The fast Fourier transform is an efficient algorithm to compute the discrete Fourier transform (DFT) and its inverse.

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How to choose a FFT algorithm?

There are many algorithms that can calculate FFT. Even with Cooley–Tukey FFT algorithm, different radix can be used and the algorithms can divided into decimation in time and decimation in frequency. ...
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32 views

How is split step fft beam propagation method constructed numerically for spatio temporal evolution of short laser pulse?

For ultrashort laser pulses the propagation in space couples both transverse space and time domain to produce transverse space dependent pulse duration. A split step fft method propagates single ...
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87 views

How can an FFT plot not be symmetrical?

From discussions all over the internet, I understand that in a 512 sample point fft, only the first 256 contain relevant information provided that my signal is real. The other half contains complex ...
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27 views

Using Matlab xcorr to find a signal hidden behind noise

I am trying to develop some code to find a transient signal in a data set I have. Before I get there however I am trying to write code using simulated data. I first am creating a signal with a ...
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67 views

Extracting features from a signal: a 21st century approach?

I have a discrete signal of random length (let's say between 500 and 800 in time). This signal has an important information which I want to collect and to put it together in a feature vector to train ...
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1answer
70 views

IFFT and OFDM upconversion

I am trying to understand some basic concepts with IFFT as represented here OFDM Modem page 50-53, and 108-114. Some basics that are represented - The base sampling rate is 8khz. The symbol period ...
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2answers
74 views

Why would you use DCT as oppose to DFT for detecting oscillation in a signal?

For my project, I needed to detect the primary oscillation frequency in a signal data. As I googled around, it turns out that DCT is the best tool to find this ...
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1answer
50 views

How to get Envelope of Audio Signal

I am having a problem understanding how the author gets the onset envelope of an audio song in this paper. The entire section in question has been quoted below: First the input sound is resampled ...
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31 views

Matlab : Reconstructing signal from cosinusoïdes : DC offset question [code + picture]

i'm trying to reconstruct a signal from amplitude and phase of cosinusoides to understand totally how DCT works. So far i've managed to get the signal very close, only by a DC offset that i don't get. ...
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54 views

Plot real-time FFT using matplotlib

I'm trying to plot fft in python. I use pyalsaaudio for capturing audio in PCM (S16_LE) format. I use the ion() and draw() functions in matplotlib to have the fft plotted in real time. This is the ...
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46 views

matlab's FFT is wrong? don't think so

i'm trying to get an FFT of a signal my FFT has 1024 samples i added two sin functions one of freq 86,13Hz (2*44100/1024) and one of freq 129,19Hz (3*44100/1024) the WAV looks like that (for 1 ...
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2answers
67 views

PSD estimate using FFT - scaling and units

I know that similar questions have already been asked – I have read the answers, yet I am still not sure if I understand the topic properly at some points. I want to use FFT to calculate PSD ...
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34 views

FFT followed by Inverse FFT : Not working. (dsPic33F)

I have a small problem, I perform an FFT then immediately an inverse FFT, but the resulting array is not the same. Please what am I doing wrong ? (the size of Frame is 128 hence the 7) ...
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34 views

DFT method for a sequence defined by recursion

I encountered the following interesting computational problem in the field of computational statistical mechanics. https://www.dropbox.com/s/8yimd9uy5u0jrdj/dftpr.pdf thank you :)
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1answer
78 views

Calculation of cosine in frequency domain instead of calculatin in time-domain followed by a FFT

I got an $N$, in my case 512, point FFT of a real-valued signal. Based on some calculation in my application I determine the parameters $k \in [1, N-1]$, the number of oscillations per period, $\phi ...
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37 views

Fourier transform of a normalized vector

I was expecting that when I take (for example, in MATLAB), s2 = fft(s1/norm(s1)), the norm of s2 is not equal one, norm(s2) $\neq$ 1. I cannot figure out why this is! Can anyone shed light on this? ...
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39 views

Moving average on the frequency domain - (Welch's Method?)

