Examination Paper For TTT4120 Digital Signal Processing: Department of Electronic Systems
Examination Paper For TTT4120 Digital Signal Processing: Department of Electronic Systems
Examination Paper For TTT4120 Digital Signal Processing: Department of Electronic Systems
Language: English.
Number of pages (front page excluded): 6
Number of pages enclosed: 2
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Problem 1 (9+5+6+3=23): Basics of filter theory and design
1a) Consider a discrete-time system with input-output relation 𝑦[𝑛] = ℋ{𝑥[𝑛]}, where operator
ℋ{∙} describes the system. Below you find four different discrete-time systems
i. 𝑦[𝑛] = (𝑛 + 1)𝑥[𝑛]
ii. 𝑦[𝑛] = 𝑥[𝑛] + 0.2𝑦[𝑛 − 1]
iii. 𝑦[𝑛] = √𝑥[𝑛 + 1], with 𝑥[𝑛] ≥ 0 ∀𝑛
iv. 𝑦[𝑛] = 0.3𝑥[𝑛] + 2
Which properties need to be fulfilled in order to fully describe a system using its
impulse response ℎ[𝑛]? List which systems above, i.-iv., have these properties? (3p)
Define the system property causality in terms of ℎ[𝑛]. List which systems above, i.-iv.,
are causal. (3p)
Define the system property stability in terms of ℎ[𝑛]. List which systems above, i.-iv.,
are stable. (3p)
1 + 𝑏𝑧 −1
𝐻(𝑧) = , 𝑎, 𝑏 ≥ 0
1 − 𝑎𝑧 −1
Give the possible values of the coefficients {𝑎, 𝑏} and the corresponding ROCs when the
system 1) is stable, 2) is causal, and 3) has minimum phase.
1d) Derive the expression for the impulse response ℎ[𝑛] = 𝒵 −1 {𝐻(𝑧)} for 𝐻(𝑧) given in 1c).
1 (4)
Problem 2 (6+6+6+2=20): Filter structures and implementations
1 1 −1 1 −1
where 𝐻1 (𝑧) = 1 − 2 𝑧 −1 , 𝐻2 (𝑧) = (1 − 4 𝑧 −1 ) , and 𝐻3 (𝑧) = (1 + 4 𝑧 −1 ) .
The system is to be implemented using fixed-point representation with 𝐵 + 1 bits (one bit is used
for the sign) and dynamic range [−1,1). Rounding is performed after each multiplication and the
rounding error 𝑒[𝑛] can be modeled as white noise with variance 𝜎𝑒2 . Consequently, each
multiplier in the fixed-point implementation is modeled as
𝑄(𝑎𝑦[𝑛 − 𝑘]) = 𝑎𝑦[𝑛 − 𝑘] + 𝑒[𝑛]
which is equivalent to adding noise sources after multipliers in the infinite-precision realization.
Rounding noise sources combine into an equivalent noise signal 𝑧[𝑛] at the filter output with
variance 𝜎𝑧2 (see hint below).
2a) Draw the cascade structure, of first-order sections, 𝐻(𝑧) = 𝐻1 (𝑧)𝐻2 (𝑧)𝐻3 (𝑧), with noise
sources due to rounding included. Determine the variance of the round-off noise at the filter
output.
2a) Draw the cascade-structure, 𝐻(𝑧) = 𝐻1 (𝑧)𝐻2′ (𝑧), with noise sources due to rounding
included, where filter 𝐻2′ (𝑧) = 𝐻2 (𝑧)𝐻3 (𝑧) is implemented as a second-order section with
real coefficients. Determine the variance of the round-off noise at the filter output.
2c) Draw the equivalent parallel-structure, 𝐻(𝑧) = 𝐻4 (𝑧) + 𝐻5 (𝑧), with noise sources due to
rounding included. Determine 𝐻4 (𝑧) and 𝐻5 (𝑧), and the variance of the round-off noise at
the filter output.
