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So, this is a picture of saccadic eye movement reflected by an EOG. How can I determine if the eye movement system is linear, causal, memoryless, and time invariant from this displayed waveform? Please explain.
|
I have position $\theta(s) = \frac{s^2+1}{s^4+2s^3+3s^2+4s+5}$ and velocity $\dot\theta(s) = \frac{s(s^2+1)}{s^4+2s^3+3s^2+4s+5}$.
How do I construct 2-state-observer with sampling time $T_s$?
$\left[
\begin{array}{c}
\theta_{k+1}\\
\dot\theta_{k+1}\\
\end{array}
\right]
=A\left[
\begin{array}{c}
\theta_{k}\\
... |
What do you think of this notch / band reject filter for a very very narrow band ?
Original signal x[n]
--> FFT
--> then I just put "0" in the FFT bin of the frequency I need to reject (notch filter)
--> inverse FFT
--> output signal y[n].
How is called such a filter ? Does it really work, or is it too simple to be ... |
I have an application that I must use a complex Morlet wavelet function (cmorfb-fc), I can run CWT (continuous wavelet transform) and it's fine, but I want to run DWT(Discrete wavelet transform) with this complex Morlet(cmorfb-fc) function too. then I want to compare the DWT and CWT results.
so Does anyone can help and... |
What's the simplest way to implement a FIR filter in C/C++ ?
(I don't want to have to use a third-party software like Matlab or Octave in order to compute the coefficiens because I have to use different filters on thousand of files, with different frequencies for each file.)
Thus, is there a ready-to-use C/C++ DSP libr... |
So I am Designing a system which would be able to send and receive stereo voice signals.( like a telephone but with full-band stereo voice signals)
here is an example to make my question clearer.
In a music recording studio two microphones are foreseen, one for singer's voice and the other for the guitar's sound. The ... |
I have a digital on/off signal sampled at a regular interval. There was a problem with the sensor that produced the signal which has sometimes caused a regular on/off ticking when the sensor should be in the off state. I.e. sometimes when the readings should be zero, there is a square wave instead (but with an unequa... |
How edges of the triangle get on both sides of fs? What is their purpose? Why fs in the center? (from Lyons Understanding DSP)
|
is that possible to find SNR of an environment ? ie., i have a outdoor environment where i cannot find the clean speech ,if so i want to measure the signal power to noise power how to do that
can i do some thing like put a voice activity detection and find voice frames and find the voice frames power to non voice frame... |
I've got an impulse response of a hall, it's 10 seconds long.
Now say I want to apply this reverb to a wav file of some singing. I would perform a convolution of the data in two wav files and that the result would be a third lot of data, this time with the reverb. Great.
But how would this actually work in a real time ... |
This is a photo of our homeworld, taken by the Juno spacecraft recently, sling-shotting on its way to Jupiter. What it gained in speed, we lost in ours, however thankfully we will not be falling into the sun.
I think the South American continent is on the left.
However, we can notice that there is a sort of artifact... |
I am doing a personal project which aims to detect infant cry. I have an infant cry audio database and extracted the MFCC from these files using MATLAB. I've come up with 12 coefficients for each frame the audio was divided into.
My goal is to compare audio obtained from a microphone with the training data (MFCC from t... |
I am using impulse responses from http://sound.media.mit.edu/resources/KEMAR.html
These are available as either raw or diffuse-field equalized HRTFs.
My question is: Which do I want? Raw, or diffuse-field equalized?
I am using these IRs to project single notes (generated synthetically using a ring resonator) to 12 loca... |
I am experimenting with HRTFs.
Specifically I'm trying to create 12 impulse responses that the brain can easily differentiate.
The more distinct they are from one another the better.
So I'm trying to figure out ways of making them as distinct as possible.
