In this paper, a system, developed for speech encoding, analysis, synthesis and gender identification is
presented. A typical gender recognition system can be divided into front-end system and back-end system.
The task of the front-end system is to extract the gender related information from a speech signal and
represents it by a set of vectors called feature. Features like power spectrum density, frequency at
maximum power carry speaker information. The feature is extracted using First Fourier Transform (FFT)
algorithm. The task of the back-end system (also called classifier) is to create a gender model to recognize
the gender from his/her speech signal in recognition phase. This paper also presents the digital processing
of a speech signals (pronounced “A” and “B”) which are taken from 10 persons, 5 of them are Male and
the rest of them are Female. Power Spectrum Estimation of the signal is examined .The frequency at
maximum power of the English Phonemes is extracted from the estimated power spectrum. The system uses
threshold technique as identification tool. The recognition accuracy of this system is 80% on average.
This method of checking the signal in the system for processing is called Polling Method. In this method, the problem is that the processor has to waste number of clock cycles just for checking the signal in the system, by this processor will become busy unnecessarily. If any signal came for the process, processor will take some time to process the signal due to the polling process in action. So system performance also will be degraded and response time of the system will also decrease.
This method of checking the signal in the system for processing is called Polling Method. In this method, the problem is that the processor has to waste number of clock cycles just for checking the signal in the system, by this processor will become busy unnecessarily. If any signal came for the process, processor will take some time to process the signal due to the polling process in action. So system performance also will be degraded and response time of the system will also decrease.
Semiconductor Memory Fundamentals
Memory Types
Memory Structure and its requirements
Memory Decoding
Examples
Input - Output Interfacing
Types of Parallel Data Transfer or I/O Techniques
The presentation covers clocked sequential circuit analysis and design process demonstrated with example. State reduction and state assignment is design is also described.
this ppt only for beginner who want to understand concept of Timer counter operation of LPC2148 step by step.
hope it may help u.
always welcoming ur suggestion.
a simple traffic lights controller and simulator implemented in VHDL
Complete project hosted at Github: https://github.com/abhishekjiitr/traffic-light-controller
The slide is about Non-deterministic Finite Automata and Deterministic Finite Automata. Which we created on our Theory of Computation course from Notre Dame University Bangladesh.
state space representation,State Space Model Controllability and Observabilit...Waqas Afzal
State Variables of a Dynamical System
State Variable Equation
Why State space approach
Block Diagram Representation Of State Space Model
Controllability and Observability
Derive Transfer Function from State Space Equation
Time Response and State Transition Matrix
Eigen Value
AN ANALYSIS OF SPEECH RECOGNITION PERFORMANCE BASED UPON NETWORK LAYERS AND T...IJCSEA Journal
Speech is the most natural way of information exchange. It provides an efficient means of means of manmachine communication using speech interfacing. Speech interfacing involves speech synthesis and speech recognition. Speech recognition allows a computer to identify the words that a person speaks to a microphone or telephone. The two main components, normally used in speech recognition, are signal processing component at front-end and pattern matching component at back-end. In this paper, a setup that uses Mel frequency cepstral coefficients at front-end and artificial neural networks at back-end has been developed to perform the experiments for analyzing the speech recognition performance. Various experiments have been performed by varying the number of layers and type of network transfer function, which helps in deciding the network architecture to be used for acoustic modelling at back end.
Semiconductor Memory Fundamentals
Memory Types
Memory Structure and its requirements
Memory Decoding
Examples
Input - Output Interfacing
Types of Parallel Data Transfer or I/O Techniques
The presentation covers clocked sequential circuit analysis and design process demonstrated with example. State reduction and state assignment is design is also described.
this ppt only for beginner who want to understand concept of Timer counter operation of LPC2148 step by step.
hope it may help u.
always welcoming ur suggestion.
a simple traffic lights controller and simulator implemented in VHDL
Complete project hosted at Github: https://github.com/abhishekjiitr/traffic-light-controller
The slide is about Non-deterministic Finite Automata and Deterministic Finite Automata. Which we created on our Theory of Computation course from Notre Dame University Bangladesh.
state space representation,State Space Model Controllability and Observabilit...Waqas Afzal
State Variables of a Dynamical System
State Variable Equation
Why State space approach
Block Diagram Representation Of State Space Model
Controllability and Observability
Derive Transfer Function from State Space Equation
Time Response and State Transition Matrix
Eigen Value
AN ANALYSIS OF SPEECH RECOGNITION PERFORMANCE BASED UPON NETWORK LAYERS AND T...IJCSEA Journal
Speech is the most natural way of information exchange. It provides an efficient means of means of manmachine communication using speech interfacing. Speech interfacing involves speech synthesis and speech recognition. Speech recognition allows a computer to identify the words that a person speaks to a microphone or telephone. The two main components, normally used in speech recognition, are signal processing component at front-end and pattern matching component at back-end. In this paper, a setup that uses Mel frequency cepstral coefficients at front-end and artificial neural networks at back-end has been developed to perform the experiments for analyzing the speech recognition performance. Various experiments have been performed by varying the number of layers and type of network transfer function, which helps in deciding the network architecture to be used for acoustic modelling at back end.
In this paper we present the implementation of speaker identification system using artificial neural network
with digital signal processing. The system is designed to work with the text-dependent speaker
identification for Bangla Speech. The utterances of speakers are recorded for specific Bangla words using
an audio wave recorder. The speech features are acquired by the digital signal processing technique. The
identification of speaker using frequency domain data is performed using backpropagation algorithm.
Hamming window and Blackman-Harris window are used to investigate better speaker identification
performance. Endpoint detection of speech is developed in order to achieve high accuracy of the system.
Emotion Recognition based on audio signal using GFCC Extraction and BPNN Clas...ijceronline
International Journal of Computational Engineering Research (IJCER) is dedicated to protecting personal information and will make every reasonable effort to handle collected information appropriately. All information collected, as well as related requests, will be handled as carefully and efficiently as possible in accordance with IJCER standards for integrity and objectivity.
