已回答
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...

8 years 前 | 0

| 已接受

已回答
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...

8 years 前 | 0

| 已接受

已回答
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 ...

8 years 前 | 0

| 已接受

已回答
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, ...

8 years 前 | 0

已回答
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...

8 years 前 | 0

| 已接受

已回答
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...

10 years 前 | 0

已回答
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...

10 years 前 | 0

| 已接受

已回答
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...

10 years 前 | 0

已回答
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...

10 years 前 | 0

| 已接受

已回答
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 ...

10 years 前 | 0

| 已接受

已回答
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...

10 years 前 | 0

已回答
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...

10 years 前 | 0

| 已接受

已回答
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...

10 years 前 | 0

| 已接受

已回答
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...

10 years 前 | 0

| 已接受

已回答
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

10 years 前 | 0

已回答
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...

10 years 前 | 0

已回答
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...

10 years 前 | 0

已回答
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...

10 years 前 | 0

| 已接受

已回答
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...

10 years 前 | 1

| 已接受

已回答
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 ...

10 years 前 | 0

| 已接受

已回答
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...

10 years 前 | 0

| 已接受

已回答
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 ...

10 years 前 | 1

| 已接受

已回答
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 ...

10 years 前 | 0

已回答
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...

10 years 前 | 0

已回答
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...

10 years 前 | 0

已回答
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...

10 years 前 | 0