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

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Using Goertzel Algorithm in under-sampling

I plan to calculate a signal's phase using Goertzel Algorithm. I have 2 signals coming to microcontroller's ADC. Need to measure the phase difference between them. Signals are 15MHz sinusoids. Sample ...
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1answer
30 views

Sample vs. sample values

Is there any difference between the terms sample and sample value in DSP? In statistics, those are very different concepts: a sample is a subset of a population: it is a set of random variables. A ...
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53 views

Recording square waves at sample frequency

For a project I am working on, I need to incorporate GPS data into an audio file. As things currently are, the easiest way for me is to record the audio in a multi-channel .wav file, and add the GPS ...
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1answer
37 views

How may samples per symbol in ieee 802.11 g

I am doing a baseband simulation for IEEE 802.11 g. The standard uses 64 IFFT length ($3.2 \mu s$) and 20 Msamples per second. This result in one sample per symbol. I am wondering what the structure ...
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1answer
16 views

Vector of random numbers with fixed sampling rate

How to create a vector of random numbers(in Matlab), when each value is [0,1] . And the sampling rate should be fixed ? thanks
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1answer
27 views

Identification of sampling frequency

I have the following vector: %create vector of sampling t=0:pi/100:2*pi; x=sin(t); What is the sampling frequency (of this vector) ? Thanks
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1answer
36 views

Is there a way to break an analog sound signal up and send it through two radio chips at double the data rate of one?

Here is a radio chip for audio that sends at a CD-level data rate. Could you sample an analog signal (for instance from a guitar pick-up or microphone) in such a way that you could transmit using two ...
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2answers
45 views

Sampling fast sinusoidal signal with low sampling rate

I have sampling problem that I am trying to solve analytically I have sampling frequency, let say 1kHz. I have number of target frequencies, let say 10kHz,100kHz,1MHz For each target frequency I ...
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1answer
50 views

What does it mean for audio to be 'sped up'? How can I undo it?

I very simply have a .wav audio file, but there is a note with it that says "This audio has been sped up by 10x". I am wondering, what does it mean exactly from a signal processing perspective for an ...
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1answer
87 views

Implementations of Low-Pass Filter

I should say which of the filters from the following image implements an ideal low-pass filter with cutoff frequency $\frac{\pi}{4}$. In the picture, Sa is a zero insertion filter, which inserts a ...
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1answer
51 views

oversampled coefficient for existing exponential smoothing

Say I have an exponential smoothing for certain $\Delta t$, $t_{i+1} = t_i + \Delta t$. In this sampling, I choose a particular $\alpha$ to filter signal $z_i$ like $$ v_1 = z_0 \\ v_{i+1} = ...
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1answer
131 views

Why can't DFT be used when samples are not equally spaced in time?

I found the following comment here The DTFT can be used when the samples are not equally spaced in time, the DFT cannot My initial thought was that this had to do with periodicity of the basis ...
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28 views

how to sample sine wave to 4 points dft output

sinusoid x(t)=1+sin(2*pi *500) sampling rate :1000 samples/s Can I have 4 points DFT output with the above data.
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1answer
37 views

Is phase and amplitude information necessarily lost when undersampling?

Is phase and amplitude information necessarily lost when undersampling if you have a constant periodic single frequency sinusoidal? My second question is: How can one determine the undersampling ...
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1answer
54 views

Ideal Reconstruction of Upsampled Signal

Problem: The signal $cos(2\pi14100t)$ is sampled at $F_s = 400 Hz$. It is then upsampled with a factor 3 and the reconstructed ideally with a new frequency $F = 500 Hz$. I now want to find the new ...
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2answers
98 views

FFT of Square Wave and Sin Wave

When we have FFT of Square wave of a pure real signal with high frequency(close to f(max), less than half of sampling frequency) we see only 2 symmetric high peaks but when we decrease the frequency ...
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1answer
32 views

Calculate Max Number Of Harmonics for given Sample Rate?

