fourier-transform | Discrete Fourier Transform in OpenCV and JavaFX
kandi X-RAY | fourier-transform Summary
kandi X-RAY | fourier-transform Summary
A project, made in Eclipse (Neon), for experimenting with the Discrete Fourier transform (and its inverse), starting from two grayscale images: a circle and the sin function. Example images are provided in the images folder. Please, note that the project is an Eclipse project, made for teaching purposes. Before using it, you need to install the OpenCV library (version 3.x) and JavaFX 8 and create a User Library named opencv that links to the OpenCV jar and native libraries. A guide for getting started with OpenCV and Java is available at
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Top functions reviewed by kandi - BETA
- Loads an image from disk
- Convert a mat object to a buffered image
- Converts a Mat object into an Image
- Places a property on the JavaFX thread on the JavaFX thread
- Updates an ImageView
- Action called when the button is applied
- Adjust the magnitude of the quaternion to re - order
- Optimizes the magnitude of the complex image
- Optimize the image
- Starts the Fourier process
- Initialize the variables
- Set the current stage
- Action called when a matrix is applied
- Starts the OpenCV library
fourier-transform Key Features
fourier-transform Examples and Code Snippets
Community Discussions
Trending Discussions on fourier-transform
QUESTION
I am trying to plot a fourier transform of a sign wave based on the scipy documentation
...ANSWER
Answered 2021-Jan-11 at 04:24import numpy as np
import matplotlib.pyplot as plt
import scipy.fft
def sinWav(amp, freq, time, phase=0):
return amp * np.sin(2 * np.pi * (freq * time - phase))
def plotFFT(f, speriod, time):
"""Plots a fast fourier transform
Args:
f (np.arr): A signal wave
speriod (int): Number of samples per second
time ([type]): total seconds in wave
"""
N = speriod * time
# sample spacing
T = 1.0 / 800.0
x = np.linspace(0.0, N*T, N, endpoint=False)
yf = scipy.fft.fft(f)
xf = scipy.fft.fftfreq(N, T)[:N//2]
amplitudes = 1/speriod* np.abs(yf[:N//2])
plt.plot(xf, amplitudes)
plt.grid()
plt.xlim([1,3])
plt.show()
speriod = 800
time = {
0: np.arange(0, 4, 1/speriod),
1: np.arange(4, 8, 1/speriod),
2: np.arange(8, 12, 1/speriod)
}
signal = np.concatenate([
sinWav(amp=0.25, freq=2, time=time[0]),
sinWav(amp=1, freq=2, time=time[1]),
sinWav(amp=0.5, freq=2, time=time[2])
]) # generate signal
plotFFT(signal, speriod, 12)
QUESTION
I was following the tutorial on this website to create a low-pass filter using NumPy and OpenCV in python. Instead of displaying the transformed image using pyplot as shown in the tutorial, I tried using OpenCV's imshow function but I am not getting the desired output.
...ANSWER
Answered 2020-Dec-19 at 12:20If you want to display data in type np.float32 you have to normalize it to 1.0
QUESTION
I am looking to implement KL Divergence in C++ efficiently. (CPU Only for now).
Much like AES or FTT (Fast Fourier transform) whereby use of a common function has lead to hardware level optimizations (Intel AES and Intel FTT). Is there anything similar for natural log, or slightly higher level efficiencies (ASM/C) that prevent bottlenecks of executing many Natural log functions in success (If they exist)?
Same use case examples:
...ANSWER
Answered 2020-Oct-10 at 21:28You can use SSE instructions
to calculate the logarithm of many values in parallel. But whether you can actually make use of those instructions depends heavily on how the rest of the calculations you are going to do depend on the logarithms you calculate, so it is not possible to give a more specific answer.
QUESTION
So I am making an AI project that classifies speech into either "up", "down", "left", right or background noise, and from this, a character in a videogame is moved.
I have made an FFT algorithm deriving it from the mathematical explanation, which I believe is correct as I have tested its output against that from this site (https://engineering.icalculator.info/discrete-fourier-transform-calculator.html)
I then have tried to generate a spectrogram and have used code based on the code from the main function of the App class from this site (Creating spectrogram from .wav using FFT in java)
I tested my code on a .wav file of me saying hello and the spectrogram generated is not what I was expecting out, see below the difference between my java made spectrogram and my python made spectrogram (ignore the colour difference).
New Java Spectrogram with SleuthEyes help
Here is the original code I have used/written:
...ANSWER
Answered 2020-Jul-24 at 10:47Reading the .wav file
The .wav file decoding included in that other question you linked is hardly a full blown decoder. It accounts for the OP's specific stereo 2bytes-per-sample use-case.
