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# DSP Lab Questions

Repeated questions might be found
1. 𝟏

1 1

ℎ1 (𝑛) = { , , },

Using Matlab, plot ℎ(𝑛) for 𝑛 = −5 to 20 if
ℎ2 (𝑛) = ℎ3 (𝑛) = (𝑛 + 1)𝑢(𝑛) and ℎ4 (𝑛) = 𝛿(𝑛 − 2), given 𝟐

4 2

ℎ2 (𝑛) 𝑥
(𝑛)

ℎ1 (𝑛)
ℎ3 (𝑛)

-

ℎ4 (𝑛)

+

+ 𝑦

(𝑛)

2.

Using DSK, prove 𝐷𝐹𝑇 {𝐷𝐹𝑇 {𝐷𝐹𝑇{𝐷𝐹𝑇{𝑥[𝑛]}}}} = 𝑁 2 𝑥[𝑛] for 𝑥[𝑛] = {1,2,3,4}

3.

Using the DSK, determine the response of the system ℎ(𝑛) = 𝑎𝑛 𝑢(𝑛), |𝑎| < 1, when the
input is a unit step sequence. Plot the response for 𝑛 = −5 to 20.

4.

Using Matlab, make a comparison between circular convolution and linear convolution:
Given 𝑥1 [𝑛] = {1, −1, −2,3, −1} and 𝑥2 [𝑛] = {1,2,3}, find the linear and circular
convolution of 𝑥1 [𝑛] and 𝑥2 [𝑛]

5.

Using DSK, obtain the impulse response for the system function given by
1
1 + z −1
4
H (z) =
.
1 −1
1
1
(1 + 2 z ) (1 + 2 z −1 + 4 z −2 )

6.

Using Matlab, consider a finite-length sequence 𝑥[𝑛] = 𝛿 [𝑛] + 2𝛿[𝑛 − 5].
a. Find the 10-point DFT of 𝑥[𝑛].
4𝜋

7.

8.
9.

b. Find a sequence that has a DFT 𝑌 [𝑘] = 𝑒 𝑗 10 𝑘 𝑋[𝑘], where 𝑋 [𝑘] is the 10point DFT of 𝑥[𝑛].
Design a high-pass FIR with the following specifications:
Stopband: 0 to 1500 Hz
Passband: 2500 to 4000 Hz
Stopband attenuation: 40dB
Passband ripple: 0.1dB
Sampling rate: 8000 Hz
Plot the filter response and determine the filter characteristics using DSK. 𝑛

Using Matlab, find the auto-correlation of a signal 𝑥(𝑛) = cos(𝜋𝑛)𝑠𝑖𝑛𝑐( 2 )
Plot the spectrum of a given real-time signal using DSK. Let the sampling frequency be
48 kHz.

Using Matlab. Using Matlab. a stopband frequency of 3kHz and stopband attenuation of 40 dB or more. 13. Find the sequence 𝑦(𝑛) whose DFT is given by 𝑌(𝑘) = 𝐼𝑚[𝑋(𝑘)] where 𝐼𝑚[𝑋(𝑘)] refers to the imaginary part of 𝑋(𝑘) and the frequency resolution of 𝑋(𝑘) is 𝜋/3. Let sampling frequency be 32kHz.1dB Sampling rate: 8000 Hz Plot the filter response and determine the filter characteristics using DSK. . 14. using the Bilinear transformation method. In a speech recording system. design a digital filter H(z) that when used in an A/D-H(z)-D/A structure gives an equivalent analog filter with the following specifications:  Passband ripple ≤ 3. Design a high-pass FIR with the following specifications: Stopband: 0 to 1500 Hz Passband: 2500 to 4000 Hz Stopband attenuation: 40dB Passband ripple: 0. The recorded speech contains information within 1800 Hz.10. ℎ2 (𝑛) = ℎ3 (𝑛) = (𝑛 + 1)𝑢(𝑛) and ℎ4 (𝑛) = 𝛿(𝑛 − 2). 12. Use Butterworth Prototype.4].2.  Stopband attenuation ≥ 15 dB. DSP Lab Questions 2𝜋𝑛 1a. find the auto-correlation of a signal 𝑥(𝑛) = 2cos( 1b. }.2kHz.  Sampling Rate = 2 kHz.3. Use DSK to remove noise. plot the impulse response of the combined system. . Use DSK to solve for 𝑦(𝑛). Using Matlab. given 𝟐 4 2 ℎ2 (𝑛) 7 ). The frequency response of a filter is described by: H(ω) = jω − π ≤ ω ≤ π Design the filter using a rectangular window.02dB and the stopband attenuation must be atleast 50dB. Given the sequence 𝑥(𝑛) = [1. design a type I low pass Chebyshev filter operating at a sampling rate of 9kHz that has a 0. In matlab.01 dB.  Stopband edge=750 Hz. Transition band of 200 Hz is allowed. passband frequency of 2. the speech is recorded in a noisy environment at a sampling rate of 8000 Hz. Take N=71.  Passband edge = 500 Hz. The filter is to be designed by performing a bilinear transformation on an analog system function.8dB ripple in the passband. Verify your result using DSK by passing a sinusoid of frequency 5000Hz to the kit. The allowable passband ripple is 0. 2a. ℎ(𝑛) for 𝑛 = −5 to 20 𝟏 1 1 if ℎ1 (𝑛) = { . 11.

