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Hi! I have a problem with IIR filter. On the output I have a input sin signal but the filter doesn't suppress the signal. It behaviour is that as rewrite input signal to output. I think that the problem is in the subtraction operation or types. Fcut=2MHz. Code below: library IEEE; use IEEE.STD_LOGIC_1164.ALL; use IEEE.STD_LOGIC_unsigned.ALL; use IEEE.NUMERIC_STD.ALL; entity dsp is Port ( clk : in STD_LOGIC := '0'; clk_adc : out STD_LOGIC := '0'; clk_dac : out STD_LOGIC := '0'; adc_data : in STD_LOGIC_VECTOR(9 downto 0) := (others => '0'); dac_data : out STD_LOGIC_VECTOR(9 downto 0) := (others => '0') ); end dsp; architecture Behavioral of dsp is constant coeff_a0: integer := 128; coefficients constant coeff_a1: integer := 256; constant coeff_a2: integer := 128; constant coeff_b0: integer := 293; constant coeff_b1: integer := 106; constant offset: STD_LOGIC_VECTOR(9 downto 0) := "1000000000"; signal adc_buffer: signed(9 downto 0) := (others => '0'); signal dac_buffer: STD_LOGIC_VECTOR(9 downto 0) := (others => '0'); signal mult_a0: signed(19 downto 0):= (others => '0'); mult registers signal mult_a1: signed(19 downto 0):= (others => '0'); signal mult_a2: signed(19 downto 0):= (others => '0'); signal mult_b0: signed(19 downto 0):= (others => '0'); signal mult_b1: signed(19 downto 0):= (others => '0'); signal tmult_a0: signed(9 downto 0):= (others => '0'); truncate registers signal tmult_a1: signed(9 downto 0):= (others => '0'); signal tmult_a2: signed(9 downto 0):= (others => '0'); signal tmult_b0: signed(9 downto 0):= (others => '0'); signal tmult_b1: signed(9 downto 0):= (others => '0'); signal add_all: signed(9 downto 0):= (others => '0'); signal pipe_a0: signed(9 downto 0) := (others => '0'); delay registers signal pipe_a1: signed(9 downto 0) := (others => '0'); signal pipe_a2: signed(9 downto 0) := (others => '0'); signal pipe_b0: signed(9 downto 0) := (others => '0'); signal pipe_b1: signed(9 downto 0) := (others => '0'); begin P0: process(clk) begin if falling_edge(clk) then adc_buffer <= signed(adc_data);  adc give data in U2 code pipe_a0 <= adc_buffer; pipe_a1 <= pipe_a0; pipe_a2 <= pipe_a1; pipe_b0 <= signed(add_all); pipe_b1 <= pipe_b0; dac_data <= dac_buffer(9 downto 0);  write to dac end if; end process; mult_a0 <= pipe_a0*to_signed(coeff_a0,10);  multiplication mult_a1 <= pipe_a1*to_signed(coeff_a1,10); mult_a2 <= pipe_a2*to_signed(coeff_a2,10); mult_b0 <= pipe_b0*to_signed(coeff_b0,10); mult_b1 <= pipe_b1*to_signed(coeff_b1,10); tmult_a0 <= mult_a0(19 downto 10);  truncate to 10 bits tmult_a1 <= mult_a1(19 downto 10); tmult_a2 <= mult_a2(19 downto 10); tmult_b0 <= mult_b0(19 downto 10); tmult_b1 <= mult_b1(19 downto 10); sum (add and substract) add_all <= tmult_a0 + tmult_a1 + tmult_a2  tmult_b0  tmult_b1; dac_buffer <= STD_LOGIC_VECTOR(add_all) + offset;  to offset binary convertion clk_adc <= clk; clk_dac <= clk; end Behavioral;
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Chris C. wrote: > if falling_edge(clk) then THIS IS NONSENSE. THERE IS NO FALLING CLOCK SENSITIVE FF IN AN FPGA. > clk_adc <= clk; > clk_dac <= clk; THIS IS BAD! CLOCKS CANNOT BE ROUTET OUT. also TRUNCATION IS TO EARLY.
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Chris C. wrote: > use IEEE.STD_LOGIC_unsigned.ALL; > use IEEE.NUMERIC_STD.ALL; Use the first one or use the second one. But never ever use both of them together. Best is to use the numeric_std and its conversions and casts. See how simple that is: http://www.lotharmiller.de/s9y/categories/16Numeric_Std Try Google translator, it's German...
