Questions tagged [fft]

The fast Fourier transform is an efficient algorithm to compute the discrete Fourier transform (DFT) and its inverse.

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440 views

Efficient calculation of correlation function every $N^{\rm th}$ point

I would like to calculate a long correlation function of length say, 1e6 points. I have a prior knowledge that the correlation peak will be in point k*1000. Is there an efficient way to apply this ...
6
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1answer
791 views

How to choose a windowing function when performing a Chirp $\mathcal Z$-transform?

I've got some data sampled at 32 Hz. The only interesting stuff is happening between about 0-8 Hz and I've got to do real time batch processing and detect events with a delay of no more than a second. ...
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2answers
3k views

Filter size vs. FFT size and Overlap add

First of all I would like to say that : I'm new to DSP and in a "learning curve" hence the "basicness" of my questions. I've read as many posts/articles as possible on the internet but still need "...
4
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2answers
344 views

Matching LPC frequency response to raw sample data frequency response

Background: The intention is feature/formant extraction from speech input. I have used an FFT on raw samples in the past and have done simple peak picking from a weighted average of frequencies over ...
4
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2answers
2k views

What is the right way to extract particular frequency phase information from FFT?

Signals acquisition: Two proximity sensors are positioned Orthogonal to each other. These would be measuring the vibrations of a rotating shaft. Problem: Each of these 2 signals has two dominant ...
4
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1answer
637 views

$1/n$ octave complex smoothing

An excellent answer to this post explains how to do $1/n$ octave energy smoothing and mentions complex smoothing can be done as well, but it's tricky business because of phase wrapping. How is phase ...
4
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1answer
1k views

STFT overlap add method - What am i doing wrong?

i am trying to implement a short time fourier transform of an audio signal to do some filtering (subtract another stft signal using spectral subtraction). I am using a hamming window function with 50% ...
4
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0answers
355 views

Extra 'fingers' found in graph after applying FFT in MATLAB

I am currently working on the FFT function in MATLAB. In this function, i require a 1024 number of data Time Domain plot and produce a 400 number of data Frequency domain plot. After applying the FFT ...
3
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0answers
57 views

Solving the PDE of a signal in an optical fiber

Disclaimer Please let me know if this is not the right place to ask this question. Disclaimer Part 2 I have no background in signals. My focus is in mathematics. In my programming class, we are ...
3
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1answer
199 views

Phase Modulation and Intermodulation Anomaly ? (Spectrum plots inside - **Updated**)

I have a MATLAB script for modelling a two tone signal (a sum of two sinusoids) going through a non linear transfer function (such as an amplifier). The amplitude of a signal is amplified non linearly,...
3
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0answers
3k views

What's spectral “tilt”?

I am looking at article Speech-in-noise intelligibility improvement based on spectral shaping and dynamic range compression. In paragraph 2.2 the article mentions "tilt" of the spectral envelope. The ...
3
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2answers
3k views

Correlation Using FFT / IFFT (Convolution in Frequency Domain) in Java

I try to find about the delay between two audio files using Cross Correlation in Java. I've already done this algorithm so far that i get a idea about how many samples is the delay. FFT x1 -> Zero ...
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0answers
410 views

Why are the frequencies associated with the STFT bins not representative of the energy?

I'm experimenting with a 30 Hz sine wave and I want to extract the frequencies of this, and have implemented the algorithm to do so. This gives a correct reading, well, gives 29 Hz. I'm now wanting ...
3
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0answers
341 views

choosing windowing function for audio signal processing

Currently I am working on chord recognition, and get confused on several aspect. One of them is choosing windowing for reduce the spectral leakage. from what I read, hanning/hamming function is a ...
3
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0answers
5k views

1/3 octave spectra from fft

I have got a signal in frequency domain. This is a frequency response function from software, so I can do nothing about it and have to leave it in frequency domain. I want to transfer the data to 1/3 ...
3
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0answers
261 views

Points of interest in a spectrogram

We have several audio tracks, all of which are different versions of the same track under one type of distortion: the speed of the track is increased by a constant factor K (hence frequences are ...
3
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0answers
137 views

Compensate for standard deviation loss

I am not sure if this question will be a little off-topic on this forum, that I will give it a try anyway, since it implies signal process arguments. By using ...
3
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0answers
773 views

What is the proper way to select the time-bandwidth product when doing multi-taper spectral estimation?

