instruction stringlengths 31 27k |
|---|
I need to alter an FIR filter which I know very little about. I have a signal which need to be altered, and looking at a Python script used for this task, I see that it uses Scipy's signal library, more specifically the firwin function.
If I input
cutoff_freq = 5400
num_taps = 51
coeffs = signal.firwin(numtaps = num... |
What filters are used to get an image like one on the left? I need to find the body pose of a person in a picture.
|
We know that that the difference equation formula for computing an output sample at time $ n$ is based upon its past/present input samples and past output samples in time domain:
$$
\begin{aligned}
y(n) &= b_0 \,x(n) + b_1 \,x(n - 1) + \cdots + b_M \,x(n - M) \\
&\qquad- a_1 \,y(n - 1) - \cdots - a_N \,y(n - N) \\ ... |
I am conducting experiments to collect wind speed data from wind anemometers placed on a moving platform. Closely, there is a fixed wind mast holding a wind vane. Prior to analysing the wind data from the moving platform, I binned the data with 1-minute average wind direction. Each bin contains a number of continuous d... |
Any idea on separating multi-panel figures?
There are 2 different sub-figures connected each other.
Thanks!
|
I have a source of noise in a duct, on either side of the noise source I have a microphone, which register the pressure, so that I get two pressure signals: $p_1$ and $p_2$. I want to get the acoustic Sound Pressure Level (SPL) spectrum which can be estimated using a periodogram approach, more specifically using the we... |
I'm trying to understand how speech recognition works, when using HMM. I know this post is very long, but that's mainly because I'm a complete beginner, and I since I have no idea what is clear and what is not, I try to write everything as clearly as possible (which takes a lot of space). Help would be really appreciat... |
In their textbook(Example 1.1), Oppenheim and Willsky provide the following method for obtaining $g(t)=x(-t+1)$ from $x(t)$ : Replace $t$ with $-t$ in $x(t+1)$. In other words, advance the signal by 1 unit and flip the result.
But there could be another interpretation as well. Based on the fact that $-t+1 = -(t-1)$, we... |
I am working on something like cartoon, but the character's mouth shape is automatically detected and matched by the word they are saying.
my guess is to use speech recognize algorithm to analyze the wav file, and get the content(word) at the exactly moment, for the exactly word, prepare corresponding mouth shape pict... |
I have a signal $x$ which is corrupted by Gaussian noise.
The signal $x$ is convolved with a filter with coefficients $h = [1, -2, 1]$, to generate signal $y$. Now the signal is taken to Discrete Fourier Transform (DFT) domain by FFT.
In MATLAB,
y = conv(x,h);
z = fft(y);
How can I compensate for the modulation caused... |
It seems there are several papers from the seventies but backtracking from the references gets quickly difficult. Who calculated for the first time a time-frequency representation of a signal?
|
I have two iron pipes, one bent and cracked, and one whole. On one end Ive attached a vibration motor (200hz) and on the other end Ive attached a vibration sensor.
I do a fft and then get out the amplitudes of the different frequencies I pick up.
Question: I want to compare the signal picked up from the broken pipe to ... |
I'm filtering a real signal with a single complex pole with a complex coefficient (a_re [real part] and a_im [imaginary part]), I also have a gain coefficient but I'm gonna leave it out for the sake of simplicity. The filter has two outputs, one for the real and another for the imaginary part. Here's the filter equatio... |
As explained in here:
Gold sequences help generate more sequences out of a pair of
m-sequences giving now many more different sequences to have multiple
users. Gold sequences are based on preferred pairs m-sequences.
For example, take the polynomials $x^5+x^2+1$ and
$x^5+x^4+x^3+x^2+x+1$:
By combining two of ... |
I'm analysing the frequencies of something I'm recording (whistling), which is resulting like this (yes, it's super mario bros' beginning song:
I'd like to filter it to make it remain like:
in order to convert them as MIDI notes after the processing.
