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Question: <p>Libraries like <a href="https://docs.scipy.org/doc/scipy/reference/signal.windows.html" rel="nofollow noreferrer">scipy</a> typically offer constructing window functions in a symmetric or asymmetric flavor. I'm aware of the rule of thumb:</p>
<ul>
<li>Use symmetric for filter analysis.</li>
<li>Use asymmet... | https://dsp.stackexchange.com/questions/95448/when-to-use-symmetric-vs-asymmetric-periodic-window-functions |
Question: <p>I have a signal (<span class="math-container">$S(t)$</span>) which is product of a Gaussian (<span class="math-container">$G(t)$</span>) and a random phase function (<span class="math-container">$e^{i\theta(t)}$</span>, here <span class="math-container">$\theta(t)$</span> is a random function), as shown be... | https://dsp.stackexchange.com/questions/54573/can-spectral-density-be-a-complex-quantity |
Question: <p>Original Question: <a href="https://ux.stackexchange.com/q/23040/16006">https://ux.stackexchange.com/q/23040/16006</a></p>
<p>I've only taken some basic signal analysis courses, so I might be missing some things.</p>
<p><strong>Purely theoretical question:</strong></p>
<p>What methods exist for represen... | https://dsp.stackexchange.com/questions/2744/what-is-the-best-way-to-represent-audio-visually-x-post-from-ux |
Question: <p>What is Spectral entropy and spectral moments? I know what the normal entropy of a signal is! And also what are some good time-frequency features for the analysis of non-stationary signals?</p>
Answer: | https://dsp.stackexchange.com/questions/49809/spectral-entropy-and-moments-and-non-stationary-signal-processing |
Question: <p>I am working on a library for generating LPC for speech synthesis. I am currently using a hamming window for the spectral analysis which goes in 200ms blocks over the signal, and does the a-to-k conversion.</p>
<p>I have read some stuff online about a technique of overlapping these windows so it would pr... | https://dsp.stackexchange.com/questions/25747/hamming-window-for-lpc |
Question: <p>I'm trying to do a spectral analysis in R. I learned it in Python from Allen Downey's ThinkDSP book.</p>
<p>What is the R equivalent of the Python numpy function, numpy.fft.fftfreq?</p>
<p>If you provide a window length and spacing, that function returns the frequency bin centers. I've been hunting throu... | https://dsp.stackexchange.com/questions/61029/fft-freqency-bin-center-in-r |
Question: <p>Hi guys i am doing a course in Digital Filters and Spectral analysis. We are given a coursework/homework, and I have absolutely no idea what to do with it. I come from Maths background, never done any signal processing before, and since I am new to the university I don't really have anyone to ask.</p>
<p>... | https://dsp.stackexchange.com/questions/54072/using-fast-fourier-transform-to-determine-musical-notes |
Question: <p>So, I've created a simple sound analysis application and one of the features I've implemented is the spectral centroid (as explained here <a href="https://en.wikipedia.org/wiki/Spectral_centroid" rel="nofollow noreferrer">https://en.wikipedia.org/wiki/Spectral_centroid</a>). In order to get reliable result... | https://dsp.stackexchange.com/questions/42452/spectral-centroid-manipulations |
Question: <p>I was reading the book "Spectral Analysis of Time Series" By Herman Koopmans.
On <a href="http://books.google.de/books?id=F09lhyXw4mcC&lpg=PP1&dq=herman%20koopman%20time%20series&pg=PA55#v=onepage&q=A%20Nonstationary%20Process%20with%20a%20Wiener%20Spectrum&f=false" rel="nofollow">Page ... | https://dsp.stackexchange.com/questions/17698/signal-plus-weakly-stationary-noise |
Question: <p>I have a couple of questions regarding windowed FFTs:</p>
<ol>
<li><p>Why is the noise floor higher with windowed FFTs (according to Wikipedia's spectral leakage page, anyway), when the whole point of windowing is to reduce side lobes?</p></li>
<li><p>I realize that different windows are better for differ... | https://dsp.stackexchange.com/questions/1357/spectrum-analysis-using-windowed-ffts |
Question: <p>What is the difference between doing a PSD estimate with data and the same data but which is logarithmically transformed before the estimate? Does it make the data more sinusoidal in nature?</p>
<p>An exercise in a book asks this question:</p>
<p>"For the lynx data, compare your spectral analysis results... | https://dsp.stackexchange.com/questions/60931/difference-of-doing-a-psd-estimate-of-data-and-logarithmic-transformed-data |
Question: <p>I have been told that for deterministic signals, it makes sense to look at their respective Fourier transforms/spectra. </p>
<p>For stochastic processes on the other hand, I am supposed to work with power spectral density in terms of qualitative analysis. </p>
<p>Why? </p>
Answer: <p>Because a stochasti... | https://dsp.stackexchange.com/questions/47740/why-look-at-power-spectral-density-for-stochastic-processes |