Utterance Based Speaker Identification Using ANNIJCSEA Journal
In this paper we present the implementation of speaker identification system using artificial neural network with digital signal processing. The system is designed to work with the text-dependent speaker identification for Bangla Speech. The utterances of speakers are recorded for specific Bangla words using an audio wave recorder. The speech features are acquired by the digital signal processing technique. The identification of speaker using frequency domain data is performed using back propagation algorithm. Hamming window and Blackman-Harris window are used to investigate better speaker identification performance. Endpoint detection of speech is developed in order to achieve high accuracy of the system.
Utterance Based Speaker Identification Using ANNIJCSEA Journal
In this paper we present the implementation of speaker identification system using artificial neural network with digital signal processing. The system is designed to work with the text-dependent speaker identification for Bangla Speech. The utterances of speakers are recorded for specific Bangla words using an audio wave recorder. The speech features are acquired by the digital signal processing technique. The identification of speaker using frequency domain data is performed using backpropagation algorithm. Hamming window and Blackman-Harris window are used to investigate better speaker identification performance. Endpoint detection of speech is developed in order to achieve high accuracy of the system.
Classification of Language Speech Recognition Systemijtsrd
This paper is aimed to implement Classification of Language Speech Recognition System by using feature extraction and classification. It is an Automatic language Speech Recognition system. This system is a software architecture which outputs digits from the input speech signals. The system is emphasized on Speaker Dependent Isolated Word Recognition System. To implement this system, a good quality microphone is required to record the speech signals. This system contains two main modules feature extraction and feature matching. Feature extraction is the process of extracting a small amount of data from the voice signal that can later be used to represent each speech signal. Feature matching involves the actual procedure to identify the unknown speech signal by comparing extracted features from the voice input of a set of known speech signals and the decision making process. In this system, the Mel frequency Cepstrum Coefficient MFCC is used for feature extraction and Vector Quantization VQ which uses the LBG algorithm is used for feature matching. Khin May Yee | Moh Moh Khaing | Thu Zar Aung "Classification of Language Speech Recognition System" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-3 | Issue-5 , August 2019, URL: https://www.ijtsrd.com/papers/ijtsrd26546.pdfPaper URL: https://www.ijtsrd.com/computer-science/speech-recognition/26546/classification-of-language-speech-recognition-system/khin-may-yee
Voice recognition is the process of automatically recognizing who is speaking on the basis of individual information included in speech waves. This technique makes it possible to use the speaker's voice to verify their identity and control access to services such as voice dialing, banking by telephone, telephone shopping, database access services, information services, voice mail, security control for confidential information areas, and remote access to computers.
This document describes how to build a simple, yet complete and representative automatic speaker recognition system. Such a speaker recognition system has potential in many security applications. For example, users have to speak a PIN (Personal Identification Number) in order to gain access to the laboratory door, or users have to speak their credit card number over the telephone line to verify their identity. By checking the voice characteristics of the input utterance, using an automatic speaker recognition system similar to the one that we will describe, the system is able to add an extra level of security.
Wavelet Based Feature Extraction for the Indonesian CV Syllables SoundTELKOMNIKA JOURNAL
This paper proposes the combined methods of Wavelet Transform (WT) and Euclidean Distance
(ED) to estimate the expected value of the possibly feature vector of Indonesian syllables. This research
aims to find the best properties in effectiveness and efficiency on performing feature extraction of each
syllable sound to be applied in the speech recognition systems. This proposed approach which is the
state-of-the-art of the previous study consist of three main phase. In the first phase, the speech signal is
segmented and normalized. In the second phase, the signal is transformed into frequency domain by using
the WT. In the third phase, to estimate the expected feature vector, the ED algorithm is used. Th e result
shows the list of features of each syllables can be used for the next research, and some recommendations
on the most effective and efficient WT to be used in performing syllable sound recognition.
Speech Analysis and synthesis using VocoderIJTET Journal
Abstract— In this paper, I proposed a speech analysis and synthesis using a vocoder. Voice conversion systems do not create new speech signals, but just transform existing one. The proposed speech vocoding is different from speech coding. To analyze the speech signal and represent it with less number of bits, so that bandwidth efficiency can be increased. The Synthesis of speech signal from the received bits of information. In this paper three aspects of analysis have been discussed: pitch refinement, spectral envelope estimation and maximum voiced frequency estimation. A Quasi-harmonic analysis model can be used to implement a pitch refinement algorithm which improves the accuracy of the spectral estimation. Harmonic plus noise model to reconstruct the speech signal from parameter. Finally to achieve the highest possible resynthesis quality using the lowest possible number of bits to transmit the speech signal. Future work aims at incorporating the phase information into the analysis and modeling process and also synthesis these three aspects in different pitch period.
One of the common and easier techniques of feature extraction is Mel Frequency Cestrum Coefficient (MFCC) which allows the signals to extract the feature vector. It is used by Dynamic Feature Extraction and provide high performance rate when compared to previous technique like LPC. But one of the major drawbacks in this technique is robustness. Another feature extraction technique is Relative Spectral (RASTA). In effect the RASTA filter band passes each feature coefficient and in both the log spectral and the Spectral domains appear linear channel distortions as an additive constant. The high-pass portions of the equivalent band pass filter effect the convolution noise introduced in the channel. The low-pass filtering helps in smoothing frame to frame spectral changes. Compared to MFCC feature extraction technique, RASTA filtering reduces the impact of the noise in signals and provides high robustness
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
Speaker Recognition System using MFCC and Vector Quantization Approachijsrd.com
This paper presents an approach to speaker recognition using frequency spectral information with Mel frequency for the improvement of speech feature representation in a Vector Quantization codebook based recognition approach. The Mel frequency approach extracts the features of the speech signal to get the training and testing vectors. The VQ Codebook approach uses training vectors to form clusters and recognize accurately with the help of LBG algorithm.