How could I do it ?
I've tried with a simple low-pass filter but it... |
I created a sinusoidal wave in some noise, and plotted the power spectrum of the signal using two periodogram estimates (welch procedure). One estimate is 'high resolution' - i.e. it uses a longer window than the corresponding 'low resolution' estimate. The result and code used to generate it are shown below. For refer... |
I don't understand how this works. Help would be really appreciated.
Let's say we have a 10 second sound input, we make a feature vector every 10ms - so we have 1000 vectors - $\mathbf{o}=o_1, \cdots,o_{1000}$.
For simplicity, let's say that all we want to know, is what sequence of phones $p=p_1,\cdots,p_n$ is the answ... |
do any of you know how to convert a FM index to a phase mod index equivalent considering they both have the same carrier and modulator frequency? Is it even possible?
This is in respect for sound synthesis and I'm talking about one sinusoid modulating another sinusoidal oscillator.
I'm being able to get very similar so... |
One way to describe a practical integrator ("leaky integrator") is
$$
H(s) = \frac{g R}{1 + sRC}
$$
I am trying to understand how to choose the values $g$, $R$ and $C$ because in practice, I will have constraints on them.
Here is my attempt: Suppose for a moment, that $R \rightarrow \infty$, then it becomes an ideal in... |
I am confused about the need to change color space for color comparison. I have read about delta E, the Lab format, and I do understand that comparisons in the RGB color space will not seem appropriate to the human eye. However, my program uses a linear color scale to calculate velocity, from a color flow Doppler signa... |
A) What are the most common methods used to remap data for current generation imaging radars and acoustic sensors?
B) What are some of the pitfalls in these methods?
For example, in the past, at least, cubic splines have been used to map slant-range image data from synthetic aperture radar to a uniform "ground" grid. ... |
I want to find points (of a processed image) that are the brightest in their local region.
Basically, I want all of the points whose 8 neighbors are all smaller but I want to have brighter maxima block out nearby dimmer maxima. So, for example, the 8-neighbor maximas could be sorted by descending brightness then yield... |
Correct me please if I am wrong...
I have just started working in the domain of signal processing using the Fourier Transform ($\mathbf{FFT}$) and Power Spectral Density ($\mathbf{PSD}$). I am trying to adjust the Frequency Response Function ($\mathbf{FRF}$) so that response $Y$ computed using relation:
$$\mathbf{FFT}(... |
Ok, I'm new to the whole Dsp thing, so sorry if the question is a lame one
I'm trying to extract frequencies from a signal provided by a radio Device (Win Radio G31DDC- IFstream).
The radio provides a 16-bit signal stream with 50MHz bandwidth and 100MHz sample rate.
when I set my FFT length to 32768 everything seems t... |
In Welch or Bartlett method, original signal is divided into L segments, of D overlapping (D=0 in Bartlett method). Then FFT is done for each segment, and the result is the average of all segments' FFTs. Averaging reduces the variance. But apart from this effect, is there anything else to take into account when choosin... |
I am currently reading Chapter 5, Applications to the Gas Markets, in Stochastic Modelling of Electricity and Related Markets by Benth, Benth and Koekebakker, World Scientific, 2008.
In the subchapter 5.3.1 Kalman filtering they assume that the detrended and deseasonalized logarithmic spot price dynamics for gas follow... |
My Matlab FFT plot keeps showing the mirrored values of my frequency of interest. In this example I am looking for 250Hz with a 3200Hz sampling freq. The FFT plots my value at 266Hz as well as 1330 Hz. I assume my sampling freq is a little off which is why it isn't exactly at 250Hz. Matlab code is below, any help is... |
Let's talk about a standard saturation model:
with bounds $bound_{lower}$ and $bound_{upper}$. So in this example the output $y$ will be:
\begin{equation}
y(u) =
\begin{cases}
bound_{lower} & u < bound_{lower} \\
u & bound_{lower} \leqslant u \leqslant bound_{upper} \\
bound_{upper... |
I have 20 audio recordings (raw format) of people reciting the same ~50 word transcript. I would like to use audio signal analysis to decode speaker gender from audio recording.
