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Question: <p>I have a device that, in the lab, takes voltage readings every 2 seconds. However in real-life application the device would only wake every 10 minutes and then take a 100 readings whilst awake. How do i change my collected data to mirror real-life application? Only then can I interpret if my device is work... | https://dsp.stackexchange.com/questions/61794/how-to-downsample-my-data-readings-from-0-5hz-to-0-001667hz-do-i-filter-and-dow |
Question: <p>This question is a part of a more general question the answer of which I don't know -
<em>How to apply a filter in the freq domain and then convert the filtered signal back to the time domain?</em> Well, I partially googled the answer that I need to</p>
<ul>
<li>convert the signal in FFT</li>
<li>multiply ... | https://dsp.stackexchange.com/questions/76023/how-to-apply-an-atmospheric-attenuation-filter-in-the-freq-domain-and-then-conve |
Question: <p>I have just implemented a Discrete Time Convolution between HRIR filter at a range of angles. I also implemented an Overlalp-and-Add method and FFT and IFFT to compute the same convolution in Frequency Domain.
The HRIRs are the same measurements which are taken from a public database for both DT and Overla... | https://dsp.stackexchange.com/questions/56755/comparisons-of-fir-causal-filters-of-type-hrir-in-discrete-time-and-frequency-do |
Question: <p>The l1 trend filtering is expressed by taking <span class="math-container">$||Dx||_1$</span> where D is the second difference matrix why is taken of size (n-2)xn rather than circulant matrix of size nxn. I have implemented in both ways but there is not much difference in the results</p>
Answer: <p>If you... | https://dsp.stackexchange.com/questions/82499/why-is-the-second-difference-matrix-of-size-n-2-x-n-in-l1-trend-filtering |
Question: <p>Suppose that I have two signals $x[n] = \left\{2,4,1\right\}$ and $p[n] = \left\{5,1,8\right\}$ and I want to multiply them.</p>
<ul>
<li>How do you do that?</li>
<li>How different is it from convolving two signals?</li>
</ul>
<p>I understand that multiplication in one domain is equal to convolution in o... | https://dsp.stackexchange.com/questions/10453/how-are-two-signals-multiplied-and-how-is-it-different-from-convolving-two-sign |
Question: <p>I am trying to filter out some square wave signal to within a limited band (1/4 or 1/8 of the original), I realized that there's a lot of ringing in the wave when I use my filter (elliptical), I also tried Butterworth, and others (given in Matlab fir1, and classic iir filters) but the only filter that seem... | https://dsp.stackexchange.com/questions/23035/low-pass-filter-with-minimal-ringing |
Question: <p>I'm confused because of filter <em>length</em>, whether (such) filters can be used to filter audio on a "per sample basis"?</p>
<p>By per sample basis I mean that I would like to filter audio one sample at a time, but vary the filter parameters even one sample at a time.</p>
<p>What then confuses me is, ... | https://dsp.stackexchange.com/questions/32104/do-filters-work-on-a-per-sample-basis |
Question: <p>I am learning about FIR filters and I'm confused.
I am trying to find out different types of FIR filters.</p>
<ol>
<li>Is direct form and n-tap FIR filter the same?</li>
<li>What does transposed FIR filter do?</li>
</ol>
Answer: <p>A finite impulse response (FIR) digital filter implements the following c... | https://dsp.stackexchange.com/questions/15412/fir-filters-direct-form-transposed-fir |
Question: <p>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 blocks, convolve them with the filter and output only the middle part (to avoid discontinuities)?</p>
<p>... | https://dsp.stackexchange.com/questions/43488/buffering-block-and-filter-length-in-real-time-processing |
Question: <p>The transfer function of a Low pass filter is H(w). From this I want to develop a high Pass filter. I read in <a href="http://books.google.co.in/books/about/Digital_Signal_Processing_Principles_Alg.html?id=CTw6GoBh-vkC" rel="nofollow">DSP by Proakis, Sec 4.5</a>, that the high pass filter can be obtained b... | https://dsp.stackexchange.com/questions/8279/frequency-response-of-low-pass-filter-and-high-pass-filter |
Question: <p>Practically speaking, if one is interested in a frequency band well-separated from the line noise (say, for example, the 10-20 Hz band, with 60 Hz line noise), would it be advisable to notch out this line noise before bandpass filtering to the desired frequency range? It seems that, given of course an acce... | https://dsp.stackexchange.com/questions/22451/notch-filtering-line-noise-outside-of-frequency-band-of-interest |