Automatic speech emotion and speaker recognition based on hybrid gmm and ffbnnijcsa
In this paper we present text dependent speaker recognition with an enhancement of detecting the emotion
of the speaker prior using the hybrid FFBN and GMM methods. The emotional state of the speaker
influences recognition system. Mel-frequency Cepstral Coefficient (MFCC) feature set is used for
experimentation. To recognize the emotional state of a speaker Gaussian Mixture Model (GMM) is used in
training phase and in testing phase Feed Forward Back Propagation Neural Network (FFBNN). Speech
database consisting of 25 speakers recorded in five different emotional states: happy, angry, sad, surprise
and neutral is used for experimentation. The results reveal that the emotional state of the speaker shows a
significant impact on the accuracy of speaker recognition.
On the use of voice activity detection in speech emotion recognitionjournalBEEI
Emotion recognition through speech has many potential applications, however the challenge comes from achieving a high emotion recognition while using limited resources or interference such as noise. In this paper we have explored the possibility of improving speech emotion recognition by utilizing the voice activity detection (VAD) concept. The emotional voice data from the Berlin Emotion Database (EMO-DB) and a custom-made database LQ Audio Dataset are firstly preprocessed by VAD before feature extraction. The features are then passed to the deep neural network for classification. In this paper, we have chosen MFCC to be the sole determinant feature. From the results obtained using VAD and without, we have found that the VAD improved the recognition rate of 5 emotions (happy, angry, sad, fear, and neutral) by 3.7% when recognizing clean signals, while the effect of using VAD when training a network with both clean and noisy signals improved our previous results by 50%.
Audio compression has become one of the basic technologies of the multimedia age. The change in the telecommunication infrastructure, in recent years, from circuit switched to packet switched systems has also reflected on the way that speech and audio signals are carried in present systems. In many applications, such as the design of multimedia workstations and high quality audio transmission and storage, the goal is to achieve transparent coding of audio and speech signals at the lowest possible data rates. In other words, bandwidth cost money, therefore, the transmission and storage of information becomes costly. However, if we can use less data, both transmission and storage become cheaper. Further reduction in bit rate is an attractive proposition in applications like remote broadcast lines, studio links, satellite transmission of high quality audio and voice over internet.
Similar to GENDER RECOGNITION SYSTEM USING SPEECH SIGNAL (20)
ANALYSIS OF EXISTING TRAILERS’ CONTAINER LOCK SYSTEMS IJCSEIT Journal
Trailers carry large containers to various destinations in the world. These are manually locked on to
trailers as they move through these long distances. Security mainly refers to the safety of a state,
organization, property, and individuals against criminal activity. The study was made to analyze the
existing trailer locks and the insecurity being experienced currently. The study also focused on creating a
background to building an automated lock system for auto-mobiles. Findings showed that there are various
container types like the General Purpose containers, the Hard-Top containers and the Open-Top among
others. Similarly, the twist locks were the ones revised for this study. The study also discussed the
weaknesses of the twist locks, most especially the non-notification on unsecured locks. This causes leads to
accidents and wastage of lives and property. The study finally proposed an automated lock system to
overcome these weaknesses to some good extent.
A MODEL FOR REMOTE ACCESS AND PROTECTION OF SMARTPHONES USING SHORT MESSAGE S...IJCSEIT Journal
The smartphone usage among people is increasing rapidly. With the phenomenal growth of smartphone
use, smartphone theft is also increasing. This paper proposes a model to secure smartphones from theft as
well as provides options to access a smartphone through other smartphone or a normal mobile via Short
Message Service. This model provides option to track and secure the mobile by locking it. It also provides
facilities to receive the incoming call and sms information to the remotely connected device and enables the
remote user to control the mobile through SMS. The proposed model is validated by the prototype
implementation in Android platform. Various tests are conducted in the implementation and the results are
discussed.
BIOMETRIC APPLICATION OF INTELLIGENT AGENTS IN FAKE DOCUMENT DETECTION OF JOB...IJCSEIT Journal
The Job selection process in today’s globally competitive economy can be a daunting task for prospective
employees no matter their experience level. Although many years of research has been devoted to job
search and application resulting in good integration with information technology including the internet and
intelligent agent-based architectures, there are still many areas that need to be enhanced. Two such areas
include the quality of jobs associated with applicants in the job search by profiling the needs of employers
against the needs of prospective employees and the security and verifications schemes integrated to reduce
the instances of fraud and identity theft. The integration of mobile, intelligent agent, and cryptography
technologies provide benefits such as improved accessibility wirelessly, intelligent dynamic profiling, and
increased security. With this in mind we propose the intelligent mobile agents instead of human agents to
perform the Job search using fuzzy preferences which is been published elsewhere and application
operations incorporating the use of agents with a trust authority to establish employer trust and validate
applicant identity and accuracy. Our proposed system incorporates design methodologies to use JADELEAP
and Android to provide a robust, secure, user friendly solution.
FACE RECOGNITION USING DIFFERENT LOCAL FEATURES WITH DIFFERENT DISTANCE TECHN...IJCSEIT Journal
A face recognition system using different local features with different distance measures is proposed in this
paper. Proposed method is fast and gives accurate detection. Feature vector is based on Eigen values,
Eigen vectors, and diagonal vectors of sub images. Images are partitioned into sub images to detect local
features. Sub partitions are rearranged into vertically and horizontally matrices. Eigen values, Eigenvector
and diagonal vectors are computed for these matrices. Global feature vector is generated for face
recognition. Experiments are performed on benchmark face YALE database. Results indicate that the
proposed method gives better recognition performance in terms of average recognized rate and retrieval
time compared to the existing methods.
BIOMETRICS AUTHENTICATION TECHNIQUE FOR INTRUSION DETECTION SYSTEMS USING FIN...IJCSEIT Journal
Identifying attackers is a major apprehension to both organizations and governments. Recently, the most
used applications for prevention or detection of attacks are intrusion detection systems. Biometrics
technology is simply the measurement and use of the unique characteristics of living humans to distinguish
them from one another and it is more useful as compare to passwords and tokens as they can be lost or
stolen so we have choose the technique biometric authentication. The biometric authentication provides the
ability to require more instances of authentication in such a quick and easy manner that users are not
bothered by the additional requirements. In this paper, we have given a brief introduction about
biometrics. Then we have given the information regarding the intrusion detection system and finally we
have proposed a method which is based on fingerprint recognition which would allow us to detect more
efficiently any abuse of the computer system that is running.