Question: <p>I´m stuck in a deduction analysis of the variance of a gaussian white noise signal in a "integrate-and-dump detector" of a baseband data transmission receiver, where <span class="math-container">$n(t)$</span> is white noise with double-sided power spectral density <span class="math-container">$N_... | https://dsp.stackexchange.com/questions/71395/white-gaussian-noise-analysis-deduction |
Question: <p>Anyone know what algorithm the Spice AC Noise Analysis uses?</p>
<p><a href="http://vision.lakeheadu.ca/eng4136/spice/noise_analysis.html" rel="nofollow noreferrer">http://vision.lakeheadu.ca/eng4136/spice/noise_analysis.html</a></p>
<p>Is it some spectral modeling synthesis? I.e. that it estimates the m... | https://dsp.stackexchange.com/questions/42488/anyone-know-what-algorithm-the-spice-ac-noise-analysis-uses |
Question: <p>My question is about the spectral resolution of a discrete signal. Each sample of my signal is made up with 2^n frames sampled at 44.1 kHz.</p>
<p>So, when I want to know the spectral resolution, I calculate : 44100/number_of_frames. With 2048 frames, my spectral resolution is around 20Hz. But, when I tak... | https://dsp.stackexchange.com/questions/49291/how-to-increase-the-spectral-resolution |
Question: <p>I have a behavioral model for a PLL that generates a chirp signal for an FMCW radar. To improve efficiency, the model outputs only the zero-crossing points at the negative edge.</p>
<p>From this zero-crossing data, I need to compute the RMS frequency error over a specific frequency range. However, since th... | https://dsp.stackexchange.com/questions/96225/fmcw-radar-signal-processing-fft-with-nonuniform-sampling-points |
Question: <p>I have a library of music tracks that I use to DJ with. Its currently about 3000 tracks that I have gather over the years. Some of it consists of low quality rips that I want to get rid of. </p>
<p>Currently I am writing a script that will look at the time/size = compression rate. Anything that is below 3... | https://dsp.stackexchange.com/questions/48027/analyzing-the-quality-of-a-music-track |
Question: <p>According to the book <em>Introduction to Spectral Analysis</em> by P. Stoica and R. Moses, the power spectral density (PSD) <span class="math-container">$P(\omega)$</span> can either be defined as the discrete-time Fourier transform (DTFT) of the covariance sequence <span class="math-container">$r(k)$</sp... | https://dsp.stackexchange.com/questions/55449/example-of-non-equivalence-of-the-two-psd-definitions |
Question: <p>All,</p>
<p>Are there any good continuing education courses of length 2-3 days to give an engineer a good background in Frequency Analysis. B&K used to teach one, but I don't think that they teach it anymore. I am looking for something that teaches sampling, aliasing, continuous vs. discrete signals... | https://dsp.stackexchange.com/questions/55853/good-continuing-education-course-in-the-basics-of-frequency-analysis |
Question: <p>What is the units of FFT, when doing Spectral Analysis of a Signal?</p>
<ol>
<li><p>For above question, the answer could be V or V/HZ for voltage signal. Which one is right? I would expect the result to be V.t or V/Hz because of dt.</p>
</li>
<li><p>I used <a href="https://www.mathworks.com/help/signal/ref... | https://dsp.stackexchange.com/questions/78188/units-of-a-fast-fourier-transform-fft-and-spectrogram |
Question: <p>I am performing a method of data analysis that requires estimating the CPSD between two measured signals. I actually have three signals, so I typically sum two of them and cross them with the third. Is there a analogous concept to the traditional cross power spectral density but for three signals at once? ... | https://dsp.stackexchange.com/questions/94802/cross-power-spectral-density-of-three-signals |
Question: <p>Here's my problem. The input signals $x$ and $y$ will be having the time value aligned with each other. However, the data are not evenly sampled. I would like to calculate CPSD of both signals.</p>
<p>The solution comes to my mind is as follow</p>
<ol>
<li>$R_{xy}$ = cross correlation of $x$ and $y$ (I'm... | https://dsp.stackexchange.com/questions/8825/cross-power-spectral-density-of-unevenly-sampled-data |
Question: <p>Do you know about Doctor Who and its screwdriver?</p>
<p>Well, I'm trying to understand how to replicate <a href="https://web.archive.org/web/20060112064455/http://www.bbc.co.uk/doctorwho/sounds/sonicscrewdriver.mp3" rel="nofollow noreferrer">this sound</a> but the spectral analysis is too way complicated... | https://dsp.stackexchange.com/questions/58961/analyze-and-reproduce-sonic-screwdriver-sound |
Question: <p>let us consider following code </p>
<pre><code>function [sca_1,sca_2,sca_3,sca_4]=calc_wavelet(y,wname,scales,freq,fs)
%y-input signal
%wname-wavelet basis name
%freq-test frequencies
%fs-sampling rate
TAB_Sca2Frq = scal2frq(scales,wname,1/fs);