Question: <p>Can you please explain in simple terms what do the input parameters indicate in the <a href="http://www.mathworks.com/help/images/ref/ordfilt2.html" rel="nofollow"><code>ordfilt2</code></a> function in matlab?</p>
<pre><code>B=ordfilt2(A,Order,Domain)
</code></pre>
<p>I have seen people use this function... | https://dsp.stackexchange.com/questions/12582/what-exactly-does-ordfilt2-do |
Question: <p>I'm currently working in a project in which I use the coefficients of a IIR Elliptical digital filter (9th order) and the voice signal from a recording of my cellphone but I'm having issues finding the best Q format to work its implemenation, any suggestions? Till now I've been analysing which are the maxi... | https://dsp.stackexchange.com/questions/66815/how-to-find-the-best-q-format-representation |
Question: <p>I'm trying to build a dataset where one of the features is a signal which has originally been sampled at 500 Hz, while another feature is a signal which was sampled at 100 Hz. I want to downsample both of them at 10 Hz and then align them (they start at different times).</p>
<p>What should I do after low-p... | https://dsp.stackexchange.com/questions/76732/whats-the-correct-way-to-align-two-signals-downsampled-at-10-hz |
Question: <p>I have two signals and want to multiply them, e.g. voltage and current are multiplied to get power. The result (e.g. power) shall be a filtered result, e.g. to see the average.</p>
<p>Is it better to filter (digitally) each signal separately before the multiplication or to filter after the multiplication... | https://dsp.stackexchange.com/questions/59425/filter-before-or-after-multiplication-of-two-signals |
Question: <p><a href="http://homepages.inf.ed.ac.uk/rbf/HIPR2/filtops.htm" rel="nofollow">This link</a> contains the following statement:</p>
<blockquote>
<p>In contrast to the frequency domain, it is possible to implement non-linear filters in the spatial domain. In this case, the summations in the convolution func... | https://dsp.stackexchange.com/questions/8995/is-it-not-possible-to-perform-non-linear-filtering-in-frequency-domain |
Question: <p>I have barometer noisy data with known variance.<br />
I studied Kalman filter but I did not find an answer to this problem:<br />
My process model is: altitude is changed because of velocity that is changed because of acceleration that is normally distributed.</p>
<p><span class="math-container">$$ s[k+1]... | https://dsp.stackexchange.com/questions/48911/how-to-use-kalman-filter-for-altitude-prediction-based-on-barometer-data |
Question: <p>I was reading the answer to <a href="https://dsp.stackexchange.com/questions/46671/decimation-and-filtering-in-the-frequency-domain">this question</a> provided by Phil Karn.</p>
<p>In the answer, it has been said:</p>
<blockquote>
<p>Ensure that the impulse response of your lowpass filter is shifted to the... | https://dsp.stackexchange.com/questions/88074/what-is-the-necessity-of-shifting-the-impulse-response-of-a-filter-to-the-front |
Question: <p>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?</p>
<pre><code>clear all;
close all;
I = double(imread('Put your path here'))/255;
I=rgb2gray(I);
sigma=1;
[M,N,s]=size(I);
f1=-fix(M/2):ceil(M/2)-1;
f2=-fix(N/2):ceil... | https://dsp.stackexchange.com/questions/44098/how-to-perform-a-gaussian-blur-using-fft |
Question: <p>A low-passed signal, bandwidth limited to 4KHz is originally sampled 10 KHz. If I want to resample it at 20 KHz, I take these steps. Are these correct? Am I missing an step?</p>
<ol>
<li><p>First we need to filter the signal. Since the bandwidth is limited to 4KHz, we need a LPF with cutoff freq. of 4KHz.... | https://dsp.stackexchange.com/questions/89492/upsampling-of-a-signal |
Question: <p>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 not going well. All i know is that zero phase filtering is filtering that do not produce any phase delay and... | https://dsp.stackexchange.com/questions/54047/what-is-zero-phase-filtering-and-forward-backward-filtering |
Question: <p>First of all thanks for your patience: it is the first time for me posting a question in this forum. I am not a DSP expert, but I should get by if you give me in depth explanation.</p>
<p>This is an example of my raw data </p>
<p><img src="https://i.sstatic.net/lqtUx.png" alt="enter image description her... | https://dsp.stackexchange.com/questions/10513/constraint-on-a-filter |
Question: <p>Investigation of the method of suppression of random noise by coherent signal accumulation</p>
<p>Purpose - to identify opportunities for the coherent accumulation for cases of stationary and quasi-stationary signal.
