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How to ignore NaN values in pcolor function?
I don't think the black lines are because of the NaNs. Its coming from the default shading method, which is faceted. Try "shad...

presque 9 ans il y a | 0

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Efficient method for getting positive half axis only from FFT
If I understand your question, you are getting the answer you want, but you are trying improve the efficiency. If you only care...

presque 9 ans il y a | 0

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Nested for loop outputs
I'm not sure if entirely understand the question... why not just do this for the last block? for j = 1000 iterations ...

presque 9 ans il y a | 0

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Eigen values differs with mchines
high condition number is not the same thing as non-deterministic. If you repeat an ill-condition computation multiple times, ...

environ 9 ans il y a | 0

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indices for zero crossing of a Sine Function
First question: why is there an extra one at end? Because of the way you handled the "missing" point at the end: drvY = [drv...

environ 9 ans il y a | 0

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wave phase adjustment between several signals
I think xcorr() may help here. xcorr() will basically try every possible time shift for you, and let you know what shift makes t...

environ 11 ans il y a | 0

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FFT don't give correct result
first, "f = Fs*linspace(-.5,.5,length(data));" is not exactly how matlab outputs the frequency vector. To get something like th...

environ 11 ans il y a | 0

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FFT analyzing of signal and signal reconstrution. (I want to reconstruct by suming a*sin(w*t-phi)
You forgot the negative frequencies. You didn't state explicitly the definition of simpleamp, but I think length(Simple) should...

environ 11 ans il y a | 0

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if the number of samples for doing fft is not n power of 2 the results accuracy would be affected or it is just matter of speed of calculations ?
There are two reasons to zero pad an fft. First reason is speed. A sample size that is a power of two is going to be the faste...

environ 11 ans il y a | 0

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how to generate a Sine wave with changable frequency in mfile?
the short answer is that you cannot just multiply f by t to get what you want. You need to integrate f with respect to time to ...

environ 11 ans il y a | 0

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how can I recover original frequencies from a data table acquired from osciloscope?
well, fft() function is a good start. I'm guessing if you are asking this, then you don't know much about fft() function. This...

environ 11 ans il y a | 0

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Power Spectral Density for HRV, need help with pwelch function.
if your input is in volts, then your output is V^2 / Hz. If your inputs are in some other unit, let's call it X, then your outp...

plus de 11 ans il y a | 0

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power spectral density of fft
well, in some sense, you can never really "get" the PSD of anything. you can only *estimate* it. And yes, the fft magnitude sq...

plus de 11 ans il y a | 0

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How do I perform zero phase filtering in Matlab?
First, your line "freq = 0:binHzConv:100;" is not correct. Frequency does not range between 0 and 100 Hz in your example, but r...

plus de 11 ans il y a | 0

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How to create impulse noise? (e.g. interference in a cable plant)
this reference might be a good start http://dsp-book.narod.ru/305.pdf

plus de 11 ans il y a | 0

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Need Help with FFT
Here's a little tutorial I wrote that may help get you started. http://www.mechanicalvibration.com/Making_matlab_s_fft_functio.h...

plus de 11 ans il y a | 0

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How can I differentiate without decreasing the length of a vector?
My standard approach is to use 2nd order centered difference for the main part of the vector, and use first order forward and ba...

plus de 11 ans il y a | 0

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How to choose the appropriate low pass filter?
if you are new to filtering, and just want a simple low pass filter to remove noise, then start with a butterworth filter. It m...

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Questions about FFT (and applying it to determine power spectral density)
"Question 1: Why isn't the scaling of ( 2 / numberOFdataPOINTS ) applied within the FFT algorithm." Matlab uses what I like t...

plus de 11 ans il y a | 1

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FFT accuracy and 'Noise Floor'
Standard double precision floating point arithmetic is good to about 16 significant digits. So trying to do what you are doing ...

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How can I determine the maximum frequency or bandwidth of a coded audio signal?
Well, on the one hand, for any digitized signal, the maximum frequency that can be represented is half the sampling rate. B...

plus de 11 ans il y a | 0

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Why is spectrum.periodogram not recommended, and how to substitute pwelch in it's place?
Well, first, is your signal a random variable? Or more of a periodic signal (i.e. a sine wave)? If it's periodic, then just a ...

plus de 11 ans il y a | 1

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How can I write a function about upsampling in DSP
resample() work might for you to upsample the data. Filtering in matlab is a two step process. First, you have to design the ...

plus de 11 ans il y a | 0

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convert time from sample to second
Well, if the time step is 0.001s and you have 300 samples, then 300 * 0.001 = 0.3, which is not 10s. So either you don't really...

plus de 11 ans il y a | 0

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FFT of NMR signal: frequency cut off
Hi Nicolai. I'm not sure exactly what you are expecting to see, but I suspect you just didn't multiply by 2 when converting f...

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Temporal waveform reconstruction from spectral magnitude only
Without the phase data, it's basically impossible to reconstruct the original time domain signal. Sorry. Cepstrum analysis wil...

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