Here in this code i am doing a stft on my wav-file. There is no problem with that. At the beginning, i am defining my parameter, afterwards using my wav file and then applying the stft. Basically what i am doing is a real-time spectral analysis. Anyway my question is, how do i a frequency band? I want my signal to be separated in LOW/MEDIUM/HIGH. I want my vector to be saved, from 0-250 Hz in the LOW-Band, 250-5000 Hz in the MEDIUM-Band, 5-22.05k Hz in the HIGH-Band. I advise you, to try my code in Matlab, if you don't understand it. Just take any wav-file. Btw my signal is plotted in the variable "Yres". Any solution is appreciated!
NFA=2; % Number is used for plotting every 2nd picture
t_seg=0.05; % Length of segment in ms
fftlen = 4096;
% Lenght of "fft",because our segment contains 2205 points
[y,fs]=audioread('UnchainMyHeart.wav');
% audioread = functions reads WAV-file
% y = A vector which contains my audio signal
% fs = sample frequency (44100)
% 'UnchainMyHeart' = WAV-file
t=linspace(0,length(y)/fs,length(y));
% linspace = Creating time vector
% 0 = Start time
% length(y)/fs = End time
% length(y) = Number of samples in y
plot(t,y)
% plotting signal in the time domain
segl =floor(t_seg*fs);
% Applying fft function on the variable "segl"
windowshift=segl/2;
% Defining the size of the window, which goes to the next "segl"
window=hann(segl);
% hann function
window=window.';
si=1;
%Start index
ei=segl;
%End index
AOS= length(y)/windowshift - 1;
% AOS is the number of "segl" we use (About 433)
f1=figure;
% Opening new window
f=0:1:fftlen-1;
f=f/(fftlen-1)*fs;
% Defining frequency vector
Ya=zeros(1,fftlen);
plot(f,Ya),axis([0 fs -90 50])
grid on
n=0;
%start variable
for m= 1:1:AOS
y_a = y(si:ei);
y_a= y_a.*window;
Ya=fft(y_a, fftlen);
n=n+1;
if n==1
Yres=abs(Ya);
else
Yres=Yres+abs(Ya);
end
if n==NFA
Yres=Yres/NFA;
n=0;
drawnow;
%Tut die Grafikobjekte immer auf den neuesten Stand updaten
figure(f1);
plot(f(1:end/2), 20*log10(abs(Yres(1:end/2))));
ylim([-90 50]);
title('Spektrum eines Audiosignal');
xlabel('f(Hz)');
ylabel('dB');
grid on;
end
si=si+windowshift;
% Updating start index
ei=ei+windowshift;
% Updating end index
end
This may not be the best answer! But this may help you get started on something. You can use spectrogram() function from MATLAB's Signal Processing Toolbox.
Let's suppose you have an audio file named ''UnchainMyHeart.wav'(in your case) with one channel. The code goes as follows:
% Reading the audio file
[y1,fs] = audioread('UnchainMyHeart.wav');
% Parameters for STFT (or spectrogram)
windowDuration = 30e-3; overlapDuration = 15e-3;
windowLength = round(windowDuration*fs); % window length
overlapLength = round(overlapDuration*fs); % overlapping of windows
nfft = 1024;
% Executing STFT for the signal
[S1,F1,T1,P1] = spectrogram(x1,hanning(windowLength), ...
overlapLength, nfft, fs, 'yaxis');
S1 and P1 contain STFT and Power Spectrum Density(PSD) of the signal for a time interval of each section with a time interval whose estimations are contained in T1.
For your question, you are looking for F1 which is a vector of cyclical frequencies expressed in terms of sampling frequency, fs. For example: if you have a sampling frequency of 48 kHz (fs) and nfft of 1024, then you will have 513 [(1024/2) +1)] frequency values spaced by (fs/nfft). i.e. 46.875. So your frequency components will be 0, 46.875, 46.875*2, ..., 46.875*512. The maximum you will have is 24 kHz due to Nyquist criterion.
Now, you can easily write a simple routine specifying the ranges as you said. The same technique can be used in your code which is an implementation of stft. I would suggest using MATLAB's built-in function unless your problem requires an implementation. Hope this helps!
If needed, I can answer why the parameters for STFT are chosen as included in the code.
