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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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 ...
1k 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. ...
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 ...
83 views

Audio sharpening

I'm working on a project where the purpose is to enhance FFT peaks in a signal (musical instruments), by other words, reduce non-harmonic components and enhance the identified FFT peaks. So far I'm ...
469 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 ...
6k 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 ...
397 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 ...
43 views

How to find the difference between maximum and minimum of a time signal, having only Fourier series cofficients

I currently know the Fourier coefficients of a signal $c_n$, from the exponential Fourier series of this form: $$f(t) = \sum^{+\infty}_{n=-\infty} c_ne^{in\omega t}$$ Using these Fourier coefficients, ...
99 views

True Envelope algorithm: how to speed it up?

Premise (since somebody may ask about the background): I am the author of the spectral modules for SynthEdit, a modular framework for realtime STFT audio processing, currently struggling since ...
91 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 ...
784 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 ...
552 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 ...
5k 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 ...
105 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 ...
2k 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'...
4k 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 ...
432 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 ...
391 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 ...
288 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 ...
147 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 ...
945 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 ...
445 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 ...
813 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 ...
37 views

Averaging of a phase flipping signal

Suppose I want to average a signal $s(t)$ which consists of several spectral components without any DC offset. I sample $M$ points in the time domain. I am interested in the power spectrum which I get ...
44 views

Frequency Interpolation of DFT to create Zero-Padded IFFT

As a response to this question I have proposed interpolating new samples in a DFT (meaning the frequency samples of an existing DFT result) sufficient to be the new samples that if we were to take the ...
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How to take the inverse FFT of Welch's power spectral density in Python?

I have calculated the power spectral density of a weakly-stationary signal using Welch's method (as implemented by scipy.signal.welch in Python), and am looking to take the inverse Fourier transform ...
318 views

Flipping of x axis values when the Fourier Transform is compared to the FFT

EDITED 3-Jun-20 I have a lorentzian lineshape $$f(z) = \frac{1+iz}{R(1+z^2)} \qquad (1)$$ where $$z=\frac{-2\pi(f - F0)}{R} \qquad (2)$$ and $R$ is the decay rate, $f$ is the frequency, $F0$ ...
268 views

Phase difference calculation for non divisible sample rate to frequency

I am learning DSP and I am trying to simulate a simple phase difference calculator between two sinusoidal signals using GNU radio. My flowchart looks like this: As can be seen on the chart, one of ...
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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 $f_s$ ...
294 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 ...
2k 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)...
81 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 ...
477 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 $4$ ...
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}{...