();

biquad.frequency().set_value(125.);



// connect the audio nodes

src.connect(&biquad);

biquad.connect(&context.destination());



// play the buffer

src.start();



// enjoy listening

std::thread::sleep(std::time::Duration::from_secs(4));

```



Check out the [docs](https://docs.rs/web-audio-api) for more info.



## Spec compliance



We have tried to stick to the official W3C spec as close as possible, but some

deviations could not be avoided:



- naming: snake_case instead of CamelCase

- getters/setters methods instead of exposed attributes

- introduced some namespacing

- inheritance is modelled with traits



## Bindings



We provide NodeJS bindings to this library over at

 so you can use this library

by simply writing native NodeJS code.



This enables us to run the official [WebAudioAPI test

harness](https://github.com/web-platform-tests/wpt/tree/master/webaudio) and

[track our spec compliance

score](https://github.com/ircam-ismm/node-web-audio-api/issues/57).



## Audio backends



By default, the [`cpal`](https://github.com/rustaudio/cpal) library is used for

cross platform audio I/O.



We offer [experimental support](https://github.com/orottier/web-audio-api-rs/issues/187) for the

[`cubeb`](https://github.com/mozilla/cubeb-rs) backend via the `cubeb` feature

flag. Please note that `cmake` must be installed locally in order to run

`cubeb`.



| Feature flag   | Backends                                                       |

| -------------- | -------------------------------------------------------------- |

| cpal (default) | ALSA, WASAPI, CoreAudio, Oboe (Android)                        |

| cpal-jack      | JACK                                                           |

| cpal-asio      | ASIO see    |

| cubeb          | PulseAudio, AudioUnit, WASAPI, OpenSL, AAudio, sndio, Sun, OSS |



### Notes for Linux users



Using the library on Linux with the ALSA backend might lead to unexpected

cranky sound with the default render size (i.e. 128 frames). In such cases, a

simple workaround is to pass the `AudioContextLatencyCategory::Playback`

latency hint when creating the audio context, which will increase the render

size to 1024 frames:



```rs

let audio_context = AudioContext::new(AudioContextOptions {

    latency_hint: AudioContextLatencyCategory::Playback,

    ..AudioContextOptions::default()

});

```



For real-time and interactive applications where low latency is crucial, you

should instead rely on the JACK backend provided by `cpal`. To that end you

will need a running JACK server and build your application with the `cpal-jack`

feature, e.g. `cargo run --release --features "cpal-jack" --example

microphone`.



### Targeting the browser



We can go full circle and pipe the Rust WebAudio output back into the browser

via `cpal`'s `wasm-bindgen` backend. Check out [an example WASM

project](https://github.com/orottier/wasm-web-audio-rs).

Warning: experimental!



## Contributing



web-audio-api-rs welcomes contribution from everyone in the form of

suggestions, bug reports, pull requests, and feedback. 💛



If you need ideas for contribution, there are several ways to get started:



- Try out some of our examples (located in the `examples/` directory) and start

  building your own audio graphs

- Found a bug or have a feature request?

  [Submit an issue](https://github.com/orottier/web-audio-api-rs/issues/new)!

- Issues labeled with

  [good first issue](https://github.com/orottier/web-audio-api-rs/issues?q=is%3Aissue+is%3Aopen+sort%3Aupdated-desc+label%3A%22good+first+issue%22)

  are relatively easy starter issues.



Unless you explicitly state otherwise, any contribution intentionally submitted

for inclusion in web-audio-api-rs by you, shall be licensed as MIT, without any

additional terms or conditions.



## License



This project is licensed under the [MIT license].



[mit license]: https://github.com/orottier/web-audio-api-rs/blob/main/LICENSE



## Acknowledgements



The IR files used for HRTF spatialization are part of the LISTEN database

created by the EAC team from Ircam.