CHAPTER

INTRODUCTION TO DIGITAL
SIGNAL PROCESSING
1
CHAPTER OUTLINE
1.1 Basic Concepts of Digital Signal Processing ......................................................................................1
1.2 Basic Digital Signal Processing Examples in Block Diagrams ..............................................................2
1.2.1 Digital Filtering ........................................................................................................... 3
1.2.2 Signal Frequency (Spectrum) Analysis ........................................................................... 4
1.3 Overview of Typical Digital Signal Processing in Real-World Applications ...........................................5
1.3.1 Digital Crossover Audio System ..................................................................................... 5
1.3.2 Interference Cancellation in Electrocardiography ............................................................ 6
1.3.3 Speech Coding and Compression .................................................................................. 6
1.3.4 Compact-Disc Recording System ................................................................................... 7
1.3.5 Vibration Signature Analysis for Defected Gear Tooth ...................................................... 9
1.3.6 Digital Image Enhancement ........................................................................................ 10
1.4 Digital Signal Processing Applications ........................................................................................... 12
1.5 Summary ...................................................................................................................................... 12

1.1 BASIC CONCEPTS OF DIGITAL SIGNAL PROCESSING

Digital signal processing (DSP) technology and its advancements have dramatically impacted our mod-
ern society everywhere. Without DSP, we would not have digital/Internet audio or video; digital re-
cording; CD, DVD, MP3 players, iPhone, and iPad; digital cameras; digital and cellular telephones;
digital satellite and TV; or wire and wireless networks. Medical instruments would be less efficient.
It would be impossible to provide precise diagnoses if there were no digital electrocardiography (ECG),
or digital radiography and other medical imaging modalities. We would also live in many different
ways, since we would not be equipped with voice recognition systems, speech synthesis systems,
and image and video editing systems. Without DSP, scientists, engineers, and technologists would have
no powerful tools to analyze and visualize data and perform their design, and so on.
The concept of DSP is illustrated by the simplified block diagram in Fig. 1.1, which consists of an
analog filter, an analog-to-digital conversion (ADC) unit, a digital signal (DS) processor, a digital-to-
analog conversion (DAC) unit, and a reconstruction (anti-image) filter.

1
2 CHAPTER 1 INTRODUCTION TO DIGITAL SIGNAL PROCESSING

Analog Band-limited Digital Processed Output Analog

input signal signal digital signal signal output
Analog DS Reconstruction
ADC DAC
filter processor filter

FIG. 1.1
A digital signal processing scheme.

As shown in the diagram, the analog input signal, which is continuous in time and amplitude, is
generally encountered in our real life. Examples of such analog signals include current, voltage,
temperature, pressure, and light intensity. Usually a transducer (sensor) is used to convert the none-
lectrical signal to the analog electrical signal (voltage). This analog signal is fed to an analog filter,
which is applied to limit the frequency range of analog signals prior to the sampling process. The
purpose of filtering is to significantly attenuate aliasing distortion, which will be explained in
Chapter 2. The band-limited signal at the output of the analog filter is then sampled and converted
via the ADC unit into the digital signal, which is discrete both in time and in amplitude. The digital
signal processor then accepts the digital signal and processes the digital data according to DSP rules
such as lowpass, highpass, and bandpass digital filtering, or other algorithms for different applica-
tions. Note that the digital signal processor unit is a special type of a digital computer and can be a
general-purpose digital computer, a microprocessor, or an advanced microcontroller; furthermore,
DSP rules can be implemented using software in general.
With the digital signal processor and corresponding software, a processed digital output signal
is generated. This signal behaves in a manner based on the specific algorithm used. The next block
in Fig. 1.1, the DAC unit, converts the processed digital signal to an analog output signal. As
shown, the signal is continuous in time and discrete in amplitude (usually a sample-and-hold signal,
to be discussed in Chapter 2). The final block in Fig. 1.1 is designated as a function to smooth the
DAC output voltage levels back to the analog signal via a reconstruction (anti-image) filter for the
real-world applications.
In general, analog signal processing does not require software, algorithm, ADC, and DAC. The pro-
cessing relies entirely on the electrical and electronic devices such as resistors, capacitors, transistors,
operational amplifiers, and integrated circuits (ICs).
DSP systems, on the other hand, use software, digital processing, and algorithms; therefore, they
have more flexibility, less noise interference, and no signal distortion in various applications. However,
as shown in Fig. 1.1, DSP systems still require minimum analog processing such as the anti-aliasing and
reconstruction filters, which are musts for converting real-world information to digital form and back
again to real-world information.
Note that there are many real-world DSP applications that do not require DAC, such as the data
acquisition and digital information display, speech recognition, data encoding, and so on. Similarly,
DSP applications that need no ADC include CD players, text-to-speech synthesis, and digital tone gen-
erators, among others. We will review some of them in the following sections.

