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If I were to make two recordings, A, and B, would A and B sound (drastically) different if:
"A" is a recording of two people talking simultaneously.
"B" is created in Audacity by "Mix and Render" on two tracks of people talking, making them "talk" simultaneously.
Assume that each person in each recording is speaking ... |
Given a .wav file that is a LTI system's response to an impulse, what would you recommend as the simplest way to plot the frequency and phase output of the system? I'm already working in Python, so python-based solutions are welcome but alternatives are fine too.
|
One technique to make a high pass filter is to subtract a lowpass filter from a wire (meaning subtract the output of the filter from the input as I have demonstrated in this post).
What lowpass can you subtract from a wire that will result in an all-pass? Specifically pass all frequencies with unity gain and modify the... |
I have a 24GHz motion detector (SMR-333 website, sparse datasheet) that gives I and Q, without signal processing. How can I use these values to detect motion? I only need to detect if a person moves past the sensor (from 2" to 48"). The direction and other aspects of the movement are not important to me.
I'm primarily ... |
In this answer, the responder claims that acquiring a measured impulse response and straightforwardly inverting it (e.g. by directly in the spectrum, after zero-padding, or by determining the inverse filter through least-squares) would sound awful and therefore, would be a poor equalizer for an audio system. Wouldn't i... |
Assume linear phase FIR symteric impulse response like [1 2 3 4 5 5 4 3 2 1],shoud resault in a frequency response with linear phase negative sloped phase. but it doesn't:
>>stem(angle(fftshift(fft([1 2 3 4 5 5 4 3 2 1]))))
causes:
But this is wrong the plot must follow the red line. Whats wrong with it?
Another exam... |
I am a civil engineer and I have a research focused on InSAR Data.
I have a basic question about analyzing the signals, please.
How can I dig into the signal I have?
Is there any code to define the pattern of the signal or just by ploting the values along the points of time? or maybe there is a specific procedure I sho... |
I have simulated a digital control system in the Z domain using MATLAB and I have got satisfactory results. However, when I converted the plant and the digital controller to difference equations and implemented them in an m-file, I did not get the same results displayed in the Z domain
In the Z domain, my system is sta... |
I'm trying to simulate a particle going from (-3,0) to (3,0) with a constant velocity and some noise (e.g. the particle is a quadcopter trying to fly at constant velocity, but may be pushed by gusts of air). The model I have come up with is: $s_{n+1}=A s_n+B u_n+G w_n$ which in my case is:
$$
\left[\begin{array}{l}
x_{... |
I have been reading lots of posts about the scaling factors in PSD, and related topics on this forum. My understanding of the topic has been greatly improved. Thankyou all. I just need a little more to drive the concepts home.
My main question now is about the definition of power spectrum. First of all, what is its mea... |
Today I learned, instantaneous power is
$$
p_x(t) = |x(t)|^2
$$
and I've known, energy is
$$
E_x = \int_{-\infty}^{\infty} |x(t)|^2 dt
$$
What I also thought was, $|x(t)|^2$ is instantaneous energy, and that $P(t) = dE(t)/dt$, but these are inconsistent. What I can't figure out:
Why is $|x(t)|^2$ defined to be instan... |
Up to now I've never found a rigorous or a formal definition of what it means for a signal to be sparse other than it means that it has a relatively low number of non-zero entries or that the cardinality of its support is small. The only thing I've found close to a somewhat formal definition of a sparse signal is that ... |
Test Setup
I have two i2s microphones (INMP441) on a common clock. They are designated as the left and right channels with a sampling rate of 22050Hz. The generated signals are stored as WAV files and are post processed using python. I am playing an audio tone of 5100Hz.
Processing
The python processing slices the data... |
As far as I understand, double sideband AM with carrier and FM can be demodulated via SDR (Software Defined Radio) without frequency or phase correction. For AM, received signal is
$$r(t) = A(t)\cos(\omega_c t) \tag{1}\label{received_am} $$
where $A(t) = 1 + m(t)$ and $-1<m(t)<1$. In ideal case we can just multiply \eq... |
Let me write down all the facts that I know of.
In context of the z plane:
Minimum phase system: All zeros and poles of such a system lie inside the unit circle.
