Questions tagged [dsp-core]

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

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45 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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17 views

Good links or tutorials to understand how a signal is represented in linear equations [closed]

Facing issues in understanding signal representation in this form x(n,k) and not able to get relation between n & k in simple terms. Any links or guidance will help in understanding mathematics ...
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22 views

About testing my code on code composer [closed]

i am a beginner in using DSP processor.i use DSP TMS230c6713. i asked for simulation program like protues to test my code..in protues using avr ,we choose atmega32 for example and load the hex file on ...
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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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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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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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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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61 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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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
178 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
33 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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2answers
875 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
44 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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55 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
220 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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58 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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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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333 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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1answer
207 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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1answer
141 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
108 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
120 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
72 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
237 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
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
269 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
82 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
706 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
305 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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51 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
117 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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1answer
160 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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74 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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1answer
208 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
44 views

Confusion regarding a library of Digital Signal Controller

I'll be using a NXP DSC with 56800Ex core and my aim is to implement a Elliptic IIR Filter in it. The User Reference Manual of the 'General Digital Filters Library' (which is available here) has ...
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1answer
381 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 ...
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2answers
505 views

SNR of a 16-bit DSP with 12-bit ADC, 40-bit accumulator

According to the 6dB-per-bit rule, a 16-bit DSP would provide 96 dB of dynamic range, approximately. If the ADC and DAC are both 12-bit long, the DSP is still considered to have 96 dB, or the ...
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1answer
64 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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1answer
184 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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1answer
280 views

Relative Computational Complexity for Channelizer Implementations

Suppose I want to separate a broadband input signal into multiple narrow subbands using an FFT-based analysis filter bank. The filter bank uses a prototype lowpass filter and is implemented using a ...
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1answer
639 views

DSP with 8+ inputs for simple beamforming

Does anyone know of a DSP that is cheap and has 8+ inputs (8 I2s mics) and that can perform beam forming calculations? I'm super new to circuit design so preferably one thats beginner friendly. ...
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1answer
83 views

Impulse property - needed mathematical proof

Prove: $$ \left. \int_{-\infty}^{\infty}\frac{d\delta(t)}{dt}\phi(t)dt=\frac{-d\phi(t)}{dt}\right|_{t=0} $$ $\delta(t)$ is impulse signal and other one is any general signal. Can anybody do the ...
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3answers
1k views

What's the difference between a DSP and microcontroller?

What are the core attributes that make a DSP and microcontroller different from each other? I have in mind a machine vision device that needs to run image processing algorithms but also control some ...
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1answer
255 views

Interpolating and Decimating CIC filter based on sinc function

How to realize a sinc function using CIC filter for decimation and interpolation ??? Can I combine interpolation and decimaton methods inorder to get the complete response of a sinc function???
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1answer
225 views

Problems with implementation of a band-stop filter on an MCU (dsPIC) using fixed-point arithmetic

I'm trying to implement a band-stop (notch) filter on a microcontroller dsPIC33EP64GS506 to filter out a $100\textrm{ Hz}$ component from an input signal. The ...
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1answer
901 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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3answers
207 views

How can define an indicator which measures the degree of similarity between two signals?

The similarity of two signals is calculated by cross correlation. But, how to define an indicator which quantitatively measures the degree of similarity between two signals? Thanks.
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1answer
570 views

Can we use Tiva C series TM4C123GXL for signal processing experiment purposes? [closed]

I've newly bought texas instruments Tiva C series TM4C123GXL processing board for developing DSP application project. Is that processing board is good for that purpose? Is that well work with Matlab? ...
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0answers
207 views

How does the ROC (Region of Convergence) related to a real world application?

In class, we are often given exercises to find the impulse response, output, and Z-transform of a system. In addition, we are often asked to define the Region of Convergence (ROC) depending on where ...
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1answer
1k views

Stereo Widening

I recently found this implementation for widening a stereo signal by Michael Gruhn. http://www.musicdsp.org/showArchiveComment.php?ArchiveID=256 I've implemented this myself as follows: ...