Hi

You have to convolve the input vectors (the pure sines) with the impulse
response from your filter (the coefficients).

Take a look at : ftp://ftp.analog.com/pub/dsp/210xx/ where some examples
exist that you might find interesting.

Thomas

"Suman" <suma_kin@yahoo.com> wrote in message
news:aac84c5a.0307150152.236b0626@posting.google.com...
> Hi all,
>
> I have the following specification for design of Low-Pass Filter (
> Butte Worth Filter)
> 1. Order of Filter =10.(N)
> 2. Cut-off Frequency =1KHz.(Fc)
> 3. Sampling Frequency = 5KHz.(Fs)
>
>  I have found out filter coefficients:
>
>              H(f) = sqrt(1/(1+(f/Fc).^(2*N)));
> Input to the filter is
> case 1: input = pure sine signal with 500Hz
>
> case 2: input = pure sine signal with 1500Hz
>
> case 3: input = pure sine signal with 500Hz + pure sine signal with
> 500Hz
>
> My problem is
> I have calculated the filter coefficients,
> How do I get the output, should I convolve the filter coefficients
> with input coeficients?
>                                        or
> should I multiply the input coefficients with the filter coefficients
>
> if any other method , please suggest me.

Hi all,

I have the following specification for design of Low-Pass Filter (
Butte Worth Filter)
1. Order of Filter =10.(N)
2. Cut-off Frequency =1KHz.(Fc)
3. Sampling Frequency = 5KHz.(Fs)

 I have found out filter coefficients:

             H(f) = sqrt(1/(1+(f/Fc).^(2*N)));
Input to the filter is
case 1: input = pure sine signal with 500Hz
    
case 2: input = pure sine signal with 1500Hz

case 3: input = pure sine signal with 500Hz + pure sine signal with
500Hz

My problem is 
I have calculated the filter coefficients,
How do I get the output, should I convolve the filter coefficients
with input coeficients?
                                       or 
should I multiply the input coefficients with the filter coefficients

if any other method , please suggest me.