EC362 -Digital Signal Processing Lab

DEPARTMENT OF
ELECTRONICS & COMMUNICATION ENGINEERING
LABORATORY MANUAL - R18
EC362-DIGITAL SIGNAL PROCESSING LAB

R.V.R & J.C COLLEGE OF ENGINEERING

(Autonomous)
(Approved by AICTE)
CHANDRAMOULIPURAM ::CHOWDAVARAM
GUNTUR-522019
EC362 -Digital Signal Processing Lab

EC362-DIGITAL SIGNAL PROCESSING LAB

Course Objectives:

1. To Understand the basic signal operations.

2. To Understand the concepts of Analog and Digital modulations.
3. To Implement the FIR and IIR filters.
4. To provide an awareness on the usage of DAQ in signal processing Applications.

Course Outcomes:

After studying this course, the students would be able to-

1. Design the basic operations of Signal processing.

2. Design Analog and Digital modulation techniques.
3. Design IIR, and FIR filters for band pass, band stop, low pass and high pass filters
4. Design and Implement the acquisition and generation of signal, Audio signal tone measurement,
equalization using DAQ.

List of Experiments

1. Generating a waveform and acquiring a same signal using DAQ

2. Generating multiple waveform signals and Acquiring a same signal using DAQ
3. Audio Equalizer using DAQ
4. Generating a signal and adding noise to the signal and removing the noise using filters
5. Generating the Audio signal and finding frequency of the tone using DAQ
6. Determination of Power Spectrum of a signal
7. Simulation of Amplitude Shift Keying
8. Simulation of Frequency shift keying
9. Simulation of Convolution and Correlation
10. Design and Simulation of FIR filter
EC362 -Digital Signal Processing Lab

Experiment 1:
Generating a Waveform Signal and Acquiring a same Signal using myDAQ
Requirements:
1. LabVIEW 2017 or Later
2. myDAQ with MSP Connector
3. Wires
Procedure:

1. Launch NI LabVIEW 2017 or Later.

2. Press Ctrl+N for New VI and Toggle the window using Ctrl+T.
3. For Generating a waveform Signal follow the below Steps.
Right Click on Block Diagram-> Express-> Simulate Signal-> Select Any Signal in
Signal Type Enter Frequency, Amplitude, Offset and Phase.
4. For Passing the Waveform Signal to myDAQ follow the below Steps
Right Click on Block Diagram->Measurement I/O-> NI DAQmx-> DAQAssistant
Place DAQ Assistant on Block Diagram then Generate Signal->Analog Output-
>Voltage->myDAQ 2-> Select AO0 then Finish.
Set Generating Mode as Continuous Samples and Tap OK.
5. For Acquiring the Same Signal follow the below Steps
Right Click on Block Diagram->Measurement I/O->NI DAQmx->DAQAssistant
Place DAQ Assistant on Block Diagram then Acquire Signal->Analog Input-
>Voltage->myDAQ 2-> Select AI0+ then Finish.
Set Acquisition Mode as Continuous Sample and Tap OK.
Connect a waveform graph indicator. See the Front Panel of aVI

Figure 1.1 Front Panel

EC362 -Digital Signal Processing Lab

Figure 1.2 Block Diagram

myDAQ Connection: Connect AO0 -> AI0(+) then AI0(-) -> AGND

VIVA Questions:
1. What does DAQ mean?

Data acquisition (DAQ) is the process of measuring an electrical or physical phenomenon such as
voltage, current, temperature, pressure, or sound with a computer. A DAQ system consists of
sensors, DAQ measurement hardware, and a computer with programmable software.

2. How does a DAQ work?

A complete data acquisition system consists of DAQ hardware, sensors and actuators, signal
conditioning hardware, and a computer running DAQ software. A sensor, which is a type of
transducer, is a device that converts a physical property into a corresponding electrical signal (e.g.,
strain gauge, thermistor).

3. What is a DAQ Assistant VI?

The DAQ Assistant is an easy-to-use graphical interface for configuring measurement tasks and
channels and for customizing timing, triggering, and scales without programming.