determine the response 𝑦[𝑛].2)𝑛 𝑢(𝑛).0. Consider the analog signal 𝑥𝑎 (𝑡) = 3 cos(2000𝜋𝑡) + 5 sin(6000𝜋𝑡) + 10cos(12000𝜋𝑡). Plot 𝑗. Plot the magnitude spectrum of the filter using DSK. Comment on the result. d.2. compute 𝑦[𝑛] using the overlap-add method. ii) 𝐹𝑠 > Nyquist rate. 0 < 𝜔 < 𝜋 Design a length-25 digital filter 𝐻 (𝜔) = { using a Hamming window.2. 𝑥[𝑛] = {1. Using DSK. Plot the response for 𝑛 = −5 to 20.3. Sketch the discrete time signal after sampling with i) 𝐹𝑠 = 5000 𝐻𝑧.3.3. where 𝑋 [𝑘] is the 10-point DFT of 𝑥[𝑛]. Use only a 5-point circular convolution. Determine the input 0. If ℎ[𝑛] = {1. Use DSK to solve for 𝑦(𝑛). −7. 4a. −7. −𝑗. 0 ≤ 𝑛 ≤ 4 Consider a system with impulse response ℎ(𝑛) = { 2 .1. c.4. determine the response of the system ℎ(𝑛) = 4(−0. A long sequence 𝑥[𝑛] is filtered through a filter with impulse response ℎ[𝑛] to yield the output 𝑦[𝑛]. 𝑦[−2] = 1. 6a. Find the sequence 𝑦(𝑛) whose DFT is given by 𝑌(𝑘) = 𝐼𝑚[𝑋(𝑘)] where 𝐼𝑚[𝑋(𝑘)] refers to the imaginary part of 𝑋(𝑘) and the frequency resolution of 𝑋(𝑘) is 𝜋/3. 𝑜𝑡ℎ𝑒𝑟𝑤𝑖𝑠𝑒 𝑥(𝑛) for 0≤𝑛≤8 that will generate the output sequence ( ) 𝑦 𝑛 = {1. Given the sequence 𝑥(𝑛) = [1. −𝜋 < 𝜔 < 0 magnitude spectrum using DSK. Find the 10-point DFT of 𝑥[𝑛]. Using the DSK.4. < 4 length 51 is used. − 4 ≤ ω ≤ 4 jω Hd (e ) = { . Plot the filter coefficients h[n]. −1.4. 3a.2. ( )𝑛 . … } 5b.3.3. if a rectangular window of π |ω| ≤ π 0.3}.7. 1 Using Matlab. Use parallel realization of first order to realize the filter. 4b.0. . Find a sequence that has the DFT 𝑌 [𝑘] = 𝑒 𝑗 10 𝑘 𝑋[𝑘].3.2}. A low-pass FIR causal filter is to be designed with the following desired frequency response: π π e−j25ω . −2 ≤ 𝑛 ≤ 20 of the system described by the second-order difference equation 𝑦[𝑛] − 𝑦[𝑛 − 1] + 𝑦[𝑛 − 2] = 𝑥[𝑛 − 3] when the input is 𝑥[𝑛] = 𝑛(𝑢[𝑛] − 𝑢[𝑛 − 4]) and the initial conditions are 𝑦[−1] = 2.4]. when the input is a unit step sequence. 5a.5. 3b.5)𝑛 𝑢(𝑛) + 2(0. Comment on the result. 4𝜋 6b.𝑥(𝑛) ℎ1 (𝑛) ℎ3 (𝑛) - ℎ4 (𝑛) + + 𝑦(𝑛) 2b. consider a finite-length sequence 𝑥[𝑛] = 𝛿 [𝑛] + 2𝛿[𝑛 − 5].2.

