Questions tagged [sampling]
In signal processing, sampling is the reduction of a continuous-domain signal to a discrete-domain signal.
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Sampling and reconstruction of signal in Matlab
I'm trying to write a program in Matlab that samples (using Nyquist theorem) and recovers signal. I got stucked on recovery part...recovery signal doesn't match with the original one (see photo).
And ...
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1answer
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Can amplitude aliasing occur in a bandlimited signal? [closed]
This question drew much controversy on whether "amplitude aliasing" can occur for a Nyquist frequency bandlimited signal - and no consensus was reached before the question was closed as &...
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Nyquist frequency for amplitude? [closed]
$f_s > 2 f_{\text{max}}$ prevents frequency aliasing for a bandlimited signal, but not amplitude aliasing:
("Actual" curve in grey; blue is what we get if taking samples (dots) "at ...
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1answer
102 views
Sample-rate, filtering, digital-filtering and aliasing
I am strugling with a question that I hope someone can help me with.
I am recording single molecule events which I detect is picoampere square deflections.
I wish to use as gentle low-pass bessel ...
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2answers
225 views
Increasing SNR and Dynamic Range using Oversampling
How much gain in dynamic range and SNR can be expected if we are to oversample a signal with fixed analog input bandwidth. For Example if I have a analog filter at the input which limits the bandwidth ...
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2answers
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Coherent Sampling vs Windowing for 50/60 applications
Let's say I need to sample 50/60 Hz signals to find RMS. I can use for example 1kHz sampling frequency. That means 100ms sampling is 5 cycles of 50Hz and 6 cycles for 60 Hz. That's coherent sampling. ...
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1answer
117 views
the number of multiplications per second required by the system
Q1. Consider a system implementing a rational sampling rate change by 5/7: for this, we cascade upsampler by 5, a lowpass filter with cutoff frequency π/7 and a downsampler by 7. The lowpass filter is ...
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63 views
Aliasing at $f_0 + kf_s$
In this question, it has been proved that $x(t) = \sin(2\pi f_0 t)$ and $x_k(t) = \sin(2\pi (f_0 + k f_s) t)$ have the same sample points. So sinusoids with frequencies $f_0$ and $f_0 + kf_s$ will ...
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Sampling frequency error in OFDM
If the sampling rates at receiver and at transmitter are different, there will be ICI.
However I don't find any practical technique to estimate and compensate this kind of error. Is it true that we ...
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2answers
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How to reconstruct a time-discrete signal
German: Sie betrachten das zeitkontinuierliche Signal $s(t)$ mit einem digitalen Oszilloskop mit Abtastperiode $T_\mathrm{s}$ und messen das zeitdiskrete Signal $\tilde{s}[n] = s(n\:T_s)$
Nehmen Sie ...
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1answer
45 views
Recoverable continuous function after average sampling
I have a personal interest in this question, but I don't know if it is unfounded.
Suppose I have a continuous function x (t), which is sampled using an average sliding window, which performs a ...
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1answer
50 views
Inverting a sampled system
I'm doing some self-study for an upcoming exam and came across the following question:
My first idea to solve it was using the bilinear transform to get some approximation of $H(Z)$ (or just using ...
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fo we need RMS in SNDR of harmonic distortion
In my system we have the following.
V_in=1+cos(2pi900KHz)
V_sample=Vin-0.01Vin^2
V_noise=20010^-6V
After the input i get in V_samples the following coefficiets of harmonics in the output as shown in ...
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2answers
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Why negative harmonics copies in sampling
i have a signal of 120Mhz and sampling at 800Mhz. we look at the region of fs/2=400MHz so obvios we have harmonics of the fundamental which is 240 360. But in the document i see that there is another ...
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1answer
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why we do multiple aliasing of high order harmonics
the general rule for aliasing.
If my sampling frequency is Fs=800Mhz
signal frequency =120Mhz.
for the 4th harmonics 480MHz
its 80MHz above nyquist frequency(400Mhz) thus its mirrored 80MhZ back to ...
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26 views
Goertzel filter for non-equidistant sampling
We have a sensor that reports a new value asynchronously only when the value changes. Therefore, we receive samples at arbitrary time stamps. We basically receive value + timestamp of the measurement ...
