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16 views

Help with literature pointers for improving estimation of underlying mean for signal with periodic noise

I am in the starting phase of a master thesis and are having some trouble finding a good starting point in the literature for my thesis. The aim of my thesis is to provide as good of an estimate as ...
0
votes
0answers
17 views

Sampling alternatingly quadrature components

Suppose, I sample a signal $z=\mathrm{e}^{\mathrm{i}\omega t}$ alternatingly in quadrature components $I=\mathrm{Re}(z(t))$ and $Q=\mathrm{Re}(\mathrm{e}^{\mathrm{i}\alpha}z(t))$. Thus my data array ...
1
vote
1answer
26 views

Train of impulses in frequency domain

Why compressing a train of impulses in time domain produce a wide-spaced train of impulses in frequency domain and vice versa? i want to know the intuition behind that.
1
vote
1answer
20 views

How can extract the cosine transform formula used for 2D by scipy.fft.dct

For the 1D cosine transform the documentation is clear in here, and I can reproduce it easily: The formula is: $$y_k= 2 \sum_{n=0}^{N-1}x_n \cos \left( \frac{\pi k(2n + 1)}{2N}\right)$$ and here it is ...
0
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0answers
12 views

What is the use of quasi logarithmic frequency distribution in multisine frequency signal in finding FFT

I'm using multisine but recommended to use quasi log frequency distribution to find out FFT. What is the advantage of using this?
0
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0answers
11 views

SVD versus PCA in signal analysis [closed]

What is the exact difference between Singular Value Decomposition and Principal Component Analysis (PCA)?
1
vote
1answer
16 views

Performing root-music, how do i retrieve frequency and phase data of a signal?

I'm trying an alternative method to the classical fft to do signal analysis. I perform root-MUSIC on a certain known signal + noise in matlab with the function 'rootmusic'. This gives gives the signal ...
0
votes
1answer
14 views

How to get a non-equally spaced FFT back into the time domain

I have a signal that I STFT and then filter using an ERB spaced filterbank. At some point after this I want to get the signal back into the time domain, how can I go about this? Using a standard iSTFT ...
0
votes
1answer
21 views

Bio-signal discontinuity after FIR filtering

I'm filtering respiration signal. I have a signal of duration equal to 10 minutes, so I decided to divided it into period of 10 s. I filtered each epoch with FIR filter, but I have discontinuities ...
0
votes
1answer
30 views

FFT of a stretched vector

Lets say I have a small size vector x=[a b c d]. Now I stretch this vector 3 times and I got x3=[a a a b b b c c c d d d]. What would be the relation between fft(x) and fft(x3)?
0
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0answers
25 views

Finding a signal piece presence in a noisy signal [closed]

I'm working on a problem where the original signal X has been trimmed in the parts and pieces of another noise signal E has been put inside. X = [x1, x2,...,xn] Y = [x1, e1, x2, e2,...,xn, en] I have ...
0
votes
1answer
26 views

Amplitude decrease in FFT at higher frequency for multisine

I am adding sine signals with different frequencies but same amplitude=100. Sampling frequency is 50MHz. I am not getting 100 in frequency domain for frequency 787KHz. At 1K, 3K I am getting 100 ...
2
votes
3answers
83 views

Implementation of CIC filter using floating point

I am trying to implement a CIC filter and was reading through the literature and found that CIC filter cannot be implemented using floating point. I am unable to understand why that would happen. The ...
0
votes
2answers
39 views

FFT padding to power of 2

I have a vector v1 of size N1, I want to pad it in a certain way until it reaches a size power of 2 (v2 of size N2). I need that the first N1 coefficients of the FFT output to stay the same as when I ...
0
votes
0answers
10 views

Source IO size “8” does not match sink IO size “4”

This is my first attempt at using GNU Radio. I am trying to do this example, which is a sine wave generator. The example works, so I am trying to connect the throttle block to the audio sink block. ...
0
votes
0answers
13 views

Python library for audio filtering, gating, and sound/silence detection

Is there a Python library that can do all of the following?: Band-pass filter; Noise gate, with different parameters set for different segments of audio file; Detect and count the number of discrete ...
0
votes
1answer
25 views

