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Dissertations / Theses on the topic 'Mel-Frequency Cepstral coefficients'

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Published: 4 June 2021

Last updated: 25 July 2025

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1

Darch, Jonathan J. A. "Robust acoustic speech feature prediction from Mel frequency cepstral coefficients." Thesis, University of East Anglia, 2008. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.445206.

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Edman, Sebastian. "Radar target classification using Support Vector Machines and Mel Frequency Cepstral Coefficients." Thesis, KTH, Optimeringslära och systemteori, 2017. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-214794.

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In radar applications, there are often times when one does not only want to know that there is a target that reflecting the out sent signals but also what kind of target that reflecting these signals. This project investigates the possibilities to from raw radar data transform reflected signals and take use of human perception, in particular our hearing, and by a machine learning approach where patterns and characteristics in data are used to answer the earlier mentioned question. More specific the investigation treats two kinds of targets that are fairly comparable namely smaller Unmanned Aer

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3

Yang, Chenguang. "Security in Voice Authentication." Digital WPI, 2014. https://digitalcommons.wpi.edu/etd-dissertations/79.

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We evaluate the security of human voice password databases from an information theoretical point of view. More specifically, we provide a theoretical estimation on the amount of entropy in human voice when processed using the conventional GMM-UBM technologies and the MFCCs as the acoustic features. The theoretical estimation gives rise to a methodology for analyzing the security level in a corpus of human voice. That is, given a database containing speech signals, we provide a method for estimating the relative entropy (Kullback-Leibler divergence) of the database thereby establishing the secu

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4

Wu, Qiming. "A robust audio-based symbol recognition system using machine learning techniques." University of the Western Cape, 2020. http://hdl.handle.net/11394/7614.

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Masters of Science<br>This research investigates the creation of an audio-shape recognition system that is able to interpret a user’s drawn audio shapes—fundamental shapes, digits and/or letters— on a given surface such as a table-top using a generic stylus such as the back of a pen. The system aims to make use of one, two or three Piezo microphones, as required, to capture the sound of the audio gestures, and a combination of the Mel-Frequency Cepstral Coeﬃcients (MFCC) feature descriptor and Support Vector Machines (SVMs) to recognise audio shapes. The novelty of the system is in the use of

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5

Candel, Ramón Antonio José. "Verificación automática de locutores aplicando pruebas diagnósticas múltiples en serie y en paralelo basadas en DTW (Dynamic Time Warping) y NFCC (Mel-Frequency Cepstral coefficients)." Doctoral thesis, Universidad de Murcia, 2015. http://hdl.handle.net/10803/300433.

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La presente Tesis Doctoral consiste en el diseño de un sistema capaz de realizar tareas de verificación automática de locutores, para lo cual se basa en el modelado mediante los procedimientos DTW (Dynamic Time Warping) y MFCC (Mel-Frequency Cepstral Coefficients). Una vez diseñado éste, se ha evaluado el sistema de forma tanto a nivel de pruebas individuales, DTW y MFCC por separado, como múltiples, combinación de ambas en serie y en paralelo, para grabaciones obtenidas de la base de datos AHUMADA de la Guardia Civil. Todos los resultados han sido vistos teniendo en cuenta la significación es

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6

Lindstål, Tim, and Daniel Marklund. "Application of LabVIEW and myRIO to voice controlled home automation." Thesis, Uppsala universitet, Signaler och System, 2019. http://urn.kb.se/resolve?urn=urn:nbn:se:uu:diva-380866.

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The aim of this project is to use NI myRIO and LabVIEW for voice controlled home automation. The NI myRIO is an embedded device which has a Xilinx FPGA and a dual-core ARM Cortex-A9processor as well as analog input/output and digital input/output, and is programmed with theLabVIEW, a graphical programming language. The voice control is implemented in two differentsystems. The first system is based on an Amazon Echo Dot for voice recognition, which is acommercial smart speaker developed by Amazon Lab126. The Echo Dot devices are connectedvia the Internet to the voice-controlled intelligent pers

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7

Larsson, Alm Kevin. "Automatic Speech Quality Assessment in Unified Communication : A Case Study." Thesis, Linköpings universitet, Programvara och system, 2019. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-159794.

