Noise Comparison in MATLAB - matlab

Assuming I have two signals (raw data as excel file) measured from two different power supplies, I want to compare the noise-levels of these signals to find out which one of them the noisier one. Both power supplies produce signals with the same frequency but the amount of data points are different. Is there a way to do this in MATLAB?

You could calculate the signal-to-noise ratio for each signal. This is just a ratio of the average signal power and average noise power, usually measured in decibels. An ideal noiseless signal would have SNR = infinity.
Recall that signal power is just the square of the signal amplitude, and to get the value x in decibels, we just take 10*log10(x).
SNR = 10*log10( mean(signal.^2)/mean(noise.^2) );
To separate the signal from the noise, you could run a low-pass filter over the noisy signal.
To get the noise you could just subtract the clean signal from the noisy signal.
noise = noisy_signal - signal;

Related

STFT(Short-Time Fourier Transform) & DFT(or FFT) in Matlab

I want to try STFT & FFT using Matlab. What I wonder is
STFT of signal computes the result that FFT(DFT) of each windowed signal and I can see the change of each frequency value over time. If I calculate the average of each frequency over the total time, can I get the same amplitude result with the result of the FFT(DFT) of the whole signal?

SNR of original signal and quantized signal - MATLAB

If I wanted to get the SNR of a quantized audio signal, would I subtract the quantized signal from the original (which would give me the noise?) and then calculate the power of this noise and compare it to the power of the original signal?
Such as, SNR(dB)=10*log(original power/noise power)
Yes, that is the quantization noise. It is not just “noise” because the input has noise as well, and you don’t include that noise in your result.

How To Make Noise Signal After Estimate Noise Variance

For Blind Source Separation (Adapative Filtering , LMS Algorithm), i need two input. a)noisy signal, 2)noise signal. But How can i make the noise signal. If i can estimate the noise variance of noisy signal, then how can i make a noise signal from noise variance in matlab. I am new in signal processing.
try this (matlab):
noise = normrnd(mean ,sqrt(variance) ,rows ,columns);
it will generate random numbers from the normal distribution (mean,variance).
rows and columns will dictate the result matrix dimensions.

FFT plot in Matlab

I have a small input signal of 60Hz sine wave from signal generator, which is corrupted with 50Hz mains supply frequency. I want to measure the amplitude of the 60Hz signal using FFT because it is very small to see in the oscilloscope.
The Matlab FFT code:
y = data;
Fs = 2048;
[r, L] = size(y);
NFFT = 2^nextpow2(L); % Next power of 2 from length of y
Y = fft(y,NFFT)/L;
f = Fs/2*linspace(0,1,NFFT/2+1);
% Plot single-sided amplitude spectrum.
plot(f,2*abs(Y(1:NFFT/2+1)))
But the FFT plot doesn't give a sharp peak at 50 and 60Hz. The plot looks like this:
The consecutive points have high and low amplitude alternatively which gives a saw-tooth like plot. Why is it so? Is the amplitude of 60Hz affected by this?
Probably there are two effects
If one measures a time window of a signal, this leads unavoidable to a phase gap between the start and endpoint of the signal modes. The FFT of a gap or a rectangular signal causes high frequency oscillations. These oscillations caused by border effects can be damped due to a window function, which smooths out the signal to the borders.
There is a discrete frequency spectra in DFT. If one measures a signal which does not match to any of these discrete modes, a lot more frequencies are necessary to reconstruct the original signal.
Your 50 Hz signal may not be a pure perfect sine wave. Any differences from a perfect sine-wave (such as clipping or distortion) is equivalent to a modulation that will produce sidebands in the spectrum.
Windowing a signal whose period is not an exact sub-multiple of the FFT length will also convolve windowing artifacts with that signal.

How to calculate user defined SNR, given only sampling frequency and a vector containing signal samples?

So I have the samples (hex values) of a sinus signal and I know the sampling frequency. Using this I can plot an fft or periodogram but then I would like to find out the SNR ratio. What would be the most accurate way to calculate the noise and signal power? I would prefer doing it in frequency domain. Is there a way to do this also in time domain?
Thanks a lot in advance!!!
So if there is noise on your signal and you know that your underlying signal is a sine wave, you can easily get your signal parameters i.e. amplitude,frequency and phase by least squares. If y(t) is your signal just minimize the L2 norm of (y(t)-A.sin(wt+b)) over A,w and b. Then you can easily get signal power from the underlying signal and the noise power from the error signal (y(t)-A.sin(wt+b)).