Questions tagged [dsp-core]

A dsp core is a processing unit specialized for digital signal processing tasks.

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1answer
55 views

What is the slope of a transfer function?

Lets say I have a Transfer function (H) plot as below. I need to find the SLOPE of the transfer function in dB/MHz. That is... i need a plot (db/MHz) vs frequency. How can i go about it ? ...
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3answers
66 views

Basic Understanding of Convolution Operation

I would really appreciate any help in understanding this. If $$y[n] = x[n]*r[n]\tag{1}$$ then what is $y[n-n_0]$ for any system? Is it $$y[n-n_0] = x[n-n_0]*r[n-n_0]\tag{2}$$ or $$y[n-n_0] = x[n-...
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1answer
28 views

what is the C/D equation in dsp?

I have one example test question by this kind: So in this ebook (Digital Signal Processing Using MATLAB) as the DSP references and not found C/D block equations on it? and this c/d search results? ...
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1answer
36 views

LOW SIGNAL NOISE RATIO FFT

I am processing an audio signal with CS4244 codec at 24 bits and 48kHz. Every 2048 samples the FFT is calculated and its amplitude. Without input signal I get FFT values ​​close to 30dB. With maximum ...
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1answer
86 views

Improvement of SNR by sampling different number of cycles using FFT of a fixed length singal

My question relates to Fourier transform of signals with different lengths in time but the same fixed length of the digitized signal with different sampling rates. Specifically, how does the number ...
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1answer
938 views

What's the difference between using DFT, IDFT or DCT to calculate cepstrum of a power spectrum?

I've seen different equations that calculate cepstrum from power spectrum, but the equations are not consistent. Some people use Fourier transform, some use the inverse Fourier transform, and some use ...
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0answers
43 views

DSPs in mobile phone?

I know about one application of DSPs ,that is a smart phone such as Samsung S4,but what about normal cell/mobile phone such as nokia3310?Is it also contains a DSP(Digital Signal Processor)?
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3answers
1k views

Mathematically Inclined Signal and Systems / Signal Processing Book Recommendations

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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0answers
17 views

Digital signal processing how audio wave and function chage with respect to each other

If an audio signal is f(t) and if i play same audio with 2× speed then what will function of 2× speed signal f(2t) or f(t/2) and why?
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1answer
24 views

DFT calculations returning different results

My coworker ran a DFT query using a software in C downloaded to his PC. His results was the following chart: I am using a JS tool from corbanbrook/dsp.js My results were very different. any ideas ...
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1answer
1k views

Natural, Forced and Total System Response - Time domain Analysis of DT

I am a self-learner of DSP by reading related books. While reading Time domain Analysis of DT (Discrete Time) system, I suddenly got introduced to difference equation representing a system. It then ...
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1answer
208 views

Preprocess sound for voice recognition in Pocket Sphinx on Raspberry Pi 3

I want to improve my voice recognition system on a Raspberry Pi 3 running Pocket Spinx. It's pretty accurate (limited recognition vocabulary) but is terrible when other noise is around. Is there any ...
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1answer
67 views

Stability of the Low pass filter after Spectral Transformation

Using a spectral transformation a filter $G(z^{cap})$ is designed to be a low pass filter from an existing lowpass filter $G_{lp}(z)$ via $G(z^{cap}) = G(z)z=H(z^{cap})$ . Which property should H(z) ...
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0answers
27 views

Skew measurement with phase deviation

I am calculating the standard deviation of skew (= phase/2*pi*fin) from the 1000 monte carlo runs of the following code. In the y0, there is no input skew., but in y1 there is a skew present. Its ...
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1answer
214 views

Polyphase decimation filter plot in Octave

I'm using de Soras hiir library for decimation and I'm trying to verify the filter design in Octave. The filter is a polyphase 2x downsampling filter with 96dB attenuation and 0.01 transition band, ...
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2answers
930 views

Career in signal processing

I am doing my MSc in electrical engineering, and there are not many courses in my university on signal processing. I want to develop my skills in signal processing such that I have a hold in one ...
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1answer
45 views

Moving from books to implementations

So after a good while of digging and practicing, I think I've managed to gain a decent intuition for some of the very core concepts related to linear systems, filter coefficients/parameters, FIR/IIR ...
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1answer
65 views

Compression Sensing for Blind Source Separation

I am new to Signal Processing, and am interested in compression sensing for audio files. CS is based on the algorithm that, given some sampling of a signal $f$ in order to obtain a smaller (compressed)...
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44 views

What is the error rate in compressed sensing?