4. What are the major components used for data acquisition?

All data acquisition systems consist of three essential elements – Sensor, Signal Conditioning, and
Analog-to-Digital Converter (ADC).
EC362 -Digital Signal Processing Lab

5. What is DAQ in LabVIEW?

Since the intention is to collect data from a DAQ card (hardware) the Hardware Input and Output
folder has been chosen followed by DAQmx, Analog Measurements and voltage. DAXmx is the
method LabVIEW communicates with the hardware. It is a low level language which sits
between LabVIEW and the hardware.

6. What is DAQmx in LabVIEW?

NI-DAQmx is an NI instrument driver that controls every aspect of your DAQ system, including
signal conditioning, from configuration to programming in LabVIEW to low-level OS and device
control

7. What is DAQ card?

Data acquisition (DAQ) is the process of measuring an event in terms of voltage, current, pressure,
temperature or sound. A DAQ system includes high-speed data acquisition measurement hardware
(a DAQ card or module), input devices such as sensors, and a computer or processor.
EC362 -Digital Signal Processing Lab

Experiment 2: Generating Multiple Waveform Signals and Acquiring a same Signal using
myDAQ
Requirements:
1. LabVIEW 2017 or Later
2. myDAQ with MSP Connector
3. Wires
Procedure:
1. Launch NI LabVIEW 2017 or Later
2. Press Ctrl+N for New VI and Toggle the window using Ctrl+T
3. For Generating a waveform Signal follow the below Steps
Right Click on Block Diagram-> Signal Processing-> Waveform
Generation->Basic Function Generator
Enter Frequency, Amplitude, Offset, Phase and Create Enum Control
for Signal Type (Sine, Square, Triangle and Sawtooth)
Based on the user Selection Signal type Waveform Will be Generated and Acquired
4. For Passing the Waveform Signal to myDAQ follow the below Steps
Right Click on Block Diagram->Measurement I/O-> NI DAQmx-> DAQAssistant
Place DAQ Assistant on Block Diagram then Generate Signal->Analog Output-
>Voltage->myDAQ 2-> Select AO0 then Finish
Set Generating Mode as Continuous Samples and Tap OK
5. For Acquiring the Same Signal follow the below Steps
EC362 -Digital Signal Processing Lab

Right Click on Block Diagram->Measurement I/O->NI DAQmx->DAQAssistant

Place DAQ Assistant on Block Diagram then Acquire Signal->Analog
Input->Voltage->myDAQ 2-> Select AI0+ then Finish
Set Acquisition Mode as Continuous Sample and Tap OK.
Connect a waveform graph indicator.

.
Figure 2.1 FrontPanel

Figure 2.2 Block Diagram

myDAQ Connection: Connect AO0 -> AI0(+) then AI0(-) -> AGND
EC362 -Digital Signal Processing Lab

Experiment3:
Audio Equalizer using myDAQ
Requirements:
1. NI LabVIEW 2017 or later
2. NI myDAQ
3. 3.5mm AUX Cable and Ear Phone
Procedure:
1. Launch LabVIEW 2017 or Later
2. Create a New VI by Pressing Ctrl+N and Toggle using Ctrl+T
3. To configure DAQ Assistant Follow these Steps
Right Click on Block diagram-> Measurement I/O-> NI DAQmx-> Select DAQAssistant
Place DAQ Assistant on Block Diagram then Select Acquire Signal-> Analog Input-
> Voltage->myDAQ 2-> Select Audio Input Left and Audio Input Right then Finish.
Select Acquisition mode as Continuous Sample and Set Samples to Read as 12k and Rate
as 32kHz
Select Voltage input as +2V and -2V for Both Channels
4. To remove Noise in the Acquired Signal use filtering techniques.
Right Click on Block Diagram-> Express-> Signal Analysis->Filter
5. Take 3 Filter Function with Low Filter with 400Hz as Cutoff Frequency, for 2nd filter as Band
Pass Filters Set Lower Cutoff frequency as 500 and Higher Cutoff Frequency as 2500 and
for 3rd filter select band pass filter with frequency range of 3000 to10000.
6. For Generate the Acquired signal.
Place DAQ Assistant on Block Diagram then Select Generate Signal-> Analog Output-
> Voltage->myDAQ 2-> Select Audio Output Left and Audio output Right then Finish.
Select Acquisition mode as Continuous Sample and Set Samples to Write as 100 and
Rate as 44.1kHz
Select Voltage input as +2V and -2V for Both Channels and Tap Ok
7. Connections: Connect AUX cable one end in Mobile to Audio IN
(myDAQ) Connect Ear phone in Audio Out(myDAQ)
EC362 -Digital Signal Processing Lab

Figure 3.1 Front Panel

Figure 3.2 Block Diagram

VIVA Questions:
1. What is difference between bass and treble?

