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Hi everyone.
I'm trying to make a simple 4 bands EQ with DSP (56k) and I would like to know what is the best FIR design methode for audio application.
Equiripple ?
Least square ?
Window ?
... ? etc
Ho yes ! You're right, y variables must be updated vis-a-vis the actual calculated filter. I never think about it before. Thank you :). Now I understand how it's hard to do an IIR filter but it still possible. But I guess that if I do it with FIR filter, it will be easier.
I looked out for FIR...
I zorro, thank for reply me.
I tested each filters one by one they work very well. The problem come from when I want assemble differents filters to "one".
Here is the code I wrote. I hope you will find why it's not work.
The first part is a do loop. I would like to implement 2 bandpass...
Yes, the gains in the bassbands are the same for the 3 filters (Gain of 0dB for each one). The result in practice was similar like a bandpass filter with a 10 kHz central frequency. :S
So I decided to remove the third filter (Bandpass filter 4kHz) to make it easier and the result was a fail...
Hello.
I actualy try to code a digital my firts equalizer with the 56k DSP (56374). I would like to know if my way to make it is good because the final response isn't look like I want.
1) take sample x(n) from ADC
2) calculate x(n) for each filter and store it.
3) Pop out results from 2) and...
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