Audio processing using Matlab. Elena Grassi. Sampling. Read values from a continuous signal Equally spaced time interval (sampling frequency). A/D (analog in/digital out). AI = analoginput('winsound'); addchannel(AI,1); set(AI,'SampleRate',44100) set(AI,'SamplesPerTrigger',4*44100)
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AI = analoginput('winsound');
data = getdata(AI);
delete(AI), clear AI
Note: dB= 20*log10 ()
specgram(y, 256, fs)
AO = analogoutput('winsound');
delete(AO), clear AO
Nyquist frequency= 2*BW
Modify frequency content of signals.
Classification according to their pass/stop bands:
Specify corner frequency(ies), normalized wrt ½ sampling frequency. Example: 2000/(fs/2) for 2000 Hz.
Classification according to their roll-off, flatness, phase:
b= numerator polynomial in z
a= denominator polynomial in z
title('Filter frequency response')