Let $x \in \mathbb{R}^n$ be a $k$-sparse vector. Given $A \in \mathbb{R}^{m \times n}$, we have a measurement vector $y$ given by $$y=Ax$$ Let $\hat{x}$ be defined as follows $$\hat{x}:=\arg\min_{z\...
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56 views

Filter Design with DSPIC

Good Morning! I am new to digital signal processing, And I am currently working with DSPIC33F. I used Matlab design Filter to extract the coefficients of the filter. My Problem is within my Data ...
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1answer
294 views

multiplication in frequency domain equals convolution in time domain mismatch

We know that a convolution in the time domain equals a multiplication in the frequency domain. As per the article below, https://www.dspguide.com/ch9/3.htm In order to multiply one frequency ...
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1answer
5k views

How to remove noise using mean and median filters?

I can remove the noise using moving average filter. The MATLAB code is shown bellow. But how can I remove the noise using mean and meadian filters?
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1answer
149 views

“Revolving” QPSK Constellation

I am trying to code a QPSK transmitter in VHDL but first I want to create one out of Simulink blocks and translate it into VHDL code. I am using an example project for the PlutoSDR as a guide. This ...
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1answer
371 views

Discrete Time Convolution Example

How did we get this final answer in example in fig attached. I understand that delta(n) is a DT unit impulse where it is 1 at n = 0 and zero elsewhere. I understand that u[n] is a unit-step where it ...
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0answers
60 views

Scipy Deconvolve help

I was playing with the deconvolve method in scipy and I can't seem to get it working properly (I am still really new to DSP/deconvolution). I convolved a gaussian with a fwhm of 2.0e-9 with a ...
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2answers
251 views

Verifying Channel Response Identity

If $x[n]$ is input signal, $y[m]$ is output signal, $h[m]$ is impulse response of the channel, then the following relation holds $\lvert Y(\omega)\rvert = \lvert H(\omega)\rvert^2 . \lvert X(\omega)\...
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2answers
101 views

What are the tradeoffs for more coefficients for fractional delay filter?

Is there a rule of thumb for how many coefficients to choose for a fractional delay filter? I figure there might be a trade off based on the signal frequency as well as my sample rate (this is ...
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1answer
117 views

Why are binary signals drawn continuous in time despite being a digital signal

The conventional definiton of a Digital signal is as follows: A digital signal is a signal that is discrete in time and quantized in amplitude. Almost all resources (textbooks, online sources,etc) ...
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1answer
122 views

Costa's loop questions, design, tracking and limitations

Why does Costa's loop PLL bandwidth need to be narrow? and why can't it track a certain frequency offset? and does that hold even if it has a filter with a pole at DC?
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1answer
253 views

Hardware requirements for ambient active noise cancallation system using adaptive LMS

Background I'm working on a hobby project to reduce ambient low-frequency noise generated from HVAC equipment within a small area of a room. The noise is predictable and concentrated between 55Hz and ...
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1answer
74 views

How can i record 1 MHz signal and do a real-time processing in a PC? [closed]

I have a 1 MHz signal from transducer that needs to be analyzed in real-time condition, so the sampling needs to be more than 2 MHz. It consists of digital filtering, turn into spectogram and feed ...
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1answer
52 views

What are some efficient algorithms for real-time calculation of eigenvalues?

