Questions tagged [discrete-signals]

A discrete signal or discrete-time signal is a time series consisting of a sequence of quantities.

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1answer
1k views

Time-reversal and delay commutativity and resulting signals

I'm reading Schaum's DSP book, and in Fig 1-3 they demonstrate why shifting and reversal are order-dependent, showing a couple of simple systems, 1) delay followed by reversal, and 2) reversal ...
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1answer
748 views

Pulse detection against various noise floors

I'm decoding a pulse amplitude modulation encoded digital signal. The goal is to detect the individual pulse against a background noise floor. The amplitude of the pulses varies depending on the ...
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0answers
164 views

how to detect the center frequency of a QRS complex in an ECG signal?

i want to design a filter that can effectively detect the QRS complex in ECG signal(whose Fs=360Hz) and reduce high frequency noise components. i want to know as to how the center frequency is ...
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1answer
309 views

3D image sample frequency

I am currently frequency filtering a 3D image in matlab, with x,y,z dimensions of different lengths. The matlab filter takes normalised frequency as an argument. What does this mean for an image and ...
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2answers
3k views

Matching 2 audio files

I'm doing research on how we can do matching between 2 audio files using Android to see if they have the same content or not. After doing lots of research, I have concluded that I must do 2 main ...
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2answers
2k views

Why the carrier frequency does not appear after AM modulation?

Okay, this might be a very basic question, but I'm completely new to the field. I have a $40$ Hz sine wave ($1000$ samples, $fs = 1$ kHz), which I have amplitude modulated to a $4$ Hz sine wave ($...
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1answer
528 views

Calculating sum of a discrete signal

I have to find a convolution of two signals $h[n] = 0.5^nu[n]$ $x[n] = u[n]-u[n-3]$ the final sum, which is correct is: $$\sum_{m=n-2}^n 0.5^mu[m] $$ note that i replaced n-k with m, that is $ m =...
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4k views

Why convolution is required, or what is the philosophy behind convolution?

I am working in digital image restoration field. I have read all things about convolution, that, for an LTI system, if we know its impulse response, then we can find its output by just using ...
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1answer
204 views

How to convert an analog signal to an impulse signal?

I have several analog signals with constant amplitude in time. What I want is a way to convert these analog signals to an impulse signal i.e. a signal that consists of impulses occurring at different ...
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2answers
206 views

Is there any way to differentiate between impulses?

I have a set of signals consisting of impulses at different instant of time i.e. its not a continuous signal. What I want is a mathematically intuitive way to distinguish between any two such signals? ...
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2answers
448 views

Calculating the Norm of Each Sample of the DFT Trasform

Given a discrete signal $ \left\{ x[n] \right\}_{n=0}^{N-1} $ and its Discrete Fourier Transform $ X[k] = DFT \left\{ x[n] \right\}[k] $. I'd like to effectively compute the norm of each sample of ...
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2answers
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How to Combine Low and High Frequencies of Two Images in MATLAB?

I would like to combine two images A and B in the following way: 1) I want to take a Fourier transform of both of them 2) For image A I want to apply a weighted filter, which gives more emphasis for ...
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1answer
214 views

Fitting fixed number of complex exponentials to complex signal

I have a complex data series, and I would like to fit a fixed number (in this case, two) of complex exponentials of the form $Ae^{jBn}$, where A, B are complex. Not interested in the phase (i.e. arg ...
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1answer
4k views

How to determine the window size and weights in Weighted Moving Average (WMA), given desired cut-off frequency?

I am trying to smooth my discrete-time data points using the method of WMA. Currently, I am using n as the window size and the weight array, ...
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3answers
280 views

Peaking Equalizer

I'm making peaking equalizer in C# using NAudio (EQ-Cookbook). Can anybody tell me what is wrong with this code? It gives me noise. Sliders ...
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1answer
159 views

Digital Image Processing Framework

i'm looking for a free/opensource solution for digital image processing with reasonable performances processing quite large images (e.g. actual smartphones have resolutions of 8-10Mpx) and being ready ...
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3answers
2k views

Detecting a single frequency in noisy data

I am not very experienced in signal processing, and it seems like this should be a simple problem. I have two computers, one plays a tone a 19 kHz at a sampling rate of 44.1 kHz, and the other listens ...
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2answers
65k views

How to apply Hamming Window?

