WebAudio | Web Audio API Playground | Audio Utils library

by cwilso JavaScript Version: Current License: MIT

X-Ray Key Features Code Snippets Community Discussions(10)Vulnerabilities Install Support

kandi X-RAY | WebAudio Summary

WebAudio is a JavaScript library typically used in Audio, Audio Utils applications. WebAudio has no bugs, it has no vulnerabilities, it has a Permissive License and it has low support. You can download it from GitHub.

Web Audio API Playground

Support

Quality

Security

License

Reuse

Support

WebAudio has a low active ecosystem.

It has 428 star(s) with 107 fork(s). There are 39 watchers for this library.

It had no major release in the last 6 months.

There are 8 open issues and 3 have been closed. On average issues are closed in 144 days. There are no pull requests.

It has a neutral sentiment in the developer community.

The latest version of WebAudio is current.

Quality

WebAudio has 0 bugs and 0 code smells.

Security

WebAudio has no vulnerabilities reported, and its dependent libraries have no vulnerabilities reported.

WebAudio code analysis shows 0 unresolved vulnerabilities.

There are 0 security hotspots that need review.

License

WebAudio is licensed under the MIT License. This license is Permissive.

Permissive licenses have the least restrictions, and you can use them in most projects.

Reuse

WebAudio releases are not available. You will need to build from source code and install.

WebAudio saves you 221 person hours of effort in developing the same functionality from scratch.

It has 540 lines of code, 0 functions and 9 files.

It has low code complexity. Code complexity directly impacts maintainability of the code.

Top functions reviewed by kandi - BETA

kandi has reviewed WebAudio and discovered the below as its top functions. This is intended to give you an instant insight into WebAudio implemented functionality, and help decide if they suit your requirements.

Load sound data .
Create audio source source
Stop the connector .
Start a new drag connector .
Creates filter .
Creates a new module element .
Mousedown handler for a connector .
Moves the drag node in the document .
This function is called when the connector has connected to the sources .
Start dragging node

Get all kandi verified functions for this library.

WebAudio Key Features

No Key Features are available at this moment for WebAudio.

WebAudio Examples and Code Snippets

No Code Snippets are available at this moment for WebAudio.

Community Discussions

Trending Discussions on WebAudio

What is an audio block?

DevTools failed to load source map

WebAudio API change volume for one of sources

Webaudio timing performance

Using wavesurfer.js in a react app cause issues

The AudioContext was not allowed to start. It must be resumed (or created)

AudioContext, getUserMedia, and websockets audio streaming

Django and JavaScript: Using last uploaded file in an object list within in a Javascript file

Django- How to get last uploaded file into template?

How to play audio without strict limitations on JavaScript?

QUESTION

What is an audio block?

Asked 2022-Feb-27 at 18:33

I've seen some sentences including a word 'block'.

Pd lightens its workload by working with samples in blocks rather than individually. This greatly improves performance. The standard block size is 64 samples, but this setting can be changed. (http://pd-tutorial.com/english/ch03.html)

Rendering an audio graph is done in blocks of 128 samples-frames. A block of 128 samples-frames is called a render quantum, and the render quantum size is 128. (https://www.w3.org/TR/webaudio/#rendering-loop)

So, what I wonder are:

(1) What is wrong with handling samples individually? Why audio samples are grouped by a block of a some size (64, 128) ?

(2) Why the block size is a power of 2? // 2^6 = 64, 2^7 = 128

(3) After grouped, to where the samples go? Are they then played by a sound card or something?

...

ANSWER

Answered 2022-Feb-27 at 03:22

A audio block is an array of floating point numbers representing audio. Where the numbers range from [1, -1] and where 0 (not 0.xxxx) is no sound.

1. What is wrong with handling samples individually? Why audio samples are grouped by a block of a some size (64, 128) ?

From my understanding handling samples in frames is better because of how web audio api runs in the browser due to performance. Different frame sizes can vary on the users performance on their PC. This is similar to fps on a video. In web audio there is node called ScriptProcesssorNode which allows you to create custom audio processing in a event handler at specific buffer/frame size. However the buffer size only ranged from 256 - 16384 or undefined for system preferred. It's simple on each function call it contains a new frame of audio. Basically it's a loop.

Now these days the ScriptProcessorNode is now deprecated/obsolete because of bad performance. This is because of the event handler onaudioprocess function is on the main thread which can block a lot of things. This is replaced with a AudioWorklet which is similar to the ScriptProcessorNode but each process call is only 128 frames and save better performance on the main thread because the AudioWorklet runs on the worker thread in background.

2. Why the block size is a power of 2? // 2^6 = 64, 2^7 = 128

I honestly don't exactly know but my best guess is that it is easier to calculate since in 8 bit alike numbers such as 8, 16, 32, 64, 128, 256 etc.

