Honored ContributorDear sir,
I am Mtech final year student. Right now I am working on ac energy parameter monitoring system. In my project I want to design filters for noise reduction like low pass , high pass , butterwort, chebysev etc... so can u plz suggest me how to make filter in fpga using Verilog. Step for implementing , any notes, materials etc....
Honored ContributorBasically what you need is rather to calculate the filter parameters and normalize them for integers. Or that is what I have done in the past. Filters are basically additions of different "volumes" over a length of time. The Code itself is not the hard part. But what I tend to do for simple filters is to simulate an ordinary RC filter like this out<=out+(k*(in-out)/256) ... K being 0-256 here.. this also works with resonance and feedback if you limit the signals and what not. Interpolation uses the same formula but with different inputs/outputs.
THis is a piece of PROTOTYPE CODE i used for a synth project.. i ended up not using it .. And for GOD SAKES keep in mind I'm a total beginner at this ... but at least it's something ,right..
reg signed yaudio ; //yaudio is output from filters (audioout is input)
reg signed ac; //accumulator
always @ (posedge clk30)
begin
reg signed b0=68; //lowpass filter response
reg signed b1=16;
reg signed b2=68; //(or 76) ,b2 is most often same as b0(could be optimized)
reg signed a1=117;
reg signed a2=-54;
reg signed in;
reg signed wa,wb,wc;
reg l; //timer limit value
reg co; //the actual timers
reg signed m; //feedback level multiplier
reg ch;
l<=(potlim/32);//frequency cutoff
m<=(potlim/256);//resonance level
ch<=ch+1; //8channels, clock through them all
in = ((audioout+(audioout*m)/32)-((yaudio*m)/8))*4; //do feedback and input
wa = ((wc * a2) + (wb * a1) + (in /2))/64; //do filtering
ac = (wa * b0) + (wb * b1) + (wc * b2); //do more filtering
if (co++>=l) begin co=0; wc <= wb; wb <= wa; end //rotate data for filter
if (co++>=l) begin co=0; wc <= wb; wb <= wa; end //faster = higher cutoff
if (co++>=l) begin co=0; wc <= wb; wb <= wa; end //at different speeds
if (co++>=l) begin co=0; wc <= wb; wb <= wa; end
if (co++>=l) begin co=0; wc <= wb; wb <= wa; end
if (co++>=l) begin co=0; wc <= wb; wb <= wa; end
if (co++>=l) begin co=0; wc <= wb; wb <= wa; end
if (co++>=l) begin co=0; wc <= wb; wb <= wa; end
end
assign yaudio = ac/256/2; //do output levelchange
assign yaudio = ac/256/2; //to be able to use all bits
assign yaudio = ac/256/2; //input also does this
assign yaudio = ac/256/2; //and center too
assign yaudio = ac/256/2;
assign yaudio = ac/256/2;
assign yaudio = ac/256/2;
assign yaudio = ac/256/2;
Basically what you need is rather to calculate the filter parameters and normalize them for integers. Or that is what I have done in the past. Filters are basically additions of different "volumes" over a length of time. The Code itself is not the hard part. But what I tend to do for simple filters is to simulate an ordinary RC filter like this out<=out+(k*(in-out)/256) ... K being 0-256 here.. this also works with resonance and feedback if you limit the signals and what not. Interpolation uses the same formula but with different inputs/outputs.
This is a piece of PROTOTYPE CODE i used for a synth project.. i ended up not using it .. And for GOD SAKES keep in mind I'm a total beginner at this ... but at least it's something ,right..
reg signed yaudio ; //yaudio is output from filters (audioout is input)
reg signed ac; //accumulator
always @ (posedge clk30)
begin
reg signed b0=68; //lowpass filter response
reg signed b1=16;
reg signed b2=68; //(or 76) ,b2 is most often same as b0(could be optimized)
reg signed a1=117;
reg signed a2=-54;
reg signed in;
reg signed wa,wb,wc;
reg l; //timer limit value
reg co; //the actual timers
reg signed m; //feedback level multiplier
reg ch;
l<=(potlim/32);//frequency cutoff
m<=(potlim/256);//resonance level
ch<=ch+1; //8channels, clock through them all
in = ((audioout+(audioout*m)/32)-((yaudio*m)/8))*4; //do feedback and input
wa = ((wc * a2) + (wb * a1) + (in /2))/64; //do filtering
ac = (wa * b0) + (wb * b1) + (wc * b2); //do more filtering
if (co++>=l) begin co=0; wc <= wb; wb <= wa; end //rotate data for filter
if (co++>=l) begin co=0; wc <= wb; wb <= wa; end //faster = higher cutoff
if (co++>=l) begin co=0; wc <= wb; wb <= wa; end //at different speeds
if (co++>=l) begin co=0; wc <= wb; wb <= wa; end
if (co++>=l) begin co=0; wc <= wb; wb <= wa; end
if (co++>=l) begin co=0; wc <= wb; wb <= wa; end
if (co++>=l) begin co=0; wc <= wb; wb <= wa; end
if (co++>=l) begin co=0; wc <= wb; wb <= wa; end
end
assign yaudio = ac/256/2; //do output levelchange
assign yaudio = ac/256/2; //to be able to use all bits
assign yaudio = ac/256/2; //input also does this
assign yaudio = ac/256/2; //and center too
assign yaudio = ac/256/2;
assign yaudio = ac/256/2;
assign yaudio = ac/256/2;
assign yaudio = ac/256/2;