Frequently programs for calculating the spectrum of a signal using the multi-taper method require the user to specify the time-bandwidth product. What considerations must be made when selecting the ...
3
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0answers
387 views

Discrete Fourier transform in a multidimensional space

I want to measure the frequencies at which a point oscillates in a multidimensional space, let's take the example of a point on a 2d-surface. For now, I naïvely split the signal in two, along the ...
3
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0answers
792 views

calculate root mean square, ratio of power, constant false alarm, … of a signal

I am new to Digital Signal Processing, and am reading a paper. I don't know how they can extract some features from signal, like root mean square (RMS), Constant False Alarm Rate - CFAR, Mean Value ...
3
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2answers
1k views

Problems Using FFT to Compute Impedance in a Model Neuron

I'm a neuroscientist currently investigating the resonance properties of a single neuron model that a colleague and I have constructed. The language we code in is Julia, which I hope is similar enough ...
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0answers
35 views

Understanding convolution of Chirp z algorithm

I dont´t understand how works the convolution part of the Chirp z. I understand how the DFT is transformed \begin{align*} x(k) = \sum_{n=0}^{N-1} x(n) W_N^{kn} \end{align*} to this expresion: \...
2
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1answer
60 views

Serial filtering using FFT

lets say I have 3 audio filters for a realtime DSP application. For simplictity each has length 256 (as well as the input signal). The filters should work in series. Starting with filter IR h1(n) and ...
2
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0answers
75 views

Correct way of derivating in frequency domain with FFT

I believe I am very close to the answer and only need a small nudge to get to the answer. What I want: I want to take a signal, use FFT to transform it to the frequency domain (FD), multiply it by $...
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0answers
56 views

N-dimensional Cooley-Tukey FFT

I am trying to formulate an n-dimension algorithm for the Cooley-Tukey FFT. In two dimensions, it is very clear: $$ Y[k_1 + k_2 n_1] = \sum_{l_2 = 0}^{n_2} \Biggl[ \Biggl( \sum_{l_1 = 0}^{n_1 - 1} X ...
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0answers
100 views

Frequency resolution - resolution BW relation for different window functions

I am working on FFT based spectrum analyser and I am not sure how to proceed calculating the RBW. According to the document (page 20) in the link, RBW described as $k \frac{f_s}N$ where $f_s$ ...
2
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0answers
144 views

Frequency translation with FFT

I'm new to DSP and I would like to make an SDR receiver. For this I want to shift bins in the frequency domain in order to bring to 0Hz any portion of the base band spectrum (I know that there are ...
2
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2answers
128 views

Higher-order Kronecker product

I am trying to generate a 2D DFT matrix in matlab, which I need for 2D compressed sensing (CS) problems. Lets say $N_1=8$, $N_2=16$, then according to the requirement of CS, to generate a 2D DFT ...
2
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0answers
68 views

Selection of transforms, windows, wavelets for current signal analysis

I am new to this topic but I already read about the propperties of different signal analysis methods. I am stuck with the obvious advantages of FFT, STFT and Wavelet transforms but can't find good ...
2
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1answer
391 views

Decimation and Upsampling

I'm currently trying to understand decimation and possible ways to retrieve back the original signal. If the signal of supposed frequency $X$ Hz was decimated by a factor of four, would there be $4$ ...
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0answers
55 views

DFT of signal with frequency-dependent time shift

I have a signal $s[n]$, and I want to compute the DFT of it with a frequency selective shift (computing all $k$ here but I really only care about $k > 0$) $$S[k] = \sum_{n=0}^{N-1} s[n-\dfrac{N}{...
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0answers
211 views

NMR spectrum deconvolution on Matlab

I am doing an NMR experiment by applying an rf signal to a sample which is mounted near a magnet and in the end I get an FID (free induction decay) and do Fourier Transform to this FID . So we have a ...
2
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0answers
64 views

Compensate for residual resampling error with a quadratic phase vs frequency- possible?

With regards to the answer in this post Is it possible to do deconvolution with two data sets that have different sampling rates? I suggest the possibility of compensating for a residual resampling ...
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0answers
80 views

Is there a way to extract the difference between the 2 signals? Fitting?