Intuitively, it seems I need to align the audio signals and then cluster them. Assuming this is correct,
For alignment, would you recommend ... |
I'm making an interpolator (filter) to remove aliasing but this doesn't work.
I see also some strange behaviour: x[i - j], but what if j > i? I simply followed FIR definition but still don't understand why this doesn't work.
const size_t M = 200;
double h[M + 1]; // h is filled with coeff
inline void process(double* x... |
I am reading a document on Image processing and couldn't analyze a statement in it.
For any $k \in \Psi_s$, the family of values $\{u[\varphi(p+l)+k-l]:l \in \Psi_s \}$ is
representative of the value of...
The variables are explained below
$k$,$p$,$l$ - pixels
$u()$ - image intensity
$\Psi_s$ - patch of s... |
Consider a sine wave having an amplitude of 9 and a frequency of 0.3 Hz. I wish to perform an fft and plot the frequency on the x-axis and the real amplitude (i.e. 9) on the y-axis. I am using the following MATLAB code:
Fs = 10;
data = xlsread('Signal.xlsx');
Y(1)=0;
Y = fft(data);
N = length(Y);
freq = 0:Fs/N:Fs/2-Fs/... |
I have the following signal for which amplitude and phase spectrums have to be computed:
This exercise also has a solution which begins by deriving the signal twice.
Next, they say the spectral density function for the given signal is:
$$X(j\omega)=\dfrac{X''(j\omega)}{(j\omega)^2} $$
In my attempt to figure out how ... |
I've just one question :
How can I write a model like y = x + w,( with w a white gaussian noise) with a fixed SNR and a noise variance equal to 1. What coefficient may I have before x ?
Thanks !
|
I used the continuous wavelet transform (CWT) to analyze the behavior of a 1-D smooth signal (extracting its modal property, damping and frequency components) in my research. Now someone asked:
Why the presented CWT algorithm is better than Ridgelet transform?
I thought it is better to ask in this forum about disadva... |
In a Wifi 802.11n context, I'm looking to find the sampling rate to perform some processing on the signal and I've something I cannot understand : in the book Next Generation Wireless LAN's: 802.11n and 802.11ac, I would like to know the meaning of this sentence :
In 802.11a, the fundamental sampling rate is 20 MHz, w... |
I wonder is there any algorithm or method that I can map two RGB images together in Matlab. I mean assume I have following images
Is there any way I find what number I should multiply each channel of one them to get the other one, or any other method.
I have tried something like: looking at destination image and see... |
i have a measured signal (e.g. voltage sine wave) and i would like to generate a sine wave which matches this meausred signal (with the generated sine wave i remove the harmonics from my measured signal), just like a PLL block in simulink.
for that i have to do the 2 steps:
1- Calculate the frequency of the measured si... |
I'm searching for a an algorithm or an idea of implementation. I need to filter a signal in real time, and want to change the length of it (number of taps) in order to change the delay.
In low signal environment I want a high order filter and care less
about delay
In high signal environment I want a low order filter i... |
I have a signal and from 'standard information' I know the predominant frequency is at $1\textrm{ Hz}$ and then there should be three harmonics at lower amplitudes. My signal however has a lower amplitude at $1\textrm{ Hz}$, higher at the first harmonic and then several harmonics, more than what should be present.
I a... |
Assume signal $x$, sampled at $f_s = 44100\; \mathrm{Hz}$. I tried to filter it using the Butterworth bandpass filter ( $30\; \mathrm{Hz} - 70\; \mathrm{Hz}$) of order $8$. However, as a result I get a vector with most elements being NaN (and some of them extremely small, approx. $-2.5 \cdot 10^{306}$`).