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Beside all the other mentioned problems: What is your main clock frequency? I've added a testbench to the see the filter effect. For me it looks like your cutoff frequency is to high. Duke
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My clock frequency > 100MHz In code below coefficient are for 4MHz cutoff frequency but I dont know if they are correct. In new code is is easy to change coefficients. I need coefficients only to test my filter. I would be gratefull if somebody generate coefficient (10 bit) to lowpass with cutoff beetwenn 110MHz. I dont know what to do with overflow bits. If I have 23bit ad_all signal the output amplitude is 4 times lower than when I have 19bit ad_all signal. It probably means that the overflow bits doesn't occur. In my opinion the sum of 5 signed 19bit signals should be stored in 23bit signal. Secondly is that if I operate on signed values I should be carefull with math operations signs (+/). If I add negative values as a constant signed, in operation of sum of the signals I must use only "+" sign. I ignored all overflow bits. It is possible that I have only wrong coefficient. On hardware filter works but not correct. My new, some improved code: library IEEE; use IEEE.STD_LOGIC_1164.ALL; use IEEE.NUMERIC_STD.ALL; entity dsp is Port ( clk : in STD_LOGIC := '0'; clk_adc : out STD_LOGIC := '0'; clk_dac : out STD_LOGIC := '0'; adc_data : in STD_LOGIC_VECTOR(9 downto 0) := (others => '0'); dac_data : out STD_LOGIC_VECTOR(9 downto 0) := (others => '0') ); end dsp; architecture Behavioral of dsp is constant coeff_a0: integer := 133; coefficients constant coeff_a1: integer := 262; constant coeff_a2: integer := 133; constant coeff_b0: integer := 164; constant coeff_b1: integer := 70; constant offset: signed(9 downto 0) := "0111111111";  +511 in U2 signal adc_buffer: signed(9 downto 0) := (others => '0'); signal dac_buffer: signed(9 downto 0) := (others => '0'); signal mult_a0: signed(19 downto 0):= (others => '0'); mult registers signal mult_a1: signed(19 downto 0):= (others => '0'); signal mult_a2: signed(19 downto 0):= (others => '0'); signal mult_b0: signed(19 downto 0):= (others => '0'); signal mult_b1: signed(19 downto 0):= (others => '0'); signal add_all: signed(19 downto 0):= (others => '0'); signal pipe_a0: signed(9 downto 0) := (others => '0'); delay registers signal pipe_a1: signed(9 downto 0) := (others => '0'); signal pipe_a2: signed(9 downto 0) := (others => '0'); signal pipe_b0: signed(9 downto 0) := (others => '0'); signal pipe_b1: signed(9 downto 0) := (others => '0'); begin P0: process(clk) begin if falling_edge(clk) then adc_buffer <= signed(adc_data);  adc give data in U2 code pipe_a0 <= adc_buffer; pipe_a1 <= pipe_a0; pipe_a2 <= pipe_a1; pipe_b0 <= add_all(19 downto 10); truncated sum pipe_b1 <= pipe_b0; dac_data <= STD_LOGIC_VECTOR(dac_buffer(9 downto 0));  write to dac (ignore MSB  offset bin conversion) mult_a0 <= pipe_a0*to_signed(coeff_a0,10);  signed * signed return 20 bit signed mult_a1 <= pipe_a1*to_signed(coeff_a1,10); mult_a2 <= pipe_a2*to_signed(coeff_a2,10); mult_b0 <= pipe_b0*to_signed(coeff_b0,10); mult_b1 <= pipe_b1*to_signed(coeff_b1,10); sum of signed results add_all <= mult_a0 + mult_a1 + mult_a2 + mult_b0 + mult_b1; signed + signed + ... dac_buffer <= add_all(19 downto 10) + offset;  to offset binary convertion (add 511) signed + signed end if; end process;  dac buffer is 11 bit long clk_adc <= clk; clk_dac <= clk; end Behavioral;
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Edited by User
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Chris C. wrote: > In code below coefficient are for 4MHz cutoff frequency but I dont know > if they are correct. Did you have a matlab? There is a toolbox available for calculating filter coeffiecients in fixed data formats. Also do this toolbox a stability analyis of your IIR filter. Best Regards,
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Chris C. wrote: > It is possible that I have only wrong coefficient. On hardware filter > works but not correct. Your coefficents are correct but your code doesn't work. If I use your coefficients with the implementation biquad, form 1 as described on this website [1], I got the expected results. Can you describe graphically the structure of your IIR filter? Duke [1] http://www.iowahills.com/4IIRFilterPage.html
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The names in code doesn't match with the schematic, but anyway: 1. move the multiplications out of your register stage (if *_edge ... end if) 2. mind the polarity signs on the big 'adder' 3. don't move the adder out of the register stage Another hint (again): Use falling_edge only if you have a good reason. Duke
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Duke Scarring wrote: > 1. move the multiplications out of your register stage (if *_edge ... > end if) > 2. .... the big 'adder' This will led to a lot of logic levels in this case  4 adders  plus 1 multiplier I would expect timing problems with the code posted by Chris. Generally it is difficult to run an IIRFilter with sampling frequency = clock frequency. Since the feed back loop has only one clock budget but you need at least two operations, add and multiply. Within a second order stage (as shown in the iir.jpg picture) there has to be performed three sequential operations, 2 additions and 1 multiplication. Either you get many logic levels (in terms of DSP blocks) or to many pipeline stages. What is the reason for your IIRFilter? Tom
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Tom, I make this filter as a project on high school. It is very possible that you're right. I spend many hours with trying to resolve problem with this filter. I obtain results which magnitude characteristic was above 10dB upper than expected which I compue in FDA Tool in MATLAB. Interestng is that I implemented FIR filter with the same way and it works very good. I saw the VHDL code where IIR filter was divided to machine state (as you described the number of operations in one clock cycle). Maybe it is a solution of my problem but I need 100MSPS. I think that the problem is in the second part of my filter, it works but 10dB above.