Questions tagged [dsp-core]

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

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701 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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64 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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180 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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177 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 ...
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0answers
287 views

DC offset removal from tetra signal

Hey I want to remove DC offset from tetra signal captured as inphase and quadrature phase samples(I and Q samples). I and and Q samples are 16 bit signed values. I tried averaging 2000 I and Q samples ...
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0answers
215 views

Realtime image-stitching using a CMOS camera array

I am very new to DSP so please excuse any brevity... I have to design a 4x4 CMOS camera array and stitch the images together for further processing... I have tried doing that with opencv but the ...
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0answers
649 views

Fast DSP with in-chip ADCs

I'm looking for a DSP, around 500MHz, with an in-chip ADC (preferably 2 but 1 will suffice) that samples around 20MSPS. Does any such DSP exist or do I have to use external ADCs?
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1answer
43 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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199 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 ...
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0answers
81 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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560 views

Hilbert Transform, filters - two different phase graphs

Relationship between magnitude response and phase response for minimum phase is: Phase plots are obtained from Hilbert transform of real magnitude values. Input to Hilbert transform was a vector (...
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0answers
160 views

Raised-Cosine filter - design

I have a requirement to design bandpass Raised-Cosine filter with the magnitude and phase response as outlined in the picture below. I have read some literature online (still learning DSP) and I have ...
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0answers
872 views

BER of OQPSK Modulation

I am unable to find out a formula for the BER for the OQPSK Modulation. Currently, I am using the berawgn function from Matlab to calculate the BER for OQPSK but I am looking for an expression for ...
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22 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
23 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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39 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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36 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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1answer
69 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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40 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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911 views

buidling a bandpass filter without using inbuilt functions

I got an assignment which require building a bandpass filter without using any inbuilt functions. I wanted to create a third order Butterworth bandpass filter ...
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58 views

Estimating the power consumption of an arbitrary, real-time DSP algorithm

Consider a digital signal processing algorithm that operates over a block of $L$ [samples], with a real-time constraint $\tau$ [s] (i.e. its throughput must be $L/\tau$ [samples/s]). The theoretical ...
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45 views

aliasing and time shifting

I feel like it is a very silly question so I feel ashamed to ask my professor. So i would appreciate it if anybody would help me understand this: We know that $x(n-a)$ shifts the function $x(n)$ to ...
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366 views

Alternatives to pole-zero cancellation

I read that pole zero cancellation is a method to get rid of poles outside the unit circle. But I believe exact cancellation could be much harder because of finite precision effects. Are there any ...
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95 views

Passband modulation at 2.4GHz

As a project requirement, I am supposed to modulate a complex baseband signal to a carrier frequency of 2.4 GHz. These are the steps I followed for the baseband modulation and demodulation: Generate ...
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383 views

Audio block-processing discontinuities

I have an algorithm that looks at an audio frame, calculates a threshold in db-rms, and decides to filter the frame whether or not the threshold is crossed. However, filtering each block is producing ...
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3k views

Hilbert Transform - C or C++ code implementation

For a minimum phase system relationship of magnitude response to phase response is: In the search for C code implementation I have found a ht.c file on www.physionet.org. However, this code specifies ...
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59 views

Parameter selection for log distance path loss model

I am currently referring to the following book MIMO-OFDM Wireless Communications with Matlab in which the log distance path loss model is given by the equation: $P_{LD}(d)[dB] = PL_F(d_0) + 10nlog(d/...
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551 views

Biquad high-pass filter - calculating filter slope

I have to design high-pass IIR Biquad filter with cutoff frequency of 80Hz, sampling frequency 24kHz, and filter slope of 24dB/oct @ 80Hz. I have used formula for HPF biquad coefficients from here. ...
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436 views

FIR Filters: determining optimal number of taps and phase slope

As a design input for an low-pass FIR filter I have 6dB@1.6kHz, 24 dB/oct, sample frequency 48kHz. I need to specify phase response, and for that I need a phase slope. I have found, from Julius O. ...
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142 views

MATLAB band-stop filter design for room reverberation?

I took acoustic measurements of a room using sine swept method, and I also used Pioneer sc-37 to obtain more accurate data. I used RoomEQ wizard software and measured the impulse response of the room,...
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44 views

A/D converter quatization noise

I read that in A/D converter we usually use rounding or two's complement truncation and not the sign magnitude or 1's complement truncation as in this case the quantization error is correlated to the ...
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1answer
112 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
83 views

Multi instance DSP C Source code

I need to design a multi instance DSP algorithm implementation. Any reference Open C source code available ? Any links on multi instance implementations. -ben