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I stumbled upon an equation system as shown in the image below while designing an orthonormal wavelet filter. I tried solving the equation (5.63,5.64,5.65 => 5.66) but no avail. I'm not sure if there's any property or parameterization method to solve this equation?
The equation below is taken from "Introduction to wav... |
I need an algorithm to detect frequency and phase of a pure sine signal. The frequency of the input signal changes between 0 and 100 Hz.
The value of the signal gets captured with a frequency of 20 kHz (so I get 20000 values per second) - this is given and cannot be changed. I need to detect frequency and phase of this... |
When using the fft function in Matlab to take the Fourier transform, one can specify the number of FFT points, nfft, and the command is written fft(x,nfft). In this case, what is the frequency vector?
How can the Fourier transform be calculated with the same nfft, but over a narrower range of frequencies? How would th... |
I'm attempting to design a FIR high pass filter than keeps signals above 200Hz and rejects signals below 60Hz with a sampling frequency of 500 samples/sec. This is my first time attempting this and I'm a little confused.
I started with looking at pole/zero placement. I'm not exactly sure how to go about this but this i... |
So I am trying to determine whether the process of up sampling (zero padding) and downsampling are linear, causal and/or time-invariant.
Based on some resources I found online, I am under the impression that it is linear, but not time-invariant. However I am unsure about causality (since it depends neither on future no... |
I'm trying to understand how a particular equation works. The scenario is: there is a load of some resistance connected onto an amplifier with some noise figure.
The thermal noise of the load alone is:
$$
\langle i_{th}^2 \rangle = 4kTB/R
$$
but since there is an amplifier before it, the noise at the input of the ampl... |
How to get the value of $y_p(n)$ using substitution method?
What exactly do we need to substitute to get the value?
|
I am working on training the output of the wavelet transform of a signal using a neural network in MATLAB. So I decided to use 3 layer db4 wave filter for its wavelet transform. I have used an array of 240 elements and after wavelet transform, I get an array l
l = [36;36;65;123;240]
Now, since I have to train my neur... |
An audio sampler (in "music" terms because I'm not experienced in dsp) is a software which loads an audio file and allows you to play it at different pitches by slowing it down and speeding it up.
The audio file should be playable at any new frequency, and the frequency at which I'm playing it can change at any time, e... |
The question is:
Are the following systems linear time invariant?
$y(t)=x(t)$, with $x(t)=u(t)$ (unit step function)
$y[n]=x[n]$, with $x[n]=\delta[n]$ (Dirac delta function)
The reason I am asking is because I don't see the relevance of the values given to the input functions. LTI is a property of the system not it... |
I was searching for a tutorial on how to implement a stereo Phaser in C++. All i´ve found was sourcecode with little or no explanations which did´nt help me much. Does anyone know a Tutorial with which i can learn and understand how to implement a stereo Phaser in C++? By the way im using the Juce Plugin.
|
I searched a lot but I didn't find a good answer. According to wikipedia SNR page, SNR can be calculated by both ratio of Power of clean signal to its noise Power either ratio of variance of clean signal to variance of its noise, but actually power and variance are two different things so I am confused how we can use b... |
I saw in a SO thread a suggestion to use filtfilt which performs backwards/forwards filtering instead of lfilter.
What is the motivation for using one against the other technique?
|
Signal Characteristics:
The IF (I/Q) signal consists of a DC component and an AC component.
DC Component: 2.52V;
AC Component: 1.72 – 2.04 V (V P-P);
BW: 20Hz - 500kHz;
Power: 20dB
Requirement: Now I want to digitize this signal for further processing. I prefer a sampling at 1 MSPS with ADC and then porting digital dat... |
simply put, let's say I have this array - [1 2 1]
I want to try and get this output in Matlab
[1 1 1
1 2 1
1 1 1]
It's like my array was 'rotated' into forming the above matrix such that if I start at any peripheral element and move through the center of the matrix (in a straight line of course) towards the other end... |
Today I have heard, that is "impossible" to make low pass butterworth filter from cascading 1 order low pass filters(RC).
I am trying to discover is that true, but I can find little to none information on their difference.
Can you please explain to me, is that true, and if it is why. Thanks?
|
I have time domain strain gauge data from a structure which is subjected to low frequency, random loading. The signal / noise ratio is pretty good, but I would like to try to improve on it as much as possible.