2.2. Take 𝑥(𝑛) = {1.4. 𝜋 A co-sinusoidal signal of frequency 100 Hz. Compute the 2𝑁-point IDFT of 𝑌(𝑘) in terms of 𝑥(𝑛).8dB ripple in the passband.0} using frequency domain approach.3. Use DSK to remove noise.2. 𝑥(𝑛) = {1. 𝑘 odd 11b. compute 62 𝑥[𝑛] = 62 {1. where 𝑥(𝑛) is the 𝑁-point IDFT of the sequence 𝑋 (𝑚 ). 7b. . The recorded speech contains information within 1800 Hz.5.02dB and the stopband attenuation must be atleast 50dB. Using DSK.2z−1 )(1+0. 11a. 10a.2kHz. using the Bilinear transformation method. 8b. a stopband frequency of 3kHz and stopband attenuation of 40 dB or more.4.Realize the function as cascade of two second-order systems. 9b.7. obtain the impulse response for the system function given by H(z) = 1 4 1+ z−1 1 2 1 2 1 4 (1+ z−1 )(1+ z−1 + z−2 ) . In a speech recording system. amplitude of 3 and phase of 4 radians is 8a.6z−1 ). obtain the impulse response for the system function given by 1−3z−4 H(z) = (1−0. Let the sampling frequency be 48 kHz. Plot the spectrum of a signal taken from signal generator using DSK.8}. Transition band of 200 Hz is allowed. 9a. 10b. Use the DSK for implementing the above.6.8} 0. sampled at frequency 1000Hz and spectrum analysed. In matlab.7a. Using the frequency domain approach. device a matlab function to determine a circular shift 𝑥((𝑛 − 3)8 ). Using Matlab.3.3. The allowable passband ripple is 0. 𝑘 even 𝑌 (𝑘 ) = { .1dB Sampling rate: 8000 Hz Plot the filter response and determine the filter characteristics using DSK.6.0. If 8 samples of the signal 𝑥[𝑛] are used for determining DFT 𝑋[𝑘]. the speech is recorded in a noisy environment at a sampling rate of 8000 Hz.Realize the function as cascade of two second-order systems. 0 ≤ 𝑘 ≤ 2𝑁 − 1. Determine the change in spectrum i) if 𝑥[𝑛] is multiplied by the sequence 𝑒 𝑗𝜋𝑛 ii) if amplitude of 𝑥[𝑛] is doubled.5. 0 ≤ 𝑚 ≤ 𝑁 − 1 and 𝑋(𝑘⁄2).7. passband frequency of 2. Using Matlab. plot the magnitude and phase response. design a type I low pass Chebyshev filter operating at a sampling rate of 9kHz that has a 0.4. given an N point sequence. Design a high-pass FIR with the following specifications: Stopband: 0 to 1500 Hz Passband: 2500 to 4000 Hz Stopband attenuation: 40dB Passband ripple: 0.

