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83 Threads found on edaboard.com: Filter Impulse Response
I have a signal cos(w0 * t). It is sampled at frequency f_s. The samples are applied to a moving average filter of length L. I want to find the amplitude of the output. So far I understand the following. Please do correct me if I am wrong; I am still learning the basics. DTFT of impulse response of the moving average (...)
Hi i have attached an image of a simple low-pass filter. To the left of the image the impulse response is something decreasing in time, while the impulse response in the right is in the sinc function. I can understand the sinc function since the input impulse is a rectangular waveform which (...)
I try to simulate a noisy signal and filter it. the signal mix some low frequency signals and some random noise. my goal is to get 14.8Hz signal. my band-pass bandwidth is 14.7Hz to 14.9Hz. function filteringTest Hd = Kaiserfilter; Fs = 4000; % Sampling frequency T = 1/Fs; % Sample time L
Build FIR filter model using Verilog-A. You can find many examples in "rfLib" of Cadence dfII.
A Linear Time-Invariant system with impulse response h1h1 is an ideal lowpass filter with cutoff frequency ωc=π/2ωc=π/2. The frequency response of the system is H1(ejω)H1(ejω). Suppose a new LTI system with impulse response h2h2 is obtained from h1h1 by (...)
what is difference between impulse response and coefficient of digital filter in FIR.
The question title involves a contradiction in terms, because a CIC decimator is a filter with finite impulse response as well. So more exactly you are asking about using different FIR characteristics than CIC as a decimation filter. The general answer is quite simple, if you want a different frequency characteristic than (...)
Hello. I am unable to understand what you want to ask. I guess you are asking on developing a finite impulse response filter.. what you need is get the filter coefficients.. if the coefficient is in time domain, you just need to convolve with the signal you need to filter..
Hi friends, I want to design a adaptive digital FIR filter using verilog. I studied the base paper. But i am not clear with the multiplication of impulse response with the input sequence. And i also dont get about the difference of Normal filters and adaptive filters. Pls explain me briefly with (...)
hello dear, i generated a target impulse response of fir low pass filter and then interpolated it , now i want to generate ternary tabs or coefficient using second order sigma delta modulation, please tell me how to generate it, if any one have code or idea
hello all , I need to implement CIC decimator, here is the matlab code using fvtool r=64; m=1; n=3; iwl=16; owl=12; IFL = 0; % Input fraction length CIC = mfilt.cicdecim(r,m,n); CIC.InputFracLength = IFL; f_in =5120; h=fvtool(CIC,'FS',f_in); I checked the impulse response using the fvtool and it gives me a complete different respo
hello all , I have a problem when designing CIC filter on matlab here is the code : r=64; m=1; n=3; IFL = 0; % Input fraction length CIC = mfilt.cicdecim(r,m,n); CIC.InputFracLength = IFL; I use fvtool to check the impusle response and I got figure 1 attached , 9330093299 I also use
I am currently designing a Manchester matched filter design in Matlab. My task requires me to create and plot the impulse response h(t) of the optimum receiver filter matched to detect symbol '1'. I have created a matched filter by using the following code: h=fliplr(p); where p is my manchester (...)
Can annyone explain to me what is the impulse response for this half band filter mean? What is the purpose to simulate this impulse respone? 89782
Guys, I have created a filter from FDAtool and want to test it on Simulink but I didn't see any response on the Scope I used discrete impulse as input What can I do to see the output ? Thanks I get this warning : Warning: The model 'FIRfilterTest' does not have continuous states, hence using the solver (...)
Hi, I'm trying to implement the overlap save method in matlab in order to clear up noise from a wav file. The function accepts the following fields: x = long sequence to be filtered (from wav file) h = impulse response of filter (loaded from a different file) N = Block length used in the algorithm ( i.e. size of the (...)
Linear phase is in fact a well defined term (other than phase change). Strictly spoken, only FIR filter with symmetrical impulse response expose linear phase (respectively constant group delay). CIC decimators belong to this class of FIR filters.
Hi, Indeed, each bin of the DFT is the result (at one specific time) of the output of the input sequence applied to a FIR filter. The "impulse response" of the filter associated to the k-th bin is a complex exponential that has k cycles in the duration of the data. Its frequency response is maximum at the (...)
i am using matlab to analyse audio signals .i am using r2008a student version and don't have many options for plotting the frequency i ended up with 'freqz' function which plots magnitude (dB) vs normalized frequency.but during filtering i need to know the actual frequency range i am filtering.say i need to filter 700 hz to 1600 hz .how
One more thing: I find that the impulse response of a bandpass filter seems to be differing depending on the damping coefficient. If we have damping coefficient less than 1 then we will be getting sine waves in the response right? In addition, also the impulse response of high- and (...)