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Speech as a medium for communication has always been important in its ability to convey our ideas, personality and emotions. It is therefore not strange that Quality of Experience (QoE) becomes central to any business relying on voice communication. Using Unified Communication (UC) systems, users can communicate with each other in several ways using many different devices, making QoE an important aspect for such systems. For this thesis, automatic methods for assessing speech quality of the voice calls in Briteback’s UC application is studied, including a comparison of the researched methods.

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8

Neville, Katrina Lee, and katrina neville@rmit edu au. "Channel Compensation for Speaker Recognition Systems." RMIT University. Electrical and Computer Engineering, 2007. http://adt.lib.rmit.edu.au/adt/public/adt-VIT20080514.093453.

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This thesis attempts to address the problem of how best to remedy different types of channel distortions on speech when that speech is to be used in automatic speaker recognition and verification systems. Automatic speaker recognition is when a person's voice is analysed by a machine and the person's identity is worked out by the comparison of speech features to a known set of speech features. Automatic speaker verification is when a person claims an identity and the machine determines if that claimed identity is correct or whether that person is an impostor. Channel distortion occurs wh

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9

Alvarenga, Rodrigo Jorge. "Reconhecimento de comandos de voz por redes neurais." Universidade de Taubaté, 2012. http://www.bdtd.unitau.br/tedesimplificado/tde_busca/arquivo.php?codArquivo=587.

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Sistema de reconhecimento de fala tem amplo emprego no universo industrial, no aperfeiçoamento de operações e procedimentos humanos e no setor do entretenimento e recreação. O objetivo específico do trabalho foi conceber e desenvolver um sistema de reconhecimento de voz, capaz de identificar comandos de voz, independentemente do locutor. A finalidade precípua do sistema é controlar movimentos de robôs, com aplicações na indústria e no auxílio de deficientes físicos. Utilizou-se a abordagem da tomada de decisão por meio de uma rede neural treinada com as características distintivas do sinal de

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10

Larsson, Joel. "Optimizing text-independent speaker recognition using an LSTM neural network." Thesis, Mälardalens högskola, Akademin för innovation, design och teknik, 2014. http://urn.kb.se/resolve?urn=urn:nbn:se:mdh:diva-26312.

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In this paper a novel speaker recognition system is introduced. Automated speaker recognition has become increasingly popular to aid in crime investigations and authorization processes with the advances in computer science. Here, a recurrent neural network approach is used to learn to identify ten speakers within a set of 21 audio books. Audio signals are processed via spectral analysis into Mel Frequency Cepstral Coefficients that serve as speaker specific features, which are input to the neural network. The Long Short-Term Memory algorithm is examined for the first time within this area, wit

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Hrabina, Martin. "VÝVOJ ALGORITMŮ PRO ROZPOZNÁVÁNÍ VÝSTŘELŮ." Doctoral thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2019. http://www.nusl.cz/ntk/nusl-409087.

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Táto práca sa zaoberá rozpoznávaním výstrelov a pridruženými problémami. Ako prvé je celá vec predstavená a rozdelená na menšie kroky. Ďalej je poskytnutý prehľad zvukových databáz, významné publikácie, akcie a súčasný stav veci spoločne s prehľadom možných aplikácií detekcie výstrelov. Druhá časť pozostáva z porovnávania príznakov pomocou rôznych metrík spoločne s porovnaním ich výkonu pri rozpoznávaní. Nasleduje porovnanie algoritmov rozpoznávania a sú uvedené nové príznaky použiteľné pri rozpoznávaní. Práca vrcholí návrhom dvojstupňového systému na rozpoznávanie výstrelov, monitorujúceho ok

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Zezula, Miroslav. "Online detekce jednoduchých příkazů v audiosignálu." Master's thesis, Vysoké učení technické v Brně. Fakulta strojního inženýrství, 2011. http://www.nusl.cz/ntk/nusl-229484.

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This thesis describes the development of voice module, that can recognize simple speech commands by comparation of input sound with recorded templates. The first part of thesis contains a description of used algorithm and a verification of its functionality. The algorithm is based on Mel-frequency cepstral coefficients and dynamic time warping. Thereafter the hardware of voice module is designed, containing signal controller 56F805 from Freescale. The signal from microphone is conditioned by operational amplifiers and digital filter. The third part deals with the development of software for th

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13

Mahajan, Mayur. "Development of a speech recognition system using the Mel Frequency Cepstrum Coefficient method." Thesis, California State University, Long Beach, 2016. http://pqdtopen.proquest.com/#viewpdf?dispub=10141515.

