equalizer | audio analysis tool for real-time | Data Visualization library
kandi X-RAY | equalizer Summary
kandi X-RAY | equalizer Summary
An audio analysis tool for real-time and choreographed visualizations. Check out the gh-pages for use.
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Trending Discussions on equalizer
QUESTION
I have an arrayList<> of strings and I added 10 strings to it.
...ANSWER
Answered 2022-Feb-22 at 05:27I'll recommend you to directly use the value of position for targetValue, inside onBindViewHolder while setting the value of text.
QUESTION
I am building an Android App which needs to create several vertical sliders in the same page for music equalizer adjustment, but I can only find horizontal sliders from the official material design documents.
I try to implement default slider from official documents and rotate it with modifier and it works, but the problem is that I am not able to adjust the height now using Modifier.
...ANSWER
Answered 2022-Feb-15 at 15:50Your rotated slider doesn't fill available height, because it gets a wrong constraint after rotation. To fix this you first of all need to swap width
and height
values.
Next, Modifier.rotate
won't change the view position, so you need to manually update it after rotation. To understand what's going on, you can run the following code:
QUESTION
To give a bit of context, i am trying to implement in matlab from scratch the following signal diagram, a Feedback Delay Network (FDN). pic: FDN
With an appropriate matrix, indifferent to delay lengths, virtually white noise comes out when fed a dirac impulse.
I've managed to do this in code, but my goal is another and hence my question. I want to apply a filter h(z) after each delay line z^-m. pic: h(z)
More specifically, i want to apply a third-octave cascaded graphic equalizer after each delay line. The purpose is to create frequency dependent attenuation on the whole structure, and consequently delay dependent. I've successfully designed the filter in the form of SOS, but my problem is: how do I apply it within the structure? I assume to use sosfilt() somewhere with what I have, but I'm not sure.
I haven't reduced the order of the system for sake of purpose. The order is 16 (16x16 matrix, 16 delay lines, 31x16 biquad filters)
The first code refers to the lossless FDN, safely runnable which generates white noise. I have commented my failed attempt to introduce the filtering in the loop saying: % Filtering
Unfortunately, I can't post all GEQ entries, but I'll leave 8 in the end corresponding to the first 8 delays.
So, the question is how do I code to filter the white noise, implementing frequency dependent attenuation in the whole FDN structure. Also, although it may be computationally inefficient, I'd prefer to apply this without higher level functions and based on what I already have, i.e: applicable in GNU Octave
Edit: Assuming you have to apply the bandpass 2nd order filtering sample by sample using the difference equation, how would you recursively do it for 31 bands in series? One is shown in the second code section.
...ANSWER
Answered 2022-Jan-24 at 03:08And so a sample by sample, rather inefficient way of filtering the noise out of a dirac impulse from a FDN would be to add 2 more buffers and means of calculating difference equations of 31 cascaded biquad filters (Any suggestions for improving calculation speed comment below)
QUESTION
I was playing with AVAudioEngine to use it in an app. I am changing pitch and equalizer (gain) parameters with sliders while music was playing.
Pitch works but equalizer is not working. I could not find the cause.
...ANSWER
Answered 2022-Jan-17 at 17:09Your AVAudioEngine
graph looks a little weird (can AVAudioPlayerNode
have two outputs?), but if you attach your nodes:
QUESTION
We are working on a project which allows us to record some sounds from a microphone with a 5k Hz sample rate with some Low-Pass filter & HighPass filter.
What we are using
We are using AvaudioEngine for this purpose.
We are using AVAudioConverter for downgrading the sample rate.
We are using AVAudioUnitEQ for the LowPass & HighPass filter.
Code
...ANSWER
Answered 2021-Dec-17 at 10:50I think the main problem with this code was that the AVAudioConverter
was being created before calling engine.prepare()
which can and will change the mainMixerNode
output format. Aside from that, there was a redundant connection of mainMixerNode
to outputNode
, along with a probably incorrect format - mainMixerNode
is documented to be automatically created and connected to the output node "on demand". The tap also did not need a format.
QUESTION
I am currently working on an equalizer which takes the input of a microphone in rust using cpal as audio backend.
I am capturing the raw data and sending it to another thread like this:
...ANSWER
Answered 2021-Aug-07 at 15:35For an equalizer I'd use an FFT algorithm implementation.
There are libraries for Rust, e.g. https://github.com/ejmahler/RustFFT. You can find more on crates.io.
In your code you are sending samples one by one, but the FFT expects a buffer (a sequence of samples during some short period of time). Given a buffer it outputs the frequencies graph in this time frame.
If you give it a sliding time window, then the repeating frequencies will be averaged, and it should show what you expect from a typical EQ. A ring buffer could be useful to collect samples in this sliding window (e.g. this one from dasp)
QUESTION
I'm trying to make an equalizer animation and would like these lines to animate from the bottom of the containing DIV as opposed to animating from the top like they are now.
...ANSWER
Answered 2021-Apr-03 at 02:57I think using transforms might suffice and be lighter than adjusting the height
property. The property that will change the orientation is the transform-origin: 0 100%
. Below the script transforms the bars by scaling from transform: scaleY(1)
to transform: scaleY(5)
.
QUESTION
I am a shiny newbie. I am trying to make an app where the user can move the slider and select the values, like in this sound equalizer example or colour picker example. But my app places the slider inputs on top of each other. Can someone help? Codes:
...ANSWER
Answered 2021-Mar-03 at 21:11As in this post, you can embed your inputs in a div(style="display:inline-block", ...)
QUESTION
I am working on an audio visualizer for the web that also lets the user "tune" the raw audio signal visualizer to a frequency. This is a feature of many hardware oscilloscopes. Basically, when a user centers on 440Hz and I have a 440Hz sine wave, the wave should stay still on the canvas and not move left or right. My plan was to move the graph to the left according to the frequency (440Hz = 1/440s to the left per second because the wave should repeat every 1/440s), but this does not work as it seems.
I could not find the units used by the Audio Analyzer Node's time domain data. I guess that it's in milliseconds, but I am not certain.
...ANSWER
Answered 2021-Jan-26 at 16:49I was able to solve this problem thanks to Raymond Toy's comment and my maths teacher (thank you Mr. Klein). The solution was Math.round((this.context.currentTime % iv) * sampleRate)
where iv
is the interval of the frequency (1/Hz
). The wave is not perfectly centered. The FFT approximation is not very accurate though. In the following example I forced the detected frequency to be the specified one.
QUESTION
I'm trying to figure out how to remove duplicates based on the url, as the aggregate query can match the same document twice if say "APPL" and "TSLA" are in stocks
and included in the same document.
ANSWER
Answered 2021-Jan-16 at 06:11You can use $group
stage after $match
stage,
$group
byurl
and get first root document using$$ROOT
, this will return document inroot
field
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