Equalization in CAR- Automotive Audio

Can someone tell me what is the purpose of CAR Equalization in audio and why before CAR Equalization ,gain adjustments are necessary ? Also why would someone use FIR filters during CAR Equalization ...
0
votes
0answers
26 views

Frequency response of chirp signal changes when extracting burst from repeating chirp waveform

I hope i can make myself clear enough, even though i might not be familiar with the right terms. Using GNU Radio, i created a chirp signal with a sawtooth signal source into a frequency mod block, as ...
0
votes
2answers
26 views

Digital Signal Quantized Processor Complexity

In “traditional” digital signal processing, the complexity is computed as a number of multiplication the operation requires, e.g. the computation of the N-point DFT via the decimation-in-frequency FFT ...
0
votes
0answers
20 views

Sampling and Demodulation question

if I have an analog modulated signal with a DSBSC modulation (modulation index 100%) as an input in my receiver and I want to convert it to a sampled signal, do I need to demodulate the signal (with ...
1
vote
1answer
26 views

Phase of the sine signal by finding FFT of it in matalb using angle

I am using sine signal with phase=0 an frequency=100KHz. Computed FFT of it. Using angle function and multiplied the result with 180/pi; But getting 90.36 at 100KHz frequency. Can anybody explain why ...
0
votes
1answer
34 views

Balance Fader in Digital Amplifiers

What is the purpose of Balance / Fader in a digital Audio amplifier? In a signal flow what is the most apt position to place balance and fader(for example after before a compressor unit or at the ...
0
votes
0answers
29 views

How to solve this question on SSB modulation? [closed]

A message signal $m(t)$ has components at frequencies of 100, 200, and 400 Hz. It is SSB-modulated on a carrier with $f_c = 100 kHz$. The resulting passband signal $u_p(t)$ is demodulated by first ...
0
votes
1answer
14 views

Data smoothing in temperature measurements from MLX90614 BCC IR sensor

I am trying to measure temperature with the MLX90614 BCC IR sensor for my project, and I am having problems with my precision requirement. I had taken data for 1 second then took the mean of these ...
0
votes
0answers
14 views

How to solve this question on communication lines?

The Question asked is as follows: A communication link operates at a bitrate of 20 Mbps. The receiver is expected to perform at a $BER$ of $10^{-6}$ or better using $QPSK$ modulation scheme. [Note: $...
0
votes
1answer
29 views

Unit step function symbol

I've been studying communications and i see that the book used the symbol $u_{-1}(t)$ for unit step function , why ?
0
votes
0answers
21 views

Taking fft of real values of positions and momenta for a 1-d fermi pasta ulam problem

I am trying to understand the code that implements taking the fft for a 1-d FPU problem. Y is a matrix of real values by real ...
0
votes
0answers
9 views

ECG SIGNAL ANALYSIS USING ADAPTIVE THRESHOLD

I am trying detect the R-peaks in ECG signal using adaptive thresholding method in MATLAB. I have done till preprocessing of the signal and find the local maxima of the signal but I am unable to ...
1
vote
1answer
33 views

Why does my zoom audio, when there are nearby microphones, seem to usually take an open loop format?

The signal from the speaker on device A gets into the microphone from device B, then the amplifier and then the speaker from device B, back into device A. The result is a sort of runaway signal that ...
0
votes
1answer
34 views

Nyquist Frequency and Window Length

The Nyquist theorem states that the sampling rate must be twice the highest frequency to be observed. How does the length of the interval play into this relationship? The main resource that I found ...
0
votes
1answer
21 views

What is the RFFT and iRFFT complexity?

What is the RFFT and iRFFT complexity for 1D and 2D? Here is some starting point. 1D RFFT The FFT is $\mathcal{O}(N\log N)$ RFFT computes $N//2 + 1$ frequencies instead of $N$. Thus, RFFT is $\...
-1
votes
1answer
26 views

Processing signal [closed]

Processing signal , Sampling of Continuous-Time Signals
0
votes
1answer
25 views

What happens to the audio of a reconstructed signal if we remove too much frequency?