[~,idxSca_1] = min(abs(TAB_Sca2Frq-freq(1)));
sca_1 = scales... | https://dsp.stackexchange.com/questions/15587/use-wavelet-for-improving-spectral-resolution |
Question: <p>I am extremely new to signal analysis. And before posting I did a lot of reading on signal analysis, FFT and windowing. I am working on my thesis which involves comparison of speech signals lets say about 100 speech samples for a given sentence. I have the recordings and all the data. I have a few question... | https://dsp.stackexchange.com/questions/16910/parameters-for-signal-analysis |
Question: <p>I am reading <a href="https://stm.sciencemag.org/content/10/431/eaap8674" rel="nofollow noreferrer">Smartphone based Blood Pressure Monitoring via the Oscillometric Finger Pressing Method</a>, which is trying to estimate blood pressure from a PPG sensor and a small applied force finger sensor. I am not an ... | https://dsp.stackexchange.com/questions/75243/estimating-average-hr-from-ppg-sensor |
Question: <p>I'm trying to compute the highest frequency (as can be sampled) in some pretty manky looking discrete time-dependent signals. My current method - a discrete fourier analysis - fails for some pretty awful looking but clearly oscillating signals (with discernable highest frequencies).</p>
<p>My current meth... | https://dsp.stackexchange.com/questions/35028/find-highest-frequency-of-a-very-manky-signal |
Question: <p>I have a fourier analysis signal as in the picture attached, where red represents the FFT of movement of the hand of a stroke subject and the blue one is the movement of a healthy subject.</p>
<p>I am doing some analysis called <strong>Spectral Arc Length</strong>, where I will calculate the spectral arc ... | https://dsp.stackexchange.com/questions/34101/effect-of-dc-component-on-the-whole-signal-comparison-between-normalised-and-n |
Question: <p>I was experimenting with sound analysis lately and from what I see when I plot spectral data of an audio file is that apart from notes that were actually picked there are some other notes with quite high local amplitude.<br>
For example I have a sample where D major chord is played with some nasty distorti... | https://dsp.stackexchange.com/questions/31151/spectral-plot-shows-more-notes-than-there-really-should-be |
Question: <p>The impulse response of an ideal low-pass filter can be determined by setting <span class="math-container">$H(\omega)=1$</span> in the Fourier-representation <span class="math-container">$$h(n) = \frac{1}{2\pi}\int_{-\omega_c}^{\omega_c} H(\omega)e^{j\omega n}d\omega$$</span></p>
<p>The solution will be a ... | https://dsp.stackexchange.com/questions/84934/deriving-the-impulse-response-of-an-ideal-low-pass-filter |
Question: <p>I'm sampling 8 bioelectric signals with a embedded board which use an 8-channels ADC (<a href="http://www.analog.com/media/en/technical-documentation/data-sheets/AD7175-8.pdf" rel="nofollow noreferrer">AD7175-8</a>). My sampling rate for every single channel is about 6250 Hz. When move the analysis in the ... | https://dsp.stackexchange.com/questions/51538/what-kind-of-signal-reppresents-this-power-spectral-density |
Question: <p>I would like to experiment with some input from a microphone and am receiving a 2 channel, 512 samples buffer in real time</p>
<p>I know the signal could be passed through a low pass filter, and then windowed before the FFT.</p>
<p>The sample rate is 44100.00, what low pass filter is needed for this, doe... | https://dsp.stackexchange.com/questions/14546/preparing-audio-data-for-fft |
Question: <p>This is my first dive in DSP. I would like to familiarize myself with frequency analysis. I have two audio tracks which should be digitized at 16bit-44.1kHz and 24bit-192kHz (music, presented as a 24bit-192kHz sample) respectively.</p>
<p>I wanted to identify the effect of the low-pass filter around the N... | https://dsp.stackexchange.com/questions/24635/how-to-interpret-these-different-fourier-analysis-of-this-audio-signal |
Question: <p>I'm looking for a Fourier analysis method that will help me with a servo position tracking problem. I'll give some background:</p>
<p>Imagine I have a control system that attempts to control a linear actuator to a submicron position. I have a following error signal in units of nanometers that I monitor. In... | https://dsp.stackexchange.com/questions/96070/what-fourier-analysis-would-be-appropriate-for-analyzing-servo-position-error-as |
Question: <p>I'm new to DSP. As I reading the textbook, I cannot understand the formula <span class="math-container">$X_{s}(f)=\frac{1}{T}\sum_{n=-\infty}^{\infty} X(f-nf_{s})$</span>. Could you please give me some keywords so I can learn the theorem and understand it?</p>
<blockquote>
<p>From spectral analysis, the or... | https://dsp.stackexchange.com/questions/86160/what-is-theorem-under-this-formula |
Question: <p>I have an audio signal and I would like to do spectral analysis/processing on it.
