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98 Threads found on Fir Iir Filter
Hi, I am interested about planar UWB antennas like vivaldi or patch antenna. These UWB antenna should be considered as what type of filter (eg fir or iir) ? After simulating in HFSS, if I save |S11 | dB 2D plot in .csv format, how may I get frequency response H(jomega) from S11.csv file in MATLAB?
hello i designed a low-pass filter in fdatool like this: 133021 when i use it in 12-bit configuration output is like below: 133022 this output looks like my input and seems to be correct. but when i use that filter in 8-bit configuration the output is like this 133023 m
Butterworth is an iir filter, the PT1 response to a step should be an exponential function. In case you are intending an fir butterworth approximation you should mention this.
Hi everyone! I wanted to design some digital filters for an ECG device based on Zynq SoC. Since The device works with 24bits of data, I can't use the fir block present in System generator. So I designed a low-pass 65th order fir filter with 250Hz cutoff frequency, and a 2nd order band-stop iir (...)
fir filters: They do not have poles (actually all poles lies on the center of the unit circle in z transform). fir filters only have zeros. Order of fir filter decide the number of zeros. fir filter is always stable because they do not have poles. There is (...)
i read in a ieee paper that FFT are use to design a filter bank instead of fir and iir filters but i am not understanding how it happens they said that FFT is used to increase the channels if necessary with same FFT can u please help me in this topic
I guess that you implies the delay count? If so, you should examine the polynomial order of the fir/iir filter. For example, a filter usually can be expressed as the following form: H(z) = A0 + A1 * z-1 + A2 * z-2 + ... + An * z-n where Y(z) = X(z)H(z).
I am multiplying a 2kHz sinewave with another 100khz sinewave to obtain a dsb-sc signal. These sinewaves were designed using DDS cores. I am also using a 50mHz clock. I would like to demodulate this dsb-sc signal. I am doing this by multiplying the dsb-sc signal with the carrier (100khz signal). Now i have to pass this signal through a
You give the explanation in the question title. The minimal fir filter order is related to the fs/fc ratio. With this sampling frequency, you can't make even a poor fir filter with less than e.g. 500 or 1000 taps. Possible solutions: - design a multirate filter with decimation before the final (...)
Read the Matlab help aboit iir design methods. fdesign.lowpass('N,Fc',...) is only valid for fir. Apart from this point I don't agree with your assumption about "better lowpass iir filter". fir is basically good, although order of 31 is insufficient for a good lowpass filter with fc/fs = 4/100.
Hello, Can someone explain the difference between the implementation of iir and fir filter with microcontroller if I am planning to implement a lowpass filter Thanks.
You can test fir versus iir implementation in Matlab. To get the same behaviour as in the microcontroller, you'll use a fixed-point simulation. You'll find out that for the given fs and fc values, a fir implementation doesn't produce a reasonable frequency response below 100, better 200 taps, which gives a strong bias towards (...)
I have read in my textbook that the motivation behind the existence of fir filter is "In particular, fir filters are used for their ease of implementation and stability." The stability issue is quite easy to explain from the transfer function along with using certain stability criterion. But I don't understand why they (...)
Is there an easy or straightforward way to analyze a frequency response plot/filter profile of a digital filter and determine it's architecture (number of taps, fir vs iir, decimation rate, coefficients, etc) For example, what information do i have about the filter and it's implementation by looking these (...)
how the direct form is most sensitivity to the coefficient quantization in iir filter. and cascade form is not much sensitive to the coefficient quantization in iir filter. how the direct form is more efficient in fir filters.
Keep in mind that iir/fir. specify the underlying filter structure, while adaptive refers to how the coefficients are updated. You probably are looking for an adaptive fir filter that uses the LMS algorithm.
Which version are you using?. I have used some fir from xilinx sys gen. However did you meet proper input and output sample rate?.
In this case, instead of thinking in fir terms, why don't you simply consider the analog filter specification (cut-off frequency, maximum pass-band error/ripple, stop band frequency, required stop band attenuation). It will lead you directly to a filter design. An "optimal" filter approach might end up in a Chebyshev (...)
Hi all!! We use a delay unit in the feed back path to implement z-1 in fir/iir system design. I am curious to know how to implement 'z' and positive powers of z if they appear in the tranfer function. Sorry if the question seems a bit stupid :-( Thanks
The order of a fir filter is the degree of its transfer function polynomial, i.e., the number of zeros. An N-order fir filter uses N+1 input samples (the last one and the N previous). The order of a iir filter is the degree of the denominator polynomial, i.e., the number of poles, provided (...)
i m doing a project on implementation of iir filters using system generator based fpga. i m using spartan 3 starter kit for this purpose however i m facing problem as it is not providing me with any iir filter response while on the same fpga fir filters are working properly, can sombody help (...)
If i use fir filter, order of HPF(remove DC) is very large. Yes, use simple first order iir.