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<p> Voice recognition systems have found widespread use in applications such as tele-shopping, tele-banking, information services, home automation, voice message security, and voice call dialing, which allows a driver to make calls safely while driving. </p><p> This project presents the development of a high performance speech recognition system using human voice models. Recognizing the behavior of the human ear, the Mel Frequency Cepstral Coefficient (MFCC) method is used to develop the system capability for feature extraction. Vector quantization optimized by the Linde Buzo Gray (LGB) algo

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14

Ulrich, Natalja. "Linguistic and speaker variation in Russian fricatives." Electronic Thesis or Diss., Lyon 2, 2022. http://www.theses.fr/2022LYO20031.

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Cette thèse présente une investigation acoustico-phonétique des détails phonétiques des fricatives russes.L'objectif principal était de détecter des corrélats acoustiques porteurs d'infor- mations linguistiques et idiosyncrasiques. Les questions abordées étaient de savoir si le lieu d'articulation, le sexe du locuteur ou son identité peuvent être prédits par des indices acoustiques et quelles mesures acoustiques représentent les indicateurs les plus fiables. En outre, la distribution des caractéristiques spécifiques au locuteur et à la variation inter et intra locuteur à travers les indices ac

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Hrušovský, Enrik. "Automatická klasifikace výslovnosti hlásky R." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2018. http://www.nusl.cz/ntk/nusl-377664.

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This diploma thesis deals with automatic clasification of vowel R. Purpose of this thesis is to made program for detection of pronounciation of speech defects at vowel R in children. In thesis are processed parts as speech creation, speech therapy, dyslalia and subsequently speech signal processing and analysis methods. In the last part is designed software for automatic detection of pronounciation of vowel R. For recognition of pronounciation is used algorithm MFCC for extracting features. This features are subsequently classified by neural network to the group of correct or incorrect pronoun

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Okuyucu, Cigdem. "Semantic Classification And Retrieval System For Environmental Sounds." Master's thesis, METU, 2012. http://etd.lib.metu.edu.tr/upload/12615114/index.pdf.

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The growth of multimedia content in recent years motivated the research on audio classification and content retrieval area. In this thesis, a general environmental audio classification and retrieval approach is proposed in which higher level semantic classes (outdoor, nature, meeting and violence) are obtained from lower level acoustic classes (emergency alarm, car horn, gun-shot, explosion, automobile, motorcycle, helicopter, wind, water, rain, applause, crowd and laughter). In order to classify an audio sample into acoustic classes, MPEG-7 audio features, Mel Frequency Cepstral Coefficients

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Assaad, Firas Souhail. "Biometric Multi-modal User Authentication System based on Ensemble Classifier." University of Toledo / OhioLINK, 2014. http://rave.ohiolink.edu/etdc/view?acc_num=toledo1418074931.

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18

Dušil, Lubomír. "Automatické rozpoznávání logopedických vad v řečovém projevu." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2009. http://www.nusl.cz/ntk/nusl-218161.

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The thesis is aimed at an analysis and automatic detection of logopaedic defects in speech utterance. Its objective is to facilitate and accelerate the work of logopaedists and to increase percentage of detected logopaedic defects in children of the youngest possible age followed by the most successful treatment. It presents methods of speech work, classification of the defects within individual stages of child development and appropriate words for identification of the speech defects and their subsequent remedy. After that there are analyses of methods of calculating coefficients which reflec

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Pešek, Milan. "Detekce logopedických vad v řeči." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2009. http://www.nusl.cz/ntk/nusl-218106.

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The thesis deals with a design and an implementation of software for a detection of logopaedia defects of speech. Due to the need of early logopaedia defects detecting, this software is aimed at a child’s age speaker. The introductory part describes the theory of speech realization, simulation of speech realization for numerical processing, phonetics, logopaedia and basic logopaedia defects of speech. There are also described used methods for feature extraction, for segmentation of words to speech sounds and for features classification into either correct or incorrect pronunciation class. In t

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Wang, Yihan. "Automatic Speech Recognition Model for Swedish using Kaldi." Thesis, KTH, Skolan för elektroteknik och datavetenskap (EECS), 2020. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-285538.

