This paper describe the morphing concept in which we convert the voice of any person into pre -analyzed or pre-recorded voice of any animals.As the user generate a pre-established voice, his pitch, timbre, vibrato and articulation can be modified to resemble those of a pre-recorded and pre-analyzed voice of animal. This technique is based on SMS. Thus using this concept we can develop many funny application and we can used this type of application in mobile device, personal computer etc. for enjoying the sometime of period.
ANALYSIS OF SPEECH UNDER STRESS USING LINEAR TECHNIQUES AND NON-LINEAR TECHNI...cscpconf
Analysis of speech for recognition of stress is important for identification of emotional state of
person. This can be done using ‘Linear Techniques’, which has different parameters like pitch,
vocal tract spectrum, formant frequencies, Duration, MFCC etc. which are used for extraction
of features from speech. TEO-CB-Auto-Env is the method which is non-linear method of
features extraction. Analysis is done using TU-Berlin (Technical University of Berlin) German
database. Here emotion recognition is done for different emotions like neutral, happy, disgust,
sad, boredom and anger. Emotion recognition is used in lie detector, database access systems, and in military for recognition of soldiers’ emotion identification during the war.
EFFECT OF DYNAMIC TIME WARPING ON ALIGNMENT OF PHRASES AND PHONEMESkevig
Speech synthesis and recognition are the basic techniques used for man-machine communication. This type
of communication is valuable when our hands and eyes are busy in some other task such as driving a
vehicle, performing surgery, or firing weapons at the enemy. Dynamic time warping (DTW) is mostly used
for aligning two given multidimensional sequences. It finds an optimal match between the given sequences.
The distance between the aligned sequences should be relatively lesser as compared to unaligned
sequences. The improvement in the alignment may be estimated from the corresponding distances. This
technique has applications in speech recognition, speech synthesis, and speaker transformation. The
objective of this research is to investigate the amount of improvement in the alignment corresponding to the
sentence based and phoneme based manually aligned phrases. The speech signals in the form of twenty five
phrases were recorded from each of six speakers (3 males and 3 females). The recorded material was
segmented manually and aligned at sentence and phoneme level. The aligned sentences of different speaker
pairs were analyzed using HNM and the HNM parameters were further aligned at frame level using DTW.
Mahalanobis distances were computed for each pair of sentences. The investigations have shown more than
20 % reduction in the average Mahalanobis distances.
Feature Extraction Analysis for Hidden Markov Models in Sundanese Speech Reco...TELKOMNIKA JOURNAL
Sundanese language is one of the popular languages in Indonesia. Thus, research in Sundanese language becomes essential to be made. It is the reason this study was being made. The vital parts to get the high accuracy of recognition are feature extraction and classifier. The important goal of this study was to analyze the first one. Three types of feature extraction tested were Linear Predictive Coding (LPC), Mel Frequency Cepstral Coefficients (MFCC), and Human Factor Cepstral Coefficients (HFCC). The results of the three feature extraction became the input of the classifier. The study applied Hidden Markov Models as its classifier. However, before the classification was done, we need to do the quantization. In this study, it was based on clustering. Each result was compared against the number of clusters and hidden states used. The dataset came from four people who spoke digits from zero to nine as much as 60 times to do this experiments. Finally, it showed that all feature extraction produced the same performance for the corpus used.
A New Method for Pitch Tracking and Voicing Decision Based on Spectral Multi-...CSCJournals
This paper proposes a new voicing detection and pitch estimation method that is particularly robust for noisy speech. This method is based on the spectral analysis of the speech multi-scale product. The multi-scale product (MP) consists of making the product of wavelet transform coefficients. The wavelet used is the quadratic spline function. We argue that the spectral of Multi-scale Product Analysis is capable of revealing an estimate of a pitch-harmonic more accurately even in a heavy noisy scenario. We evaluate our approach on the Keele database. The experimental results show the robustness of our method for noisy speech, and the good performance for clean speech in comparison with state-of-the-art algorithms.
Broad phoneme classification using signal based featuresijsc
Speech is the most efficient and popular means of human communication Speech is produced as a sequence
of phonemes. Phoneme recognition is the first step performed by automatic speech recognition system. The
state-of-the-art recognizers use mel-frequency cepstral coefficients (MFCC) features derived through short
time analysis, for which the recognition accuracy is limited. Instead of this, here broad phoneme
classification is achieved using features derived directly from the speech at the signal level itself. Broad
phoneme classes include vowels, nasals, fricatives, stops, approximants and silence. The features identified
useful for broad phoneme classification are voiced/unvoiced decision, zero crossing rate (ZCR), short time
energy, most dominant frequency, energy in most dominant frequency, spectral flatness measure and first
three formants. Features derived from short time frames of training speech are used to train a multilayer
feedforward neural network based classifier with manually marked class label as output and classification
accuracy is then tested. Later this broad phoneme classifier is used for broad syllable structure prediction
which is useful for applications such as automatic speech recognition and automatic language
identification.
ANALYSIS OF SPEECH UNDER STRESS USING LINEAR TECHNIQUES AND NON-LINEAR TECHNI...cscpconf
Analysis of speech for recognition of stress is important for identification of emotional state of
person. This can be done using ‘Linear Techniques’, which has different parameters like pitch,
vocal tract spectrum, formant frequencies, Duration, MFCC etc. which are used for extraction
of features from speech. TEO-CB-Auto-Env is the method which is non-linear method of
features extraction. Analysis is done using TU-Berlin (Technical University of Berlin) German
database. Here emotion recognition is done for different emotions like neutral, happy, disgust,
sad, boredom and anger. Emotion recognition is used in lie detector, database access systems, and in military for recognition of soldiers’ emotion identification during the war.
EFFECT OF DYNAMIC TIME WARPING ON ALIGNMENT OF PHRASES AND PHONEMESkevig
Speech synthesis and recognition are the basic techniques used for man-machine communication. This type
of communication is valuable when our hands and eyes are busy in some other task such as driving a
vehicle, performing surgery, or firing weapons at the enemy. Dynamic time warping (DTW) is mostly used
for aligning two given multidimensional sequences. It finds an optimal match between the given sequences.