Suppose that the input is the mixture of observed signal and a random white noise (i.e. n... | https://dsp.stackexchange.com/questions/10995/can-somebody-help-me-solve-this-signal-accumulation-problem |
Question: <p>Given the signal shown below, what is the best way to remove the steps and local maximas it contains. The signal contains some steps which can last up to 100 Samples before they return to about the same value as before the step (Marked with red circles). There are also some peaks which last only for one sa... | https://dsp.stackexchange.com/questions/12672/detecting-removing-steps-in-signal |
Question: <p>I'm reading a material where it says that a filter mask or kernel can be separable if the matrix of the filter mask has a rank 1. The two slides which describes this are as below:<img src="https://i.sstatic.net/vAIiA.png" alt="enter image description here"></p>
<p><img src="https://i.sstatic.net/Lv2AJ.png... | https://dsp.stackexchange.com/questions/15850/role-of-the-rank-of-the-filter-mask-matrix-in-image-processing |
Question: <p>I've made a simple first order IIR highpass filter with a zero at z = 1 and a pole at z = 0.9. Its frequency response looks like this:</p>
<p><img src="https://i.sstatic.net/bDtTx.png" alt="enter image description here" /></p>
<p>Now, I filter a DC signal using this filter. Here's the MATLAB code I use to ... | https://dsp.stackexchange.com/questions/16869/removing-transients-in-highpass-filtering-with-matlab |
Question: <p>Consider the following black-and-white image. It depicts a freehand sketch.</p>
<p><img src="https://i.sstatic.net/TMV3q.png" alt="enter image description here"></p>
<p>I wish to characterize the "density" of sketch strokes. For e.g. the hair strokes are densely grouped together. So are strokes near the ... | https://dsp.stackexchange.com/questions/24295/texture-like-measures-for-quantifying-density-of-data-in-binary-images |
Question: <p>I generate random spike data that represent the ouput from a rotary encoder. Here the output of my algorithm:
<a href="https://i.sstatic.net/lJIYf.jpg" rel="nofollow noreferrer"><img src="https://i.sstatic.net/lJIYf.jpg" alt="enter image description here"></a>
The first plot is position, then velocity and ... | https://dsp.stackexchange.com/questions/26248/derive-velocity-and-acceleration-from-position |
Question: <p>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. </p>
<p>Here's a picture of the transition itself:<br>
<a href="https://i.sstatic.net/KrqYc.png" rel="nofo... | https://dsp.stackexchange.com/questions/33805/how-do-i-smooth-over-transition-from-a-time-series-generated-by-one-system-to-a |
Question: <p>I have some data from a position encoder, so naturally i want to estimate its speed. However, the data is very quantized, so it's difficult to smooth enough to differentiate easily: </p>
<p><img src="https://i.sstatic.net/eMRcm.png" alt="data"></p>
<p>Each step level is about 70-140 data points long on ... | https://dsp.stackexchange.com/questions/33821/smoothing-a-staircase |
Question: <p>I want to compare the performance of a Wiener Filter and the Kalman filer to estimate the value of a constant $d$ using mesurements corrupted by a white noise. That is, my measurements are of the form
$$x(n) = d + v(n)$$
where $v(n)$ have a normal distribution with mean $0$ and known variance $\sigma^2$. <... | https://dsp.stackexchange.com/questions/34707/wiener-filter-to-estimate-constant-signal |
Question: <p>I am planning the following.</p>
<p>First sample the 20 MHz WiFi channel (WiFi channel-1 in figure).</p>
<p>Put band pass filters (5 MHz wide) around each of the ZigBee center frequencies (11, 12, 13, 14).</p>
<p>Re-sample the chunks to 4 MHz.</p>
<p>Is there anything wrong with this approach ?</p>
<p... | https://dsp.stackexchange.com/questions/37016/extracting-narrow-band-zigbee-signals4-mhz-from-a-wide-band-wifi-signal20-mhz |
Question: <p>If I apply a digital Bessel filter to a perfect step function, I get something that looks like the following:</p>
<p><a href="https://i.sstatic.net/iMDYj.png" rel="nofollow noreferrer"><img src="https://i.sstatic.net/iMDYj.png" alt="enter image description here"></a></p>
<p>The green line is the input st... | https://dsp.stackexchange.com/questions/37301/analytical-expression-for-step-response-of-digital-bessel-filter |
Question: <p>I am rather new to the world of signal processing, and am struggling to understand a fundamental concept: How are filters actually implemented?</p>
<p>I have read a significant portion of <a href="http://www.dspguide.com/" rel="nofollow noreferrer" title="DSP for Scientists and Engineers">this online book... | https://dsp.stackexchange.com/questions/43143/filter-implementation |
Question: <p>I am processing the signal from MPU6050. Applying FIR filter in the frequency domain and then taking IFFT to get filtered signal but getting some spikes in the output signal. Searched a lot about it and found here exit spectral leakage. Found some solutions like zero paddings and windowing but nothing is w... | https://dsp.stackexchange.com/questions/59678/real-time-fft-ifft-with-low-pass-filter |
Question: <p>Apologies in advance for asking something so basic, this is not my field at all so I'm a bit lost where else to find this info. I'm trying to understand this piece of code that detects sudden drops in a signal, it goes something like this:</p>