1.2 BASIC DIGITAL SIGNAL PROCESSING EXAMPLES IN BLOCK DIAGRAMS

We first look at digital noise filtering and signal frequency analysis, using block diagrams.
1.2 BASIC DIGITAL SIGNAL PROCESSING EXAMPLES IN BLOCK DIAGRAMS 3

1.2.1 DIGITAL FILTERING

Let us consider the situation shown in Fig. 1.2, depicting a digitized noisy signal obtained from dig-
itizing analog voltages (sensor output) containing useful low-frequency signal and noise that occupy all
of the frequency range. After ADC, the digitized noisy signal x(n), where n is the sample number, can
be enhanced using digital filtering.
Since our useful signal contains low-frequency components, the high-frequency components above
the cutoff frequency of our useful signal are considered as noise, which can be removed by using a
digital lowpass filter. We set up the DSP block in Fig. 1.2 to operate as a simple digital lowpass filter.
After processing the digitized noisy signal x(n), the digital lowpass filter produces a clean digital signal
y(n). We can apply the cleaned signal y(n) to another DSP algorithm for a different application or con-
vert it to analog signal via DAC and the reconstruction filter.
The digitized noisy signal and clean digital signal, respectively, are plotted in Fig. 1.3, where the top
plot shows the digitized noisy signal, while the bottom plot demonstrates the clean digital signal
obtained by applying the digital lowpass filter. Typical applications of noise filtering include acqui-
sition of clean digital audio and biomedical signal and enhancement of speech recording and others
(Embree, 1995; Rabiner and Schafer, 1978; Webster, 2009).

x(n) DSP y(n)

Digitized noisy input Digital filtering Clean digital signal
FIG. 1.2
The simple digital filtering block.

Noisy signal
2

1
Amplitude

−1

−2
0 0.005 0.01 0.015 0.02 0.025 0.03

1
Amplitude

−1

−2
0 0.005 0.01 0.015 0.02 0.025 0.03
Time (s)
FIG. 1.3
(Top) Digitized noisy signal. (Bottom) Clean digital signal using the digital lowpass filter.
4 CHAPTER 1 INTRODUCTION TO DIGITAL SIGNAL PROCESSING

1.2.2 SIGNAL FREQUENCY (SPECTRUM) ANALYSIS

As shown in Fig. 1.4, certain DSP applications often require that time domain information and the fre-
quency content of the signal be analyzed. Fig. 1.5 shows a digitized audio signal and its calculated
signal spectrum (frequency content), defined as the signal amplitude vs. its corresponding frequency
for the time being via a DSP algorithm, called fast Fourier transform (FFT), which will be studied in
Chapter 4. The plot in Fig. 1.5A is the time domain display of a recorded audio signal with a frequency
of 1000 Hz sampled at 16,000 samples per second, while the frequency content display of plot
(B) displays the calculated signal spectrum vs. frequencies, in which the peak amplitude is clearly
located at 1000 Hz. Plot (C) shows a time domain display of an audio signal consisting of one signal

Analog
Time domain display
input x(n)
Analog DSP
ADC
filter Algorithms
Frequency content display

FIG. 1.4
Signal spectral analysis.