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With the development of intelligent era, speech recognition has been a hottopic. Although many automatic speech recognition(ASR) tools have beenput into the market, a considerable number of them do not support Swedishbecause of its small number. In this project, a Swedish ASR model basedon Hidden Markov Model and Gaussian Mixture Models is established usingKaldi which aims to help ICA Banken complete the classification of aftersalesvoice calls. A variety of model patterns have been explored, whichhave different phoneme combination methods and eigenvalue extraction andprocessing methods. Word E

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Лавриненко, Олександр Юрійович, Александр Юрьевич Лавриненко та Oleksandr Lavrynenko. "Методи підвищення ефективності семантичного кодування мовних сигналів". Thesis, Національний авіаційний університет, 2021. https://er.nau.edu.ua/handle/NAU/52212.

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Дисертаційна робота присвячена вирішенню актуальної науково-практичної проблеми в телекомунікаційних системах, а саме підвищення пропускної здатності каналу передачі семантичних мовних даних за рахунок ефективного їх кодування, тобто формулюється питання підвищення ефективності семантичного кодування, а саме – з якою мінімальною швидкістю можливо кодувати семантичні ознаки мовних сигналів із заданою ймовірністю безпомилкового їх розпізнавання? Саме на це питання буде дана відповідь у даному науковому дослідженні, що є актуальною науково-технічною задачею враховуючи зростаючу тенденцію д

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Bekli, Zeid, and William Ouda. "A performance measurement of a Speaker Verification system based on a variance in data collection for Gaussian Mixture Model and Universal Background Model." Thesis, Malmö universitet, Fakulteten för teknik och samhälle (TS), 2018. http://urn.kb.se/resolve?urn=urn:nbn:se:mau:diva-20122.

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Voice recognition has become a more focused and researched field in the last century,and new techniques to identify speech has been introduced. A part of voice recognition isspeaker verification which is divided into Front-end and Back-end. The first componentis the front-end or feature extraction where techniques such as Mel-Frequency CepstrumCoefficients (MFCC) is used to extract the speaker specific features of a speech signal,MFCC is mostly used because it is based on the known variations of the humans ear’scritical frequency bandwidth. The second component is the back-end and handles thes

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Sklar, Alexander Gabriel. "Channel Modeling Applied to Robust Automatic Speech Recognition." Scholarly Repository, 2007. http://scholarlyrepository.miami.edu/oa_theses/87.

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In automatic speech recognition systems (ASRs), training is a critical phase to the system?s success. Communication media, either analog (such as analog landline phones) or digital (VoIP) distort the speaker?s speech signal often in very complex ways: linear distortion occurs in all channels, either in the magnitude or phase spectrum. Non-linear but time-invariant distortion will always appear in all real systems. In digital systems we also have network effects which will produce packet losses and delays and repeated packets. Finally, one cannot really assert what path a signal will take, and

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Tang, Chu-Liang, and 唐曲亮. "Improved Mel Frequency Cepstral Coefficients Combined with Multiple Speech Features." Thesis, 2015. http://ndltd.ncl.edu.tw/handle/57856949340151071584.

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碩士<br>國立中央大學<br>電機工程學系<br>103<br>This thesis studies the speech feature extracting and feature compensation in speech recognition. Several speech features are selected for combinations. The best one is cascading Linear Prediction Cepstral Coefficients (LPCC) and Mel-Frequency Cepstral Coefficient (MFCC). The MFCCs used here are obtained by utilizing a Gaussian Mel-Frequency band instead of using a triangular filter bank. And by experiments, it is found that the best combination ratio of LPCC and MFCC is 1:1. The thesis also showed that further improved performance is possible if Cepstral Mean

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Kuo, Yo-zhen, and 郭又禎. "Improved Mel-scale Frequency Cepstral Coefficients for Keyword Spotting Technique." Thesis, 2014. http://ndltd.ncl.edu.tw/handle/27592493670347223949.

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碩士<br>國立中央大學<br>電機工程學系<br>102<br>In the speech recognition system, Mel frequency cepstral coefficients (MFCCs) are the feature parameters that are used widely. Because of the wide applications of MFCC in the audio signal processing, lots of studies on the improvement of MFCCs were presented. In this study, we use particle swarm optimization algorithm to optimize the weight of MFCC filter bank. We utilize the difference between voice training database’s energy statistical curve and MFCC filter bank’s envelope as fitness function. Experimental results show that the proposed MFCCs method improves

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Lin, Shih-Fen, and 林士棻. "Bird songs recognition using two-dimensional Mel-scale frequency cepstral coefficients." Thesis, 2006. http://ndltd.ncl.edu.tw/handle/94553686394732089037.

