I have generated three identical waves with a phase shift in each. For example:
t = 1:10800; % generate time vector
fs = 1; % sampling frequency (seconds)
A
For two sinusoidal signal the phase of the complex correlation coefficient gives you what you want. I can only give you an python example (using scipy) as I don't have a matlab to test it.
x1 = sin( 0.1*arange(1024) )
x2 = sin( 0.1*arange(1024) + 0.456)
x1h = hilbert(x1)
x2h = hilbert(x2)
c = inner( x1h, conj(x2h) ) / sqrt( inner(x1h,conj(x1h)) * inner(x2h,conj(x2h)) )
phase_diff = angle(c)
There is a function corrcoeff in matlab, that should work, too (The python one discard the imaginary part). I.e. c = corrcoeff(x1h,x2h) should work in matlab.
The Matlab code to find relative phase using cross-correlation:
fr = 20; % input signal freq
timeStep = 1e-4;
t = 0:timeStep:50; % time vector
y1 = sin(2*pi*t); % reference signal
ph = 0.5; % phase difference to be detected in radians
y2 = 0.9 * sin(2*pi*t + ph); % signal, the phase of which, is to be measured relative to the reference signal
[c,lag]=xcorr(y1,y2); % calc. cross-corel-n
[maxC,I]=max(c); % find max
PH = (lag(I) * timeStep) * 2 * pi; % calculated phase in radians
>> PH
PH =
0.4995
If you know the frequency and just want to find the phase, rather than use a full FFT, you might want to consider the Goertzel algorithm, which is a more efficient way to calculate the DFT for a single frequency (an FFT will calculate it for all frequencies).
For a good implementation, see: https://www.mathworks.com/matlabcentral/fileexchange/35103-generalized-goertzel-algorithm and https://asp-eurasipjournals.springeropen.com/track/pdf/10.1186/1687-6180-2012-56.pdf
There are several ways that you can measure the phase shift between signals. Between your response, the comments below your response, and the other answers, you've gotten most of the options. The specific choice of technique is usually based on issues such as:
Depending on your answer to these questions, you could consider the following techniques:
Cross-Correlation: Use the a command like [c,lag]=xcorr(y1,y2);
to get the cross-correlation between the two signals. This works on the original time-domain signals. You look for the index where c
is maximum ([maxC,I]=max(c);
) and then you get your lag value in units of samples lag = lag(I);
. This approach gives you the average phase lag for the entire recording. It requires that your signal of interest in the recording be stronger than anything else in your recording...in other words, it is sensitive to noise and other interference.
Frequency Domain: Here you convert your signals into the frequency domain (using fft
or cpsd
or whatever). Then, you'd find the bin that corresponds to the frequency that you care about and get the angle between the two signals. So, for example, if bin #18 corresponds to your signal's frequency, you'd get the phase lag in radians via phase_rad = angle(fft_y1(18)/fft_y2(18));
. If your signals have a constant frequency, this is an excellent approach because it naturally rejects all noise and interference at other frequencies. You can have really strong interference at one frequency, but you can still cleanly get your signal at another frequency. This technique is not the best for signals that change frequency during the fft analysis window.
Hilbert Transform: A third technique, often overlooked, is to convert your time-domain signal into an analytic signal via the Hilbert transform: y1_h = hilbert(y1);
. Once you do this, your signal is a vector of complex numbers. A vector holding a simple sine wave in the time domain will now be a vector of complex numbers whose magnitude is constant and whose phase is changing in sync with your original sine wave. This technique allows you to get the instantaneous phase lag between two signals...it's powerful: phase_rad = angle(y1_h ./ y2_h);
or phase_rad = wrap(angle(y1_h) - angle(y2_h));
. The major limitation to this approach is that your signal needs to be mono-component, meaning that your signal of interest must dominate your recording. Therefore, you may have to filter out any substantial interference that might exist.
Here is the little modification of your code: phi = 10
is actually in degree, then in sine function, phase information is mostly expressed in radian,so you need to change deg2rad(phi)
as following:
t = 1:10800; % generate time vector
fs = 1; % sampling frequency (seconds)
A = 2; % amplitude
P = 1000; % period (seconds), the time it takes for the signal to repeat itself
f1 = 1/P; % number of cycles per second (i.e. how often the signal repeats itself every second).
y1 = A*sin(2*pi*f1*t); % signal 1
phi = deg2rad(10); % phase shift
y2 = A*sin(2*pi*f1*t + phi); % signal 2
phi = deg2rad(15); % phase shift
y3 = A*sin(2*pi*f1*t + phi); % signal 3
YY = [y1',y2',y3'];
plot(t,YY)
then using frequency domain method as mentioned chipaudette
fft_y1 = fft(y1);
fft_y2 = fft(y2);
phase_rad = angle(fft_y1(1:end/2)/fft_y2(1:end/2));
phase_deg = rad2deg(angle(fft_y1(1:end/2)/fft_y2(1:end/2)));
now this will give you a phase shift estimate with error = +-0.2145
With the correct signals:
t = 1:10800; % generate time vector
fs = 1; % sampling frequency (seconds)
A = 2; % amplitude
P = 1000; % period (seconds), the time it takes for the signal to repeat itself
f1 = 1/P; % number of cycles per second (i.e. how often the signal repeats itself every second).
y1 = A*sin(2*pi*f1*t); % signal 1
phi = 10*pi/180; % phase shift in radians
y2 = A*sin(2*pi*f1*t + phi); % signal 2
phi = 15*pi/180; % phase shift in radians
y3 = A*sin(2*pi*f1*t + phi); % signal 3
The following should work:
>> acos(dot(y1,y2)/(norm(y1)*norm(y2)))
>> ans*180/pi
ans = 9.9332
>> acos(dot(y1,y3)/(norm(y1)*norm(y3)))
ans = 0.25980
>> ans*180/pi
ans = 14.885
Whether or not that's good enough for your "real" signals, only you can tell.