5 6
Signal amplitude

Signal spectrum

0 1000 Hz
2

−5 0
0 0.005 0.01 0 2000 4000 6000 8000
(A) Time (s) (B) Frequency (Hz)

10 6

5
Signal amplitude

Signal spectrum

1000 Hz
4

0
3000 Hz
2
−5

−10 0
0 0.005 0.01 0 2000 4000 6000 8000
(C) Time (s) (D) Frequency (Hz)
FIG. 1.5
Audio signals and their spectra. (A) 1000 Hz audio signal. (B) 1000 Hz audio signal spectrum. (C) Audio
signal containing 1000 and 3000 Hz frequency components. (D) Audio signal spectrum containing 1000 and
3000 Hz frequency components.
 1.3 OVERVIEW OF TYPICAL DIGITAL SIGNAL PROCESSING 5

× 104 Speech data: "We lost the golden chain."

2

Speech amplitude
1

−1

−2
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2
Sample number × 104

200
Amplitude spectrum

150

100

50

0
0 500 1000 1500 2000 2500 3000 3500 4000
Frequency (Hz)
FIG. 1.6
Speech samples and speech spectrum.

of 1000 Hz and another of 3000 Hz sampled at 16,000 samples per second. The frequency content dis-
play shown in plot (D) gives two locations (1000 and 3000 Hz) where the peak amplitudes reside, hence
the frequency content display presents clear frequency information of the recorded audio signal.
As another practical example, we often perform spectral estimation of a digitally recorded speech or
audio (music) waveform using the FFT algorithm in order to investigate the spectral frequency details
of speech information. Fig. 1.6 shows a speech signal produced by a human in the time domain and
frequency content displays. The top plot shows the digital speech waveform vs. its digitized sample
number, while the bottom plot shows the frequency content information of speech for a range from
0 to 4000 Hz. We can observe that there are about 10 spectral peaks, called speech formants, in the
range between 0 and 1500 Hz. Those identified speech formants can be used for applications such
as speech modeling, speech coding, speech feature extraction for speech synthesis and recognition,
and so on (Deller et al., 1999).

1.3 OVERVIEW OF TYPICAL DIGITAL SIGNAL PROCESSING

IN REAL-WORLD APPLICATIONS
1.3.1 DIGITAL CROSSOVER AUDIO SYSTEM
An audio system is required to operate in an entire audible range of frequencies, which may be beyond
the capability of any single speaker driver. Several drivers, such as the speaker cones and horns, each
covering a different frequency range, can be used to cover the full audio frequency range.
6 CHAPTER 1 INTRODUCTION TO DIGITAL SIGNAL PROCESSING

Gain Tweeter:
Digital The crossover passes
Digital highpass filter high frequencies
audio x(n)

Gain Woofer:
Digital The crossover passes
lowpass filter low frequencies

FIG. 1.7
Two-band digital crossover.

Fig. 1.7 shows a typical two-band digital crossover system consisting of two speaker drivers: a
woofer and a tweeter. The woofer responds to low frequencies, while the tweeter responds to high fre-
quencies. The incoming digital audio signal is split into two bands using a digital lowpass filter and a
digital highpass filter in parallel. Then the separated audio signals are amplified. Finally, they are sent
to their corresponding speaker drivers. Although the traditional crossover systems are designed using
the analog circuits, the digital crossover system offers a cost-effective solution with programmable
ability, flexibility, and high quality. This topic is taken up in Chapter 7.

1.3.2 INTERFERENCE CANCELLATION IN ELECTROCARDIOGRAPHY

In ECG recording, there often exists unwanted 60-Hz interference in the recorded data (Webster, 2009).
The analysis shows that the interference comes from the power line and includes magnetic induction,
displacement currents in leads or in the body of the patient, effects from equipment interconnections,
and other imperfections. Although using proper grounding or twisted pairs minimizes such 60-Hz
effects, another effective choice can be the use of a digital notch filter, which eliminates the 60-Hz
interference while keeping all the other useful information. Fig. 1.8 illustrates a 60-Hz interference
eliminator using a digital notch filter. As shown in Fig. 1.8, the acquired ECG signal containing
the 60-Hz interference passes through the digital notch filter. The digital notch filter eliminates the
60-Hz interference and only outputs the clean ECG signal. With such enhanced ECG recording, doctors
in clinics could provide accurate diagnoses for patients. This technique can also be used to remove
60-Hz interference in audio systems. This topic is explored in depth in Chapter 8.