I am interested in having frequency bins that approach a BarkScale rather than being equidistant.</p>
<p>First, I should mention that I am an amateur in DSP, please take it into account when answering.</p>
<p>I did some resea... | https://dsp.stackexchange.com/questions/96208/how-to-get-fft-bins-of-an-audio-signal-to-approach-a-barkscale |
Question: <p>Perhaps someone can help me resolve something - this is my understanding:</p>
<p>In deterministic signal analysis, for a continuous signal <span class="math-container">$x(t)$</span> the <a href="https://en.wikipedia.org/wiki/Energy_(signal_processing)" rel="nofollow noreferrer">signal energy</a> is defined... | https://dsp.stackexchange.com/questions/65963/inconsistency-between-the-units-of-power-spectral-density-and-the-definition-tha |
Question: <p>I am using the following formula to calculate SNR of a real world complex baseband signal sampled at 1x Nyquist.</p>
<pre><code>SNR = Rxy(tm)^2 / [ Px*Py - Rxy(tm)^2 ]
SNR (dB) = 10*log10(SNR)
</code></pre>
<p>where</p>
<pre><code>Rxy(tm) = peak of the cross correlation at time delay, tm
Px = power i... | https://dsp.stackexchange.com/questions/69380/how-to-calculate-time-domain-snr-using-known-sequence |
Question: <p>What is the advantage of using a polyphase filter bank (PFB) for spectral analysis over just using the FFT? In the standard <a href="http://cnx.org/contents/3dea9cf9-32b6-4bf2-940b-cf8d251a0a84@15/Uniformally_Modulated_%28DFT%29_Fi" rel="nofollow">"critically sampled" uniform DFT filterbank</a>, the polyph... | https://dsp.stackexchange.com/questions/24166/purpose-of-using-polyphase-filter-bank-pfb |
Question: <p>I would like to know whether there is a condition in selecting the number of channels <span class="math-container">$M$</span> and filter length <span class="math-container">$N$</span>. As of now I am trying to design a channelizer where the data length <span class="math-container">$M$</span> is much less t... | https://dsp.stackexchange.com/questions/91992/filter-length-in-maximally-decimated-polyphase-channelizer |
Question: <p>I have several signals that consist of repetitive units. In the figure you'll clearly see the variability of the signals, that increases top down. The first signal is super repetitive and units are indicated with green lines.
In the third, you'll see peaks and in the middle a little insertion that I know c... | https://dsp.stackexchange.com/questions/22306/detect-repetitive-units-within-signals |
Question: <p>I am using an implementation of the averaged cyclic periodogram (section 3.2.4 in Antoni, Jérôme. "Cyclic spectral analysis in practice." Mechanical Systems and Signal Processing 21.2 (2007): 597-630.). The cyclic spectrum is estimated based on two <span class="math-container">$L$</span> ln
<span... | https://dsp.stackexchange.com/questions/83096/how-to-eliminate-a-cyclic-spectrum-estimation-window-artifact |
Question: <p>I want to calculate vibro-acoustic analysis of plate in program Ansys workbench Mechanical.</p>
<p>Tell me please how to convert Sound Pressure Level [dB] in Power Spectral Density [Watt/Hz] ?</p>
Answer: <p>Tricky.</p>
<p>Assuming a level of <span class="math-container">$L_{SPL}$</span> in dB and that yo... | https://dsp.stackexchange.com/questions/96486/how-to-convert-spl-db-in-psd-watt-hz |
Question: <p><a href="https://en.wikipedia.org/wiki/Autoregressive_integrated_moving_average" rel="nofollow">ARMA</a>
models are afaik just filters with transfer function
$ {MA(z) \over AR(z)} \equiv {FIR(z) \over IIR(z)} $ .<br>
However forecasters of stock prices, market trends ...<br>
seem to be mainly statisticia... | https://dsp.stackexchange.com/questions/23606/forecasting-with-arma-models-from-a-filter-point-of-view |
Question: <p>I have an auto-correlation function that was generated from a signal, and I am trying to extract its 'repetition rate' in order to calculate the dominant frequency of the pulse, but I am not exactly clear how to do this. </p>
<p>Here are two cases, labelled 'good' and 'bad' to mark best/worst case scenari... | https://dsp.stackexchange.com/questions/2119/extraction-of-non-sinusiodal-repetition-rates |
Question: <p>I am working on trying to apply a low and high pass filter to an audio file that contains a set of exhalations over a microphone. The inhalations have been cut out of the file, and the exhalations are stitched together in the file. I am attempting to replicate this <a href="https://jamanetwork.com/journals... | https://dsp.stackexchange.com/questions/76605/paper-replication-validating-the-proper-way-to-pass-wav-audio-breathing-data-t |
Question: <p>I am currently reading through <a href="https://www.cs.cmu.edu/~rbd/papers/dannenberg-goto-structure-2009.pdf" rel="nofollow noreferrer">Music Structure and Analysis from Acoustic Signals</a> and am having some difficulty in understanding how the modified Kullback-Leibler distance is calculated. (I am jus... | https://dsp.stackexchange.com/questions/37279/kullback-leibler-distance-of-spectral-data |