13b. A long sequence 𝑥[𝑛] is filtered through a filter with impulse response ℎ[𝑛] to yield the output 𝑦[𝑛]. 14a. The frequency response 𝜋 0. 12b. Using DSK.01 dB.12a. Comment on the result. Use Butterworth Prototype.2. . the value of the autocorrelation is the signal power. 15b. < |𝜔| < of an ideal bandpass filter is given by 0. Design a high-pass FIR with the following specifications: Stopband: 0 to 1500 Hz Passband: 2500 to 4000 Hz Stopband attenuation: 40dB Passband ripple: 0. −1} and 𝑥2 [𝑛] = {1. −7. Use only a 5-point circular convolution. Plot the characteristics of 𝑟(𝑛) = 𝑠(𝑛)ℎ(𝑛). Plot the filter 2𝜋10𝑛 characteristics.9.100) and 𝑥2 = 𝑟𝑎𝑛𝑑𝑛(1. Using DSK. design a digital filter H(z) that when used in an A/D-H(z)-D/A structure gives an equivalent analog filter with the following specifications:  Passband ripple ≤ 3. 14b.7. prove Parseval’s relation for 𝑥(𝑛) = {1.3}.4. Design a 51-order FIR low-pass filter. −1.3.0. 0 ≤ |𝜔| ≤ 8 𝐻(𝜔) = 𝜋 3𝜋 8 3𝜋 8 1.0. ≤ |𝜔| ≤ 𝜋 { 8 Determine its impulse response using matlab.4. −2. make a comparison between circular convolution and linear convolution: Given 𝑥1 [𝑛] = {1. −1. 𝑥1 = 𝑟𝑎𝑛𝑑𝑛(1.2}.3. Multiply the filter coefficients with 𝑠(𝑛) = cos( 𝑁 ).1dB Sampling rate: 8000 Hz Plot the filter response and determine the filter characteristics using DSK. Determine the cross correlation of the two signals and plot. compute 𝑦[𝑛] using the overlap-save method.4} 13a.3}. Using Matlab.  Sampling Rate = 2 kHz. generate two vectors.100). find the auto-correlation of the signal 𝑥 (𝑛) = −4 sin (2𝜋𝑛 ) and show that at zero 36 shift.4.3. The filter is to be designed by performing a bilinear transformation on an analog system function.4. 15a. If ℎ[𝑛] = {1.  Passband edge = 500 Hz. 𝑥[𝑛] = {1.5. find the linear and circular convolution of 𝑥1 [𝑛] and 𝑥2 [𝑛] DSP Lab Questions 1a. −7. In Matlab.7.3.  Stopband edge=750 Hz. 𝑥1 and 𝑥2 using the following code fragment.  Stopband attenuation ≥ 15 dB. Using DSK. ℎ(𝑛) using rectangular window.