My best attempt so far is to use HRIRs on the horizontal plane ... |
Thanks to all the people who helped me here, I finally achieved to implement a notch filter (*), with coefficients given by :
http://www-users.cs.york.ac.uk/~fisher/mkfilter/trad.html
When I do order=2 (number of poles=4, according to this website), all is working very well with :
y[n] = a0*x[n-4] + a1*x[n-3] + a2*x[n-... |
This document on HRTFs, A Spherical Far Field HRIR/HRTF Compilation, of the Neumann KU 100, talks about headphone compensation filters.
Searching Google returns http://www.ak.tu-berlin.de/fileadmin/a0135/Publikationen/2010/Brinkmann_2010_On_the_effect_of_individual_headphone_compensation_in_binaural_synthesis.pdf
These... |
I am trying to create or find a sound that localises well.
I read that emergency vehicles are changing their sirens; people found it difficult to localise the sound of the old siren (a sinusoidal sweep), and apparently this led to a lot of accidents. Apparently a single frequency is difficult to localise.
I've experime... |
So I found this answer on one of the questions. I wanted to ask how the complex part can be related with the phase shift. I wanted to ask this on the same question but it didn't allow me. Could anyone please help me understand this concept. Even if I imagine this as a spiral with the value of the imaginary part changi... |
In error correction for example for Reed-Solomon codes there are computed syndromes for received data in decoding process. What is intuitively a syndrome (I don't mean the mathematical expression for it)?
|
Related question: What are the practically relevant differences between various image resampling methods?
Bilinear and bicubic interpolation for image resampling seem to be fairly common, but biquadratic is in my experience rarely heard of. To be sure, it's available in some programs and libraries, but generally it doe... |
I have two signals, one whisper data and one normal voice data of the same sentence.
I'm trying to build a voice conversion, but just want to start off with some simple signal manipulations.
I wanted to get the excitation of several vowels first to apply the excitation to the whispered data; basically to get a Electrol... |
This question follows from Designing a sound that localises well
I am developing an experimental system for pitch training. I am giving each of the 12 pitch-classes {C CD D DE E F FG G GA A AB B } an associated consonant phoneme: {d b r ng m v sh z h l k t }
So I could sing a chromatic scale 'da ba ra nga ... etc'
My ... |
How is it possible to draw an audio frequency response graph (x-axis : freq from 20 to 20 000hz, y-axis : dB) from a simple equation like
y[n] = x[n] - x[n-486]
(Example: with this equation I get a nice comb filter that removes frequency 197,5Hz and all its harmonics, for an audio file with sampling rate 96 Khz)
How ... |
I will try to explain what is my level of understanding of this problem, please correct me if I'm wrong:
RMS is the Root Mean Square, it represent the mean value of the input signal.
PSD is the measurement of the responses that shows me at which frequencies most of the energy is concentrated.
The area below a curve is... |
In this question, we consider the Hilbert space of vectors in $\mathbb{R}^4$ with one of its basis $\{v_0, v_1, v_2, v_3\}$.
Given the first three basis vectors
$$v_0=\left[\begin{array}{c}
1/2\\
1/2\\
1/2\\
1/2
\end{array}\right]v_1=\left[\begin{array}{c}
1/2\\
1/2\\
-1/2\\
-1/2
\end{array}\right]
v_2=\left[\begin{arr... |
Does any one of you know what topic to teach people related to signal processing.
The audience is mainly for a group of software programmer working on a telecommunication product who have limited knowledge of signal processing.
Thanks!
|
I am trying to find the amplitude and phase plots of the saw tooth waveform pictured.I have already computed the Fourier series of the waveform but I don't know how to derive the amplitude and phase plots from the sawtooth's Fourier series.
$$
x(t) = \begin{cases}
\frac{V}{T}t + V & \text{$-T < t < 0$} \\... |
Currently I'm working out plotting biquads in a VST/AU plugin and have problems with the graphics part. I posted in several forums but haven't got a really useful answer for this case.
First, tell me if I'm right how to get the magnitude response. For this I use this equation
http://rs-met.com/documents/dsp/BasicDigit... |
It seems that 2D Wigner-Ville has been used to test the compression of data.