I'm very new to signal processing and have been struggling this far. I have a device that measures the signal due to the magnetization of iron particles. In order to calculate the response of the ...
2
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0answers
136 views

Constructing $X(\omega/2)$ from the FFT of $x[n]$

Let $X(\omega)$ be the Discrete Time Fourier Transform (DTFT) of $x[n]$, I want to construct $X(\omega/2)$. Precisely, I use FFT function to compute the samples of $X(\omega)$ in one period, say $[0,...
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0answers
4k views

Calculate 1D Power Spectrum from 2D Images

Imagine satellite images, these are irregular sampled in X and Y direction and the shapes are of course are oddly off. We now want to estimate a 1D power spectrum from the whole image to estimate the ...
2
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0answers
198 views

How to iplement fast DCT-III (for inverse of DCT-II) using FFT or iFFT

I have to implement DCT-II and its inverse which is equivalently given by a forward DCT-III. All from real to real. I have been checking some paper where it is specified how to do it. However, I ...
2
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1answer
297 views

How to make sense of audio file's $\textrm{dBFS}$?

I'm trying to build a song visualizer using the FFT method. When I analyze the audio, I get the $\textrm{dBFS}$ value for each frequency range. From my understanding, $\textrm{dBFS}$ is in the range ...
2
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0answers
96 views

Calculating Phase Response for 2 microphones given the input

I want to measure and compare the phase response of two identical microphones. To do this, I have a pressure chamber, which allows to apply the same pressure to both microphones. I apply a sweeped ...
2
votes
3answers
1k views

Aquila DSP C++ Library - Wave file FFT analog frequency off by factor of 4?

I'm working with the Aquila C++ DSP library. I'm computing the FFT of a wave file (16 bit depth, single channel, 44100 sample rate). I am using a window size of 16384 to calculate the FFT spectrum. I'...
2
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0answers
102 views

Low-rank approximation of the STFT matrix : good audio compression technique?

I recently heard about this interesting math tool: With a matrix $A$, compute the SVD ($ A = U \Sigma ^tV$ factorization, where $\Sigma$ is a diagonal matrix containing the "singular values") Keep ...
2
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3answers
4k views

Interpreting the FFT results of a structural vibration problem

System: Cantilever beam type models, with an accelerometer on the end. The beam base is excited at various frequencies in harmonic fashion on a shake table. This results in vibration in the beam. The ...
2
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0answers
544 views

How do I interpolate between bins on an FFT in python?

I have four frequency peaks, which I have after applying FFT. Now I want to know precise values of these frequency peaks. there are different interpolation methods. How can I use this method of ...
2
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0answers
242 views

MFCC help: Same sound, different coefficients?

Problem: I'm getting totally different MFCC coefficients for identical and near identical sounds like several finger snaps or several desk taps. What am I missing? I thought since the timbre doesn't ...
2
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0answers
204 views

Noise analysis with periodogram. problem with different resolutions

I don't understand what happened in my simulation. I would like to perform a trade off between 2 ways to estimate power in frequency bands. I want to detect spurious frequencies. First method: I ...
2
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0answers
90 views

Signal Relationships and Conversion: Transforming one Signal into another?

I'm trying to relate a near shore tidal signal (point A) to 3 points along a long model boundary (points B C D). I want to possibly have a relationship between B C D with which we can convert A ...
2
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0answers
126 views

How to properly express periodogram over analog frequency?

Assume a signal $x[n]$ with $N$ samples and its $N$-point FFT $X[k]$. Assume also the simplest power spectral density estimator: $$ S_x[k] = \frac{\left| X[k] \right|^2}{N} $$ With this estimator, ...
2
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0answers
2k views

Phase of an image in the frequency domain

The following picture is amplitude and phase of an image filtered using Gabor filter in the frequency domain. I'm trying to reproduce the results, I'm getting valid results for the amplitude I think, ...
2
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0answers
881 views

Matlab: Estimating power spectral density of an experimental data?

How do I estimate PSD in matlab ? There are lots of methods to estimate, depending on application. I am using an experimental data to identify faults ( e.g. interturn fault of electric motor), which ...
2
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0answers
479 views

Averaging frequency spectra gives incorrect RMS

I am averaging several FFT results together in an attempt to get the average frequency spectrum of a track of audio. I'm attempting to use Parseval's Relation to get the RMS of the audio from the ...