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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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'...
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2answers
89 views

Checking Parseval's Theorem for Gaussian Signal by Using Scipy

I'm trying to check Parseval's theorm for Gaussian signal. It's well known that fourier transform of $\exp(-t^2)$ is $\sqrt{\pi}\exp(-\pi^2 k^2)$. So I implement it by using quad and simps. I think ...
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1answer
34 views

The frequency-domain spectrum obtained using MATLAB's FFT function seems inaccurate

I am using MATLAB R2018b on a Windows 10 PC. In order to obtain the frequency domain information of a signal, I'm applying MATLAB's FFT function. The signal of interest is obtained from an experiment ...
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1answer
148 views

Question About Linear and Circular Convolution - 1D and 2D

We know that applying a filter on an image is called correlation or convolution depending on the filter angle. In Gonzalez I have read that we can apply linear convolution on an image. Here I have a ...
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1answer
239 views

How to Combine 8 N/8 FFT's into one N FFT

I need to make in FPGA (using Verilog) an FFT. Input data is N=8192 points at 1 GSPS. However, the FPGA operates at 125 MHz, therefore the data is split into 8 channels (each one at 125 MHz). This ...
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1answer
117 views

Distinguish between sinusoidal oscillations and rythmic pulses

I want to distinguish between a sinusoidal oscillations and rythmic pulses. Which method can i use to do that? I want to implement a detector that can distinguish between this two types of signals. ...
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31 views

Is the whole data needed for the fast fourier transform with an ongoing signal?

I sample an signal with around 840k values a second, to check the spectrum around <400kHz. But i can't save the whole data as it is just to much for my microcontroller. I know that for the ...
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2answers
5k views

Avoiding “RuntimeWarning: invalid value encountered in log10” when converting FFT data to dBFS

I've attached some code below that I'm having some issues with. Basically, I need to write a function to get the dBFS value of a signal at a specified Hz value, at a given channel in a .wav file. ...
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1answer
31 views

Measuring group delay through a linear phase filter

I'm having a difficult time wrapping my head around a phase/ group delay problem and it corresponding to where you start sampling so I built up a system to test some of these issues. The system passes ...
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0answers
33 views

How do I find the peak frequency of a wav file in python?

I am fairly new to python and signal processing and I was given a task to record audio for 'x' seconds and then find the peak frequency in the audio file. So far I have successfully implemented the ...
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1answer
54 views

Why would FFT interpolation using zero-padding undershoot the true frequency of a single tone sinusoid?

Why would FFT interpolation by zero-padding or using the Chirp Z-Transform produce a maximum at a bin that corresponds to a frequency less than the input frequency of a single tone sinusoid? I am ...
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1answer
28 views

Is fft2 in MATLAB unitary? Some differences happen

I meet a problem when implementing fft2 in MATLAB. The question is I try to simulate the realistic measurements $Y = |FCXF^H|^2$ - the intensity of Fourier domain of object $X$, where $F$ denotes ...
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1answer
50 views

What measurement errors occur in an inteferogram?

This question is bordering on physics, but it is the signal side that I am interested in so I hope this is the correct forum. I am very familiar with instrumental noise sources in CCD based ...
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0answers
17 views

Normalizing FFT after recording with Pyaudio

I have a USB microphone that has a 24 bit ADC, however, Pyaudio doesn't allow for 24 bit encoding so I used 16 bit. I hope this is not an issue. Having said that, I hope to make sure I'm applying the ...
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1answer
153 views

Filter length in a Filtered-OFDM system

In a filtered-OFDM system, lets say we have the following parameters: ...
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1answer
377 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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2answers
51 views

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.
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1answer
188 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
498 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
85 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
292 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
34 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
37 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
18k 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
37 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
96 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
58 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
985 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
75 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 ...
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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
26 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
31 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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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
46 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
65 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
56 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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22 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
415 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
49 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
36 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
54 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
42 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
72 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
51 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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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
78 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
635 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
112 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: ...