Question: <p>I'm currently working with physiological signals (PPG and GSR) for emotion recognition but, from my research, I've found out that almost everyone in that area use a PSD analysis over a FFT analysis. I've been reading about them and found out that PSD helps with giving a clearer view of the spectrum despite... | https://dsp.stackexchange.com/questions/70103/why-is-it-used-the-power-density-spectrum-psd-over-an-anylisis-with-the-fast-f |
Question: <p>This is my first ever question here so the help is really appreciated.</p>
<p>I am performing FFT on a signal. I want to perform windowing, 50% overlapping and averaging to the signal. There is a function <code>scipy.signal.welch</code> to perform this automatically but the output is in power spectral dens... | https://dsp.stackexchange.com/questions/85303/fft-of-signal-data-with-windowing-overlapping-and-averaging |
Question: <p>Let <span class="math-container">$x(t)$</span> be a periodic signal with period <span class="math-container">$T>0$</span>. Suppose we sample <span class="math-container">$x(t)$</span> with sample rate <span class="math-container">$f_s\in\mathbb N$</span> in the interval <span class="math-container">$[0,... | https://dsp.stackexchange.com/questions/96427/basic-questions-about-spectral-leakage |
Question: <p>I am doing a project on ECG arrythmia analysis using matlab.</p>
<ol>
<li><p>I have designed notch filter for removing 50 Hz noise but don't know how to add a 50 Hz powerline interference noise to a clean ECG signal? </p></li>
<li><p>Also, I want to check whether noise is reduced in the filtered signal. W... | https://dsp.stackexchange.com/questions/6103/adding-noise-to-an-ecg-signal |
Question: <p>In a lot of time-series analysis references I find (written by mathematicians or statisticians rather than engineers), I find the following signal decomposition for a stochastic process, termed the "Cramér representation" (e.g. eqn 8.11 of this <a href="https://www.stat.tamu.edu/%7Esuhasini/teach... | https://dsp.stackexchange.com/questions/68936/why-cram%c3%a9r-spectral-representation-and-not-dtft-for-stochastic-process |
Question: <p>I have been reading a paper on the <a href="https://ieeexplore.ieee.org/document/7251907" rel="nofollow noreferrer">"Single pass spectrogram inversion"</a> </p>
<p>and I came across this in the Introduction part.</p>
<blockquote>
<p>In many applications, the analysis and modification of the Short-Time
... | https://dsp.stackexchange.com/questions/58680/when-do-phases-not-exist-for-spectrograms |
Question: <p>I am working with a new Digital-to-Analog Converter (DAC) design in simulation and I'm trying to analyse the output. The device takes in an ideal 14-bit digital representation of a sine wave and outputs through an ideal Butterworth filter (a̶n̶t̶i̶-̶a̶l̶i̶a̶s̶i̶n̶g̶ anti-imaging).</p>
<p>In the simple ana... | https://dsp.stackexchange.com/questions/73872/analysing-dac-spectra-transient-noise-analysis |
Question: <p>I am look into CSPE. "<a href="http://jssunderlin.pbworks.com/f/13449.pdf" rel="nofollow noreferrer">Signal Analysis Using the Complex Spectral Phase Evolution (CSPE) Method</a>"</p>
<p>The method is simple. It compares the original signal's FFT and shifted signal FFT in phase domain so that it can get an... | https://dsp.stackexchange.com/questions/57905/complex-spectral-phase-evolution-cspe-performance-depending-on-signal-windowin |
Question: <p>I'm interested in papers which are about auto-correlations of periodic <strong>time series</strong> signals.All relevant papers and applications are interesting to me, as I am studying the properties of the auto-correlation of periodic, digital time signals.</p>
<p>The reason, I am asking you for this hel... | https://dsp.stackexchange.com/questions/43529/auto-correlation-of-time-signals |
Question: <p>We've been working on a machine learning project for pattern recognition, using time-domain features such as kurtosis, mean, standard deviation, variance, skewness, and peak-to-peak values.</p>
<p>Background:</p>
<p>Initially, we trained our data after applying a high-pass filter at 1 kHz. The results were... | https://dsp.stackexchange.com/questions/88988/issues-with-ml-pattern-recognition-after-bandpass-filtering |
Question: <p>The title might be unclear, but the problem is this. I have a signal sampled 1500 times with a rate of 60/s, and a sensor array 512 units large. There is a lot of noise, echo and other frequencies being picked up, but I am interested in only one. First I do a spike removal, then a bandpassfilter (butterwor... | https://dsp.stackexchange.com/questions/16432/finding-the-best-principle-component |
Question: <p>I am following the book The Intuitive Guide to Fourier Analysis & Spectral Estimation with MATLAB. I am trying to selflearn the fourier analysis in matlab.
