Synopsis on

Performance Analysis of Combined Wavelet

Transform and Artificial Neural Network for
Isolated Marathi Digit Recognition
By

Atul Dattatraya Narkhede

Under the Supervision of

Dr. Milind Nemade

Faculty of Engineering
PACIFIC ACADEMY OF HIGHER EDUCATION AND
RESEARCH UNIVERSITY, UDAIPUR.
1

SUMMARY

Abstract

Introduction

Review of Literature

Research Gaps

Scope of Research

Research Objectives

Hypothesis

Tools & Techniques

Research Plan

Tentative Chapter Flow

References

ABSTRACT

Speech processing is useful for various applications such as

mobile applications, healthcare, automatic translation,
robotics, video games, transcription, audio and video
database search, household applications, language learning
applications etc.

A speech recognition system has two major components,

namely, feature extraction and classification.

There are two dominant approaches of acoustic

measurement. First is a temporal domain or parametric
approach and second approach is nonparametric frequency
domain approach.

The objective of our research is to investigate the

combined performance of wavelet transform and artificial
neural network (ANN) for isolated Marathi digits so as to
improve accuracy of speech recognition system.

We proposed in to derive effective, efficient, and noise

robust features from the frequency subbands of the frame.
Each frame of speech signal is decomposed into different
frequency subbands using discrete wavelet transform
(DWT) and each subband is further classified using (ANN).

The effective speech features and classification would

improve quality of the speech, represent speech signal in
terms of frequency and bandwidth and improves speech
recognition.
4

INTRODUCTION

The performance of a speech processing system is

usually measured in terms of recognition accuracy.

All speech recognizers include an initial signal

processing front end that converts a speech signal into
its more convenient and compressed form called
feature vectors.

Feature extraction method plays a vital role in speech

recognition task.

The wavelet transform, with its flexible time-frequency

window, is an appropriate tool for the analysis of non
stationary signals like speech.

In speech signal, high frequencies are present very

briefly at the onset of a sound while lower frequencies
are present latter for long period.

DWT resolves all these frequencies well.

The DWT parameters contain the information of

different frequency scales.

This helps in getting the speech information of

corresponding frequency band.

Artificial Neural Network (ANN) is an efficient pattern

recognition mechanism which simulates the neural
information processing of human brain.

The computational intelligence of neural networks is

made up of their processing units, characteristics and
ability to learn.

In learning the system parameters of NN vary over

time and are characterized by their ability of local and
parallel computation, simplicity and regularity.

A wavelet transform is an elegant tool for the analysis

of non-stationary signals like speech.

The results have shown that this hybrid architecture

using Discrete wavelet transforms and Neural
networks could effectively extract the features from
the speech signal for automatic speech recognition.

REVIEW OF LITERATURE
Remarkable Observations in the review of work are as
follows

Speech features which are usually obtained via Fourier

transforms (FTs), Short time Fourier transforms (STFTs) or
Linear Predictive Coding techniques are used for some kind
of Automatic speech/speaker recognition (ASR). They may
not be suitable for representing speech/voice.

In spite of the improvement on the computation time for

FFT, the recognition time of the proposed systems is still
too long to be used for real-time applications.

Multi-core and parallel processing for the speech

recognition algorithm are necessary to further improve the
recognition time and is worthwhile to examine in the
research.

REVIEW OF LITERATURE
Remarkable Observations in the review of work are as
follows

The conventional approaches like Mel frequency cepstrum

coefficients (MFCC) and linear predictive coefficients (LPC)
focus on spectral features limited to lower frequency bands.

Best recognition was found from the DWT decomposition

when compared to the MFCCs for speaker independent and
speaker dependent tasks respectively.

Wavelet transform approaches provided good results in

clean, noisy and reverberant environments and also has a
much lower computational complexity.

REVIEW OF LITERATURE
Remarkable Observations in the review of work are as
follows

Wavelet decomposition results in a logarithmic set of

bandwidths, which is very similar to the response of human ear
to frequencies.

Wavelet transform efficiently locates the spectral changes in

speech signal as well as beginning and end of the sounds can
also be located.

Results show that hybrid architecture using discrete wavelet

transforms and neural networks could effectively extract the
features from the speech signal for automatic speech
recognition.