Linear phase FIR filter: For every zero ($ z = z_0 $), its inverse ($ z = \frac{1}{z_0} $) is also a zero of this filter.
Real value coefficients FIR filter:... |
I am having trouble understanding why the system matrix of an LTI system is Toeplitz. I am following an Edx online course by Professor Richard Baraniuk of Rice University, named discrete-time signals and systems. In the course, the instructor showed that given a system matrix $H$, $H(n,m) = H(n'+q,m'+q)$ where $n'=n-q$... |
I am looking for alternate solutions to determine the IIR filter coefficients from the impulse response, or more specifically determine the closest IIR filter for a given FIR filter. Much of this is covered at this related post, but synchrosqueezing was not mentioned, and as suggested and referenced by OverLordGoldDrag... |
Consider the original signal (just a sine) and the same signal interpolated with zeros:
The Fourier transform of them:
In this example I inserted only one zero between the samples, as if I sampled the original signal with $2f_s$ (where $f_s$ is the sampling for original signal), but added zero after each sample.
Her... |
I am trying to resample a signal using Matlab's resample function. The documentation explains that conceptually, it is upsampling by p, then applying an interpolation filter, then downsampling by q, although it implements it using polyphase filters. If my original sampling rate is $f_s$, the upsampled signal is at rate... |
I want to convert analog signal to digital form using Matlab. I know that this will need two steps (sampling and quantization) but I am confused which command should be used for quantization? quant or quantiz?
I have read online official documentation of both commands quant & quantiz.
The documentation for quantiz`` ha... |
I am quite new to signal processing, so I would need some help regarding a project.
I am interested in increasing the bass of a song from Spotify. At first I was thinking that applying a FFT to it to get the frequency content and then adding a Low-Pass Filter would be enough. But I started reading from here: https://en... |
Note: this question has been edited to refer to a sample that does not use dithering.
I recently asked my first question on here, about what I saw in an uncompressed audio file written out as integers. I learned about dithering.
This time, rather than silence, I'm examining a sine wave I generated. In Audacity, I gener... |
Let's say I have a Kalman filter with this simple state model:
$$\begin{pmatrix} x^0_{k+1}\\ x^1_{k+1}\\ \end{pmatrix} = \begin{pmatrix} 1 & \Delta t\\ 0 & 1\\ \end{pmatrix} \begin{pmatrix} x^0_{k}\\ x^1_{k}\\ \end{pmatrix} + \mathbf{w_k} \tag 1$$
where $x^1$ is the time derivative of $x^0$, and $\mathbf{w_k}$ is proc... |
I am training deep learning models (i.e., CNNs, convolutional deep neural nets) on Fourier transformed images, i.e. the neural net receives as input a 2-channel (real and imaginary) tensor of shape e.g. batchsize x 2channels x 64 x 64 which is a DFT of an image.
The network struggles to learn even more to generalize an... |
I would like to convert PDM (Pulse Density Modulation) microphone data into PCM (Pulse Code Modulation) data. I understand that there are 2 steps in this process.
application of a low pass filter on the PDM data stream of binary pulses
decimation process to reduce the sample rate to the desired PCM data rate
encode th... |
I am collecting data for a project, where I am measuring the room impulse response by playing a sine sweep from a speaker (QSC K8 1KW Active Loudspeaker) and measuring it from the other side of the room using a microphone array. Even though the sine sweep is very clean (bottom image), there appear to be these "ghost sw... |
I am using the basic way to calculate the PAPR of OFDM system using MATLAB, the strange thing I noticed that I get almost the same PAPR effect even when increasing the modulation order from 4 to 64. I couldn't explain the cause of that issue. Here is the code I use:
N = 256;
M = 4;
PAPR_ofdm = zeros(1, 10^5); %init... |
I am designing a low pass filter which takes a 19 bit input.
\begin{align}
\log_2(\max(\mbox{LPF_COEFF})) &= 16.64\\
\log_2(\mbox{sum}(\mbox{LPF_COEFF})) &= 17.87
\end{align}
Now should I take the output as 19+17 =36 bit output or 19+18=37 bit output.
|
I am new to matlab.