<pre><code>filter = [ones(1,10)*-0.05 ones(1,10)*0.05];
filter... | https://dsp.stackexchange.com/questions/75095/help-identifying-filter |
Question: <p>I'm filtering a raw PPG signal sampled at 100sps</p>
<p>After applying a bandpass filter my signal looks like this
<a href="https://i.sstatic.net/C25nN.png" rel="nofollow noreferrer"><img src="https://i.sstatic.net/C25nN.png" alt="PPG with trend" /></a></p>
<p>As you can see, It has a trend (a low pass com... | https://dsp.stackexchange.com/questions/76067/detrending-ppg-signal |
Question: <p>Is it possible, and how could it be done, to make an extension of filters that converts the signal in the filter's stop-band into noise with no information content or at least with zero correlation to the input signal in that frequency range, with only an arbitrarily small degradation of the signal in the ... | https://dsp.stackexchange.com/questions/77018/a-filter-removing-forbidden-information-in-the-stop-band |
Question: <p>Update: the yellow area in the graph below can be ignored, it shows power produced. I'm only interested in the blue line, and how to separate power consumed by the heating system from the rest.</p>
<p>I have data for a signal (blue line) showing overall power consumption. The spikes come from a heating sys... | https://dsp.stackexchange.com/questions/85536/method-for-splitting-time-sampled-signal-into-two-signals |
Question: <p>I know overlap save and overlap add are used for long data sequence filtering. Are there any other similar or better techniques like these? </p>
Answer: <p>The main alternative that I can think of is the hybrid method proposed by <a href="http://alumni.media.mit.edu/~billg/projects.html#conv" rel="nofollo... | https://dsp.stackexchange.com/questions/8771/other-techniques-like-overlap-save-overlap-add |
Question: <p>I need to know what is iteration and divergence in anisotropic diffusion filter technique.</p>
<blockquote>
<ul>
<li><p><strong>Isotropic diffusion</strong> $$\frac{\partial I(x, y, z)}{\partial t}={\rm div}\left[c\cdot \nabla I\left(x, y, z\right)\right], \quad
\text{where } c \text{ is the diffusion... | https://dsp.stackexchange.com/questions/14658/anisotropic-diffusion-filter-intuition-behind-parameters |
Question: <p>Let's say I have a 2 second data set taken at 220Hz sample rate and I would like to filter out the frequency bands associated with the EEG Spectrum:
$$\begin{align}
\Delta:& [1,3]\text{ Hz}\\
\theta:& [4,7]\text{ Hz}\\
\alpha_1:& [8,9]\text{ Hz}\\
\alpha_2:& [10,12]\text{ Hz}\\
\beta_1:&... | https://dsp.stackexchange.com/questions/28965/filtering-frequency-bands-out-of-a-signal |
Question: <p>Let`s say I have a signal </p>
<p>$m(t)=\cos(4\pi t) + \cos(6\pi t)$</p>
<p>so we can say the signal is containing frequencies $f_1= 2\mathrm{Hz}$ and $f_2= 3\mathrm{Hz}$. The cut-off frequency of the low pass filter is equal to $f_c= 3.5\mathrm{Hz}$, and the sampling frequency is $5\mathrm{Hz}$.</p>
<p... | https://dsp.stackexchange.com/questions/29880/a-question-about-sampling-theory |
Question: <p>I have a question about using the Dirichlet kernel as a filter. Let us suppose that I have samples of a continuous function sampled with frequency <span class="math-container">$F_s=10 \,\texttt{Hz}$</span>. The function is band-limited and the sampling frequency is well above the Nyquist frequency, but the... | https://dsp.stackexchange.com/questions/93959/using-the-dirichlet-kernel-as-a-lowpass-filter |
Question: <p>I can't find this answer anywhere. I have a couple satellite modem manuals and they refer to digital filtering functions that they do, but they say almost nothing about their sample rate. I always thought, without considering it too much, that all the modems I've worked with were only sampling at a rate ... | https://dsp.stackexchange.com/questions/93664/sample-rate-of-digital-modems-how-do-they-do-digital-filtering-if-sampling-belo |
Question: <p>I'm trying to convolve an input signal <span class="math-container">$x[n]$</span> with two FIR filters <span class="math-container">$h_1[n]$</span> and <span class="math-container">$h_2[n]$</span> in sequence, using block-based overlap-save FFT processing. I want my output to be:
<span class="math-containe... | https://dsp.stackexchange.com/questions/96583/how-to-apply-two-fir-filters-in-sequence-with-a-single-overlap-save-fft-step |
Question: <p>I would like to filter some data in an online sense i.e.</p>
<p><span class="math-container">$$y(t) = a0 + a1*y(t-1) + a2*y(t-2) + ... $$</span></p>
<p>the order not important.</p>
<p>My understanding of the SG is that it is really a smoother - I have to take some point and use points around it, in this ca... | https://dsp.stackexchange.com/questions/83038/savitzky-golay-filtering-not-smoothing-in-real-time |
Question: <p>Has anyone attempted to use savitzky-golay filters in conjunction with interpolation of missing observations? It seems very logical to do so, but I was wondering if there are any good reasons for not doing it since my search has yielded no results.</p>
Answer: <p>Savitzky-Golay filters, based on polynomia... | https://dsp.stackexchange.com/questions/95291/savitzky-golay-with-missing-observations |
Question: <p>On <a href="http://www.cs.toronto.edu/%7Efidler/slides/2019/CSC420/lecture2.pdf" rel="nofollow noreferrer">page 22 of the slide</a>, the 1D array is [1, 1, 1, 1, 1], but the dots are uneven. Then why their values are the same?