The interference is however a problem.
Can someone send me a link to a code where this has been done?
I want to study the code and change it little study the compression and Wigner distribution.
Here, seems to be basic steps to make 512x512 m... |
I've recently started playing with the Kalman filter for a simple 2D (x,y,dx,dy) tracking toy problem. But I seem to have some misunderstanding on what I can expect from the filter. I'm interested in plotting the uncertainty ellipse from the corrected covariance matrix but noticed a few observations:
The covariance de... |
For a system such that $y[n]=\cos(x[n])$, I'd like to determine whether the system is causal or not.
Definition 1: A system is causal if and only if the output $y[n]$ is a function of a linear combination of inputs $x[n-k]$ such that $k \ge0 $.
We can see that the system is causal by this definition because $y[n]$ onl... |
Wikipedia - Wiener Deconvolution shows this formula:
$$
\ G(f) = \frac{H^*(f)S(f)}{ |H(f)|^2 S(f) + N(f) }
$$
Could someone explain the derivation?
Specifically, where does the squaring ($|H(f)|^2$) come from?
|
Can anyone suggest an introduction to speech processing,
class notes or books
accompanied by programs that can be run to see what the algorithms do ?
For example, in Burrus et al., Computer-Based Exercises for Signal Processing Using Matlab,
20 of the 400 pages are on speech processing. But it's from 1994;
there must b... |
Does the entropy of a symbol represent the most optimal average number of bits that can be used to represent a symbol? For example take the example of tossing a coin:
$$
H_{source} = -p_{head} log_2 p_{head} - p_{tail} log_2 p_{tail}
$$
so if
$$
p_{head} = p_{tail} = \frac{1}{2}
$$
then H = 1 bit, but if
$$
... |
I'm new to audio (and signal processing in general), and I've been learning about the AAC codec. I've read that it supports 1-48 synchronous channels in a single stream. My question is: what need is there for 48 channels in a single audio stream? I understand that if you have a 7.1 speaker setup you can send a separate... |
FFT of some signal $x$ has two parts: $|x|$, which is the magnitude spectrum and $arg(x)$, which is the phase spectrum. I want each frequency to have the same phase. Is there a way to accomplish this?
|
I was not quite sure what the right SE for this was, so I posted this also here on SciComp. Please tell me which one to remove :)
Problem statement
I have a few hundred unrelated time series, say $P_i(t)$, with $i=1,2,...,N$. The sample times are all unequally spaced.
It is known that during certain (unknown) interva... |
I have a processing unit that does a nice EQ filtering on soundfiles.
The only thing I can do is : I give the original .wav file, and the processing unit gives me back the filtered .wav. I have no other information than this.
I'd like to know the frequency response graph of this processing unit. How to do it ?
1) feed... |
I just have a small theoretical question that buggers me to be honest.
I have a sound file that I found its spectrum on linear and on logarithmic scale. Alright, I have done this, it's a plain standard procedure. The problem is that I cant really understand the plot on logarithmic scale. (it just looks like a typical ... |
I am trying to understand how you use an FFT in matlab to measure/retrieve the magnitude of the input signal.
For my understanding I have put this script together, creating a 15Hz sine wave with a sampling frequency of 1kHz.
fftsize = 2048; % How many samples for FFT
fs = 1000; % Sampli... |
I'm trying to code the Haar wavelet approximation of a function.
I'm a complete newbie.
Edit
Below is a pseudo-code of my algorithm for $f: [0,10] \to \mathbb R$, $f(x)=x$.
I picked it here and there on google.
1) Haar wave formula of order $N$ for $x \in [0, \infty)$
$$
\psi_{N,j}(x) = \frac{1}{\sqrt{N}} \cdot\begin... |
How might one model the propagation of a sound wave through a 3D environment with obstacles in it?
My initial thought would be to use freely moving particles that repel one another; with sufficiently many particles I see no reason why the simulation wouldn't work.