If I try the ... |
I'm trying to resample a digital communication signal in SciPy or Matlab. Suppose the signal was sampled at 3 samples per second, and I want to resample it to 2 samples per second, the thing is I also want to be able to change the new sampling point so the new samples doesn't necessarily start at the same point as the ... |
SNR is defined as:
$$SNR=10 \log_{10}\left(\frac{\sigma_s^2}{\sigma_n^2}\right)$$
For instance, when we say the SNR of 10 dB is 10 times higher than that of 20 dB, it is higher in what unit? Amplitude, power or what?
Also, could you relate as to why it's 10 times higher for the above example.
|
Gaussian noise with different SNR levels are usually used in research works to simulate a realistic environment. How can researchers guarantee that Gaussian noise can simulate the reality of a System?
|
I have implemented a python butterworth filter but it seems to shift the data a lot, the x location of my data is very important to me for my analysis. Is there a way to correct for this shift precisely?
Does it shift all the data uniformly? Or does certain areas get shifted further/less far that others?
|
I have written a piece of code in MATLAB to generate Frank codes - Pastebin link for pulse compression.
I know that Frank codes are stepped frequency approximations to Linear Frequency modulation(LFM).
LFM Parameters are obtained here:
PW = 100e-6;% Pulse Width
PCR = 100; % Pulse Compression Ratio
CPW = PW/PCR;% Compre... |
I have two approximately sine truncated waveforms of different amplitudes and I want a numerical measure of how similar they are to each other. I am sending a sonic pulse train whose multiple reflections I am receiving. I want to know which of those reflections most closely correlate with each other in terms of wave sh... |
1)Suppose that we have low-pass filter with cf equal to 0.1. How frequency response would look like if I multiply filter kernel by sine with frequency say 0.3? How this is related to converting
low-pass filter algorithm into bandpass filter?
2)What is the difference between multiplying filter kernel by sine and cos... |
I have a multispectral satellite image and I would like to "transform" each of the image bands into one with integer values. For example to stretch? the image between 0 and 255. Not sure how to proceed with this. I also thought I could make a histogram and replace each pixel value with the bin edge in which it falls in... |
In my research area, instantaneous power in a specific frequency band is commonly estimated by the following procedure:
Apply a bandpass filter on the raw signal (e.g. 80-90Hz bandpass).
Estimate the analytical signal by DFT (in Matlab: hilbert())
Take the absolute of the result.
I'm wondering how far this procedure... |
Some posts evocate the computation of convolution or cross-correlation using FFT and zero-padding the temporal signal.
I want to compute the autocorrelation of a 3D array using FFTW (2 dimensions are periodic). The file is very large (dimension 256*256*49995, double precision, 25GB).
1 : Is padding necessary for period... |
I am writing a lens correctionion code and the process should do several things:
Inverse fisheye projection to obtain rectilinear image
Barrel/pincushion correction
Vignetting correction
Lateral chromatic aberration correction
The problem is - in what order apply these steps?
Should the image be first "de-fished" or ... |
Suppose I have an input signal which represents a vehicle front suspension response that has been measured using an accelerometer. It is seen that the measured rear suspension response is effectively just a time delayed version of the front suspension response and shifting the rear signal forward in time by the delay c... |
I want to find the orientation of 2x2 checkerboard cubes, like these:
That is to say, I want to recover their positions and rotations within the image relative to the camera.
In my particular case, I care more about the rotation than the position (and the depth barely matters at all). I know the colors ahead of time, ... |
Does anyone know what the heck is a coupled-air microphone? I know that they were used for recording internal body sounds in papers like this and this. I googled the term, but I couldn't find any circuitry schema or place where to buy one. Are they some kind of diaphragm, condenser or contact microphone?
Any help would... |
I wanna experiment k-means clustering method on different kind of images, so I am trying to find different kind of images used in image segmentation such as MRI images.I want to gather some more categories.
