Questions tagged [filtering]

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1answer
37 views

The result if order of two filter are reversed

Apply the Composite Laplacian Filter first, then apply the gaussian filter. Apply the gaussian filter first, then apply the composite laplacian filter. My work as below: Here we assume the original ...
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1answer
69 views

Respiration (breathing) signal movement artifact removal

Situation: I have breathing data of participants who have done different things during the study which caused their breathing change; watching mindless video to which resulted in unconscious regular ...
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1answer
190 views

DC Spur in DAC output

I have a design in which a signal from ADC is filtered and then played by DAC. In FPGA the real signal is converted to quadrature using DDS with 70 MHz frequency. After filtering, the baseband signal ...
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2answers
307 views

How Butterworth low-pass filter can be applied on a digital signal (i.e. discrete representation of a signal)?

I would like to know how to apply a low pass filter (Butterworth) to a digital signal. So, I have some values of the signal, let's say $S(t)$. Those values are equally spaced in time. I have read the ...
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0answers
170 views

Avoiding transients during filtering: Padding or Gustafsson's method?

I'm pretty uneducated when it comes to filtering, more of a stats guy, but I have to bandpass some signals and after a bunch of reading today I understand that the ...
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0answers
64 views

Noise Reduction: How can I filter out saw or other tool noises?

I recently came across old recording from the 1930's. Unfortunately not only is the quality low, it's a bunch of carpenters talking as they saw things and make all sorts of other noises. I've tried: ...
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1answer
354 views

How to calculate the Fourier transform of a mean filter in Matlab?

In Matlab, how can I calculate the discrete-space Fourier transform of a mean which takes the average of 4 adjacent points, with this kernel $$\begin{pmatrix} 0 &1& 0\\ 1 &0&...
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0answers
93 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 ...
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0answers
160 views

Electromyography: is this interference from the heart/how to remove it?

I'm using electromyography to evaluate the activity of lumbar erector spinae at various levels of effort. For this I also intend to sample the activity at complete rest (in prone lying). In my first ...
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0answers
38 views

Multitapering in time domain to obtain smoother filtered signals?

Suppose we want to approximate the instantaneous power $z(n) = |y(n)|^2$ of the discrete-time signal $y(n)$, where $y(n)$ is the result of filtering $x(n)$ with a given window $h(n)$, $y(n) = x(n) \...
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0answers
96 views

Fastest way to subtract known music tracks from live videogame audio stream

I want to isolate video game "action" sounds from the video game soundtrack. Given all ~20 songs, I'd like to filter them out from any live audio stream during gameplay. What is a fast and robust way ...
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1answer
108 views

Filter results from force measurements

I was provided with several force measurements. Precisely: The vertical compression force exerted from the milling tool on a working peace. The plot of the force looks reasonable and of course it ...
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2answers
4k views

How to apply low-pass filter to an I/Q signal to get baseband signal?

Can anyone help me with the demodulation technique of a modulated signal? I mixed it with I/Q signal. But I am very confused about low-pass filtering. I am trying to apply a low-pass filter, but I am ...
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1answer
488 views

Bandpass filtering an audio signal with MATLAB's $\tt kaiserord$: unknown new frequency component

I am trying to filter an audio signal with kaiserord in MATLAB, but I'm having a problem. Here is my code: ...
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0answers
39 views

How do I smooth-over transition from a time-series generated by one system to a time-series generated by another system?

So I have generated a time-series with length on 2N samples, where the first N are generated by one auto-regressive system and the N+1 to 2N are generated by another, similar, auto-regressive system. ...
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3answers
301 views

Adaptive filter with two inputs

I have general theoretical questions: Is it true that an adaptive filter with two inputs (one normal and one delayed by the single time increment) can completely get rid of any single frequency ...
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1answer
3k views

Choosing correct filter parameters for IMU sensor datas

I have a IMU sensor breakoutboard using ITG3701, LSM303D LinkProduct. This involves accelerometer, gyroscope and magnetometer. I have connected the sensors to a Arduino and sending the datas to ...
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0answers
50 views

Proximal operator of image Laplacian

Let $\alpha, \beta > 0$ and $\Delta := \nabla^T \nabla$ be the discrete laplacian operator, $$\nabla: \mathbb{R}^{n_x \times n_y \times n_z} \rightarrow \mathbb{R}^{3 \times n_x \times n_y \times ...
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0answers
2k views

Implementing a 2d butterworth filter for image processing?