The treble is the highest sound in music while the bass is the lowest sound in music. The treble is
located on the line in the staff that is a space higher than the bass while the bass is located on the line in
the staff that is a space lower than the treble.

2. What is speaker treble?

Bass and Treble is a two-band Equalizer. The Bass control is a low-shelf filter with the half gain
frequency at 250 Hz. The Treble control is a high-shelf filter with the half gain frequency at 4000 Hz.
All slider controls have a gain range of +/- 30 dB.
EC362 -Digital Signal Processing Lab

3. What is mid bass and treble?

Mid frequencies are strong for things like guitars, vocals, piano etc. You can control the mid by changing the
bass and treble. Increasing the bass and treble will leave the mid frequencies reduced. Reducing the treble and
bass will effectively mean the mid frequencies are boosted.

4. Does treble affect bass?

Bass' refers to all the sounds having a low frequency or low pitch. This includes sounds the bass guitars
and bass drums in music or a male voice. rock music sounds best with a higher middle. 'Treble'
includes all the high pitched sounds.
5. What is audio equalizer?

Equalization, or EQ for short, means boosting or reducing (attenuating) the levels of different
frequencies in a signal. The most basic type of equalization familiar to most people is the treble/bass
control on home audio equipment. The treble control adjusts high frequencies, the bass control adjusts
low frequencies.
EC362 -Digital Signal Processing Lab

Experiment 4:

Generating a Signal and adding Noise to the Signal and removing the noise using Filters
Requirements:
1. NI LabVIEW 2017 or Later
Procedure:
1. Launch LabVIEW 2017 or Later
2. Create a New VI by Pressing Ctrl+N and Toggle using Ctrl+T
3. To Generate Signal with Noise Select Simulate Signal with Add Noise from Block
Diagram-> Express-> Input-> SimulateSignal
4. To remove Noise in the Acquired Signal use filtering techniques.
1.1. Right Click on Block Diagram-> Express-> Signal Analysis-> Filter
5. Take Filter Function with Low Filter with 100Hz as Cutoff Frequency

Figure 4.1 Front Panel and Block Diagram

EC362 -Digital Signal Processing Lab

Experiment 5: Generating the Audio Signal and Finding Frequency of the

Tone
Requirements:
1. NI LabVIEW 2017 or later
2. NI myDAQ
3. 3.5mm AUX Cable and Ear Phone
Procedure:
1. Launch LabVIEW 2017 or Later
2. Create a New VI by Pressing Ctrl+N and Toggle using Ctrl+T
3. To configure DAQ Assistant Follow these Steps
Right Click on Block diagram-> Measurement I/O-> NI DAQmx-> Select DAQAssistant
Place DAQ Assistant on Block Diagram then Select Acquire Signal-> Analog Input->
Voltage->myDAQ 2-> Select Audio Input Left then Finish.
Select Acquisition mode as Continuous Sample and Set Samples to Read as 100 and Rate as
1kHz
Select Voltage input as +2V and -2V for Both Channels
4. To Find Frequency of the Acquired Signal use Tone Measurements that available in Block
Diagram-> Express-> Signal Analysis-> Tone Measurements select Frequency and create
numeric Indicator.
5. Connections: Connect Microphone to AudioIn(myDAQ)
EC362 -Digital Signal Processing Lab

Figure 5.1. Front Panel and Block Diagram

EC362 -Digital Signal Processing Lab

Experiment 6: Determination of Power Spectrum of a signal

Requirements

1. Lab VIEW 2017 or Later

Procedure:

1. Launch NI LabVIEW 2017 or Later

2. Press Ctrl+N for New VI and Toggle the window using Ctrl+T
3. For Generate a signal source follow the below Steps
• Right Click on Block Diagram->Functions->Express->input->simulate signal
• Enter Frequency, Amplitude, Offset, and Phase.(Adjust these parameters as per our requirement)
4. Wire the output of Simulated signal to the spectral measurement(Configure the spectral window as power
spectrum for one case and PSD for another case)
5. To display the output signal for both the cases, place a waveform graph
(Controls>>Modern>>Graph>>waveform graph) on the front panel.

