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DSP Implementation of Voice Recognition Using Dynamic Time Warping

Algorithm
Fariha Muzaffar, Bushra Mohsin, Farah Naz
Lecturer Farooq Jawed
Department Of Electronics Engineering, NED University of Engineering & Technology, Karachi

Abstract Voice recognition system performs two fundamental

Voice recognition is a process of an automatic system to
perceive speech. The paper discusses voice recognition
using Cepstral Analysis and DTW of a set of five words.
The software model was designed using DSP block
library in Simulink. The developed model provide us
with the necessary tools to record, filter, and analyze
different voice samples and compare them with the
archived sample.
The paper concludes with discussion about the
implementation of the speech recognition algorithm on a
DSP processor. The algorithm was implemented on the
target DSP via Embedded target for TI DSP toolbox and
real time Workshop (RTW).
Keywords: DTW, Cepstrum analysis
operations: signal modeling and pattern matching. Signal
modeling represents process of converting speech signal
into a set of parameters. Pattern matching is the task of
finding parameter set from memory which closely
matches the parameter set obtained from the input speech
signal. We have designed a very basic Speaker-
dependent Voice recognition system that identifies
isolated spoken words using a limited vocabulary of five
words. In this report we briefly discuss the signal
modeling approach for speech recognition. It is followed
by overview of basic operations involved in signal
modeling. Further commonly used temporal and spectral
analysis techniques of feature extraction are discussed in
detail.

