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 finitelength sequence 𝑥[𝑛] = 𝛿 [𝑛] + 2𝛿[𝑛 − 5].
a. Find the 10point 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 highpass 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 autocorrelation of a signal 𝑥(𝑛) = cos(𝜋𝑛)𝑠𝑖𝑛𝑐( 2 )
Plot the spectrum of a given realtime 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/DH(z)D/A structure gives an equivalent analog filter with the following specifications: Passband ripple ≤ 3. Design a highpass 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 autocorrelation 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 overlapadd method. ii) 𝐹𝑠 > Nyquist rate. 0 < 𝜔 < 𝜋 Design a length25 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 10point DFT of 𝑥[𝑛]. Use only a 5point 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 10point 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 lowpass 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 secondorder 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 finitelength sequence 𝑥[𝑛] = 𝛿 [𝑛] + 2𝛿[𝑛 − 5].2.
2.2. Take 𝑥(𝑛) = {1.4. 𝜋 A cosinusoidal signal of frequency 100 Hz. Compute the 2𝑁point 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 𝑁point 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 secondorder 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 secondorder 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 highpass 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 highpass 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 5point 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/DH(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 51order FIR lowpass 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 overlapsave method.4} 13a.3}. Using Matlab. Sampling Rate = 2 kHz. generate two vectors.100). find the autocorrelation 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 autocorrelation 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 highpass 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 lowpass 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 secondorder 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 length25 digital filter 𝐻 (𝜔) = { using a Hamming window. Compute the 2𝑁point 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 10point 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 secondorder 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 finitelength 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 secondorder 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 𝑁point IDFT of the sequence 𝑋 (𝑚 ).5. The recorded speech contains information within 1800 Hz. Find the 10point 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 cosinusoidal 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 highpass 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 autocorrelation 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/DH(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 51order FIR lowpass 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 length25 digital Hilbert transformer using a Hamming window. find the linear and circular convolution of 𝑥1 [𝑛] and 𝑥2 [𝑛] 4b. Find the 10point DFT of 𝑥[𝑛].1a. ℎ3 (𝑛) = 𝑛𝑢(𝑛) and ℎ4 (𝑛) = 𝑢(𝑛 − 2). 7a. consider a finitelength 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 2𝑁point 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 realtime signal using DSK. 11a. In matlab. where 𝑥(𝑛) is the 𝑁point 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/DH(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 secondorder 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 cosinusoidal 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 autocorrelation 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 discretetime 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 highpass FIR with the following specifications: Stopband: 0 to 1500 Hz Passband: 2500 to 4000 Hz Stopband attenuation: 40dB Passband ripple: 0.