From the FFT, I know that the dominant loading and response is below a frequency of around 1 Hz, but my sign... |
I’m new to DSP programming and I’m trying to learn a bit about convolution and deconvolution and FFTs. I have a project where I am taking a signal (a sine sweep) and convolving it with a shorter Impulse Response. Then I take the convolved signal and deconvolve it with the original signal to recover the IR. I’m writin... |
fm=20000 and I am sampling following signal at 65000 Hz:
x=0.5*sin(2*pi*(fm-2500)*t)+0.6*sin(2*pi*(fm-1250)*t*1.05)+0.7*sin(2*pi*fm*t*1.1)+0.8*sin(2*pi*(fm+1250)*t*1.15)+0.9*sin(2*pi*(fm+2500)*t*1.2);
and looking at frequency spectrum is not as it should look like.
Please find attachment.
It has code and result also.
|
I am simulating a feature detector in noisy environment which consist of AWGN Noise and Impulsive Noise. But I am getting a strange result as Signal with Impulsive+AWGN noise has better detection probability then with Signal with only AWGN. I know its wrong at some point. How Is it possible that signal with more noise ... |
I'm working on a software defined radio project on a PIC32 MCU. I'm using a CMX 973 quadrature frontend to provide I and Q. I'm independently low passing I and Q with carefully matched analog filters to reduce the overall bandwidth to 192 kHz, then digitizing it with an audio codec.
I'm going to be receiving a couple o... |
I am trying to find the 2-D derivative of an elongated Gaussian density.
The Gaussian has standard deviations $\sigma_x$ and $\sigma_y$. How can I get the scale-normalized 2-D Gaussian derivative in this case? Normally, I'd multiply by $\sigma^2$, but what to do when the standard deviation in different in each dimensio... |
Of the thousands to millions of frames that a video might have, how is a particular representative frame chosen to be the thumbnail?
|
I need to identify the coefficients of a linear, causal, time-invariant physical system that can be described by a classical state-space formulation.
For the sake of the example, suppose that the system has two states, only one of which is observed, and one input. Suppose that this state-space is described by the follo... |
I am working with a signal acquired by a pulse oximeter, i.e. a device that records the change of light intensity reflected by a specific tissue depending on its blood perfusion.
Since I am interest in the heart rate whose physiological frequency is in between let's say 30 beats per minute up to 300, I would like to b... |
I am well versed in cross-correlations, as my masters thesis heavily relied on them for music classification and beat detection. So this question focuses more on the signals that I am running an xcorr on as opposed to the actual cross correlation process.
Here is a simplified version of my setup:
Suppose I have the ori... |
I'm looking everywhere for solutions for audio downsampling. However for one reason or another they are incomplete, sometimes buggy, and sometimes are in a programming language I cannot understand.
So here's my use case:
I have an array containing two channels (left-right, it is a microphone) with Float32 data that get... |
i have been segmented some image of retina. but the result is not good as i hope. for one image i can get good segmentation, like this image
my code is from some reference in internet, in below i will write my code and this code is start when i have got a filtered image (mean filtering) .
prewitthor = [-1 0 1; -1 0 1... |
When the frequency domain's independent variable is labeled as a fraction of the sampling rate, the values along the horizontal axis always run between $0$ and $0.5$, since discrete data can only contain frequencies between DC ($0$) and half the sampling rate.
But why is the highest frequency pegged at exactly $0.5$ of... |
For the analog S-plane, the Q factor depends on the angle of the complex pole $p$ from the horizontal axis, $\theta = \arg p$:
damping factor $\zeta = \frac{1}{2Q}$, and $\zeta = \cos(\theta)$, and $\theta = \arg(p)$ so $Q = \frac{1}{2 \cos(\theta)}= \frac{1}{2 \cos(\arg(p))}$ Right?
But in the Z plane, the curves of... |
I have seen some devices which specify their internal noise in RMS or Peak-to-peak value. How does one calculate RMS or p2p values for white noise?
|
Is there a way to decompose IIR filter into set of poly phase components, that are All pass. E.g. a an IIR filter of order M, can be decomposed into a cascade of All zero filter of order 2*M( which can be implemented as polyphase) and all pole of same order( as Direct form), by rationalizing the Denominator. But, I ha... |
Essentially I'm looking for an explanation how the second image results from a capture of the original. Possibly you've seen something like this before and you can tell just from looking at the result what's going on.
The original image: https://i.stack.imgur.com/GBJ6P.jpg
The output image: https://i.stack.imgur.com/u... |
I had this question in the exam without any further explanation.
Why the time translation invariance is an important feature for time-frequency distributions?