It looks like you stumbled upon other decoding issues while trying to adapt it to a different use case. As a general piece of advice, I'd suggest to use a more complete .wav decoder which would take into account the number of channels, the number of bytes-per-sample, etc.
If on the other hand you want to craft your own decoder (for example as a learning exercise), then a slightly more robust implementation may look like the following:
QUESTION
When I am using Intel IPP's ippsFFTFwd_RToCCS_64f and then ippsMagnitude_64fc I get a massive peak at zero index in magnitudes array.
My sine wave is long and main component I am interested is somewhere between 0.15 Hz and 0.25 Hz. I take the sample with 500Hz sampling frequency. If I reduce mean from the signal before FFT I get really small zero component not that peak anymore. Below is a pic of magnitudes array head:
Also the magnitude scaling seems to be 10 times the magnitude I see in the time series of the signal e.g. if amplitude is 29 in magnitudes it is 290.
I Am not sure why this is so and my question is 1. Do I really need to address the zero index peak with mean reduction and 2. Where does this scale of 10 come?
...ANSWER
Answered 2020-Jul-23 at 02:17Do I really need to address the zero index peak with mean reduction?
For low frequency signal analysis a small bias can really interfere (especially due to spectral leakage). For sake of illustration, consider the following low-frequency signal tone
and another one with a constant bias tone_with_bias
:
QUESTION
Source code
...ANSWER
Answered 2020-Jul-07 at 16:09You need to use reshape
:
QUESTION
I am actually want to convert this blur detection into C++. As a beginner in OpenCV, I am actually following this for conversion, But maybe I am getting it wrong. Here is my approach. I have to use DFT instead of FFT in C++.
...ANSWER
Answered 2020-Jun-29 at 12:43I think this comes close to the original Python code
QUESTION
I am receiving data from a vibration sensor in the form of two numpy arrays. The first array represents the actual values for the vibration measurement and the second array is the corresponding time information (timestamp). For example:
...ANSWER
Answered 2020-Jun-19 at 16:45This is really a signal processing question, not a Python one. You have several options here:
- if your data is uniformly sampled - you can ignore the timestamps altogether. All the information you need is in the data, and the (constant) sampling frequency: f_s = 1.0 / (timestamps[1] - timestamps[0])
- if not, you can either:
- use Non-uniform DFT (here is one implementation, haven't tried)
- interpolate the data between non-uniform timestamps so it becomes uniform. Note that effectively, this applies a low-pass filter to your data, which may be not what you want (more on the effects of interpolation here).
In all cases, when you perform FFT, time information is not required anymore, as you are in the frequency domain.
QUESTION
I'm writing an audio plugin and I would like to convert a series of real numbers representing sample values into a complex array representing frequency and phase. Then I want to be able to do the opposite, turning a complex array of frequency and phases to a series of real numbers, reconstructing the original data.
I'm using Intel MKL and I see only the possibility to perform real->real conversions or complex->complex conversions. Here's the reference I'm using: Intel MKL FFT Functions.
In that reference, there are two overloaded functions: DftiComputeForward and DftiComputeBackward. So, I would like to use these to do the conversions. In the reference for DftiCreateDescriptor
, the only options available for DFTI_FORWARD_DOMAIN
are DFTI_COMPLEX
and DFTI_REAL
, but no option for mixed conversions.
Edit
I found that the phase is actually equal to atan(imaginary/real)
. I don't want to mislead anyone getting information from questions.
Edit
I just learned that it's best to use atan2(imaginary,real)
. More information is in the comments.
ANSWER
Answered 2020-Jun-07 at 15:50Every real number is a complex number: ℝ ⊂ ℤ. So going forward from float
or double
in time domain to complex
is trivial. The language does that for you!
QUESTION
I have a pandas data frame with one column containing lists. I wish to divide each list element in each row by a scalar value in another column. In the following example, I wish to divide each element in a by b:
...ANSWER
Answered 2020-May-28 at 09:03Community Discussions, Code Snippets contain sources that include Stack Exchange Network
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Install fourier-transform
You can use fourier-transform like any standard Java library. Please include the the jar files in your classpath. You can also use any IDE and you can run and debug the fourier-transform component as you would do with any other Java program. Best practice is to use a build tool that supports dependency management such as Maven or Gradle. For Maven installation, please refer maven.apache.org. For Gradle installation, please refer gradle.org .
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