Related
How can I change the pitch of an imported signal logarithmically / exponentially over time?
Please note that the imported signals that will be used are not single frequencies so a simple sweep or a chirp command will not work since I will be importing vocal audio files, I just created the examples below so they would work and could be tested / show the issues I'm having.
I can change the pitch of a signal over time linearly which works great see part 1 of test code and frequency plot below. Thanks to Sheljohn for the code
%Sweep question part 1
clear all,clf reset,tic,clc
pkg load signal %load packages
%%%----create signal
start_freq=500;
end_freq=20;
fs=22050
len_of_sig=7; %in seconds
t=linspace(0,2*pi*len_of_sig,fs*len_of_sig);
orig_sig1=.8*sin(start_freq*t);
wavwrite([orig_sig1(:)] ,fs,16,strcat('/tmp/0_sig.wav')); % export file
%%%---import signal
[ya, fs, nbitsraw] = wavread('/tmp/0_sig.wav');
orig_total_samples=length(ya); %make this the same length as signal wave
t_import=linspace(0,2*pi*(orig_total_samples/fs),orig_total_samples);
%%%%----Begin linsweep
x = ya(:);
fac=(end_freq-start_freq)/length(x); %linear slope
n = numel(x); % number of timepoints
m = mean(x); % average of the signal
k = transpose(0:n-1); %
h = hilbert( x - m ); % analytic signal
env1 = abs(h); % envelope
sweep=fac*pi*k.^2/(fs); %linearly increasing offset original %alter curve here
p = angle(h) + sweep; % phase + linearly increasing offset original
y = m - imag(hilbert( env1 .* sin(p) )); % inverse-transform
wavwrite([y(:)] ,fs,16,strcat('/tmp/0_sweep.wav')); % export file
%%%----------Used for plotting
z = hilbert(y);
instfreq = fs/(2*pi)*diff(unwrap(angle(z))); %orginal
t_new=t_import/(2*pi); %converts it to seconds
plot(t_new(2:end),instfreq,'-r')
xlabel('Time (secnds)')
ylabel('Frequency (Hz)')
grid on
title('Instantaneous Frequency')
Issues with the code I have below are:
1) The frequency doesn't start or end at the correct frequency.
2) It doesn't have the correct slopes
I believe it has to do with the variables fac and sweep I'm just not sure how to calculate them correctly.
fac=log(start_freq/end_freq)/length(x); %slope
sweep=-(start_freq)*exp(fac*k); %alter curve here
-
%-----------------Sweep question part 2
clear all,clf reset,tic,clc
pkg load signal %load packages
%%%----create signal
start_freq=500;
end_freq=20;
fs=22050
len_of_sig=7; %in seconds
t=linspace(0,2*pi*len_of_sig,fs*len_of_sig);
orig_sig1=.8*sin(start_freq*t);
wavwrite([orig_sig1(:)] ,fs,16,strcat('/tmp/0_sig.wav')); % export file
%%%---import signal
[ya, fs, nbitsraw] = wavread('/tmp/0_sig.wav');
orig_total_samples=length(ya); %make this the same length as signal wave
t_import=linspace(0,2*pi*(orig_total_samples/fs),orig_total_samples);
%%%%----Begin linsweep
x = ya(:);
fac=log(start_freq/end_freq)/length(x); %slope
n = numel(x); % number of timepoints
m = mean(x); % average of the signal
k = transpose(0:n-1); %
h = hilbert( x - m ); % analytic signal
env1 = abs(h); % envelope
sweep=-(start_freq)*exp(fac*k); %alter curve here
p = angle(h) + sweep; % phase + increasing offset
y = m - imag(hilbert( env1 .* sin(p) )); % inverse-transform
wavwrite([y(:)] ,fs,16,strcat('/tmp/0_sweep.wav')); % export file
%%%----------Used for plotting
z = hilbert(y);
instfreq = fs/(2*pi)*diff(unwrap(angle(z))); %orginal
t_new=t_import/(2*pi); %converts it to seconds
plot(t_new(2:end),instfreq,'-r')
xlabel('Time (seconds)')
ylabel('Frequency (Hz)')
grid on
title('Instantaneous Frequency')
The slopes I'm trying to get are when the start frequency starts at 500hz and goes to 20hz. And when the start frequency starts at 20hz and it goes to 500hz. See plots below: Note: These frequency will change so I'm trying to get the correct formula / equation that will calculate these slopes when needed.
