Signal Processing Stack Exchange
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Questions tagged [sampling]

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

A common example is the conversion of a sound wave (x(t), a continuous-time signal) to a sequence of samples (x[n], a discrete-time signal).

A sample refers to a value or set of values at a point in time and/or space.

A sampler is a subsystem or operation that extracts samples, x[n]=x(nT), from a continuous-domain signal. A theoretical ideal sampler produces samples equivalent to the instantaneous value of the continuous-domain signal at specified sampling instances.

Source: Wikipedia.

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If humans can only hear up to 20 kHz frequency sound, why is music audio sampled at 44.1 kHz?

I read in some places that music is mostly sampled at 44.1 kHz whereas we can only hear up to 20 kHz. Why is it?
Soham De
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"Complex sampling" can break Nyquist?

I have heard anecdotaly that sampling complex signals need not follow Nyquist sampling rates but can actually be gotten away with half Nyquist sampling rates. I am wondering if there is any truth to this? From Nyquist, we know that to unambiguously…
Spacey
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What is meant by "stochastic sampling"?

What exactly is meant by "stochastic sampling" and is it profoundly different from the regular Nyquist-Shannon sampling theorem? Is it related to sampling a stochastic process?
Phonon
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When is aliasing a good thing?

In Hamming's book, The Art of Doing Science and Engineering, he relates the following story: A group at Naval Postgraduate School was modulating a very high frequency signal down to where they could afford to sample, according to the sampling…
datageist
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How to Extrapolate a 1D Signal?

I have a signal of some length, say 1000 samples. I would like to extend this signal to 5000 samples, sampled at the same rate as the original (i.e., I want to predict what the signal would be if I continued to sample it for a longer period of…
PearsonArtPhoto
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Why do we choose 44.1 kHz as recording sampling rate?

Peoples' ears can hear sound whose frequencies range from 20 Hz to 20 kHz. Based on the Nyquist theorem, the recording rate should be at least 40 kHz. Is it the reason for choosing 44.1 kHz?
new_comer_forever
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Why would I leave a signal oversampled?

I can't think of a better way for asking this question so I will start with an example. Suppose that I have an input signal with a max frequency of 50Hz (sampled at 100Hz). Now the signals of interest lie in the range 0-5Hz, so I can add a low-pass…
anasimtiaz
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What sampling frequency should I use if Nyquist is not available?

I have the following homework question that confuses me: We have an audio emitter that can emit two signals: It either emits a sine wave at 23 kHz or it emits a sine wave at 25 kHz. The receiver has the following sampling frequencies available:…
NN amateur
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Downsampling an image by an integer factor

When downsampling an image by an integer factor $n$, the obvious method is to set the pixels of the output image to the average of the corresponding $n \times n$ blocks in the input image. I remember vaguely having read somewhere that this method is…
Styg Oldenbaum
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What are the advantages, if any, of derivative sampling?

In Five short stories about the cardinal series $[1]$, the author makes the following comment: Interestingly enough, Shannon goes on to mention that other sets of data can also be used to determine the band-limited signal--for example, the…
datageist
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Converting frequency from $\textrm{Hz}$ to radians-per-sample

In MATLAB I have to pass cut-off frequency for designing a filter. But this Cut-off frequency is in radians-per-sample. How do I convert my analog Cut off frequency in $\textrm{Hz}$, into the required radians-per-sample for MATLAB?
gpuguy
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Is there such a thing as band-limited non-linear distortion?

So if you generate a square wave by just switching a signal between two values, at sample boundaries, it produces an infinite series of harmonics, which alias and produce tones below your fundamental, which is very audible. The solution is…
endolith
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Why is telephone audio sampled at 8 kHz?

When did we decide to sample telephone at $8$ kHz? Has this always been the case? Why did we do that? Is it because higher bit rates can't be transferred as quick? And do these reasons still count? And if not, why isn't there a new standard yet? Is…
Lewistrick
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The current state of the art in audio signal interpolation

Three questions: What are all the metrics one can use to measure audio interpolation quality, objectively? (but also in terms of psychoacoustics if possible) By those metrics, what is the current state of the art in audio interpolation? Suppose I…
Bent Rasmussen
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Can we have a Digital Anti Aliasing filter?

I am working on a board that has no antialisaing filter at the input of the ADC. I have option to I implement my own filter using RC + Opamp circuit. But is it also possible to implement Anti Aliasing filter after sampling by ADC and processing in…
gpuguy
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