I have a noisy dataset represented as a signal on which I have to compute the fft and plot the one-sided spectrum, however I'm quite unsure if my data's signal length I assume that is denoted by the size of my vector (4x10000double).
Just a picture of a piece of how my data looks:
Fs = 1000; ... |
I'm stuck trying to filter a two-channel signal using sosfilt() with the zi initial state. I need to use the zi parameter so I can carry the filter state over multiple buffers.
Omitting zi works as expected
import numpy as np
from scipy.signal import sosfilt
>>> x = np.arange(20, dtype=np.float32).reshape(2, -1)
>>> ... |
I am working on a set of Images captured by an Industrial camera. However, I am not sure if I need to apply any Denoising (e.g. Gaussian or Laplacian etc) on it ?
Is there any metric that I can used to arrive at the conclusion ? Certain blogs I happened to read suggest using PSNR or MSE technique.
A sample image is att... |
DESCRIPTION OF SIGNALS
I have several sung melodies under each of nine conditions. The melodies are all of slightly different lengths (ranging more or less from 3s to 12s), and each is encoded as a discretised melodic contour (pitch profile), at one sample per ms (fs=1000 Hz). It is those, rather than the original audi... |
I am learning about SQNR and when i google i get its formula that it is equal to
6.02 db(per bit)+1.76 db
But in our Lab/class, we saw different formula ,which is following formula when using MATLAB:
SQNR=10*log10(sum(x.^2)/sum(E.^2))
Where x is original signal and E is error/noise signal
I am confused which formula i... |
Is there a relationship between the distance between the transceiver and the receiver and the AWGN?
When the distance between the transceiver and receiver increases, the AWGN level decreases?
|
I have an input signal of 30000 samples. I perform the Stationary wavelet transform in python to get the 4th level od decomposition. Then I perform an array mathematical operation over the decomposed level and convert it back into time. When I perform that conversion back in time, I get the exact double length (60000) ... |
I have frequency data from a VNA. I am trying to convert to time domain, and I can't figure out why I am getting complex valued time domain data. This is for ISAR measurements, and I can't get passed this simple step.
ifft_points = length(frequency)*1;
% S11 time domain
for i = 1:1:s_param_number
temp_data = data(:... |
I am back with another question.
Context for the question: I am trying to smooth out the angular velocity data from an encoder. The encoder has 720 ppr and the rough angular speed of the wheel is around 327 rpm (5.45 Hz). Thus, the maximum update rate of the encoder is around 4000 Hz (encoder gives velocity update ever... |
Which command should be used for quantization of a signal in MATLAB?
Please check above question link especially answer of @Hilmar, especially the first paragraph of answer and especially the highlighted statement:
"Every signal that's represented as a vector or a matrix in Matlab is already quantized. You can't reall... |
Does the change of phase of the signal affect the data rate?
Actually, I consider the RIS(Reconfigurable Intelligent Surfaces) between the transceiver and receiver. I wonder if adjusting the phase shift matrix of RIS can improve the datatate.
|
I am reading book"Signals and Systems Laboratory with Matlab" and i am studying about discrete time signals
I have also attached a snap and highlighted a line:
The definition of the time. For discrete-time signals, time is defined by using step 1
Does that means when we define a discrete time signal in MATLAB ,its inc... |
Suppose I have a cheap camera that takes 1280x720 photos which are a little bit blurry.
Now suppose I stand somewhere and take 100 photos of the same scene, more or less from the same spot.
Is it possible to combine the data from these images to produce a sharp 1920x1080 image?
(Please note, I'm not asking for a "give ... |
I have the following transform for $t>0, a_i>0$
$$f(t)=\sum_{i=0}^d a_i \exp(-t a_i)$$
And I need to invert it for a set of target values $b$:
Find $(t_0,t_1,\ldots,t_d)$ such that $f(t_0),f(t_1),\ldots,f(t_d)=(b_0,b_1,\ldots,b_d)$
Is there a way to express either forward or backward operation in terms of some standard... |
Recently I study radar alot and was wondering if there is a possibility to generate SAR like pictures with MIMO radar.
Under the condition that the MIMO radar is not moving and the scene does not change.