I am new in matlab and signal processing. The time series that have been used are obtained from accelerometer in a building. As far as I understand both the time series' length and window function ...
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1answer
152 views

DSP related websites and blog posts [duplicate]

Do you guys know good DSP related websited and blog posts? I know that for example there is this site called Minutify: http://www.minutify.com which is blog posts regarding all areas of electrical ...
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1answer
3k views

Power or Energy Signal?

Can you tell me whether the following signal is a energy or power signal? $$ x(n) = e^{j (n \pi/2 + \pi/8) } $$ I've solved it and found it as it was neither power nor energy signal was that right?
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NAudio real-time equalizer

I have designed 9 IIR bandpass filters (chebyshev type 1, order 4) using fdatool in Matlab. Then I use a and b filter coefficients to apply it on differential equation. So, my question is how to apply ...
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2answers
321 views

What does the uncertainty principle say about recursive filters?

The uncertainty principle is usually stated as a relationship between a continuous signal and that signal's Fourier transform, and says that $$ \int_{-\infty}^{\infty} \! x^2 f(x) \ \mathrm{dx} \int_{-...
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1answer
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Implementation of basic 2-pole low-pass filter

I know this question has been asked to hell and back from many people, and that it will probably get downvoted until it's deleted, but please bear with me. I am designing a sound engine for one of my ...
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2answers
847 views

Why integrate over $2\pi$ in inverse DTFT?

In DTFT of a signal, the spectrum of a sequence is periodic with period $2\pi$ and all the information needed for derivation of the original signal from its spectrum is contained in $\pi <\omega &...
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2answers
2k views

Why the power of periodic signal is square?

I have DSP in my academics and while going through the video lectures i am stuck regarding power of periodic signal. Its mentioned as $$P_{\text{avg}}= \frac{1}{N}\sum_{n=0}^{N-1}|x(n)|^2$$ my ...
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2answers
90 views

Multiple filters or not?

I'm processing audio signal sampled at 44.1 kHz. After some other processing, I need to filter the signal between ~5-50 Hz. Achieving this for a signal with such a high sampling frequency appears to ...
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1answer
281 views

'Best fit' motion curve for an unachievable profile

I have a discrete function that represents a 1d position in space over time. I have a motion system that would like to achieve this curve, but, due to constraints on jerk (3rd derivative) and ...
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3answers
1k views

Mathematically inclined Signal and Systems/Signal Processing book?

I'm an electronics engineering student with high inclination to analysis and pure mathematics. I was just wondering if there was any book ( or any resource ) that treats signal and systems and signal ...
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1answer
951 views

How to derive stability of analogue and digital systems

I'm trying to find out why digital systems are stable iff poles oh their transfer function are inside the unit circle (or are there any others conditions?). I understand analogous systems are stable ...
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2answers
4k views

Main differences to take into account between continuous and discrete time signals

I started studying DSP and the first things that came out were the differences between continuous and discrete time signals. So, I was wondering if I understood well these concepts before I keep going ...
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1answer
2k views

Discretizing a continuous time signal

This may be a very straightforward question but I've been trying to understand this for a long time and can't wrap my head around it. How would one go about discretizing the following continuous time ...
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1answer
122 views

Why is a feedback loop not represented by the least order transfer function?

