Prelab Questions

Experiment # 7

1. At 1kHz cosine wave is sampled by a series of impulses spaced T seconds apart.

s(nT ) = cos(2 1000 t ) (t nT )
n =

where T=1/fs (sampling frequency). If fs = 5,000 Hz find the positive frequency spectrum of the sampled signal (amplitude only). The spectrum of s(nT) is the spectrum of cos(2*1000*t) convoluted with the spectrum of the impulse train. the magnitude spectrum above is also the magnitude spectrum of s(nT) An nth order low pass filter is used to reconstruct the sampled signal in question #1 and has an amplitude response of:
H( f ) = 1 f 1+ f3
2n

where n = the order of the filter and f3 = 3dB cutoff =1 kHz. If a 4th order filter is used, what would be the lowest sampling frequency that would cause less than 1% distortion due to aliasing? (Hint: Assume that the single aliasing impulse that is closest to the 1 kHz signal contributes the most to the aliasing error.) If it is assume that the single aliasing impulse that is closest to the 1 kHz signal contributes the most to the aliasing error, then this problem is equivalent to find the frequency such that the |H(f)| is 1% of the value at 1 kHz. And from this frequency, the lowest sampling frequency that would cause less than 1% distortion due to aliasing is compute by adding 1000Hz to it. So, first we have to find the frequency such that the |H(f)| is 1% of the value at 1 kHz |H(1000)| = 2 -0.5 = 0.7071 Want to find f such that |H(f)| = 0.7071*0.01 = 0.007071
0.007071 = 1
8

f 1 + 1000 solve the above equation for f and get f = 3448.5 Hz

Therefore, the lowest sampling frequency must be 3448.5 + 1000 = 4448.5 Hz

2. If an 8th order filter is used, what would be the lowest sampling frequency that would cause less than 1% distortion? Do the same for problem #2 but this time change n to 8, then the equation that need to be solved is
0.007071 = 1
16

f 1 + 1000 solve the above equation for f and get f =1857.0 Hz

Therefore, the lowest sampling frequency must be 1857.0 + 1000 = 2857.0 Hz 3. A 4-bit A-D converter has a sampling rate of 16 kHz. This system samples a 1 kHz triangle wave that has an amplitude that exactly spans the converters quantization range. Sketch the unfiltered D-A output, if this signal is the A-D input. Create one period of triangle signal in MATLAB with command: t=[0:(16*10^3)^-1:10^-3]; s=[t(1:9) t(8:-1:1)]/.5e-3*2-1; The quantizer was configured with these parameters, Quantization partition: p=[-1+.125:.125:.875] Quantization codebook: c=[p-.125/2 .875+.125/2]