8
votes
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Are IIR filters (and specifically Butterworth filter) causal?
Yes, Butterworth are IIR. The decay from an impulse technically lasts forever.
Yes, all [implementable] IIR are causal.
Yes, because of #1 and #2.
Don't use ...
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7
votes
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Convert RGB Image to Grayscale and Display It (Python + Matplotlib)
You need to set the cmap properties of the imshow() function:
https://matplotlib.org/devdocs/api/_as_gen/matplotlib.pyplot....
- 39.3k
6
votes
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Why is Scipy implementation of Hilbert() function different from Matlab implementation of the function?
It works fine for me:
...
- 14.9k
6
votes
Accepted
Designing digital low pass filter with low pass-band group delay
Minimum phase filters will not give you a near constant group delay. You can design a non-linear phase FIR filter with a linear desired passband phase with a specified group delay that is smaller than ...
- 79.2k
5
votes
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Calculating values of frequency bins in Python
This is simply how Discrete Fourier Transform (i.e. Fourier Transform theory applied on sampled signal) works.
You get an output of length N if your input has ...
- 191
5
votes
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How to generate random samples of Gaussian distribution directly in the frequency domain?
You can, but... you'll need to keep symmetry if your original time-domain signal is real-valued.
If a signal $x$ is real-valued, then its DFT $X$ will exhibit complex-conjugate symmetry:
$$
X[k] = X^*...
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5
votes
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How to Apply Gaussian Filter to Raw Audio Files in Python?
Do it using the following steps:
Look for the optimal frequency for the Low Pass Filter. Usually close as possible to the bandwidth of your desired signal.
Design an LPF filter according to the ...
- 39.3k
5
votes
Accepted
Scipy resample, "fourier method" explanation
Suppose you have initially a real-valued sequence x of length N. The function is basically doing this:
To upsample, it ...
- 4,842
4
votes
Issue with the time vector returned by $\tt signal.spectrogram$ function
The default parameters of signal.spectrogram are:
nperseg = 256
noverlap = nperseg/8 = 32
This means that:
The length of ...
- 10.4k
4
votes
Butterworth band pass filter
You need to specify your filter design specifications parameters consistently for either an analog or a digital filter. With your posted code, the butterord ...
- 1,732
4
votes
Welch spectrogram
Reading the documentation for scipy.signal.spectrogram I noticed that it does not do any kind of periodogram averaging. It simply splits up the signal into (...
- 4,004
4
votes
Do DCT results contain phase spectrum?
A DCT is equivalent to a DFT of real data that is doubled and mirrored, thus rendering it symmetric. The DFT of any symmetric real signal has a phase of zero (its all cosines, no antisymmetric sine ...
- 33.8k
4
votes
Do DCT results contain phase spectrum?
Not really, as the transform is real. However, one could interpret the sign as a poor man's phase, being "quantized" or restricted to values $0$ or $\pi$. In other words, $1 = 1.e^{0.\imath}$ and $-1 =...
- 29.6k
4
votes
Decreasing Sample Rate in DFT (FFT) for Audio Analysis
There are 2 different things here that we are mixing:
Bin Width
As you mentioned, the bin width is a function of the ratio between the sampling rate and the number of samples. If you increase the ...
- 39.3k
4
votes
Checking Parseval's Theorem for Gaussian Signal by Using Scipy
There is a simple idea behind Parseval's Theorem.
Unitary transform preserves the $ {L}_{2} $ Norm.
The Fourier Transform (All its variants) are indeed Unitary when the correct factor is in place.
...
- 39.3k
4
votes
Delay in Savitzky-Golay filtering
Standard Savitzky-Golay filters are linear phase (type I) FIR filters. So they have an odd number of filter coefficients $2N+1$, and the delay equals $N$.
For a good overview of Savitzky-Golay ...
- 79.2k
4
votes
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Apply FFT to positive spiky signal
When calculating the frequency correctly as the inverse of the period, instead of the inverse of the period times 2 pi, the fft does indeed give the expected answer.
- 143
4
votes
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Invertible low-pass (Butterworth) filter in python (scipy)?
Is there an invertible low-pass filter
No
is there something particularly difficult about inverting a low-pass filter?
Yes. Digital low pass filters (in the most common sense) have a zero at ...
- 30.6k
4
votes
Accepted
IIR design: SciPy CMSIS-DSP coefficient format
scipy.signal returns a sos matrix when you set output='sos', which is cascaded second-order sections and has a shape of ...
- 2,350
3
votes
Accepted
Python different autocorrelation with FFT and non-FFT
By converting to np.int16, you're hitting integer overflow and wrapping the values around. I assume this is not intentional, since the output looks much more sane ...
- 14.9k
3
votes
Why is Scipy implementation of Hilbert() function different from Matlab implementation of the function?
You have to detrend the signal. If you have a DC component the behauviour changes completely.
...
- 41
3
votes
Morphing between two FFTs
maybe you can do it from bottom to top or top to bottom using something like the following.
a notation about notation:
$x[n]$ is the input sample for the $n$-th sample. this originally comes from ...
- 16.7k
3
votes
Accepted
How to filter out those outliers
Have you tried just median filtering the data ? That's what I'd try as a first step. Winsorizing really doesn't work on a data set that has a changing mean.
It looks like python has ...
- 21.7k
3
votes
Accepted
How to calculate the PSD from the complex calculated STFT?
In general, if you have complex spectrum and need PSD in dB the mathematical equation is
$$P_{xx} = 20\cdot\log_{10}|X_{x}|,$$
where $P_{xx}$ is your PSD in dB and $X_{x}$ is your complex STFT ...
- 76
3
votes
How to calculate the PSD from the complex calculated STFT?
I normally use the Welch method to calculate and plot the PSD of a signal.
...
- 63
3
votes
Why does a higher sampling frequency mess up my bandpass filter?
With an IIR filter, the higher the ratio is of sampling frequency to bandpass filter frequency, the closer the poles are to the complex unit circle in the Z-transform. Poles close to the unit circle ...
- 33.8k
3
votes
Accepted
Why does a higher sampling frequency mess up my bandpass filter?
As others said in the comments, this looks like numerical error. 3rd-order filters are not typically prone to this, but the higher your sampling frequency, the closer the poles move to +1:
You might ...
- 14.9k
3
votes
Accepted
How to convert a spectrogram back to a signal
I am not a good Python coder, but did similar processing in Matlab in the past. The subject has been discussed in SE.DSP in several instances, for instance Librosa stft + istft - Understanding my ...
- 29.6k
3
votes
Python Hilbert does not seem to calculate the envelope
Hm well, technically it is some kind of envelope: it oscillates between hilbert(x) and -hilbert(x). Your examples (dashed lines ...
- 2,268
3
votes
What Are The Semantics Of Wav-File Sample Values?
I don't know much about this semantics? of WAV files but their numerical format is the following. (assuming mono)
Given a recording with 8-bit per sample precision, then those samples are unsigned ...
- 26.5k
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