Any suggestion would be gratefully appreciated.
|
There is a lot of tools for designing FIR filters. Using one such tool I have designed a filter that satisfies my needs. It's coefficients are as follows:
0.000106519966569472
0.000164942422133671
0.000154198611135164
-0.000051344150552581
-0.000527514219352601
-0.001228193882673074
...
0.176493072494376790
...
... |
I'm working on creating a simple program to render spectrograms like this one. In this plot, the X-axis is time, the Y-axis is frequency, and the color represents the magnitude of the DFT at that frequency and time.
I have most of this working, but I'm not exactly sure how to interpret the magnitude value. I'm computin... |
I am using MUSIC (Multiple Signal Classification) to determine Direction Of Arrival (DOA) and frequency of signals impinging on an Antenna Array. I am using a function MUSIC(U,V) which accepts left and right singular vectors of a matrix X. U is used to determine frequencies whereas V is used to determine the DOA (Metho... |
Assume i have a discrete signal, which i want to up-sample by factor 100.
Up-sampling using poly-phase algorithm sounds like a bad idea, in terms of cache low-efficiency due to low space-locality of the data.
Do you have any suggestions?
Thanks,
Itay
|
trying simply to create femtosecond pulse in MATLAB exactly like in the attached image,
the carrier frequency is around 374THz, and my sampling frequency is 10 times the carrier.
The results of the fft yields nothing understandable... tried to change some of the variables, fs,t-around zero, fftshift...
but could not... |
I have generated sinsuoids containing fundamental with harmonics using C++. Fundamental frequency is 50 Hz and each of the harmonic has phase offset of 60 degrees. I have applied flattop window before performing and N point FFT. Magnitude repose is correct. Phase reponse is wrong. I should expect 60 degree in all 100,1... |
I have to compute the some operations (say sum and subtraction) between 2 signals.
I have data in PDM domain:
x1_pdm(t)=[0101011....];
and
x2_pdm(t)=[0101011....];
What I do now is:
convert x1_pdm and x2_pdm in PCM domain (low pass filter and decimation factor M=64) and that compute the sum and difference (or any other... |
I'm working on creating an application which will render a spectrogram of the audio in an MP3 file. I'm having some trouble, though - after running the audio samples through my program, the resulting magnitudes are all very close together, and don't seem to be very patterned like one would expect. I'm coloring my spect... |
In control systems, the Laplace transform is often used to analyze the stability and the performance of LTI system. For instance, the LTI system is stable if and only if the transfer function, which is the quotient between the Laplace transform of the output of the system and the Laplace transform of the input, has all... |
What would be my options to decimate a large input signal available only in sections (or blocks) due to DMA type of interface? The FIR should be about order of 18 only and the total decimation factor of 8. I've made a quick MATLAB function to illustrate this issue.
function decimFilt(varargin)
% Multi-Stage Block-By-Bl... |
I am currently doing analysis on Photoplethysmograph (PPG) data and I want to know the frequency (heart rate) at every time point so a windowed FFT might not be the best option. I am looking at wavelet to generate frequency and time information. I have been working with matlab example code however I have trouble determ... |
Are there any algorithms out there that can do a FFT of 2D or XY data (not sure of the exact terminology here), I don't mean XY data like in a graph I mean XY/2D data points like a circle or a smiley face for example.
Thanks
Dave
|
I have two images: A and B. Image A is a screenshot of image B, the original image. While there is noise in image A, the original features are still distinguishable.
Histogram intersection tells us that:
$$K(a,b) = \sum_{i=1}^{n}\min(a_i,b_i)$$
where if $K$ is large, $a$ and $b$ are similar.
If we process image A to cr... |
I am trying to implement GCC-PHAT in python.
The approach is similar to the following two links:
link1 and link2
It seems the only difference between GCC-PHAT and normal cross-correlation using FFT is the division by the magnitude.