What's the value of the point that the arrow is pointing at? Though it gives Mov... | https://dsp.stackexchange.com/questions/69055/how-does-filter-works-in-noise-reduction |
Question: <p>I'm working on a dsPIC33, using Audio Codec Board - Proto to read audio samples and for reproduction.</p>
<p>In order to implement a relatively simple signal processing algorithm, I'm reading audio samples and I'm supposed to process them and pass them to DAC for reproduction.</p>
<p>The dsPIC33 I'm using ... | https://dsp.stackexchange.com/questions/72661/dsp-signal-filtering-and-number-formats |
Question: <p>I'm trying to do some simple filtering for an audio signal using a window-sinc low pass filter. Supposing that my buffer has M values and the filter kernel size is N, after convolving these two arrays I would obtain an output of m+n-1 values. Which part should I take into consideration? I'm asking this bec... | https://dsp.stackexchange.com/questions/8380/how-to-apply-convolution-on-a-buffer |
Question: <p>The question says it all. In typical (wavelet-like) decomposition of a signal, why is only the low pass component chosen for successive decomposition ?</p>
Answer: <p>Wavelets decomposition separates out the details/fluctuations/high-pass information from the image or signal. At each step details are sepa... | https://dsp.stackexchange.com/questions/18850/why-is-successive-decomposition-of-a-signal-performed-only-on-low-pass-component |
Question: <p>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 the filtering. So the requirement here is, after mixing I have to compensate for amplitude loss, i.e. the o... | https://dsp.stackexchange.com/questions/23419/amplitude-compensation-after-filtering-and-mixing-audio |
Question: <p>Why median filter is considered as good for removal of salt and pepper noise? What are the other filters used for the same?</p>
Answer: <p>Median filter is considered good because unlike averaging filter which ruins the edges of an image by blurring it to remove the noise, median filter removes only the n... | https://dsp.stackexchange.com/questions/27147/median-filter-for-salt-and-pepper-noise-removal |
Question: <p>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 a pulse shaping filter (RRC or RC).</p>
<p>As far as I know RC/RRC filters have a real impulse response, t... | https://dsp.stackexchange.com/questions/28087/pulse-shaping-and-baseband-filtering |
Question: <p>I have an input sequence $x=\{x_1, x_2 , ... x_n\}$ of reals, where $n=2^m$ for some $m$. I wish to calculate FFT of $x$.</p>
<p>$X=FFT(x)$</p>
<p>However, before I calculate the FFT, the signal $x$ gets corrupted with noise $\eta$, so $\hat{x}=x+\eta$, and calculated FFT, $\hat{X}$ is FFT of $\hat{x}$,... | https://dsp.stackexchange.com/questions/28230/filtering-noise-from-fft-where-noise-is-known-to-be-shifted-delta-function |
Question: <p>I am trying to understand what filter may be suitable for the following HMM:</p>
<p>The signal is a Wright-Fisher one-dimensional diffusion characterised by the SDE</p>
<p>$$dX_{t}=\frac{1}{2}\left(\alpha(1-x)-\beta x\right)dt+\sqrt{X_{t}(1-X_{t})}dB_{t}$$</p>
<p>with unknown parameters $\alpha$ and $\b... | https://dsp.stackexchange.com/questions/41578/filtering-for-wright-fisher-hmm |
Question: <p>I have a 192kHz IQ signal from an RF receiver, and i'm trying to remove signals in the negative (or positive) frequency spectrum.</p>
<p>I see that the negative frequency signals are -90 degrees phase shifted from I, where positive frequencies have the usual +90 degrees shift I vs Q.</p>
<p>Here's the pr... | https://dsp.stackexchange.com/questions/46301/how-to-eliminate-negative-frequencies-from-iq-signal |
Question: <p>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 are as follows:</p>
<p>\begin{equation}
y(n) = x(n) + \sum_{D=1}^kA(n)x(n-D)+v(n)\;\;\;\;\; (1)
\end{equatio... | https://dsp.stackexchange.com/questions/49105/implementing-fir-iir-alike-filter-in-r |
Question: <p>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-direction, and a 1D gaussian in z-direction).</p>
<p>Here is my MATLAB implementation:</p>
<pre><code>[y... | https://dsp.stackexchange.com/questions/49155/how-to-filter-a-sequence-of-frames |
Question: <p>Reading the ARMA model for the first time, and I'm confused.</p>
<p>Let's say I have a time series</p>
<pre><code>x = [1, 2.1, 2.9, 3, 4.1]
</code></pre>
<p>According to the ARMA model, <span class="math-container">$X_t$</span> is a linear combination of previous values and errors, something like</p>
<... | https://dsp.stackexchange.com/questions/52563/arma-ma-methods-how-do-you-know-the-error-terms |
