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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Is the value of a frequency bin of a DFT-output the average of the 'real' frequency values within that bin's range?
I am wondering whether the value of a frequency bin with a certain resolution is the average of the fourier transform values of the 'real' frequencies within that bin's range.
3
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1answer
194 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,...
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3answers
525 views
real time overlapping buffer for FFT
I am currently using overlap-and-add method for a noise cancellation project. Everything works well so far except that when it comes to the real time, I need to overlap the input signals for FFT input....
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2answers
90 views
Half Window Functions and Formulas
I am looking at implementing different window functions in an application I am making. I have found the formulas for all major symmetrical windows, however I would like to implement half windows.
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1answer
294 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 ...
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1answer
38 views
Approximating Lorentzian Fourier transform with FFT
I have a Lorentzian frequency distribution
$F(w) = \frac{1+iz}{1+z^2}$
Where
$z = \frac{w-\Omega}{R}$
With $\Omega$ being the peak frequency and R the decay constant. I know that analytically ...
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1answer
39 views
Does FFT magnitude of a sound drop to 1/10 of its value if sound level is decreased by 20dB?
I understand decibel = 20 * log (magnitude/reference value). To truly appreciate the meaning of this, if I have a single frequency tone that corresponds to a certain FFT magnitude shown on a PC that's ...
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3answers
20k views
Apply Low pass Butterworth filter in Python
I want to use a low pass Butterworth filter on my data but on applying the filter I don't get the intended signal. Here is the dummy code:
Signal A:
...
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0answers
39 views
Scaling issue after doing manipulations in FFT
I have two filters which must apply (in series) to same input data $x[n]$ e.g of length 256 (real time asio input buffers). One filter is e.g. an impulse response $h_1[n]$ of a speaker and other one ...
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1answer
112 views
Using fourier coefficients to reconstruct data in matlab
When doing a discrete fourier transform on some data using matlab's fft function, its output is a set of fourier coefficients but I was wondering how do I go about converting these into an and bn so I ...
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2answers
7k views
Logarithmic fourier transform(LFT) on audio signal
I am trying to analyze the musics as possible as precisely.
Of course I tried FFT, but got some problems.
I found low frequencies have very low resolution than human's hearing. I tried very long time ...
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4answers
61 views
Is frequency resolution (“bin size” inverse) always directly proportional to sampling duration (window length)?
I am new to DSP and signal processing in general (digital or analogue). In implementing an FFT/Goertzel I have noticed that the resolution ("bin size" inverse) is directly proportional to the sampling ...
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1answer
1k views
How to plot Full spectrum with negative frequencies
I'm trying to do frequency analysis on signal of a rotating shaft. The signal is generated in two measurement planes i,e X-position and Y-Position. I was able to use fft and transform into frequency ...
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1answer
81 views
Is it possible to approximate given skewness and kurtosis values using an IFFT?
I have time domain data for a signal that behaves randomly but has non-random (non-Gaussian?) skewness and kurtosis values of ~0.9 and ~7.4, respectively.
The FFT of the signal shows that it's not ...
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% ...
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1answer
27 views
Does FFT magnitude of recorded sound depends on the sound card used?
Based on some reading, here's my understanding of how sound recording works:
Microphone outputs a voltage based on sound pressure. The microphone voltage is amplified and the amplified voltage goes ...
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1answer
37 views
For a 20 MHz channel, with 1200 usable subcarriers(18 MHz occupied), what should be the size of the FFT on the transmitter before IFFT is applied?
I am trying to simulate the SC-FDMA transmission and I read H Myung's paper and research on the topic but I am not sure exactly on what should be the size of FFT at the transmitter.
For a 20 MHz ...
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0answers
16 views
Using KNN for audio classification based on FFT
I need to classify whether a product sound "good" and "bad" based on FFT of its audio recording. The FFT magnitudes are shown for frequencies from 0 to 7khz, with a frequency resolution of 5 hz, so ...
