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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1answer
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Finding Peaks in an Autocorrelation Function

I'm trying to find the period of a signal. I've used FFT to compute the autocorrelation of the signal. As can be seen from the autocorrelation function (plotted below) I obtained, there are 70 sample ...
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1answer
3k views

Filtering PCM data in java

I'm working on an audio based application for android platform , I'm trying to use a low pass filter to a fixed or variable max frequency "depends on the cost of development". first I've been playing ...
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793 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 ...
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FFT of High Frequency and Low Frequency?

When we plot FFT in a matlab it doesn't ask for Frequency , all we need to input the sequence and it simply shows the output using stem command. If i want to compute two different signal using FFT, ...
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6k views

breaking the input signal into segments while calculating DFT

Im reading a book about DSP and there is an example about investigating the sound that travel through the ocean. The sound was recorded as the time-domain signal and they tried to analyze it in ...
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62 views

how to break the input signal into segments while calculating DFT [duplicate]

Possible Duplicate: breaking the input signal into segments while calculating DFT Im reading a book about DSP and there is an example about investigating the sound that travel through the ocean. ...
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2answers
295 views

How does Fourier Transform manipulates image technically?

I need a little bit more explanation on technical details regarding Fourier Transform in Image Processing, mainly on the frequency aspect. I see that we can use Discrete Fourier Transform to change ...
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1answer
316 views

Can FFT Substitute Correlation for This Case?

I'm trying to determine frequency of a repeating pattern in a time series. The sampling rate 100Hz and the pattern repeats itself with periods between 0.5 second and 1.25 second. This corresponds to a ...
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3answers
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Speech Recognition - Project Idea

Wondering if someone could offer me any advice or help. Sorry if this question is too broad or anything. I'm wanting to complete an algorithm in basic Speech analysis. The algorithm will detect ...
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1answer
691 views

Demodulation of BFSK signal with frequency hopping using Goertzel Algorithm

I am implementing a BFSK algorithm with frequency hopping on an embedded system and I am using the Goertezel algorithm in my demodulation approach. Basically, I use the goertzel alorithm to determine ...
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2answers
868 views

change the frequency component of a spectrum in an audio file

I am working on an audio dynamic range compressor using Matlab. After calculating the FFT of the audio file, I use several bandpass filters to amplify specific bands separately. What I want to do then ...
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1answer
603 views

My vocoder sounds bad

I'm writing a program to emulate a vocoder and it obviously works, but any speech that comes through is very difficult to understand. I'm going to explain exactly how my implementation works and then ...
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Time resolution of the Short Time Fourier Transform (STFT)

I'm struggling to figure out how the time points of an STFT are calculated, and I can't find a definitive answer. Let's say I have a 4Hz stationary signal and I'm going to use a 64 second window with ...
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2answers
227 views

Why this two ways of FFT are different?

I have a small question that I am confused. N=16; x=[-pi:2*pi/N:pi-2*pi/N]; y=1.5*cos(x) z=fft(y)/N we will get negtive Fourier coefficients, however when we do ...
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3answers
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Demodulation of a BFSK frequency hopping signal using FFT

I am not very familiar with DSP techniques, and I am currently working on my research project to develop the firmware of a DSP processor, using C/Linux. It involves the modulation and demodulation of ...
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1answer
291 views

How to calculate the gain in a bivariate fft in R?

In Statistica gain is defined as follows: Gain. The gain value is computed by dividing the cross-amplitude value by the spectrum density estimates for one of the two series in the analysis. ...
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Can edge detection be done in the frequency domain?