Question: <p>Given a input sequence <span class="math-container">$x = {x_1, ... x_n}$</span> and a filter with l elements <span class="math-container">$h = [h_1, ..., h_l]$</span> with l < n. We want to filter the input sequence with the specified filter. </p>
<p>My first thought was to compute the convolution betw... | https://dsp.stackexchange.com/questions/55106/applying-a-filter-on-an-input-sequence-in-matlab |
Question: <p>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. </p>
<p>I've tried:</p>
<ul>
<li>Various filters </li>
<li>Audacity's native Noise Reducer</li>
</ul>
<p>and yet no l... | https://dsp.stackexchange.com/questions/55828/noise-reduction-how-can-i-filter-out-saw-or-other-tool-noises |
Question: <p>Given some difference equation for a filter like </p>
<p><span class="math-container">$y[k] = ax[k] + bx[k-1]+ cy[k-1]$</span>, </p>
<p>how would you initialize it? Since it needs an old output value (feedback) to calculate the new output value, it seems like the equation will keep on chasing its own tai... | https://dsp.stackexchange.com/questions/56084/what-is-used-to-initialise-difference-equations-that-require-data-from-the-outpu |
Question: <p>I am trying to implement basic homomorphic filtering but I can't seem to understand what happens in the frequency domain when you take the natural logarithm of the function.</p>
<p>In Matlab I'm getting not a number or infinity everywhere. </p>
Answer: <p>Plus is the simplest operation : <span class="mat... | https://dsp.stackexchange.com/questions/58781/how-does-the-frequency-spectrum-of-a-signal-change-in-the-logarithmic-space |
Question: <p>I have implemented FIR filter using tapped delay line method. I start getting output as soon as first input sample is passed to it, I am wondering from which sample I will get the proper output(without transient outputs) from filter.</p>
<p>Does it depend on location of highest magnitude tap or it's just ... | https://dsp.stackexchange.com/questions/62337/meaningful-output-of-fir-filter-output |
Question: <p>I am going through digital filter design. But I really get confused with so many </p>
<ol>
<li><p><strong>Digital filters</strong> : </p>
<ul>
<li>IIR </li>
<li>FIR </li>
<li>Moving Average</li>
<li>Linear phase filter</li>
<li>Allpass filter</li>
<li>Comb filter </li>
</ul></li>
<li><p><strong>Analo... | https://dsp.stackexchange.com/questions/42146/digital-filter-selection |
Question: <p>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 proper type of filters (like Chebyshev, Butterworth filter, etc)?</p>
Answer: <p>The easiest approach would ... | https://dsp.stackexchange.com/questions/74874/digital-filters |
Question: <p>I am having a LM35 sensor which gives $10\textrm{ mV}$ signal per degree rise of temperature. Now I have one question, which digital filter will be the best for this ?</p>
<p>This sensor will be use in very noisy industrial environment. Also digital filter have some cutoff frequency what should be the cut... | https://dsp.stackexchange.com/questions/42230/digital-filter-selection-for-sensor |
Question: <p>I am a beginner to study about the filter notion and property</p>
<p>Being a real digital filter,
(here "real filter" I means that its impulse response is real-valued)</p>
<p>this formula is established.
But I have no idea how to prove it</p>
<p>$$ |H(\pi +w )|=|H(\pi -w)|$$</p>
<p>What should be $H(... | https://dsp.stackexchange.com/questions/29659/real-digital-filter-property |
Question: <p>Would upscaling from a DVD to 4k on a TV be considered Digital Filtering?
Would an AI trained to suppress noise and enhanced desired signal be considered a Digital Filter?</p>
<p>I'm trying to figure out what counts as Digital filtering and what does not.</p>
<blockquote>
<p>In signal processing, a digital... | https://dsp.stackexchange.com/questions/82575/what-is-considered-a-digital-filter |
Question: <p>I am given a notch digital filter with the <span class="math-container">$z$</span>-transform being:
<span class="math-container">$$W(z)=MF(z)F(z^{*})^{*}=M\frac{z-q}{z-p}\frac{z-q^{*}}{z-p^{*}}$$</span>
where <span class="math-container">$M$</span> is the normalisation factor, <span class="math-container">... | https://dsp.stackexchange.com/questions/87124/notch-digital-filter |
Question: <p>How to subtract a digital filter from another one, if their lengths are different. How to make the length equal.</p>
Answer: <p>If (and only if) the filters (either FIR or IIR) are linear, then you can subtract their coefficients term-wise, supposing they are correctly aligned (below, on the $0$ index), t... | https://dsp.stackexchange.com/questions/43331/subtract-a-digital-filter-from-another-filter |
Question: <p>First of all, I'm new to DSP so excuse my simplified words.