Here is my code:
import numpy as np
import matplotlib.pyplot as plt
from scipy.fftpack... |
Hello What are the numbers of Daubechies and Complex Gaussian are called?
I dont know what to call db4 or cgau8.
|
I want to the start time and end time for the peak shown below.
The start and end are shown by the marked circles in the fig
As there are number of such plots and want to know start and end time for each plot. So if its an automatic procdure thats good.
Can any one help me with this
Thanks
|
I'd like approach to Legendre polynomial.
But i don't make concept with Legendre polynomial.
So if you know where and when does L.P used and how L.P used, would you please let me know about this? To more understand it.
Hmm.. specifically in the application.
|
I know that for a given signal, the sampling frequency $F_s$ must be twice or more than maximum frequency of the signal $F_m$. It is easy to understand the concept for a 1D signal. But I don't know how to calculate sampling frequency or Nyquist rate for a 2D image.
Also what is effect of over-sampling and under-samplin... |
I am exploring methods for reducing computations required by MUSIC (Multiple Signal Classification) Algorithm. Most of the computational power is spent on calculating SVD.
My approach is to take fourier transform of signal matrix and perform filtering to make it sparse; calculating SVD on sparse matrix takes much less... |
It is said, do not use derivative operator in a the control system. because it will amplify the noise effect. Can someone explain it and give some mathematical reference about it please?
Also I heard, we shouldn't use logarithm operator too, for the same reason. How come?
Thanks in advance.
|
I want to generate a BPSK modulated signal with non zero mean. I don't know how to add a non zero mean.
|
As i know, taylor series is used for analysis f(x).
So if i have many orders, i can approximate f(x).
In this point of view, what is difference Legendre Polynomials?
|
If I have the histogram of an input image as Gaussian probability density function of the form:
$$P_r(r)=\dfrac{1}{\sqrt{2\pi}\sigma}e^{-\dfrac{(r-\mu)^2}{2\sigma^2}} $$
where: $\mu$ and $\sigma$ are the mean and standard deviation of the Gaussian PDF respectively.
The approach is to let $\mu$ and $\sigma$ be the measu... |
I am a self-learner of DSP by reading various books. I have accomplished basic understanding of Signals - CT and DT and a few transforms. I recently started to learn FIR / IIR Filters.
The part that I cannot understand is that they are termed as 'Filters' which for me logically means blocking / allowing from a threshol... |
I want to carry out a binary classification/segmentation on a grayscale image with very little texture. The only prior knowledge that's supposed to be available is that the object of interest is rather dark (i.e. low gray values) and comprises a significant portion of the image (maybe around 30%-70%). The object of int... |
I have a system with limited memory. So basically the question is this: If I have (say) 24KB of data space and 4096 data samples of single precision floats (16KB) can I do an FFT on the data? More generally, what is the minimum data space required to do an FFT?
|
I have an intensity image, similar to the image below. I would like to very quickly, segment this image, into separate peaks. Where each peak is labelled, and segmented. A notional segmentation of 3 blobs is given below by the black lines.
Definition of a peak segment is: The contour that just crosses the highest... |
Here I am again (Implementation of the constant Q transform + property questions) :)
I was reading this paper on a constant-q transform algorithm, where they managed to turn the following equation for the constant-q transform:
into this:
(where X[k] is the fourier transform of x[n], and K*[k, kcq] is the complex conj... |
For example, when playing piano, at the same time, print out the key notes by analyzing the signal of piano sound.
How to do some pre-processing to remove the noise? When calculate with FFT, noise will influence the result?
|
I am applying simple FFT to estimate the frequencies of the oscillations. The real values of frequencies are known to me as I made a synthesis signal for simulation.Thus, I can calculate the error simply. The signal contains 4 different frequencies. FFT can estimate these components fairly accurate.
when, the noise is... |
I am a little new to matlab and would like to get some help on how to write a program or a function that can calculate the energy/power of a signal depending on whether the signal is an energy or a power signal.