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林士棻. "Bird songs recognition using two-dimensional Mel-scale frequency cepstral coefficients." Thesis, 2006. http://ndltd.ncl.edu.tw/handle/38302762655714685237.

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碩士<br>中華大學<br>資訊工程學系(所)<br>94<br>We propose a method to automatically identify birds from their sounds in this paper. First, each syllable corresponding to a piece of vocalization is segmented. The average LPCC (ALPCC), average MFCC (AMFCC), Static MFCC (SMFCC), Two-dimensional MFCC (TDMFCC), Dynamic two-dimensional MFCC (DTDMFCC) and TDMFCC+DTDMFCC over all frames in a syllable are calculated as the vocalization features. Linear discriminant analysis (LDA) is exploited to increase the classification accuracy at a lower dimensional feature vector space. A clustering algorithm, called progressi

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HUANG, CHUAN-HAO, and 黃川豪. "Multi-feature Speaker Verification Based on Mel-frequency cepstral coefficients and Formants." Thesis, 2019. http://ndltd.ncl.edu.tw/handle/4nbqev.

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Xu, Sheng-Bin, and 徐勝斌. "Continuous Birdsong Recognition Using Dynamic and Temporal Two-Dimensional Mel-Frequency Cepstral Coefficients." Thesis, 2009. http://ndltd.ncl.edu.tw/handle/21749503795140776068.

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碩士<br>中華大學<br>資訊工程學系(所)<br>97<br>In this paper, we will propose an approach for the classification of bird species using fixed-duration sound segments extracted from continuous birdsong recordings. First, each sound segment is divided into a number of overlapped texture windows. Each texture window will be individually classified and then a fusion approach is employed to determine the classification result of the input segment. The features derived from static, transitional, and temporal information of two-dimensional Mel-frequency cepstral coefficients (TDMFCC) will be extracted for the class

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CHIANG, MING-DA, and 蔣明達. "Speaker Recognition Using Mel-Scale Frequency Cepstral Coefficients by Time Domain Filtering method." Thesis, 2008. http://ndltd.ncl.edu.tw/handle/13444981721982290438.

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碩士<br>中華技術學院<br>電子工程研究所碩士班<br>96<br>ABSTRACT According to past papers, we find that the algorithm based on Mel-frequency cepstral coefficients (MFCCs) has a better performance than any other algorithms which based on the other feature parameters [1-7]. The Mel-frequency cepstral coefficients are taken by following procedures, including: framing, multiplied by the Hamming Window, taking the fast Fourier transform (FFT), filtered in frequency domain by Mel-frequency triangular filter bank, calculating the logarithmic energy of filter outputs, and taking discrete cosine transform (DCT) to obtain

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Lin, Bo-Zhi, and 林柏志. "Speaker Recognition Algorithm Using Mel-Scale Frequency Cepstral Coefficients with Two Stages Linear Prediction Filters." Thesis, 2006. http://ndltd.ncl.edu.tw/handle/18209732501243789128.

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碩士<br>中華技術學院<br>電子工程研究所碩士班<br>94<br>The development of computer and communication technologies hastens the application requirements of speaker recognition and speech recognition. The purpose of this paper is to present a new algorithm to promote the performance of speaker recognition. The algorithm uses two stages linear prediction error filters to estimate the spectrogram of the processed speech signal. Then, the algorithm uses Mel-scale triangle bandpass filters bank to obtain the Mel-scale frequency cepstral coefficients（MFCC）to build the needed Gaussian mixture model for speaker recognitio

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Yang-Ming, Cheng, and 鄭陽銘. "A Mel-Scale Frequency Cepstral Coefficients Speaker Recognition Algorithm Based on Linear Prediction Spectrum Estimation." Thesis, 2005. http://ndltd.ncl.edu.tw/handle/38345345070598427641.