I'm testing the performance of a digital filter on a (partly) noncontinuous signal:</p>
<p><img src="https://i.sstatic.net/X5ECG.png" alt=""></p>
<p>As you can see, the signal is not continuous at some points (like it is stopping and starting ov... | https://dsp.stackexchange.com/questions/9701/digital-filter-performance-with-noncontinuous-signal |
Question: <p>I have following digital signal which has been retrieved via sampling of an analog signal with sampling period <span class="math-container">$T_s = 100\,\mu\mathrm{s}$</span></p>
<p><a href="https://i.sstatic.net/JfBcY72C.png" rel="nofollow noreferrer"><img src="https://i.sstatic.net/JfBcY72C.png" alt="ente... | https://dsp.stackexchange.com/questions/95524/digital-filter-output-understanding |
Question: <p>I was just studying an old circuit analysis textbook that was describing how to design a Butterworth filter, and that seemed easy enough.. then, I started to wonder if I can take this analog filter and convert it into a digital filter. Its not really an exercise in the textbook, i was just curious how to c... | https://dsp.stackexchange.com/questions/63538/converting-a-simple-analog-butterworth-filter-into-a-digital-filter |
Question: <p>I'd like to find some book or books to get information about how digital filter is built depending on specifications. Like depending on price, or speed and etc. As I know it could be built by some D flip-flops and summation blocks, but there are more kind of them depending on specs, so if it's possible to ... | https://dsp.stackexchange.com/questions/70010/digital-filter-as-physical-device |
Question: <p>Having only dealt with digital filter scarcely, the question dawned to me when I used the firls function in matlab to design an equalizer with a certain gain response. </p>
<p>In general, can we prescribe an arbitrary shape as the equalizer filter response and hope to obtain a FIR/IIR filter that matches ... | https://dsp.stackexchange.com/questions/42816/digital-filter-design-accuracy |
Question: <p>Why is it that we Z-transform a difference equation to get a the transfer function of an digital filter?</p>
<p>How come a digital filter is given in the Z-domain, and what is the Z-domain?</p>
<p>And for that sake, why do analog filters operate in the S-domain, and what is the S-domain?</p>
Answer: <p>... | https://dsp.stackexchange.com/questions/13612/digital-filters-and-the-z-transform |
Question: <p>I'm new to digital filters. So I'm trying to get things right and I can't find an explicit answer to my question on the internet.</p>
<p><strong><em>Question: Digital filters only accept samples as input?</em></strong> I mean the input can not be zeros nor ones. It can only be samples. If that's the case,... | https://dsp.stackexchange.com/questions/30851/digital-filters-deal-only-with-samples-right |
Question: <p><em>To introduce my situation:</em> I'm developing a digital synthesizer in a form of a C++ library, working with low level APIs like WASAPI, ASIO, ALSA etc. It's probably not very practical and I'm mostly "reinventing the wheel" but my intention is to learn about digital synthesis in depth. So far I have ... | https://dsp.stackexchange.com/questions/51755/how-does-a-digital-filter-work |
Question: <p>I am designing a High pass Digital filter. I calculated the filter coefficients using fdatool and got some negative filter values.</p>
<p>Now, I need to convert them into values from 0 to 255.</p>
<p>In case of low pass filter , I used </p>
<pre><code>value = value*255/sum(value)
</code></pre>
<p>I ha... | https://dsp.stackexchange.com/questions/1427/normalizing-negative-filter-coefficients-for-digital-filter-design |
Question: <ul>
<li>What tools to use for practicing elementary filter design?</li>
<li>Is MATLAB all there is?</li>
<li>Do I need some specific toolboxes?</li>
<li>What functions do I need?</li>
</ul>
<p>I'm starting from the ground up in digital filter design and I thought that I need to decide on a program that lets ... | https://dsp.stackexchange.com/questions/31153/basic-tools-for-digital-filter-design |
Question: <p>i have the following question:
in digital filter design what's the difference between the methods of transformations :
bilinear vs impulse invariance vs Euler vs step invariance.
thank you!</p>
<hr />
<p><a href="https://i.sstatic.net/FwRTY.png" rel="nofollow noreferrer"><img src="https://i.sstatic.net/FwR... | https://dsp.stackexchange.com/questions/86536/transformation-methods-for-digital-filters |
Question: <p>I am getting conflicting advise regarding how to clean my EEG data:</p>
<p>1) Manually remove artefacts first and then apply digital filters</p>
<p>OR</p>
<p>2) Apply digital filters first and then manually remove artefacts</p>
<p>The reason given for 1) is because artefacts are more visible and avoids... | https://dsp.stackexchange.com/questions/52050/manual-filter-or-digital-filter-first-for-eeg |
Question: <p>I have 2 simultaneous signals that both are designed to measure eye movements.
They are sampled at 250 Hz.
We have 12 subject recordings.
For 3 stable periods in each subject, we choose 256 points and did an FFT.