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1answer
48 views
Why autocorrelation can be more efficiently calculated using the fft
Can anyone explain why autocorrelation can be more efficiently calculated using the fft ?
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2answers
66 views
Is the FFT magnitudes of multiple audio sources additive?
Is the FFT magnitude of two audio signals when played together the sum of their individual FFT magnitudes across frequency bins? Note I'm talking only about the magnitude, ignoring phase.
For ...
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3answers
61 views
Analyzing a signal that contains frequency content at Fs/2 doesn't seem to work unless there is a phase shift
I am trying to write a basic program that samples a 4 kHz sinewave at a sampling rate of 8 kHz and takes the FFT of the signal and plots it.
From everything I have read, as long as the signal you are ...
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0answers
25 views
Discrete vs Continuous Power Spectrum Analysis
Let's suppose we have a time signal of watt units which in the frequency domain transforms into a sinc-like spectrum. The amplitude of that spectrum will likely be also of watt units. In order to find ...
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1answer
435 views
1D convolution and deconvolution using FFT
The task: there is some original signal, and there is some response function. I need to convolve them using FFT and then do deconvolution to restore original signal.
The task graphical illustration ( ...
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1answer
61 views
fft function in R vs spec function from 'seewave' package? They don't give a similar frequency spectra
I understand that spec will give me the frequency and the corresponding amplitude of that component, whereas fft will compute the DFT of the signal and throw the complex numbers for each component ...
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1answer
38 views
Phase measurement with FFT for DMM Keithley 7510
I am currenly working on a measurement driver for the DMM Keithley 7510. I implemented a harmonics measurement using FFT (from MathNet library).
First I simply sample my input voltage signal and ...
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2answers
57 views
How can I graph the amplitude for a given frequency window over time?
I have simple audio files of single plucked notes from a guitar:
I want to graph the amplitude of each partials over time, rather than just have this color-coded "dB" intensity graphing I have here.
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0answers
44 views
ACF expression using FFT method
It is known that the ACF of a 2D digital image can be obtained by computing the inverse Fourier transform ...
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1answer
82 views
How can I “smooth” square wave/pulse data to remove/reduce harmonics?
First, I have reviewed close-match questions, but can't find something that seems to match what I'm looking for. Also, it's been 30 years since college, so forgive me for forgetting stuff.. :)
I am ...
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3answers
58 views
What's the difference between calculating the FFT of a constant sound for 5 seconds versus every 1 second and taking the average?
I'm interested in learning the frequency spectrum of a constant sound. However, there is background noise. My plan is to record the sound for 1 second, take the FFT, and then doing this 5 times and ...
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0answers
76 views
SNR computation in frequency domain with scipy.fftpack.fft
I have a noisy time series (gaussian coloured noise) to which I add a signal. I call the sum of noise+signal a segment. In the picture you can see an instance:
I want to compute the signal-to-noise ...
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0answers
37 views
How to remove sum and difference terms while computing frequency from fft?
I am currently operating in vhf band AND trying to detect frequencies using Fast Fourier transform thresholding method. While detection of multiple frequencies , i received spurs(May not appropriate ...
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1answer
79 views
How to group similar patterns together that are shifted along the x axis?
I have a set of time series plots that are shifted along the x-axis. I am looking for an algorithm using which I can group them together. My end goal is to average them.
While reviewing the ...
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6answers
676 views
Clarification on defining FFT bin sizes
When reconstructing the frequency domain for an FFT, what is the most self-consistent way to do this -- i.e. how is it best to define the bin width, $\Delta f$?
For example, previously I thought it a ...
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2answers
136 views
Understanding the phase in DFT
Although using regularly the FFT algorithm to compute DFTs, I don't really understand how the phase part works.
e.g. in Matlab:
...
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2answers
110 views
Interpolation of estimated frequency between fft bins: what about phase?
I am developing an additive re-synthesier, which involves identifying partials from FFT frames.