I am writing to ask whether anyone can please explain what is time translation invariance? And why is it important feature?
|
When attempting to spot activities on an EEG waveform given a stimuli, it is recommended that several trials be taken and their average is done to increase the SNR.
Intuitively, since noise is not stationary or localized, averaging different trials will have a smearing effect on the noise.
But how exactly does it incre... |
A channel has a bandwith of 5MHz. If signal strength to spectral noise density S/N0 is 107 Hz, calculate the channel capacity C and R. The problem
is that I can use Shannon-Hartley theorem because instead of S/N or SNR I have
S/N0.
|
I'm trying to show that $y[n]=x[n]*h[n]$ turns into $Y(z) = X(z)H(z)$ in Z-domain by first applying convolution then by taking the inverse Z-transform of the $Y(z)$, stating that it's the same sequence after all. However, in power series expansion an extra term appears which kills the equation. You can find the questi... |
For instance, I believe we cannot detect global phase shifts. And of course frequencies above ~20kHz or below ~20Hz.
I've heard of "frequency masking", but don't know much about it and would appreciate a brief explanation.
What else is there?
|
In circuit analysis, I understand the use of Laplace Transforms to obtain the impedance of a linear RLC circuit, ie transforming from the time domain to the frequency domain. In most texts I have seen and classes I have taken, the Bode plot looks at the rational transfer function of $H(j\omega)$, which is effectively t... |
The Welch lower bound on the maximum cross-correlation (inner product) of $m$ vectors, each of $n$ elements, is given by $c_{max} = f(m, n, k)$ (eg, see Wikipedia Welch bounds). It is always assumed that $m \gt n$ (otherwise the bound is 0). Apparently the Welch bound has been widely applied in the restricted case tha... |
What would be the most significant difference when using either a sobel filter or a gaussian-derivative filter, apart from the obvious difference of the size of the 2 filters?
|
Here is a question I have been trying to solve:
Estimate the "essential bandwidth" of a rectangular pulse
$$ g(t) = \operatorname{rect}\left(\frac{t}{T}\right), $$ with $T>0$, where this "essential" bandwidth contains 90% of the rectangular pulse energy.
What I have so far is that the Fourier Transform of $\operatorn... |
I am new to signal processing and and having difficulty understanding how to implement an IIR filter. The reference is athttp://www.dxatlas.com/rocky/Advanced.asp in the Band scope section. It concerns applying an IIR filter to a power spectrum with this equation:
FilteredSpectrum[i] = FilteredSpectrum[i]*(1-gain) + ... |
I am working with a 30-state, 14-input linear model that is described by a state-space model:
model_state_space = ss(A, B, C, D);
The model is extremely slow (it describes the thermal behavior of a building), so when I discretize it I choose a sample time of 900 seconds and plot the step response from one of the input... |
I want to do one threshold method on high pass components of a wavelet frame decomposition approch,for edge detection in image. I have 14 high pass components.
for thresholding, I need keep the lowest 50% of the coefficients in absolute value of the high pass parts or in other words keep only the lowest 50% largest co... |
I am currently trying to detect heads in a sequence of real-footage images and am using HOG feature descriptor and SVM as classifier.
Currently I am using Dalal's HOG implementation code in MATLAB found in this link:
http://www.mathworks.com/matlabcentral/fileexchange/46408-histogram-of-oriented-gradients--hog--code-us... |
I want to know " Whether there is any Tool Box in Mathematica (MMA) for the Time-Frequency (TF) Signal Analysis".
I am well-versed in MMA Programming, so want to do TF Signal Analysis in MMA. I think if there is toolbox of TF Analysis, then it will be of very much great help in long programming.
Thanks in Advance!!!!
|
If I take the FFT of a sinusoid I will get a plot whit all the energy of the signal concentrated at the sinusoid frequency. But what happens if I have a signal in which the frequency keeps changing?(like the chirp: linear and exponential)
Is it possible to compute the FFT of a chirp signal? And what should I expect as... |
I have a question regarding Modulation and Demodulation.
In question 5.21 (b) from the "Discrete-time Signal Processing" book by Oppennheim, he asks to find out the type of filter used in the attached figure:
The question is what is the effect of $(-1)^n$ on the original filter and y[n].
