Why are the results of my spectrogram sinusoidal?

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Harvey 2016 年 4 月 21 日
コメント済み: Harvey 2016 年 4 月 22 日
I am trying to determine the amplitude envelope of specific frequencies over time, from a sample of an instrument (a trumpet). I use the spectrogram function to find the amplitude of each frequency bin over time. Knowing that a harmonic lies within a certain bin, I can plot that particular bin against time to produce the following figure:
Although I can make out the general envelope, as marked in red, this method is much less accurate than what I was expecting.
I can't understand why the results are sinusoidal. Is this a result of aliasing, or something else entirely?
If we compare this plot, to a plot of all frequencies, we notice that there is no sinusoidal fluctuation in the power of the particular harmonic (The harmonic plotted in the first figure has been boxed):
So where is difference in the values used to plot the consistent power we see here, and the values that produce the sinusoidal plot?
For reference I have attached the audio file I am using. Bellow are the functions I used to get these results. Row 13 refers to the bin containing the specific harmonic;
Any help would be extremely gratefully received!


J. Webster
J. Webster 2016 年 4 月 21 日
編集済み: J. Webster 2016 年 4 月 21 日
You don't want just the real part of the fft, you want the magnitude of it.
s_real = real(s)
s_mag = abs(s)
and use that in your plot.
  3 件のコメント
Harvey 2016 年 4 月 22 日
Thank you!


その他の回答 (1 件)

Star Strider
Star Strider 2016 年 4 月 21 日
I am not certain what you want to do. Just filtering the fundamental frequency gives it as a function of time, and the envelope is relatively easy to calculate:
t = linspace(0, length(y), length(y))/fs; % Time Vector
Fn = fs/2; % Nyquist Frequency
Wp = [0.2 0.4]*1E+3/Fn; % Filter Passband
Ws = [0.5 2.0].*Wp; % Filter Stopband
Rp = 10; % Passband Ripple
Rs = 45; % Stopband Ripple
[n,Wn] = buttord(Wp, Ws, Rp, Rs); % Filter Order
[b,a] = butter(n,Wn); % Transfer Function Coefficient Vectors
[sos,g] = tf2sos(b,a); % Second-Order-Section (For Stability)
freqz(sos, 4028, fs); % Filter Bode Plot
yf = filtfilt(sos,g, y); % Filter Signal (Phase-Neutral)
env = abs(hilbert(yf)); % Calculate Envelope
plot(t, y)
title('Unfiltered Signal')
axis([0 0.2 ylim]) % Axis Resolution (Comment-Out To See Entire Signal)
plot(t, yf)
hold on
plot(t, env, '-r', t, -env, '-r')
hold off
title('Bandpass-Filtered Signal')
axis([0 0.2 ylim]) % Axis Resolution (Comment-Out To See Entire Signal)
xlabel('Time (s)')
It’s a signal with a fundamental frequency and (apparently 13) harmonics, so you would have to filter each one, and calculate the envelope for each one. They seem to be evenly-spaced, so you could determine the spacing and then design a bank of filters (see How to shape the spectrum of an audio waveform? to filter each frequency.
  3 件のコメント
Harvey 2016 年 4 月 21 日
I will try the filter bank method. Strange that the spectrogram appears steady in figure 2, but could be the windowing.



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