However, if any particle could collide with any other t... |
I have a 3D data set sampled on a regular grid shown in the Figure below. This data can be interpreted as an image where the point altitude is equivalent to the pixel intensity value.
In the middle of my 3D data, I have a peak, that becomes a bright spot when the data is interpreted as an image.
My goal is to remove th... |
Ok, so the situation is that I have a DFII biquad with some filter coefficients:
\begin{align}
w[n] &= x[n] - a_1*w[n-1] - a_2*w[n-2]\\
y[n] &= b_0*w[n] + b_1*w[n-1] + b_2*w[n-2]
\end{align}
While the filter is running, I change the coefficients and I believe my filter is going unstable. I want to build in a check t... |
I have the image below. This image contains sets of points that are aligned horizontally and vertically.
How can i find these sets of points? based 1) on their alignment and 2) on the ** **inter-line and Inter-word distances.
Do I have to use these 2 arguments, alignment and distance? or is one of them sufficient?
I... |
I use a simple biquad IIR filter :
if (n >= 2)
y[n] = (b0/a0)*x[n] + (b1/a0)*x[n-1] + (b2/a0)*x[n-2]
- (a1/a0)*y[n-1] - (a2/a0)*y[n-2]
else
y[n] = x[n] // here it probably needs to be modified
(thanks to http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt)
It works very well.
My problem... |
This question follows from Modelling propagation of sound wave by particle simulation
This last fortnight I have started experimenting with Binaural audio; it really blows everything else out of the water.
The main reason it hasn't taken off, I think, is that every individual needs to get their own HRTF calculated, whi... |
I would like to map the envelope of a sound (which has max dB level = dbmax, and min dB level = dbmin) from [dbmin,dbmax] to [-inf,dbmax].
Here is an example :
It's like some sort of dynamic compression (or rather dynamic range 'expansion' to be precise), but here the mapping is linear from [dbmin,dbmax] to [-inf,db... |
I just want to ask, whenever you reduce a bandwidth of signal by filtering, I wonder why do you need to improve the number of bits used to represent that signal.
The number of bit growth is
$$\frac{1}{2}\log_2\left(\frac{BW_{in}}{BW_{out}}\right)$$
Is it to represent to small noise due to filtering.
Thanks!
|
I'm trying to use the goertzel algorithm to detect a pair of frequencies in a signal.
The sample rate I'm using is 8000 hz, and the size of the incoming sample buffer is 60 samples.
I have a large array of 600 samples in size, so each time a new buffer comes in I move the data 60 samples to the right in the large buf... |
I am working on an accelerometer on a mobile device. When an event occurs, I record around 30 acceleration values. The accelerometer, being extremely noisy is very unreliable. What makes it worse is that my motion during the acquisition time is also of the same order as the noise. Is there a good method to estimate the... |
With the following code, I get the "envelope" of a sound signal.
However in the "envelope" there are small oscillations (high frequency) : it is not smooth enough...
I imagine that it's possible to apply a low-pass to the "envelope signal" in order to remove the high frequencies oscillations, but I also imagine it's ... |
Suppose $x[n]$ and $y[n]$ are two nonzero signals(i.e., $x[n] \neq 0$ for at least one value of n and similarly for $y[n]$).Can the convolution between $x[n]$ and $y[n]$ result in an identically zero signal? In other words, is it possible that $\displaystyle\sum_{k = -\infty}^{k = +\infty}x[k]y[n-k] = 0$ for all n.
|
I have just started learning some DSP, I keep seeing things like memory in regard to filters. I am unsure exactly what this means and why it is useful. I understand that increasing the number of taps increases the memory etc.. and a tap is essentially just a coefficient for changing the signal.
|
What are the advantages / disadvantages of bilateral filter and anisotropic diffusion for noise reduction?
EDIT: To be more specific - what are the practical differences in terms of quality, speed or flexibility? How should I choose whether to use AD or BF for noise reduction?
|
I am running Wavelet packet analysis for a 1-D signal. I want to estimate dominate frequencies in the original signal.
