Does anyone has experienced any issues on Macos? I have set the frequency to exactly 562 hz with a detune of exactly -700 cents which should result in a perfectly steady sine wave. It's a software issue as the windows version is running in a VM and has no problems. The waveform seems to flip every other frame and I don't know why.

This is the visualizer on Windows:

Windows 11

And this is the visualizer on macos:

MacOS Sonoma

​

Just in case anyone will find this useful in their own projects… I wanted to promote a package I’ve published called earwurm:

• GitHub repo
• Demo site

I know there are already competent alternatives in this space, so to quickly summarize the purpose of this specific package:

Earwurm is an opinionated and minimal-scope solution for loading short audio files via the Web Audio API. Intended for playback of UI sound effects in modern browsers.

Minimal React example:

import {Earwurm, type LibraryEntry} from 'earwurm';

const entries: LibraryEntry[] = [
  {id: 'beep', path: 'assets/beep.webm'},
  {id: 'zap', path: 'assets/zap.mp3'},
];

const manager = new Earwurm();
manager.add(...entries);

// Optional: pre-fetch/decode each asset ahead of time
// so the browser is ready for immediate playback.
entries.forEach(({id}) => manager.get(id)?.prepare());

async function handlePlaySound(id = '') {
  const stack = manager.get(id);
  const sound = await stack?.prepare();

  sound?.play();
}

function Page() {
  return (
    <div>
      <button onClick={() => handlePlaySound('beep')}>Play beep</button>
      <button onClick={() => handlePlaySound('zap')}>Play zap</button>
    </div>
  );
}

An example of the above code can be tinkered with in this CodeSandbox. Better yet, the source code for the Demo site is included in the repo.

Earwurm doesn’t try to solve for every use case, and is instead limited to what I believe is an expected set of patterns for UI sound effects.

That being said, I do consider this an MVP. There are other features I intend to add in the future, such as the ability to reverse playback, as well as adjust pitch. All of this is building towards having a tool that empowers me build richer user experiences.

So, just in case anyone else finds this interesting, please give it a shot and feel free to report any issues you may encounter! 😁

Please forgive me I know this is going to be long and hopefully not too redundant but I have been battling this issue for almost two years and I swear there is no solution or it's the easiest thing that I keep stepping over. I would be forever in your debt for a miracle.

So I'm a piano player and aside from Instagram Youtube the usual I love going on omegle because it's face to face. For the past and I swear I'm not kidding close to two years I have been having this audio issue that for the life of me I can't figure out and I am pretty technically savvy. If I stream through obs to a specific streaming site like youtube or Instagram my sound is great aside from usual hums and buzzes that I'm trying to get rid of between my yamaha Electric keyboard and computer. When you are streaming on omegle you really aren't streaming on omegle you are using OBS as a virtual camera and as a virtual audio you know what I mean. Except the differences you are going through an Internet browser and you just tell the browser which virtual stuff to use botta Bing bang everything should be good Pretty much how Most streamers do it from what I understand. The issue I have when I am using my live setup which is my keyboard and microphone Which we know works correctly because it works when I stream outside of an Internet browser but when I go through the browser my sound just cuts off after a few seconds slowly like it's being muffled and then if I stop for a while it's like the noise suppression releases and I can play a few more notes on the piano and then it slowly dies out again into a weird muffily every 3rd 4th note is heard until I stop and then you can literally hear the usual static return so you know you are at full volume if you understand what I'm saying. When I say I have tried everything I mean all the way to fresh install of windows installing a sound blaster sound card all of the usual windows sound settings let things take exclusive control don't let them take exclusive control the communications tab to do nothing but ultimately because it works when I am not using a browser I know it's not a hardware issue 99% because I also bought a brand new computer and have nothing on it but my streaming stuff and I have gone into real tech audio control panel made sure my steam account wasn't doing anything mature my NVIDIA RTX 4070 isn't suppressing anything but again because If I try and directly go through an Internet browser my sound gets suppressed. Because I watch Harry Mac and other streamers so much I keep saying to myself he can do it so I know other people do it so I No it can be done. I have tried the top three Browsers and I have gone into the flag experimental settings and shut off any kind of audio interference and I even found an extension that I tried because here's what physically happens sort of with my setup. After getting the new computer going on for the first time praying that it was going to work I noticed as soon as I went on the browser and the browser took control of my microphone and camera because it gets permission that my microphone volume as soon as I played a note on my piano would automatically slowly increase all the way up to 100 so this leads to some kind of conclusion that when I'm going online it doesn't think I'm loud enough because it starts suppressing me and then after it suppresses my sound it tries to make my volume higher which in turn makes it suppress my sound. Forgive me if I sound like a crazed maniac but I have tried so many things that it literally is the number one thorn in my side and has been for almost two years. Because I have two computers I can test it by going on omegle with myself using a crazy keyword But then I started thinking I'm on the same Internet line maybe it's bouncing back and forth between my two computers but it literally happens if I do a microphone test online I can see all of the bars start to get lower and lower and lower on the little equalizer thing. My God if you are reading all of this I'm already grateful because I just feel so alone in this. I'm to the point where I think it's my AT&T fiber doing something because there are only so many common denominators now that I'm using a new computer but something is suppressing my sound and I would literally pay somebody $100 to help me and that's a lot of money to me I promise. I know you guys know everything I guess I have the option of just giving up but maybe somebody would go on omegle with me and test it we just have to put in a keyword like submarine or something off the wall but we would have to time it. Sorry to be a pain But at least you would hear what I'm talking about. I hear other people play music and piano and synthesizers and violins LOLI can go on and on I know other people do it and I am a pretty smart dude but something is choking my sound and I am on my hands and knees begging for help. Ohh wise Internet gods hear me please send me my magic wand that I am so desperate for. Barry.

