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0
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1answer
43 views

ARMA & MA methods: how do you know the error terms?

Reading the ARMA model for the first time, and I'm confused. Let's say I have a time series x = [1, 2.1, 2.9, 3, 4.1] According to the ARMA model, $X_t$ is a ...
1
vote
1answer
126 views

How to remove ECG artifacts from EMG data?

I'm new to signal processing and try to analyze EMG data from pectoralis major and other respiratory muscles. Unfortunately, there are strong ECG artifacts in all recordings and I don't know how to ...
0
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0answers
15 views

Integrating values in a matrix in MATLAB- Audio Loudness Meter

I'm trying to implement an algorithm in MATLAB as specified by ITU-R BS.1770-3. The goal is to measure loudness of an audio file in 3 different ways: un-gated LKFS, gated LKFS, and true peak. So far I ...
0
votes
1answer
45 views

Manual filter or digital filter first for EEG?

I am getting conflicting advise regarding how to clean my EEG data: 1) Manually remove artefacts first and then apply digital filters OR 2) Apply digital filters first and then manually remove ...
2
votes
3answers
41 views

How to Classify Image Filter as Low Pass Filter (LPF) or High Pass Filter (HPF)?

I am currently reading on the topics of image filtering and convolution kernels. While the process of convolution is clear to me, I do not understand, how one arrives at the statement, that a certain ...
1
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1answer
77 views

What are the advantages and disadvantages separability of Gaussian 2D filter?

I know separability of a Gaussian filter enhances the computational complexity from $\mathcal{O}(L^2*N*M) \to \mathcal{O}((L*N*M)$. How this really reflect on my program? I suppose the number of for ...
2
votes
2answers
137 views

Extended Kalman Filter (EKF) for Non Linear (Coordinate Conversion - Polar to Cartesian) Measurements and Linear Predictions

I'm new to Kalman filtering and state estimation and I'd like some guidance on EKFs. Currently, I'm trying to use a linear prediction model coupled with nonlinear measurements to estimate the state ...
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0answers
23 views

How to properly filter an image in the frequency domain using matlab

my problem is the next: I´m starting to understand the world of image processing and I´ve been kind of bugged with the idea of filtering images in frequency domain. I have made a test programm in ...
-1
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1answer
59 views

What do poles do for a filter?

What are the disadvantages of having too many poles? Thanks.
0
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1answer
28 views

When applying a bandpass filter designed in Matlab's DSP system toolbox, is it necessary to do any normalization?

I'm applying the next bandpass filter to the data of an impulse response coming from .wav; it fades data below 10 Hz and above 400 Hz: ...
0
votes
0answers
59 views

filtering 1D signal with very low SNR(<2)

I have recorded speech using a microphone(with 16bit A/D converter). The signal has a frequency range of 300Hz to 1000Hz, I have recorded 3.5 seconds of the speech and 1.5 seconds of the baseline(no ...
0
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0answers
33 views

How to filter a sequence of frames?

I'm have a sequence of 2-dimensional MxN frames. I have concatenated these frames to form a 3-dimensional MxNxT matrix. Now i want to filter this 3D volume by 2 types of filters (a 2D log-Gabor in xy-...
0
votes
1answer
105 views

Implementing FIR/IIR “alike” filter in R

I am new to signal processing. I am trying to simulate something similar to IIR/FIR filter with $k$ delays to imitate acoustic echo reflection. The difference equations for FIR and IIR respectively ...
0
votes
2answers
184 views

How to use Kalman filter for altitude prediction based on barometer data?

I have barometer noisy data with known variance. I studied Kalman filter but I did not find an answer to this problem: My process model is: altitude is changed because of velocity that is changed ...
0
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0answers
35 views

audio filter designing help

buzzer frequency range recording of audio signals(environmental sounds with speech) with buzzer soundI have audio signals(environmental sounds with speech) with buzzer sound, I have to attenuate only ...
1
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1answer
71 views

Do integration/differentiation processes work as simple filters?