Ps: I'm using Octave 4.0 which is similar to Matlab.
Please note that the imported signals that will be used are not single frequencies so a simple sweep or a chirp command will not work since I will be importing vocal audio files, I just created the examples below so they would work and could be tested / show the issues I'm having.
I can get the sweep to look like the plot you are interested in by making the following changes to your code. Some of them are just cosmetic for my sake (e.g. I like my time variables to remain in units of seconds throughout).
Relevant changes:
From:
t=linspace(0,2*pi*len_of_sig,fs*len_of_sig);
orig_sig1=.8*sin(start_freq*t);
fac=log(start_freq/end_freq)/length(x); %slope
To:
t=linspace(0,len_of_sig,fs*len_of_sig);
orig_sig1=0.8*sin(start_freq*t*2*pi);
fac=log(end_freq/start_freq)/length(x);
sweep=(start_freq*2*pi/fs)*exp(fac*k); %alter curve here
Here was some other changes I made,
y = env1.*sin(p);
% and later for consistency
t_import=linspace(0,orig_total_samples/fs,orig_total_samples);
t_new=t_import; %t is seconds
The fac, in my mind, is going to be the difference from your start and end, so it would be: log(endFreq)-log(startFreq) or log(endFreq/startFreq) with the additional normalization for the length. This can be flipped with a negative sign in front.
One issue with the sweep may be happening when you use it to calculate p=angle(h)+sweep; where angle(h) is in radians.
The radians vs Hz units issue may be causing some of the difficulty.
I will keep the explanation, how my codes works, very short. I advise you to try this code yourself, so perhaps you understand it better that way. I have an audio-file and read it in my code. Now i switch from the time domain to the frequency domain by using the function FFT. But the only difference is, that i am performing an STFT on my audio signal. I do it every 30ms, until to the length of my signal. I am aware, that there are many different function in matlab, which also can perform this easily, but there are not giving me the results i need. Now, i am plotting many different frequency spectrums every 30ms. But i split up my signal in three frequency bands. They are called LOW, MEDIUM and HIGH. Basically, this means I have 3 different spectrums plotting every 30ms. The next step I do, is summing all the magnitudes from ONE frequency spectrum together, this means I have ONE VALUE per frequency spectrum, which are being squared.
Now, i have the power from every spectrum ! And all of these values are being plotted in my code. I am only plotting the power values, otherwise my code performance time would be extremely slow.
Btw, the code looks long, but there are two for loop. In the first, i read the low spectrum and when it is finished, the second starts with the medium and high spectrum. Basically they are the same. I am aware, i can probably do that with findpeaks or something similar. But how can i write/pull that of? Or what the necessary steps to do that. At the end, i included a file, Hopefully you can see that.
I want to measure the peaks and get the distance between them from the red plot.
EDIT:
Ok, i got the peaks, but not in the way i imagined. I want to show the peaks, which are above 5000-line. Sorry for not being clear at the beginning. See my plot, what i mean. I want to say my code, that only the peaks should be measured, which are above the 5000-line.
[pks, locs] = findpeaks(ValuesOfYc);
p=plot(x,ValuesOfYc,'r-' ,x(locs), pks,'ob');
This is, what I did above, in my first loop. How should i go on from there?