2
[Hint:] Assuming noise source 𝑒𝑖 [𝑛] with variance 𝜎𝑒𝑖 acts as input to (sub-)filter ℎ𝑖 [𝑛] that
terminates at the output, the variance of the noise signal 𝑧𝑖 [𝑛], due to 𝑒𝑖 [𝑛], is given by
1
2
𝜎𝑧𝑖 2
= 𝜎𝑒𝑖 2
𝑟ℎ𝑖ℎ𝑖 [0] = 𝜎𝑒𝑖 ∑ ℎ𝑖2 [𝑘] = 𝜎𝑒𝑖
2
∫ |𝐻𝑖 (𝑓)|2 𝑑𝑓
𝑘 0
𝑒𝑖 [𝑛] 𝑧𝑖 [𝑛]
𝐻𝑖 (𝑧)
2
𝜎𝑒𝑖 = 𝐸{𝑒𝑖2 [𝑛]} 2
𝜎𝑧𝑖 = 𝐸{𝑧𝑖2 [𝑛]}
2 (4)
Problem 3 (4+8+8=20): Parametric modeling and Wiener filtering
𝑊[𝑛] 𝑆[𝑛]
𝐻(𝑧)
A wide-sense stationary (WSS) stochastic process 𝑆[𝑛] is generated by filtering a white noise
process 𝑊[𝑛], with autocorrelation sequence 𝛾𝑊𝑊 [𝑙] = 0.36𝛿[𝑙], through a causal and stable
filter with the following system function
1
𝐻(𝑧) =
1 − 0.8𝑧 −1
3a) Design the whitening filter 𝐻𝐼 (𝑧) so that the cascaded structure 𝐻(𝑧)𝐻𝐼 (𝑧) produces a white
output sequence, when 𝑊[𝑛] is the input sequence. Provide the resulting ROC for 𝐻𝐼 (𝑧).
3b) Compute the output spectrum Γ𝑆𝑆 (𝑓) of signal 𝑆[𝑛] and show/verify that the corresponding
autocorrelation sequence is given by 𝛾𝑆𝑆 [𝑙] = 0.8|𝑙| .
3c) The signal 𝑆[𝑛], with known 𝛾𝑆𝑆 [𝑙], is then corrupted by additive white noise resulting in
another (noisy) sequence 𝑋[𝑛] = 𝑆[𝑛] + 𝑉[𝑛], with 𝑉[𝑛] being uncorrelated white noise
with power 𝜎𝑉2 = 1 which is independent of 𝑆[𝑛]. We wish to filter the noisy signal 𝑋[𝑛]
using an FIR Wiener filter with length 𝑀 = 2. Find the coefficients of the FIR filter, i.e., the
filter that minimizes 𝜎𝐸2 = 𝐸{(𝑆[𝑛] − ∑𝑀−1 2
𝑘=0 ℎ[𝑘]𝑋[𝑛 − 𝑘]) }.
Hint: Remember that the inverse of a 2-by-2 matrix is given by the following formula
𝑎 𝑏 −1 1 𝑑 −𝑏
( ) = ( )
𝑐 𝑑 𝑎𝑑 − 𝑏𝑐 −𝑐 𝑎
3 (4)
Problem 4 (4+16=20): Sampling, filtering, and rate-conversion
𝑡𝑛 = 𝑛𝑇𝑥
𝑥𝑎 (𝑡) 𝑥[𝑛] 𝑥̃[𝑛] 𝑣[𝑘] 𝑤[𝑘] 𝑦[𝑚]
𝑔[𝑛] ↑𝐼 ℎ[𝑘] ↓𝐷
𝐼
Rate: 𝐹𝑥 Rate: 𝐹𝑦 = 𝐹𝑥
𝐷
𝐼
Fig. 2: Rate-conversion by a fractional factor 𝐷
The continuous-time signal 𝑥𝑎 (𝑡) = 2 cos 400𝜋𝑡 + 4 cos 800𝜋𝑡 is sampled at every 𝑇𝑥 second
to generate the discrete-time sequence 𝑥[𝑛] = 𝑥𝑎 (𝑡)|𝑡 =𝑛𝑇𝑥 .
4a) What is the minimum sampling rate, or the Nyquist rate, that ensures that 𝑥𝑎 (𝑡) can be
reconstructed from samples 𝑥[𝑛]? Plot the spectrum of the discrete-time sequence 𝑥[𝑛]
obtained by sampling 𝑥𝑎 (𝑡) at Nyquist rate, i.e., 𝑋(𝑓) = ∑∞
𝑛=−∞ 𝑥[𝑛]𝑒
−𝑗2𝜋𝑓𝑛
with 𝑓 = 𝐹𝑇𝑥
being the normalized frequency.
