Questions tagged [sampling]

In signal processing, sampling is the reduction of a continuous-domain signal to a discrete-domain signal.

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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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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
45 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 ...
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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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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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3answers
75 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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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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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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down sample and up sample

Does these codes hold true for up sampling and down sampling processes? ...
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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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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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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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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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39 views

The impact on the frequency of adding zero samples and non zero samples

Can someone give me a brief explanation of what is going on in here? I struggle to understand the full differences between the impact on the frequency of added zero samples and adding samples which ...
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Cosine of amplitude 1 but GNU Radio shows FFT has amplitude > 30,000

I am doing an FFT on a cosine wave which has an amplitude of 1. The FFT is amplitude is not 1 but over 30,000. What am I missing here? It's from the GNU Radio FFT wiki page:
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How to save Fourier series approximated signal to a WAV file

I changed this Matlab/Octave code to approximate square wave by using combination of Fourier series and Fejér taper: ...
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4G LTE Sampling Frequency Offset

Could someone please explain how the 4G LTE OFDM measures and corrects the sampling frequency offset? I can see a number of algorithms proposed in the literature, but I am trying narrow it down to the ...
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Find the frequencies present when a non-sinusoidal wave is reconstructed

A 1.6kHz periodic (not sinusoidal) signal contains all its harmonics, both even and odd. The signal is sampled at 3.6kHz and ideally reconstructed. What frequencies are present in the output? I know ...
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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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Filtering a noise of F[sig] frequency [Example 20Khz] in measurement system

I am developing a system where there is inherent noise of F[sig] frequency. This is a DC measurement system and want to get rid of the noise by sampling at F[samp] samples per rate and taking multiple ...
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47 views

Where does the imaginary part of a singnal come from? [duplicate]

So I recently watched this lovely overview of DSP/SDR explaining how demodulation of AM and FM becomes trivial by turning a signal into a complex number. (AM being the magnitude of the number and FM ...
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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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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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How to calculate sine wave SNR from oscilloscope

I'm trying to calculate the SNR of a sine wave produced with a 12 bit DAC. I'm new when it comes to calculating/measuring noise in signals. My idea was to capture the signal using an oscilloscope (...
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1answer
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Interpreting the cross correlation results

I am new to this forum. I am just trying to understand how to calculate delay (in terms of samples) between two signals using cross-correlation function. I wrote a small Matlab code in order to ...
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How to sample along diagonal in 2D FFT

I aquire typically 2D time traces which I process by a 2D Fourier transform. The aquisition is done line by line. First, I let the signal evolve for t1, then an action is performed that possibly ...
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1answer
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Convolution : Anechoic Speech signal and Impulse response

I am trying to perform convolution in MATLAB of an anechoic speech signal (2.1 s) and a impulse response (20 s) using the following code: ...
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Sampling of bandpass signal with bandpass filter

Trying to solve the following problem about bandpass signal sampling. I have a signal whose Fourier transform is such that $X(w)= 0$ if $w>w_h$ or $w<w_l$. The reconstruction is done using a ...
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42 views

Frequency response and sampling theorem for triangular function

The triangular function is defined as follows: $h_l(x) = \begin{cases}1-|x|,&|x|<1;\\0&\text{otherwise}.\end{cases}$ According to ccrma.stanford.edu: "If the output of the interpolator ...
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34 views

Max frequency of cosine in SDR

I would like to transmit a 56 MHz real cosine and receive it using SDR with max complex sample rate equals 61.44MSPS and an analog filter with BW = 56MHz (the transceiver is AD9361). Can I do it? How ...
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Impulse Invariant method for digital filter design

One of the known methods for discretizing analog filters is impulse response invariant. We get the impulse response in time domain, discretize it and then get the Z transform. What I am trying to ...
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1answer
84 views

Find continuous signal given a condition on its samples

Let $x(t)$ be band-limited with $B = \omega_m$. Sampling gives us $$x(nT_s) = \begin{cases} 1, & n = 0 \\ 0, & n \not = 0 \end{cases}$$ And $\omega_s = 2\omega_m = \frac{2\pi}{T_s}$. Find ...
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Reconstruct $x(t)$ from $y(t)$ and $z(t)$

