dsp课程设计（DSP curriculum design）.docVIP

dsp课程设计（DSP curriculum design） Design of%FIR low pass filter Fp=1000; fs=1200; Determination of index% As=120; Fs=22050; Wp=2*pi*fp/Fs% for the normalized frequency; Ws=2*pi*fs/Fs; Wo=ws-wp% for transition zone; Belta=0.1102* (As-0.87);% for Keyser window function parameters N=ceil ((As-8) /2.285/wo);% by the decision of the N value of transition zone Wc= (wp+ws) /2; B=fir1 (N, WC, Kaiser (N+1, belta)); [Hw, w]=freqz (B, 1); Figure (1) Plot (w*Fs/2/pi, 20*log10 (ABS (Hw)));% frequency curve drawing Grid; Title (FIR low pass filter ); Ylabel (amplitude /db); Xlabel (/Hz frequency); Axis ([08000, -200,10]); X=wavread (D:\0072.wav);% sampling of the original speech using the function wavread Sound (x, 22050);% of the original speech signal playback Y=filter (B, 1, x);% speech signal filter Sound (y, 22050);% play filtered signal Figure (2) Subplot (2,1,1) Plot (x (: 1)); impulse response output% of the original speech signal Title (input waveform); Xlabel (f/Hz); Ylabel (range); Subplot (2,1,2) Plot (Y (: 1));% of the original speech signal output by the impulse response filter Title (output waveform); Xlabel (f/Hz); Ylabel (range); Subplot (2,1,2) Figure (3) X=fft (x, 1024);% Fu Liye transform to the filtered signal and 1024 point sampling Y=fft (y, 1024); Fu Liye transform of the original speech% and 1024 point sampling F=Fs*[0:1023]/1024; Subplot (2,1,1) Plot (F, ABS (X));% output speech signal spectrum the original map Title (FFT of the original signal waveform); Ylabel (range); Xlabel (/Hz frequency); Axis ([08000,0,4]); Subplot (2,1,2) Plot (F, ABS (Y)); Title (original speech signal by the filter after the FFT); Ylabel (range); Xlabel (/Hz frequency); Axis ([08000,0,4]); % band-pass filter design Fb1=1200; fc1=1000; Determination of index% Fb2=3000; fc2=3200; As=120; Fs=22050; Wp1=2*pi*fc1/Fs% for the normalized frequency; Ws1=2*pi*fb1/Fs; Wp2=2*pi*fc2/Fs; Ws2=2*pi*fb2/Fs; Wo1=wp1-ws1% for transition zone; Wo2=wp2-ws2; Wo=max (wo1, WO2); Beta=0.1102* (As-0.87);%