The distance between the aligned sequences should be relatively lesser as compared to unaligned
sequences. The improvement in the alignment may be estimated from the corresponding distances. This
technique has applications in speech recognition, speech synthesis, and speaker transformation. The
objective of this research is to investigate the amount of improvement in the alignment corresponding to the
sentence based and phoneme based manually aligned phrases. The speech signals in the form of twenty five
phrases were recorded from each of six speakers (3 males and 3 females). The recorded material was
segmented manually and aligned at sentence and phoneme level. The aligned sentences of different speaker
pairs were analyzed using HNM and the HNM parameters were further aligned at frame level using DTW.
Mahalanobis distances were computed for each pair of sentences. The investigations have shown more than
20 % reduction in the average Mahalanobis distances.
Feature Extraction Analysis for Hidden Markov Models in Sundanese Speech Reco...TELKOMNIKA JOURNAL
Sundanese language is one of the popular languages in Indonesia. Thus, research in Sundanese language becomes essential to be made. It is the reason this study was being made. The vital parts to get the high accuracy of recognition are feature extraction and classifier. The important goal of this study was to analyze the first one. Three types of feature extraction tested were Linear Predictive Coding (LPC), Mel Frequency Cepstral Coefficients (MFCC), and Human Factor Cepstral Coefficients (HFCC). The results of the three feature extraction became the input of the classifier. The study applied Hidden Markov Models as its classifier. However, before the classification was done, we need to do the quantization. In this study, it was based on clustering. Each result was compared against the number of clusters and hidden states used. The dataset came from four people who spoke digits from zero to nine as much as 60 times to do this experiments. Finally, it showed that all feature extraction produced the same performance for the corpus used.
A New Method for Pitch Tracking and Voicing Decision Based on Spectral Multi-...CSCJournals
This paper proposes a new voicing detection and pitch estimation method that is particularly robust for noisy speech. This method is based on the spectral analysis of the speech multi-scale product. The multi-scale product (MP) consists of making the product of wavelet transform coefficients. The wavelet used is the quadratic spline function. We argue that the spectral of Multi-scale Product Analysis is capable of revealing an estimate of a pitch-harmonic more accurately even in a heavy noisy scenario. We evaluate our approach on the Keele database. The experimental results show the robustness of our method for noisy speech, and the good performance for clean speech in comparison with state-of-the-art algorithms.
Broad phoneme classification using signal based featuresijsc
Speech is the most efficient and popular means of human communication Speech is produced as a sequence
of phonemes. Phoneme recognition is the first step performed by automatic speech recognition system. The
state-of-the-art recognizers use mel-frequency cepstral coefficients (MFCC) features derived through short
time analysis, for which the recognition accuracy is limited. Instead of this, here broad phoneme
classification is achieved using features derived directly from the speech at the signal level itself. Broad
phoneme classes include vowels, nasals, fricatives, stops, approximants and silence. The features identified
useful for broad phoneme classification are voiced/unvoiced decision, zero crossing rate (ZCR), short time
energy, most dominant frequency, energy in most dominant frequency, spectral flatness measure and first
three formants. Features derived from short time frames of training speech are used to train a multilayer
feedforward neural network based classifier with manually marked class label as output and classification
accuracy is then tested. Later this broad phoneme classifier is used for broad syllable structure prediction
which is useful for applications such as automatic speech recognition and automatic language
identification.
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.
Speech Emotion Recognition is a recent research topic in the Human Computer Interaction (HCI) field. The need has risen for a more natural communication interface between humans and computer, as computers have become an integral part of our lives. A lot of work currently going on to improve the interaction between humans and computers. To achieve this goal, a computer would have to be able to distinguish its present situation and respond differently depending on that observation. Part of this process involves understanding a user‟s emotional state. To make the human computer interaction more natural, the objective is that computer should be able to recognize emotional states in the same as human does. The efficiency of emotion recognition system depends on type of features extracted and classifier used for detection of emotions. The proposed system aims at identification of basic emotional states such as anger, joy, neutral and sadness from human speech. While classifying different emotions, features like MFCC (Mel Frequency Cepstral Coefficient) and Energy is used. In this paper, Standard Emotional Database i.e. English Database is used which gives the satisfactory detection of emotions than recorded samples of emotions. This methodology describes and compares the performances of Learning Vector Quantization Neural Network (LVQ NN), Multiclass Support Vector Machine (SVM) and their combination for emotion recognition.
Wavelet Based Noise Robust Features for Speaker RecognitionCSCJournals
Extraction and selection of the best parametric representation of acoustic signal is the most important task in designing any speaker recognition system. A wide range of possibilities exists for parametrically representing the speech signal such as Linear Prediction Coding (LPC) ,Mel frequency Cepstrum coefficients (MFCC) and others. MFCC are currently the most popular choice for any speaker recognition system, though one of the shortcomings of MFCC is that the signal is assumed to be stationary within the given time frame and is therefore unable to analyze the non-stationary signal. Therefore it is not suitable for noisy speech signals. To overcome this problem several researchers used different types of AM-FM modulation/demodulation techniques for extracting features from speech signal. In some approaches it is proposed to use the wavelet filterbanks for extracting the features. In this paper a technique for extracting the features by combining the above mentioned approaches is proposed. Features are extracted from the envelope of the signal and then passed through wavelet filterbank. It is found that the proposed method outperforms the existing feature extraction techniques.