ℎ2 (𝑛) = ℎ3 (𝑛) = (𝑛 + 1)𝑢(𝑛) and ℎ4 (𝑛) = 𝛿(𝑛 − 2). Sketch the discrete time signal after sampling with iii) 𝐹𝑠 = 5000 𝐻𝑧.3. find the auto-correlation of a signal 𝑥(𝑛) = 2cos( 4b.1.5. Using Matlab. Comment on the result. plot the impulse response of the combined system. ( )𝑛 . 𝑦[−2] = 1.2. . 3a. Consider the analog signal 𝑥𝑎 (𝑡) = 3 cos(2000𝜋𝑡) + 5 sin(6000𝜋𝑡) + 10cos(12000𝜋𝑡).2. if a rectangular window of π |ω| ≤ π 0. − 4 ≤ ω ≤ 4 jω Hd (e ) = { .3.1dB Sampling rate: 8000 Hz Plot the filter response and determine the filter characteristics using DSK. Design a high-pass FIR with the following specifications: Stopband: 0 to 1500 Hz Passband: 2500 to 4000 Hz Stopband attenuation: 40dB Passband ripple: 0.2. 0 ≤ 𝑛 ≤ 4 Consider a system with impulse response ℎ(𝑛) = { 2 . Take N=71. Comment on the result. 1 7 ). Using Matlab. }. - ℎ4 (𝑛) + + 𝑦(𝑛) Using DSK. 5a. Plot the magnitude spectrum of the filter using DSK.3. A low-pass FIR causal filter is to be designed with the following desired frequency response: π π e−j25ω .2.4]. determine the response 𝑦[𝑛]. .3. Let sampling frequency be 32kHz. ℎ(𝑛) for 𝑛 = −5 to 20 𝟏 1 1 if ℎ1 (𝑛) = { .0. … } Given the sequence 𝑥(𝑛) = [1. Verify your result using DSK by passing a sinusoid of frequency 5000Hz to the kit. 3b. iv) 𝐹𝑠 > Nyquist rate. −2 ≤ 𝑛 ≤ 20 of the system described by the second-order difference equation 𝑦[𝑛] − 𝑦[𝑛 − 1] + 𝑦[𝑛 − 2] = 𝑥[𝑛 − 3] when the input is 𝑥[𝑛] = 𝑛(𝑢[𝑛] − 𝑢[𝑛 − 4]) and the initial conditions are 𝑦[−1] = 2. < 4 length 51 is used. given 𝟐 4 2 ℎ2 (𝑛) 𝑥(𝑛) ℎ1 (𝑛) ℎ3 (𝑛) 2b. Plot the filter coefficients h[n]. Use DSK to solve for 𝑦(𝑛). 2𝜋𝑛 4a.1b. The frequency response of a filter is described by: H(ω) = jω − π ≤ ω ≤ π Design the filter using a rectangular window. 2a. 5b. 𝑜𝑡ℎ𝑒𝑟𝑤𝑖𝑠𝑒 𝑥(𝑛) for 0≤𝑛≤8 that will generate the output sequence 𝑦(𝑛) = {1. Find the sequence 𝑦(𝑛) whose DFT is given by 𝑌(𝑘) = 𝐼𝑚[𝑋(𝑘)] where 𝐼𝑚[𝑋(𝑘)] refers to the imaginary part of 𝑋(𝑘) and the frequency resolution of 𝑋(𝑘) is 𝜋/3. Determine the input 0.

determine the response of the system ℎ(𝑛) = 3(−0.1. Using DSK.2. 0 < 𝜔 < 𝜋 Design a length-25 digital filter 𝐻 (𝜔) = { using a Hamming window. Compute the 2𝑁-point IDFT of 𝑌(𝑘) in terms of 𝑥(𝑛). Plot 𝑗. Using Matlab. In matlab. 9b. Using Matlab.2.3. Use DSK to remove noise.6a.8} 0. where 𝑋 [𝑘] is the 10-point DFT of 𝑥[𝑛]. In a speech recording system. Using the frequency domain approach. passband frequency of 2. 𝑘 even 𝑌 (𝑘 ) = { . 10b. Let the sampling frequency be 48 kHz. obtain the impulse response for the system function given by 1−3z−4 H(z) = (1−0. −𝜋 < 𝜔 < 0 magnitude spectrum using DSK. given an N point sequence.6. device a matlab function to determine a circular shift 𝑥((𝑛 − 5)10 ).Realize the function as cascade of two second-order systems. −2. f.4.3}. 0 ≤ 𝑘 ≤ 2𝑁 − 1. The allowable passband ripple is 0. −𝑗. e.8dB ripple in the passband. 8a. 9a.7.0. using the Bilinear transformation method.6z−1 ). Using the DSK. 𝑘 odd 11b. Plot the response for 𝑛 = −5 to 20. 11a. 0 ≤ 𝑚 ≤ 𝑁 − 1 and 𝑋(𝑘⁄2).2. make a comparison between circular convolution and linear convolution: Given 𝑥1 [𝑛] = {1.4. 4𝜋 6b. design a type I low pass Chebyshev filter operating at a sampling rate of 9kHz that has a 0. the speech is recorded in a noisy environment at a sampling rate of 8000 Hz. compute 62 𝑥[𝑛] = 62 {1.9}. 10a. Use parallel realization of first order to realize the filter. Transition band of 200 Hz is allowed.0} using frequency domain approach.3.6. Using Matlab. consider a finite-length sequence 𝑥[𝑛] = 𝛿 [𝑛] + 2𝛿[𝑛 − 5]. 7b. 8b. 𝑥(𝑛) = {5. −1. obtain the impulse response for the system function given by H(z) = 1 4 1+ z−1 1 2 1 2 1 4 (1+ z−1 )(1+ z−1 + z−2 ) . find the linear and circular convolution of 𝑥1 [𝑛] and 𝑥2 [𝑛] .02dB and the stopband attenuation must be atleast 50dB.2kHz.1)𝑛 𝑢(𝑛) + (0.Realize the function as cascade of two second-order systems. when the input is a unit step sequence. a stopband frequency of 3kHz and stopband attenuation of 40 dB or more.4.0. where 𝑥(𝑛) is the 𝑁-point IDFT of the sequence 𝑋 (𝑚 ).5. The recorded speech contains information within 1800 Hz. Find the 10-point DFT of 𝑥[𝑛]. −1} and 𝑥2 [𝑛] = {1.2z−1 )(1+0.2. Find a sequence that has the DFT 𝑌 [𝑘] = 𝑒 𝑗 10 𝑘 𝑋[𝑘]. Plot the spectrum of a signal taken from signal generator using DSK. 7a.4. Using DSK. Take 𝑥(𝑛) = {1.25)𝑛 𝑢(𝑛).3.4.9.