Partial identification requires amplitude peaks. The simplest way to do this is to just look at the ...
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2answers
38 views
(FFT) Is it good practice to ignore each bin's phase when characterizing a signal
I have been recently working with the FFT and realized that each frequency within a bin corresponds to a phase. I understand why a phase that corresponds to each frequency is needed because it allows ...
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3answers
5k views
How to Average Complex Responses (and Justification)?
I am developing software that calculates the response of a system by comparing the FFT of input and output signals. The input and output signals are divided into windows and, for each window, the ...
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1answer
66 views
Power and Energy of an Audio file
I have a question about calculating the power and Energy of an audio recording.
I have imported a waveform of an audio recording in WAV format into Matlab and then calculated the energy and power in ...
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1answer
30 views
What does imaginary Component of a bin's phase in the Fourier Transform mean?
I have been recently working with the FFT and realized that each frequency within a bin corresponds to a phase. This concept I have no problem with because if you wanted to construct the summation of ...
0
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1answer
23 views
psd in matlab, problem with the coefficient $\frac{1}{2\pi}$
Using the fft to mimic the Fourier Transform I have this question.
The definition that I use for the PSD $S(f)$ is:
$S(f)=\frac{1}{2\pi}\int_{\infty}^{\infty}exp(-i\tau2\pi f)r(\tau)d\tau$
where r(t)...
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0answers
25 views
Classifier for detecting abnormal noise using FFT
As a simple machine learning application for quality evaluation, I want to detect abnormal sound in a product. A good product would make a signature humming sound, a bad product might make noises due ...
0
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1answer
38 views
Unexpected frequency components found in Periodogram
I am trying to make sense of Digital Signal Processing with the help of R language. So, the issue came up when I tried to create a periodogram for a simple sine wave. The regular periodogram shows 2 ...
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4answers
1k views
Convolution and windowing using a buffer - how do I do overlap add?
I'm reading chunks of audio signal (1024 samples) using a buffer, applying a Hanning window, doing an FFT on it, then reading an Impulse Response, doing its FFT and then multiplying the two (...
0
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1answer
48 views
Is the sum of all FFT bins from Audacity equal to RMS amplitude?
Say I open an audio segment with Audacity. Then I perform a size 4096+ FFT with a rectangular window. The program outputs bin magnitude values in dB. If I add them all together like this:
sum = 10*...
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0answers
29 views
how to cut off frequencies above 10 Hz with Welch method
This is the first time I have worked with any code so I am rather unfamiliar with all the lingo and processes involved. I am trying to convert a signal to the frequency domain and want to find the ...
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2answers
49 views
The DC bin of an FFT should be the mean of all the other bins in the spectrum, why does the FFT produce 0?
I am experimenting a bit with FFT's. From what I have read the first bin of an FFT is called the DC bin -- that is it is the mean of the other components in the FFT. I have also read that it should be ...
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1answer
29 views
Should you scale the FFT bins by $1/N$ where $N$ is the number of points in a transient signal?
I am going over the following tutorial. In section 2.1 the author says
To calculate the $N$ point FFT the Matlab algorithm 1 can be used. Here, after taking the FFT, its magnitude is calculated ...
2
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1answer
91 views
MATLAB FFT Amplitude
I am trying to understand FFT, DFT through Matlab and I am fairly inexperienced in all of these. I have a time vector t and a corresponding simple sine wave x = 0.25*sin(t). Now I would like to view ...
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1answer
486 views
How to calculate LF/HF power ratio from heart rate RR-intervals
I've been searching everywhere, but I'm having trouble finding a step-by-step guide for someone with little knowledge of signal processing.
I have a file with a list of RR-intervals in milliseconds. ...
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2answers
83 views
Have you seen these kind of DFT artifacts?
we've been working with the Spiral DFT Implementation on an FPGA and managed to get it working.
Unfortunately we see these strange artifacts and have no idea where they come from. Our input is a ...