Imagine I have following signal:
fn=[0.4 0.6 1.0];E=[0.03 0.05 0.1];A=[1.0 2.0 3.0];
wn=2*pi.*fn;a=E.*wn;wd=wn.*sqrt(1-E.^2);
y=A(1)*exp(-a(1)*t).*sin(wd(1)*t)+A(2)*exp(-a(2)*t).*...
sin(wd(2)*t... |
What is the difference between the Fourier transform, short-time Fourier transform and wavelets?
|
I know that for a given system, the Fourier transform of its impulse response gives its frequency response.
I want to find where this property comes from, but haven't been able to find if it's a definition or if there's a mathematical proof available, for both continuous and discrete-time systems.
|
I'm using a C# library that has the FIR filter implemented and I'd like to know what's the meaning of the ${z^{-1}}$ operation (delay) mathematically.
The triangle is multiplication, the circle is a sum, but what does the delay means in terms of mathematical operations?
|
I'd like to implement a digital notch filter, which has linear phase, in C++ or C#.
Example : a notch filter that removes 400hz frequency on a .wav file (16 bit, 44.1 Khz, stereo).
I'm a bit lost with many examples, I don't achieve to implement them.
Maybe some of you has a working example?
Thank you in advance.
|
I have a set of data in excel like this:
column A is time in ns and column B is the electric field strength.
now I want to convert the time-domain signal from these data to a frequency-domain signal with DSP toolbox in matlab.
to calculate Fs (sampling frequency) in order to use in signal analyse i use this formula:
t... |
I have a signal in matlab like this:
I have designed a band-pass filter in 16-32 MHz range:
and applied the filter to the signal,
the result is like this :
My Question:
what is the easiest way to convert this time-domain signal into a frequency based signal?
can I export the signal into workspace and use the data v... |
Why I get a different response from the same system (e.g. three phase inverter with LC filter) in state space form and in transfer function (Laplace) form when using the same PI controller values ($K_p$ and $T_i$)?
First I start with differential equations for each phase $a$, $b$ and $c$ to get state space model
\begi... |
So I just revised my pitch calculation algorithm using a harmonic product spectrum algorithm. I was just curious about why this explanation of Harmonic Product Spectrum states that you need to implement a Hanning Window to the data set. What would be the effect of implementing other Window functions on a data set (and ... |
I am programming a voice activity detection algorithm and I found a paper that recommends the use of HHT. I have been trying to program it and understand how to calculate $m1(x)$, but the shifting process confuses me.
Could someone explain what exactly I need to do after I calculate the mean value between the two spli... |
The top answer to this question: Power spectral density vs Energy spectral density states:
However if random process is stationary, then it is for sure that it
has some finite power over some band of frequencies
Why is that? Also, is stationarity a necessary condition for a process to have finite power over a band ... |
On one hand, there is the statistical definition:
$R_x(\tau) = \frac{\operatorname{E}[(X_t - \mu)(X_{t+\tau} - \mu)]}{\sigma^2}$
and on the other hand, there is this other definition (e.g. see Dilip's comment here):
$R_x (\tau) = \int_{-\infty}^{\infty} x(t) x(t+\tau) {\rm d}t$
Why are these two definitions different? ... |
I am trying to write an algorithm that would automatically segment a piece of audio with bird calls recordings. My input data are 1 minute-long wave files and on the output I would like to get separate calls for further analysis. Problem is that signal-to-noise ratio is quite terrible due to environmental conditions an... |
The following picture is amplitude and phase of an image filtered using Gabor filter in the frequency domain. I'm trying to reproduce the results, I'm getting valid results for the amplitude I think, but I'm struggling with the the phase, I'm just getting a noisy image that does not resemble the picture. I'm referencin... |
I am programming a Voice Activity Detection algorithm. After researching several methods, I implemented a simple method that determines whether or not the person is speaking based off the RMS of the signal. However, I want to improve the algorithm to determine whether the sound coming into the microphone is actually a ... |
I apologize if this comes as a basic question, but I am struggling to understand why the PSD can only be estimated and not directly computed. For example, this thread discusses several such PSD estimation methods.