PERFORMANCE ANALYSIS OF FINGERPRINTING EXTRACTION ALGORITHM IN VIDEO COPY DET...IJCSEIT Journal
A video fingerprint is a recognizer that is derived from a piece of video content. The video fingerprinting
methods obtain unique features of a video that differentiates one video clip from another. It aims to identify
whether a query video segment is a copy of video from the video database or not based on the signature of
the video. It is difficult to find whether a video is a copied video or a similar video, since the features of the
content are very similar from one video to the other. The main focus of this paper is to detect that the query
video is present in the video database with robustness depending on the content of video and also by fast
search of fingerprints. The Fingerprint Extraction Algorithm and Fast Search Algorithms are adopted in
this paper to achieve robust, fast, efficient and accurate video copy detection. As a first step, the
Fingerprint Extraction algorithm is employed which extracts a fingerprint through the features from the
image content of video. The images are represented as Temporally Informative Representative Images
(TIRI). Then, the second step is to find the presence of copy of a query video in a video database, in which
a close match of its fingerprint in the corresponding fingerprint database is searched using inverted-filebased
method. The proposed system is tested against various attacks like noise, brightness, contrast,
rotation and frame drop. Thus the performance of the proposed system on an average shows high true
positive rate of 98% and low false positive rate of 1.3% for different attacks.
Effect of Interleaved FEC Code on Wavelet Based MC-CDMA System with Alamouti ...IJCSEIT Journal
In this paper, the impact of Forward Error Correction (FEC) code namely Trellis code with interleaver on
the performance of wavelet based MC-CDMA wireless communication system with the implementation of
Alamouti antenna diversity scheme has been investigated in terms of Bit Error Rate (BER) as a function of
Signal-to-Noise Ratio (SNR) per bit. Simulation of the system under proposed study has been done in M-ary
modulation schemes (MPSK, MQAM and DPSK) over AWGN and Rayleigh fading channel incorporating
Walsh Hadamard code as orthogonal spreading code to discriminate the message signal for individual
user. It is observed via computer simulation that the performance of the interleaved coded based proposed
system outperforms than that of the uncoded system in all modulation schemes over Rayleigh fading
channel.
FUZZY WEIGHTED ASSOCIATIVE CLASSIFIER: A PREDICTIVE TECHNIQUE FOR HEALTH CARE...IJCSEIT Journal
In this paper we extend the problem of classification using Fuzzy Association Rule Mining and propose the
concept of Fuzzy Weighted Associative Classifier (FWAC). Classification based on Association rules is
considered to be effective and advantageous in many cases. Associative classifiers are especially fit to
applications where the model may assist the domain experts in their decisions. Weighted Associative
Classifiers that takes advantage of weighted Association Rule Mining is already being proposed. However,
there is a so-called "sharp boundary" problem in association rules mining with quantitative attribute
domains. This paper proposes a new Fuzzy Weighted Associative Classifier (FWAC) that generates
classification rules using Fuzzy Weighted Support and Confidence framework. The naïve approach can be
used to generating strong rules instead of weak irrelevant rules. where fuzzy logic is used in partitioning
the domains. The problem of Invalidation of Downward Closure property is solved and the concept of
Fuzzy Weighted Support and Fuzzy Weighted Confidence frame work for Boolean and quantitative item
with weighted setting is generalized. We propose a theoretical model to introduce new associative classifier
that takes advantage of Fuzzy Weighted Association rule mining.
DETECTION OF CONCEALED WEAPONS IN X-RAY IMAGES USING FUZZY K-NNIJCSEIT Journal
Scanning baggage by x-ray and analysing such images have become important technique for detecting
illicit materials in the baggage at Airports. In order to provide adequate security, a reliable and fast
screening technique is needed for baggage examination.This paper aims at providing an automatic method
for detecting concealed weapons, typically a gun in the baggage by employing image segmentation method
to extract the objects of interest from the image followed by applying feature extraction methods namely
Shape context descriptor and Zernike moments. Finally the objects are classified using fuzzy KNN as illicit
or non-illicit object.
META-HEURISTICS BASED ARF OPTIMIZATION FOR IMAGE RETRIEVALIJCSEIT Journal
The proposed approach avoids the semantic gap in image retrieval by combining automatic relevance
feedback and a modified stochastic algorithm. A visual feature database is constructed from the image
database, using combined feature vector. Very few fast-computable features are included in this step. The
user selects the query image, and based on that, the system ranks the whole dataset. The nearest images are
retrieved and the first automatic relevance feedback is generated. The combined similarity of textual and
visual feature space using Latent Semantic Indexing is evaluated and the images are labelled as relevant or
irrelevant. The feedback drives a feature re-weighting process and is routed to the particle swarm
optimizer. Instead of classical swarm update approach, the swarm is split, for each swarm to perform the
search in parallel, thereby increasing the performance of the system. It provides a powerful optimization
tool and an effective space exploration mechanism. The proposed approach aims to achieve the following
goals without any human interaction - to cluster relevant images using meta-heuristics and to dynamically
modify the feature space by feeding automatic relevance feedback.
ERROR PERFORMANCE ANALYSIS USING COOPERATIVE CONTENTION-BASED ROUTING IN WIRE...IJCSEIT Journal
In Wireless Ad hoc network, cooperation of nodes can be achieved by more interactions at higher protocol
layers, particularly the MAC (Medium Access Control) and network layers play vital role. MAC facilitates
a routing protocol based on position location of nodes at network layer specially known as Beacon-less
geographic routing (BLGR) using Contention-based selection process. This paper proposes two levels of
cross-layer framework -a MAC network cross-layer design for forwarder selection (or routing) and a
MAC-PHY for relay selection. Wireless networks suffers huge number of communication at the same time
leads to increase in collision and energy consumption; hence focused on new Contention access method
that uses a dynamical change of channel access probability which can reduce the number of contention
times and collisions. Simulation result demonstrates the best Relay selection and the comparative of direct
mode with the cooperative networks. And also demonstrates the Performance evaluation of contention
probability with Collision avoidance.
