Stack Exchange Network

Stack Exchange network consists of 175 Q&A communities including Stack Overflow, the largest, most trusted online community for developers to learn, share their knowledge, and build their careers.

Visit Stack Exchange

Questions tagged [dsp-core]

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

0
votes
0answers
25 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 ...
0
votes
1answer
32 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 ...
0
votes
0answers
17 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 ...
0
votes
0answers
41 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\...
0
votes
1answer
49 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 ...
1
vote
1answer
124 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, ...
0
votes
1answer
66 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 ...
1
vote
1answer
56 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) ...
1
vote
1answer
77 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?
0
votes
1answer
60 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 ...
1
vote
1answer
139 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 ...
0
votes
1answer
48 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 ...
1
vote
3answers
142 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 ...
0
votes
1answer
59 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 ...
0
votes
1answer
404 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: ...
2
votes
2answers
237 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]...
0
votes
0answers
32 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 ...
-1
votes
1answer
79 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 ...
-2
votes
1answer
88 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) ...
2
votes
0answers
61 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 ...
1
vote
1answer
177 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 ...
0
votes
1answer
42 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 ...
0
votes
1answer
286 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 ...
5
votes
2answers
411 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 ...
1
vote
0answers
50 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)...
2
votes
1answer
170 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 ...
0
votes
1answer
234 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 ...
2
votes
1answer
457 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. ...
1
vote
1answer
82 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 ...
1
vote
3answers
698 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 ...
0
votes
1answer
215 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???
5
votes
1answer
173 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 ...
3
votes
1answer
652 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 ...
2
votes
3answers
182 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.
0
votes
1answer
430 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? ...
2
votes
0answers
156 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 ...
2
votes
1answer
849 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: ...
1
vote
0answers
184 views

How to find frequency and phase shift in the recorded wave( .wav file )

I am working on BPSK at frequecies near 20 kHz with a sample rate of 44100 Hz. We can't afford sample rates above than this due to computational complexities. Have found that the frequency of ...
1
vote
1answer
224 views

What can we understand when we increase the $N$-point DFT for a given signal?

I have an input sequence $$x(n)=\cos(0.48\pi n)+\cos(0.52\pi n).$$ I am determining its spectrum (amplitude of DFT values) based on the finite number of samples. For Example : Taking the first $10$ ...
1
vote
0answers
76 views

STM32 Discovery dev platforms [closed]

I'm still researching this but though I'd post asking about experiences. Looks like st.com offers a variety of STM32 Discovery dev boards for ARM CPUs to, among other things, play with DSP. But ...
3
votes
1answer
203 views

FFT of a $N$-length real sequence via FFT of a $N/2$-length complex sequence

I am doing FFT. The one way I did is to put real part in one array and the imaginary part as zeros and then calculated the FFT. But this take up lots of resources. Now I want to calculate FFT by ...
1
vote
2answers
666 views

Why is this FFT code not working to extract frequencies from a wav audio?

I'm quite new to audio processing and i'm trying to extract frequencies from a wav file. As research papers stated i put the wav audio into a byte array and segmented and trying to apply FFT to get ...
1
vote
1answer
4k 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?
2
votes
3answers
984 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 ...
1
vote
1answer
557 views

FPGA vs DSP for MAC operations

I'm new to SP Stack Exchange . I am doing a project on implementing a handwriting recognition machine using Neural Networks in Real Time . The Image Processing Part involves convolving a 4*5 kernel ...
0
votes
1answer
36 views

Is digitization of discrete valued signal necessary for it to be processed?

While reading the book DSP by Proakis and Manolakis i came across the concept of digitization of signals. The Author states that Digitization(rounding it off to nearest integer) of signal is ...
1
vote
1answer
431 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 ...
1
vote
1answer
83 views

Multiplier in digital signal processing?

whenever we start working on Discrete Cosine Transform(DCT), Loeffler algorithm a key role. Furthur if we approach hardware implementation of this algorithm, we find methods based on Distributed ...
2
votes
0answers
166 views

Alternative to scipy maximum_filter to find peaks?

I'm using scipy maximum_filter together with binary_erosion to find peaks in my spectrum. I just wondering if there are any ...
0
votes
1answer
916 views

Finding peaks in an Audio spectrogram

I'm using gist.github.com/sixtenbe/1178136#file-peakdetect-py to find maximum peaks in my spectrogram. I used ...