I am curious about your ideas and papers if you have some.
|
I have three random variables:
$Y$: my data
$Y_n$: my data corrupted by additive white Gaussian noise (AWGN)
$Y_{nc}$: my noisy data corrupted by a non-linear transformation $\mathcal{C}$.
I have computed the histograms of the noise terms:
$N = Y_n - Y$: the AWGN
$N_c = Y_{nc} - Y_n$: the noise added by the transfor... |
I am working on speech enhancement and I want to compare the phase and magnitude between two signals. I compute the STFT for both, and comparing the magnitudes is easy via mean-squared error or some other metric. However, for the phase, there seems to be no precedent in the literature for how to check if the two phases... |
I wanted to make wavelet transform and represent the frequencies as function of time instead of the scale of the wavelet as function of time. This example that uses ssqueezepy does it however I would like to understand precisely first what are the frequencies that it calculates (is it the frequency of the signal?) and ... |
Let $x(t) = \cos(2\pi\times15t)+\cos(2\pi\times22t)+\cos(2\pi\times35t)+\cos(2\pi\times42t)$ and $\forall t\in\mathbb{R} :w(t) = 1$. We sample $x(t)$ and $w(t)$ with $F_s = 92 \ \text{Hz}$. So we obtain the discrete-time signals $x[n]$ and $w[n]$. The product $y[n] = x[n]w[n]$ is the windowed version of $x[n]$. Let $N$... |
According to my understanding the FFT operation for a vector whose length is $N$ has a complexity of $ \frac{N}{2}\log_{2}{N}$ complex multiplication and $ N\log_{2}{N}$ complex addition. I was wondering if I performed that FFT operation using another way, will that have the same complexity too?
For example, let have t... |
I have real-valued samples of a signal at a negative frequency (was mixed with a cosine of a higher frequency than itself). I want to quadrature modulate it up from $-\omega_c$ to 0. There seem to be two ways to do it. Both seem to make sense. I've tried and plotted both and they seem to give very similar results, but ... |
Suppose I have a sound recording of a tone whose instantaneous frequency keeps falling (continuously, not in steps) in inverse proportion to the running time -- i.e. the initial frequency is 1000 Hz, then 1 second later the frequency is 500 Hz, 2 seconds later it's ~333.333 Hz, 3 seconds later it's 250 Hz and so on. Th... |
I've heard, that "windowed Fourier transform" is but one perspective on STFT, and that STFT is fundamentally convolutions of windowed complex sinusoids with the input, i.e. bandpass filtering. Yet the standard implementation and perspective on STFT is, take a segment of input x_seg = x[start:end], window it x_seg * win... |
I'm back again with another probably very basic question, but I searched a lot about the minimum length of signal for a filter to work, and mine seems to satisfy this (from filtfilt requirement in MATLAB), but still the filter doesn't seem to work. Please help.
Context : Dealing with an encoder signal for angular veloc... |
I'm sorry if this kind of question isn't allowed, but I'm starting to learn Fourier series and I'm still not entirely sure what's going on... in this specific case, I'm trying to find the Continuous Time FS representation of this signal:
From what I gather, the period of this signal is $T=6$, and in order to find the ... |
As far as I know, for any signal the first $N/2$ frequency samples and second $N/2$ frequency samples are equal (by magnitude), so (sorry, if I made mistake in indices):
$$X[k]=\sum_{n=0}^{N-1}x[n]e^{-j2 \pi kn/N}, \tag{1}$$
$$X^*[1]=X[N-1], X^*[2]=X[N-2], ... \qquad \text{if} \ x[n]\in \mathbb{R} \ \forall n\in\mathbb... |
I am working on experimental data, where I need to choose Q in Kalman filter. How to intuitively understand:
What does selecting low value of Q indicates in Kalman filter?
