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I have made some research but I couldn't find what I am exactly looking for. At the moment, I have to send channel values by com port.

For example:

  • the content of file freqs.ini

    low=0-xx khz; 
mid=xx-yy khz; 
high=yy-zz khz;

Then I will get values by percentage like

  • the expecting values

    lowPercent  = 10;
midPercent  = 77;
highPercent = 53;

So, I will be able to send these values by rs232 and my room will turn into club :) (I am using this code to illuminate LED strips). I have found some spectrum analyser projects but they all have 9 channels, that is, 3*3 combinations from low-low to high-high.

I know how to communicate with com port, but how can I get integer values of 3 frequency range I have set before?

John Saunders
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Yağız Elesković
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  • Your question is not totally clear, but if you have code that divides audio into 9 bands, why not simply sum the results from the bands in groups to get only three? low = a1 + a2 + a3 , mid = a4 + a5 + a6, high = a7 + a8 + a9; – Bjorn Roche Mar 19 '13 at 01:33
  • I will enter a frequency range and my program will give me some values.I am DJing, I have a dj setup at my home.I have 3 strip leds.Mixers has 3 main frequency knobs.I want my leds act according to audio channels.For ex, when I turned **low** knob to 0, my first strip led will not bright because there will not be any low frequency audio.So, I need to get real-time values of audio – Yağız Elesković Mar 19 '13 at 01:35
  • Perhaps you should read this question: http://stackoverflow.com/questions/1794010/how-to-use-numpy-with-portaudio-to-extract-bass-mid-treble I'm sure there's a library for audio filtering in C#. If not, you can write one: http://blog.bjornroche.com/2012/08/basic-audio-eqs.html – Bjorn Roche Mar 19 '13 at 01:42
  • I have read it all but couldn't find anything to help :( – Yağız Elesković Mar 19 '13 at 02:04

1 Answers1

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I don't know if you still need of that but....

Do you want to know how to get a real-time spectral analsys of sound?

1.implement a queue to take a buffer of audio samples

2.take the product of buffer and a proper window function (tipically , Hamming or Hann) calculated by your program as float array

3.do FFT of yelded array: there are may algortihms out there for every language....find the best one for you, use it and take the square module from each output coefficent ( Real_part^2 + Imaginary_part^2 , if FFT returns you algebrical representation of coefficients)

  1. sum coefficients across your bands: to know what coefficient is associated to a frequency you've just got to know that the k-th coefficient is at (SampFrequency/BufferLength)*k Hz.....so it's easy to find band boundaries

  2. if you need to normalize in [0 , 1] interval, you have to do nothing but divide each of 3 yelded bands value for maximum value between the 3

  3. pop your buffer queue by a Shift value that is Shift <= BufferLength and start again

the number of coefficients coming from FFT alg is equal to BufferLength (this is beacause the Discrete Fourier Transform definition) so, the frequency resolution is better when you select a long buffer, but the program goes slower. The light intensity wont' vary after BufferLength audio frames, buf after Shift audio frames.....and high ratio beetwen BufferLength gives you slowly light changes....so you must select parameters that fits your desires, remembering that you have just to turn on & off some light....make your alg fast and lo-fi!

The last thing to do is discover freqeuncy bands from your mixer's eq knobs....i don't remember if this information was on mixers handbooks

Carmine Ingaldi
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