# 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?
3answers
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### 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)...
3answers
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 ...
2answers
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### 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 ...
1answer
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### 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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### 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 ...
2answers
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### 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 ...
1answer
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### 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 ...
3answers
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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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### 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: ...
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 ...
2answers
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### 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 ...
3answers
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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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### 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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### 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 ...
1answer
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### 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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### 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 ...
1answer
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### 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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### 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 ...
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. ...
1answer
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### 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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### 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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### 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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### 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 ...
2answers
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### 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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### 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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### Filter length in a Filtered-OFDM system

In a filtered-OFDM system, lets say we have the following parameters: ...
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### 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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### 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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### 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,...
3answers
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### 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....
2answers
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### 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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### 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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### 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 ...
1answer
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### 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 ...
1answer
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### 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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### 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 ...
3answers
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### 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: ...