clear;
clc;
%% MATLAB
%% read file
%_________________________________________
[y,fs]=audioread('Undertale - Megalovania.wav');
% audioread = read wav -file
% y = contains the audio signal
% fs = 44100
% 'UnchainMyHeart' = name of the wav-file
%_________________________________________
%% PARAMETER FOR STFT
%_________________________________________
t_seg=0.03; % length of segment in ms
fftlen = 4096; %FFT-Points
% Defining size of frequency bands
f_low= 1:200; %lower frequencies
f_medium= 201:600; %medium frequencies
f_high= 601:1000; %higher frequencies
%__________________________________________
%% CODE
segl =floor(t_seg*fs);
windowshift=segl/2;
% defining the size of the window shift
window=hann(segl);
% apply hann function on segment length (30 ms)
window=window.';
% transpose vector
si=1;
% defining start index
ei=segl;
% defining end index
N=floor( length(y)/windowshift - 1);
% Calculates the number, how often the window has to shift
% until to length of the audio signal
f1=figure;
% Generating new window
f=0:1:fftlen-1;
f=f/fftlen*fs;
% defining frequency vector
Ya=zeros(1,fftlen);
ValuesOfYc = NaN(1,N);
ValuesOfYd = NaN(1,N);
ValuesOfYe = NaN(1,N);
x =(1:N)*windowshift/fs;
% defining x-axis
for m= 1:1:N
y_a = y(si:ei);
% a segment is taken out from audio signal length(30ms)
y_a= y_a.*window;
% multiplying segment with window (hanning)
Ya=fft(y_a, fftlen);
% Applying fft on segment
Yb=abs(Ya(1:end/2)).^2;
% Squaring the magnitudes from one-sided spectrum
drawnow; % Updating the graphical values
figure(f1);
% Showing the power values
%% frequency bands
y_low = Yb(f_low); % LOW frequency spectrum
Yc=sum(y_low);
% Summing all the power values from one frequency spectrum together
% so you get one power value from one spectrum
ValuesOfYc(m) = Yc;
%Output values are being saved here, which are generated from the for
%loop
% m = start variable from for loop
[pks, locs] = findpeaks(ValuesOfYc);
subplot(2,1,1)
p=plot(x,ValuesOfYc,'r-', x(locs(pks>=5000)), pks(pks>=5000),'ob');
p(1).LineWidth =0.5;
xlabel('time (Audio length)')
ylabel('Power')
grid on
si=si+windowshift;
% Updating start index
ei=ei+windowshift;
% Updating end index
end
for o= 1:1:N
y_a = y(si:ei);
% a segment is taken out from audio signal length(30ms)
y_a= y_a.*window;
% multiplying segment with window (hanning)
Ya=fft(y_a, fftlen);
% Applying fft on segment
Yb=abs(Ya(1:end/2)).^2;
% Squaring the magnitudes from one-sided spectrum
drawnow; % Updating the graphical values
figure(f1);
% Showing the power values
[![enter image description here][1]][1]
%% frequency bands
y_medium = Yb(f_medium); % MEDIUM frequency spectrum
y_high = Yb(f_high); % HIGH frequency spectrum
Yd=sum(y_medium);
Ye=sum(y_high);
% Summing all the power values from one frequency spectrum together
% so you get one power value from one spectrum
ValuesOfYd(o) = Yd;
ValuesOfYe(o) = Ye;
%Output values are being saved here, which are generated from the for
%loop
% m = start variable from for loop
subplot(2,1,2)
p=plot(x, ValuesOfYd,'g-', x, ValuesOfYe,'b-' );
p(1).LineWidth =0.5;
xlabel('time (Audio length)')
ylabel('Power')
grid on
si=si+windowshift;
% Updating start index
ei=ei+windowshift;
% Updating end index
end
I apologize in advance, if the title is confusing. Basically, i have a Audio-file, on which I perform a STFT every 50ms. My File is about 11 seconds long (10.8526s), which i have cut off from a soundtrack. Btw, i am not allowed to use the built-in function in Matlab for the STFT. I am aware it is much more easier. Anyway, after i run my code, every 50 ms a STFT is performed and the picture is being plotted.
Now i want to seperate it in 3 different plots. In the first plot i have the lower frequencies (0-300Hz), in the second plot medium frequencies(300-5kHz) and in the last plot i have high frequencies(5Khz-fs/2). fs=44100 --> Further explanations below in the code. How can i define now the areas?
