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248 Threads found on edaboard.com: Impulse Response
Hi , I am trying to design comb generator of 30 MHz to 1 GHz using multisim 12.0. I got the impulse response but its amplitude is decreasing as the frequency increases as shown in figure attached. What can i do to get smooth comb like output in spectrum analyzer?
At the final expression of H(s), the exponential has -s*T1 which represents a delay. impulse response=1 from 0 to T1 and then 0 from T1 to infinity. Step response = linear ramp from 0 to T1 and then is T1 from T1 to infinity. - - - Updated - - - Please I would appreciate if som
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 filter: H(w) = (1/L) * (1 -
Hi, I simulated a UWB vivaldi antenna in HFSS. I would export S11 as .csv file from HFSS and import it in MATLAB. The purpose is to find out impulse response h(t) of UWB antenna. After importing .csv file of s11 from HFSS, what I need to perform to get impulse response h(t) of that antenna? Thanks
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 is made up of (...)
So say i want to input a impulse signal to my systems then my input is nothing but a rectangular waveform??? Am i correct about this???Correct. You have to consider mainlobe of Spectrum. Spectrum of Rectangular waveform is Sinc function. Mainlobe of Spectrum has to be far wider than bandwidth of your system.
There's nothing wrong with the filter. But you see that the second filter still passes part of the unwanted signals. Furthermore, you need to wait at least the threefold time for complete settling of the output signal. Finally there's a relation between the length of the impulse response (Fs*filter order) and the minimal achievable bandwidth.
Hi, Is there a block in virtuoso where I can input the time domain impulse to get the required characteristics? The reason why I am asking is because I am trying to model an optical channel which has a gaussian impulse response and modeling the transfer function using svcvs is a bit tedious. I have the time domain impulse (...)
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 (...)
Convolution is used when your system has the following two properties 1 Linear Time Invariant: The output of the system does not change if you apply same input at another time. If your input is combination of inputs like x1 + x2 then the output should also be the sum of two individual outputs y1 + y2. 2 Finite impulse response: If you apply an im
Hi, I am little confused about channel propagation of a wireless waveform. I know that y= h*x + n -------- in time domain Y=HX+N --------in frequency domain where h is the channel impulse response which can be rayleigh or rician if small scale fading is considered. But what happens in large scale fading consideration? H t
Hello, please it is possible to find the impulse response of a telecommunication system in HFSS Thanks in advance
Hi, How can I represent the convolution process between a signal comes from Inverse Discrete Wavelet Transform (IDWT) and the channel impulse response? Is that like the convolution between signal and channel impulse response in time domain? Regards
what is difference between impulse response and coefficient of digital filter in FIR.
Hello, I'm trying to determine the doppler spectrum for a specific tap, using channel impulse response. There is how I proceed to determine the channel impulse response : - I have real and imaginary part from ray field E - magnitude = abs(real + imaginary) - phase = angle(real + imaginary) - channel (...)
1. Find the impulse response of the system by taking the inverse fourier transform of the frequency response. 2. Convolute your input with the impulse response. Or just use a transient simulation.
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 (sin(x)/x)^n, you won't
Hi everyone, After somme researchs on the web, I don't find the answer of my problem (or I don't understand it) and I hope this post will succeed. I'm working on a real-time FFT convolution algorithm (C++) which split the impulse response in several increasing size blocks (64/128/256/...). To realise the convolution, I have to split the incomi
All basic signals have specific purpose impulse -used to get frequency response of system step- used to get time response of system and it also used to extract some portion of signal signum-? ramp-? my i knowthat use or application of above two signals.Thank you in advance.
Hi all, I am trying to implement STTC-OFDM (2Tx-1Rx) over frequency selective channels(2-taps or 3-taps and so on). I need a modification on my code over single tap quasistatic flat fading environment. My question is how can I use the channel impulse response for multiple tap channels on the maximum likelihood decoder to calculate branc metrics.