《Audio processing using Matlab》.pptVIP

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) set(AI,TriggerType,Manual) start(AI) trigger(AI) data = getdata(AI); delete(AI), clear AI Spectrogram Short time Fourier transform Tradeoff frequency/time resolution. Note: dB= 20*log10 () D/A (digital in/analog out) AO = analogoutput(winsound); addchannel(AO,1); set(AO,SampleRate,22050) set(AO,TriggerType,Manual) putdata(AO,x) start(AO) trigger(AO) waittilstop(AO,5) delete(AO), clear AO Aliasing When sampling is too slow for a signal’s BW, high frequency content cannot be observed and it leaks into lower frequencies, thus distorting the signal. Minimum sampling required to capture the signal accurately: Nyquist frequency= 2*BW If not possible, apply antialiasing filter. Filters Modify frequency content of signals. Classification according to their pass/stop bands: Lowpass (smoothing filter) Highpass Bandpass Stopband Specify corner frequency(ies), normalized wrt ? sampling frequency. Example: 2000/(fs/2) for 2000 Hz. Example Filter Types Classification according to their roll-off, flatness, phase: Bessel: linear phase, preserves wave shape. Butterworth: flat and monotonic, sacrifice roll-off steepness. Chebyshev I: equiripple in passband and monotonic in stopband. Chebyshev II: monotonic in passband and equiripple in stopband, roll off slower than type I. Example Filter frequency response h= impz(b,a,N); H=(abs(fft(h))); fscale= fsi/N*(1:N/2); plot(fscale,H(1:N/2),r) xlabel(f [Hz]) title(Filter frequency response) Filter order Related to complexity (hardware or numerical) and how many samples of data are used. Higher order - Steepness Trade off with complexity/numerical stability * specgram(y, 256, fs) title(Spectrogram [dB]) [b,a]= butter(6,2000*2/fsi,low); b= numerator polynomial