Questions tagged [dsp-core]

**Do NOT use this tag for general questions about digital signal processing!** A DSP core is a processing unit specialized for digital signal processing tasks.

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4
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2answers
109 views

DSP Book with More Applications

The process of my DSP class used the Oppenheimer DSP book as text. Although I can easily understand the math in the book, I found it lack examples and applications, hence I still cannot build a ...
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1answer
45 views

Is there any way to align temporally two signals in spectrogram?

I am keen to know that If I have two signals say : Farend_signal (y[n]) and Nearend_signal (x[n]) and x[n] also contains an echo of y[n] with some delay. Is there any technique to align these two ...
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0answers
29 views

How to subtract echoed signal from mixture of voice and echo signal?

I am working on an echo removal project. So far, I have successfully identified the far-end signal of length 21 ms at a sampling rate of 48000Hz whose echo is present in my near-end signal of 21ms. I ...
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1answer
71 views

Echo removal from a chunk of PCM signal

I have implemented a paper regarding echo detection in real-time using MFCC coefficients. Now, I want to remove the echo from the mixed signal. Let me explain it as: There are two signals, NearEnd ...
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5answers
124 views

Effect of One Sample Latency in Audio Application

I am dealing with a customized processor that has a processing latency of one sample (20.8 microseconds) at 48kHz Sampling rate . What I would like to know is ...does this latency of one sample have ...
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3answers
81 views

QAM Amplitude Levels

I'm working on a project to stream video between 2fpgas by using QAM. While mapping symbols with different amplitude pulses(PAM) a question popped up in my mind. In real life how does communicating ...
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1answer
109 views

number of multiplications per second required by the system

Consider a system implementing a rational sampling rate change by 5/7: for this, we cascade upsampler by 5, a lowpass filter with cutoff frequency π/7 and a downsampler by 7. The lowpass filter is a ...
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0answers
74 views

minimum oversampling factor for D/A converter

Consider a D/A converter for audio signals consisiting of a zero-order-hold interpolator followed by a continuous-time lowpass filter with positive passband between 0 and 20KHz and stopband starting ...
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1answer
230 views

the number of multiplications per second required by the system

Q1. Consider a system implementing a rational sampling rate change by 5/7: for this, we cascade upsampler by 5, a lowpass filter with cutoff frequency π/7 and a downsampler by 7. The lowpass filter is ...
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1answer
38 views

Detect static in audio with Python

I have a program running in python. It samples the demodulated FM output from an SDR. When there is no transmission, the input to the program is completely static. How can I go about squelching the ...
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0answers
40 views

How can I determine the fundamental period of the signals?

How can I find the fundamental period of the given signal? Can anyone help me, I am stuck. $x(t) = cos(2πt) + cos(4πt)$ $y(t) = cos(2πt) + cos(2t)$ Thank you in advance.
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2answers
57 views

Hardware kits for DSP Lab?

I have been assigned responsibility as Lab instructor of DSP at undergrad Level and so i have to equip my lab,please recommend which processors should i purchase for hardware implementation of dsp ...
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0answers
28 views

Efficient way to calculate eigen-values of a 2x2 matrix

I'm working on an algorithm that works based on finding the eigen-values of a 2x2 complex covariance matrix. However, from a fixed-point implementation point of view, finding eigen-value of a matrix ...
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3answers
228 views

Confusion understanding digital signal processor?

I am reading wikipedia article about digital signal processor and as shown in end of attached snapshot ." Most general-purpose microprocessors can also execute digital signal processing ...
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1answer
31 views

Why SRS signalling is used in 5g?

Since DMRS is also being used as a reference signal in PUCCH and PUSCH channel, so why an extra SRS signalling is required.
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1answer
115 views

general purpose microprocessor vs DSP(digital signal processor)?