Prior to the FFT, the data were mean-centered and detrended with a 2nd
order polynomial. They... | https://dsp.stackexchange.com/questions/60098/reverse-engineering-a-digital-filter |
Question: <p>I want to design a digital filter for pulse shaping. Pulses are of 100us Fall time. and the sampling rate is 100MegaSamples/sec. and the Shaping time is 5us. What should my coefficients be??? And how to obtain them using matlab or any other related software.</p>
Answer: | https://dsp.stackexchange.com/questions/2760/finding-the-coefficients-of-the-digital-filter |
Question: <p>By negative frequency, I refer to Fourier transform. Often, the frequency response of a digital filter is only displayed for positive frequencies. For a linear IIR digital filter, what happens for negative frequencies? Are frequency response for negative frequencies mirror images of what happens for positi... | https://dsp.stackexchange.com/questions/24386/for-linear-iir-digital-filter-what-happens-for-negative-frequencies |
Question: <p>I am reading a chapter on digital filter design from analog filter design using difference equations. What they do first of all is that they map <span class="math-container">$s$</span> (Laplace variable) to <span class="math-container">$z$</span> (<span class="math-container">$z$</span>-transform) by the f... | https://dsp.stackexchange.com/questions/53218/digital-filter-design-using-difference-equations |
Question: <p>Of the four classic analog filter types: Butterworth, Chebyshev, Elliptic and Bessel- are any of these relegated to obsolescence for purposes of digital filter design in comparison to optimized algorithms such as least squares (<code>firls</code>), Parks-McClellan (<code>firpm</code> or <code>remez</code>)... | https://dsp.stackexchange.com/questions/79400/mapping-of-classic-filters-for-digital-filter-design |
Question: <p>I want learn digital filter design. My knowledge of math is at high school level. I can learn math through the Internet. Then, what fields of math do I have to learn? </p>
Answer: <p>If you have the balls to learn math by yourself. The two fields of Mathematics that you need to dominate in order to do fil... | https://dsp.stackexchange.com/questions/26582/fields-of-math-needed-for-digital-filter-design |
Question: <p>I have a system that performs wireless sampling of data about every 7.5ms (133Hz). Due to it being wireless, I get occasional data drop out. I want to construct a LP butter filter with cut-off frequency of 10Hz using Python's scipy butter method and then downsample everything to a lower frequency. </p>
<p... | https://dsp.stackexchange.com/questions/40697/python-butter-filter-choosing-between-analog-and-digital-filter-types |
Question: <p>I am very new in signal processing and using digital filters. I have to use a low-pass filter to analyze my data in LabVIEW and have a question about it. Any help and advice is appreciated.</p>
<p>I am trying to simplify my problem here:</p>
<p>Let’s say there is a digital sine wave (made by LabVIEW) wit... | https://dsp.stackexchange.com/questions/37591/digital-filters-in-labview |
Question: <p>I am new to DSP and I am trying to find the cutoff frequency of a HP digital filter. I know the equation that describes the system, its frequency response: <span class="math-container">$H(e^{jω})= 1 - \frac{e^{jω}}{2} - \frac{1}{2e^{jω}}$</span> and of course amplitude response / phase diagrams.</p>
<p>I ... | https://dsp.stackexchange.com/questions/67670/calculating-the-cutoff-frequency-of-a-highpass-digital-filter |
Question: <p>I am new to the world of digital filters and am educating myself with the book <em>Introduction to Digital Filters by J.O Smith III,</em>. The author derives the frequency response of a very simplistic filter:</p>
<p><span class="math-container">$$y(n) = x(n) + x(n-1)$$</span>
<span class="math-container">... | https://dsp.stackexchange.com/questions/80795/issue-understanding-time-delay-of-a-digital-filter |
Question: <p>One of the known methods for discretizing analog filters is impulse response invariant. We get the impulse response in time domain, discretize it and then get the Z transform.</p>
<p>What I am trying to understand is why the freq response of the resulting digital filter has a freq response magnitude scaled... | https://dsp.stackexchange.com/questions/68903/impulse-invariant-method-for-digital-filter-design |
Question: <p>Beginning from the basic definition of decibel that expresses the ratio of two amplitudes as <span class="math-container">$20\log_{10}(A_{2}/A_{1}) $</span>, how do we arrive at the expression <span class="math-container">$-20\log_{10}(\delta_{s})$</span> measured in dB, for the stopband deviation of a dig... | https://dsp.stackexchange.com/questions/91648/expression-for-stopband-deviation-of-a-digital-filter |
Question: <p>In an ideal design, a digital filter has a target gain in the passband and a zero gain (−∞ dB) in the stopband. In a real implementation, a finite transition region between the passband and the stopband, which is known as the transition band, always exists. The gain of the filter in the transition band is ... | https://dsp.stackexchange.com/questions/46454/transition-bands-and-passband-gain-in-digital-filter-design |
Question: <p>I am strugling with a question that I hope someone can help me with.</p>
<p>I am recording single molecule events which I detect is picoampere square deflections.</p>
<p>I wish to use as gentle low-pass bessel filtering as possible.</p>
<p>The lowest filter settings my amplifier allow are 10 kHz and 100... | https://dsp.stackexchange.com/questions/55053/sample-rate-filtering-digital-filtering-and-aliasing |
Question: <p>I am making a 9th order digital bandpass filter with lower and upper corners of 200 kHz and 40 MHz respectively. I am using this filter to filter a 1D time domain signal which is 64k samples long sampled at a frequency of 150MHz. </p>
<p>I have done some digital filtering before in university, so I know ... | https://dsp.stackexchange.com/questions/18323/implementing-digital-filter-by-padding-with-zeros |
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