1dB Sampling rate: 8000 Hz Plot the filter response and determine the filter characteristics using DSK. If 8 samples of the signal 𝑥[𝑛] are used for determining DFT 𝑋[𝑘]. amplitude of 3 and phase of 𝜋 4 radians is sampled at frequency 1000Hz and spectrum analysed.  Passband edge = 500 Hz.  Sampling Rate = 2 kHz. In Matlab. ℎ(𝑛) using rectangular window.  Stopband attenuation ≥ 15 dB. 14b.4. the value of the autocorrelation is the signal power. Multiply the filter coefficients with 𝑠(𝑛) = cos( 𝑁 ).  Stopband edge=750 Hz.5. Plot the filter 2𝜋10𝑛 characteristics. Determine the cross correlation of the two signals and plot.100). The 𝐻(𝜔) = 𝜋 1. 15b. 𝑥1 = 𝑟𝑎𝑛𝑑𝑛(1.7. 15a. A co-sinusoidal signal of frequency 100 Hz.01 dB.12a. prove Parseval’s relation for 𝑥(𝑛) = {1. 8 < |𝜔| < 3𝜋 filter is given by 8 3𝜋 8 ≤ |𝜔| ≤ 𝜋 0. Design a high-pass FIR with the following specifications: Stopband: 0 to 1500 Hz Passband: 2500 to 4000 Hz Stopband attenuation: 40dB Passband ripple: 0. Plot the characteristics of 𝑟(𝑛) = 𝑠(𝑛)ℎ(𝑛). Determine the change in spectrum i) if 𝑥[𝑛] is multiplied by the sequence 𝑒 𝑗𝜋𝑛 ii) if amplitude of 𝑥[𝑛] is doubled. 𝑥1 and 𝑥2 using the following code fragment. generate two vectors. 14a. 13b. frequency response of an ideal bandpass 𝜋 0. The filter is to be designed by performing a bilinear transformation on an analog system function.100) and 𝑥2 = 𝑟𝑎𝑛𝑑𝑛(1. Using DSK. 0 ≤ |𝜔| ≤ 13a. 12b. find the auto-correlation of the signal 𝑥 (𝑛) = −4 sin (2𝜋𝑛 ) and show that at zero 36 shift. Comment on the result. Using DSK. Using Matlab. design a digital filter H(z) that when used in an A/D-H(z)-D/A structure gives an equivalent analog filter with the following specifications:  Passband ripple ≤ 3.3.0. { 8 Determine its impulse response using matlab. Use the DSK for implementing the above. Use Butterworth Prototype.9. Design a 51-order FIR low-pass filter. plot the magnitude and phase response.4} DSP Lab Questions .