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碩士<br>中華技術學院<br>電子工程研究所碩士班<br>93<br>According to the past research, we know that the spectrum estimation based on linear prediction is more robust than the spectrum estimation based on FFT in the case of lower SNR. In this paper, we propose a new speaker identification algorithm based on linear prediction spectrum estimation. In this algorithm, the spectrum estimation algorithm based on short time faster Furrier transform is replaced by the linear prediction spectrum estimation algorithm, then, the Mel-scale frequency cepstral coefficients are obtained by using the Mel-scale frequency triangle

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Bowman, Casady. "Perceiving Emotion in Sounds: Does Timbre Play a Role?" Thesis, 2011. http://hdl.handle.net/1969.1/ETD-TAMU-2011-12-10656.

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Acoustic features of sound such as pitch, loudness, perceived duration and timbre have been shown to be related to emotion in regard to sound, demonstrating that an important connection between the perceived emotions and their timbres is lacking. This study investigates the relationship between acoustic features of sound and emotion in regard to timbre. In two experiments we investigated whether particular acoustic components of sound can predict timbre, and particular categories of emotion, and how these attributes are related. Two behavioral experiments related perceived emotion ratings with

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Wu, Sunrise, and 吳尚叡. "Design Time Domain Filter Banks Using Least Squares Method to Calculate the Mel-Frequency Cepstral Coefficients for Speaker Recognition." Thesis, 2008. http://ndltd.ncl.edu.tw/handle/08178129842426697899.

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碩士<br>中華技術學院<br>電子工程研究所碩士班<br>96<br>Up to now, the best speaker recognition technique is based on Mel-frequency cepstral coefficients (MFCCs) [1-4,11] method. The main procedures on taking MFCCs are undergone by: framing, Hamming windowing, multiplied by FFT（Fast Fourier Transform）[7], filtered by Mel-scale triangular filter bank, taken the logarithmic energies of outputs, and transformed by DCT (Discrete Cosine Transform)[1-8]. After these processes, the MFCCs are obtained. The main topic of this thesis is we replace previous procedures of FFT [7] and filtering using a frequency-domain Mel-sc

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Yuan, Hor, and 原禾. "Design Time Domain Filter Banks Using Least Squares Method to Calculate the Mel-Frequency Cepstral Coefficients for Non-Continuous Speech Recognition." Thesis, 2009. http://ndltd.ncl.edu.tw/handle/76162451347630250736.

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碩士<br>中華技術學院<br>電子工程研究所碩士班<br>97<br>In speech recognition, the Mel frequency cepstral coefficients (MFCC) is currently popular to be used in speech recognition and speaker recognition[2,8-11,14,15]. To obtain the MFCC, the main procedures are filtering the speech signal by a set of triangular Mel-scale Filter Bank in the frequency domain to obtain the logarithm of the output powers of filter bank, and then taking Discrete Cosine Transform to obtain the MFCC. In this paper, the frequency domain triangular Mel-scale filter bank is replaced by a new designed time domain triangular Mel-scale filte

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Sujatha, J. "Improved MFCC Front End Using Spectral Maxima For Noisy Speech Recognition." Thesis, 2005. https://etd.iisc.ac.in/handle/2005/1506.

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Sujatha, J. "Improved MFCC Front End Using Spectral Maxima For Noisy Speech Recognition." Thesis, 2005. http://etd.iisc.ernet.in/handle/2005/1506.

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Lei, Ying, and 雷穎. "Chip Design of Mel Frequency Cepstral Coefficient for Speech Recognition." Thesis, 2006. http://ndltd.ncl.edu.tw/handle/69494964042006361607.

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碩士<br>國立暨南國際大學<br>電機工程學系<br>94<br>This paper proposed the chip design of speech recognition for multimedia system. It is composed by three cores: a low-power high performance fast fourier transform (FFT) processor, a Mel-scale frequency cepstral coefficient (MFCC) circuit, and a dual-ALU digital signal process (DSP) processor with dynamic time warping speech recognition algorithm. The DSP processor had been implemented by previous researcher. In this paper, we mainly proposed the FFT processor and MFCC chip. In the FFT processor, we proposed a novel register array based pipelined radix-2 struc

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Singh, Jatindra Kumar, and Sujit Behera. "Speaker verification using Mel Frequency Cepstral Coefficient and Artificial Neural Network." Thesis, 2012. http://ethesis.nitrkl.ac.in/3745/1/final_yr_project__thesis.pdf.

