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

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basic question about OFDM

I'm going to simulate OFDM system and reighley channel in MATLAB. I create array of 128 symbols (for example) and then get IFFT, add CP. passing through AWGN channel in easy, but in Reighley there is ...
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2answers
70 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
31 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
40 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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23 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
19 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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33 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
48 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
53 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
56 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
169 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
47 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
46 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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2answers
43 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
88 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
82 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
252 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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0answers
10 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
68 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
83 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
115 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
71 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 ...
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1answer
42 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
39 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
34 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 ...
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88 views

Expression to represent alternating pulse train?

I have a question regarding an alternating pulse train signal, p(t), and a system which looks like this: Immediately, I'm supposed to be able to extrapolate that pt=p1(t)-p1(t-Δ) How do you ...
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1answer
141 views

Minimum Sampling frequency for this signal to avoid aliasing

I would like to compute the minimum sampling frequency for the signal $x(t)$ given below: $x(t)= sinc ^3(300t) * sinc ^5(600t)$ and $ *$ stands for convolution. where the sinc function is defined ...
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34 views

Undersampling of a non periodic pulse signal

I am starting getting familiar with electronics, consider me as a newbie.. I am facing a sampling trouble with my DAQ system. I have a non periodical pulsed signal in the range of 0.3 to 1kHz. The ...
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1answer
61 views

Sampling Theory in Signal Processing

In signal processing, when you sampling an analog signal, the sampling theory says that the sampling frequency needs to be at least twice the maximum frequency in the analog signal content. I ...
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26 views

Discretization of Sensor Noise for Robotic Simulations

What is the best way to perform discretization of sensor noise for simulations in Matlab? I have a MEMS accelerometer and the datasheet specifies the following: ...
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50 views

Sampling Clock Offset estimation

I want to estimate sampling clock offset. I am currently referring this paper " CRAMER-RAO BOUNDS FOR DATA-AIDED SAMPLING CLOCK OFFSET AND CHANNEL ESTIMATION ". Is there any other method to estimate? ...
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1answer
208 views

Why do we read that Hilbert transform can be used for envelope detection?

We can often read that Hilbert transform is useful for envelope detection (e.g. Hilbert transform to compute signal envelope?) I have done some tests with various soundfiles, and ...
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1answer
114 views

Why only take 205 sample when using a sample rate of 10khz or 8khz?

I've often seen i my DSP book, that when they record a sound, they use a sample rate at 10khz or 8 khz, which makes sense since the fs is 2 times bigger than the sampling signals max freq. What i ...
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1answer
159 views

What is the frequency of $\cos x -\sqrt 2\cos\sqrt 2 x$?

Question Considering that $1\over{2\pi}$ is the frequency of $\cos x$ and also of $\cos x - 2\cos 2x$, what is the frequency of $\cos x - \sqrt 2\cos\sqrt 2 x$? Thoughts Perhaps "frequency" isn't ...
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2answers
145 views

Reducing sampling rate by a non-integer factor

I have data that is sampled at 12 kHz. The downsample function in software such as MATLAB only allow you to downsample by an integer factor, i.e. from ...
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1answer
107 views

Envelope follower not smooth enough

With the following code, I get the "envelope" of a sound signal. However in the "envelope" there are small oscillations (high frequency) : it is not smooth enough... I imagine that it's possible to ...
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1answer
79 views

Map the Amplitude Envelope from [dbmin,dbmax] to [-inf,dbmax]

I would like to map the envelope of a sound (which has max dB level = dbmax, and min dB level = dbmin) from [dbmin,dbmax] to ...
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0answers
25 views

Spectrum of “sampled” white noise [duplicate]

Theoretically, no sampling frequency is good enough to represent a white noise signal since its power is spread across the entire frequency axis. Let's say we started out with a discrete ...
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4answers
118 views

What this triangles mean in frequency aliasing depiction?

How edges of the triangle get on both sides of fs? What is their purpose? Why fs in the center? (from Lyons Understanding DSP)
3
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2answers
291 views

Minimum period of a signal

What is the minimum period P (in samples) of the signal $e^{j(\frac{M}{N} )*2πn}$ for the following values of M and N? M=1,N=3 M=5,N=7 M=35,N=15 I have got the answer for the first ...
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1answer
598 views

Preparations to implement FFT on a wav file (code Java, not Library)

I am making preparations to implement FFT on a wav File.wav Only Java. I'm reluctant to use matlab I have a File wav stereo, 16 bits. I need to calculate the FFT, but I have confusing. My File has ...
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3answers
83 views

What sample rate should I use on my ADC?

I have an ADC that will have a guitar track as an input. I want to find which notes are hitting via the frequency domain. Given that guitar frequency range is from 82Hz to 1046Hz which sample rate ...
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68 views

Can somebody help me solve this signal accumulation problem?

Investigation of the method of suppression of random noise by coherent signal accumulation Purpose - to identify opportunities for the coherent accumulation for cases of stationary and ...
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1answer
77 views

Can you help me determine what section of digital signal processing discussed in this screenshot?

what kind of filtering is it? How signal/noise ratio in the output mix of the duration of accumulation would look like, knowing Signal/noise ratio=0.1 ,The number of cycles of accumulation of up to ...
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93 views

Stochastic process inference from partial observations

Consider a set $U$. My signal is a piece-wise constant "function" $Sig: t \mapsto s$, i.e. the signal at time $t$ equals to some subset $s \subset U$. One can see $Sig(t)$ as a stochastic process. ...
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62 views

understanding discrete wavelets

I was reading section 4.3.3 from chapter 4 of "A Wavelet Tour of Signal Processing", "...Let $\bar{f}(t)$ be a continuous time signal defined over $[0,1]$. Let $f[n]$ be the discrete signal obtained ...