pitch: 3 sinusoidal partials of 500, 1000 and 1500 Hz duration: 100 ms (including 5 ms rise and fall times) amplitude: 80 dB
Sound 2:
like Sound 1, but with a cutting out of 10 ms (5 ms fall / rise included) from the middle of the sound
This is my third try
%%configure signal settings duration = 0.1; % duration in seconds amplitude = 0.8; % amplitude f1 = 500; % frequency in Hertz f2 = 1000; f3 = 1500; phi = 2*pi*0.5; % phase offset, e.g.: 2*pi*0.25 = 1/4 cycle %%configure output settings fs = 44100; % sampling rate T = 1/fs; % sampling period t = 0:T:duration; % time vector %%create the signal omega1 = 2*pi*f1; % angular frequency in radians omega2 = 2*pi*f2; omega3 = 2*pi*f3; partial1 = cos(omega1*t + phi)*amplitude; % sinusoidal partial 1 partial2 = cos(omega2*t + phi)*amplitude; % sinusoidal partial 2 partial3 = cos(omega3*t + phi)*amplitude; % sinusoidal partial 3 signal = (partial1 + partial2 + partial3)/3; %%plot the signal plot(t, signal); xlabel('Time (seconds)'); ylabel('Amplitude'); title('Complex Signal'); %%play the signal sound(signal, fs); %%save signal as stereo wave file stereo_signal = [x1; x1]'; wavwrite(stereo_signal, fs, 'test.wav');
w = ((1+sin(pi*(t-b)/2/e))/2.*(t>b-e)-1 ) .*(t<=b+e)+ 1;
This equals 0 for t<b, then increases and equals 1 for t>e. Also note that Matlab has a large number of windows available.
And note that the way you're implementing the amplitudes is very different form the decibels you specify!
soundsc(
scales the values of the audio signal y
)y
to fit in the range from –1.0 to 1.0, and then sends the data to the speaker at the default sample rate of 8192 hertz. By first scaling the data, soundsc
plays the audio as loudly as possible without clipping. The mean of the dynamic range of the data is set to zero.
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