I know that DSPs are specialized processors but i want to know what will happen if DSPs are replaced by general purpose processor in their applications? I know one application,mobile phone,will that ...
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1answer
41 views

Why is it that the input to an IFFT is first reordered in an OFDM Modulator

I am back with my questions! I recently came across a paper titled "Low Latency IFFT Design for OFDM Systems Supporting Full-Duplex FDD" where they claim a new method of IFFT input mapping ...
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1answer
50 views

simulator for dsp kit tms320c6713?without using actual hardware kit tms320c6713?

As you know, most universities across the globe are closed due to pandemic and people are unable to acess research facilites ,so in such a scenario one wants to improve his dsp hardware programming ...
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1answer
33 views

Selecting a Digital Signal Processor for a Specific Image Sensor

I am working on developing a camera and I am trying to select hardware for the camera. I believe that I may need a digital signal processor to compress image data before it is transmitted to a ...
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0answers
20 views

What is the intuitive idea behind FIR filter order and filter parameters? [duplicate]

I know that if we decrease stop-band and pass-band attenuation, and also decreasing the transition band, the filter order will increase as well as filter performance. But what is the intuitive idea ...
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1answer
116 views

What does librosa.load return?

I'm new to signal processing and I'm working with librosa library quite a bit. I would like to know what is the information that is returned by librosa.load function when I read a audio(.wav) file ...
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1answer
323 views

How to Find spectral Peaks from 2D array of Spectrogram? [closed]

I have computed a Spectrogram 2D array as : How to develop a spectrogram (2D array) from audio signal? Now, I want to find spectral peaks to find the respective temporal displacement. In python using ...
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2answers
66 views

Q format doubts

Is my understanding correct w.r.t Q format. signed Q15.1 format means, 15 bits for fractional and 1 for integer bits SO max range is - 2^15 / 2^15 to + (2 ^ 15 - 1) / 2^15 signed Q7.0 means, 7 bits ...
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1answer
37 views

Is there a counterpart for zero forcing in communications in DSP?

If yes, what is forced to zero?
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1answer
41 views

Why does my signal magnitude increase after adaptive filtering?

I am using a series-cascade of multiple NLMS adaptive filters each with step size 0.0040, leakage factor 1.0, and 100 filter taps. My signal gains magnitude at each step of the filtering, say the peak ...
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0answers
60 views

Removal of EEG noise using cascading adaptive filtering technique

I am required to filter out noise from EEG data using preferably Python or MATLAB. My dataset contains values for 64-electrode EEG along with their time-corresponding HEOG (horizontal eye movement), ...
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1answer
95 views

Help in designing filter for DUC (Digital Up-Converter) and DDC (Digital Down-Converter) using the built-in DUC and DDC blocks in MATLAB Simulink

Can anyone please help me in simulating a simple BPSK transceiver system that correctly receives a signal through a DUC and DDC? I have attached the screenshot of my Simulink system along with ...
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0answers
33 views

Relation between Gaussian and bi-Gaussian distribution

A Gaussian Doppler Spectrum can be seen as a bi-Gaussian Doppler spectrum in wich only one Gaussian function is present ( the second having a zero power gain) ? any help will be really appreciated
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0answers
135 views

How to compute position by FFT?

I have FFT data of x, y, z - axis, Now I'm trying to find the position through FFT data. Step 1 : For that, applied corresponds to an ordinary frequency i.e rad·s−1, ...
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2answers
57 views

Optimization of execution time of audio capture call in c/c++ (ALSA)

I am working on a Audio Deep Learning Project. In this C++ project I need to capture real-time audio and then do pre-processing and the pre-processed output is fed to the Deep Learning Network. I am ...
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0answers
143 views

Early-Late Symbol Synchronizer on Non-Triangular Pulses

I've been studying the basics of digital communications using the free book at desktopsdr.com. One of the examples is a 4-QAM demodulator which where symbol timing recovery is done with some Early-...
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3answers
83 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
203 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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1answer
36 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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0answers
199 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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0answers
18 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
28 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
122 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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1answer
349 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
38 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
2k 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
60 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
706 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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0answers
97 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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0answers
46 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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1answer
625 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
287 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
273 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
198 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
201 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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