M-FISH KARYOTYPING - A NEW APPROACH BASED ON WATERSHED TRANSFORMIJCSEIT Journal
Karyotyping is a process in which chromosomes in a dividing cell are properly stained, identified and
displayed in a standard format, which helps geneticist to study and diagnose genetic factors behind various
genetic diseases and for studying cancer. M-FISH (Multiplex Fluorescent In-Situ Hybridization) provides
color karyotyping. In this paper, an automated method for M-FISH chromosome segmentation based on
watershed transform followed by naive Bayes classification of each region using the features, mean and
standard deviation, is presented. Also, a post processing step is added to re-classify the small chromosome
segments to the neighboring larger segment for reducing the chances of misclassification. The approach
provided improved accuracy when compared to the pixel-by-pixel approach. The approach was tested on
40 images from the dataset and achieved an accuracy of 84.21 %.
Steganography is the technique of hiding a confidential message in an ordinary message and the extraction
of that secret message at its destination. Different carrier file formats can be used in steganography.
Among these carrier file formats, digital images are the most popular. For this work, digital images are
used. Here steganography is done on the skin portion of an image. First skin portion of an image is
detected. Random pixels are selected from that detected region using a pseudo-random number generator.
The bits of the secret message will be embedded on the LSB of these random pixels. An analysis is done to
check the efficiency and robustness of the proposed method. The aim of this work is to show that
steganography done using random pixel selection is less prone to outside attacks.
A NOVEL WINDOW FUNCTION YIELDING SUPPRESSED MAINLOBE WIDTH AND MINIMUM SIDELO...IJCSEIT Journal
In many applications like FIR filters, FFT, signal processing and measurements, we are required (~45 dB)
or less side lobes amplitudes. However, the problem is usual window based FIR filter design lies in its side
lobes amplitudes that are higher than the requirement of application. We propose a window function,
which has better performance like narrower main lobe width, minimum side lobe peak compared to the
several commonly used windows. The proposed window has slightly larger main lobe width of the
commonly used Hamming window, while featuring 6.2~22.62 dB smaller side lobe peak. The proposed
window maintains its maximum side lobe peak about -58.4~-52.6 dB compared to -35.8~-38.8 dB of
Hamming window for M=10~14, while offering roughly equal main lobe width. Our simulated results also
show significant performance upgrading of the proposed window compared to the Kaiser, Gaussian, and
Lanczos windows. The proposed window also shows better performance than Dolph-Chebyshev window.
Finally, the example of designed low pass FIR filter confirms the efficiency of the proposed window.
CSHURI – Modified HURI algorithm for Customer Segmentation and Transaction Pr...IJCSEIT Journal
Association rule mining (ARM) is the process of generating rules based on the correlation between the set
of items that the customers purchase.Of late, data mining researchers have improved upon the quality of
association rule mining for business development by incorporating factors like value (utility), quantity of
items sold (weight) and profit. The rules mined without considering utility values (profit margin) will lead
to a probable loss of profitable rules.
The advantage of wealth of the customers’ needs information and rules aids the retailer in designing his
store layout[9]. An algorithm CSHURI, Customer Segmentation using HURI, is proposed, a modified
version of HURI [6], finds customers who purchase high profitable rare items and accordingly classify the
customers based on some criteria; for example, a retail business may need to identify valuable customers
who are major contributors to a company’s overall profit. For a potential customer arriving in the store,
which customer group one should belong to according to customer needs, what are the preferred functional
features or products that the customer focuses on and what kind of offers will satisfy the customer, etc.,
finds the key in targeting customers to improve sales [9], which forms the base for customer utility mining.
AN EFFICIENT IMPLEMENTATION OF TRACKING USING KALMAN FILTER FOR UNDERWATER RO...IJCSEIT Journal
The exploration of oceans and sea beds is being made increasingly possible through the development of
Autonomous Underwater Vehicles (AUVs). This is an activity that concerns the marine community and it
must confront the existence of notable challenges. However, an automatic detecting and tracking system is
the first and foremost element for an AUV or an aqueous surveillance network. In this paper a method of
Kalman filter was presented to solve the problems of objects track in sonar images. Region of object was
extracted by threshold segment and morphology process, and the features of invariant moment and area
were analysed. Results show that the method presented has the advantages of good robustness, high
accuracy and real-time characteristic, and it is efficient in underwater target track based on sonar images
and also suited for the purpose of Obstacle avoidance for the AUV to operate in the constrained
underwater environment.
USING DATA MINING TECHNIQUES FOR DIAGNOSIS AND PROGNOSIS OF CANCER DISEASEIJCSEIT Journal
Breast cancer is one of the leading cancers for women in developed countries including India. It is the
second most common cause of cancer death in women. The high incidence of breast cancer in women has
increased significantly in the last years. In this paper we have discussed various data mining approaches
that have been utilized for breast cancer diagnosis and prognosis. Breast Cancer Diagnosis is
distinguishing of benign from malignant breast lumps and Breast Cancer Prognosis predicts when Breast
Cancer is to recur in patients that have had their cancers excised. This study paper summarizes various
review and technical articles on breast cancer diagnosis and prognosis also we focus on current research
being carried out using the data mining techniques to enhance the breast cancer diagnosis and prognosis.
FACTORS AFFECTING ACCEPTANCE OF WEB-BASED TRAINING SYSTEM: USING EXTENDED UTA...IJCSEIT Journal
Advancement in information system leads organizations to apply e-learning system to train their employees
in order to enhance its performance. In this respect, applying web based training will enable the
organization to train their employees quickly, efficiently and effectively anywhere at any time. This
research aims to extend Unified Theory of Acceptance and Use Technology (UTAUT) using some factors
such flexibility of web based training system, system interactivity and system enjoyment, in order to explain
the employees’ intention to use web based training system. A total of 290 employees have participated in
this study. The findings of the study revealed that performance expectancy, facilitating conditions, social
influence and system flexibility have direct effect on the employees’ intention to use web based training
system, while effort expectancy, system enjoyment and system interactivity have indirect effect on
employees’ intention to use the system.