I need to calculate eigenvalues of a square matrix for a real-time implementation using microcontrollers or dSPACE. Obviously, I can find the eigenvalues of a 2by2 matrix analytically. I am wondering ...
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3answers
286 views

How to Improve Signal Processing Skills? [duplicate]

I've heard that signal processing is fundamental field that can be applied to many other fields (MRI, EEG, audio, speech recognition, communication even to economics) So, I want to contribute myself ...
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1answer
168 views

The Spectrum of the Impulse Response of Linear Time Invariant (LTI) System

Consider an LTI system with impulse response $$h[n] = \frac{1}{5^n} u[n].$$ (a) Determine and sketch the magnitude and phase response $|H(\omega)|$ and $\arg(H(\omega))$ respectively. (b) ...
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1answer
744 views

Confused over Stopband attenuation and stopband ripple

I have designed this bandstop filter to a desired specification: Stopband attenuation = 40db Passband ripple = 2db etc ... I understand that the stopband ripple can be given by the following: ...
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2answers
323 views

What is so special about this filter coef = [1,2,1]?

Can anyone tell what kind of filter is this? I can see it's smoothing/averaging the input signal, but why specifically with [1,2,1] instead of something else? What is so special about this coef=[1,2,1]...
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0answers
52 views

DSP Hardware recommendation or is it even needed?

I'm new in embedded programming. Trying to choose a dsp, they seem very feature rich for my planned usage. I want to analyze 18 signals (simultaneously or near simultaneously) real time. I want to do ...
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1answer
211 views

Suggestion for a DSP board to implement a Kalman filter in real-time at a rate of around 4MHz or higher

I have been attempting to implement a Kalman filter on an FPGA in VHDL and have got to the point that it works. However I have the issue that each variable requires 64 bits to be represented to get ...
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1answer
498 views

Saliency Map based on itti model

i want to create a Saliency Map based on itti model via OpenCV(Python). what i'm doing is that: 1- first i change image color space from RGB to LAB color space 2- for each color channel, i create a ...
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1answer
122 views

On-Chip DFT Engine vs Software DFT

I'm new to signal processing. I'm trying to build a network analyzer. So far I was only considering getting the output signal through a 16-bit ADC and then do its corresponding windowing and DFT to ...
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0answers
76 views

Finding specifications for digital signal processors (DSPs) for given audio application

Note: This topic has been moved to https://electronics.stackexchange.com/posts/345289 since it fits better, please post any answers/comments there. I'm finding difficulties choosing an adequate ...
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0answers
85 views

Why do I have to duplicate the labels used for the classification process?

I'm using the BCI Competition dataset IV 2a. I'm working on Left & right hand movement. I extracted data of channels (Fz,Cz,Pz). I filtered data [8-30], then normalized it. I calculated the ...
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3answers
4k views

What does 'canonical' mean?

What does 'canonical' mean? e.g. 'write IIR or FIR in canonical form' I get the gist of it but maybe someone here has a precise definition. In a non-DSP context it would mean something like '...
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2answers
25k views

Intuitive explanation of cross-correlation in frequency domain

According to the cross-correlation theorem : the cross-correlation between two signals is equal to the product of fourier transform of one signal multiplied by complex conjugate of fourier transform ...
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3answers
1k views

Speech corpora for VAD

I'm doing a student research project connected with VAD. And I want to find speech corpora with added noise. I found many references to the NIST Speaker Recognition Corpora. But unfortunately, all ...
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2answers
526 views

Hilbert transform of a bandpass signal [closed]

I have a bandpass signal x. I am taking hilbert transform of it. I want to reconstruct amplitude and phase of the signal.Therefore, i am using Hilbert transform. ...
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1answer
1k views

FIR filter design and implementation - sample rate and number of taps

I have been designing and using IIR filters to process audio for a while now but want to design some FIR filters for delay equalization. I have done some preliminary work using the following approach: ...
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1answer
402 views

Variance of N samples of Noise

What is the variance of $N$ samples of white Gaussian noise? I have an Equation for a continuous time which includes an $\mathrm{SNR}$ term inside it. Now when I am going to sample it, what is the ...
2
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1answer
188 views

High performance DSP processor with better on-chip memory

I am currently using a combination of 4 ADSP 21469 because of the complexity and huge chunk of code that I currently have. BOOT and application will execute from external Flash memory. The code for ...
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2answers
245 views

Why call it discrete time signals as difference equations?

There are lots of representations of discrete time signals. You represent it in block diagram or as difference equations. We call it block diagrams as there are elementary structures in the diagrams. ...