1.3.3 SPEECH CODING AND COMPRESSION

One of the speech coding methods, called waveform coding, is depicted in Fig. 1.9 (top plot), describing
the encoding process, while Fig. 1.9 (bottom plot) shows the decoding processing. As shown in Fig. 1.9
(top plot), the analog signal is first filtered by an analog lowpass filter to remove high-frequency noise
components and is then passed through the ADC unit, where the digital values at sampling instants are
captured by the digital signal processor. Next, the captured data are compressed using data compression
rules to reduce the storage requirement. Finally, the compressed digital information is sent to storage
 1.3 OVERVIEW OF TYPICAL DIGITAL SIGNAL PROCESSING 7

Digital notch filter for

eliminating 60-Hz
ECG signal interference
with 60-Hz
interference ECG recorder with
the removed 60 Hz
interference
60-Hz
interference
ECG
preamplifier

FIG. 1.8
Elimination of 60-Hz interference in electrocardiography (ECG).

Analog
input Storage
Analog DSP
ADC media
filter Compressor
Analog
output
Storage DSP Reconstruction
DAC
media Decompressor filter

FIG. 1.9
(Top plot) Simplified data compressor. (Bottom plot) Simplified data expander (decompressor).

media. The compressed digital information can also be transmitted efficiently, since compression re-
duces the original data rate. Digital voice recorders, digital audio recorders, and MP3 players are products
that use compression techniques (Deller et al., 1999; Li et al., 2014; Pan, 1995).
To retrieve the information, the reverse process is applied. As shown in Fig. 1.9 (bottom plot), the
digital signal processor decompresses the data from the storage media and sends the recovered digital
data to DAC. The analog output is acquired by filtering the DAC output via a reconstruction filter.

1.3.4 COMPACT-DISC RECORDING SYSTEM

A compact-disc (CD) recording system is described in Fig. 1.10 (top plot). The analog audio signal is
sensed from each microphone and then fed to the anti-aliasing lowpass filter. Each filtered audio signal
is sampled at the industry standard rate of 44.1 kilo-samples per second, quantized, and coded to 16 bits
8 CHAPTER 1 INTRODUCTION TO DIGITAL SIGNAL PROCESSING

Left mic
Anti-aliasing 16-bit
LP filter ADC
Encoding
Optics and
Multiplex modulation
Right mic recording
synchronization
Anti-aliasing 16-bit
LP filter ADC

Amplified
left speaker

14-bit Anti-image
DAC LP filter
Optical pickup 4x
demodulation Over-
error correction sampling
14-bit Anti-image
DAC LP filter
CD
Amplified
right speaker
FIG. 1.10
(Top plot) Simplified encoder of the CD recording system. (Bottom plot) Simplified decoder of the CD recording
system.