4b) Signal 𝑥𝑎 (𝑡) is now sampled at every 𝑇𝑥 = 0.625 ms. You want to remove the high-
frequency component and, thereafter, change the sampling rate in digital domain to 𝐹𝑦 =
1200 Hz. Figure 2 shows the schematic that will perform this task. Filter
ℎ[𝑘] is a lowpass filter with frequency response
1
1, |𝑓𝑣 | ≤ 2 max(𝐼,𝐷)
𝐻(𝑓𝑣 ) = {
0, otherwise
Assume that filter 𝑔[𝑛] is an ideal lowpass filter. Provide the range of frequencies
that can be used as valid cutoff frequency so that 𝑥̃[𝑛] only contains the low-
frequency sinusoidal sequence. (2p)
Is the sequence 𝑥̃[𝑛] periodic or aperiodic? Motivate. (2p)
Provide the smallest integer values of 𝐼 and 𝐷 needed for this rate conversion? (2p)
Sketch the spectra of signals 𝑥̃[𝑛], 𝑣[𝑘], 𝑤[𝑘] and 𝑦[𝑚]. (8p)
Comment on whether any information is lost in the rate conversion. Motivate. (2p)
4 (4)
Appendix: TTT4120 Table of formulas, 2018
A. Sequences:
1−𝛼𝑁
∑𝑁−1 𝑛
𝑛=0 𝛼 = 1−𝛼
1 1
|𝛼| < 1 ⇒ ∑∞ 𝑛
𝑛=0 𝛼 = 1−𝛼 and − ∑−∞ 𝑛
𝑛=−1 𝛼 = 1−𝛼
1−𝛼𝑁 𝑁𝛼𝑁
∑𝑁−1 𝑛
𝑛=0 (𝑛 + 1)𝛼 = (1−𝛼)2 − 1−𝛼 ; 𝛼≠1
1
|𝛼| < 1 ⇒ ∑∞ 𝑛
𝑛=0(𝑛 + 1)𝛼 = (1−𝛼)2
B. Linear convolution:
𝑌(𝑧) = 𝐻(𝑧)𝑋(𝑧)
𝑌(𝑓) = 𝐻(𝑓)𝑋(𝑓)
C. Transforms:
Z-transform: 𝐻(𝑧) = ∑∞
𝑛=−∞ ℎ[𝑛]𝑧
−𝑛
DTFT: 𝐻(𝑓) = ∑∞
𝑛=−∞ ℎ[𝑛]𝑒
−𝑗2𝜋𝑓𝑛
D. Sampling theorem:
Given an analog signal 𝑥𝑎 (𝑡) sampled at 𝐹𝑠 = 1/𝑇. The DTFT of the resulting discrete-time
sequence 𝑥[𝑛] = 𝑥𝑎 (𝑡)|𝑡=𝑛𝑇 is given by
𝑋(𝑓) = 𝑋(𝐹/𝐹𝑠 ) = 𝐹𝑠 ∑∞
𝑘=−∞ 𝑋([𝑓 − 𝑘]𝐹𝑠 )
𝑣(𝑚𝑇𝑦 ) = ∑∞
𝑘=−∞ ℎ[(𝑚𝐷 − 𝑘)𝑇𝑥 ]𝑥(𝑘𝑇𝑥 ) 𝑚∈ℤ
𝑦(𝑙𝑇𝑦 ) = ∑∞
𝑛=−∞ ℎ[(𝑙 − 𝑛𝐼)𝑇𝑦 ]𝑥(𝑛𝑇𝑥 ) 𝑙∈ℤ
𝐷
Rate conversion where 𝑇𝑦 = 𝐷𝑇𝑣 = 𝐼 𝑇𝑥
𝑦(𝑙𝑇𝑦 ) = ∑∞
𝑚=−∞ ℎ[(𝑙𝐷 − 𝑚𝐼)𝑇𝑣 ]𝑥(𝑚𝑇𝑥 ) 𝑙∈ℤ
Given a wide-sense stationary and ergodic sequence 𝑋[𝑛] with infinite energy