How do I calculate the maximum number of harmonics a complex wave can have based on the sample rate? Say I Have a 20hz fundamental and a sampling rate of 44100 hz. My instinct is to take 44100/2 = ...
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1answer
139 views

Sampling frequency of modulated signal

A very naive question... A signal with bandwidth of 20 MHz is modulated using a carrier of 1 GHz. What will be the sampling frequency? I was asked this question during an interview. I answered 2 GHz ...
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1answer
74 views

Does audio upsampling create noise/artifacts or degrade the signal?

I've read that upsampling performed in digital music playback can color the sound, produce artifacts, etc. For example, an audio file ripped from a CD might be 44.1Khz/16-bit, and then upconverted to ...
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0answers
31 views

Calculate the intersample peak of a periodic sequence?

Similar to Calculating the PDF of a waveform from its samples, but for periodic sampled signals, and I just want the peak. Is there a way to calculate the highest inter-sample peak of the ...
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2answers
126 views

Computing shifted signal without first reconstructing

Looking for a solution to the following problem: A signal $x(t)$ is band limited to $B$ Hz, and sampled above the Nyquist rate, with corresponding $f_s = 1/T$. If the sampled signal is given ...
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1answer
44 views

How to calculate BW ofdm?

HI I have these specification for an OFDM program I was wondering, what is the bandwidth and the sampling frequency? I'm not a student of Digital signal processing, hence I am so confused. ...
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2answers
101 views

Nonuniform sampling band-limited gaussian white noise

I'd like to sample band-limited Gaussian white noise non-uniformly. One way to approximate this would be to filter Gaussian white noise with a lower cutoff frequency and non-uniformly pick samples ...
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1answer
34 views

correcting delay between two samples

I'm sampling two signals with fundamental frequency of 50 Hz. I want to sample them at the same instant, but there's a delay between the first and the second conversion. I don't keep the previous ...
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1answer
56 views

What does “two signals aliasing to each other” mean?

So I just read something that two signals are aliasing to each other. I know what a single signal is being aliased - that is when sampling rate is not higher than or equal to the Nyquist rate. But I ...
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0answers
26 views

Matlab:Sampling Curve into Frames and Plotting

I have a Output Curve of following characteristics http://i.imgur.com/hABfsiC.jpg Following is the data which represents the output curve cited above ...
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1answer
23 views

Why do we observe band when recording CRT monitors or Flouroscent lamp via video recorder?

I understand that this has something to do with difference between refresh rate of monitor (or line frequency used by the ballast in case of Fluoroscent lamp) and frame capture rate of video recorder. ...
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42 views

Interpolating on the borders of differently-resolved images

I'm creating a three-dimensional model of the earth based on SRTM height data. The data set is pretty huge, so only a small fraction of the data is available at any given time. The height data is ...
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1answer
99 views

Java amplitude processing

I have some code that collects amplitudes from a complex sound like a song. I now want to plot a graph of amplitude vs time, for the graph plotting I use JavaFX. However I get about 9 million ...
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1answer
73 views

Why need an low pass filter after up-sampling? [duplicate]

I understand why there need an low pass filter before down-sampling, the sample frequency need to be at least twice of the max frequency signal, else there will be ...
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1answer
65 views

Dealing with Random Errors in Sampling Rates?

I wonder what generalization I can make about what happens when I introduce some error in my sampler spacing. For example, If I samples once every second I would want my sampler to take a signal at ...
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2answers
171 views

What's the Model for a DAC followed by a sampler

I have a situation where a sampled signal is passed through a DAC and re-sampled by a switched capacitor circuit. I can't see a model for this in the literature. The standard model of a DAC is the ...
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1answer
62 views

How to prove energy is preserved in sampling using Parseval's Relation?

I have a continuous signal,xc(t) which is bandlimited to =pi/T , and x[n] = xc(nT) are samples of this signal. I need to show there is a Parseval's relation between the original signal and the sampled ...
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1answer
58 views

Standard deviation of Gaussian corresponding to perfect low pass filter [1,1]

For a time discrete signal, the rect-filter (for 1D signal [1,1]) is a perfect low pass filter. I was wondering, what is the best gaussian approximation of this filter? For a current problem, I ...
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44 views

Using under-sampled data, what determinations can be made about the Nyquist rate (correct sampling rate)?