Comparison of Feature Extraction MFCC and LPC in Automatic Speech Recognition...TELKOMNIKA JOURNAL
Speech recognition can be defined as the process of converting voice signals into the ranks of the
word, by applying a specific algorithm that is implemented in a computer program. The research of speech
recognition in Indonesia is relatively limited. This paper has studied methods of feature extraction which is
the best among the Linear Predictive Coding (LPC) and Mel Frequency Cepstral Coefficients (MFCC) for
speech recognition in Indonesian language. This is important because the method can produce a high
accuracy for a particular language does not necessarily produce the same accuracy for other languages,
considering every language has different characteristics. Thus this research hopefully can help further
accelerate the use of automatic speech recognition for Indonesian language. There are two main
processes in speech recognition, feature extraction and recognition. The method used for comparison
feature extraction in this study is the LPC and MFCC, while the method of recognition using Hidden
Markov Model (HMM). The test results showed that the MFCC method is better than LPC in Indonesian
language speech recognition.
An Introduction to Various Features of Speech SignalSpeech featuresSivaranjan Goswami
An overview of various temporal, spectral and cepstral features of speech signal used in digital speech processing.
For more tutorials visit:
https://sites.google.com/site/enggprojectece
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.
Isolated words recognition using mfcc, lpc and neural networkeSAT Journals
Abstract Automatic speech recognition is an important topic of speech processing. This paper presents the use of an Artificial Neural Network (ANN) for isolated word recognition. The Pre-processing is done and voiced speech is detected based on energy and zero crossing rates (ZCR). The proposed approach used in speech recognition is Mel Frequency Cepstral Coefficients (MFCC) and combine features of both MFCC and Linear Predictive Coding (LPC). The back-propagation is used as a classifier. The recognition accuracy is increased when combine features of both LPC and MFCC are used as compared to only MFCC approach using Neural Network as a classifier.. Keywords: Pre-processing, Mel frequency Cepstral Coefficient (MFCC), Linear Predictive Coding (LPC), Artificial Neural Network (ANN).
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.
Can You Qualify for Low Income Housing Tax Credits: A Guide for Land Developr...Curley & Rothman, LLC
Low income housing tax credits (LIHTCs) were created by the Tax Reform Act in 1986. LIHTCs provide a dollar-for-dollar tax credit program to reduce the federal taxes of a developer or investor in real estate. Learn more about low income housing tax credits in this presentation.
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.
Speech Emotion Recognition is a recent research topic in the Human Computer Interaction (HCI) field. The need has risen for a more natural communication interface between humans and computer, as computers have become an integral part of our lives. A lot of work currently going on to improve the interaction between humans and computers. To achieve this goal, a computer would have to be able to distinguish its present situation and respond differently depending on that observation. Part of this process involves understanding a user‟s emotional state. To make the human computer interaction more natural, the objective is that computer should be able to recognize emotional states in the same as human does. The efficiency of emotion recognition system depends on type of features extracted and classifier used for detection of emotions. The proposed system aims at identification of basic emotional states such as anger, joy, neutral and sadness from human speech. While classifying different emotions, features like MFCC (Mel Frequency Cepstral Coefficient) and Energy is used. In this paper, Standard Emotional Database i.e. English Database is used which gives the satisfactory detection of emotions than recorded samples of emotions. This methodology describes and compares the performances of Learning Vector Quantization Neural Network (LVQ NN), Multiclass Support Vector Machine (SVM) and their combination for emotion recognition.
Wavelet Based Noise Robust Features for Speaker RecognitionCSCJournals
Extraction and selection of the best parametric representation of acoustic signal is the most important task in designing any speaker recognition system. A wide range of possibilities exists for parametrically representing the speech signal such as Linear Prediction Coding (LPC) ,Mel frequency Cepstrum coefficients (MFCC) and others. MFCC are currently the most popular choice for any speaker recognition system, though one of the shortcomings of MFCC is that the signal is assumed to be stationary within the given time frame and is therefore unable to analyze the non-stationary signal. Therefore it is not suitable for noisy speech signals. To overcome this problem several researchers used different types of AM-FM modulation/demodulation techniques for extracting features from speech signal. In some approaches it is proposed to use the wavelet filterbanks for extracting the features. In this paper a technique for extracting the features by combining the above mentioned approaches is proposed. Features are extracted from the envelope of the signal and then passed through wavelet filterbank. It is found that the proposed method outperforms the existing feature extraction techniques.
Comparison of Feature Extraction MFCC and LPC in Automatic Speech Recognition...TELKOMNIKA JOURNAL
Speech recognition can be defined as the process of converting voice signals into the ranks of the
word, by applying a specific algorithm that is implemented in a computer program. The research of speech
recognition in Indonesia is relatively limited. This paper has studied methods of feature extraction which is
the best among the Linear Predictive Coding (LPC) and Mel Frequency Cepstral Coefficients (MFCC) for
speech recognition in Indonesian language. This is important because the method can produce a high
accuracy for a particular language does not necessarily produce the same accuracy for other languages,
considering every language has different characteristics. Thus this research hopefully can help further
accelerate the use of automatic speech recognition for Indonesian language. There are two main
processes in speech recognition, feature extraction and recognition. The method used for comparison
feature extraction in this study is the LPC and MFCC, while the method of recognition using Hidden
Markov Model (HMM). The test results showed that the MFCC method is better than LPC in Indonesian
language speech recognition.
An Introduction to Various Features of Speech SignalSpeech featuresSivaranjan Goswami
An overview of various temporal, spectral and cepstral features of speech signal used in digital speech processing.
For more tutorials visit:
https://sites.google.com/site/enggprojectece
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.
Isolated words recognition using mfcc, lpc and neural networkeSAT Journals
Abstract Automatic speech recognition is an important topic of speech processing. This paper presents the use of an Artificial Neural Network (ANN) for isolated word recognition. The Pre-processing is done and voiced speech is detected based on energy and zero crossing rates (ZCR). The proposed approach used in speech recognition is Mel Frequency Cepstral Coefficients (MFCC) and combine features of both MFCC and Linear Predictive Coding (LPC). The back-propagation is used as a classifier. The recognition accuracy is increased when combine features of both LPC and MFCC are used as compared to only MFCC approach using Neural Network as a classifier.. Keywords: Pre-processing, Mel frequency Cepstral Coefficient (MFCC), Linear Predictive Coding (LPC), Artificial Neural Network (ANN).
