Hello guys,

I want to read an analogue audio signal with a microcontroller and filter it digitally (by writing the proper algorithm). Any ideas on how to implement a bandpass digital filter?

 

Checkout
http://www.dspguide.com

 

You can also try this introduction to digital filters.

hgmjr

 

Thanks for the links but do you know any transfer function which makes the function of a bandpass filter?

 

You select a low pass function for one end of the pass band, and a high pass for the other. The low pass rolls off the upper end of the band; the high pass rolls off the lower end. The range of frequencies between the -3 dB points is the passband.

 

You select a low pass function for one end of the pass band, and a high pass for the other. The low pass rolls off the upper end of the band; the high pass rolls off the lower end. The range of frequencies between the -3 dB points is the passband.

Yes but i have never built a digital filter before so i would like to know how we get the transfer function of a low pass filter for example according to the cutoff frequency, the Q factor etc we want.

 

Check this out. It describes really simple one-pole IIR digital filters that can be implemented in small microcontrollers.
http://www.psocdeveloper.com/uploads/tx_piapappnote/an2099.pdf

Here is an applet that allows you to play with coefficients for up to four biquad filters.
http://www.digitalfilter.com/products/iircalwav/iircalwav.html

Hope this helps.

 

Thanks Caveman

 

If you have access to the Filter Design Toolbox in MATLAB I would highly recommend fdatool. Otherwise, Filter Free is my favorite analog filter design tool that also does digital filter design. It's available for download from http://www.filter-solutions.com/