Oops! I reversed the n and k. Try this instead:
N
y(k) = sum x(k-n)*h(n)
n=0
--
Jim Thomas Principal Applications Engineer Bittware, Inc
jthomas@bittware.com http://www.bittware.com (703) 779-7770
Better to ask a stupid question than to make a stupid mistake.
Reply by Jim Thomas●August 6, 20042004-08-06
Cristiano wrote:
> In that page I only seen the filter design, but as I said, I need to know
> how to convolve the input signal and the filter kernel.
N
y(k) = sum x(n-k)*h(n)
n=0
For various implementations, go to dspguru.com and read the FIR FAQ.
--
Jim Thomas Principal Applications Engineer Bittware, Inc
jthomas@bittware.com http://www.bittware.com (703) 779-7770
Better to ask a stupid question than to make a stupid mistake.
Reply by Cristiano●August 6, 20042004-08-06
axlq wrote:
> In article <D3wQc.76974$OR2.4340761@news3.tin.it>,
> Cristiano <cristiano.pi@NSquipo.it> wrote:
>> I'm trying to write (in C++) a low-pass filter for audio frequencies.
>
> This may help:
>
> http://unicorn.us.com/alex/buttercrit.html
In that page I only seen the filter design, but as I said, I need to know
how to convolve the input signal and the filter kernel.
Cristiano
Reply by axlq●August 5, 20042004-08-05
In article <D3wQc.76974$OR2.4340761@news3.tin.it>,
Cristiano <cristiano.pi@NSquipo.it> wrote:
>I'm trying to write (in C++) a low-pass filter for audio frequencies.
I'm trying to write (in C++) a low-pass filter for audio frequencies.
I used this:
H[i]= sin(2*PI*FC * (i-M/2)) / (i-M/2) *
* 0.54 - 0.46*cos(2*PI*i/M)
where M= 100 points and FC is the cutoff frequency (0 to 0.5), but I guess I
don't know how to use it because the filtered music is really bad.
I have the real and the imaginary part of the transformed input signal.
Could anybody tell me how to convolve the input signal and the filter kernel
H[]?
Thanks
Cristiano