I saw the answer to be high-... |
I'm new to dsp. I've got a complex signal that was sampled at 192kHz then an fft of 16384 bins created. This now needs to be plotted on a graph for a real time spectrum display. The problem is if you want to plot the whole spectrum consisting of a 192kHz span, you will be plotting 16384 points on a display that many... |
If we have a linear system, represented in State Space and the A matrix is singular (det(A) == 0), can we expect any special properties from the system?
|
Does removing the mean value of the signal effect the estimation of power spectral density(PSD)? If it effects, how?
What is the advantage we can get by removing the mean of the signal in estimating the PSD ?
Can the advantage be in dealing with the signal points whose magnitude is very large and we can not see the ch... |
Does anyone know , how the spread spectrum sequences are used in airports. I've attended a presentation this morning regarding spread spectrum in my college, the author mentioned that the sequences are helpful in landing of the aircraft, and also to mitigate interference between air crafts. How is that possible? Is it ... |
I would like to ask how to design a digital filter to remove $50$ Hz interference on an ECG signal. Do I need to use notch filter?
Thanks.
|
I am working on a FMCW radar and I want to sample the I/Q signals from the 2 receiving antenna (total 4 channels) simultaneoulsy and send the digital data to computer to extract range and velocity information using FFT.
The below diagram is a overall plan.
In the diagram I have shown a ADC with 1MSPS but thats not bas... |
The famous https://www.w3.org/TR/audio-eq-cookbook/ offers a set of [real] biquad filter calculation formulas that generally work fine.
However, when filter's frequency approaches Nyquist frequency the Q (bandwidth) specification of a filter becomes greatly distorted - usually it shrinks a lot (even though the author m... |
What does the graph of audio signal tells? what are negative samples?
|
Say I have an FIR filter with desired magnitude characteristics. I could imagine using an allpass IIR in series with the FIR to alter the group delay characteristic to provide both a 'magnitude equalization' and 'delay equalization' curve custom for the application.
I imagine this would be simpler in the frequency doma... |
I am applying 8 samples and triangular window to my fft engine. I am getting the results but not accurate (approximately equal with some tolerance).
I am calculating MATLAB reference as below :
a=[1,2,3,4,5,6,7,8];
b=triang(8);
c=a*b;
d=fft(c)
Is it the correct way to apply window for signal processing unit?
|
I'm designing a QC-LDPC decoder in VHDL. It uses a layered decoding method - only a single block row of parity equations is evaluated each iteration (the bit values are still updated every iteration). Bit values are stored as log-likelihoods with the sign representing the respective bit symbol.
A valid codeword is indi... |
For a project I’m currently working on, I composed a irregular ocean wave (function of time) by the summation of n regular waves. For the validation and further processing I’d like to produce the following graphs:
Power Spectral Density (PSD) spectrum of the wave (single sided) with:
Units: x-axis: [Hz], y-axis: [m^2/... |
What does the output signal look like when you insert $2000 Hz$ square pulses to chebyshev filter with parameters: (Matlab syntax). I tried to do it like this but I don't understand where it goes wrong.
%Making the square pulses
Fs = 1000000;
t = [0:1/Fs:50e-4];
f = 2000;
%square pulse with 50% duty cycle
xt = 1*s... |
If a system operates on a signal with a time-domain IIR or FIR, why would an acquisition system chunk the audio into powers of two? I can understand filling a buffer by a power of two for an FFT operation. Is there a difference?
I am still confused by the difference between block processing and sample by sample proces... |
My data is sampled at $f_s = 32kHz$, and for a particular analysis I'm only interested in signals with certain frequencies from $f_{highpass} = \text{0.1Hz}$ to $f_{lowpass} = \text{300 Hz}$.
When I build and apply a 4th order butterworth filter to my data I get an error and there's not much left of the signal. Here's... |
Our team develops an application that, in some feature, connects to a scanner (via TWAIN or WIA) and scans a document, and makes a PDF file out of it. The application is mostly used to process mail or documents with few images.
We recently noticed that the PDF files generated by our application are four times larger (s... |
I have the following problem:
I need to detect changes between two more or less similar image.
In that case, I want to detect the missing parts.
Lets assume that the displacement is no more that 2px, and there is no significant change in light (but there is some) or camera pose.
What I want as an output, is a binary ... |
I hope someone can help me understand the meaning of variance in this case. I am reading an old research paper by Ephraim and Malah (for my DSP project) and it says:
Knowing that $D_k$ is the FFT of white noise, what I am trying to find out is $\lambda_d(k)$. Can anyone explain to me the meaning of $E\lbrace|D_k|^2\rb... |
I want to design a Filter with a variable Cutoff Frequency and Resonance. However in the source I'm using there is a Bi-Quad Filter with two IIR and two FIR Filters designed like this :
Furthermore I found that I can compute the coefficients for the different Filter types as stated in this table :
Is there a way t... |
I would like to know how $\frac T2$ came for time period in the image below.Because I think the frequncy is 1 so time period should be 1.Is that correct? or can anyone help me..