Figure 1.1: Block diagram for a VI

EC362 -Digital Signal Processing Lab

Figure 1.2: Front panel for a VI

VIVA Questions:

1. What is the power spectrum of a signal?

The power spectrum of a time series describes the distribution of power into frequency components
composing that signal The statistical average of a certain signal or sort of signal (including noise) as
analyzed in terms of its frequency content, is called its spectrum
2. What is power spectral density used for?
A Power Spectral Density (PSD) is the measure of signal's power content versus frequency. A PSD is
typically used to characterize broadband random signals. The amplitude of the PSD is normalized by
the spectral resolution employed to digitize the signal

3. What is the relation between power spectrum density and auto-correlation function?
Power spectrum density is basically Fourier transform of auto-correlation function of power signal. This
property is helpful for calculating power of any power signal.

the property is -

P = Rff(0). , where P = power of signal and Rff(0) = the value of auto-correlation(Rff)

EC362 -Digital Signal Processing Lab

of signal at t=0.

If Rff is Real and Even then itsPower spectrum density( PSDf) also Real and Even .

4. What are the properties of PSD.

PSD is a real function of frequency

sxx(w) = sxx(−w)
EC362 -Digital Signal Processing Lab

Experiment 7: Simulation of Amplitude Shift Keying

Requirements

1. LabVIEW 2017 or Later

Procedure:

1. Launch NI LabVIEW 2017 or Later

2. Press Ctrl+N for New VI and Toggle the window using Ctrl+T
3. For Generating a waveform Signal follow the below Steps
• Right Click on Block Diagram-> Express-> input->Simulate Signal.
• Enter Frequency, Amplitude. (neglect Offset, Phase)
• Use this signal as carrier.
4. Similarly generate the another signal using array with boolean controls.(used as digital input)
5. Based on our requirement, perform the multiplication operation. Adjust the scaling factors to see the
various modulated outputs.(See the block diagram)
6. To display the output signal , place a waveform chart.
(Controls>>Modern>>Graph>>waveform chart) on the front panel.

Figure 2.1:Block diagram for a VI

EC362 -Digital Signal Processing Lab

Figure 2.2:Front Panel for a VI

VIVA Questions:

1. What is modulation?
Modulation is the process of changing any one parameter (amplitude, frequency or phase) of a relatively
high frequency carrier signal in proportion with the instantaneous value of the modulating signal or message
signal.
2. Define amplitude Modulation.
Amplitude Modulation is the process of changing the amplitude of a relatively high frequency carrier signal
in proportion with the instantaneous value of the modulating signal.
3. Define Modulation index and percent modulation for an AM wave.
Modulation index is a term used to describe the amount of amplitude change present in an AM waveform.
It is also called as coefficient of modulation.
Mathematically modulation index is m=Em/Ec
where m= Modulation coefficient,
Em = Peak change in the amplitude of the output waveform voltage.
Ec = Peak amplitude of the unmodulated carrier voltage.
4. Give the bandwidth of AM?
Bandwidth (B) of AM DSBFC is the difference between highest upper frequency and lowest lower side
frequency. fm(max) – maximum modulating signal frequency.
EC362 -Digital Signal Processing Lab

5. Give the expression for modulation index in terms of Vmax and Vmin.
m = Vmax – Vmin / Vmax + Vmin
EC362 -Digital Signal Processing Lab

Experiment 8: Simulation of Frequency shift keying

Requirements

1. LabVIEW 2017 or Later

Procedure:

2. Launch NI LabVIEW 2017 or Later

3. Press Ctrl+N for New VI and Toggle the window using Ctrl+T.
4. For Generate a signal source follow the below Steps
• Right Click on Block Diagram->Functions->Express->input-> simulate signal
➢ Configure the upper simulated signal with the frequency of f1 and also the lower
simulated signal with the frequency of f2.(both are of different frequencies)
5. Create one Boolean array. Use this sequence as message.
6. Connect the VI as per the following block diagram.
7. Change the Boolean values ,given in the array and see the response.
8. To display the output signal for both the cases, place a waveform graph
(Controls>>Modern>>Graph>>waveform graph) on the front panel.

