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98 Threads found on edaboard.com: 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
what does the order of a filter (fir or iir) signifies How to determine the order of filter
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 (...)