I got lost in one passage in the demonstration that states that the FT of the ACF function is the square of the DTFT of the signal. I have attached... | https://dsp.stackexchange.com/questions/59380/the-demonstration-that-states-that-the-ft-of-the-acf-function-is-the-square-of-t |
Question: <p>My question is on the aliasing cancellation of the OLA method when spectral modification is involved. The book related to this question is given by <a href="https://ccrma.stanford.edu/%7Ejos/sasp/Constant_Overlap_Add_COLA_Cases.html" rel="nofollow noreferrer">this link</a>.</p>
<p>As stated by the webpage,... | https://dsp.stackexchange.com/questions/81687/strong-vs-weak-cola-constant-overlap-add |
Question: <p>I am referring to the work of Stephen A. Billings on "<a href="https://eprints.whiterose.ac.uk/87212/1/acse%20research%20report%2056.pdf" rel="nofollow noreferrer">Identification of a class of nonlinear systems using correlation analysis</a>" from the year 1978, where it is mentioned that the <sp... | https://dsp.stackexchange.com/questions/91730/i-textth-dimensional-autocorrelation-function |
Question: <p>I have a few questions regarding cepstral analysis, that the numerous articles and papers I've read on the topic didn't answer.</p>
<p><strong>What I understood:</strong> The cepstrum captures the periodicity of harmonics in a spectrum.</p>
<p><strong>My questions:</strong>
In articles treating about defau... | https://dsp.stackexchange.com/questions/89115/questions-on-cepstral-analysis |
Question: <p>I am building an application that would "listen" to the microphone input, analyse it, and compare the analysis to a pre-analysed and pre-classified sound bank (small - maximum 20 sounds). It will then show the user what sound it was.</p>
<p>Now, I have a vague idea on how to implement this. I would like t... | https://dsp.stackexchange.com/questions/21961/feature-extraction-for-sound-recognition-and-classification |
Question: <p>This is a cross posting from the crossvalidated stack exchange as I thought this may be a better forum to ask.</p>
<p>I have a dataset consisting of respiratory time series signals of different lengths obtained from different groups of patients. I want to either classify or cluster the patients using these... | https://dsp.stackexchange.com/questions/71917/help-with-denoising-signal-and-periodogram-analysis-resources |
Question: <p>I have data from a LIDAR unit that I would like to get the spectral density of. Unfortunately, the only thing I remember from my Fourier analysis class are the methods that I know will not work.</p>
<p>The data comes from a 1D LIDAR scan of a (mostly flat) surface, which returns radial distance at evenly ... | https://dsp.stackexchange.com/questions/10207/right-algorithm-for-fourier-transform-on-physical-heights |
Question: <p>Given a (FFT-sized) frame of data, and detection of a spectral component statistically above the noise floor in the FFT of this window, what characteristics or signal analysis could be used to determine that this spectral component is more likely to be a linearly swept sinusoid, rather than one that is sta... | https://dsp.stackexchange.com/questions/18950/detecting-a-frequency-swept-sinusoid-and-its-parameters |
Question: <p>BRISQUE compares your image with a pre-learned model with opinion scores. I am not sure about this but would image resolution affect the result of BRISQUE?</p>
<p>Moreover, if I have 2 filters and I would like to compare the results of using either filter to get the better result, can I use BRISQUE to quan... | https://dsp.stackexchange.com/questions/71033/can-i-use-brisque-to-compare-different-filtering-techniques-for-the-same-acquire |
Question: <p>Is it possible to design an LPF that has an output identical to the input at specific points in time domain (the rest of the input waveform can get filtered/distorted)?
Is there a general name/technique for this kind of thing (assuming it is possible), so that I can search for more information on this topi... | https://dsp.stackexchange.com/questions/21774/lpf-signal-values-unaffected-at-specific-times |
Question: <p>my name is niladri, I am new to image processing(actually this is my first code). I want to implement shock filter using structure tensor. I have rough idea of what structure tensor is and implemented in MATLAB. But to design a shock I need to calculate the sign using the dominating eigenvector. But as for... | https://dsp.stackexchange.com/questions/32409/shock-filtering-using-structure-tensor |
Question: <p>I am reading about <a href="http://rads.stackoverflow.com/amzn/click/0133457117" rel="nofollow noreferrer">estimation theory</a>, including topics like Bayesian Estimation (e.g. Wiener Filtering).</p>
<p>It seems that we usually define a filter in terms of tis frequency response (e.g. High Pass, Low Pass)... | https://dsp.stackexchange.com/questions/35526/how-is-bayesian-estimation-related-to-filtering |
Question: <p>I am the author of an amateur radio application that produces a waterfall display across a range of frequencies. Time is on the $x$-axis, frequency is on the $y$-axis, and the relative strength of each signal is depicted by the intensity of color.</p>
<p>Some of the signals have, what are called, <em>"key... | https://dsp.stackexchange.com/questions/36589/how-to-filter-key-clicks |
Question: <p>i am trying to create a low pass filter in matlab for a project in signals and systems class but i couldn't. cut off frequency is 500 , sampling frequency is 10000 and the low pass filter's band width is 150 Hz. If you help me i would be very appreciated</p>
Answer: <p>filterDesigner Filter Designer
... | https://dsp.stackexchange.com/questions/41619/how-can-i-create-a-low-pass-filter-in-matlab |
Question: <p>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 to analyse graphics.
I already concluded that the low-pass filter passes the zones of the graphic that ha... | https://dsp.stackexchange.com/questions/45193/recommendation-for-studying-filters |
Question: <p>I am trying to supply a solution for recording a court room and then, use some smart algorithms in order to automatically convert the speech to text.</p>
<p>In order to do that I have three boom microphones, one is near the judge, the second is on A side and the third one on the B side. this way i get 3 d... | https://dsp.stackexchange.com/questions/53871/filter-multiple-sources-from-each-other |
Question: <p>Can filter "depth" be adjusted by mixing dry and wet signals? </p>
<p>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 that has been bandpass filtered at 1kHz and the filter has around the same shape as the bandshelf/pea... | https://dsp.stackexchange.com/questions/24831/can-filter-depth-be-adjusted-by-mixing-dry-and-wet-signals |
Question: <p>I'm making a phonographic filter or simulator to make one of my songs to sound as if it was recorded on a phonographic cylinder.</p>
<p>Two important things about phonographic recoding is that sound gets more treble when it's louder and also that the recording of the wave is made top to bottom so the negat... | https://dsp.stackexchange.com/questions/84182/how-to-design-a-phonographic-sound-filter-in-python |
Question: <p>I have a a channel matrix "H" that is circulant.