2. VOICE PRODUCTION
1. INTRODUCTION We speak using pulmonic aggressive airstream (i.e. when
we breathe out). This airstreams sets the vocal folds in
Voice recognition is the ability of machines to respond to motion, producing voicing. Moving the active
spoken commands. Among the earliest applications for articulators (tongue, lips, mandible, uvula, and posterior
speech recognition were automated telephone systems pharyngeal wall) against the non-movable structures or
and medical dictation software (Transcription). Voice passive articulators (teeth, palate) changes the shape of
recognition is being used today by thousands of people the supralaryngeal vocal tract and modifies the sound
everyday. Such systems as calling cards and phone being produced. The result in sounds that are very similar
banking services use speech recognition by prompting to other sounds found in nature but which humans
the user to answer questions in voice rather than pressing perceive as speech because other humans articulate them.
digits. Voice Recognition Information Systems have
become so advanced and mainstream that business and
health care professionals are turning to Voice recognition
solutions for everything from providing telephone
support to writing medical reports. Technological
advances have made voice recognition software and
devices more functional and user friendly, with most
contemporary products performing tasks with over 90
percent accuracy.
Voice Recognition systems can be classified into two
categories speaker-dependent and speaker-independent.
Speaker-dependent systems works by comparing a whole
word input with a user-supplied pattern. These patterns
are developed by user during the training exercise.
Accuracy rates are typically less than 90%. Fig.1 Graphical Model of Human Speech Production
Speaker–independent systems require no training through Vocal Tract
sessions.
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3. ANNOTATION
Some fundamental ideas about the articulatory
production of numbers from 0 to 9, call & end & also
major classification into which their sounds are divided
One:
Rounded half close back vowel: There are three possible
resonators involved in the articulation of a vowel: the
oral cavity, the labial cavity, and the nasal cavity. If the
lips are pushed forward and rounded, a third, labial
resonator is formed. Rounded vowels means labial
resonator active and back vowels means tongue body in
the post-palatal or velar region
Two:
Voiceless dental or alveolar stop: The tongue makes
contact with the alveolar ridge directly above front teeth.
Two comes in the class of the aspirated word which
means if feel a breath of air as you say the word.
Three:
Voiceless dental or alveolar stop: The tongue makes
contact with the front teeth. Three is also an aspirant.
Four & Five:
Voiceless, labiodentals, fricative. The lower lip is
brought close to the upper teeth, occasionally even
grazing the teeth with its outer surface, or with its inner
surface, imparting in this case a slight hushing sound
4. VOICE ANALYSIS
A speech analysis is done after taking an input through
microphone from a user. The design of the system
involves manipulation of the input audio signal. At
different levels, different operations are performed on the
input signal such as Pre-emphasis, Normalizing,
Windowing, Cepstrum analysis and Recognition of the
spoken word.
Fig. 2 Block diagram of speech recognitions system
4.1 Pre-Emphasis
The speech signal obtained from the microphone is
digitized and is put through a filter, to spectrally flatten
the signal. After that the signal is run through a pre-
emphasis network. The pre-emphasizer can be either
fixed or slowly adaptive. For Pre-emphasis a fixed first
order system is recommended. The most widely used
pre-emphasis network is
Where,
Let x (k) be the digitized filtered input signal. The output
is then related to the input signal by the following
equation:
.
The value for the variable ‘a’ is usually chosen to be
around 0.95. This means that about 95% of any one
sample is presumed to originate from the previous
sample.
4.2 Normalization
After pre-emphasis, each word has it's energy
normalized. Based on the energy distribution along the
temporal axis, it is computed the center of gravity, and
this information is used as reference for temporal
alignment of the words. In practice normalization is done
so power in signal after windowing is approximately
equal to the power of signal before windowing.
4.3 Windowing
When we analyze signals we tend to do so on only small
portions at a time. The reason for this is that we typically
assume that the signal is constant over the time-span of
our analysis. Since speech is actually changing rapidly,
we have to cut it into small parts for this assumption to
hold. Hence we windowed our speech signal before
analysis. Windowing can be seen as multiplying a signal
by a window which is zero everywhere except for the
region of interest, where it is one. Since we pretend that
our signals are infinite, we can discard all of the resulting
zeros and concentrate on just the windowed portion of
the signal. A rectangular window is called so because of
its shape. One problem with this kind of window is the
abrupt changes at the edge, which can cause distortion in
the signal being. To reduce this distortion we used a
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smoother window shape called a Hamming window. This
window is zero at the edges and rises gradually to be 1 in
the middle. When this window is used the edges of the
signal are de-emphasized and the edge effects are
reduced.
Each individual frame is windowed to minimize the
signal discontinuities at the borders of each frame. If the
window is defined as w[n], 0 < n < N-1, then the
windowed signal is
Where, 0 < n < N-1.
4.4 Cepstral Analysis
The source in voiced speech is the vibration of the vocal
folds in response to airflow from the lungs. In unvoiced
speech the sound source is not a regular vibration but
rather vibrations are caused by turbulent airflow due to a
constriction in the vocal tract. The filter in speech
production is the vocal tract As with any other filter, this
tube has a characteristic spectrum. In a real speech
spectrum the overall filter shape and the location of the
formants is often drowned out by the effects of the source
spectrum. Therefore, we removed the source effects by
taking the Cepstrum of the signal and then studied the
two spectra separately to achieve a more accurate picture
of the precursors of the speech sound.
Cepstral Analysis technique is very useful as it provides
methodology for separating the excitation from the vocal
tract shape. In the linear acoustic model of speech
production, the composite speech spectrum, consist of
excitation signal filtered by a time-varying linear filter