Assuming we have an audio with noise embedded in it, I was wondering what happens theoretically if we remove (filter out) too much frequency when we pass LPF with a cutoff frequency lower than the ...
1
vote
0answers
30 views

Accelerometer Bias estimation with kalman filter

I have to estimate biases of a 3-axes accelerometer by modyfying an existent kalman filter mounted on a drone. The biases are assumed constant. The filter has 9 states: position (xyz), velocity (xyz) ...
1
vote
1answer
25 views

What phase rotation occurs when you take the derivative of an audio signal?

If you take the derivative of an audio signal, it provides a 6 dB/oct upward sloping filter (increasing high frequencies / cutting low frequencies) all the way across the spectrum. What is the result ...
4
votes
3answers
151 views

How to find peak value of an analog signal efficiently after sampling in the digital domain?

I have a bandlimited analog signal for which I want to find the peak value in real time. The signal is sampled and processed digitally at just enough sampling rate. Since the peak of the analog signal ...
1
vote
2answers
987 views

Does the Shannon theorem not apply when the amplitude of a wave is changed faster than half the time period of the wave?

Shannon's version of the sampling theorem states that if a function contains frequencies all strictly less than $B$ hertz, then it is completely determined by giving its ordinates at a series of ...
1
vote
1answer
34 views

What is meant by “omega suppression” in a lock in amplifier context?

Can anyone explain what is meant by "Omega suppression" in lock-in amplifier discussions or specifications. I'm usually very good at web-searching, but have been unable to find anything ...
0
votes
1answer
29 views

Digital Filters

If I want to remove the baseline drift in my ECG signal, which digital filter should be used without distortion and shift in my filtered output? What are the necessary things I have to look for the ...
0
votes
0answers
5 views

Which Image pre-processing method is the best for Histopathology images? (H and E stained image)

I am working on classification of breast cancer histology images (Dataset:ICIAR BACH challenge 2018). But I don't really know which method is the best way to enhance the features of H&E stained ...
1
vote
0answers
39 views

Short Time Fourier Transform has different frequencies than Fourier Transform?

The reason we do the STFT is so that we can analyse for short segments of time how much of the components in the frequencies of the FT are present. However, it may be possible that completely ...
0
votes
1answer
20 views

Magnitude and phase of frequency response from Z-transform

I have been given a signal which I have completed the Z-transform for and have got: $H(z)=1-z^{-1}$. I need to find the magnitude and phase of this transform but am unsure on how to actually do this. ...
1
vote
0answers
51 views

QPSK demodulation

My objective is to demodulate QPSK signal. At this moment, my constellation in the best case looks like in the picture below: and the corresponding eye-pattern: Eyes are completely closed, which ...
1
vote
0answers
19 views

In OFDM, does the parameter time offset is important if there is only one mobile in the cell?

I am learning about OFDM, and in my reading, I have noticed that (mentioned everywhere) synchronization is a very critical aspect of OFDM. Here, I am writing in the context of time synchronization (...
0
votes
1answer
28 views

What is the difference between sampling over $-4\pi$ to $4\pi$ and 0 to $8\pi$

I see no difference in magnitude, but the phase plots vary significantly while using NumPy and MatPlotLib to plot. Looking at the phase plots, I feel like I'm missing something important. Also, I'm a ...
1
vote
2answers
45 views

Analysis of the ratio of two moving average filters

I have created a basic segmentation algorithm for 1-D signals (e.g. audio) based on a ratio of energy averages. The classification rule involves comparing the signal energy measured over a short time ...
0
votes
3answers
126 views

There is an implant in me, is there a way to identify the frequency it is receiving signals on?

There is an implant in me, is there a way to identify the frequency it is receiving signals on? I don't want to share the evidence and my reasoning for believing there is an implant in me, you can ...
0
votes
1answer
39 views

Reconstructing input from the output of a known discrete-time IIR filter

I have known discrete-time IIR filter. I pass a signal x(n) through the filter and obtain the output y(n). How would I design a filter to obtain x(n) from y(n)? In general, is it possible to design ...
-1
votes
0answers
33 views

get DFT from impulse response

I've been trying to get DFT from an impulse response, such as $(n+1)[u[n]-u[n-5]]$, but I don't have any clue of where I should start from Thanks
1
vote
1answer
25 views

Moving average frequency response over an image

I'm studying image denoising by linear filtering with cross-correlation, in particular with a moving-average kernel (K x K kernel of all equal elements which sum is 1). For clarity, I'd like to refer ...

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