I have data blocks.
I want to add the channel effects to the signal. </p>
<p>When H was only a vector of channel coefficients I would say:</p>
<pre><code>%Going Through The Channel
After_channel= filter(H,1,Data);
</code></pre>
<p>but now that H is a ma... | https://dsp.stackexchange.com/questions/25501/adding-channel-effects-to-a-signal |
Question: <p>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 suspect.</p>
<p>Can you explain, in lay terms, to satisfy the intuitions of a listener, why the matched filte... | https://dsp.stackexchange.com/questions/70173/an-intuitive-explanation-as-to-why-a-matched-filter-is-time-reversed |
Question: <ol>
<li>Apply the Composite Laplacian Filter first, then apply the gaussian filter.</li>
<li>Apply the gaussian filter first, then apply the composite laplacian filter.</li>
</ol>
<p>My work as below:
Here we assume the original image is function <span class="math-container">$f(x,y)$</span>
<span class="math... | https://dsp.stackexchange.com/questions/73111/the-result-if-order-of-two-filter-are-reversed |
Question: <p>this is a more general version of a <a href="https://dsp.stackexchange.com/q/38108/4298">question which i asked previously</a>.</p>
<p>From what I understand, the purpose of a filter is to change the amplitude of a specific frequency band.</p>
<p>As I went through the analytic calculation of a filtered (... | https://dsp.stackexchange.com/questions/38171/does-a-filter-add-oscillations-to-a-signal |
Question: <p>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 the current increases and therefore the amplitude of the sinusoidal gets bigger. I am looking for the maxi... | https://dsp.stackexchange.com/questions/46281/remove-high-resolution-spurious-peaks-from-sinusoidal-signal |
Question: <p>I have been given a sound file of a plane passing over a rain forest filled with birds. I am supposed to filter out the sound of the plane as it flies over. I've accomplished this with various types of filters in MATLAB, but I always run into one problem. I can either cut out all of the plane and lose som... | https://dsp.stackexchange.com/questions/19485/filtering-overlapping-frequencies-in-sound-file |
Question: <p>I'm wondering how I would go about filtering out a regular sine-wave signal with a constant frequency and voltage.</p>
<p>I'm really new to the whole DSP-area but imagine I would have a regular sinewave at a specific frequency (let's say 30Hz), and on top of that signal I have a bunch of other small peaks... | https://dsp.stackexchange.com/questions/24782/filtering-out-a-specific-frequency-in-an-analog-signal |
Question: <p>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 rivers can be modeled as a linear impulse-response system. Transport along a river reach changes as the upst... | https://dsp.stackexchange.com/questions/46302/use-of-an-impulse-response-system-to-model-sediment-transport-along-a-river-reac |
Question: <p>I mean what for the window FIR filter the filtered signal is truncated at the end because impulse response of the filter is symmetric. For example the code from here <a href="https://github.com/tmk/tmk_keyboard/blob/master/tmk_core/tool/mbed/mbed-sdk/workspace_tools/dev/dsp_fir.py" rel="nofollow noreferrer... | https://dsp.stackexchange.com/questions/55372/how-the-zero-phase-filter-without-filtered-signal-truncation-at-the-end-can-be |
Question: <p>I'm in a debate with a peer who says that filtering accelerometer signals at the chip-level has nothing to do with mitigating the problem of drift</p>
<p>Often, additional software-level filtering is employed to smooth noise, depending on the application. But My question pertains to any filtering or proces... | https://dsp.stackexchange.com/questions/72732/accelerometer-drift-what-hardware-level-signal-conditioning-operations-are-bein |
Question: <p>I am using Naudio open source library and I am trying to do some simple filtering. The problem is that I hear some "clicks", not too loud. The library offers me the possibility to use at least two buffers, so the computing time doesn't introduce a delay between them. Because in the most of the time I am d... | https://dsp.stackexchange.com/questions/8272/strange-noises-when-filtering-audio-signal |
Question: <p>I need help solving the following blurring function question.Assume an image $f(x,y)$ is moving in front of a camera so that $𝑥_0(𝑡)$ and $𝑦_0(𝑡)$ are the time-varying components of motion in the x- and y- directions respectively. The camera’s shutter opens at 𝑡 = 0 and closes at 𝑡 = 𝑇. Assume that ... | https://dsp.stackexchange.com/questions/16457/blurring-transfer-function-of-image |
Question: <p>I have signal, periodic with amplitude 10 with frequency about 10 kHz. It is hidden in Gaussian noise with standard deviation 100. Some idea how to extract periodic signal?