What does selecting high value of Q indicates in Kalman filter?
|
Suppose I sample an audio signal with a sampling frequency > twice the highest audio frequency.
ie, the frequency transform of the sampled signal looks like the image on top here, where there is no overlap of copies:
Also, suppose that all the high image copies are outside the audible range. Now is there any need for ... |
We know that we can sample a real function such as g(x) = sin(2pi*f*x) with a (approximate) critical sampling rate expressed as max[dg(x)/dx/pi] = 2f, assuming that x in the range of [-x0,x0]. But what's the critical sampling frequency of a complex function, such as g(x) = cos(a*x^3)*exp(i*2pi*f*x) (here 'a' is a const... |
Is there a way to extract only the content related to the main and important frequencies when we get the FFT and have the information related to the important frequencies only (amplitude and frequency especially)?
with help of @dan boschen so it became clear that such a work is acceptable with a good approximation tha... |
In real world, signals are always finite, which means that any aperiodic signal can be periodic signal by repeating themselves. Then, why we don't just use Fourier series for those finite aperiodic signals?
|
So I currently have a time series signals which I need to classify each point on each signal.
Points are taken as follow:
For a given timestamp. Make a measure
Each meassure can randomly come from a different source (not equiprobable but I ignore the probability) or be an outlier. (See picture for an example).
The da... |
Approximately bandlimiting, or maybe exactly if x is exactly - example:
rfft(abs(x)**2)[210]
(-7.993605777301127e-15+0j)
I notice the original signal must be bandlimited by a certain amount for this to happen, and there appears to be a definite relationship.
Is there a formula that relates fft(|x|^2) and |fft(x)|^2 ... |
Assume I have a FIR, stable and causal system. I want to know the deepness of non-causality on the inverse of my FIR system. It's obvious that the system is non-minimum-phase, since minimum-phase inverse is always stable and causal.
Definition: The system with $h(n)$ having non-zero elements on $$n> -L$$ have deeper no... |
I have a brain activity simulator that is capable of receiving various stimuli. Both generated signals and input stimulus are causal: a single sample is created every time step. I use a sinusoidal stimulus that I would like to CONVOLVE (not superpose) with Gaussian noise ~N(1,s).
I tried to multiply the stimulus spectr... |
I applied Butterworth filters (order=5) that have different cutoff frequencies to a simple signal composed of four sinusoidal oscillations and computed its PSD.
% Simulated signal
data = np.zeros((27000, 62))
t = np.arange(data.shape[0])
for ch in range(62):
for f in [1.2, 12, 24, 36]:
data[:, ch] += np.sin... |
Ignoring normalization factors, we can define the squared Root Mean Square (RMS) of a discret-time signal $x[t+n], n=0,1,...$, at time $t$ as:
$$\mathrm{RMS}^2_t = \sum_{n=0} ^{N-1} |x[t+n]|^2$$
which, by virtue of the Parseval's Theorem, can be shown to be equivalent to its frequency-domain analogue:
\begin{equation... |
The problem:
I have a signal that contains multiple, relatively stable frequency components and I want to extract only one of them. I attached a plot that illustrates the situation with a simple example. The top plot shows the raw signal and the lower one the spectrogram for the signal.
The emphasis here is on "relati... |
I want to find the impulse response of the linear time invariant system given by
$y''(t)+4y'(t)+3y(t)=x'(t)+2x(t)$
using time domain analysis.
As solved in the textbook "Continuous and Discrete Signals and Systems" by Soliman and Srinath (1990, Prentice Hall International), Section 2.5, page 86, the impulse response is... |
So I have an image that has horizental lines (noise), in other to get ride of them I worked with the Fourier Transform, however this is the result I get.
Here is my code:
figure, imshow(im3);
frequencyImage = fftshift(fft2(im3));
% Take log magnitude so we can see it better in the display.
amplitudeImage = log(abs(fre... |
Could anyone explain the action of interpolation control in symbol synchronization? What the underflows mean? And how it provides the fractional part? I know how other components work but don’t understand the process of interpolation control. I appreciate it if anyone could help me out.
Now my question is that the o... |
I'm doing the gain budget for a receiver that feeds into an ADC. I want to know what the best input range should be into the ADC so that I can detect down to the MDS (minimum discernible signal) while minimizing ADC spurious output content in my detection bandwidth.