Is there a name for a filter that is using a moving average on the frequency domain? I am searching for such a filter to filter out long "pads" (synths) from music. Do any of you know about such ...
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1answer
57 views

How Radix-8 and Radix-2^3 are related

How are Radix-8 and Radix-2^3 FFT are related? Are they same? I got some more details in IEEE paper by S.He and M. Torkelson, "Designing pipeline FFT processor for OFDM (de)modulation," in Proc. ...
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37 views

Why the radar (moving) target return is not centered at zero frequency

In Radar signal processing, one has to separate clutter(unwanted signals) from moving target signals. The picture given below is an example of Radar Signal Power Spectum(taken from the book Radar ...
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2answers
62 views

Design Filter Banks for every 100 Hz

I need to study a wave file for every 100 Hz. However, I dont know how can I design filer for each 100 Hz. Let's say 0-100 Hz, 100-200 Hz, ... , ((fs/2)-100) _ (fs/2) Hz. Is it any MATLAB code for ...
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15 views

How distinct should MFCC vectors be?

I'm mid way through creating an embedded speaker verification system and I'm unsure if I'm using/calculating my MFCC vectors correctly. I've hit a brick wall. Essentially given two speakers with ...
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2answers
71 views

Audio compression with STFT

Let STFT[frame, freqbin] be the short-time Fourier transform of some audio data. (here frame is the index in time, and ...
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1answer
47 views

Method for modifying frequency-domain signal to produce predetermined time-domain signal.

We are performing a convolution of audio signal, with 511th order FIR filters, using fast convolution methods: overlap-add or overlap-save. Filters are being designed in frequency domain several ...
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1answer
56 views

Why does this FFT based signal deliver a sinusoidal result?

I have noticed that there is an identical pattern between two signals: one is a signal of n identically spaced samples over a single period of a sinusoid the other is a signal constructed of the ...
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34 views

2D version of Shift Theorem

What is the 2D version of Shift Theorem?
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glitches when performing EQ on an OLA FFT

i'm trying to filter a signal with a FFT. I window each block of signal (i use a 4096 points signal) and then remove some high frequencies by just multiplying the magnitude response. the result is : ...
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59 views

Why is there a difference between MATLAB's FFT of a zero-padded signal compared to the non-padded case

I created a data signal and calculated the spectrum via the fft-command in MATLAB. The data-signal itself is not periodic but it is created in a way that it can be treated as one period of a periodic ...
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163 views

Store a FFT with a minimal amount of data

I have an array x of length 1024 (stored as 16 bits integers, named for example np.int16 in numpy/python), i.e. the size of x is ...
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28 views

Online FFT of simulated signal

it is a couple of weeks that I am thinking about the following: I have a tool which is able to simulate the dynamic behavior a steel tower exhibites due to the coupled action of wind and waves. ...
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1answer
55 views

MFCC, uncertain of my results and algorithm

I've started to implement an MFCC algorithm on an embedded DSP, using the CMSIS DSP library. However the results I'm getting don't appear consistent. This is my algorithm so far: Gather 32ms PCM ...
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33 views

Obtaining an expression for the transfer function when using FFT

Say I have two signals, a(t) and b(t) where the former is the input and the latter in the output. These signals are both recorded by sampling at every 0.01s. The Fast Fourier Transform was applied to ...
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51 views

Why does the inverse fourier transform of a lowpass filter have complex components in matlab?