Figure 3.1:Block diagram for a VI

EC362 -Digital Signal Processing Lab

Figure 3.2:Front panel for a VI

VIVA Questions:

1. What is the difference between coherent & non-coherent digital modulation techniques.
Coherent systems need carrier phase information at the receiver and they use matched filters to detect and
decide what data was sent , while non coherent systems do not need carrier phase information and use
methods like square law to recover the data.
2. What does amplitude shift keying mean?
Amplitude-shift keying (ASK) is a form of amplitude modulation that represents digital data as variations
in the amplitude of a carrier wave. In an ASK system, the binary symbol 1 is represented by transmitting
a fixed-amplitude carrier wave and fixed frequency for a bit duration of T seconds
3. What is meant by frequency shift keying?

Frequency-shift keying (FSK) is a frequency modulation scheme in which digital information is

transmitted through discrete frequency changes of a carrier signal. ... The simplest FSK is
binary FSK (BFSK). BFSK uses a pair of discrete frequencies to transmit binary (0s and 1s) information
EC362 -Digital Signal Processing Lab

4. Why we are not preferred ask over PSK and FSK?

We generally don't prefer ASK because it is much prone to noise than FSK and PSK. But, still ASK had
better spectral efficiency compared to FSK and simpler demodulation structure compared to FSK.
Both PSK and FSK receivers usually employ correlators, which consist of an analog multiplier, followed
by AM integrator.
5. Why FSK is required more bandwidth than ask?
This method is less susceptible to errors than ASK. It is mainly used in higher frequency radio
transmission. Frequency spectrum: FSK may be considered as a combination of two ASK spectra
centered around fc1 and fc2, which requires higher bandwidth
EC362 -Digital Signal Processing Lab

Experiment 9: Simulation of Convolution and Correlation

Requirements

1. LabVIEW 2017 or Later

Procedure:

2. Launch NI LabVIEW 2017 or Later

3. Press Ctrl+N for New VI and Toggle the window using Ctrl+T.
4. For Generate a signal source follow the below Steps
• Right Click on Block Diagram->Functions->Express->input-> simulate signal
• Enter Frequency, Amplitude, Offset, and Phase.(Adjust these parameters as per our requirement)
5. Connect the VI as per the following block diagram.
6. To display the output signal for both the cases, place a waveform graph
(Controls>>Modern>>Graph>>waveform graph) on the front panel.

Figure 4.1:Block diagram for a VI

EC362 -Digital Signal Processing Lab

Figure 4.2:Front panel for a VI

VIVA Questions:

1. What are the two methods used for the sectional convolution?
The two methods used for the sectional convolution are 1) Overlap-add method and 2) Overlap-save
method.
2. What is difference between correlation and convolution?

convolution is a technique to find the output of a system of impulse response h(n) for an input x(n) so
basically it is used to calculate the output of a system, while correlation is a process to find the degree of
similarity between two signals.

3. What is the difference between convolution and multiplication?

Convolution, for discrete-time sequences, is equivalent to polynomial multiplication which is not the same
as the term-by-term multiplication. ... The key point of Fourier analysis is that term-by-
term multiplication in one domain is the same as convolution in the other domain.
EC362 -Digital Signal Processing Lab

4. What is circular convolution in DSP?

The circular convolution, also known as cyclic convolution, of two aperiodic functions (i.e. Schwartz
functions) occurs when one of them is convolved in the normal way with a periodic summation of the
other function. That situation arises in the context of the circular convolution theorem

5. What is the convolution sum?

Convolution sum and product of polynomials— The convolution sum is a fast way to find the coefficients
of the polynomial resulting from the multiplication of two polynomials

6. What is zero padding? What are its uses?

Let the sequence x(n) has a length L. If we want to find the N-point DFT(N>L) of the sequence x(n), we
have to add (N-L) zeros to the sequence x(n). This is known as zero padding. The uses of zero padding
are 1) We can get better display of the frequency spectrum. 2) With zero padding the DFT can be used in
linear filtering.
EC362 -Digital Signal Processing Lab

Experiment 10:

Design and Simulate a low pass filter having the following specifications: pass band
response=0.1dB, pass band frequency=1200Hz,Stop band attenuation=30dB,stop band
frequency=2200Hz,and sampling rate=8000Hz.