representing the vocal tract shape as shown in figure
Fig.3. Vocal tract filter
Hence in log domain the excitation and the vocal tract
shape are superimposed, and can be separated. Cepstrum
is computed by taking inverse discrete Fourier transform
(IDFT) of logarithm of magnitude of discrete Fourier
transform finite length input signal. In log domain the
excitation and the vocal tract shape are superimposed,
and can be separated.
Fig.4. Block diagram of Cepstrum Analyzer
4.5 Recognition block
Several techniques can be used for the recognition of
speech. The recognition technique we have used is that
of Dynamic Time Warping.
4.5.1 Dynamic Time warping
This technique allows for the incoming word of speech
and the template to be of different sizes. The size
difference is due to variations in the length of words in
normal spoken speech. These variations cause the frames
to be out of alignment. Every word or phrase is stored in
its own separate template. Speech input must be stored in
these templates before recognition is to take place. DTW
operates by selecting which frames of the reference
template best match each frame of the input such that the
resulting error between them is minimized. Therefore the
speaker must pre-program the recognizer with his or her
own speech. By allowing multiple frames of one to be
matched against a single repeated frame of the other,
DTW can compress or expand relative time.
The difference between a new & a stored sample is called
a Euclidean distance & is given by:
Min [ di = (x – Zi)T(x – Zi) ]
Dynamic programming is an approach to implicit storage
of all possible solutions to a problem requiring
minimization of global error. Applied to template
matching for speech recognition, imagine a matrix ‘D’ in
which the rows correspond to frames of reference
template & the column corresponds to an input template.
For each matrix element in ‘D’ a cumulative distortion
measure is given by:
D(i,j) = d(i,j) + min{D[p(i.j) + T[(i,j), p(I,j)]}
Where,
‘d’ is the local distance measure between frame ‘i’ of
input & frame ‘j’ of reference stored template
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P(i,j) is the coordinates of the possible previous points on
the matching trajectory between the two templates
T( ) is a term for cost associated with any particular
transition
5. SOFTWARE & HARDWARE
PLATFORM
5.1 Software approach:
The following tools were used for the design &
implementation of speech recognition algorithm
5.1.1 MATLAB.
The name MATLAB stands for matrix laboratory.
MATLAB is a high-performance language for technical
computing. It integrates computation, visualization, and
programming in an easy-to-use environment where
problems and solutions are expressed in familiar
mathematical notation. MATLAB has extensive facilities
for displaying vectors and matrices as graphs, as well as
annotating and printing these graphs. It includes high-
level functions for two-dimensional and three-
dimensional data visualization, image processing,
animation, and presentation graphics.
5.1.2 Simulink.
Simulink is a software package that enables you to
model, simulate, and analyze systems whose outputs
change over time. Such systems are often referred to as
dynamic systems. Simulink can be used to explore the
behavior of a wide range of real-world dynamic systems.
The DSP Blockset brings the full power of Simulink to
DSP system design and prototyping by providing key
DSP algorithms and components in the adaptable block
format of Simulink.
5.1.3 Data Acquisition Toolbox
The Data Acquisition Toolbox is a collection of M-file
functions and MEX-file dynamic link libraries (DLLs)
built on the MATLAB® technical computing
environment. The toolbox provides a framework for
bringing live, measured data into MATLAB using PC-
compatible, plug-in data acquisition hardware. It
provides support for analog input (AI), analog output
(AO), and digital I/O (DIO) subsystems including
simultaneous analog I/O conversions. It also supports
popular hardware vendors/device such as the Windows
sound cards.
5.1.3a Softscope
Softscope is the the Data Acquisition oscilloscope. It is
an interactive graphical user interface (GUI) for
streaming data into a display. When the Softscope is
opened hardware, math, and reference channels are
displayed. With the Scale Channel option data can be
scaled horizontally and vertically according to the
requirement. The Triggering option helps to control the
initialization of data acquisition. Using the Export Data
option we can save channel data or measurements to the
workspace, a figure, or a MAT-file.
5.1.4 Embedded target for TIC6000 DSP
Embedded Target for TI C6000 DSP use Simulink to
model digital signal processing algorithms from blocks in
the DSP Blockset, and then use Real-Time Workshop to
generate C code targeted to Texas Instruments Code
Composer Studio Integrated Development Environment
(CCS IDE). The Embedded Target for TI C6000 DSP
takes the generated C code and uses Texas Instruments
tools to build specific machine code depending on the TI
board you use. The build process downloads the targeted
machine code to the selected hardware and runs the
executable on the digital signal processor. After
downloading the code to the board, the digital signal
processing application runs automatically on the target.
When using this target, the build process creates a new
project in Code Composer Studio and populates the
project with the required files.
The following are blocks of this toolbox, ADC, DAC,
LED, Switch.
5.2 Hardware Approach
Digital Signal Processors such as the TMS320C6x family
of processors are like fast special purpose
microprocessors with a specialized type of architecture &
instruction set appropriate for signal processing. The
architecture of C6x digital signal processor is very well
suited for numerical intensive calculations, based on
VLIW architecture.
5.2.1 C6x DSP’s Architecture.
The C67x processor consists of three main parts: CPU,
peripherals, and memory. The CPU consists of eight
functional units, which operate in parallel. It is divided
into two sides A & B. Each side has a so called
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M unit (Used for multiplication operation) coefficients are obtained. Finally, using the Euclidean
L unit (used for logical & arithmetic operations) distance formula the difference between the words is
S unit (used for branch, bit manipulation, & arithmetic calculated. The single word of testing phase is compared
operations) with the five words which were stored during the training
D unit (used for loading, storing & arithmetic operations) phase and five Euclidean distances are obtained in a
similar manner. The Euclidean distances are then
1 8V 16M 128K Daughter compared amongst themselves to determine the smallest
Card I/F value, which is then displayed as the recognized word.