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.
Can You Qualify for Low Income Housing Tax Credits: A Guide for Land Developr...Curley & Rothman, LLC
Low income housing tax credits (LIHTCs) were created by the Tax Reform Act in 1986. LIHTCs provide a dollar-for-dollar tax credit program to reduce the federal taxes of a developer or investor in real estate. Learn more about low income housing tax credits in this presentation.
Before undertaking a project or buying land, commercial real estate developers need to do their due diligence to protect their interests. Sellers have only limited disclosure requirements, so commercial developers must ensure the land is fit for their desired purpose. Learn more in this presentation.
Aspectos relevantes de la cultura china como el significado de la bandera, escudo, himno, modeda y billetes, orígenes y dinastias, arquitectura, arte, muralla china, aportes de la ciencia y la literatura china.
Intrusion Detection Systems (IDSs) have become widely recognized as powerful tools for identifying, deterring and deflecting malicious attacks over the network. Intrusion detection systems (IDSs) are designed and installed to aid in deterring or mitigating the damage that can be caused by hacking, or breaking into sensitive IT systems. . The attacks can come from outsider attackers on the Internet, authorized insiders who misuse the privileges that have been given them and unauthorized insiders who attempt to gain unauthorized privileges. IDSs cannot be used in isolation, but must be part of a larger framework of IT security measures. Essential to almost every intrusion detection system is the ability to search through packets and identify content that matches known attacks. Space and time efficient string matching algorithms are therefore important for identifying these packets at line rate. In this paper we examine string matching algorithm and their use for Intrusion Detection. Keywords: System Design, Network Algorithm
Sommer, Sonne, Zärtlichkeit: In den schönsten Wochen des Jahres tun sich Männer und Frauen mit der Kontaktaufnahme oft deutlich leichter. Besonders in der locker-entspannten Atmosphäre einer Urlaubsreise sind viele gerne schon mal zu einem kleinen Flirt bereit.
Einer aktuellen Studie im Auftrag der L’TUR Tourismus AG zufolge hat sich jeder dritte Deutsche (31 Prozent) schon einmal im Urlaub verliebt
Introduction
As the Jubilee administration nears the half-way mark in its term, Ipsos’ most recent survey included questions on a number of issues that are continuing to occupy public, and government, attention. In this Media Release, we cover the following in the Education sector: (1) The laptop project, (2) School-ranking, (3) Teachers’ Pay Demands.
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.
Audio/Speech Signal Analysis for Depressionijsrd.com
The word “depressed†is a common everyday word. People might say "I am depressed" when in fact they mean "I am fed up because I have had a row, or failed an exam, or lost my job", etc. These ups and downs of life are common and normal. Most people recover quite quickly. Depression is identified by different methods. Here we are identified depression by MFCC (Mel Frequency Ceptral Coefficient) method. There are different parameters used for the identification of depressed speech and normal speech, but MFCCs based parameter is the most applicable information then other parameter because depressive speech or audio signal can contain more information in the higher energy bands when compared with normal speech.
Broad Phoneme Classification Using Signal Based Features ijsc
Speech is the most efficient and popular means of human communication Speech is produced as a sequence of phonemes. Phoneme recognition is the first step performed by automatic speech recognition system. The state-of-the-art recognizers use mel-frequency cepstral coefficients (MFCC) features derived through short time analysis, for which the recognition accuracy is limited. Instead of this, here broad phoneme classification is achieved using features derived directly from the speech at the signal level itself. Broad phoneme classes include vowels, nasals, fricatives, stops, approximants and silence. The features identified useful for broad phoneme classification are voiced/unvoiced decision, zero crossing rate (ZCR), short time energy, most dominant frequency, energy in most dominant frequency, spectral flatness measure and first three formants. Features derived from short time frames of training speech are used to train a multilayer feedforward neural network based classifier with manually marked class label as output and classification accuracy is then tested. Later this broad phoneme classifier is used for broad syllable structure prediction which is useful for applications such as automatic speech recognition and automatic language identification.
The primary goal of this paper is to provide an overview of existing Text-To-Speech (TTS) Techniques by highlighting its usage and advantage. First Generation Techniques includes Formant Synthesis and Articulatory Synthesis. Formant Synthesis works by using individually controllable formant filters, which can be set to produce accurate estimations of the vocal-track transfer function. Articulatory Synthesis produces speech by direct modeling of Human articulator behavior. Second Generation Techniques incorporates Concatenative synthesis and Sinusoidal synthesis. Concatenative synthesis generates speech output by concatenating the segments of recorded speech. Generally, Concatenative synthesis generates the natural sounding synthesized speech. Sinusoidal Synthesis use a harmonic model and decompose each frame into a set of harmonics of an estimated fundamental frequency. The model parameters are the amplitudes and periods of the harmonics. With these, the value of the fundamental can be changed while keeping the same basic spectral..In adding, Third Generation includes Hidden Markov Model (HMM) and Unit Selection Synthesis.HMM trains the parameter module and produce high quality Speech. Finally, Unit Selection operates by selecting the best sequence of units from a large speech database which matches the specification.
Effect of Dynamic Time Warping on Alignment of Phrases and Phonemeskevig
Speech synthesis and recognition are the basic techniques used for man-machine communication. This type
of communication is valuable when our hands and eyes are busy in some other task such as driving a
vehicle, performing surgery, or firing weapons at the enemy. Dynamic time warping (DTW) is mostly used
for aligning two given multidimensional sequences. It finds an optimal match between the given sequences.
The distance between the aligned sequences should be relatively lesser as compared to unaligned
sequences. The improvement in the alignment may be estimated from the corresponding distances. This
technique has applications in speech recognition, speech synthesis, and speaker transformation. The
objective of this research is to investigate the amount of improvement in the alignment corresponding to the
sentence based and phoneme based manually aligned phrases. The speech signals in the form of twenty five
phrases were recorded from each of six speakers (3 males and 3 females). The recorded material was
segmented manually and aligned at sentence and phoneme level. The aligned sentences of different speaker
pairs were analyzed using HNM and the HNM parameters were further aligned at frame level using DTW.