The following are informative examples of iir filter design using PICs: Chapter 3: Infinite Impulse Response (iir) filters Microchip Appnote AN852: Implementing fir and iir Digital filters Using PIC18 Mic
Hello, I am trying to show the difference between the outputs of fir and iir filters. However the phase distortion in the iir filter output is not evident. Here are the scripts: (...)
95 kHz sampling rate sounds reasonable. What do you mean exactly with "but above 9KHz it is not stable"? Strictly spoken, fir filters are stable by nature. Did you verify the filter operation in Matlab with the numeric resolution used in the FPGA implementation? If you are implementing a Chebyshev filter of moderate (...)
Hi guys, anyone can help me on this? Thanks
Current Processors can compute 1 output per cycle for a 10 tap fir filter, may take 3-4 cycles per 1 output for a 10 tap iir filter. Hope that helps........
Hi, I am doing survey on "Hardware modules of Signal Processing", Anyone has any material related to fir/iir digital hardware using MACs? Useful links will also be helpful. Thanks, Regards, Naveed
How long is the wiring between the sensor and the micro? If it's long (like 40cm or so) then you might want to use a shielded cable for that. And what is the time constant or cutoff frequency of that RC filter? It might be just bad component calculation. Other solution would be to implement an iir or fir filter inside MCU, (...)
There's some code involved like CIC filter, fir filter, iir filter ,Cordic algorithm and FFT in the book "Digital Signal Processing with Field Programmable Gate Arrays".Hope it would be helpful.
Hi saatwik, I'm not specialized in audio technology (I guess you are speaking about audio applications), but I can guess that if fir filter are not used, this is because the same result is obtained more simply with iir. Of course, peak and notch filters can be made in fir. But: a) if the (...)
hai,, i am rashmi,, doing mtech,, can u plz guide me to write c code for fir filter design ,,, any 1 plz reply....... where can i get gudance about c code to fir filter design:oops8-O
Hi! i have build ECG acquisition hardware and then acquired this signal in MATLAB and LabVIEW using DAQmx(Data Acquisition device). the sampling ratewas 500Hz. is this rate ok? please help me with further filtering process, either iir or fir My design is output is here [
Hello Experts, The output of the fir filters is convolution of the input signal and the filter kernel. In that case, the length of the output signal should be greater than input signal by M-1 points where M is the length of the filter kernel. x=ecg(500)'+0.25*randn(500,1); %noisy waveform (...)
the coefficients play a large role in the filter design. Any fir filter can be normalized (not including quantization issues). stable iir filters can also be normalized (and excluding quantization issues).
You can implement that with a fir or iir digital filter in Matab. The iir is better
firstly try only spectrogram(x), then check other options if needed.
Hi, Can you please elaborate further on what kind of a filter you wish to implement? iir or fir? And what is the sampling frequency for your problem? Also is the higher frequency cutoff 40KHz or 40MHz? Mohit
There is nearly nothing to add - perhaps one fir property which in many cases is important: fir filters have a linear phase function (opposite to iir filters).
Clearly the answer depends on the type of impulse response. To represent an arbitrary impulse response, you would need a respective number of fir coefficients. Some classes of responses may be reproducable with a rather short iir structure, others not at all. Generally, the advantage of iir shows with representing time-continuous (...)
i am doing project in dsp on multirate and multistage filter design and application is below attachment,can anyone do the matlab/simulink for me pleaseeeeeeeeee
I am a little confused? What do you mean it works at lower frequency? One possible cause could be the implementation errors in fir/iir approximation. As i understand, Simulink is a time domain simulation tool, so the filter block must be doing some fir/iir for the time domain response. now (...)
Hi You are given an iir filter and now you have to approximate the given iir filter with an fir one.
Hello all, I would want to implement in PIC18 a digital filter (iir or fir...), I read an interesting application note to integrate the filter in a pic; ( ) but I don't know how I could calculate coefficients of fir or iir filter and what is the method to build a digital (...)
is there any algorithm written in C to filter out the data The simplest approach would be to take the moving average of the last n samples. More sophisticated approach would be the fir or iir filter.
For EMG you can implement a fir filter with the order that you want and can conservate the phase mainly if you apply an iir will destroy the phase. with a MC56F8025 is enough to implement this in realtime the EMG signal are slow signal in comparison with the DSP capability. pls review the following link for a cheap CPU. www.freescale.
When creating interpolation simulink model, I got two different result from these two models. Assume I want to create upsampling 2x system. The first system, upsampling 2x then halfband fir filtering. The second system, only use one polyphase iir filter. I think these two systems should be the same (...)
hi i need a code for fir ,iir filter, i'm using ccs3.3 320c5x/6x. i cant create a code. could u pls send the download link or send a file to my mail.thankx ram
Hi, I have some data in tow arrays, containing the data and timestamp. For example data(1)=10 timestamp(1)=100 data(2)=11 timestamp(2)=110 et cetera I want to filter the data using fir/iir implementations. For this two things need to be done * Extend the data series, so for t=100...109 data=100 * Do the (...)
first you can use fdatool for matlab then mention the type of filter you would like to design fir or iir ? see the signal processing toolbox in matlab