In signal processing, a filter is a device or process that transforms a signal by selectively choosing specific frequencies and leaving others with appropriate gain or as it is.

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GCC-PHAT cross correlation

I am trying to implement GCC-PHAT in python. The approach is similar to the following two links: link1 and link2 It seems the only difference between GCC-PHAT and normal cross-correlation using FFT ...
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Filtering and decimating input as sections in multiple stages

What would be my options to decimate a large input signal available only in sections (or blocks) due to DMA type of interface? The FIR should be about order of 18 only and the total decimation factor ...
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How to handle encoder wraparound in an alpha-beta filter

I want to use an alpha beta filter to estimate speed from an encoder that's mounted on a motor shaft. The encoder position wraps around to zero every 8192 counts; it gives the absolute angle of the ...
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2answers
54 views

Reconstruct FIR frequency response from the coefficients

There is a lot of tools for designing FIR filters. Using one such tool I have designed a filter that satisfies my needs. It's coefficients are as follows: ...
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1answer
38 views

correct for shift from butterworth filter?

I have implemented a python butterworth filter but it seems to shift the data a lot, the x location of my data is very important to me for my analysis. Is there a way to correct for this shift ...
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4answers
85 views

The reasons for filter explosion

Assume signal $x$, sampled at $f_s = 44100\; \mathrm{Hz}$. I tried to filter it using the Butterworth bandpass filter ( $30\; \mathrm{Hz} - 70\; \mathrm{Hz}$) of order $8$. However, as a result I get ...
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23 views

comb filter - attenuation

I have a signal and from 'standard information' I know the predominant frequency is at 1Hz and then there should be three harmonics at lower amplitudes. My signal however has a lower amplitude at 1Hz, ...
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2answers
33 views

variable length no-refrence filter algorithm

I'm searching for a an algorithm or an idea of implementation. I need to filter a signal in real time, and want to change the length of it (number of taps) in order to change the delay. In low ...
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1answer
40 views

Problem with a code of FIR Lowpass filter

I'm making an interpolator (filter) to remove aliasing but this doesn't work. I see also some strange behaviour: x[i - j], but what if ...
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43 views

Designing a practical integrator

One way to describe a practical integrator ("leaky integrator") is $$ H(s) = \frac{g R}{1 + sRC} $$ I am trying to understand how to choose the values $g$, $R$ and $C$ because in practice, I will ...
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20 views

What filter should I pass to these signals to make them constant on every piece

I'm analysing the frequencies of something I'm recording (whistling), which is resulting like this (yes, it's super mario bros' beginning song: I'd like to filter it to make it remain like: in ...
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1answer
136 views

Real and Imaginary Frequency Responses of a single complex pole

I'm filtering a real signal with a single complex pole with a complex coefficient (a_re [real part] and a_im [imaginary part]), I also have a gain coefficient but I'm gonna leave it out for the sake ...
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56 views

How to reconstruct the signal which is subjected to FIR filtering - Matlab

I have a signal $x$ which is corrupted by Gaussian noise. The signal $x$ is convolved with a filter with coefficients $h = [1, -2, 1]$, to generate signal $y$. Now the signal is taken to Discrete ...
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45 views

Difference Equation - Time Response analysis of Discrete Systems

We know that that the difference equation formula for computing an output sample at time $ n$ is based upon its past/present input samples and past output samples in time domain: $\displaystyle ...
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3answers
68 views

Are reconstruction filter always needed?

in mixed (digital and analogue application) where you have to convert a signal from continuous to digital time domain and then back again to continuous time an anti-aliasing LPF is needed and in ...
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1answer
37 views

How to calculate the time delay of a signal by a first-order filter

lets assume following signal $$x=a_1 \sin(2\pi f_1 t) + a_2 \sin(2\pi f_2 t).$$ This is passed through a simple single-pole filter with cut-off frequency $f_c$. How is $x$ delayed by the filter (peak ...
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12 views

