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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How to interpret integer coefficients for a filter generated by Matlab

I have always used high-performance 32-bit microcontrollers with hardware FPU support. To calculate the coefficients of digital filters, I used the fdatool package in Matlab. I used single-precision ...
red15530's user avatar
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Is it possible to work with DSD audio on MATLAB?

Just for starters: I am very new to audio processing I was looking for audio data that is in 1-bit format and I stumbled upon DSD. From what I was able to read it is old but kind of controversial ...
Sahil Sharma's user avatar
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250 views

Computing Phase detector in PLL

I have been trying to implement PLL in a STM32 microcontroller. Starting with the first step, I want to implement a phase detector. I tried to first simulate in Python so that I can later use in STM32....
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Computing phase difference between two sinusoidal signals

Based on this post here, I tried to calculate the phase difference between sine waves. Here is my code. ...
Rima's user avatar
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FIR code implementation question [closed]

I have found this piece of code in Infenion DSP Library (asm code): Fir_Blk_16() ... ;(ACC)=(ACC)+h(i)*x(n-i)<<1 as comment I don't understand why the result ...
Anton S's user avatar
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Given a signal at one scale, how do I scan a differently-scaled signal to see if it contains that subsignal?

Suppose I have a signal with a characteristic shape - say, a right angle ‘cliff’ where it is high, drops like a rock, and stays low. I now have a bunch of stock data that are at completely different ...
dnb's user avatar
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I have a doubt regarding FFT size in 5G NR 60 MHz Bandwidth

maximum RBs for 60MHz with SCS = 30 kHz is defined to be 162 RBs, so NSubcarriers is 162 x 12 = 1944, 1944 < 2048. Then NFFT size for 60 MHz bandwidth with SCS = 30 kHz should be 2048 instead of ...
Eliza's user avatar
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Find the impulse response of a causal 52 day moving average system to be used in stock market [closed]

Moving average system………………………. How to solve this one?
Akhil Singh Andotra's user avatar
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Minimum Sampling Frequency

For a continuous time signal with spectrum $$X_c(j\Omega)=1 \quad\text{for} \quad10\pi < |\Omega| < 50 \pi$$ Choose the minimum sampling frequency so that there is no overlap in the spectrum of ...
Basit Mujtaba's user avatar
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DSP Clock & Sample Rate Problems (ADAU1172, PDM Microphone)

I'm working on ADAU1772 evo.board with a PDM MEMS microphone. Microphone's clock is 3.072MHz. DSP Master Clock is 12.288MHz. DSP sampling rate is at 192kHz and all DSP modules work with 192kHz. 1st ...
BugraKezan's user avatar
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Efficient CIC Decimation Filter in C

Does anyone have advice on efficient CIC Decimation filter implementation. I designed a basic 4th order CIC filter with a 64 decimation ratio in C (below). I am currently testing this on a Sharc SC589 ...
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Difference between lattice and wiener FIR filter

I know the structure of lattice and wiener filter. But I though FIR filter is always FIR (wiener, tap + delay) filter, But what is the advantage of lattice structure, which in the CMSIS DSP filter ...
mohammadsdtmnd's user avatar
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OBC DA FIR Filter Implementation

I understand the theory of Distributed Arithmetic and its application in FIR Filter implementation to avoid the excessive power consumption in the multiplication operations But a question here, when I ...
digitalman's user avatar
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1 answer
283 views

STM32F7 Discovery board vs dsp kit tms320c6713 vs Rasberry Pi?

We want to design dsp kit Labs for our undergraduate students for following tasks/topics. 1)Sampling and Quantization of real time Audio Signals on DSP Kit ; analyze the effect of Aliasing and ...
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Designing an FIR filter with Gain = 1 on TI DSP processor

I'm trying to design an FIR filter with gain = 1 on DSP processor that does not support floating representation. I'm using MATLAB to design the filter and then transport the taps into my code. The ...
Euler's user avatar
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How can I verify my code?

...
john's user avatar
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What's practical interpretation of showing DC Offset in frequency domain?

In homodyne reciever, we read about LO Leakage causing DC Offset. Say my output at downconversion is in analog voltage range +5 and -2, and there is a DC offset of 1 unit So, it can be added right? To ...
Reddi Suresh's user avatar
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2 answers
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What applications do DSP processors primarily target?

I've recently been reading up on Analog Device's SHARC floating point DSP. It seems cool reading about DSP specific processors, however, I'm curious what applications actually leverage DSPs. For ...
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Verification of intermediate outputs in Signal processing chain of FPGA

I have two ADC interfaces with FPGA and 32-bit processed output data from FPGA. I have an IO signals to verify the Inputs and Outputs of FPGA. But I dont have any test points to verify the ...
srihari's user avatar
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Why we sometime use some values of fixed point and floating point both in dsp?

I have been writing kernels for a communication dsp device, which uses cmac/cmad operation on fixed and floating point numbers. I am not very clear why both types is used in the same operation.
Nits r's user avatar
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Digital Filters

If I want to remove the baseline drift in my ECG signal, which digital filter should be used without distortion and shift in my filtered output? What are the necessary things I have to look for the ...
Kesavaraja C's user avatar
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How to replicate/simulate experiments done on dsp kit 6713 without using hardware?

Due to covid, our university is now offering online classes to under grad students, even labs are conducted online. But we are bit confused regarding evaluation? We make videos of demonstration of dsp ...
cvz's user avatar
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Options for DSP hardware for Under Grad Lab?

We have TI 6713 dsp kits in our Lab but since those kits are very old(They were started/released near 2000) we are now considering to get new dsp kits ,so our under grad students may clear their ...
fgj's user avatar
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1 vote
1 answer
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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 ...
Khubaib Ahmad's user avatar
1 vote
5 answers
266 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 ...
Clauvunate's user avatar
1 vote
2 answers
134 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 ...
LifeIsBeatiful's user avatar
1 vote
1 answer
314 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 ...
dsp's user avatar
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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 ...
dsp's user avatar
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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 ...
dsp's user avatar
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1 answer
341 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 ...
Lilrags16's user avatar
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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.
Hammad Ahmed's user avatar
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2 answers
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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 ...
Matlab_lover's user avatar
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0 answers
54 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 ...
vmontazeri's user avatar
2 votes
3 answers
390 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 ...
DSP_CS's user avatar
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1 vote
1 answer
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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.
Nits r's user avatar
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1 answer
618 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 ...
DSP_CS's user avatar
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1 answer
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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 ...
Amit Sravan's user avatar
1 vote
1 answer
332 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 ...
DSP_CS's user avatar
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2 votes
1 answer
162 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 ...
CMH12's user avatar
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1 vote
1 answer
431 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 ...
thileepan's user avatar
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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 ...
Khubaib Ahmad's user avatar
0 votes
2 answers
619 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 ...
ksi's user avatar
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0 votes
1 answer
47 views

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

If yes, what is forced to zero?
BusyP3rs0nal1ty's user avatar
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1 answer
103 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 ...
Saprativ Ray's user avatar
-1 votes
1 answer
345 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 ...
Amit Sravan's user avatar
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0 answers
130 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
Abby_DSP's user avatar
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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, ...
SamDev's user avatar
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1 vote
2 answers
278 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 ...
Saket's user avatar
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1 vote
0 answers
355 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-...
Manuel Menzella's user avatar
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3 answers
153 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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