%AUDIO-FILE
%______________________________________________________
[y,fs]=audioread('UnchainMyHeart.wav');
% audioread = Reads Audio file
% y = A vector, which contains the audio signal
% fs = sample rate
% 'UnchainMyHeart' = Audio file
%______________________________________________________
% Paramter for the real-time spectral-analysis
%______________________________________________________
NFA=2;
% Every second picture is being plotted
% Don't need every picture
t_seg=0.05;
%Length of the audio signal on which is a STFT performed
fftlen = 4096;
% Length of the FFT, frequency resolution
TPF= 300;
BPF= 5000;
HPF= 22050;
% Trying to define the frequencies areas
% Isn't working right now
LOW=((TPF*fftlen)/fs);
MEDIUM=((BPF*fftlen)/fs);
HIGH=((HPF*fftlen)/fs);
% Contains the number of FFT points in the frequency
%_______________________________________________________
segl =floor(t_seg*fs);
windowshift=segl/2;
window=hann(segl);
window=window.';
si=1;
% Start Index
ei=segl;
% End Index
AOS= length(y)/windowshift - 1;
f1=figure;
f=0:1:fftlen-1;
f=f/(fftlen-1)*fs;
Ya=zeros(1,fftlen);
n=0;
for m= 1:1:AOS
y_a = y(si:ei);
y_a= y_a.*window;
Ya=fft(y_a, fftlen);
n=n+1;
if n==1
Yres=abs(Ya);
else
Yres=Yres+abs(Ya);
end
if n==NFA
Yres=Yres/NFA;
n=0;
drawnow;
%Updates the graphical objects which are being plotted every 50ms
figure(f1);
plot(f(1:end/2), 20*log10(abs(Yres(1:end/2))));
ylim([-90 50]);
title('Spektrum of audio signal');
xlabel('f(Hz)');
ylabel('dB');
grid on;
end
si=si+windowshift;
% Updating Start Index
ei=ei+windowshift;
% Updating End index
end
I am not able to run you code as I do not have your audio file but I will try to explain conceptually, and use pseudo code.
Frequency brick-wall
If you just want to separate the frequencies for visual purposes you can just use brick-wall filters.
Perform an fft of the full signal. Define a frequency vector.
SigFD = fft(signal);
n = length(signal); % number of samples
fs = 44100; % sampling rate
deltaF = fs/n; % frequency resolution
F = [0:floor(n/2)-1, -(floor(n/2)):-1]*deltaF; % frequency vector
Slice the signal based on the frequency range that you want.
lowF = 0;
highF = 500;
part1Range = abs(F)>lowF&abs(F)<highF;
Fpart1 = F(part1Range);
Sig1FD = SigFD(part1Range);
Note that I am unable to test the code on your waveform so this should be considered more of pseudo code!
I'm trying to understand how the FFT in matlab works, particularly, how to define the frequency range to plot it. It happens that I have read from matlab help links and from other discussions here and I think (guess) that I'm confused about it.
In the matlab link:
http://es.mathworks.com/help/matlab/math/fast-fourier-transform-fft.html
they define such a frequency range as:
f = (0:n-1)*(fs/n)
with n and fs as:
n = 2^nextpow2(L); % Next power of 2 from length of signal x
fs = N/T; % N number of samples in x and T the total time of the recorded signal
But, on the other hand, in the previous post Understanding Matlab FFT example
(based on previous version of matlab), the resulting frequency range is defined as:
f = fs/2*linspace(0,1,NFFT/2+1);
with NFFT as the aforementioned n (Next power of 2 from length of signal x).
So, based on that, how these different vectors (equation 1 and final equation) could be the same?
If you can see, the vectors are different since the former has n points and the later has NFFT/2 points! In fact, the factor (fs/n) is different from fs/2.
So, based on that, how these different vectors (equation 1 and final equation) could be the same?
The example in the documentation from Mathworks plots the entire n-point output of the FFT. This covers the frequencies from 0 to nearly fs (exactly (n-1)/n * fs). They then make the following observation (valid for real inputs to the FFT):
The first half of the frequency range (from 0 to the Nyquist frequency fs/2) is sufficient to identify the component frequencies in the data, since the second half is just a reflection of the first half.
The other post you refer to just chooses to not show that redundant second half. It then uses half the number of points which also cover half the frequency range.
In fact, the factor (fs/n) is different from fs/2.
Perhaps the easiest way to make sense of it is to compare the output of the two expressions for some small value of n, says n=8 and setting fs=1 (since fs multiplies both expressions). On the one hand the output of the first expression [0:n-1]*(fs/n) would be:
0.000 0.125 0.250 0.500 0.625 0.750 0.875
whereas the output of fs/2*linspace(0,1,n/2+1) would be:
0.000 0.125 0.250 0.500
As you can see the set of frequencies are exactly the same up to Nyquist frequency fs/2.