The article from where I got this image is here
|
I am writing some code which takes a signal of length $n$, and processes it in 20 ms chunks with 5 ms of overlap... So basically I have a buffer of 20 ms worth of sampled data, and I am trying to apply a Hamming window to it. For some reason, when I listen back to the audio generated from this, I am getting nothing-- ... |
I have a signal with components at 1200 and 1800 Hz. When I mix this signal with a 1500 Hz signal to create components at -300 and +300 Hz (then LPF) a DC offset presents itself when the input signal is going form 1200 to 1800 Hz at a period of 1200 Hz (msk modulated signal). This strange affect only occurs with spec... |
I am trying to design a digital ButterWorth filter for the given specifications.
rp=3;
rs=15;
FS=1;
wp=0.5*pi;
ws=0.75*pi;
pwp=2*FS*tan(wp/2);
pws=2*FS*tan(ws/2);
[n,wn]=buttord(pwp,pws,rp,rs,'s')
[b,a]=butter(n,wn,'s');
[bn,an]=bilinear(b,a,FS);%error
freqz(bn,an,512,FS);
If I design a filter manually i am getting di... |
I'm having a hard time trying to understand signal modulation graphics and how can I calculate/find the properties of the signal. The properties I'm talking are:
Max. amplitude
Carrier frequency
the time it takes to code a bit
Can I do it just observing this two graphics? If not, can I do it using matlab? I'm a lit... |
I need help figuring out a way to automatically synchronize two audio files that are both different recordings of the same source. More info below.
This example is from a concert, where there is only one audio source - the band.
In the first track of the image below, we have three minutes of audio
recorded from the ... |
I have the output of a time series y(t) which is a vector for t = 1,2,...m denoting the number of data points.
Q1: If E[y(t)] = mean of a process y. This can be invoked using the command mean(y) or
writing the statement
m = size(y,1);
Expectation_y = sum(y)/m;
Then, how
do I find the Expectation of the following ... |
I am having a very hard time to implement the DCT algorithm. I have quite a few requirements like it has to work with NxN matrix or at least power of 2, it has to be 2D, it has to produce same output as FFTW fftwf_plan_r2r_2d(FFTW_REDFT10) it has to use real data and I need DCT I, II and III. And it has to be fast!
1) ... |
I would like to hear how aliasing sounds like. How can I get this with Audacity?
(I was told that I should adjust the Fourier coefficients in Audacity. I have no idea where I can do so / how I should adjust them.)
|
I have some noisy data that looks like this:
I want to find the components that are symmetric and antisymmetric with respect to magnetic field. The first thing I tried was just using:
Symmetric part $= \displaystyle \frac{1}{2}\left[y(x) + y(-x)\right]$
Antisymmetric part $= \displaystyle \frac{1}{2}\left[y(x) - y(-x)... |
I need to compute the amplitude of a certain frequency band as the input to a PID algorithm. My system looks like this:
[Sine Generator] -> [Actuator] -> [Physical complex coupling] -> [Sensor]
^ |
| ... |
Determine $X(\omega)$.
$g(t)$: I understand how to create a box from [-1,1] of amplitude 1/2.
$x(t) = g(t) * g(t)$
$X(\omega) = G(\omega)G(\omega)$
the solution I am seeing says that $G(\omega) = \frac{2\sin(\omega)} {2\omega}$
I don't understand where $\sin$ came from and that the values of the 2s correlate to. I'... |
I have software that tracks an object moving (in the x-dimension only) across a video shot from a stationary camera. I need to find the velocity and acceleration of the object as functions of time. This calculation can be done offline, so there are no real-time constraints. The motion of the object is also pretty regul... |
I have implemented a pre-emphasis filter by the following (pseudo) code:
a = 0.5;
s1[0] = s[0];
for (n = 1; n < N; n++) {
s1[n] = (s[n-1] * a) + s[n]
}
The problem I am finding is that due to summing, there is clipping happening. I thought the solution would be to find the difference between the max value in s1[n]... |
I am trying to implement an algorithm proposed in an old research paper by Ephraim and Malah for my DSP course project. I hope someone can help me with this difficult question. For the clean signal Amplitude Estimator, they used the following equation:
I understand most of the variables in the equation except the Bes... |
For STFT, we impose window of certain size onto the original signal, then we perform fft on each window. The uncertanty about frequency and time is determined by the width of the window, however, I can't understand what is the point of having overlap windows...