PROBABILISTIC INTERPRETATION OF COMPLEX FUZZY SETIJCSEIT Journal
The innovative concept of Complex Fuzzy Set is introduced. The objective of the this paper to investigate
the concept of Complex Fuzzy set in constraint to a traditional Fuzzy set , where the membership function
ranges from [0, 1], but in the Complex fuzzy set extended to a unit circle in a complex plane, where the
member ship function in the form of complex number. The Compressive study of mathematical operation of
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1. International Journal of Computer Science, Engineering and Information Technology (IJCSEIT), Vol.2, No.1, February 2012
DOI : 10.5121/ijcseit.2012.2101 1
GENDER RECOGNITION SYSTEM USING SPEECH
SIGNAL
Md. Sadek Ali1
, Md. Shariful Islam1
and Md. Alamgir Hossain1
1
Dept. of Information & Communication Engineering
Islamic University, Kushtia 7003, Bangladesh.
E-mail :{ sadek_ice, afmsi76, alamgir_ict}@yahoo.com
ABSTRACT
In this paper, a system, developed for speech encoding, analysis, synthesis and gender identification is
presented. A typical gender recognition system can be divided into front-end system and back-end system.
The task of the front-end system is to extract the gender related information from a speech signal and
represents it by a set of vectors called feature. Features like power spectrum density, frequency at
maximum power carry speaker information. The feature is extracted using First Fourier Transform (FFT)
algorithm. The task of the back-end system (also called classifier) is to create a gender model to recognize
the gender from his/her speech signal in recognition phase. This paper also presents the digital processing
of a speech signals (pronounced “A” and “B”) which are taken from 10 persons, 5 of them are Male and
the rest of them are Female. Power Spectrum Estimation of the signal is examined .The frequency at
maximum power of the English Phonemes is extracted from the estimated power spectrum. The system uses
threshold technique as identification tool. The recognition accuracy of this system is 80% on average.
KEYWORDS
Gender Recognition, Feature Extraction, First Fourier Transform (FFT), Font-end, Back-end.
1. INTRODUCTION
“Speech” according to Webster’s Dictionary is the “communication or expression of throughout
in speaker words”. Speech signal not only caries the information that is need to communicate
among people but also contents the information regarding the particular speaker. The
nonlinguistic characteristics of a speaker help to classify speaker (male or female). Features like
power spectrum density, frequency at maximum power carry speaker information. These speaker
features can be tracked well varying the frequency characteristics of the vocal tract and the
variation in the excitation. The speech signal also carry the information of the particular speaker
including social factors, affective factor and the properties of the physical voices production
apparatus for which human being are able to recognize whether the speaker is a male or a
female easily, during telephone conversation or any hidden condition of the speaker [1][2]. With
the current concern of security worldwide gender classification has received great deal of
attention among of the speech researchers. Also a rapidly developing environment of
computerization, one of the most important issues in the developing world is gender recognition.
Gender recognition, which can be classified into two different tasks: Gender identification and
Gender verification. In the identification task, or 1: N matching, an unknown speaker is compared
against a database of N known speakers, and the best matching speaker is returned as the
recognition decision. The verification task, or 1:1 matching, consists of making a decision
whether a given voice sample is produced by a claimed speaker. An identity claim (e.g., a PIN
2. International Journal of Computer Science, Engineering and Information Technology (IJCSEIT), Vol.2, No.1, February 2012
2
code) is given to the system, and the unknown speaker’s voice sample is compared against the
claimed speaker’s voice template. If the similarity degree between the voice sample and the
template exceeds a predefined decision threshold, the speaker is accepted, and otherwise rejected.
The rest of the paper is organized as follows. In section II shows the related works. Section III
describes the mathematical tools and techniques for gender recognition systems. Section IV
describes speech recording and feature extraction process. Computations of power and frequency
spectrum are described in section V. System implementation is detailed in section VI.
Recognition and experimental results are given in section VII. Finally, section VIII concludes the
paper.
2. RELATED WORKS
Gender recognition is a task of recognizing the gender from his or her voice. With the current
concern of security worldwide speaker identification has received great deal of attention among
of the speech researchers. Also a rapidly developing environment of computerization, one of the
most important issues in the developing world is speaker identification. Speech processing based
several types of research work have been continuing from a few decade ago as a field of digital
signal processing (DSP). The most efficient related work is “Speaker recognition in a multi-
speaker environment” was submitted in Proc. 7th European Conference on Speech
Communication and Technology (Eurospeech 2001) (Aalborg, Denmark, 2001), pp. 787–90[3].
Another related work is “Spectral Feature for Automatic Voice-independent Speaker
Recognition” was developed in the department of Computer Science, Joensuu University, Finland
2003 [4]. From the study of different previous research works it was observed that among the
different features the power spectrum results in best classification rate. Based on the power
spectrum, we have computed frequency spectrum from maximum power of speech signal. We
have implemented a complete gender recognition system to identify particular gender
(male/female) using frequency component. In addition to description, theoretical and
experimental analysis, we provide implementation details as well.
3. MATHEMATICAL TOOLS AND TECHNIQUES
For digital communication or digital signal synthesis, it is necessary to convey the analog signal
such as speech, music etc. as a sequence of digitized number, which is commonly done by
sampling the speech signal denoted by Xn (t) periodically to produce the sequence
X(n) = Xn (nt) α < n < α ……………………………………………...(1)
Where n0 have only integer value. In this paper, we have used pulse code modulation (PCM)
technique to digitize speech signal.
The sampled data are operated to find out the different parameter. Discrete Fourier Transform
(DFT) computes the frequency information of the equivalent time domain signal [5]. Since a
speech signal contains only real point values, we can make use of this fact and use a real-point
Fast Fourier Transform (FFT) for increased efficiency. The resulting output contains both the
magnitude and phase information of the original time domain signal. The Short time Fourier
analysis of windowed speech signal can produce a reasonable feature space for recognition [6].
