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67 Threads found on edaboard.com: Freq Sampling
add a ramp signal as the source signal. waht's the relation between the freq of ramp and the freq. of sampling? i means what freq of ramp is suitable for testing the static performance . and what about the freq of sin wave for testing the dynamic performance, what's the coherent freq?
Hello everyone, Thanks in advance for your kindly help. I'm to use a bandpass filter (preferably Gaussian window filter with 2.4 MHz and 5 MHz cutoff freq.s) to filter out my discrete samples array (8017 samples, sampling rate= 100 MHz), and then obtain the output in C language. Using Matlab fdatool, I obtained the necessary filter coeffici
Hi, To convert speech signal into binary, perform the steps - sampling, QUANTIZING AND ENCODING. Step 1: Sample the signal according to sampling theorem i.e., use a sampling frequency greater than the highest freq. component of the message u'r case, the highest freq. component of (...)
Why delay line is used before ADC? Is it for sampling pupose? How much delay should be in a sytem whose external freq is 1 MHz and internal 6 to 7KHz?
hi all, if i have a discrete integrator with t.f of Z^-1/(1-Z^-1), can i implement in veriloga using the Z-Transform Filters module? if so, what should i use as the sampling period of filter? the switch freq which samples the signal? thank you.
1. There is no relation, unless you limit your discussion to a particular type of scope. For example I have a scope with 500 MHz bandwidth and 5 GHz sample freq. I have another scope with 15 GHz bandwidth and about 200 kHz sample freq. My analog scopes don't sample at all. 2. Need more info about the particular scope, and which characteristic of
I think, Its wrong to say that "ordinary sampling carried out at twice the frequency i.e Nyquist criterion?" In fact, IMHO, bandpass sampling is what comes in Nyquist criterion. B'cos it says twice the freq of bandwidth not twice the absolute freq. To be more clear bandpass sampling means (...)
How to do PSS analysis in cadence spectre..I had given PSS run for SC filter with sampling freq=10MHz and fun=3KHz.Its taking very long time to simulate..I am not ablet debug the problem.plz help me?
hey...here is wat i think may help u.... 1. a filter before the a/d is always advisable.... but u need to know abt the carrier freq to filter it. 2. schmith trigger can be used only if u can make up with certain constrains...like u need to clearly know the delay wid whcih the signal is comin to ur samplin point and even the sampling (...)
An analog, low-pass filter prior to the input of the ADC eliminates or significantly reduces aliased noise. The corner frequency of the filter must be lower than fsampling/2. Increasing the order of the filter can minimize the noise.
If you don't mean sample freq, it is the setting time of SHA. It depends on the resolution you want, (for ADC or something) .
isnt DTFT the fourier transform of a discrete time signal,so isnt that also discrete in the frequency domain too.. and incase my signal is of and 5 Khz1Khz and sampling freq is 10KHz ,so will a 10 point DFT accurately capture both the frequency components
hi thank you for sharing the files Added after 8 minutes: how to design -a LPF(fir) by window method sampling freq=100hz and cut off=20khz..without using FFT builtin function. plz can u give me a code(program) to implement this using matlab softwear.please. u can use thes
Can ne1 help me how to design a LPF(fir) by window method sampling freq=100hz and cut off=20khz..without using FFT
Be clear that in MATLAB you can do only FFT which is a efficient form of DTFT. now lets take a analog signal x(n)= A sin (2*pi*f*t); ( in a PC nothing is analog)... so lets make it Discrete ... and the sampling freq fs = ( 1/Ts) lets and fs > 2*fm .. fm = max freq in our input signal lets take Ts=0.1 for a period (...)
Try this FFT in spice or spectre: N/(FFT point) = Fin/(sampling freq) N prime number 7,9,13,17,19,23,29,31............. FFT point : power of 2 , 1024,2048,4096,32768,65536 , larger will give a lower FFT noise floor Fin : your input signal
hi all, does anyone knows how to convert actual freq to normalized freq from ? Eg: Actual freq=500hz. sampling freq=5khz. thanks regards scdoro
I need circuits that can delay a signal with certain freq (1Hz - 100kHz) so the output signal is the same shape and form and frequency as the input but with delayed output appearance Bottom line is i want to reduplicate a signal with variable frequency so that the output is delayed X seconds from the input Any idea?? :cry::cry:
"unmatched local osc" means the the RF upsampling oscillator at TX and the RF downsampling oscialltor at RX.. for eg WLAN OFDM (802.11a ) its in the range of 5GHz ... the difference in sampling point and difference in sampling frequency is what which causes the CFO.
zero padding can not increase sampling rate in freq domain, it increase resolution of frequency domain.