How these processes are different from simple IIR 1 order filtering, FIR filters in terms of amplitude and phase characteristics?
1
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1answer
743 views

Impulse response of ideal filters

I am aware that an ideal low-pass filter in both continuous time and discrete time has a $\mathrm{sinc}$ impulse response. What would the impulse response of an ideal high-pass or band-pass filter ...
1
vote
2answers
143 views

BIBO Stability and the convergence of the frequency response of a system

It is my understanding that an LTI system is BIBO stable if and only if its impulse response $h(t)$ is absolutely integrable. This also happens to be one of the Dirichlet conditions for the ...
0
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2answers
116 views

Bandpass filtering IQ samples from USRP

I have an USRP (B205mini) with an RX frequency of 900 MHz and a bandwidth of 100 MHz. I want to distinguish between two raw, unmodulated signals at respectively 875 and 925 MHz. My approach would be ...
3
votes
1answer
90 views

Use samples of Fourier transform as DFT?

Consider the LTI system given by: $H(z) = 1 - \frac{1}{2}z^{-1}+\frac{3}{4}z^{-2}$ Let $x[n] = (\frac{1}{2})^nu[n]$ be the input to the system. We want to find the output for $n = 0,1,...,N_a$, using ...
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0answers
203 views

How to implement real time wavelet transform

I have a real time system where incoming data rate is 512 Hz. This signal needs to be filtered in real time. My algorithm has a mix of wavelet filters and FIR filters. During my algorithm development, ...
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0answers
47 views

Is there a proper filter for my application?

I am new to DSP and am working with a signal that has a floating baseline. The signal baseline roughly centered around 10^7 and the baseline can float upwards or downwards 2000 counts in a matter of ...
4
votes
1answer
2k views

Apply butterworth filter (lowpass) to a signal

I'm working on a DSP lab and I can't figure out how to apply the filter I've created to a sound. The only thing I found online that did something to my sound was ...
1
vote
2answers
333 views

What is the difference between the sampling frequency of signal and sampling frequency of filter

I believe that there is no connection between the sampling frequency used for converting an analogue filter to digital filter and the one used to sample a signal that the filter will be used on. But I ...
0
votes
1answer
315 views

Extracting the carrier signal from AM

I am new to DSP and leaning about amplitude modulation. I am trying to demodulate a 5 Hz signal that is modulated by a 50 Hz carrier. I am trying to extract the carrier, not the the 5 Hz signal. A ...
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0answers
13 views

Removing real time signal mean with sliding DFT/iDFT [duplicate]

I'm looking for a way to remove the mean of a real-time data stream while keeping very low frequency data intact. Would it be feasible to use a sliding DFT to generate the frequency components, the ...
1
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0answers
88 views

Real-time oscillation-like noise filtering

I am making real-time measurements, which are affected by some noise. This noise has oscillating pattern, i.e. the total signal tends to oscillate around the mean value. The examples of measurements ...
3
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0answers
96 views

Use of an impulse-response system to model sediment transport along a river reach

I am new to signal processing and have come into the subject from the study of rivers and basic geophysics. I am trying to test an idea previously put forward that the sediment transport response of ...
0
votes
1answer
174 views

How to eliminate negative frequencies from IQ signal

I have a 192kHz IQ signal from an RF receiver, and i'm trying to remove signals in the negative (or positive) frequency spectrum. I see that the negative frequency signals are -90 degrees phase ...
0
votes
1answer
88 views

Remove high resolution spurious peaks from sinusoidal signal

I have the current from a LEM transducer measured. The measurement is taken on the output of a transformer. The signal is a 50 Hz signal, measured at 100kHz. When the demand of the system is increased ...
0
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0answers
33 views

How can I get to filter out the sawtooth signal to ramp signal

The figure is the result in arduino. I used arduino uno and 30V dc gear motor. I wanted to read the actual angle rotation of the motor by using arduino uno. But the result is unexpected and wrong ...
1
vote
0answers
47 views

Processing digital sound in Audiokit to sound like the 1960's

I am looking to imitate 70's analog equipments sound. I was wondering is there a way to create tape saturation / distortion inside Audiokit? I want to add that and randomized tape crackles and ...
1
vote
1answer
731 views

Pre-emphasis filter for speech recognition

I'm trying to design a: $$H(z)=1 - 0.95z$$ FIR filter, for speech signal. I am using this arm_fir_example_f32.c code shown below. And I need to change the filter ...
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0answers
19 views