Can we take advantage of the fact that high frequency components in the FFT of an image generally correspond to edges, to implement an edge detection algorithm in the fourier domain? I did try ...
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4answers
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Can FFT be used to find peaks and pits on audio files

I'm able to read wav files and its values. I need to find peaks and pits positions and their values. First time, I tried to smooth it by $\frac{(i-1 + i + i + 1)}{3} $ formula, then searching on ...
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0answers
684 views

Get bit stream from known radio frequency

I'm a beginner in radio demodulation, and several days can't solve my problem. The problem is to get bit stream from known frequency. I made I C# application that gets the data in real time from ...
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1answer
1k views

Analysing 2500 frequencies using FFT with an input vector of 2048 samples?

I am currently reading the paper A Highly Robust Audio Fingerprinting System and on page 4 one can read about the technical parameters they use: Sampling rate of 5000 Hz, frames of 2048 samples as ...
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3answers
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FFT for line detection

I am trying use the FFT in a different way then most people ask about. I want to be able to take a picture of a graph with regular repeating vertical lines, and to process the image to determine how ...
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1answer
2k views

Audio processing FFT of chunks of audio

Im new to Matlab. I need to split the time dimensions of an audio track into frames of 10mS and for every 10mS frame have to compute the FFT to generate the spectrograph. I need help on this.
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Do FFT-based filtering methods add intrinsic latency to a real time algorithm?

In a current audio processing application, I am working entirely in the time domain using IIR filters. This is so I can use an audio buffer of just 3 or 4 samples and can guarantee that there will be ...
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3answers
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Fast Fourier transform- non-integer number of cycles in the FFT aperture

There are a few excellent discussion threads and answers on this site (eletronics.se) on the theory of Fourier transforms. I tried implementing the same in a simulation tool (MS Excel :)). I have a ...
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2answers
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How to do prediction using frequency domain data?

Both linear regression and Kalman filtering can be used to estimate and then predict from a time domain sequence of data (given some assumptions about the model behind the data). What methods, if ...
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2answers
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Generating wavelet using amplitude spectra

Equation (4) of a recent paper (An improved peak-frequency-shift method for Q estimation) shows how to generate a wavelet using a zero-phase Ricker wavelet. $|B_1(f)| = |B_0(f)|\exp(-\pi t f / Q)$ ...
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What is a good FFT window function to reject DC?

I'm using an FFT to analyze what is essentially the power envelope of a signal (see here for info on the containing project), and, since power numbers are always positive, to eliminate the DC ...
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1answer
263 views

How Can I practically implement the FFT with size unequal to a power of two?

Can anybody explain how I can implement the Fast Fourier Transform (FFT) of length unequal to a power of two in hardware, e.g. FPGA? What is the algorithm for this implementation? Does anybody know ...
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766 views

Is it valid to increase amplitude (and presumably FFT quality) by simply scaling the data?

I'm using a version of "KISS FFT" by Mark Borgerding. It accepts an array of 16-bit fixed-point input values and produces a 32-bit float result array. I've discovered that if the input amplitudes ...
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1answer
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low pass filter and FFT for beginners with Python

I am new to signal processing and especially to FFT, hence I am not sure if I am doing the correct thing here and I am a bit confused with the result. I have a discrete real function (measurement ...
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2answers
19k views

Improving spectrogram resolution in Python?

I'm using the specgram() function in matplotlib to generate spectrograms of speech wave files in Python, but the output is ...
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3answers
578 views

FFT on non-rectangular part of image

I need to do a FFT on an image for noise reduction, but the problem is that I do not need the complete image, but only a circle in the middle. The borders are a fixed rig, thus I am not interested in ...
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3answers
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I need advice about how to make an audio frequency analyzer

I just need some clarification of some very basic aspects of creating a frequency analyzer. I am writing a c++ program, so far it is just a media player, with a custom explorer. You can make ...
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1answer
5k views

How to get coefficients for sine/cosine function from complex FFT?

I'm working on a control system that measures the movement of a vibrating robot arm. Because there is some deadtime, I need to look into the future of the somewhat noisy signal. My idea was to use the ...
5
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1answer
1k views

FFT window phase syncronising

I'm currently devising a solution for demodulating a DPSK signal in C#. So far I've got a working FFT with phase detection, BUT: I now have an issue of aligning the window of the FFT to synchronise ...
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3answers
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What's the best criterion for determining a frequency peak?

