LMU08208

DIGITAL SIGNALING PROCESSING

Sub – enabling Outcomes:

a) Apply knowledge of different process and control system in measuring instrument
b) Apply knowledge of compensation in controlling measuring instruments.
c) Apply knowledge and skill of feedback and transfer function for measuring
instrument.
d) Construct and interpret bode plots of the given function.
e) Derive a relationship for convolution sum for discrete.
f) Apply the Z-transform signal function to control the behavior of signals.
g) Construct running average filters and recursive filters.
f) Use direct method to design 11R filters.
g) Calculate the initial location of poles and zero
h) Apply the fine tuning to stabilize the signal processing systems
i) Design IIR filters through analog filters
j) Convert an analog filter to its digital equivalent process “discretization”
k) Describe the bilinear transform the impulse invariant method and the pole-
zero matching technique.
l) Inspect the frequency needed for pre-warping before carrying out the bilinear
transformation.
m)Design band stop filter using the frequency sampling technique
Digital signal processing (DSP) technology has helped in the
advancements of the following
i. Digital/Internet
ii. Audio or video;
iii.Digital recording; CD, DVD, MP3 players, iPhone, and
iPad;
iv.Digital cameras;
v. Digital and cellular telephones;
vi.Digital satellite and TV; or
vii.Wire and wireless networks
 Advantages of digital signal processing
i. DSP improves much medical instruments to be more efficient in
providing precise diagnoses with help of digital
electrocardiography (ECG), or digital radiography and other
medical imaging modalities.
ii. DSP enhances voice recognition systems, speech synthesis
systems, and image and video editing systems.
iii. DSP has helped scientists, engineers, and technologists to have
powerful tools in analyzing and visualizing data and performing
their design.
 General concept of the DSP
DSP system consists of an
a. Analog filter,
b.An analog to digital conversion (ADC) unit,
c. A digital signal (DS) processor,
d.A digital to analog conversion (DAC) unit,
and
e. A reconstruction (anti-image) filter.

The concept of DSP is illustrated by the simplified block

The analog input signal is continuous in time and amplitude it is
derived from our real life. Examples of such analog signals include
current, voltage, temperature, pressure, and light intensity.
A transducer (sensor) is used to convert the nonelectrical 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.

The band-limited signal is 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
applications.
Digital signal processor unit is a special type of a digital computer can
either be general-purpose digital computer, a microprocessor, or an
advanced microcontroller; furthermore,
DSP rules can be implemented using software in general
The processed digital output signal is generated when the specific
logarithm is defined into the digital signal processor.
The generated signal is behaving in similar manner of the defined the
specific algorithm
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
In general, analog signal processing does not require software, algorithm,
ADC, and DAC. The processing relies entirely on the electrical and
electronic devices such as resistors, capacitors, transistors, operational
amplifiers, and integrated circuits (ICs)

A reconstruction (anti-image) filter to smooth the DAC output

voltage levels back to the analog signal which is compatible to the
real-world applications
DSP systems, on the other hand, use software, digital processing, and
algorithms; therefore,
 they have more flexibility,
 less noise interference, and
 no or less signal distortion in various applications.
However, 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.
There are some 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.

DSP applications that need no ADC include

CD players,
Text-to-speech synthesis, and
Digital tone generators, among others.
BASIC DIGITAL SIGNAL PROCESSING EXAMPLES IN BLOCK
DIAGRAMS
Digital noise filtering and signal frequency analysis, using block
diagrams

DIGITAL FILTERING
Depicting a digitized noisy signal obtained from digitizing 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
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.
The digitized noisy signal x(n), the digital lowpass filter produces a
clean digital signal y(n).
The cleaned signal y(n) is applied to another DSP algorithm for a
different application or convert it to analog signal via DAC and the
reconstruction filter.

The simple digital filtering

The noise filtering process include acquisition of clean digital audio and
biomedical signal and enhancement of speech recording.