Here is the program the I tried:
function [ ] = power_or_energy( s,t)
%UNTITLED Summary of this function goe... |
I've been trying to get my head around digital plls from knowing nothing and seen that most examples only use the positive zero crossing, not the negative. I've tried to find the answer but the best i could find is a textbook that said "complications arise when using the negative zero crossing so this type of PLL isn't... |
I'm trying to understand the discrete-time convolution for LTIs and its graphical representation. standard explanations (like: this one) start with the idea of decomposing an input signal $x[t]$ into a sum of unit impulses. We have unit impulse function $\delta$:
$$
\delta[n]=
\begin{cases}
1,& \text{if}\ n = 0\\
... |
For multi-pitch detection, I mean for example pressing several piano keys at the same time. For signal pitch detection (pressing one key at a time), after FFT, mostly the frequency with the largest magnitude is desired frequency. However, for multi-pitch detection, let's just say press two piano keys, the two frequenc... |
given this block diagram:
can someone please clarify how to interpret the second delay (the one prior to $Y$)? to convert this block diagram into a simple equation, I interpret the diagram as:
bottom path: take input at $n$, multiply by $-1$ and delay by $1$, gives: $-1x[n-1]$
top path: take input at $n$ and sum it b... |
I came across this below question, (which was a homework assignment question for Signal Processing class, which my friend mailed me for help solving), mulled over it for an hour and had no idea how to proceed with solving it.
Let $C(x) = A(x)B(x)$ where:
$$A(x)=\sum_{n=0}^{N_1}a(n)x^n$$
$$B(x)=\sum_{n=0}^{N_2}b(n)x... |
The histogram in Photoshop for a 16-bit image downsamples to 8 bits, showing a tonal range of 0 to 255. I appreciate to show tonal ranges for a 16 bit image a horizontal zoom function would be necessary.
However the mean, std deviation,... statistics also get downsampled to 8 bits with a corresponding loss of accuracy.... |
The background:
I conducted an experiment where I rotated an object (with three-fold symmetry) many times and recorded the projection of the object. I should be able to tomographically reconstruct the object using an inverse Radon transform.
However, first, I need to know the exact angle of the object at each step. U... |
I am new to the field of signal processing. I am wondering what is the difference between DFS(Fourier Series) vs. DFT(Fourier Transform).
For common applications, usually we get a segment(length N) of digital waveform(like a audio segment), and then we apply FFT(DFT) and then do post-analysis with it.
I am wondering i... |
What is the actual meaning of frequency? what if the signal is aperiodic??
In general, we say that frequency means the repeatition of an event in unit time. But what about a speech signal? how can we say that a speech signal is having 4KHz Bandwidth(say)?
|
I use the following code to calculate the fft of 1024 samples of a 200 Hz windowed sinusoid (code taken from aurio touch apple example app):
//take fft
vDSP_ctoz((DSPComplex *)input.data(), 2, &inputComplex, 1, mFftSize);
vDSP_fft_zrop(fftSetup, &inputComplex, 1, &output.complex, 1, log2n, FFT_FORWARD);
//scale
float ... |
I have video with text which can be rotated in any direction. If I trained a neural network on images of each letter at all possible degrees of rotation, would it be possible to classify the characters accurately? Alternatively would another technique like HAAR, or KNN be better?
|
In short I'm recreating a program that reads 3 sinusoidal signals through an ADC performs some manipulations and then reproduces these signals through a DAC. In order to reproduce these signals I need the instantaneous phase of one of these manipulations. So I recreated a charge pump zero crossing pll in software.
I d... |
With the intention of converting a single rate LPF FIR filter to multi rate equivalent: what is the largest possible downsampling factor? I found one source that said it was the sampling frequency divided by the frequency at the centre of the transition band, but am seeking verification.
|
Subsets and Splits
No community queries yet
The top public SQL queries from the community will appear here once available.