3. After grouped, to where the samples go? Are they then played by a sound card or something?

Web audio api is node graph system where sample frames (a.k.a 128 frame block) is passed through different nodes such as effects like BiquadFilterNode or custom processing blocks like AudioWorkletNode, etc. There is a node called AudioDestinationNode which is the speaker hardware output of the users PC. Any connection is connected to this particular node, (if there is sound), It will produce sound from that connection. Think of these nodes as a connect the dots with point A is the start and point B is the end. Each dot is a processing block like the AudioWorklet, etc, where point A is the source like a mic or a mp3/wav file and point B is the speaker destination. Connect them together, boom! You got amazing sounds.

I hope this makes sense and what your looking for :)
Feel free to correct me if have it wrong.

Source https://stackoverflow.com/questions/71215976

QUESTION

DevTools failed to load source map

Asked 2022-Feb-05 at 08:12

I'm aware of many posts regarding these warnings in Chrome dev tools. For all of them, the solution is to turn off notifications "Enable javascript source maps" and "Enable CSS source maps".

What I want to know is how to FIX this and what is the reason under the hood that causes these warnings.

I'm currently developing a Vue JS app that uses Twilio Js SDK and I'm getting tons of warnings when the app is built in stage mode by using sudo npm run build -- --mode staging

Any advice will be appreciated.

...

ANSWER

Answered 2022-Feb-05 at 08:12

Twilio developer evangelist here.

Do you need to turn on sourcemaps in webpack, like in this GitHub issue?

Source https://stackoverflow.com/questions/70994328

QUESTION

WebAudio API change volume for one of sources

Asked 2021-Dec-25 at 21:13

I'm reading this article

My goal is to play two sounds at the same time. One sound is in a different volume. Having regular "audio" tags is not a solution because it's not working well on mobile devices. So I started to dive into Web Audio API.

I wrote the code below, that works well across all devices. The single issue - I can't figure out how to control the volume. Code from example is not working(

Please help 🙏

...

ANSWER

Answered 2021-Dec-25 at 21:13

The connect() method returns an AudioNode, which must then have connect() called on it so the Nodes are chained together:

Source https://stackoverflow.com/questions/70480176

QUESTION

Webaudio timing performance

Asked 2021-Nov-02 at 12:49

The file below uses ToneJS to play a steam of steady 8th notes. According to the log of the timing, those 8th notes are precisely 0.25 seconds apart.

However, they don't sound even. The time intervals between the notes are distinctly irregular.

Why is it so? Is there anything that can be done about it? Or is this a performance limitation of Javascript/webaudio-api? I have tested it in Chrome, Firefox, and Safari, all to the same result.

Thanks for any information or suggestions about this!

...

ANSWER

Answered 2021-Nov-02 at 12:49

For a scheduled triggerAttackRelease, you should pass the time value as the third argument.

Source https://stackoverflow.com/questions/69804550

QUESTION

Using wavesurfer.js in a react app cause issues

Asked 2021-Oct-17 at 22:19

Got an issue using wavesurfer. I have a demo here using react but I'm using Nextjs. I could not get codesandbox to work with their nextjs package so I used react instead. I have two issues:

I'm getting ReferenceError: self is not defined (node_modules/wavesurfer.js/dist/wavesurfer.js (15:4))
Everytime I make changes, wavesurfer duplicates. Can be seen in demo should you make a change.

Code:

...

ANSWER

Answered 2021-Oct-17 at 22:19

To make sure Wavesurfer does not duplicate, you need to destroy the wavesurfer instance when the component unmounts which can be done in the useEffect's cleanup phase.

Source https://stackoverflow.com/questions/69605050

QUESTION

The AudioContext was not allowed to start. It must be resumed (or created)

Asked 2021-May-31 at 15:56

So my script for some reason is now giving me the current errors when i run it. Its a selenium project that opens a browser headless.It was working just fine.

...

ANSWER

Answered 2021-May-31 at 15:56

So i found the answer to the ERROR and FOR THE INFO:CONSOLE.

For the ERROR i just added :

Source https://stackoverflow.com/questions/67770491

QUESTION

AudioContext, getUserMedia, and websockets audio streaming

Asked 2021-Apr-25 at 20:26

I am trying to make an as-simple-as-possible Javascript frontend that will allow me to receive audio from a user's mic on a mouse click within a web browser using getUserMedia, modify it to a custom sample rate and monochannel, and stream it over a websocket to my server where it will be relayed to Watson Speech API.

I have already built the websocket server using autobahn. I have been trying to make an updated client library drawing on whisper and ws-audio-api but both libraries seem outdated and include much functionality I don't need which I am trying to filter out. I am using XAudioJS to resample the audio.

My current progress is in this Codepen. I am stuck and having trouble finding more clear examples.

