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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3answers
560 views

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 ...
6
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1answer
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. ...
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2answers
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 ...
4
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0answers
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 ...
4
votes
2answers
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 ...
4
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0answers
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 ...
4
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0answers
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 ...
3
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0answers
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, ...
3
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0answers
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 ...
3
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0answers
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 ...
3
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1answer
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 ...
3
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3answers
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 ...
3
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0answers
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 ...
3
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0answers
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 ...
3
votes
3answers
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'...
3
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2answers
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 ...
3
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0answers
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 ...
3
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0answers
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 ...
3
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0answers
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 ...
3
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0answers
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 ...
3
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0answers
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 ...
3
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0answers
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 ...
3
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0answers
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 ...
2
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1answer
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 ...
2
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0answers
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 ...
2
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0answers
70 views

Multiple 1D gaussian filter

Given $$ \begin{cases} &f_0(x)=1 \\ &f_{n+1}(x)= (\varphi*(f\mathbb{I}_{[a_n,b_n]}))(x)=\int_{-\infty}^{+\infty}\varphi(x-t)f_n(t)\mathbb{I}_{[a_n,b_n]}(t)dt = \int_{a_n}^{b_n}\varphi(x-t)f_n(...
2
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0answers
83 views

How do large tap counts effect FIR Filters for block data?

I am using the ARM CMSIS DSP library for a signal processing application. I find good filter performance when using tap counts that are in excess of 10% of the length of my data block. I understand ...
2
votes
0answers
57 views

How to compensate the energy of a spectrum after using a Tukey window

I am trying to minimise the noise in a spectrum of a variable. I have divided the time history in equal parts, I have multiplied every part of the time history by a Tukey window, done a FFT of each ...
2
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0answers
62 views

Blow sound recognition without using machine learning

I am building an experience in Unity that will recognize when a user blows into the microphone. I have a working implementation that recognizes sounds above a certain loudness threshold, but I want to ...
2
votes
2answers
250 views

Comparing two small audio files (e.g., hand clap) Android

Dears, I am new in signal processing and I am dealing with an Android application that should do some audio files (having the same "small" duration) comparison in real time. My main focus is ...
2
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0answers
131 views

Understanding Phase noise of a Hard Limited Sine Wave

I need help in understanding the frequency spectrum of a sine wave with phase noise that has been hard limited. Background: Typically, textbooks define phase noise for a sinewave, however in many ...
2
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2answers
243 views

Radar for Vital Signs – Matlab Simulation

Interested to design and simulate FMCW radar for Vital Signs sensing. The initial signal parameters are(1): When using the following Matlab code to generate the waveform: ...
2
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0answers
330 views

Arbitrary precision FFT and IFFT

For evaluating the performance of algorithms that use double-precision floating point numbers, or to pre-calculate double-precision floating point data to the best precision, it would sometimes be ...
2
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0answers
94 views

Strange behavior in fixed-point FFT

This is a problem I encountered some time ago while doing my thesis, which still now, I am not sure to understand. By then, I was implementing the circular correlation in an FPGA using this Xilinx FFT ...
2
votes
2answers
392 views

Where is the Nyquist frequency in an FFT spectrum?

I've been working with even length audio for a while, so I've had no problem finding the Nyquist frequency in the FFT spectrum, as it's shape is: $$(\text{dc_component}, f_1, f_2, ..., f_{N/2-1}, f_\...
2
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0answers
536 views

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 ...
2
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1answer
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 $ ...
2
votes
1answer
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 ...
2
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0answers
78 views

fft formula convention

I try to code up an algorithm from a scientific paper where they solve a differential equation using Fourier transforms. In the paper, they use the FT convention $\rho(x_i) = \sum_{k=-m}^m r_k e^{i2\...
2
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0answers
65 views

Understanding convolution of Chirp z algorithm

I dont´t understand how works the convolution part of the Chirp z. I understand how the DFT is transformed \begin{align*} x(k) = \sum_{n=0}^{N-1} x(n) W_N^{kn} \end{align*} to this expresion: \...
2
votes
1answer
153 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 ...
2
votes
2answers
273 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 ...
2
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0answers
117 views

Correct way of derivating in frequency domain with FFT

I believe I am very close to the answer and only need a small nudge to get to the answer. What I want: I want to take a signal, use FFT to transform it to the frequency domain (FD), multiply it by $...
2
votes
0answers
171 views

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 $f_s$ ...
2
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0answers
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 ...
2
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4answers
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)...
2
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0answers
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 ...
2
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0answers
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 $4$ ...
2
votes
0answers
98 views

DFT of signal with frequency-dependent time shift

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}{...
2
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0answers
281 views

NMR spectrum deconvolution on Matlab

I am doing an NMR experiment by applying an rf signal to a sample which is mounted near a magnet and in the end I get an FID (free induction decay) and do Fourier Transform to this FID . So we have a ...

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