The Fourier Transform for a discreet time signal f kT( ) is given by
3. International Journal of Computer Science, Engineering and Information Technology (IJCSEIT), Vol.2, No.1, February 2012
3
Fn f kT e j Nnk
k
N
( ) ( ) / .
= −
=
−
∑ 2
0
1
π
………………………………………………………………….. (2)
Equation (2) can be written as
F n f k WN
nk
k
N
( ) ( )= −
=
−
∑0
1
……………………………………………………………………… (3)
Where f k f kT( ) ( )= andW eN
j N
= 2π /
, WN is usually referred to as the kernel of the transform.
There are several algorithms that can considerably reduce the number of computations in a DFT.
DFT implemented using such schemes is referred to as Fast Fourier Transform (FFT). Among the
FFT algorithms there are two popular algorithms: decimation-in-time and decimation- in
frequency. Through out this paper, the DFT is computed using decimation-in-time algorithm.
4. SPEECH RECORDING AND FEATURE EXTRACTION
Human speech signal contains significant amount of energy within 2.5 KHz. So, we have taken
the sampling rate of speech signal as 8 KHz, 8 bit mono, which is sufficient for representing
signals up to 4 KHz without aliasing effect. To record speech signal we have used Intel(r)
Integrated sound card, a normal microphone and windows default sound recorder software.
Speech was recorded in room environment. The recorded sound was stored in PCM (.wav) sound
file format. The file header is stored at the beginning of the PCM file and occupied 44 bytes [7].
We also know that the actual wave data are stored after 58 bytes from the beginning. So, to
extract wave data, we first discard 58 bytes from the beginning of the wave file and then read
wave data as character. This data are stored in a text file (.txt) as integer data.
Feature extraction is the process of converting the original speech signal to a parametric
representation that gives a set of meaningful features useful for recognition. Feature extractions is
the combination of some signal processing steps including the computation of drive data from
wave sound, computation of Fast Fourier Transform (FFT), Power spectrum, the sample point at
maximum power and finally compute the frequency.
5. COMPUTATION OF POWER AND FREQUENCY
Power spectrum estimation uses an estimator called peridogram [5][6]. The power spectrum is
defined at N/2+1 frequency as
[ ] ( )P f N F F k Nk k N k( ) | | | | , ,.......,= + − − − − − − − − − = −−1 1 2 2 12 2 2
Where fk is defined only for the zero and positive frequencies
f k N f k N k Nk c≡ = − − − − − − − =∆ 2 0 1 2, ,....,
To compute the power spectrum of speech, the speech data is segmented into K segments of
N=2M points. Where N is the length of a window, taken as a power of 2 for the convenient
computation of FFT. Each segment is FFTed separately and the resulting K periodograms are
averaged together to obtain a Power Spectrum estimate at M frequency values between 0 and fc.
The figure 1 and 2 shows the signal waveform (fig1.a and fig2.a), power spectrum (fig1.b and
fig2.b) of male and female speaker for phoneme “A”.
4. International Journal of Computer Science, Engineering and Information Technology (IJCSEIT), Vol.2, No.1, February 2012
4
(a) (b)
Figure 1. The signal waveform and power spectrum of a male speaker for Phoneme “A”.
(a) (b)
Figure 2. The signal waveform and power spectrum of a female speaker for phoneme “A”.
0
0.01
0.02
0.03
0.04
0.05
0 200 400 600 800
Frequencyin Hz
PowerMagnitude
0
0.002
0.004
0.006
0.008
0.01
0 200 400 600 800
Frequency in Hz
PowerMagnitude
5. International Journal of Computer Science, Engineering and Information Technology (IJCSEIT), Vol.2, No.1, February 2012
5
The computation steps for frequency can be summarized as shown in figure 3.
Figure 3. The sequence of operations in converting a speech signal into Frequency features
6. SYSTEM IMPLEMENTATION
The following figure 4 shows the abstraction of a gender recognition system. Regardless of the
type of the task (classification or verification), gender recognition system operates in two modes:
training and recognition modes. In the training mode, a new gender person’s voice is recorded
and analysis. The recognition mode, an unknown gender person gives a speech input and the
system makes a decision about the speaker’s identity. Both the training and the recognition modes
include feature extraction, sometimes called the front-end of the system. The feature extractor
converts the digital speech signal into a sequence of numerical descriptors, called feature vectors.
The features provide a more stable, robust, and compact representation than the raw input signal.
Feature extraction can be considered as a data reduction process that attempts to capture the
essential characteristics of the speaker with a small data rate.
Speech Signal
Drive data from recorded
Wave sound
FFT
Log10. 2
Sample point at maximum
power spectrum
Power Spectrum
Frequency determination at max power
6. International Journal of Computer Science, Engineering and Information Technology (IJCSEIT), Vol.2, No.1, February 2012
6
Figure 4. Block diagram of the Gender Recognition System
The software system contains the following steps:
1. To code and record the speech data as a disk file
2. To compute the Fast Fourier Transform of the data from recorded file
3. To compute the power spectrum from transformer data of step 2.
4. To compute the frequency from power spectrum of step 3.
5. To identify the Speaker as a male person or as female person or give a message that indicate
the speaker is neither a male person nor a female person.
The developed system is applied to store and analyze English phonemes “A” and “B”. The speech
data are recorded for male and female speaker, FFT and power spectrum is calculated using these
data files by the developed system. The results are shown in tabular form in Table1:
Training Phase
Input
Speech
Feature
Extraction
Pattern
Matching
Decision LogicDecision
Recognition phase
Feature
Extraction
Store Extracted
Feature (data) in
Database.
Input
Speech
7. International Journal of Computer Science, Engineering and Information Technology (IJCSEIT), Vol.2, No.1, February 2012
7
Table 1. Frequency at maximum power of some male and female corresponding English
phonemes “A” and “B”.