If we have a signal, for instance, why can't we just divid... |
Do you know of any algorithms for audio processing that make use of peaks in auto-correlation function? I am currently involved in research on mapping methodology for dynamic algorithms. I came recently over an algorithm for detection of fundamental frequency in audio signals, which has a dynamic structure I am looking... |
I am studying the meaning of the power spectral density (PSD) and I am wondering whether I can see from the PSD of the input signal and output signal if a system is a LTI system?
|
I've been lurking around these forums for a while and they've been a great help to me in the past. However, I've been researching this problem for the past 2 days or so and can't seem to find a solution even though I feel it should be fairly simple. What I want to do is read in the audio stream, and essentially ignore ... |
I have used an MMSE STSA estimator to obtain the magnitude of an audio signal. The original signal is combined with white noise and I used an algorithm given in an old research paper by Ephraim and Malah to estimate the amplitude of the clean signal. Now I just have the magnitude information and I don't have the phase.... |
A very common homework question. You are given an audio file(.wav) which has the recording of the keys pressed during a telephone call. You have to identify the telephone number from this audio file when you are given the sampling frequency & the duration for which one key is pressed.
From the given sampling frequenc... |
Following http://www.codeproject.com/Articles/729759/Android-Sensor-Fusion-Tutorial , I get two orientation values. Then, I transform those values to rotation matrices R1, R2.
I think the relative rotation matrix is $$R_{12} = R_1*R_2^T.$$
I test one case using matlab calibration toolbox. The ground truth for $R_{12}$ ... |
In my Android App,I have to detect text lines in images.I want to use Hough Transform to detect lines.You can see an image below:
I got edges in image by canny filter,then I applied Hough Transform on canny image:
Size size = canny.size();
Mat lines = new Mat();
int lineGap = 20;
double sum1 = 0;
Imgproc.HoughLinesP(... |
My understanding is that if I have a real signal, it can be called as the 'I' component and its Hilbert transform can be called the 'Q' component. Due to the property of Hilbert transform, 'I' and 'Q' will have 90 degrees phase difference.
Now consider the case of a pure sine wave with constant amplitude. Let us say w... |
I am currently in the process of taking on implementing the Gold-Rabiner pitch tracking algorithm, and I came across what seemed like a good explanation of it in David Meehan's "A Code Excited Linear Predictive Coder".
However, I am a bit confused with M5, which states: "the magnitude of the difference in amplitude be... |
In some papers/books I found infos about a signal given as RMS value.
What is the RMS value and what information does it contain?
How can I compute it?
Moreover, if they say that the signal is a white noise with RMS $100$ band limited onto the interval $[0, 2000] Hz$, how can I use the RMS information to produce the ... |
When calibrating a camera with a chessboard, are more squares better or worse? When would you use more or less? And how does the size of the blocks effect the calibration results?
|
I am looking to transition some of my MATLAB algorithms to C/C++ for deployment on target (PowerPC and/or ARM). Mostly 1D FIR/IIR filter in this instance.
In other domains, there are well-regarded general libraries (both speed and quality) with liberal licencing. For example, Linear Algebra has LAPACK and Eigen (among... |
In reading literature on the construction of impulse responses from sine sweeps (e.g. papers by Farina), I see it stated over & over that the way the impulse response is constructed from a (exponential) sine sweep is to take the original sine sweep signal $S(t)$, time reverse it ( $S(T-t)$ ), scale it somewhat in magni... |
Do you have any idea about how we can find the principal eigenvectors of an unknown matrix ${H}$?
The elements of $H$ are unknown in general. If you are familiar with channel estimation procedure in wireless communications, $H$ is channel matrix, and every element $(i,j)$ represents the channel gain between antenna ele... |
I'm trying to create my own program for speaker identification (text-independent) and have used MFCC features for that. But result of recognition is very poor. My supervisor told me that using MFCC for such problem is not the best solution because we are loosing a lot of data about speaker and MFCC mostly used for spee... |
I am trying to reconstruct a signal from a noisy speech using an MMSE algorithm proposed long time ago by Ephraim and Malah (1984). After going through the algorithm, I got a matrix A which represents the magnitude of the reconstructed signal. With the help of this group, I now know that I need to use this magnitude wi... |
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