for each digital sample in each channel. The two channels are further multiplexed and encoded, and
extra bits are added to provide information such as playing time and track number for the listener. The
encoded data bits are modulated for storage, and more synchronized bits are added for subsequent re-
covery of sampling frequency. The modulated signal is then applied to control a laser beam that illu-
minates the photosensitive layer of a rotating glass disc. When the laser turns on and off, the digital
information is etched on the photosensitive layer as a pattern of pits and lands in a spiral track. This
master disc forms the basis for mass production of the commercial CD from the thermoplastic material.
During playback, as illustrated in Fig. 1.10 (bottom plot), a laser optically scans the tracks on a CD
to produce digital signal. The digital signal is then demodulated. The demodulated signal is further
oversampled by a factor of 4 to acquire a sampling rate of 176.4 kHz for each channel and is then passed
to the 14-bit DAC unit. For the time being, we consider the oversampling process as interpolation, that
is, adding three samples between every two original samples in this case, as we shall see in Chapter 11.
After DAC, the analog signal is sent to the anti-image analog filter, which is a lowpass filter to smooth
the voltage steps from the DAC unit. The output from each anti-image filter is fed to its amplifier
and loudspeaker. The purpose of the oversampling is to relieve the higher-filter-order requirement
for the anti-image lowpass filter, making the circuit design much easier and economical
(Ambardar, 1999).
Software audio players that play music from CDs, such as Windows Media Player and RealPlayer,
installed on computer systems, are examples of DSP applications. The audio player has many advanced
features, such as a graphical equalizer, which allows users to change audio with sound effects including
boosting low-frequency content or emphasizing high-frequency content to make music sound more
entertaining (Ambardar, 1999; Embree, 1995; Ifeachor and Jervis, 2002).
 1.3 OVERVIEW OF TYPICAL DIGITAL SIGNAL PROCESSING 9

1.3.5 VIBRATION SIGNATURE ANALYSIS FOR DEFECTED GEAR TOOTH

Gearboxes are widely used in industry and vehicles (Spectra Quest, Inc.). During the extended service
lifetimes, the gear teeth will inevitably be worn, chipped, or missing. Hence, with DSP techniques,
effective diagnostic methods can be developed to detect and monitor the defected gear teeth in order
to enhance the reliability of the entire machine before any unexpected catastrophic events occur.
Fig. 1.11A shows the gearbox, in which two straight bevel gears with a transmission ratio of 1.5:1 in-
side the gearbox are shown Fig. 1.11B. The number of teeth on the pinion is 18. The gearbox input shaft

Output shaft

Gear
Input shaft

Pinion Pinion and gear

(A) Gearbox (B)
Tachometer bracket

Missing of
a tooth
Tachometer
(C) Spectra quest’s gearbox dynamics simulator (GDS) (D) Damaged pinion

FIG. 1.11
Vibration signature analysis of the gearbox. (A) Gearbox, (B) Pinion and gear, (C) Spectra Quest’s Gearbox
Dynamics Simulator (GDS), (D) Damaged pinion.
Courtesy of SpectaQuest, Inc.
10 CHAPTER 1 INTRODUCTION TO DIGITAL SIGNAL PROCESSING

0.5

Amplitude (V)
0

−0.5
0 2 4 6 8 10 12 14
Time (s)
Amplitude (VRMS)

0.02
Meshing frequency
0.01

0
0 1000 2000 3000 4000 5000 6000 7000
Frequency (Hz)
Amplitude (VRMS)

0.02 Sideband frequency fm-fi

Meshing frequency
0.01

0
250 260 270 280 290 300 310 320 330 340 350
Zoomed frequency (Hz)
FIG. 1.12
Vibration signal and spectrum from the good condition gearbox.
Data provided by SpectaQuest, Inc.

is connected to a sheave and driven by a “V” belt drive. The vibration data can be collected by triaxial
accelerometer installed on the top of the gearbox, as shown in Fig. 1.11C. The data acquisition system
uses a sampling rate of 12.8 kHz. Fig. 1.11D shows that a pinion has a missing tooth. During the test, the
motor speed is set to 1000 rpm (revolutions per minute) so the meshing frequency is determined as
fm ¼ 1000(rpm)  18/60 ¼ 300 Hz and input shaft frequency is fi ¼ 1000(rpm)/60 ¼ 16.17 Hz. The base-
line signal and spectrum (excellent condition) from x direction of the accelerometer are displayed in
Fig. 1.12, where we can see that the spectrum contains the meshing frequency component of 300 Hz and
a sideband frequency component of 283.33 (300–16.67) Hz. Fig. 1.13 shows the vibration signature for
the damaged pinion in Fig. 1.11D. For the damaged pinion, the sidebands (fm  fi, fm  2fi…) become
dominant. Hence, the vibration failure signature is identified. More details can be found in Robert Bond
Randall (2011).