Speaker
ID
Phonemes No.of
sample
Frequency at
maximum power
(Male voice)
Frequency at
maximum power
(Female voice)
1 A 1 570 672
2 A 1 588 720
3 A 1 498 672
4 A 1 462 729
5 A 1 432 687
7. SYSTEM RESULTS
7.1. RECOGNITION RESULTS
The detailed recognition results are shown in Table 2.
Table 2. For sample #1
No. of
speaker
No. of accuracy
of Gender
Recognize
Percentage (%)
1 Male 100
2 Male 100
3 Male 100
4 Male 100
5 Male 100
6 Female 100
7 Female 100
8 Female 100
9 Female 100
10 Female 100
1 B 2 432 618
2 B 2 342 402
3 B 2 543 666
4 B 2 366 612
5 B 2 510 414
8. International Journal of Computer Science, Engineering and Information Technology (IJCSEIT), Vol.2, No.1, February 2012
8
Table 2. For sample #2
No. of
speaker
No. of accuracy
of Gender
Recognize
Percentage (%)
1 Male 100
2 Male 0
3 Male 100
4 Male 0
5 Male 100
6 Female 100
7 Female 0
8 Female 100
9 Female 100
10 Female 0
7.2. EXPERIMENTAL RESULTS
The system was tested with 10 speakers (5 male people and 5 female people).The speech
(utterance) of word (“A” and “B”) of 10 speakers was recorded separately. The threshold
technique was used in recognition. The objective of the experiment was to study the recognition
performance of the system with different speakers. The percentage of accuracy rate of recognition
has been calculated using following equations:
No. of accurately recognized gender
Recognition Accuracy (%) = X 100...................... (4)
No. of correct gender expected
From recognition result, number of accurately recognized gender 16, number of correct gender
expected 20.The recognition accuracy is = 16*100/20 = 80%.
8. CONCLUSION
In this paper the main goal was to develop a gender recognition system using speech signal. The
feature selection is one of the most important factors in designing a gender recognition system.
From the study of different previous research works it was observed that among the different
features the power spectrum results in best classification rate. Thus the power spectrum has been
selected as the feature for classification. Among the different technique, the statistical analysis
and threshold technique is simple in computation and produces very good results. This is why this
method was selected for pattern comparison in recognition process to obtain improved
performance. The average recognition accuracy is 80 %. From experimental result, it can be seen
that recognition rate decreases as the number of speaker increases. Therefore, the system
efficiency is decreases, if the number of reference speakers in the speaker database increases. For
recognition constant thresholds have been used. If we could use dynamic threshold for
recognition it might produce more accurate and better recognition results.
9. International Journal of Computer Science, Engineering and Information Technology (IJCSEIT), Vol.2, No.1, February 2012
9
REFERENCES
[1]. Prabhakar, S., Pankanti, S., and Jain, A. “Biometric recognition: security and privacy concerns”
IEEE Security and Privacy Magazine 1(2003), 33-42.
[2]. Huang X., Acero, A., and Hon, H.-W. “Spoken Language Processing: a Guideto Theory, Algorithm
and System Development” prentice-Hall, New Jersey, 2001.
[3]. Martin, A., and Przybocki, M. Speaker recognition in a multi-speaker environment.In Proc. 7th
European Conference on Speech Communication and Technology (Eurospeech 2001) (Aalborg,
Denmark, 2001), pp. 787–90.
[4]. Tomi Kinnunen “ Spectral Feature for Automatic Voice-independent Speaker Recognition”
Depertment of Computer Science, Joensuu University,Finland. December 21, 2003.
[5]. John R. Deller, John G Proakis and John H. L. Hansen, “Discrete- Time Processing of Speech
Signals” Macmillan Publishing company, 866 Third avenue, New York 10022.
[6]. Rabiner Lawrence, Juang Bing-Hwang, “Fundamentals of Speech Recognitions”, Prentice Hall New
Jersey, 1993, ISBN 0-13-015157-2.
[7]. Md. Saidur Rahman, “Small Vocabulary Speech Recognition in Bangla Language”, M.Sc. Thesis,
Dept. of Computer Science & Engineering, Islamic University, Kushtia-7003, July-2004.
Biography
Md. Sadek Ali received the Bachelor’s and Master‘s Degree in the Department of
Information and Communication Engineering (ICE) from Islamic University, Kushtia,
in 2004 and 2005 respectively. He is currently a Lecturer in the department of ICE,
Islamic University, Kushtia-Bangladesh. Since 2003, he has been working as a
Research Scientist at the Signal and Communication Research Laboratory,
Department of ICE, Islamic University, Kushtia, where he belongs to the spread-
spectrum research group. He has three published paper in international and one
national journal in the same areas. He has also two published paper in international
and one national conference in the same areas. His areas of interest include Signal processing, Wireless
Communications, optical fiber communication, Spread Spectrum and mobile communication.
Md. Shariful Islam received the Bachelor’s and Master‘s Degree in Applied Physics,
Electronics & Communication Engineering from Islamic University, Kushtia,
Bangladesh in 1999, and 2000 respectively. He is currently Assistant Professor in the
department of ICE, Islamic University, Kushtia-7003 Bangladesh. He has five
published papers in international and national journals. His areas of interest include
signal processing & mobile communication.
Md. Alamgir Hossain received the Bachelor’s and Master‘s Degree in the Dept. of
Information and Communication Engineering (ICE) from Islamic University, Kushtia,
in 2003 and 2004, respectively. He is currently Lecturer in the department of ICE,
Islamic University, Kushtia-7003, and Bangladesh. He was a lecturer in the Dept. of
Computer Science & Engineering from Institute of Science Trade & Technology
(Under National University), Dhaka, Bangladesh. from 23th October, 2007 to 18th
April 2010. Since 2003, he has been working as a Research Scientist at the
Communication Reasearch Laboratory, Department of ICE, Islamic University, Kushtia, where he belongs
to the spread-spectrum research group. He has two published paper in international and one national journal
in the same areas. His current research interests include Wireless Communications, Spread Spectrum and
mobile communication, Signal processing, Data Ming.