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1answer
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Multiplying signals in discrete-time vs continuous-time
Given two discrete-time signals $a[n]$, $b[n]$ and its product $c[n]=a[n] b[n]$. The ideally interpolated, continuous-time version of $c[n]$ is
\begin{align}
c_1(t)&=\sum_{n=-\infty}^{\infty} a[n] ...
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Downsampling vs. ADC with lower Sample-Rate
I would like to know in general where the advantages and disadvantages are:
Downsampling of a high sampled continuous analog signal in FPGA or µC and
Direct use of a lower sampling rate, i.e. using ...
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2answers
91 views
OFDM sampling frequency offset
how the 4G LTE Or 5G NR OFDM measures and corrects the sampling frequency offset? if there is an algorithm using matlab please help
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1answer
71 views
Sampling Dirac function and a DC signal
Can we sample the Dirac function?
$$ x(t) = \delta(t) $$
Can we sample a DC signal?
$$ x(t) = 1$$
I think that we can't sample $x(t) = 1$ because the Fourier Transform of $x(t) = 1$ is $2\pi \delta(...
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What is the multiplication rate in FIR filter
Consider a system implementing a rational sampling rate change by 5/7: for this, we cascade upsampler by 5, a lowpass filter with cutoff frequency pi/7 and a downsampler by 7. The lowpass filter is a ...
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0answers
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Showing that filtering a signal with bandwidth B with a brickwall filter of bandwidth W>B has no effect in time domain
The time-domain representation of $G(f) H(f)$, where $H(f)$ is an ideal brickwall filter of bandwidth $1/(2T)$ is:
$$
\int g(\tau) \operatorname{sinc}\left(\frac{t-\tau}{T}\right) d\tau
$$
I want to ...
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1answer
100 views
Is spectral density sometimes normalized by sampling rate rather than bin size?
I'm a scientist conducting an experiment that requires some signal processing. My expertise is not in signal processing, thus here I am. We've basically re-created an experiment conducted by other ...
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1answer
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Partial column approximation error
Can someone help me understanf how to plot the next partial column in MATLAB:
while N=2001
|M|<700
ak =
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2answers
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Can I sample at Nyquist rate if I know that my samples are taken only at the signal's maxima or minima?
I know that in general the sampling rate, $f_s$, must be greater than twice the highest frequency of the signal, $f$.
If I sample at the Nyquist rate, it can lead to the following:
However, if the ...
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1answer
518 views
Are resolution increase and noise reduction from oversampling mutually exclusive?
Oversampling a signal means sampling it with a significantly higher sampling frequency than the Nyquist rate. As far as I know, there are three advantages:
Easier design of anti alias filter
Increase ...
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2answers
71 views
D/A converter with
I am trying to figure out how to consider digital-to-analog conversion and how to consider oversampling based on a course I am taking online and figuring out the logic behind this. Unfortunately there ...
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1answer
65 views
number of multiplications per second required by the system
Consider a system implementing a rational sampling rate change by 5/7: for this, we cascade upsampler by 5, a lowpass filter with cutoff frequency π/7 and a downsampler by 7. The lowpass filter is a ...
1
vote
3answers
94 views
How to distinguish between FFT power leak or close frequencies
This week I was having an interview with a Professor for Grad School. He asked a bunch of questions related to signal processing that lead up to some questions about power leak on the FFT spectrum. So ...
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How to characterize the randomness of an event using it's PSD?
I have the power spectral density function of a stochastic phenomenon.
how can I generate a signal (time series) representing the randomness of this event over time?
How can I draw the probability ...
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minimum oversampling factor for D/A converter
Consider a D/A converter for audio signals consisiting of a zero-order-hold interpolator followed by a continuous-time lowpass filter with positive passband between 0 and 20KHz and stopband starting ...
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2answers
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Exponential moving average cut-off frequency
I am trying to implement low pass filter from this example.
What is the cut-off frequency for this type of filter? Is it $F_s \frac{1-\alpha}{2\pi\alpha}$, where $F_s$ is sampling frequency?
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1answer
28 views
Modulation upsampling method
I try to implement a qpsk modulator in Octave.
