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DSP Sample Sheet

Digital Signal Processing

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19 views4 pages

DSP Sample Sheet

Digital Signal Processing

Uploaded by

a1185569145
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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Download as PDF, TXT or read online on Scribd
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D i g i t a l S i g n a l P ro c e s s i n g 英 文 试 卷 样 卷

该试卷从吴镇杨老师编写《数字信号处理》参考
书后翻译,不代表真实考试难度,仅做题型参考

课程名称 数字信号处理 考试学期 Sample 得分

适用专业 信息工程 考试形式 半开卷 考试时间长度 120 分钟

开卷、半开卷可带的资料:
(4 页单面或 2 页双面 A4 纸)

Section A (10%): True or False (1 mark per question,10 marks in total)


1. For FIR filters designed with Kaiser window, the transition band gets narrower
with reducing β. ( )
自 2. The poles of all-pass system must be within the unit circle, and its zeros are
觉 conjugate reciprocals of the poles. ( )
遵 3. If we have 𝐻1 (𝑧) = 𝐻2 (𝑧) , we must have ℎ1 [𝑛] = ℎ2 [𝑛] , where ℎ1 [𝑛] and

ℎ2 [𝑛] are the inverse z-transform of 𝐻1 (𝑧) and 𝐻2 (𝑧), respectively. ( )

4. IIR System must have recursive structure, i.e., feedback from the output. ( )

5. Direct form structure of digital IIR filters is less sensitive to quantization effect

of coefficients, as compared with the cascade and parallel structures. ( )
线


6. The sampling points of Chirp-z transform do not have to locate on the unit
circle. ( )
如 7. With sampling frequency 𝑓𝑠 = 5000 Hz, 2000-point DFT samples the
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考 spectrum with frequency interval 2.5 Hz. ( )



8. A N -point sequence can be uniquely determined by N samples from its

frequency spectrum. ( )

9. Equiripple approximation (Parks-McClellan algorithm) is an optimal FIR filter
此 design strategy that minimize the maximum error. ( )
答 10. The overflow oscillation is caused by round off noise. ( )

无 Section B (10%): Multiple Choice(1 mark per question,10 marks in total)


效 1. To evaluate the linear convolution of two N-point sequences using DFT algorithms,
what is the minimum length for the DFT computations?
学号

a. N b. 2N-1 c. 2N+1

共 4 页 第 1 页
2. Which of the following IIR filter has ripple for both passband and stopband?
a. Elliptic b. Chebyshev c. Butterworth
3. Which of the following settings on ℎ[𝑛], 0 ≤ 𝑛 ≤ 𝑁 − 1 , can be used for
designing a linear phase lowpass FIR filter?
a. h[n] is even symmetric, N is even b. h[n] is odd symmetric, N is odd
c. h[n] is odd symmetric, N is even
4. Consider a sequence x[n] with sampling frequency 𝑓𝑠 and its DFT represented
by X[k], 0 ≤ k ≤ N − 1. The value X[2] represents the frequency sample at ( ).
a. 𝑓𝑠 /𝑁 b. 2𝑓𝑠 /𝑁 c. 𝑓𝑠 /2𝑁
5. How is sequence x[2n] related with sequence x[n]?
a. x[2n] is obtained from x[n] by dividing its amplitude by 2
b. x[2n] is obtained from x[n] by multiplying its amplitude with 2
c. x[2n] is obtained from x[n] by extracting the sample points with even indices.

6. A system whose output depends only on the past and current inputs, but not the
future inputs, is
a. stable system b. causal system c. time-invariant system

(𝑛−𝑝)2
7. Taking 16 point FFT of the Gaussian sequence 𝑥[𝑛] = exp[− ]. Which of
𝑞

the following parameters will lead to the most significant low frequency component?
a. 𝑝 = 8, 𝑞 = 2 b. 𝑝 = 8, 𝑞 = 8 c. 𝑝 = 14, 𝑞 = 8

8. Consider a digital filter, taking the DTFT of its impulse response gives:
a. its power spectrum b. its frequency response c. its amplitude spectrum.

9 . The quantization noise of the Nth order FIR filter is____________.


a. proportional to N, b. proportional to N2, c. proportional to N3.

10. In general, the ________ of the decimation-in-time FFT algorithm is in bit-reversal


order.

a. input b. output c. input and output

共 4 页 第 2 页
Section C (32%): Calculations(8 marks per question,32 marks in total)
1. Calculate the output of the following MATLAB codes:

2. If 𝐻(𝑧) is a low pass filter, which type of filter does 𝐻(−𝑧) belong to? Let 𝜔𝑝
and 𝜔𝑠 denote the passband and stopband edge frequency, respectively. Find the
passband and stopband edge frequency of 𝐻(−𝑧) and the relationship between
the impulse response of these two filters.

3. Consider the analog signal


xa(t)=3cos(2000πt)+5sin(6000πt)+10cos(12000πt)
(a) What is the Nyquist rate for this signal?
(b) If we sample this signal using a sampling rate fs=5000Hz. What is the discrete-time
signal obtained after sampling?

4. Design a 3rd order digital Butterworth filter using bilinear transformation method,
with sampling frequency 9𝑘𝐻𝑧 and passband cutoff frequency 3𝑘𝐻𝑧.
Note: the transfer function of the 3rd order digital Butterworth filter prototype is
1
Ha  s 
1  2  s / c   2  s / c    s / c 
2 3

Section D (48%): Problems(12 marks per question,48 marks in total)


1. Consider a finite-length complex sequence 𝑓[𝑛] of length 𝑁 , which is
constructed by a real sequence 𝑥[𝑛] of length 2𝑁, i.e., 𝑓[𝑛] = 𝑥[2𝑛] + 𝑗𝑥[2𝑛 +
1]. Denote the DFT of 𝑓[𝑛] and 𝑥[𝑛] by 𝐹[𝑘] and 𝑋[𝑘], respectively. Find the
expression of 𝑋[𝑘] in terms of 𝐹[𝑘].

共 4 页 第 3 页
2. Find the best cascade structure of the filter with system function 𝐻(𝑧) =
1
, such that the round-off noise at the output is minimized, and
(1−0.6𝑧 −1 )(1−0.9𝑧 −1 )

calculate the round-off noise variance at the output.


3. Use frequency sampling method to design a linear-phase highpass FIR digital filter,
with ideal frequency response
0, 0    0.8
H d  e j   
1, 0.8    
Take N=16 sample points and give the sample values 𝐻[𝑘] and the impulse response
ℎ[𝑛].

4. x[n] is a 8-point real sequence, X  k   X  k  e j X k 


 DFT  x  n  and the first 5

points in X[k] are 4, 2  3 j, 1  2 j, 4  j, 2 ;y[n] is obtained from x[n] by


circular shift of 4 positions; Y  k   Y  k  e  DFT  y  n .
jY  k 

7 7

 x[n] ,  x[n]
2
a). Find x[0], 。
n0 n0

(2) Is X [k ]  Y [k ] true? How is  x  k  related with  y  k  ?

共 4 页 第 4 页

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