Gaussian kernel standard deviation and spacing

Suppose I know the normalized gaussian weights for a kernel of standard deviation 1.0 evaluated in the sample positions $O = \{-2, -1, 0, 1, 2\}$. Let's call these G. I know that the same values G ...
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1answer
43 views

Advice on filtering a CW signal for processing

This question is related to my other recent questions. In the last question, I learned that I should filter about 4 * WPM (words per minute) around the carrier frequency if I want to recover Morse ...
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36 views

Feature selection of Gabor filter

I am working with Gabor filter with level set method here. As you known, gabor filter gave many filter output. Some of them is not necessary. Could you suggest to me some method that can choose best ...
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1answer
35 views

How to decode Morse code (CW)

I have two related questions on this topic. First, about how wide should the filter be around the carrier frequency? What is used to determine this? Second, in Python, I must use 'numtaps' when ...
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1answer
55 views

Quantizing a filtered signal

In the process of trying to learn dialup, I've managed to learn/figure out everything except how to convert a demodulated/filtered signal back into the original data. Here I have a random two-bit ...
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111 views

How to estimate PSD, and time delay of a random signal?

I'm trying to estimate the power spectrum and time delay for random signals. the signals were taken from two sensors, 10 cm apart. My aim is to estimate the PSD first then estimate the time delay in ...
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3answers
66 views

How to filter noise from continuous path in image

Problem statement In the dataset above, you can notice a continuous path and then a noise-band at around 320-400 on the x-axis. I want to be able to apply a filter that amplifies the continuous ...
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25 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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13 views

Extraction of a decision boundary (LDA) after a systematic querying of the feature space and convolution with Sobel filter (examples in numpy)

I am doing some experiments with LDA (Linear Discriminant Analysis), in python. Now I am at the point in which I would like to display the separation planes in the 3-dimensional feature space. I ...
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1answer
79 views

Frequency response of a filter - fft vs. freqz

I have designed my filter using firhalfband(N,Fo,A), which gives me the filter coefficients. Then I try to see the frequency response of it. I use these two ...
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1answer
61 views

Low pass filter to maintain edge information

I am looking for a kernel acting as a low pass filter that satisfies these conditions: $$K(-\mathbf{u})=K(\mathbf{u}) \tag{1}$$ $$K(\mathbf{u}) \ge K(\mathbf{v}), \;\; \text{if} \;\; ...
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51 views

Design and implementation of causal band-pass filter for biosignals - what to consider? [closed]

I currently work in a project of clinical software (C#) which deals with clinical biosignals, specifically EMG and other low-frequency signals (load-cells, goniometers, and other biomechanical ...
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1answer
28 views

What are some example of applications for LTI systems

I'm giving a lecture on LTI systems. I encountered some questions: for a discrete LTI system H with impulse response h, is the system applied on signal x(t) equals x*h - normal discrete convolution ...
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2answers
62 views

How to visualize and understand Gabor filter using matlab

I am reading a paper, in which the author defines gabor kernels as : $$ \psi_{\mu,\nu}(x,y)=\frac{\lVert \mathbf{k}_{\mu,\nu} \rVert}{\sigma^{2}}\exp (-\frac{\lVert ...
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1answer
72 views

Filtering out unwanted square wave (Radio: DCS/DPL signal)

I've need to remove a DPL/DCS signal from a recording. Roughly speaking, a DCS signal is a low frequency (67.15Hz) square wave sent at the same time, over the same carrier, as speech. Because it is a ...
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2answers
70 views

What is wrong with this method of detecting the presence of a signal?

I am trying to decode Morse code from an audio (wav) file. Here is the technique I am using to detect the presence of the signal: I have a window of size (sampling_freq/tone), where tone is the ...
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2answers
67 views

Filtering a Morse code (CW, OOK) signal

I am building a morse code decoder in Python. I am using SciPy and Numpy for FFT, filter, etc. It is coming along ok, but the signal is noisy. I am extracting the dominant frequency from the FFT of ...
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57 views

Any way to compensate for fractional delay filter's weakness between taps?