I'm writing an algorithm that analyzes results of discrete Fourier transform (DFT). The algorithm should detect amplitude peaks in predefined ranges of frequencies. This is an example of such a peak: ...
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2answers
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Does down-mixing a signal before taking the fourier transform aid in signal detection?

Let us assume that we have a narrowband signal $s[n]$, at frequency $f_1$, and let us further assume that this $f_1$ is just below $\frac{f_s}{2}$, where $f_s$ is the sampling rate. Furthermore, let ...
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154 views

Help me understand this simple code snippet to get the magnitude frequency response

I inherited the following FFT code to get the magnitude frequency response of a sample window. I understand what fft() does. What I don't understand are the lines marked with a "???". What do they do? ...
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2answers
538 views

Embedding digital ID in audio signal, and retrieving it

The problem I am working on an interesting challenge -- to encode a 32 bit ID blip in an audio stream, and subsequently retrieve it. Something like this: http://gizmodo.com/5883215/why-lady-gaga-...
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1answer
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Find similar Music using FFT Spectrums

I have been experimenting some weeks to find a way to match/find similar Songs in a Library containing different Genres of Music. My first try was to Detect Features like Tempo or how much Bass there ...
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1answer
484 views

From a 44kHz audio signal, how could I extract 32 evenly spaced frequency bands?

Assume I have an array of floating points that contain the last 8192 samples played. How would I convert that array to 32 frequency bins that are spaced from 0Hz - Nqyst?
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1answer
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Can I use FFT to interpret accelerometer gestures?

I have the need to detect two different gestures that occur when looking at accelerometer data. Here's a run down (as brief as I can make it): Lets say an iPhone is being oscillated back and forth ...
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1answer
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FFT of image data: “mirroring” to avoid boundary effects

I load and display an image of some rice in Matlab: g = imread('rice.png'); imshow(g); I take the FFT of this image and shift it: ...
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2answers
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How can I implement a binary chirp signal containing zeros and ones?

I build a signal, $x$, which is linear in the frequency domain as (in MATLAB syntax) x = 1:0.1:1e3; Then I do: ...
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1answer
6k views

How do you plot an SPL vs frequency curve for a given set of pressure data?

I have a set of pressure data at a sampling rate of 262144 Hz for 3 seconds making it 786432 samples of pressure data. I want to plot a curve of the SPL (sound pressure level) vs frequency. I first ...
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1answer
2k views

How to estimate the signal-to-noise ratio of a waveform?

I have a signal: $f_i(t_i=i\Delta t)$, where $i = 0\ldots n-1$. The signal seems to vary quickly around a slower varying "trend". I am assuming that the quickly varying part is noise and the slowly ...
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1answer
214 views

Is there a “first” Fourier transform?

Every time I've read "first Fourier transform", I've assumed it to be a misspelling of the fast Fourier transform. Recently though, I saw the term in both the title and keywords of an article from a ...
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What data should I use to test an FFT implementation, and what accuracy should I expect?

I'm involved with an effort to implement an FFT algorithm, and am curious what the recommended advice is for the input test data to use -- and why! -- and what accuracy to expect. On test inputs, I'...
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3answers
609 views

How to efficiently calculate only the low coefficients of a zero-padded FFT

I've got an algorithm that zero pads a sequence to 4N, does an FFT, and only uses the lowest frequency N points out of the generated 4N. This seems like a lot of wasted work, any ideas how this can ...
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2answers
273 views

Is there any way to make a sampling rate reduction in the frequency domain?

Is there any way to take advantage of the frequency domain to make a clean/nice sampling rate reduction effect ? and how ? Basically, deducted from my tests: if I set to 0 the magnitude uppon a ...