I have a feedback loop with transfer $L(z)= \frac{H(z)C(z)}{1+H(z)C(z)}$. $$H(z) = h\quad \text{and} \quad C(z) = \frac{K}{z-\alpha}.$$ If I manually calculate the transfer function, I get: $$L(z) =...
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0answers
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Simulink continuous to discrete question

I'm new to Matlab/Simulink and would like to convert the continuous system below to discrete for eventual conversion to C code. Matlab has a c2d function, but I don't know how to apply it to a ...
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3answers
936 views

Determining the mean of of a sampled signal (improvements to time averaging method)

I am sampling a signal at 1kHz, which is essentially a single-frequency sinusoid with a DC offset (and a small amount of noise). I don't know the precise frequency of the sinusoid (it is at most 10Hz),...
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1answer
361 views

Cross Power Spectral Density of Unevenly Sampled Data

Here's my problem. The input signals $x$ and $y$ will be having the time value aligned with each other. However, the data are not evenly sampled. I would like to calculate CPSD of both signals. The ...
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0answers
341 views

Kalman Filter: continuous state space, discrete observations

This is just an idea. How can we model the kalman filter to get the state representation in continuous space when the observations to the system are actually from the discrete space. The discrete ...
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2answers
902 views

One Source Signal Separation

I have a wav file which is gained by playing 2 different wav files with 2 speakers and recorded with mic. If I want to get one of the original signals using the resulting wav file and the other ...
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1answer
358 views

How to automatically detect blurred document images

The question I have is related to image quality evaluation. Suppose you are supposed to perform OCR (Optical Character Recognition) on a set of document images. Some images, however, are very blurring,...
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11k views

How to compute dBFS?

I asked myself how to compute dBFS (dB full scale) from a value of sample between 1 and -1? and in general?
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1answer
159 views

Format location from LPC magnitude

I am confusing when I find the lpc roots magnitude, f1=0.9313, f2=0.9308, f3=0.91 and f4=0.95 but when I look at LPC envelope the f1 is higher than f4 in term of magnitude in dB. Could anybody explain ...
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1answer
127 views

doing sine regression to recover the paramters from a given signal

I have a discrete signal. I want to do sinusidal regression to estimate the parameters such as amplitude,phase, frequency etc It is important to note I only have the signal values and ...
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2answers
128 views

What is the theoretic difference in the ways to compute the spectrogram?

I have noticed that the spectrogram for a audio signal is often calculated in different ways when the result from the DFT is transformed into real numbers; some tend to compute $|k|$ or $|k|^2$ while ...
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2answers
366 views

Calculate interbeat intervals

I'm currently reading a paper and I can't seem to make sense of a certain part. A link to the paper: removed The part I don't quite get is on page 286 (after the part where they explain their filters)...
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1answer
272 views

Banding in the derivative of a lock-in amplifier signal

I'm analysing several measurements taken with a SR830 lockin amplifier. These measurements look similar to this one. Since I'm interrested in the derivative of the signal I took the numerical ...
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3answers
11k views

Finding the dominant tone in a signal

I am given 256 samples at a sampling rate of 48k and am asked to find the frequency of dominate tones and the amplitude of the largest tone. The data looks like this when plotted: I assume I must ...
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1answer
874 views

How to calculate the energy levels of frequency intervals of an audio signal?

Using the Fourier transformation I can translate a signal from the time-domain to the frequency-domain. If I divide the frequency axis into intervals, how can I calculate the audios signals energy for ...
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1answer
1k views

Determine the type of filter from the transfer function

i got that system $x_n \to x_n-x_{n-1}$, so $h_n=[.....,0,1,-1,0,...]$, with $h_0=1$ and $h_1=-1$, so the transfer function given by: $$\sum_{i=-\infty}^{\infty} h_ne^{-jwn} = h_0e^{-jw(0)}+h_1e^{-jw(...
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2answers
932 views

Are there any measurements or test for whether an audio signal is stationary?

I need to divide an audio signal into frames in which the signal can be assumed to be stationary. How can I measure whether an audio signal is stationary in a frame?
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140 views

How to measure the short-time change in an audio signal?

The choice of frame length for calculating the Mel Frequency Cepstral Coefficients of a signal, depend on the how much the signal change over a small time scale. I guess that human speech change ...
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1answer
79 views

Finding Frequency Differences In Differing Amplitude Signals

I have two audio signals, both with supposedly matching frequencies at each time window. What I would like to do is to verify that the frequencies are identical. However, one was created at a higher ...