Mahalanobis distances were computed for each pair of sentences. The investigations have shown more than
20 % reduction in the average Mahalanobis distances.
A Comparative Study: Gammachirp Wavelets and Auditory Filter Using Prosodic F...CSCJournals
Modern automatic speech recognition (ASR) systems typically use a bank of linear filters as the first step in performing frequency analysis of speech. On the other hand, the cochlea, which is responsible for frequency analysis in the human auditory system, is known to have a compressive non-linear frequency response which depends on input stimulus level. It will be shown in this paper that it presents a new method on the use of the gammachirp auditory filter based on a continuous wavelet analysis. The essential characteristic of this model is that it proposes an analysis by wavelet packet transformation on the frequency bands that come closer the critical bands of the ear that differs from the existing model based on an analysis by a short term Fourier transformation (STFT). The prosodic features such as pitch, formant frequency, jitter and shimmer are extracted from the fundamental frequency contour and added to baseline spectral features, specifically, Mel Frequency Cepstral Coefficients (MFCC) for human speech, Gammachirp Filterbank Cepstral Coefficient (GFCC) and Gammachirp Wavelet Frequency Cepstral Coefficient (GWFCC). The results show that the gammachirp wavelet gives results that are comparable to ones obtained by MFCC and GFCC. Experimental results show the best performance of this architecture. This paper implements the GW and examines its application to a specific example of speech. Implications for noise robust speech analysis are also discussed within AURORA databases.
Effect of MFCC Based Features for Speech Signal Alignmentskevig
The fundamental techniques used for man-machine communication include Speech synthesis, speech
recognition, and speech transformation. Feature extraction techniques provide a compressed
representation of the speech signals. The HNM analyses and synthesis provides high quality speech with
less number of parameters. Dynamic time warping is well known technique used for aligning two given
multidimensional sequences. It locates an optimal match between the given sequences. The improvement in
the alignment is estimated from the corresponding distances. The objective of this research is to investigate
the effect of dynamic time warping on phrases, words, and phonemes based alignments. The speech signals
in the form of twenty five phrases were recorded. The recorded material was segmented manually and
aligned at sentence, word, and phoneme level. The Mahalanobis distance (MD) was computed between the
aligned frames. The investigation has shown better alignment in case of HNM parametric domain. It has
been seen that effective speech alignment can be carried out even at phrase level
EFFECT OF MFCC BASED FEATURES FOR SPEECH SIGNAL ALIGNMENTSijnlc
The fundamental techniques used for man-machine communication include Speech synthesis, speech
recognition, and speech transformation. Feature extraction techniques provide a compressed
representation of the speech signals. The HNM analyses and synthesis provides high quality speech with
less number of parameters. Dynamic time warping is well known technique used for aligning two given
multidimensional sequences. It locates an optimal match between the given sequences. The improvement in
the alignment is estimated from the corresponding distances. The objective of this research is to investigate
the effect of dynamic time warping on phrases, words, and phonemes based alignments. The speech signals
in the form of twenty five phrases were recorded. The recorded material was segmented manually and
aligned at sentence, word, and phoneme level. The Mahalanobis distance (MD) was computed between the
aligned frames. The investigation has shown better alignment in case of HNM parametric domain. It has
been seen that effective speech alignment can be carried out even at phrase level.
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.
This paper proposes a voice morphing system for people suffering from Laryngectomy, which is the surgical removal of all or part of the larynx or the voice box, particularly performed in cases of laryngeal cancer. A primitive method of achieving voice morphing is by extracting the source's vocal coefficients and then converting them into the target speaker's vocal parameters. In this paper, we deploy Gaussian Mixture Models (GMM) for mapping the coefficients from source to destination. However, the use of the traditional/conventional GMM-based mapping approach results in the problem of over-smoothening of the converted voice. Thus, we hereby propose a unique method to perform efficient voice morphing and conversion based on GMM, which overcomes the traditional-method effects of over-smoothening. It uses a technique of glottal waveform separation and prediction of excitations and hence the result shows that not only over-smoothening is eliminated but also the transformed vocal tract parameters match with the target. Moreover, the synthesized speech thus obtained is found to be of a sufficiently high quality. Thus, voice morphing based on a unique GMM approach has been proposed and also critically evaluated based on various subjective and objective evaluation parameters. Further, an application of voice morphing for Laryngectomees which deploys this unique approach has been recommended by this paper
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.
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.
Emotion Recognition Based on Speech Signals by Combining Empirical Mode Decom...BIJIAM Journal
This paper proposes a novel method for speech emotion recognition. Empirical mode decomposition (EMD) is applied in this paper for the extraction of emotional features from speeches, and a deep neural network (DNN) is used to classify speech emotions. This paper enhances the emotional components in speech signals by using EMD with acoustic feature Mel-Scale Frequency Cepstral Coefficients (MFCCs) to improve the recognition rates of emotions from speeches using the classifier DNN. In this paper, EMD is first used to decompose the speech signals, which contain emotional components into multiple intrinsic mode functions (IMFs), and then emotional features are derived from the IMFs and are calculated using MFCC. Then, the emotional features are used to train the DNN model. Finally, a trained model that could recognize the emotional signals is then used to identify emotions in speeches. Experimental results reveal that the proposed method is effective.
About
Indigenized remote control interface card suitable for MAFI system CCR equipment. Compatible for IDM8000 CCR. Backplane mounted serial and TCP/Ethernet communication module for CCR remote access. IDM 8000 CCR remote control on serial and TCP protocol.
• Remote control: Parallel or serial interface.
• Compatible with MAFI CCR system.
• Compatible with IDM8000 CCR.
• Compatible with Backplane mount serial communication.
• Compatible with commercial and Defence aviation CCR system.
• Remote control system for accessing CCR and allied system over serial or TCP.
• Indigenized local Support/presence in India.
• Easy in configuration using DIP switches.