1 2 5 plot ℎ(𝑛) for 𝑛 = −5 to 20 𝟏 1 1 ℎ1 (𝑛) = { . - ℎ4 (𝑛) + + 𝑦(𝑛) Read samples of a sinusoid using ADC in DSK and send alternate samples to DAC output. −2. Generate a signal as sum of two sine waves of amplitude 5 V and 2 V and frequency 100 Hz and 2000 Hz. 𝑥(𝑛) = {1. Design a length-25 digital Hilbert transformer using a Hamming window. find the linear and circular convolution of 𝑥1 [𝑛] and 𝑥2 [𝑛] 4b. Find the 10-point DFT of 𝑥[𝑛].1a. ℎ3 (𝑛) = 𝑛𝑢(𝑛) and ℎ4 (𝑛) = 𝑢(𝑛 − 2). 7a. consider a finite-length sequence 𝑥[𝑛] = 𝛿 [𝑛] + 2𝛿[𝑛 − 5]. Find a sequence that has a DFT 𝑌 [𝑘] = 𝑒 𝑗 10 𝑘 𝑋[𝑘]. 6a.3}. . Plot the response for 𝑛 = −5 to 20. when the input is a unit step sequence. 1 −1 1 −1 1 −2 (1 + 2 z ) (1 + 2 z + 4 z ) 5a. Apply the signal to the filter and plot the output. h. Using Matlab.3. Let sampling frequency be 32kHz. make a comparison between circular convolution and linear convolution: Given 𝑥1 [𝑛] = {1.6. Comment on your result. Using DSK. −1} and 𝑥2 [𝑛] = {1.7. determine the response of the system ℎ(𝑛) = 𝑎𝑛 𝑢(𝑛). 5 . 3b. Using Matlab. |𝑎| < 1. device a matlab function to determine a circular shift 𝑥((𝑛 − 3)8 ). Using the DSK. The frequency response of a filter is described by: H(ω) = jω − π ≤ ω ≤ π Design the filter using a rectangular window.3. given an N point sequence. g.4. 4𝜋 6b. where 𝑋 [𝑘] is the 10point DFT of 𝑥[𝑛]. 10} . obtain the impulse response for the system function given by 1 1 + 4 z −1 H (z) = . 2a. Using Matlab. . given ℎ2 (𝑛) 𝑥(𝑛) ℎ1 (𝑛) ℎ3 (𝑛) 2b. 3a.2. Design a suitable IIR filter that will attenuate the high frequency component atleast by 30 dB. 1b.2. 4a. if 𝟐 4 2 9 ℎ2 (𝑛) = {5 . Verify your result using DSK by passing a sinusoid of frequency 5000Hz to the kit.5. Use this program to demonstrate aliasing.8}. Using the frequency domain approach. 5b. Take N=71. 6 . }. −1.