The confusion is perhaps arising from the fact that the two examples which you have referenced are plotting results of the fft differently. Please refer to the code below for the references made in this explanation.
In the first example, the plot is of the power spectrum (periodogram) over the frequency range. Note, in the first plot, that the periodogram is not centered at 0, meaning that the frequency range appears to be twice that of the Nyquist sampling frequency. As mentioned in the mathworks link, it is common practice to center the periodogram at 0 to avoid this confusion (figure 2).
For the second example, taking the same parameters, the original plot is of amplitude of the fourier spectrum with a different normalization than in the first example (figure 3). Using the syntax of Matlab's full frequency ordering (as commented in the code), it is trivial to convert this seemingly different fft result to that of example 1; the identical result of the 0-centered periodogram is replicated in figure 4.
Thus, to answer your question specifically, the frequency ranges in both cases are the same, with the maximum frequency equal to the Nyquist sampling frequency as in:
f = fs/2*linspace(0,1,NFFT/2+1);
The key to understanding how the dfft works (also in Matlab) is to understand that you are simply performing a projection of your discrete data set into fourier space where what is returned by the fft() function in matlab are the coefficients of the expansion for each frequency component and the order of the coefficients is given (in Matlab as in example 2) by:
f = [f(1:end-1) -fliplr(f(1,2:end))];
See the Wikipedia page on the DFT for additional details:
https://en.wikipedia.org/wiki/Discrete_Fourier_transform
It might also be helpful for you to take the fft omitting the length as a power of 2 parameter as
y = fft(x).
In this case, you would see only a few non-zero components in y corresponding to the exact coefficients of your input signal. The mathworks page claims the following as a motivation for using or not using this length:
"Using a power of two for the transform length optimizes the FFT algorithm, though in practice there is usually little difference in execution time from using n = m."
%% First example:
% http://www.mathworks.com/help/matlab/math/fast-fourier-transform-fft.html
fs = 10; % Sample frequency (Hz)
t = 0:1/fs:10-1/fs; % 10 sec sample
x = (1.3)*sin(2*pi*15*t) ... % 15 Hz component
+ (1.7)*sin(2*pi*40*(t-2)); % 40 Hz component
% Removed the noise
m = length(x); % Window length
n = pow2(nextpow2(m)); % Transform length
y = fft(x,n); % DFT
f = (0:n-1)*(fs/n); % Frequency range
power = y.*conj(y)/n; % Power of the DFT
subplot(2,2,1)
plot(f,power,'-o')
xlabel('Frequency (Hz)')
ylabel('Power')
title('{\bf Periodogram}')
y0 = fftshift(y); % Rearrange y values
f0 = (-n/2:n/2-1)*(fs/n); % 0-centered frequency range
power0 = y0.*conj(y0)/n; % 0-centered power
subplot(2,2,2)
plot(f0,power0,'-o')
% plot(f0,sqrt_power0,'-o')
xlabel('Frequency (Hz)')
ylabel('Power')
title('{\bf 0-Centered Periodogram} Ex. 1')
%% Second example:
% http://stackoverflow.com/questions/10758315/understanding-matlab-fft-example
% Let's redefine the parameters for consistency between the two examples
Fs = fs; % Sampling frequency
% T = 1/Fs; % Sample time (not required)
L = m; % Length of signal
% t = (0:L-1)*T; % Time vector (as above)
% % Sum of a 3 Hz sinusoid and a 2 Hz sinusoid
% x = 0.7*sin(2*pi*3*t) + sin(2*pi*2*t); %(as above)
NFFT = 2^nextpow2(L); % Next power of 2 from length of y
% NFFT == n (from above)
Y = fft(x,NFFT)/L;
f = Fs/2*linspace(0,1,NFFT/2+1);
% Plot single-sided amplitude spectrum.