10

REVIEW OF LITERATURE
Remarkable Observations in the review of work are
as follows

Artificial neural network performance depends on the

size and quality of training samples.

The simplification of the ANN architecture without

reducing the recognition rate can also speed up the
recognition time.

Improving the recognition accuracy of the system by

combining the multiple classifiers.

11

RESEARCH GAPS

Feature extraction and classification are major components,

plays vital role in speech recognition systems. So efficient
representation of speech features & its classification is required
for speech recognition systems.

To improve accuracy of speech recognition system, we can use

hybrid architecture consist of Wavelet Transform (WT) and
Artificial Neural Network (ANN).

ANN architecture
recognition rate.

Isolated Marathi Digit should be recognised speedily by speeding

up the recognition time.

can be simplified without reducing the

12

SCOPE OF RESEARCH

The scope of our research is limited to investigate the

combined performance of Wavelet Transform (WT) and
Artificial Neural Network (ANN) for feature extraction
and classification for isolated Marathi digits.

TOOLS & TECHNIQUES

Matlab / Simulink Programming Language

13

RESEARCH OBJECTIVES
The objective of our research is to investigate the combined
performance of Wavelet Transform (WT) and Artificial Neural
Network (ANN) for Isolated Marathi Digits so as to improve
accuracy of speech recognition system.

To derive effective, efficient, and noise robust features

from the frequency sub bands of the frame using discrete
wavelet transform.

Each frame of speech signal is decomposed into different

frequency sub bands using discrete wavelet transform.

Classification of each sub band using artificial neural

network (ANN).

Determination of accuracy of speech recognition system.

14

ISOLATED
DIGIT
RECOGNITI
ON

15

RESEARCH
METHODOLOGY

16

Hypothesis
The objective of our research is to investigate the
combined performance of wavelet transform and artificial
neural network (ANN) for isolated Marathi digits so as to
improve accuracy of speech recognition system.

Tentative Chapter Flow

1.

Introduction to Speech Processing, Wavelet Transform

& ANN

2.

Speech Feature Extraction using Wavelet Transform

(WT)

3.

Speech Feature Classification using Artificial Neural

Network (ANN)

4.

Performance analysis of Speech Feature Extraction

and Classification Techniques

17

RESEARCH PLAN
Activity

Phas Phas Phas Phas Phas Phas

e
e II e III e IV
eV
e VI
I

Literature Survey
Study of Software Tools like
MATLAB/SIMULINK, Neural
Network Toolbox and its
MATLAB link
Survey of Existing Methods and
Algorithms
Suggesting techniques for
removing limitations in existing
algorithms
Simulation of combined
strategies
Comparing results of developed
strategies with existing
algorithms
Performance evaluation and
implementation
Documentation
Review & Research Paper
Preparation &
Presentation/Publication

18

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Thank you

25

Training phase accepts speech samples from different people and trains the
system to create acoustic models for each word in vocabulaey.TP undergoes
through two stages Data preparation & Recording data.
Verification Phase display some random numbers then check for pronouns
number.
Some time system consists of speech processing inclusive of digit boundary
and recognition which uses zero crossing and energy techniques. Mel
Frequency Cepstral Coefficients (MFCC) vectors are used to provide an
estimate of the vocal tract filter. Meanwhile dynamic time warping (DTW) is
used to detect the nearest recorded voice.
The general methodology of audio classification involves extracting
discriminatory features from the audio data and feeding them to a pattern
classifier. Different approaches and various kinds of audio features were
proposed with varying success rates. The features can be extracted either
directly from the time domain signal or from a transformation domain
depending upon the choice of the signal analysis approach. Some of the audio
features that have been successfully used for audio classification include Mel
Frequency Cepstral Coefficients (MFCC).

26

MFCCs are commonly derived as follows:

1.Take the Fourier Transform of (a windowed
excerpt of) a signal.
2.Map the powers of the spectrum obtained
above onto the Mel Scale, using triangular
overlapping windows.
3.Take the logs of the powers at each of the Mel
frequencies.
4.Take the discrete cosine transform of the list
of Mel log powers, as if it were a signal.
5.The MFCCs are the amplitudes of the resulting
spectrum.

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