# Questions tagged [sampling]

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

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### 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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### 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 ...
61 views

### 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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### 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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### 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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### 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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### ZOH non ideal sampling loss formula

Hello i am trying to calculate the amplitude loss of non ideal sampling. My signals amplitude is 10 at 70 frequency. When we sample it at Fs=400 we have al 0.511 loss as shown in plot bellow. How can ...
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### CTFT to DTFT why can't you always just substitute $\Omega = \omega/T_S$

This is something I've always wondered about in DSP class, but just accept as a fact because I never really fully understand why this is the case: Given CTFT: X_s(j\Omega) = 6000 \pi \sum \limits_{...
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### Difference between frame rate and sampling rate?

Below are the two different methods of generating time to create an audio sine wave. While framerate is being used as steps in one method, the inverse of sampling rate is being used as steps in the ...
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### Contradictory results from power spectral densities and autocorrelation in shot noise

I have a white noise generator circuit which involves a Zener diode meant to undergo breakdown across a resistor $R1$, and am trying to compare power spectral densities and autocorrelation of the ...
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### Time stretching allowing for pitch change

I have implemented the WSOLA time stretching algorithm for audio data, but there are still many artifacts from the stretching. I don't actually care about maintaining pitch and am totally ok with ...
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### Reconstruction using sinc

The signal that I produce above. What is the reason for it to slide to the right? In oversampling at Nyquist rate can I make like below picture ? do you think the signal i produced at nyquist rate is ...
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### Sampling and Reconstruction of digital signal in Matlab

I'm trying to write a program in Matlab that samples (using Nyquist theorem) and recovers signal. However, I cannot write sampling part for sum of 2 signal. ...
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### Any DVB-S2 sample streams publicly available on-line?

I'm working on a DVB-S2 receiver and I'd like to be able to test a part of the DSP chain with real-world signals, but getting good sample streams from a digitizer is not terribly easy. Can anyone ...
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### Downsampling issues

I have this data that need to be downsampled so that it can be fed into a classifier. The issue is that I need to downsample from 1000Hz into 20Hz. If you see the graph below, the blue line is the ...