I am at the moment trying to implement a bandpass butter worth filter which i then can use to apply on my original image. Problem is though My implementation of it using OpenCV doesn't seem to be ...
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0answers
168 views

amplitude compensation after filtering and mixing audio

I am working on a mixer which lets me mix dry and wet signal in a user defined ratio. Wet signal is basically just the all pass filtered signal. The mixing takes place in frequency domain and so does ...
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1answer
1k views

How does kernel work in unsharp masking?

I applied the kernel below to the picture for unsharp masking k = [-1 -1 -1; -1 8 -1; -1 -1 -1] I got ! Then I changed 8 to 14. And I got How does kernel work ...
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0answers
960 views

Instantaneous power estimation by discrete hilbert transform - how far does it smooth?

In my research area, instantaneous power in a specific frequency band is commonly estimated by the following procedure: Apply a bandpass filter on the raw signal (e.g. 80-90Hz bandpass). Estimate ...
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2answers
4k views

preprocessing raw ECG data

I need to preprocess raw ecg data in R, here is a sample already standardized. I'm not an expert in signal processing nor experienced in working with medical data, so I need concrete answers ideally ...
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2answers
80 views

Scipy Butterworth highpass filter results in distortion and bloated output

Trying to use Scipy Butterworth filter to filter frequencies below 54 Hz from audio files. Audio files are stored as WAVs. Sample rate is 44100. When I try using this filter the result is distorted (...
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5answers
171 views

An intuitive explanation as to why a matched filter is time reversed?

It easy enough to study correlation, and matched filters. But the challenge I see unmet anywhere to date and which I struggle to meet myself is a simpler presentation, to with more difficult task I ...
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1answer
147 views

Filter: IIR Filter

How IIR Filter coefficients formula derived? for LPF, HPF, BPF, BSF there are coefficients formula for a0, a1, a2, b0, b1. How it came? I would like to know the derivation for all those coeffs w.r.t ...
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2answers
58 views

DSP signal filtering and number formats

I'm working on a dsPIC33, using Audio Codec Board - Proto to read audio samples and for reproduction. In order to implement a relatively simple signal processing algorithm, I'm reading audio samples ...
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2answers
91 views

Distortion of OFDM signal when using RRC filter

I am trying to transmit OFDM signal in pass-band based on RRC filter, but what I noticed that the signal before the RRC filter is completely different compared with signal after the RRC filter (It ...
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2answers
189 views

Is There Any Filtering Technique with Element Wise Product Only Instead of Convolution?

Generally we apply a filter to reduce noise from signal. For example if the received signal is $S$ we would get a signal $Y$ with reduced noise applying a filter $W$ with the operation Y= convolution(...
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1answer
7k views

What is zero-phase filtering and forward-backward filtering?

I am pretty familiar with IIR and FIR filtering and I have implemented them in my several projects. However, recently i found something called zero phase filtering. I have tried to understand it but ...
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1answer
855 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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1answer
337 views

Relative Computational Complexity for Channelizer Implementations

Suppose I want to separate a broadband input signal into multiple narrow subbands using an FFT-based analysis filter bank. The filter bank uses a prototype lowpass filter and is implemented using a ...
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2answers
49 views

Adding channel effects to a signal

I have a a channel matrix "H" that is circulant. I have data blocks. I want to add the channel effects to the signal. When H was only a vector of channel coefficients I would say: ...
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3answers
90 views

I can't wrap my mind around the concept of filtering and the unit circle ? any help? [closed]

Can someone elaborate this theory where there is a link between the stability of a filter and the zeros or nulls being on the unit circle?
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1answer
112 views

Filtering in the Frequency Domain - DIP [closed]

I am trying to do some filtering with a gray scale image in the frequency domain. I am just getting into matlab after some time away from any signal processing or coding and can't quite seem to get my ...
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2answers
536 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 ...
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1answer
45 views

Is this a robust method for analysing correctness of a filtered signal? Is there a more robust method?