If I have a deterministic signal with a fixed number of samples, shouldn't I be able to directly determin... |
I understood that SNR is the ratio of signal power to the noise power. In terms of images, how the original image is affected by the added noise. In PSNR, we take the square of the peak value in the image (in case of an 8 bit image, the peak value is 255) and divide it by the mean square error. The SNR and PSNR are use... |
Sensor fusion algorithms can give a more precise 3D orientation (and possibly postion?) of a device by combining readings from an accelerometer, gyroscope, and magnetometer.
Can anyone explain, or provide links, explaining the details behind this? I want to understand the math and physics, such that if I have raw 9-DO... |
I am trying to understand how the Windowed Fourier transform works, so I used Matlab to create an artificial signal, and plotted the Windowed Fourier transform using a gaussian window.
$x(t) = \left\{\begin{matrix}
\sin(2\pi100t)& \text{for }0\leq{t}<\frac{1}{4}\\
\sin(2\pi50t)& \text{for }\frac{1}{4}\leq{t}<\frac{... |
I have a general task of discriminating images of shaded cylinders that significantly vary in 3d orientation, and almost insignificantly in size and shape.
I have treated it as a regression problem on a dataset of these oriented cylinders, but my problem is with feature computation for each sample.
Basically, what is a... |
I have a dozen source sound waves; each is about one second duration. I want to put each one through a digital filter (I'm truncating it to 128 taps).
Currently I'm doing this with a bog standard convolution. It is chewing up a lot of CPU.
I would like to switch to fast convolution.
Could someone outline the steps?
I'm... |
I'm struggling to implement a sampling scheme where arithmetic decoding is used to generate the samples.
This is based on excercise 6.3 from David MacKay's book, where you compare rejection sampling with arithmetic decoding sampling. For each symbol you want to generate with rejection sampling you have to generate an ... |
I'm trying to create a digital filter from a first order analog filter with transfer function
$$H(s)=\frac{1}{1+\tau s}$$
with time constant $\tau=.1\text{s}$, and sampling rate $f_s=1000\text{Hz}$.
Applying the bilinear transform in Matlab however appears to yield a filter with the a different 3dB point than expecte... |
I am going to be straight forward and say this is one of the problem sheets and I am stuck on one of the questions. I already know the answer as I have been given them but I am trying to work backwards to see how to derive from the answer. so the question is:
Find another cosine y(t), whose frequency is as close as ... |
Are the results of applying Kalman filter and recursive linear MMSE estimation process the same ?
|
While studying an OFDM system, I have encountered statements like coherence bandwidth is the reciprocal of delay spread and coherence time is the reciprocal of the Doppler spectrum.
Can somebody help me to understand this mathematically?
How is coherence bandwidth related to delay spread, and how is coherence time rela... |
I'm trying to use Savitzky-Golay Filter to smooth experimental data. Savitzky-Golay computes $c_{n}$ in the following equation
$$ g_{i} = \displaystyle\sum\limits_{n=-n_{L}}^{n_{R}} c_{n}f_{i}+n$$
According to Numerical Recipes book by William H. Press. I've bought the library. The next step to calculate $g_{0}$ which... |
I have a problem that is similar to the state space model in Kalman filter but the observation matrix $G_t$ of $$y_t=G_tx_t +w_t,$$ is random. The elements of $G_t$ are i.i.d. random variables with a given distribution. Is it possible to use the Kalman filter to this problem? If it has already been used please provide ... |
I'm just beginning to study Fourier transforms and I'm having trouble getting this concept down.
I know what a low/high pass filter does to a signal.
What is the relationship between Fourier magnitudes (dB) and frequency (Hz)?
