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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What is wrong with band passing by inverse Fourier transforming a section of a signal's frequency spectrum? [duplicate]

If the question is unclear, here is what I mean in Python using scipy.fftpack: ifft(fft(arr)[x:y], len(arr)) What is wrong with this method of band passing?
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549 views

How do I calculate peak amplitude of the signal components after zero padding and FFT?

I am learning about DFT and trying to apply it to some audio processing. I am new to DSP but experienced in programming and have some background in math and physics. The FFT algorithm I use (lomontFFT)...
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18k views

Calculating values of frequency bins in Python

I have to use FFT to determine the period of waves inside a signal. After applying FFT on a window of 10000 point from a signal, I get something like this: What I don't understand is that FFT is ...
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53 views

How to select the sampling frequency when the input signal frequency is not known

I am trying to observe the noise produced by UPS present in our lab, under no load (a normal running condition when its power is on). As I don't know its frequency range, I have randomly chosen 4 ...
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204 views

What Are the Suggested Resources for Extensive Knowledge About the Fourier Transform (DFT, FFT, etc…)? [closed]

I would like to improve my knowledge about Fourier Transform related subjects. What are the recommended resources in order to self study about it?
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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 ...
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82 views

Multiple different length Windows for high resolution realtime FFT

I am looking at trying to achieve a 'realtime' (as quick as possible to acquire, process and present data) FFT application to analyse audio. I have setup my app to acquire a set of samples, apply a ...
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577 views

frequency domain zero padding in OFDM

I have a matlab code for OFDM system, in this code the auther added a zero padded subcarriers before the IFFT stage, in the middle of the raws, and added zero padded guard interval after IFFT, my ...
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How do I perform Spectral Analysis (FFT, Windowing, Detrending) on Sonic Anemometer Data?

I am having trouble with all of the forums that I have looked at thus far. I have one day's worth of Sonic Anemometer data. I want to see the spectral analysis of this data. When the anemometer data ...
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1answer
49 views

FFT and Power Spectrum Normalization

On many websites, including MathWorks, it was suggested to normalize the fft spectrum (MATLAB or numpy) by dividing it by the total number of samples ($N$). For a sinusoidal signal, for example: $$x(...
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227 views

Applying a filter kernel defined by a continuous equation in the frequency domain

I have a filter kernel $K(\omega)$ completely defined in the frequency domain as a continuous function of angular frequency $\omega$. I know $K(\omega)$ defined as a continuous mathematical equation. ...
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100 views

Shifting signal smaller than discrete step

I have a image that I need to shift with less than a pixel. My plan was to do a Fourier transform and multiply the signal with $e^{-aiu-biv}$ where $a, b$ are the shifts in x and y direction. This ...
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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 ...
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2k 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 "...
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3answers
87 views

Reverse engineering a digital filter

I have 2 simultaneous signals that both are designed to measure eye movements. They are sampled at 250 Hz. We have 12 subject recordings. For 3 stable periods in each subject, we choose 256 points and ...
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improving performance of KNN classifier on FFT

I need to classify whether a product is passing or failing based on a noise check. I have 100 labeled "good" product and 100 "bad" product. For each, I recorded the sound for 3 seconds and each chunk ...
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317 views

Why is pitch detection necessary in pitch correction?

I am trying to understand the process of changing the pitch of a signal. After performing the FFT on a windowed signal, we have the frequency information for the signal. Why can we not simply change ...
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943 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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275 views

Calculate the harmonics in the power supply of $50\textrm{ Hz}$ using FFT

I need to calculate the harmonics in the power supply of $50\textrm{ Hz}$. If suppose I need to calculate up to 20 harmonics. Then my sampling frequency by Nyquist criteria should be : $$f_s \ge 2 ...
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1answer
315 views

How to smooth (or interpolate) phase of FFT and reduce data points

I am writing some code for audio analysis, and have currently got two signals with FFT performed on them. I get the phase of my complex array by using: ...
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3answers
59 views

Is spectral leakage normal for FFT?

I record a sample of 2000 data points, with a sampling rate of 40khz, so freq. resolution is 20hz. I multiplied the chunk of data with a Hanning window prior to doing the FFT. I recorded a 1khz test ...
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87 views

Intermittent Harmonics, Is there a physical explanation?

I am doing some frequency analysis of a vibration signal from a spinning rotor, I am expecting to see peaks corresponding to the rotational frequency of the rotor. So the fundamental frequency ...
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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'...
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78 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
32 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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135 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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235 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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116 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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29 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
29 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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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
49 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
33 views

Multi Resolution STFT for audio

I am looking at trying to achieve a multi resolution STFT analysis on a log sine sweep audio measurement. I have an impulse, and I would like to perform two different windowed length FFTs on the ...
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1answer
26 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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313 views

What is the proper way to implement a real time spectrogram?

I am trying to implement a real time spectrogram and I am not sure that I am implementing it the right way. First I decided to use Matlab, I was taking a short recording of lets say 0.1 seconds and ...
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1answer
49 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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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
152 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
370 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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48 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
180 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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466 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
82 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
13 views

LOW SIGNAL NOISE RATIO FFT

I am processing an audio signal with CS4244 codec at 24 bits and 48kHz. Every 2048 samples the FFT is calculated and its amplitude. Without input signal I get FFT values ​​close to 30dB. With maximum ...
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1answer
42 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 ...
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1answer
291 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
31 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
15k 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: ...