Technical Specifications
Indigenized remote control interface card suitable for MAFI system CCR equipment. Compatible for IDM8000 CCR. Backplane mounted serial and TCP/Ethernet communication module for CCR remote access. IDM 8000 CCR remote control on serial and TCP protocol.
Key Features
Indigenized remote control interface card suitable for MAFI system CCR equipment. Compatible for IDM8000 CCR. Backplane mounted serial and TCP/Ethernet communication module for CCR remote access. IDM 8000 CCR remote control on serial and TCP protocol.
• Remote control: Parallel or serial interface
• Compatible with MAFI CCR system
• Copatiable with IDM8000 CCR
• Compatible with Backplane mount serial communication.
• Compatible with commercial and Defence aviation CCR system.
• Remote control system for accessing CCR and allied system over serial or TCP.
• Indigenized local Support/presence in India.
Application
• Remote control: Parallel or serial interface.
• Compatible with MAFI CCR system.
• Compatible with IDM8000 CCR.
• Compatible with Backplane mount serial communication.
• Compatible with commercial and Defence aviation CCR system.
• Remote control system for accessing CCR and allied system over serial or TCP.
• Indigenized local Support/presence in India.
• Easy in configuration using DIP switches.
Water scarcity is the lack of fresh water resources to meet the standard water demand. There are two type of water scarcity. One is physical. The other is economic water scarcity.
COLLEGE BUS MANAGEMENT SYSTEM PROJECT REPORT.pdfKamal Acharya
The College Bus Management system is completely developed by Visual Basic .NET Version. The application is connect with most secured database language MS SQL Server. The application is develop by using best combination of front-end and back-end languages. The application is totally design like flat user interface. This flat user interface is more attractive user interface in 2017. The application is gives more important to the system functionality. The application is to manage the student’s details, driver’s details, bus details, bus route details, bus fees details and more. The application has only one unit for admin. The admin can manage the entire application. The admin can login into the application by using username and password of the admin. The application is develop for big and small colleges. It is more user friendly for non-computer person. Even they can easily learn how to manage the application within hours. The application is more secure by the admin. The system will give an effective output for the VB.Net and SQL Server given as input to the system. The compiled java program given as input to the system, after scanning the program will generate different reports. The application generates the report for users. The admin can view and download the report of the data. The application deliver the excel format reports. Because, excel formatted reports is very easy to understand the income and expense of the college bus. This application is mainly develop for windows operating system users. In 2017, 73% of people enterprises are using windows operating system. So the application will easily install for all the windows operating system users. The application-developed size is very low. The application consumes very low space in disk. Therefore, the user can allocate very minimum local disk space for this application.
Student information management system project report ii.pdfKamal Acharya
Our project explains about the student management. This project mainly explains the various actions related to student details. This project shows some ease in adding, editing and deleting the student details. It also provides a less time consuming process for viewing, adding, editing and deleting the marks of the students.
CFD Simulation of By-pass Flow in a HRSG module by R&R Consult.pptxR&R Consult
CFD analysis is incredibly effective at solving mysteries and improving the performance of complex systems!
Here's a great example: At a large natural gas-fired power plant, where they use waste heat to generate steam and energy, they were puzzled that their boiler wasn't producing as much steam as expected.
R&R and Tetra Engineering Group Inc. were asked to solve the issue with reduced steam production.
An inspection had shown that a significant amount of hot flue gas was bypassing the boiler tubes, where the heat was supposed to be transferred.
R&R Consult conducted a CFD analysis, which revealed that 6.3% of the flue gas was bypassing the boiler tubes without transferring heat. The analysis also showed that the flue gas was instead being directed along the sides of the boiler and between the modules that were supposed to capture the heat. This was the cause of the reduced performance.
Based on our results, Tetra Engineering installed covering plates to reduce the bypass flow. This improved the boiler's performance and increased electricity production.
It is always satisfying when we can help solve complex challenges like this. Do your systems also need a check-up or optimization? Give us a call!
Work done in cooperation with James Malloy and David Moelling from Tetra Engineering.
More examples of our work https://www.r-r-consult.dk/en/cases-en/
Vaccine management system project report documentation..pdfKamal Acharya
The Division of Vaccine and Immunization is facing increasing difficulty monitoring vaccines and other commodities distribution once they have been distributed from the national stores. With the introduction of new vaccines, more challenges have been anticipated with this additions posing serious threat to the already over strained vaccine supply chain system in Kenya.
NO1 Uk best vashikaran specialist in delhi vashikaran baba near me online vas...Amil Baba Dawood bangali
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Sachpazis:Terzaghi Bearing Capacity Estimation in simple terms with Calculati...Dr.Costas Sachpazis
Terzaghi's soil bearing capacity theory, developed by Karl Terzaghi, is a fundamental principle in geotechnical engineering used to determine the bearing capacity of shallow foundations. This theory provides a method to calculate the ultimate bearing capacity of soil, which is the maximum load per unit area that the soil can support without undergoing shear failure. The Calculation HTML Code included.
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Animal Voice Morphing System
Siddhapara Milan v.
milansiddhapara@gmial.com
Gujarat Technological University,
Gujarat, India.
Abstract:This paper describe the morphing concept in which we convert the voice of any person into pre -analyzed or pre-
recorded voice of any animals.As the user generate a pre-established voice, his pitch, timbre, vibrato and articulation can
be modified to resemble those of a pre-recorded and pre-analyzed voice of animal. This technique is based on SMS. Thus
using this concept we can develop many funny application and we can used this type of application in mobile device,
personal computer etc. for enjoying the sometime of period.
Keywords: Voice morphing, Animal.