Both whisper and ws-audio-api initialize the AudioContext on page load, resulting in an error in at least Chrome and iOS as audio context must now be initialized as a response to user interaction. I have tried to move the AudioContext into the onClick event but this results in my having to click twice to begin streaming. I am currently using audio_context.resume() within the onClick event but this seems like a roundabout solution and results in the page showing it is always recording, even when it's not, which may make my users uneasy. How can I properly initiate the recording on click and terminate it on click?
I have updated from the deprecated Navigator.getUserMedia() to MediaDevices.getUserMedia() but not sure if I need to alter the legacy vendor prefixes on lines 83-86 to match the new function?
Most importantly, once I get a stream from getUserMedia, how can I properly resample it and forward it to the open websocket? I am a bit confused by the structure of bouncing the audio from node to node and I need help with lines 93-108.

...

ANSWER

Answered 2021-Apr-25 at 20:26

I found help here and was able to build a more modern JavaScript frontend based on the code from vin-ni's Google-Cloud-Speech-Node-Socket-Playground which I tweaked a bit. A lot of the existing audio streaming demos out there in 2021 are either outdated and/ or have a ton of "extra" features which raise the barrier to getting started with websockets and audio streaming. I created this "bare bones" script which reduces the audio streaming down to only four key functions:

Open websocket
Start Streaming
Resample audio
Stop streaming

Hopefully this KISS (Keep It Simple, Stupid) demo can help somebody else get started with streaming audio a little faster than it took me.

Here is my JavaScript frontend

Source https://stackoverflow.com/questions/67118642

QUESTION

Django and JavaScript: Using last uploaded file in an object list within in a Javascript file

Asked 2021-Mar-21 at 07:42

Edit: this question started as a question about copying files in Django but it turned out that the better way to achieve my aim of accessing files in JavaScript could be achieved directly.

Original question

I want to copy the latest uploaded mp3 file from the object list in my first model (which uses the default media folder) to a new folder called ‘latest’ and I also want to rename the new copy ‘latest.mp3’. This is so that I have a known filename to be able to process the latest uploaded file using Javascript. I wish to also keep the original, unaltered, uploaded file in the object list of my first model.

The below is what I have so far but it doesn’t work: I don’t get any traceback error from the server or from the browser. However, the copy isn’t made in the ‘latest/’ folder. I believe I am doing more than one thing wrong and I am not sure if I should be using CreateView for the SoundFileCopy view. I am also not sure when the process in SoundFileCopy view is triggered: I am assuming it happens when I ask the browser to load the template ‘process.html’.

I am using Django 3.1.7

If anyone can help me by letting me know what I need to put in my models, views and template to get this to work I would be very grateful.

Currently my models are:

...

ANSWER

Answered 2021-Mar-20 at 12:38

Give this a try,

Source https://stackoverflow.com/questions/66715524

QUESTION

Django- How to get last uploaded file into template?

Asked 2021-Mar-17 at 15:08

I am making a site where users can upload sound files. They can see the list of user-uploaded sound files and play them.

This much works. I have made a for loop of the object list with audio elements.

However, additionally, I also want to be able to place only the most recently added sound file on its own at the bottom of the template so that it can be dealt with separately (i.e. I want to put it in an audio element separate from those in the list and also be able to access it on its own in the template to process it using WebAudio API).

I know I need to use a filter and I keep reading that Model.objects.latest(‘field’) should do the job (I assume in views.py) but I am doing something very wrong as whatever I put in my view and my template creates errors. I am using class-based views and Django version 3.1.7

If anyone can show me with what I need to put into my model, my view, and my template so that I can get just the last added sound file into my template from the object list. I would be very grateful.

My model looks like this:

...

ANSWER

Answered 2021-Mar-17 at 15:08

You are passing only a single SoundFile object to the template, then you are trying to iterate that single object in a for-loop, that's why you are getting a 'SoundFile' object is not iterable error.

the ListView is used when you have to display a list of objects. Here, in your case, you have only one object to represent, So it is highly recommended to use a Detilview.

But If you still want to use ListView, then try this

Source https://stackoverflow.com/questions/66672938

QUESTION

How to play audio without strict limitations on JavaScript?

Asked 2021-Mar-15 at 15:42

I'm making a game made of bare JavaScript code and I want to manipulate audio in JavaScript. Specifically, I want to let the page start playing music when some div element is clicked and when using some function in the JavaScript code.

I know audio tag of HTML5 has a limitation that a music file associated with Audio element cannot be played without user clicking the play button of the audio element, so I cannot do like this:

...

ANSWER

Answered 2021-Mar-15 at 13:13

You're correct. You can only be sure that the playback will not be blocked by the autoplay policy if the playback is started in response to a user gesture. A button with a click handler is the classic example but the click handler could be attached to a div as well. Let's say your div has an id called my-div. In that case the following should work.

Source https://stackoverflow.com/questions/66637044

Community Discussions, Code Snippets contain sources that include Stack Exchange Network

Vulnerabilities

No vulnerabilities reported

Install WebAudio

You can download it from GitHub.

Support

For any new features, suggestions and bugs create an issue on GitHub. If you have any questions check and ask questions on community page Stack Overflow .

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