Using under-sampled data, what determinations can be made about the Nyquist rate (correct sampling rate)? If my instrumentation limits me to sampling data at a fixed but insufficient rate what, if ...
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1answer
48 views

doubt regarding sampling frequencies

I have a speech file that was recorded with a sampling frequency of 8kHz. I read the file in Matlab as [x,fs,nbits]=wavread('speech.wav). Now I play the file in ...
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2answers
93 views

Sample and hold - LTI filter

I've found a material explaining the sample and hold operation pretty well, but there's one thing I can't understand. ...
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1answer
87 views

How do you deconvolve irregularly sampled points?

$f:\mathbb{R}^2\rightarrow\mathbb{R}$ is an unknown signal, and $g:\mathbb{R}^2\rightarrow\mathbb{R}$ is a known point-spread function (in my particular case it's just a Gaussian). A system samples ...
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7answers
425 views

Nyquist Frequency Phase Shift

The above figure shows in dashed lines sinusoidal signals of the same frequency at three different phase shifts. The signals are then sampled such that the sinusoidal frequency is exactly a half of ...
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12 views

Finding lowest gradient path through multiple sets of samples

I have a noisy 2d signal I have sampled at multiple orientations, let's say 4 . I know that the orientation will either catch the signal accurately, or be much too high or too low. If the true ...
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0answers
17 views

Effect of an AA filter on sampling time

An AA filter is used in control schemes to filter out the non band limited behaviour that is mostly due to the sensors noise. This avoids aliasing by eliminating these unnecessary frequencies. What I ...
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0answers
33 views

Mathematical effect of reordering samples to known period, or equivalent-time sampling (ETS), and the interpretation or analysis of its result

I measure a natural raw signal over time, whose value is dependend on many factors of which several are periodic. I am interested in the effect of one such periodic factor and want to look at ist ...
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1answer
76 views

How does an equalizer prevent a sample and hold effect?

How does an equalizer prevent a sample and hold effect? Does a reconstruction filter only smooth out the output of an DAC (+S&H) or does it have another function?
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1answer
99 views

Calculating an incoherence property from sub-optimal sampling patterns

EDIT (after comments and subject matter review) CS is based on a choice of a sensing basis $\Phi$ relative to a representation basis $\Psi$. Using an "Incoherence Property" $\mu$ that measures the ...
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2answers
138 views

Why is dirac delta used in continuous signal sampling?

Why this concept is the most widely accepted model of signal sampling? By multiplying the continuous signal value with the dirac delta we get an infinite value. However if we perform convolution of ...
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2answers
88 views

efficient spectrum/fft analysis of sparse signals (found in radar applications)

consider the classic scenario of identifying a radar frequency. You typically have a very short pulse of very high RF frequency. For example, a 50ns pulse repeating itself every 1MHz or so (PRF 1MHz ...
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1answer
65 views

Inverse of a summing function

I am an experimental research engineer working on a Laser experiment. The details do not matter but due to practical limitations my data is influenced by the following transformation during data ...
1
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1answer
54 views

Which antialiasing filter before equispaced sampling

I am using MIT-BIH arrhythmical database where I have a digital signal of 1200 Hz so 1200 samples per second. This means that analog filters have already applied to remove the frequencies over Nyquist ...
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1answer
44 views

Filtering of a sampled signal

Let's say we have 2 signals of frequencies 4 kHz and 8 kHz. If we sample both signals with sampling frequency 10 kHz and pass the signal through a low pass filter with cut off frequency of 10 kHz, ...
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1answer
36 views

Odd sampling for symmetric functions?

I have just started my first proper course in DSP, and am a little confused about a statement in my textbook. The book states: "Notice that when the Nyqust-frequency is defined as $\pi/\Delta t$, we ...