I. Introduction
In general morphing means “conversion of some stages into some another stages”. Similarly in voice morphing us
convert the voice generated from the any source into pre-analyzed or predict target voice in other word voice morphing means
the transition of one speech signal into another. Like image morphing, speech morphing aims to preserve the shared
characteristics of the starting and final signals, while generating a smooth transition between them. In voice morphing
processone speech signal should smoothly change into another, keeping the shared characteristics of the starting and ending
signals but smoothly changing the other properties. The major properties of concern as far as a speech signal is concerned are
its pitch and envelope information. These two reside in a convolved form in a speech signal. Hence some efficient method for
extracting each of these is necessary.The main goal of the developed voice morphing methods is the smooth transformation from
one sound to another, thus, the combination of two sounds to create a new sound with an intermediate timbre. Most of these methods
are based on the interpolation of sounds parameterizations resulting from analysis/synthesis techniques, such as the Short-timeFourier
Transform (STFT), Linear Predictive Coding (LPC) or Sinusoidal Models.
In this paper we focus on specific concept, we convert the voice of any person into predict animal voice to generate
some joyful environment. In this paper user convert the pitch, articulation and also able to control the degree of morphing to
achieve the desire objective. Morphing take place in following stages, first system understand the voice of source (person) and
find same sound Pieces from the target voice, then interpolate the selected attributes and produced the morphed voice In this
paper [1] the implantation is discussed.
2. KNOWLEDGECUDDLE PUBLICATION
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II. Voice Morphing System
Fig.1 shows the general block diagram of voice imitator system. The underlying analysis technique is SMS.Also in
system recognition and alignment module was added for synchronizing the user’s voice with the target voice before the
morphing is done.Before we can morph a particular voice we have to supplyinformation about target information(predict animal
voice ) and voice to be morphed. The system requires the phonetic transcription of the lyrics, the melody as MIDI data, and the actual
recording to be used as the target audio data. Thus, a good impersonator of any animal that originally Voice has to be recorded. This
recording has to be analysed with SMS, segmented into “morphing units”, and each unit labelled with the appropriate note and
phonetic information of the song. This preparation stage is done semi-automatically, using a non-real time application developed for
this task. The first module of the running system includes the real-time analysis and the recognition/alignment steps. Each analysis
frame, with the appropriate parameterization, is associated with the phoneme of a specific moment of the voice and thus with a target
frame. The recognition/alignment algorithm is based on traditional speech recognition technology, that is, Hidden Markov Models
(HMM) that were adapted to the singing voice [2]
The source sound are segmented into number of frames and each frame are individual analyzed. The recognitionalignment
algorithm is based on hidden Markov models (HMM) [3] and when the source frame matched with target frame then we
morphed them and get the desired output.
III. Spectral Modeling Synthesis
We use the term SMS to describe our ongoing research on the development of techniques and software applications for the
analysis, transformation and synthesis of musical sounds based on spectral models [2].Information on SMS and public domain
software applications based on it can be found in our Web site (http://www.iua.upf.es/~sms).
IV. Analysis Using SMS
Analysis processes can be categorized as follows.
Preprocessing or representation conversion: This involves processes like signal acquisition in discrete form and
windowing.
Cepstral analysis or Pitch and Envelope analysis: This process will extract the pitch and formant information in the
speech signal.
Morphing which includes Warping and interpolation.
3. KNOWLEDGECUDDLE PUBLICATION
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Signal re-estimation.
The traditional SMS analysis output is a collection of frequency and amplitude values that represent the partials of the
sound (sinusoidal component), and either filter coefficients with a gain value or spectral magnitudes and phases representing
the residual sound (non sinusoidal component) [4]
The major improvement is done SMS model is alignment and analysis. A major improvement to SMS has been the real-time
implementation of the whole analysis/synthesis process, with a processing latency of less than 30 milliseconds.
V. Phonetic recognition/ alignment
This part of the system is responsible for recognizing the voice that is being generated from the userthat a similar segment
can be chosen from the target information.
The huge amount of research is carry out in speech recognition.The recognition systems work well when tested in the well-
controlled.The recognition systems work reasonably well when tested in the well-controlled environment of the laboratory.
However, phoneme recognition rates decay miserably when the conditions are adverse.
A. SMorphing
Speech morphing can be achieved by transforming the signal.To prepare the signal for the transformation, it is split
into a number of 'frames' - sections of the waveform. The transformation is then applied to each frame of the signal. This
provides another way of viewing the signal information. The new representation (said to be in the frequency domain) describes
the average energy present at each frequency band. Further analysis enables two pieces of information to be obtained: pitch
information and the overall envelope of the sound. A key element in the morphing is the manipulation of the pitch information.
If two signals with different pitches were simply cross-faded it is highly likely that two separate sounds will be heard. This
occurs because the signal will have two distinct pitches causing the auditory system to perceive two different objects. A
successful morph must exhibit a smoothly changing pitch throughout. The pitch information of each sound is compared to
provide the best match between the two signals' pitches. To do this match, the signals are stretched and compressed so that
important sections of each signal match in time. The interpolation of the two sounds can then be performed which creates the
intermediate sounds in the morph. The final stage is then to convert the frames back into a normal waveform.
However, after the morphing has been performed, the legacy of the earlier analysis becomes apparent. The conversion of the
sound to a representation in which the pitch and spectral envelope can be separated loses some information. Therefore, this
information has to be re-estimated for the morphed sound. This process obtains an acoustic waveform, which can then be
stored or listened to.
Depending on the voice of person (user), a unit from the target is selected. Each frame from the user is morphed with
a different frame from the target, advancing sequentially in time. Then the user has the choice to interpolate the different
parameters extracted at the analysis stage, such as amplitude, fundamental frequency, spectral shape, residual signal, etc. In
general the amplitude will not be interpolated, thus always using the amplitude from the user and the unvoiced phonemes will
also not be morphed, thus always using the consonants from the user. This will give the user the feeling of being in control.In
most cases the durations of the user and target voice to be morphed will be different. If a given user’s voice is shorter than the
one from the target the system will simply skip the remaining part of the target voice and go directly to the articulation portion.
In the case when the user voiceisa longer than the one present in the target data the system enters in the loop mode. Each
voiced phoneme of the target has a loop point frame, marked in the preprocessing, non-real time stage. The system uses this
frame to loop-synthesis in case the user sings beyond that point in the phoneme. Once we reach this frame in the target, the rest
of the frames of the user will be interpolated with that same frame until the user ends the phoneme.