I am quite confused whether the following numerical differences i find are just severe round off errors made by matlab, or something i am doing wrong. The following happened when trying to see what a ...
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53 views

Comparing an audio signal with a previous 'taught/profiled' signal

I'm working on a audio sample recognition project, where first you profile a sound and then run the sound detection function. Currently during the 'profile' stage I'm grabbing copies of the ...
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1answer
52 views

Detecting a curve with certain characteristics in a spectrogram

I have a spectrogram built by breaking a signal into chunks and taking the FFT of these chunks. An example spectrogram is: In this spectrogram I am looking for a curve (aka pitch contour) that: ...
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68 views

A question about spectral leakage

I have read about spectral leakages in other posts here. From what i understand, it occurs because you dont have an integer number of time periods in your sampled data. By default a rectangular window ...
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2answers
133 views

Trying to understand downsampling and then upsampling

I am trying to understand how this works, specifically, what the DTFT of each step looks like in each step of the chain (for understanding). I am not looking for an answer like because the input ...
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2answers
68 views

Problems calculating 8-point FFT of an 8-point sine wave by hand

I need a small help for my assignment, which is to solve the 8-point FFT without using Matlab or C. Preliminary information: 8-point sine wave: [0, 0.7071, 1, 0.7071, 0, -0.7071, -1, -0.7071] ...
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1answer
69 views

understanding complex fft results

i use this for complex fft. Output expected $fft[3].real= 32$ (peak at 3rd bin) $fft[61].real= 32$ ((peak at negative frequency pair of 3rd bin)) All other values negligibly small The input is ...
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91 views

How does amplitude scale when FFT size is smaller then signal duration?

I have one quite simple question: how does the amplitude of the signal scale when we do FFT where FFT window length is shorter then duration of the signal? Does this means that if FFT size is 12 time ...
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88 views

How to get the 10-5 most dominant frequency in an audio signal

I have this wav file that I recorded using my laptop, It contains the sound of a G chord of a guitar. Since the guitar have 6 strings the sound should contain different frequencies I want to determine ...
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53 views

possibilty of extracting a tone from MIDI values

I have a project of recording human's sound via Android's SoundRecoder API, and then displaying the extracted note on The screen. It is specific for Android platform (can't use Jmusic or Jfugue API as ...
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155 views

problems calculating phase propagation (in C)

I'm coding a phase vocoder in C for time-stretching audio signals. The only piece that is currently not working is the phase propagation. I take overlapping frames from an input signal, apply a ...
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2answers
134 views

Measure the SNR of a signal in the frequency domain?

How do you measure the SNR of a sinewave in the frequency domain? (Assuming no filtering.) Suppose: $$x(n) = s(n) + n(n)$$ Create the signal (nCycles of a sinewave) in MATLAB...: ...
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98 views

Fourier Transform for frequency analysis

I applied Fourier transform on a sound (PCM generated digitally with sin function) and frequency detection is accurate. However, if I play the sound and record it, and then apply a Fourier transform ...
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4answers
305 views

How do I compare chords played on a guitar with a known sample?

I want to create a project that will recognized the chords that i played in guitar. I recorded a raw sound to be compare to the one that the user will input. Example I have a recorded sound chordA ...
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2answers
112 views

FFT plot in Matlab, accuracy

I have a small input signal of 60Hz sine wave from signal generator, which is corrupted with 50Hz mains supply frequency. I want to measure the amplitude of the 60Hz signal using FFT because it is ...
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46 views

FFT of complex envelope

I have I and Q samples of a time domain signal, sampled at a rate T, both of length L. I create the complex envelope, C = sampleI+j*sampleQ. Then take the FFT of ...
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126 views

SPL values from FFT of microphone signal

I am trying to get a frequency response vs dB SPL plot for recorded sounds. I have a microphone with a flat response, a sensitivity of 4mV/Pa, and 1 Pa = 94dB. To get the dB SPL, I have the following ...
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44 views

Audio Steganography using Phase Encoding Technique

I referred the below papers to understand the working of Phase Encoding,but the explanation is not proper.Just a few steps have been given in brief. ijcsi.org www.ijarcce.com Can someone please ...
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1answer
47 views

filtering fft spectrum further?

I am working on bearing faults.I did the high pass filter and then calculated the envelope of the signal in the time domain. I then found its fourier. The fft spectrum is still not clear. My ...