The chain is the state of the art I guess: Mapper -> Upsampling x4 -> Root raised cosine filter.
My main question is about the upsampling.
I try to ...
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0answers
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down sample and up sample
Does these codes hold true for up sampling and down sampling processes?
...
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2answers
79 views
Correction of amplitude after zero padding for upsampling purposes
I have time sequence in which the data is sampled at 0.8 Hz. The data is related to chromatography (chemical analysis), that is why the sampling frequency is relatively low. The instrument cannot ...
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Confusion in property of Z transform of ideal sampled waveform
I was reading about z transfom of ideal sampled signals and one of the properties of Z transform of sampled signal that surprised me,here it is (image)
So here this property of Z transform is quite ...
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1answer
152 views
Continuous wavelet transform subsampling at lower frequencies?
I am not completely confident in my understanding of wavelet theory, but since I am currently in the process of creating a CWT scalogram tool, I have to ask:
Is there a theorem allowing us to ...
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1answer
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Sampling the Wave Equation for String Vibration
I am having a difficult time understanding the basic principles when sampling the function describing the vibration of a string. Mainly I am confused on given a sampling rate how many samples to take ...
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1answer
320 views
Choosing a Sampling Rate and a Cutoff frequency
I have an assignment:
You wish to generate a pure 1000 Hz tone digitally using a computer. How would you choose a sample rate that assures that you could generate the tone and use the same sample ...
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3answers
3k views
Is sampling at double the desired reproduction frequency accurate?
This is probably a general principles question, though I'm thinking specifically in relation to sound, which is commonly sampled at a rate of 44.1 kHz in part because the maximum frequency the average ...
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0answers
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Variable Window Size of Blackman FIR filter
I am trying to implement an FIR filter with a Blackman window function. The solution I got for my signal filtering problem is to have different window sizes for certain characteristics of the input ...
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2answers
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What is the difference between the sampling frequency of signal and sampling frequency of filter
I believe that there is no connection between the sampling frequency used for converting an analogue filter to digital filter and the one used to sample a signal that the filter will be used on. But I ...
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1answer
55 views
Oversampling, decimation and Nyquist
I know this is a bit of a fundamental DSP question but I want to finally understand this and I think I need some help to finally put it all together.
Let's say I have an RF complex sampling ADC with a ...
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1answer
100 views
Sub nyquist sampling, required number of samples for time sparse grouped signals
Question: Does it make sense to perform compressed sampling if the non zero samples are grouped in time? If so, what is the minimal length of the vector x that should be acquired to allow full signal ...
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1answer
130 views
How to make low pass filter using frequency sampling method?
https://www.allaboutcircuits.com/technical-articles/design-of-fir-filters-using-frequency-sampling-method/
So there is two main equation:
I wish to filter out frequency $\le 10000Hz$, for example. So ...
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1answer
109 views
Signals sampling
I have a simple question, but sadly I'm kind of "noob" in signals theory.
A signal having 4 harmonics at the following frequencies: 1 kHz, 2 kHz, 3.5kHz and 4.2 kHz.
(How can a signal have harmonics "...
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1answer
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Multisampling of an ADC channel in a PWM cycle in TI C2000
I want to sample an ADC channel multiple times in one PWM cycle.
I can trig any ADC channel in anywhere in my PWM timer counter now. But I want to trig and sample same channel in many times within a ...
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1answer
276 views
Low-Pass Filtering of not evenly sampled signal
I have a signal sampled unevenly over 1 million ns.
the signal is sampled over 1GHZ clock and the samples are as the following:
0-100 ns - sample every 1 ns.
100-1000 ns - sample every 10 ns.
1000-...
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1answer
91 views
Representing a positive 0-limiting continuous time domain function, in terms of spectral samples, outside of a window
Bit of an odd question
Assume you have a known function $x(t)$ which is continuous, but approaches but is not equal to, 0, on the interval $[-\infty,\infty]$ . You know also that it approaches 0 ...
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1answer
51 views
Multiband undersampling
In Practical Signal Processing (Marc Owen), an exercise explore the topic of undersampling:
[...] Suppose you know that an audio signal is a sine wave with a frequency that might be in one of two ...