VI. CONCLUSION
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The final purpose of the system is to make joyful voice from the user’s (source) voice. Means using this we developed the
many funny application but we need the some voice sample of target voice. For instance the voice of Karan is translated into
voice of cat. But Limitation is that we need a clear voice from the source so we can easily recognize or analyzed it. If there is
no clear voice of user then sample of that voice will converted into some another sound pieces of target animal. So if we have
clear voice then we easily convert into target vice and enjoy the sometime of period.
REFERENCES
[1]. Cano, P., A. Loscos, J. Bonada, M. de Boer, X. Serra.2000. “Singing Voice Impersonator Application for PC”.
Proceedings of the ICMC 2000.
[2]. Serra, X. 1997. “Musical Sound Modeling with Sinusoids plus Noise”. G. D. Poli and others (eds.), MusicalSignal
Processing, Swets & Zeitlinger Publishers, 1997.
[3]. Loscos, A., P. Cano, J. Bonada. 1999. “Low-Delay Singing Voice Alignment to text”. Proceedings ofthe ICMC 1999.
[4]. Serra, X. 1997. “Musical Sound Modeling with Sinusoids plus Noise”. G. D. Poli and others (eds.), Musical Signal
Processing, Swets & Zeitlinger Publishers, 1997.
![KNOWLEDGECUDDLE PUBLICATION
International Journal of Research in Advance Engineering (IJRAE) – Volume-1 Issue -1
1
www.knowledgecuddle.com
Animal Voice Morphing System
Siddhapara Milan v.
milansiddhapara@gmial.com
Gujarat Technological University,
Gujarat, India.
Abstract:This paper describe the morphing concept in which we convert the voice of any person into pre -analyzed or pre-
recorded voice of any animals.As the user generate a pre-established voice, his pitch, timbre, vibrato and articulation can
be modified to resemble those of a pre-recorded and pre-analyzed voice of animal. This technique is based on SMS. Thus
using this concept we can develop many funny application and we can used this type of application in mobile device,
personal computer etc. for enjoying the sometime of period.
Keywords: Voice morphing, Animal.
I. Introduction
In general morphing means “conversion of some stages into some another stages”. Similarly in voice morphing us
convert the voice generated from the any source into pre-analyzed or predict target voice in other word voice morphing means
the transition of one speech signal into another. Like image morphing, speech morphing aims to preserve the shared
characteristics of the starting and final signals, while generating a smooth transition between them. In voice morphing
processone speech signal should smoothly change into another, keeping the shared characteristics of the starting and ending
signals but smoothly changing the other properties. The major properties of concern as far as a speech signal is concerned are
its pitch and envelope information. These two reside in a convolved form in a speech signal. Hence some efficient method for
extracting each of these is necessary.The main goal of the developed voice morphing methods is the smooth transformation from
one sound to another, thus, the combination of two sounds to create a new sound with an intermediate timbre. Most of these methods
are based on the interpolation of sounds parameterizations resulting from analysis/synthesis techniques, such as the Short-timeFourier
Transform (STFT), Linear Predictive Coding (LPC) or Sinusoidal Models.
In this paper we focus on specific concept, we convert the voice of any person into predict animal voice to generate
some joyful environment. In this paper user convert the pitch, articulation and also able to control the degree of morphing to
achieve the desire objective. Morphing take place in following stages, first system understand the voice of source (person) and
find same sound Pieces from the target voice, then interpolate the selected attributes and produced the morphed voice In this
paper [1] the implantation is discussed.](https://image.slidesharecdn.com/1animalvoicemorphing-150423102433-conversion-gate01/85/Animal-Voice-Morphing-System-1-320.jpg)
![KNOWLEDGECUDDLE PUBLICATION
International Journal of Research in Advance Engineering (IJRAE) – Volume-1 Issue -1
2
www.knowledgecuddle.com
II. Voice Morphing System
Fig.1 shows the general block diagram of voice imitator system. The underlying analysis technique is SMS.Also in
system recognition and alignment module was added for synchronizing the user’s voice with the target voice before the
morphing is done.Before we can morph a particular voice we have to supplyinformation about target information(predict animal
voice ) and voice to be morphed. The system requires the phonetic transcription of the lyrics, the melody as MIDI data, and the actual
recording to be used as the target audio data. Thus, a good impersonator of any animal that originally Voice has to be recorded. This
recording has to be analysed with SMS, segmented into “morphing units”, and each unit labelled with the appropriate note and
phonetic information of the song. This preparation stage is done semi-automatically, using a non-real time application developed for
this task. The first module of the running system includes the real-time analysis and the recognition/alignment steps. Each analysis
frame, with the appropriate parameterization, is associated with the phoneme of a specific moment of the voice and thus with a target
frame. The recognition/alignment algorithm is based on traditional speech recognition technology, that is, Hidden Markov Models
(HMM) that were adapted to the singing voice [2]
The source sound are segmented into number of frames and each frame are individual analyzed. The recognitionalignment
algorithm is based on hidden Markov models (HMM) [3] and when the source frame matched with target frame then we
morphed them and get the desired output.
III. Spectral Modeling Synthesis
We use the term SMS to describe our ongoing research on the development of techniques and software applications for the
analysis, transformation and synthesis of musical sounds based on spectral models [2].Information on SMS and public domain
software applications based on it can be found in our Web site (http://www.iua.upf.es/~sms).
IV. Analysis Using SMS
Analysis processes can be categorized as follows.
Preprocessing or representation conversion: This involves processes like signal acquisition in discrete form and
windowing.
Cepstral analysis or Pitch and Envelope analysis: This process will extract the pitch and formant information in the
speech signal.
Morphing which includes Warping and interpolation.](data:image/gif;base64,R0lGODlhAQABAIAAAAAAAP///yH5BAEAAAAALAAAAAABAAEAAAIBRAA7)