I’m writing a radio app. The hardware is on a local network. I have a socketio server in python that collects raw audio PCM data which is sent via socket to a client along with power spectrum data to be rendered as sound and real-time power spectrum display. The client is an electron app using electron-forge. To render sound I use BufferSource and AudioBuffers. All this works great except the Audio API buffers. It slowly builds up a cache of PCM data. It’s a small but annoying effect. After several minutes I typically have several seconds of buffer. First off the power spectrum display is out of sync with the sound which I could likely fix by buffering that data as well. That aside, how can I limit the buffering of sound to be less than say 0.2 seconds. Anything less than a second would be great.

I'm a startup founder building a prototype generative music app using Web Audio. I would love to hear your advice about where to start from programming standpoint: pure Javascript? Use libraries like Tone.js + Tuna.js, Howler.js, Wad.js, XSound? I'd prefer to well-supported libraries. I'm happy to trade some functionality or performance for stability.

The app we are building needs to support sample loop playback, SoundFont libraries, simple subtractive synthesis (not Serum-quality), and real-time audio processing (Tuna.js appears to have everything we need). Ideally, it would be efficient enough to playback 8 stereo 44.1Khz samples simultaneously with Tuna effects in Safari/Firefox on MacOS, or ideally, Safari on iOS on a <3 year old iPhone.

I would very much appreciate your advice. And, DM me if you are interested in freelance work.

Thanks in advance.

It seems to be rather lame question after looking at posts here, butI'm trying to make cue-point functionality in my web app so I can seamlessly switch to another time of track

I've found this post on stackoverflow: https://stackoverflow.com/questions/59815825/how-to-skip-ahead-n-seconds-while-playing-track-using-web-audio-api

But yet it looks complicated to me. I use HLS, and looks like it makes things even more complex... Can you give any advice on this?

up. seems hls was actual problem in ios safari

Hello
I search partners create an online service for musicians, for example a system to retrieve the acapella on a piece.

Motivated people?

If so leave a message with your experience.

Lots of experience in audio processing with ableton live.

3 years of experience in javascript.

Lets say I have a chain of FX nodes between a source and a destination. I want to dynamically insert or remove nodes at runtime. What is the correct way to approach this? Do I need to basically maintain saved state of the chain, destroy it and reconnect everything? It's a bit unclear to me, thanks.

I want to play many audio channels ( around 16) that synced to a video (maybe around 60s) . And I want to be able to mute or unmounted them like a mixer? Is their any way to do this?

This is the first in a short series of tutorials on how to do this:
https://www.johncotterell.me/2023/01/tutorial-creating-scratchable-audio.html

Shortcut to a working example here:
https://www.johnc.pro/scratchableaudio/

And the original proof of concept here:
https://www.johnc.pro/boris_scratch/

I am making a little web browser looper thingy, and I have been attempting to use .ogg files.

I was using .wav in my prototyping, and shipped it today, only to discover iOS doesn't support .ogg files :/

any ideas are welcome.

Hi

I'm wondering if someone here can help me. I have a web app that generates and plays back rhythmic musical notation and I have been trying to build "play along mode". e.g. the user clicks/taps to play along during playback and their accuracy is assessed along the way.

In order to achieve this I am measuring the currentTime of the Web Audio API AudioContext at each user interaction, resolving this to a position within the measure(s) of music.

No matter how I build it I can't seem to achieve 100% accuracy. The milisecond difference between the notes I'm trying to detect is often very small and the problem seems to be the latency caused by JS event handlers such as 'onClick', 'onPointerDown' etc.

The interaction is always read slightly late and inconsistently late each time so that I can't reliably account for this latency.

Here is a codesandbox link recreating the issue using Tone JS: https://codesandbox.io/s/festive-sound-w7zz22?file=/src/App.js

I'd realy appreciate any help, thanks!

I have an app that uses web audio api heavily, and I am always struggling with how to test it on an efficient way.

I am using jest for unit test and cypress for integration tests but they always fell short. I have to mock too many stuff on jest for web audio, and in cypress I have no way to know what an audio event is really happening as I expect it happens.

How are you testing your code?

I am trying to find the frequency data in an audio file. I have used the analyzer node for this.

const element = document.getElementById("track")

const AudioContext = window.AudioContext
const audioContext = new AudioContext()

const track = audioContext.createMediaElementSource(element)
const analyzer = audioContext.createAnalyser()

track.connect(analyzer)
let dataArray = new Uint8Array(analyzer.frequencyBinCount);
analyzer.getByteFrequencyData(dataArray);

When I print out the contents of dataArray, I just get zeros. If I use the floatFrequencyData then I get an array of negative infinity values. According to the Mozilla documentation, this means that the audio source is silent. The max decibels for this sample is -30db. I have been able to play the file and that works fine. Any ideas on why I am not able to get the frequency data?