A little background: I have an FPGA (running at 200MHz) that I am creating multipath events using buffers, all of that works. I take RF, mix down to IF, sample, make adjustments, out the DAC, and ...
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1answer
25 views

Parameters of standard audio channel filters

I want to do some simulations of speech recorded on microphones. In order to simulate the channel response, are there any standard (FIR) filter parameters available that I can use?
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53 views

Filter that is like low-pass filter but for intensity?

Is there a type of filter for processing spectrum data that will subtract spikes in intensity? I was thinking of something like a low-pass filter but that takes away data if the intensity is too high. ...
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1answer
61 views

Correlating accelerometer data

I am getting a signal from a device. This signal contains a non-random noise signal in it. I want to remove that noise from the signal. This noise is correlated to the motion the device is having. So ...
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47 views

Correlating two signals [duplicate]

I am getting a signal from a device. This signal contains a non-random noise signal in it. I want to remove that noise from the signal. This noise is correlated to the motion the device is having. So ...
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33 views

Separating fast decaying signal from a slowly decaying signal

I have a discrete signal $x[n]$ which is exponentially decaying but the decay constant is not known. This signal is observed over a certain time period, say $T$. The only known detail about $x[n]$ is ...
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digital filter and epoch length

A reviewer has asked me to re-filter my data to remove baseline drift. Each sweep is 170ms sampled at 1000Hz and the reviewer wants it high-pass filtered at 0.5Hz. The original bandpass settings on ...
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1answer
79 views

Best way to inverse Hilbert on FPGA?

I have a real signal that is being sampled via ADC and passed to a Xilinx V6 FPGA. I then do a Hilbert transform on it using the FIR compiler core and get the complex signal (I and Q) to tweak the ...
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1answer
95 views

filtfilt giving unexpected results

I am trying to filter EEG signals using butterworth filter and filtfilt. I have gone through a lot of documentation and these 2 commands seem sufficient for filtering. However, the results are ...
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3answers
89 views

Does perceived ADC resolution, range and precision depend on fixed point format? What is the best way to deal with this for signal processing?

I am working on a system that is implementing a digital filter that was designed for another system that had an FPU and a 12bit ADC. The system that I am working on is using the same filter without an ...
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1answer
25 views

How to model the dispersion which is introduced by an electro-absorption modulator (EML)?

I need to model the dispersion which is introduced by an EML in Matlab. The parameters are: Lambda: 1310nm PAM4 power: Pmax = 1.09 mW Pmin = 0.13 mW Bandwidth = 56Gbps
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2answers
76 views

Why impulse response

Why do people calculate impulse response? Generally systems are designed after modelling the possible input signals and calculating the appropriate filters for the desired outputs. Where does impulse ...
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1answer
33 views

Difference between delayed down-converted signal and a down-converted delayed signal

I am having trouble understand what I think should be a pretty simple concept. Conceptually I can understand that a signal that has been down-converted, and then delayed in time is different than a ...
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1answer
74 views

Wiener Filters - Obtaining image and noise spectrum

We all know the famous Wiener Filter, $$G(\omega_1 , \omega_2)=\dfrac{H^*(\omega_1 , \omega_2)}{|H(\omega_1 , \omega_2)|^2 + \dfrac{N(\omega_1 , \omega_2)}{S(\omega_1 , \omega_2)}}$$ How to we use ...
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1answer
55 views

Filtering method that minimizes ringing in time domain, implementable in Python

I am acquiring data through an oscilloscope, which I am then filtering. I am doing all of this in Python. I need a bandpass filter with stop bands from 0 to .4MHz, and 4MHz and up. My pass band needs ...
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2answers
54 views

Masking the mains hum to improve anonymity

Today, a news blurb appeared that claimed that attackers could identify the location in which an audio record was taken by matching the mains hum to known data. Sadly, I fear that this might be ...
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2answers
77 views

Could multiple echo reflections be combined to improve signal to noise ratio?

I am working on a system processing ultrasonic echo reflection. The transmitter creates a short ultrasonic pulse and the receiver analyses the reflection. If I send two pulses (back to back or with ...