1.3.6 DIGITAL IMAGE ENHANCEMENT

We can look at another example of signal processing in two dimensions. Fig. 1.14A shows a picture of
an outdoor scene taken by a digital camera on a cloudy day. Due to this weather condition, the image is
improperly exposed in natural light and comes out dark. The image processing technique called his-
togram equalization (Gonzalez and Wintz, 1987) can stretch the light intensity of an image using the
digital information (pixels) to increase the image contrast, therefore, detailed information can easily be
seen in the image, as we can see in Fig. 1.14B. We will study this technique in Chapter 13.
 1.3 OVERVIEW OF TYPICAL DIGITAL SIGNAL PROCESSING 11

Amplitude (V)
0

−2
0 2 4 6 8 10 12 14
Time (s)
Amplitude (VRMS)

0.04
Meshing frequency
0.02

0
0 1000 2000 3000 4000 5000 6000 7000
Frequency (Hz)
Amplitude (VRMS)

0.02 Sidebands Meshing frequency

Sidebands
0.01

0
250 260 270 280 290 300 310 320 330 340 350
Zoomed frequency (Hz)
FIG. 1.13
Vibration signal and spectrum from the damaged gearbox.
Data provided by SpectaQuest, Inc.

Original image Enhanced image

(A) (B)
FIG. 1.14
Image enhancement. (A) Original image taken on a cloudy day. (B) Enhanced image using the histogram
equalization technique.
12 CHAPTER 1 INTRODUCTION TO DIGITAL SIGNAL PROCESSING

Table 1.1 Applications of Digital Signal Processing

Digital Audio and Speech
Digital audio coding such as CD players, MP3 players, digital crossover, digital audio equalizers, digital stereo and
surround sound, noise reduction systems, speech coding, data compression and encryption, speech synthesis and
speech recognition
Digital telephone
Speech recognition, high-speed modems, echo cancellation, speech synthesizers, DTMF (dual-tone multifrequency)
generation and detection, answering machines
Automobile Industry
Active noise control systems, active suspension systems, digital audio and radio, digital controls, vibration signal
analysis
Electronic Communications
Cellular phones, digital telecommunications, wireless LAN (local area networking), satellite communications
Medical Imaging Equipment
ECG analyzers, cardiac monitoring, medical imaging and image recognition, digital X-rays and image processing
Multimedia
Internet phones, audio and video, hard disk drive electronics, iPhone, iPad, digital pictures, digital cameras, text-to-
voice, and voice-to-text technologies

1.4 DIGITAL SIGNAL PROCESSING APPLICATIONS

Applications of DSP are increasing in many areas where analog electronics are being replaced by DSP
chips, and new applications depend on DSP techniques. With the decrease in the cost of digital signal
processors and increase in their performance, DSP will continue to affect engineering design in our
modern daily life. Some application examples using DSP are listed in Table 1.1.
However, the list in the table by no means covers all the DSP applications. Many application areas
are increasingly being explored by engineers and scientists. Applications of DSP techniques will con-
tinue to have profound impacts and improve our lives.

1.5 SUMMARY
1. An analog signal is continuous in both time and amplitude. Analog signals in the real world include
current, voltage, temperature, pressure, light intensity, and so on. The digital signal contains the
digital values converted from the analog signal at the specified time instants.
2. Analog-to-digital signal conversion requires an ADC unit (hardware) and a lowpass filter attached
ahead of the ADC unit to block the high-frequency components that ADC cannot handle.
3. The digital signal can be manipulated using arithmetic. The manipulations may include digital fil-
tering, calculation of signal frequency content, and so on.
4. The digital signal can be converted back to an analog signal by sending the digital values to DAC to
produce the corresponding voltage levels and applying a smooth filter (reconstruction filter) to the
DAC voltage steps.
5. DSP finds many applications in areas such as digital speech and audio, digital and cellular tele-
phones, automobile controls, vibration signal analysis, communications, biomedical imaging,
image/video processing, and multimedia.