Let $x(t)$ be band-limited signal with $X(j\omega) = 0$ for $|\omega|\gt \omega_M$. We use $$s(t) = \sum_{k =-\infty}^{+\infty}(-1)^k\delta(t - \frac{kT}{2})$$ for sampling. So we have $z(t) = x(t)s(t)...
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60 views

Fourier transform of the sampled signal

I want to calculate Fourier transform of the sampled signal in two ways. Let $$s(t) = \sum_{k = -\infty}^{\infty}\delta(t - kT)$$And $z(t) = x(t)s(t)$. So we have $$z(t) = \sum_{k = -\infty}^{\infty}x(...
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How to convert ofdm symbol to signal?

Hello may I know how to convert ofdm symbol generated from ifft block to an OFDM signal for the purpose of transmitting it?
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Meaning of sampling frequency?

Say there is a communication system, which can manage sampling frequency up to 4 MHz (given by software). Can the system be able to process signals with maximum frequency more than 2 MHz? Because ...
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going from output of D/C to input of D/C

$$y_c[n] \longrightarrow \boxed{D/C} \longrightarrow y(t)$$ If the following equation describes the IDEAL D/C converter: $$y_c(t) = \sum \limits_{n=-\infty}^{\infty} y_d[n]~ \text{sinc}\Big(\pi (t - ...
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Sample and Hold Circuit at Digital to Analog Converters

Does DACs contain sample-and-hold circuits? I am a bit confused thanks
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Aperture Jitter in Digital to Analog Converters

I have noticed in the literature that there is an internally induced timing jitter inside Analog to digital converters called (aperture jitter). It is caused by the sample-and-hold circuit of the ADC. ...
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Ideal low pass filter output at given sampling frequency

Consider the signal $\cos(30t)$ sampled at $w_s=40 rad/s$ using a unit impulse train. The sampled signal is filtered with an ideal low pass filter with unity gain and cutoff frequency $w_c = 40rad/s$. ...
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How should I sample the signal $f(t) = \sin(24t)^3\chi_{[0,0.52]}$ on the interval $[0,1.25]$?

I am from an applied math/PDE background and I don't usually deal with proper sampling and signal processing. I just take roughly what seems to be a good number of samples. But I would like to be ...
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1answer
39 views

Oversampling a signal

I’ve built a circuit with an analogue sensor (70 Hz bandwidth) connected to a 24 bit ADC, which is oversampling the signal at 10 kHz. Nyquist sampling theorem implies I should sample by at least 140 ...
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Oversample or undersample correlated noise?

I wish to measure nontrivial (non-Gaussian) broadband current noise for its distribution and power spectral density. The noise is amplified (with transimpedance amplifier) with some characteristic ...
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DFT without knowing the exact value of sampling rate?

I have a sequence of (real) numbers that represent the magnitude of a certain natural event. I know that the samples are equispaced in time, but not the exact value of the spacings. So does that mean ...
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1answer
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Number of samples in a discrete signals after sampling a continuous periodic signal should be integer?

If a continuous periodic signal is sampled with time interval Ts , and T being time period of continuous signal -- For the resultant discrete signal to be periodic following must be true-- the ratio ...
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Upsampling implementation

I have problem with upsampling implementation in the following way: My code: ...
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How are these 2 frequencies aliases of each other?

I'm trying to understand this solved example from a book, given an analog signal $x(t) = 3\cos(100\pi t) $. This signal has a frequency of $50hz$. If sampled at $F_S=75Hz$ ,I obtain the discrete time ...
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Sampling: question regarding sampling period

what exactly happens when I increase the sampling period ? Providing some Matlab plots so that I can be more clear. Using the signal $x(t) = \sin(10\pi t)$ and periods $T_s=0.02 ,\; 0.05 ,\; 0.1$ ...
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Multichannel sampling with aliasing

if i have 2 sensors, sensor1 and sensor2, that sample a signal on complementary points so that sensor2 samples always between sensor1s sampling points. can I achieve the doubled sampling rate with the ...

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