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.
Low Pass Filter design using ADSP21061 EZkit Lite
Started by ●July 15, 2003
Reply by ●July 15, 20032003-07-15
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.