Suppose for discussion sake that the ADC has a 1 volt... |
The real-time constant-Q transform are possible using sliding DFT (sDFT) because unlike FFT, sDFT can have arbitrary frequency scale and variable per-bin window size, and of course, the realtime CQT can also be more efficiently approximated using analog-style IIR bandpass filters. However, it requires the audio data st... |
I have two sensor data in time series as an array, when I plot we have this image. The peak you see in sensor 1 is something I am interested in classifying, It is easy to see that there is a similarity in both waves i.e when a heart beats the sensor 1 goes two peaks (up and down)
and immediately after some gap sensor ... |
A digital signal is commonly described as a signal which is discrete both in time and value/amplitude/magnitude.
I understand what is "discrete in time" (say, the signal is received each 1 second instead of continuously/sequentially or all the time), but I don't understand what is the meaning of the signal being discre... |
I'm confused about these terms: frequency shift, frequency offset, phase offset, and phase noise. My understanding that frequency shift and frequency offset are the same and caused by Doppler shift. Phase noise is caused by the instability of the local oscillator, and it changes per symbol. However, some books say this... |
I am trying to compute the MTF (modulation transfer function - Fourier Transform) of the following filter: $ [1, 1, 1] $. Here are my steps:
$$ X_k = \frac{1}{N} \sum_{n=0}^{2} e^{-\frac{2 \pi i kn}{3}} = \frac{1}{3} \left( 1 + 2\cos{\left(\pi k \right)} \cos{\left(\frac{\pi k}{3} \right)} - 2i \sin{\left(\pi k \right)... |
A superposition of two signals with different frequencies will never produce visible (i.e. stationary) interference patterns. Such waveforms will produce spatiotemporal beat patterns, but they rapidly propagate through space such that a stationary detector (e.g. screen or antenna array) never sees any interference fri... |
I made a previous post about what I am trying to do
Previous Post
I have band-limited data in the frequency domain from a VNA and I am trying to convert it into the time domain. VNA data saved in .s2p in Real/Imaginary format
VNA settings:
I extend the data to create a complete FFT. Zero padding is used for this proce... |
It would be greatly appreciated if the usage of the python package scipy's filter (e.g. butter) analog=True argument could be explained. I don't understand what is meant by this (any signal being processed by scipy in python on a computer is discrete and will always be digital?). I an pretty familiar with DSP but have ... |
I'm very, very new to this stuff, so please forgive me if the answer is obvious. I haven't been able to find any clear answers to this issue, even after looking through source code for various filter implementations and scouring the net for similar questions.
I have a function $f(x)=y$, such that:
$x$ is a frequency i... |
so I am a little bit confused about the formula of the step size in a uniform quantizer.
from what I have studied:
StepSize = $\displaystyle\frac{maxValue-minValue}{2^n}$, where n is the number of bits.
so for example, if we have a signal that changes from (-6 to 6) and the number of bits is 3 bits (also suppose it's m... |
A baseband analog message signal $x(t)$ with bandwidth 20 kHz, power
$10^{-3}$, and $|x(t)| \leq 1$ is waveform encoded and transmitted
through a channel of bandwidth 160 kHz. Besides, let the sampled data
of $x(t)$ be denoted by $x_n$ and suppose that $$ | x_n - x_{n-1}|
> \leq 10^{-0.55 f_s / f_N}, $$ where $f_s$ de... |
I'm making musical 'woodwind' type oscillators using digital delays with feedback, similar to these: A Meta-Wind-Instrument Physical Model, and a Meta-Controller for Real-Time Performance Control. I have successfully implemented my own delay/feedback loops using block diagrams programmed in C in the Arduino IDE (but am... |
I know that Kalman filter is optimal filter under some assumption like process and measurement noise are Gaussian. But if the process and measurement noise is non-Gaussian, the estimation of the Kalman filter will start to be inaccurate, especially with noisy measurements. Can you suggest adaptive Kalman filter algorit... |
I want to be use kalman filter for the state estimation but I don't know about process and measurement noise. How can I estimate process and measurement noise and use this information for kalman filter?
|
Does someone know, how to convert the Sweep signal into Impulse response
I am testing that comparing software and hardware compressor with sweep signal
I generated a sweep signal from REW software and then correct the output value sweep signal from the compressor.