I have a recorded signal which contains several frequency components with stochastic phase and amplitude changes and want to extract the true signal with the other frequency components and noise ...
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2answers
163 views

In fourier space, how to apply transfer function with n frequencies to input data with m>>n frequencies

I have a transfer function in Fourier space with $N=2028$ frequencies $(\frac {0, 1}{(N\cdot dx)} \dots ) $ Where $dx = 0.1m$. I need to apply this transfer function to a signal with 20000 samples (...
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1answer
164 views

Pulse shaping and Baseband filtering

I'm slightly confused about baseband pulse shaping. Let's assume I have a complex data vector in an arbitrary complex constellation (QAM for example). I would like to pass this complex vector through ...
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2answers
105 views

Can filter “depth” be adjusted by mixing dry and wet signals?

Can filter "depth" be adjusted by mixing dry and wet signals? I.e. can I simulate e.g. a +6dB bandshelf/peak filter at 1kHz by mixing in some of the dry unequalized signal and some of a wet signal ...
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2answers
199 views

Matlab: is a highpass filter what I want, to obtain my required spectrogram? How do I design/apply this? (pic included)

The axes are wrong in the spectrogram. I have an audio wav file, and I know the sampling rate. this is read into the audioData variable. The audio is 1 channel. selected window length I believe (can't ...
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2answers
755 views

filtering in frequency domain

I am performing decorrelation of two channels of audio using all pass filtering. The two all pass filters for two channels are of very high order (>1000 but with not all coefficients non zero). I have ...
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1answer
516 views

Filtering Passband Signals using Complex Baseband Filtering

So I'm given a pass band filter with specific transfer function $H_p(f)$, I want to implement this via baseband processing. I already know how to take the input signal $u(t)$ and process it such ...
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2answers
155 views

Why is successive decomposition of a signal performed only on low pass component?

The question says it all. In typical (wavelet-like) decomposition of a signal, why is only the low pass component chosen for successive decomposition ?
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2answers
436 views

Apply Singular Value Decomposition on Gabor filters

According to my question in this link, I applied (with a big help from lennon310) the separability of Gabor filters of any orientation. The only disadvantage of this method is the convolution in the ...
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1answer
40 views

Next steps for analyzing this PWM algorithm and choosing a low pass filtering technique?

The Broadcom BCM2711 GPIO controller used by the Raspberry Pi has a hardware PWM implementation option using an algorithm that provides an average value defined by a rational fraction $N/M$ defined as ...
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1answer
164 views

A filter with low phase shift or no phase shift at all? Which one? [duplicate]

I'm seeking a good advice for a filter. Can you help me? Right now I made my own simple filter. Not sure what I'm doing, but it seems to work. ...
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1answer
35 views

How can I plot Wiener filter's behaviour for different taps (memory filter)?

What's the expected behaviour of Wiener Filter for different taps (memory filter) , when input signal is a sinusodial tone? Do you have any idea how I can plot or simulate Wiener Filter's behaviour (...
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1answer
53 views

How to quantify phase distortion of a filter specific to a signal?

For example, a EKG input signal is filtered with a known transfer function that does not have linear phase. How do you quantify the phase distortion of the output signal given you have the input? I ...
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2answers
80 views

Downsides of higher order filtfilt's

I understand that with a standard filter, higher order will yield sharper frequency selection at the cost of increased phase delay. However, with so-called zero-phase-distortion filters like that ...