subplot(2,2,3)
plot(f,2*abs(Y(1:NFFT/2+1)),'-o')
title('Single-Sided Amplitude Spectrum of y(t)')
xlabel('Frequency (Hz)')
ylabel('|Y(f)|')
% Get the 0-Centered Periodogram using the parameters of the second example
f = [f(1:end-1) -fliplr(f(1,2:end))]; % This is the frequency ordering used
% by the full fft in Matlab
power = (Y*L).*conj(Y*L)/NFFT;
% Rearrange for nicer plot
ToPlot = [f; power]; [~,ind] = sort(f);
ToPlot = ToPlot(:,ind);
subplot(2,2,4)
plot(ToPlot(1,:),ToPlot(2,:),'-o')
xlabel('Frequency (Hz)')
ylabel('Power')
title('{\bf 0-Centered Periodogram} Ex. 2')
I have a question while computing the spectrum of a time series in Matlab. I have read the documentations concerning 'fft' function. However I have seen two ways of implementation and both wgive me different results. I would appreciate to have some answer about this difference:
1st Method:
nPoints=length(timeSeries);
Time specifications:
Fs = 1; % samples per second
Fs = 50;
freq = 0:nPoints-1; %Numerators of frequency series
freq = freq.*Fs./nPoints;
% Fourier Transform:
X = fft(timeSeries)/nPoints; % normalize the data
% find find nuquist frequency
cutOff = ceil(nPoints./2);
% take only the first half of the spectrum
X = abs(X(1:cutOff));
% Frequency specifications:
freq = freq(1:cutOff);
%Plot spectrum
semilogy(handles.plotLoadSeries,freq,X);
2nd Method:
NFFT = 2^nextpow2(nPoints); % Next power of 2 from length of y
Y = fft(timeSeries,NFFT)/nPoints;
f = 1/2*linspace(0,1,NFFT/2+1);
% % Plot single-sided amplitude spectrum.
% plot(handles.plotLoadSeries, f,2*abs(Y(1:NFFT/2+1)))
semilogy(handles.plotLoadSeries,f,2*abs(Y(1:NFFT/2+1)));
I thought that it is not necessary to use 'nextpow' function in 'fft' function in Matlab. Finally, which is the good one?
THanks
The short answer: you need windowing for spectrum analysis.
Now for the long answer... In the second approach, you are using an optimised FFT algorithm useful when the length of the input vector is a power of two. Let's assume that your original signal has 401 samples (as in my example below) from an infinitely long signal; nextpow2() will give you NFFT=512 samples. When you feed the shorter, 401-sample signal into the fft() function, it is implicitly zero-padded to match the requested length of 512 (NFFT). But (here comes the tricky part): zero-padding your signal is equivalent to multiplying an infinitely long signal by a rectangular function, an operation that in the frequency domain translates to a convolution with a sinc function. This would be the reason behind the increased noise floor at the bottom of your semilogarithmic plot.
A way to avoid this noise increase is to create manually the 512-sample signal you want to feed into fft(), using a smoother window function instead of the default rectangular one. Windowing means just multiplying your signal by a tapered, symmetric one. There are tons of literature on choosing a good windowing function, but a typically accurate one with low sidelobes (low noise increase) is the Hamming function, implemented in MATLAB as hamming().
Here is a figure illustrating the issue (in the frequency domain and time domain):
...and the code to generate this figure:
clear
% Create signal
fs = 40; % sampling freq.
Ts = 1/fs; % sampling period
t = 0:Ts:10; % time vector
s = sin(2*pi*3*t); % original signal
N = length(s);
% FFT (length not power of 2)
S = abs(fft(s)/N);
freq = fs*(0:N-1)/N;
% FFT (length power of 2)
N2 = 2^nextpow2(N);
S2 = abs(fft(s, N2)/N2);
freq2 = fs*(0:N2-1)/N2;
t2 = (0:N2-1)*Ts; % longer time vector
s2 = [s,zeros(1,N2-N)]; % signal that was implicitly created for this FFT
% FFT (windowing before FFT)
s3 = [s.*hamming(N).',zeros(1,N2-N)];
S3 = abs(fft(s3, N2)/N2);
% Frequency-domain plot
figure(1)
subplot(211)
cla
semilogy(freq,S);
hold on
semilogy(freq2,S2,'r');
semilogy(freq2,S3,'g');
xlabel('Frequency [Hz]')
ylabel('FFT')
grid on
legend( 'FFT[401]', 'FFT[512]', 'FFT[512] with windowing' )
% Time-domain plot
subplot(212)
cla
plot(s)
hold on
plot(s3,'g')
xlabel('Index')
ylabel('Amplitude')
grid on
legend( 'Original samples', 'Windowed samples' )