2kHz. In a speech recording system.8} 0. 8b. 𝑘 odd 11b.7b. Use DSK to remove noise. Compute the 2𝑁-point IDFT of 𝑌(𝑘) in terms of 𝑥(𝑛). ny) that produces the total result of linear convolution between two sequences x and y.7.  Passband edge = 500 Hz.  Stopband attenuation ≥ 15 dB. Write a modified linear convolver. 10a. Transition band of 200 Hz is allowed. 0 ≤ 𝑘 ≤ 2𝑁 − 1.5. 𝑦[−4] = 1. Using DSK. amplitude of 3 and phase of 4 radians is sampled at frequency 1000Hz and spectrum analysed. The recorded speech contains information within 1800 Hz. upper cut off frequency 2 KHz. Plot the spectrum of a given real-time signal using DSK. 11a. In matlab. where 𝑥(𝑛) is the 𝑁-point IDFT of the sequence 𝑋 (𝑚 ). design a type I low pass Chebyshev filter operating at a sampling rate of 9kHz that has a 0.4. the speech is recorded in a noisy environment at a sampling rate of 8000 Hz. The allowable passband ripple is 0.  Sampling Rate = 2 kHz. plot the magnitude and phase response. If 8 samples of the signal 𝑥[𝑛] are used for determining DFT 𝑋[𝑘]. Take 𝑥(𝑛) = {1.6.2.3. design a digital filter H(z) that when used in an A/D-H(z)-D/A structure gives an equivalent analog filter with the following specifications:  Passband ripple ≤ 3. 12a. Amplitude modulate a 15 KHz carrier signal with a modulating frequency of 2 KHz. 0 ≤ 𝑚 ≤ 𝑁 − 1 and 𝑋(𝑘⁄2). Determine the change in spectrum i) if 𝑥[𝑛] is multiplied by the sequence 𝑒 𝑗𝜋𝑛 ii) if amplitude of 𝑥[𝑛] is doubled. 9b.02dB and the stopband attenuation must be atleast 50dB. 10b. conv_m (x. Using Matlab. using the Bilinear transformation method.8dB ripple in the passband. y. −2 ≤ 𝑛 ≤ 20 of the system described by the second-order difference equation 𝑦[𝑛] − 𝑦[𝑛 − 1] + 𝑦[𝑛 − 2] = 𝑥[𝑛 − 3] when the input is 𝑥[𝑛] = 𝑛(𝑢[𝑛] − 𝑢[𝑛 − 4]) and the initial conditions are 𝑦[−3] = 2. Use the DSK for implementing the above.01 dB. 𝜋 A co-sinusoidal signal of frequency 100 Hz. determine the response 𝑦[𝑛]. Obtain the spectra of the modulated waveform. nx. Sampling frequency is 16 KHz. passband frequency of 2. . 9a. Let the sampling frequency be 48 kHz. Plot the waveform using MATLAB. a stopband frequency of 3kHz and stopband attenuation of 40 dB or more.  Stopband edge=750 Hz. 𝑘 even 𝑌 (𝑘 ) = { . Using DSP board implement an FIR band stop filter for lower cut off frequency 1 KHz. 8a.

Use Butterworth Prototype.6.The filter is to be designed by performing a bilinear transformation on an analog system function.5. find the auto-correlation of the signal 𝑥 (𝑛) = −4 sin (2𝜋𝑛 ) − 2cos(2𝜋𝑛 ) and show 36 40 14a.2. Using DSK. Using matlab. Given the sequence 𝑥(𝑛) = [1.3.7𝑦(𝑛 − 1) − 𝑦(𝑛 − 2). Use DSK to solve for 𝑦(𝑛). that at zero shift.0.1dB Sampling rate: 8000 Hz Plot the filter response and determine the filter characteristics using DSK. Find the sequence 𝑦(𝑛) whose DFT is given by 𝑌(𝑘) = 𝐼𝑚[𝑋(𝑘)] where 𝐼𝑚[𝑋(𝑘)] refers to the imaginary part of 𝑋(𝑘) and the frequency resolution of 𝑋(𝑘) is 𝜋/3. The frequency response 𝜋 0. −1} and 𝑥2 [𝑛] = {1. the value of the autocorrelation is the signal power.2} 15b. Is the system stable? 14b. prove 𝐷𝐹𝑇 {𝐷𝐹𝑇 {𝐷𝐹𝑇{𝐷𝐹𝑇{𝑥[𝑛]}}}} = 𝑁 2 𝑥[𝑛] for 𝑥 [𝑛 ] = {1.3.3}. 15a. plot the impulse response of the system. Using Matlab. Using DSK. find the linear and circular convolution of 𝑥1 [𝑛] and 𝑥2 [𝑛] .8. 12b.2. Consider the discrete-time system given by the difference equation 𝑦(𝑛) = 𝑥(𝑛) − 4𝑥 (𝑛 − 1) + 3𝑥 (𝑛 − 2) + 1. 13a. 8 < |𝜔| < 3𝜋 of an ideal bandpass filter is given by 3𝜋 8 0. −2. ≤ |𝜔| ≤ 𝜋 { 8 Determine its impulse response using matlab. 0 ≤ |𝜔| ≤ 8 𝐻(𝜔) = 𝜋 1. 13b.9.3.4]. make a comparison between circular convolution and linear convolution: Given 𝑥1 [𝑛] = {1. −1. Design a high-pass FIR with the following specifications: Stopband: 0 to 1500 Hz Passband: 2500 to 4000 Hz Stopband attenuation: 40dB Passband ripple: 0.