But i lost the way that how to convert the sweep signal... |
Given the DFT magnitude vector of an 1-D image, I want to understand if we can calculate the size and pitch of repeating patterns in the image. Is this possible? I took a few test images and calculated DFT magnitude using openCV https://docs.opencv.org/3.4/d8/d01/tutorial_discrete_fourier_transform.html. Then I tried t... |
I am currently working on a guitar training progam that compares guitar input from a guitar to a song as you play, to give a score of how well you're playing. The reference song is converted from MIDI to a waveform. The user input is taken in by cable as an audio stream.
I want to compare the user input to the referenc... |
https://www.quora.com/How-is-signal-filtering-done-with-convolution
What is the relation between convolution and a filter?
Is it mainly that convolution process is used to determine the output of an LTI system and since filters are also LTI systems?
A related statement mentioned in above link and also shown in attached... |
I need to summarize a set of samples representing an audio to achieve a smooth zoom animation for a spectrogram in my app.
In the time domain (waveform) I achieve this by resummarizing the samples each time the zoom changes, where my current summarization goes like this: I know the audio's sampling frequency and I have... |
Does someone know, how to convert the sweep signal into the impulse response?
What kind of results will come out?
I generated a sweep signal from REW software and then correct the output value sweep signal from the compressor.
How can I convert the sweep signal into an impulse response with MATLAB?
|
Considering a finite-length impulse response $h[n]$ of length $M $ (which amounts to considering $h[n] = 0,$ for $n \geq M$). The data model, with additive disturbance, is then written as:
$$y[n] = y_M[n] + \epsilon[n]$$
where
$$y_M[n]= \sum_{k=0 }^{ M-1} h[k]x[n-k].$$
We seek to estimate the impulse response $h[n]$, i... |
I am currently working on a project which requires evaluating slow changes over time in the magnitude of a signal's high frequency (over 10 kHz) components.
This signal was acquired in tests with a 25.6 kHz sampling rate and in "batches": every minute, one second of data was acquired. This was done in order to save sto... |
This question is perhaps related to the semantics or jargon of signal processing. I have checked plenty of advanced books/monographs on multivariable calculus and signal processing but could find an explicit answer.
Suppose we have a simple function $y$=$\sin(x)$. In short, if we have sampled a sine wave as a function ... |
I am trying to understand a MATLAB code of convolution animation.
Please check above link especially code in lines 15 and 16. I think author has mistakenly swapped comments of line 15 and 16. Please correct me if i am wrong
I am also placing the code here
close all
clear all
clc
% Initialization
t = -400:400; % Time r... |
I have the transfer function: $$H(z) = \frac{z^2 + 0.75z + 0.125}{z^2+0.5625}, |z| > 0.75 = \frac{(z-0.5) (z-0.25)}{(z - 0.75j) (z + 0.75 j)}$$
I attempted partial fraction expansion in order to use my lookup table for z-transformations. The issue is my poles are complex conjugate pairs. I have never had to deal with a... |
I have a touch signal on a plate recorded with 3 accelerometers with a sample rate of 150kHz over a duration of 5 seconds (so 750k samples). The goal is to localize the position of the touch in the end. But for that, I need to separate the reflections from the direct wave. And therefore need to do some signal analysis.... |
I have a simple $0-40 \texttt{Hz}$ IIR bandpass filter.
With pure sine inputs I see $60 \,\texttt{Hz}$ is completely gone in the output FFT as expected by looking at the filter frequency response.
However I fed it with non sine signal+noise (i.e real world recorded signal), and I see the FFT shows a peak at $60 \,\text... |
I'm implementing 2FSK and 4FSK modulation schemes for an experimental system. I'm using GNURadio, although that shouldn't affect the question.
I'm trying to figure out how the center frequency should be selected. Many 2FSK examples use a center frequency of 1700Hz, with a deviation of 500Hz, for a "space" frequency of ... |
I have an audio file that I encrypted and decrypted using the AES algorithm.
After the decryption is complete I output the audio file to a new file.
I wanted to calculate the root mean square between the original file and the newly decrypted file. It came out as 0. Why is that? Shouldn't the processing have introduced ... |
I have an In-phase signal collected by an FMCW radar. I am trying to reconstruct the analytical signal by applying the Hilbert Transform to the In-phase signal.
Are there certain conditions that need to be satisfied in order to correctly, reconstruct the analytical signal using Hilbert Transform ? If yes, could you lis... |
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