How would Fourier magnitudes get affected when a low/high pass filter is passed on a signal... |
I am confused between symbol rate offset, symbol timing offset and sampling clock offset. Can anyone tell be the difference between these terms.
|
Professor asked me what filter is better -1/3 2/3 -1/3 or -1/4 1/2 -1/4, the answer was second. But what those coefficients means? It was during discussion of linear filter but i'm not sure was it about it.
|
What are filter banks and what's their application? What's the meaning of $H(z)$ in descriptions of filter banks?
|
I have implemented a QPSK modulator and demodulator using classic text-book examples: Square Root Raised Cosine Impulse shaping and Matched Filtering.
The problem is that I've been told that I could only used square waves for the modulation instead of sine waves on the emitter side (only +/-Vmod) and this really degrad... |
What happens if I ignore Nyquist? I have a 16.368MHz digital signal coming out of a GPS front-end chip. My microcontroller (which is reading that digital stream of data) operates at a maximum of 32MHz so sampling at ~16MHz seems unlikely. If I undersample my signal with say M = 3, then I have a 5.456MHz signal which... |
I'm trying to create a spectrogram of a complex signal 'y', but I'm running into a problem where the spectrogram that I create appears to have its spectral features "displaced" in frequency space, and I think that it has something to do with not having pre-processed the signal correctly.
I know the signal's:
sampling r... |
I went through the answer Kalman filter in practice and it seems we must know all the first and second order properties of random variables to apply the Kalman filter.
But when I only have a set of data $y_1,\dots,y_n$ and a given model $G$ and $F$ with assumed noise processes, could someone please explain to me how to... |
In most examples and FFT code that I've seen, the output (frequency magnitudes) of the forward DFT operation is scaled by N -- i.e. instead of giving you the magnitude of each frequency bin, it gives you N times the magnitude.
Operationally, this is simply because the DFT is calculated by taking the inner product of th... |
I have a vector of zeros and ones with length ~2000. I need to detect and filter out the once, that appear periodically. There may be none to 3 such periodic signal's components. Typical periods are in range 10-90, but due to measuring noise, some spikes may be shifted by 1-3 steps in any direction.
I assume there is... |
I'm trying to simulate the damping of the hand on a guitarstring on an already recorded/sampled open guitarstring sound. I've been trying to use low pass filter and had a moving frequency range but that didn't make it sound like a palm muted string, just the loss of higher frequencies.
Could someone help me find good m... |
I want to implement a Time-Freqency analysis by wavelet packets. I don't know what level and sampling rate should I use. I've tried different value for level(3,4,5,6,). but I couldn't get an accurate frequency identification. here is my little scrip and results, hope one can say whats the problem, Thanks in advance.
... |
Is there a method in order to compute some sort of edge detection / convex hull of a signal x[n] (in orange here). Here of course the orange signal is not convex nor concave, but only piecewise...
(I would like to know if this could be an alternative way of computing the envelope of a signal, alternative to the traditi... |
I’m trying to figure out how to go about differentiating between a single-voice speech and a multiple-voice conversation in which voices overlap. In other words – the moment of overlap is what I’m interested in. I want to be able to listen to a vocal conversation between two people, and detect when voices collide/overl... |
I have found several sources describing how to use Doppler velocimetry/anemometry to measure the Doppler shift with analog signals, but i cannot find much about using this method for digital signals such as Bluetooth. I was wondering if it were possible to use Doppler velocimetry/anemometry with digital signals? If so... |
I am dealing with a problem similar to principal component analysis. Aka, I have a matrix and i want to recover the 'most efficient basis' to exaplin the matrix variability. With a square matrix these are the eigenvectors, weighted by the eigenvalues.
Originally, I was dealing with square matrices, and I used eigendec... |
I'm an absolute beginner.
I'm doing a project that involves estimating the time delays between the sound signal arriving at different microphones in an array (the project is a bit complicated but I'll find my way through). I'm planning to use a panda board and connect the microphone array to it. But before going about ... |
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