Digital
lowpass
filter
The simple digital filtering block.
 SIGNAL FREQUENCY (SPECTRUM) ANALYSIS

Signal spectral analysis

DSP applications often require that time domain information and the
frequency content of the signal be analyzed.
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)
 SIGNAL FREQUENCY (SPECTRUM) ANALYSIS

Signal spectral analysis

The plot A is the time domain display of a recorded audio signal with a
frequency of 1000Hz sampled at 16,000 samples per second, while
The plot B is the frequency (domain) content displays the calculated
signal spectrum vs. frequencies, in which the peak amplitude is clearly
located at 1000Hz.
Plot (C) shows a time domain display of an audio signal consisting of
one signal
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
Speech samples
and speech
spectrum.
 DIGITAL SIGNAL PROCESSING IN REAL-WORLD
APPLICATIONS
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.
A woofer and a tweeter.
The woofer responds to low frequencies, while
the tweeter responds to high frequencies.
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.
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.
 INTERFERENCE CANCELLATION IN
ELECTROCARDIOGRAPHY
In ECG recording, there is unwanted 60-
Hz interference in the recorded data
The causes of interference in ECG
comes from the power line which
includes
i. Magnetic induction,
ii. Displacement currents in leads or in
the body of the patient,
iii. Effects from equipment Elimination of 60-Hz
interconnections. interference in
electrocardiography (ECG)
 INTERFERENCE CANCELLATION IN
ELECTROCARDIOGRAPHY

Method can be used to eliminates

interferences
i. Proper grounding or twisted
pairs minimizes such 60-Hz
effects
ii. The use of a digital notch filter,
which eliminates the 60-Hz Elimination of 60-Hz
interference. interference in
electrocardiography (ECG)
Illustrates a 60-Hz interference eliminator using a digital notch filter.
i. The use of a digital notch filter, which eliminates the 60-Hz
interference.
ii.It keeps all the other useful information.
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.
Enhanced ECG recording, helps doctor to provide accurate diagnoses for
patients.
This technique can also be used to remove 60-Hz interference in audio
systems.
 SPEECH CODING AND COMPRESSION
The speech coding methods, called waveform coding, is depicted in (top plot),
describing the encoding process, while (bottom plot) shows the decoding
processing.

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.
The captured data are compressed using data compression rules to reduce the
storage requirement.
The compressed digital information is sent to storage
The compressed digital information can also be transmitted
efficiently, since compression reduces the original data rate.
Digital voice recorders, digital audio recorders, and MP3
players are products that use compression techniques
 COMPACT-DISC RECORDING SYSTEM
A compact-disc (CD) recording system.
The analog audio signal is sensed from each microphone and then fed to the anti-
aliasing lowpass filter.
The filtered audio signal is sampled at the industry standard rate of 44.1kilo-
samples per second, quantized, and coded to 16 bits 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 recovery of sampling
frequency.
The modulated signal is then applied to control a laser beam that
illuminates 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.
 COMPACT-DISC RECORDING SYSTEM

During playback,
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.
The oversampling process is like interpolation, that is, adding three
samples between every two original samples.
 COMPACT-DISC RECORDING SYSTEM

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
 COMPACT-DISC RECORDING SYSTEM

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
 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. A shows the gearbox, in which two straight bevel gears with a
transmission ratio of 1.5:1 inside the gearbox are shown Fig. B.
The number of teeth on the pinion is 18.
The gearbox input shaft is connected to a sheave and driven by a “V”
belt drive
 The baseline signal and spectrum
(excellent condition) from x direction
of the accelerometer are displayed in

The vibration data can be collected by triaxial accelerometer installed

on the top of the gearbox.
The data acquisition system uses a sampling rate of 12.8kHz.
Fig. D shows that a pinion has a missing tooth.
During the test, the motor speed is set to 1000rpm (revolutions per
minute) so the meshing frequency is determined as
and input shaft frequency is
During the test, the motor speed is
set to 1000rpm
Meshing frequency is determined
18
as fm 1000  
rpm   300Hz
60

1000  rpm
Inputf shaft   16.17Hz is
frequency
i
60

Fig. where we can see that the

spectrum contains the meshing
frequency component of 300Hz and
a sideband frequency component of
283.33 (300–16.67) Hz.
Fig. shows the vibration signature for the damaged pinion in Fig.
For the damaged pinion, the sidebands become dominant.
Hence, the vibration failure signature is identified
 DIGITAL IMAGE ENHANCEMENT
Example of signal processing
in two dimensions.
Fig. A 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 histogram equalization can stretch the
light intensity of an image using the digital information (pixels) to increase the
image contrast.
 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