Requirements

1. LabVIEW 2017 or Later

2. DFD Toolkit

Procedure:

1. Launch NI LabVIEW 2017 or Later

2. Press Ctrl+N for New VI and Toggle the window using Ctrl+T.
3. To place the DFD classical filter Design,
• Right Click on Block Diagram->signal processing- >Digital filter design->filter design> classical
filter.
4. The magnitude response of the filter and zero/pole plot are displayed based on the specifications in the
dialog box.
5. Here the equiripple method is chosen in the design method.
6. Once this Express VI is configured, its label is changed based on the filter type specified. The filter type
gets displayed on the block diagram.
7. Enter the specifications of the filter in the configuration dialog box which appears when placing this
express VI.
8. To place the DFD filter analysis,
a. Right Click on Block Diagram->signal processing- >Digital filter design-> Analysis> filter
Analysis.
9. Additional information on the designed filter such as phase, group delay, impulse response,unit
response, frequency response and zero/plot can be seen by using the DFD filter analysis.
➢ Functions>Digital filter design>filter analysis>DFD filter analysis
10. Connect the reaming VI as per the following block diagram.
11. To display the output, place a waveform graph
(Controls>>Modern>>Graph>>waveform graph) on the front panel.
EC362 -Digital Signal Processing Lab

12. Similarly connect the block diagram for IIR filter(But in the configuration dialog box select design
method as Butterworth)

Figure 5.1: Block diagram for a VI

Figure 5.2: Front Panel for a VI

EC362 -Digital Signal Processing Lab

VIVA Questions:

1. What are the different types of filters based on impulse response?

Based on impulse response the filters are of two types
1. IIR filter

2. FIR filter

The IIR filters are of recursive type, whereby the present output sample depends on the present input,
past input samples and output samples. The FIR filters are of non recursive type, whereby the present
output sample depends on the present input sample and previous input samples.
2. Distinguish between FIR filters and IIR filters.

FIR filter These filters can be easily designed to have perfectly linear phase. FIR filters can be realized
recursively and non-recursively. Greater flexibility to control the shape of their magnitude response.
Errors due to round off noise are less severe in FIR filters, mainly because feedback is not used. IIR filter
These filters do not have linear phase. IIR filters are easily realized recursively. Less flexibility, usually
limited to specific kind of filters. The round off noise in IIR filters is more.

3. What are the design techniques of designing FIR filters?

There are three well known methods for designing FIR filters with linear phase .
They are (1) Window method (2) Frequency sampling method (3) Optimal or minimax design.
4. What is Gibb’s phenomenon?
One possible way of finding an FIR filter that approximates H(w) would be to truncate the infinite Fourier
series at n=±(N-1/2). Direct truncation of the series will lead to fixed percentage overshoots and
undershoots before and after an approximated.
5. Give the equations specifying the following windows.
a. Rectangular window
b. Hamming window
c. Hanning window
d. Bartlett window
e. Kaiser window
a. Rectangular window:
The equation for Rectangular window is given by
W(n) = 1 for 0 ≤ n ≤ M-1
= 0 otherwise
EC362 -Digital Signal Processing Lab

b. Hamming window: The equation for Hamming window is given by

WH(n) = 0.54-0.46 cos (2 n/M-1) for 0 ≤ n ≤ M-1
= 0 otherwise
c. Hanning window: The equation for Hanning window is given by
WHn(n) = 0.5[1- cos (2 n/M-1 ] for 0 ≤ n ≤ M-1
= 0 otherwise
d. Bartlett window: The equation for Bartlett window is given by
WT(n) = 1-{2|n-(M-1)/2|}/(M-1) for 0 ≤ n ≤ M-1
= 0 otherwise
e. Blackman window: The equation for Blackman window is given by
WH(n) = 0.42-0.5 cos (2 n/M-1)+0.08 cos (4 n/M-1) for 0 ≤ n ≤ M-1
= 0 otherwise