Thanks in advance.</p>
Answer: <p>Rearrange the signal into a matrix containing one period in each row and average down the columns. ... | https://dsp.stackexchange.com/questions/16511/extract-signal-from-big-gaussian-noise |
Question: <p>I am trying to design a filter whose magnitude is the same as that of a given signal. The given signal is wind turbine noise, so it has significant low-frequency content. After designing the filter, I want to filter white Gaussian noise so as to create a model of wind turbine noise. The two signals, that i... | https://dsp.stackexchange.com/questions/16781/how-to-design-a-filter-with-a-certain-magnitude-response |
Question: <p>I am working on my paper about comparing between mean filter and ordering statistics filter, The mean filter is contraharmonic mean filter and the ordering statistic filter is alphatrimmed mean filter. So, I compare contraharmonic mean filter and alphatrimmed mean filter</p>
<p>and i've read a slide prese... | https://dsp.stackexchange.com/questions/24064/difference-between-mean-filter-and-order-statistic-filter |
Question: <p>I have two signals, $x$ and $y$. I know that
$F(x)=F(s)+F(n)$ and $F(y)=F(n)$, where $s$ is the 'clean' signal, $n$ is the added noise and $F$ donate Fourier Transform.</p>
<p>To obtain the clean signal, I am trying the following: $s=F^{-1}[F(x)-F(y)]$. Is there any better way that I should try?</p>
Ans... | https://dsp.stackexchange.com/questions/24870/if-a-signal-is-added-to-another-in-the-freq-domain-how-to-filter-it-out |
Question: <p>I have a transfer function in Fourier space with $N=2028$ frequencies $(\frac {0, 1}{(N\cdot dx)} \dots ) $</p>
<p>Where $dx = 0.1m$. </p>
<p>I need to apply this transfer function to a signal with 20000 samples (also $dx=0.1m$).
When I transform this signal to Fourier space I get 20000 frequencies.
So... | https://dsp.stackexchange.com/questions/29783/in-fourier-space-how-to-apply-transfer-function-with-n-frequencies-to-input-dat |
Question: <p>I am studying the Kalman filter and its basic implementation, and it was asked to use the filter to estimate a signal observed in noise
$$y(n) = x(n) + v(n)$$
where $v(n) \sim \mathcal{N}(0, \sigma^2)$ and $x(n)$ is modeled as an ARMA(2,2) process
$$x(n) = b_0 u(n) + b_1 u(n-1) - a_1 x(n-1) - a_2 x(n-2).$$... | https://dsp.stackexchange.com/questions/35159/practical-examples-of-arma-model |
Question: <p>I do not have any background in Electrical Engineering and I have recently started a project which involves signals captured from sensors.</p>
<p>Lowpass filtering: The way I understand it is: The values below the <code>CUTOFF</code> are allowed to pass through, above the <code>CUTOFF</code> are simply fi... | https://dsp.stackexchange.com/questions/36661/understanding-the-lowpass-filtering-of-a-digital-signal |
Question: <p>I'm trying to record voice from a microphone into my laptop. Unfortunately, the laptop does not have a microphone jack, so I have to use the USB input. This doesn't present a problem when I record voice from my gaming headset, but when I record from the condenser microphone over USB I get a very pronounc... | https://dsp.stackexchange.com/questions/36725/how-do-i-eliminate-line-noise-from-usb-microphone |
Question: <p>I am trying to simulate a distributed sensing system and I need to filter only frequencies lesser than 500 Hz(low pass filter) from acquired signal with sample rate 1250000 Samples/sec using below mentioned program:</p>
<pre><code> %Time specifications:
Fs = 1250000; % samples per sec... | https://dsp.stackexchange.com/questions/38120/digital-lowpass-butterworth-filter-with-cut-off-500hz-and-sampling-rate-1-25msps |
Question: <p>How these processes are different from simple IIR 1 order filtering, FIR filters in terms of amplitude and phase characteristics?</p>
Answer: <p>Yes, integration and differentiation can be linear filters.
You can start from <code>laplace properties</code> that say:</p>
<p>$ \int_{0}^{t} {x(t)dt} \longrig... | https://dsp.stackexchange.com/questions/48584/do-integration-differentiation-processes-work-as-simple-filters |
Question: <p>What are the disadvantages of having too many poles?</p>
<p>Thanks.</p>
Answer: <p>Poles in a filter come from recursions. Consider a discrete filter</p>
<p>$$y(k) = x(k) + \alpha y(k-1)$$</p>
<p>where $y(k)$ describes the output of the system, $x(k)$ the input and $y(k-1)$ the output of the system the... | https://dsp.stackexchange.com/questions/49830/what-do-poles-do-for-a-filter |
Question: <p>Let's say I have a signal like x[k] = [-20 -50 -30 50 30 -60 60 -60 60 10 5 10 5 5], and I want to apply a lowpass and a highpass filter to this signal (separately). For example the impusle response of the filters are as follows:</p>
<p>Lowpass: h[k] = [-1 2 6 2 -1], k = -1,0,...,3</p>
<p>Highpass: g[k] ... | https://dsp.stackexchange.com/questions/59843/how-to-apply-discrete-filters-to-a-signal |
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