Adaptive Facial blurring

I got one problem related to facial blurring. The problem is Input: Image with facial (can be any location, not aligned, multiple faces) Know the bounding box: ellipse bounding box I would like ...
4
votes
1answer
271 views

Polyphase Filter decomposition. It is not working

So, I want to do a polyphase implementation of a filter bank (I had problems with it since the beginning and I already asked for help to the community in the past, indeed: Norm MPEG-1 Layer III (Mp3) ...
0
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0answers
26 views

signal transformed using FFT, remove some magnitude in certain frequency, and back transformed using IFFT, is it possible? [duplicate]

i am so curious, filter usually using FIR and IIR to remove all frequency except a certain range frequency (lets say form 0-100 Hz, i want a signal from 20-50 Hz). If i have a signal or a discrete ...
0
votes
1answer
280 views

Convolution in spatial domain is multiplication in frequency domain

I have to prove convolution in spatial domain = multiplication in frequency domain using two matrices. $$ x(m, n) = \begin{bmatrix} 1 && 2 \\ 3 && 4 \end{bmatrix} $$ $$ h(m, n) = \...
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0answers
58 views

How can I implement Two Pass Split Window Algorithm in Matlab

I'm new in DSP & matlab.I'm trying to implement TPSW algorithm to filter some noise.Can someone give me some hints about implenting this algorithm?
4
votes
2answers
98 views

Filtering without using any transforms

A composite signal is defined as $$x(t) = \sin(2\pi 10t) + \sin(2\pi 40t) + \sin(2\pi 60t) + \sin(2\pi 50t)$$ Separate the four frequencies without using any of the transform techniques.
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0answers
38 views

What to consider using a PLC for data acquisition?

for a project I have to read out data from a force sensor using the background task of a PLC (programmable logic controller). I'm now facing two problems: My sample rate is not constant due to ...
0
votes
1answer
361 views

Implementation of the filter function (Matlab) C/C++

I am trying to apply some filtering in C/C++ code, I have designed an algorithm using Matlab and it works well, so now I am moving it to native code. I am ...
2
votes
2answers
60 views

Recommendation for studying filters

I have a project to do about what happens to a periodic function when we pass it through a low-pass, high-pass and band-pass filter. I have no expressions for the filters or the function, I just have ...
0
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0answers
43 views

How much does a filter realistically attenuate?

I've got a legacy software which filters (with a third party library) a given signal (a sinusoidal wave plenty of noise) with a passband filter which is really, really narrow; the filter cut-off ...
5
votes
1answer
177 views

Is there a z-transform like for variable sampling rate signals?

I'm working with signals with variable sampling rate (the time space between samples is not constant). I know the delay between samples but I don't wont to interpolate the signal. Is there a way to ...
1
vote
1answer
48 views

position of useful filtered data inside DFT based filtered output

I'm trying to filter a 400 samples signal with various bandstop FIR filters (constant group delay), one at a time to see which one gives me the desired result. Every filter was build using the Kaiser ...
1
vote
1answer
213 views

Image zoom out using Gaussian blur and downsampling

I am trying to implement a simple zoom out algorithm using two steps: The image is blurred using gaussian convolution of standard deviation $0.8\sqrt{{\tt zoomfactor}^2-1}$ as we saw in class. The ...
1
vote
1answer
305 views

How to perform a gaussian blur using FFT?

I am trying to implement a gaussian filter with matlab.Here is my implementation: So far All I get as output is a black image. Any hints? ...
0
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0answers
21 views

How does one delay a signal and allow its phase to propagate vs time with a specific slope or rate?

How does one delay a signal and allow its phase to propagate vs time with a specific slope or rate? How can one -digitally- simulate the effect of a propagation line, in the sense that, depending on ...
0
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0answers
95 views

Implementing a ram-lak filter in javascript implementation of fast fourier transform?

Having a non-mathematical background I'm trying to understand how to implement filtering of data in the frequency domain as part of FFT. A javascript implementation of a discrete FFT is found here. ...
0
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1answer
37 views

Buffering block and filter length in real-time processing

I have a signal sampled at 256 Hz, which I want to filter with a 50 taps long FIR filter in real time. Would it be a problem, if my data block size is only 32 samples? And should I then concatenate 3 ...