Par

TMS320 C
Po
D.
Po C
Us

Re Three
3 3V 16-bit codec
Emula Line Level Input
Line Level
Fig. 5 DSK TI c6711

5.2.2 HILS (Hardware-in-the-loop-simulation)

The purpose of the HILS is to test an embedded

system. The proof that the embedded system passed Fig 6 Simulink simulation model of voice recognizer
its test is that its outputs were correct for the inputs
that it was given. HILS can be applied very
effectively in other areas too. For instance, machine 6.2 Embedded Model
control and motion control are two areas where it's
hard to completely test the software before the
expensive, fragile, and often unique machine is built.
HILS proves to be quite useful in testing the
software in such cases. Some limitations of HILS are
that it cannot easily stop-if you pause the hardware-
in-the-loop simulator, all the components that it's
attached to, including the embedded program in your
system-under-test, keep running. Furthermore, A
HILS cannot tell what's going on inside the Fig. 7 Embedded model of voice recognizer
embedded software. It can only read the embedded
system's outputs The above figure shows an embedded model. This voice
recognition model is embedded on the Target DSP
c6711. Results from hardware are obtained using
hardware-in-the-loop-simulation (HILS). The DSP with
6. SIMULINK MODELS
the program embedded on it is plugged into the
backplane of the PC. The PC runs the user interface and
6.1 Simulation Model data logging code and the DSP runs your simulation and
the analog I/O. The input of the DSP is connected to the
The figure shows the simulation model of voice “line out” of the DSP and the output of the DSP is
recognizer. Input words, for training and testing, are connected to the “line in” of the DSP. During the training
taken through the softscope present in Data Acquisition phase wav files of the designed vocabulary are played on
toolbox. These words are saved in Matlab workspace as the PC. The DSP takes it as the input and calculates its
arrays. During simulation these words from the Cepstral Co-efficient. These co-efficient are fed into the
workspace are taken as input. Different operations are line in of the PC and are acquired with the help of
performed on them at different stages and their cepstral
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softscope present in the Data Acquisition toolbox of

Matlab. Similarly, another wav file is played during the
testing phase and its Cepstral co-efficient are obtained in
the same manner. These co-efficient are saved in the
Matlab workspace. In order to recognize the word spoken
during the testing phase, the formula of Euclidean
distance is applied on these co-efficient and they are fed
into the decision making block

7 RESULTS:

Fig.11 waveform of the digit “one” after Windowing

Fig. 8 waveform of the digit “one”

Fig.12 waveform of the digit “one” after Cepstrum

Fig. 9 waveform of the digit “one” after pre-emphasis

Fig.12 waveform after calculating Euclidean distance

8. TESTING:
For the tests we used a training set consisting of 15
occurrences of the digit “1” by 3 speakers (i.e., 5
occurrences per speaker). All the speakers were female.
A range was set and it was found that the error rate was
less than 30% (more than 70% correct classifications).
Fig. 10 waveform of the digit “one” after overlap Table 8.1 and 8.2 displays the error and efficiency for
analysis every speaker.
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DIGITS SPEAKER 1 SPEAKER SPEAKER

2 3
ONE 50 % 75% 75%
TWO 75 % 75% 75%
THREE 75 % 75% 75%
FOUR 75 % 75% 50%
FIVE 75 % 50% 50%
Table 8.1

TOTAL EFFICIENCY
SPEAKER 1 70 %
SPEAKER 2 70%
SPEAKER 3 65%
SYSTEM 68.33%
Table 8.2

9. CONCLUSION
We have discussed the implementation of speech
recognition algorithms using Simulink rather than using
C codes. Working with C is tedious and time consuming
and have used Cepstrum Analysis and Dynamic time
Warping which give 68% accurate results.

References.
[1] Ben Gold, Nelson Morgan, Speech and Audio Signal
processing processing

[2] Ralph Chassaing, DSP Applications using C and the

TMS320C6x DSK 2002

[3] Texas Instruments, Implementation of an HMM-

Based, Speaker-Independent Speech Recognition System
on the TMS320C2x and TMS320C5x, 1996

[4] Texas Instruments, Automated Dialing of Cellular

Telephones Using Speech Recognition, 1994

[5] Texas Instruments, TMS320C6000 Technical Brief

SPRU197D February 2000

[6] Texas Instruments,TMS320C6201/6701 Evaluation

module user’s guide 1998

[7] Mitra, Digital Signal Processing using MATLAB