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Speaker recognition is defined as to make sure that if the person is the same person he claims to be or not. This technique is one of the biometric recognition techniques useful in all most all areas where security is a concern. Speaker recognition can be divided into speaker identification and speaker verification. Speaker identification decides if a speaker is a specific person or is from a group. In speaker verification, a person makes an identity claim (e.g., by entering a pin number with the debit/credit card at ATM). There are two main stages in this technique, feature extraction and f

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Liu, Yi-Ming, and 劉益銘. "The Chip Design of Reconfigurable FFT-Based Mel Frequency Cepstrum Coefficient." Thesis, 2008. http://ndltd.ncl.edu.tw/handle/51081196289542485757.

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碩士<br>國立暨南國際大學<br>電機工程學系<br>96<br>This thesis proposed the chip design of speech recognition for multimedia system. It is composed by three cores: a reconfigurable fast Fourier transform (FFT) processor, a Mel-scale frequency cepstrum coefficient (MFCC) circuit, and a dual-ALU digital signal process (DSP) processor with dynamic time warping speech recognition algorithm. In this FFT processor, we used the reconfigurable architecture to design our FFT processor. We proposed a novel register array based pipelined radix-22 structure to reduce power consumption and computation cycles. The FFT proce

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Liu, Yi-Ming. "The Chip Design of Reconfigurable FFT-Based Mel Frequency Cepstrum Coefficient." 2008. http://www.cetd.com.tw/ec/thesisdetail.aspx?etdun=U0020-1607200815503000.

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(6642491), Jingzhao Dai. "SPARSE DISCRETE WAVELET DECOMPOSITION AND FILTER BANK TECHNIQUES FOR SPEECH RECOGNITION." Thesis, 2019.

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<p>Speech recognition is widely applied to translation from speech to related text, voice driven commands, human machine interface and so on [1]-[8]. It has been increasingly proliferated to Human’s lives in the modern age. To improve the accuracy of speech recognition, various algorithms such as artificial neural network, hidden Markov model and so on have been developed [1], [2].</p> <p>In this thesis work, the tasks of speech recognition with various classifiers are investigated. The classifiers employed include the support vector machine (SVM), k-nearest neighbors (KNN), random forest (RF

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Chen, Chia-Yu, and 陳佳妤. "The Investigation of Chinese Vowel Recognition for Mel-Frequency Cepstrum Coefficient Feature." Thesis, 2011. http://ndltd.ncl.edu.tw/handle/9jf6bu.

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碩士<br>國立中興大學<br>統計學研究所<br>99<br>This paper is mainly to discuss the speech vowel recognition of 337 isolated mandarin words for dependence. We use the features of Mel-Frequency Cepstrum Coefficient(Mfcc). We consider three factors such as “the length of frame”, “the duration of the consonant” and “the dimension of speech feature”in the experiment. The method of K-nearest neighbor is used for recognition and the optimal combination can be found for the three experimental factors. In the experimental result, the best recognition can be up to 98.5%.

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Weng, Yu-Sheng, and 翁育生. "The Chip Design of Mel Frequency Cepstrum Coefficient for HMM Speech Recognition." Thesis, 1998. http://ndltd.ncl.edu.tw/handle/80385881218692562375.

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碩士<br>國立成功大學<br>電機工程學系<br>86<br>MFCC倒頻譜係數是語音特徵參數的一種，主要從人類聽覺的物理特性所得來，非常能夠代表語音，除了計算方式較LPCC倒頻譜參數更加直接外，也有相當不錯的辨識率。此外語音辨識近來則以隱藏式馬可夫模型（HMM ）為主要架構，因為在許多相關應用方面都有相當不錯的效果，所以使用該模型作為參數求取後的語音辨識驗證。因此我們希望將MFCC演算法以硬體實現，並做為未來整個辨識系統中參數計算的模組。本論文首先詳細分析整個原始MFCC演算法所需計算量，使用簡化過之餘弦查表方法（ simplified cosine table-lookup method）降低一半的記憶體，也將乘法運算減少為原來二分之一；其次，利用Mel 頻率座標轉換之特性，將加權能量頻譜所需之相關乘法運算與記憶體同時降為原來一半；最後更採用修改後之分割式查表法（modified partitioned logarithm table look-

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Chen, Wan-Yu, and 陳宛余. "The Investigation of Capturing Mel-Frequency Cepstrum Coefficient Features on Mandarin Consonant Word Recognition." Thesis, 2011. http://ndltd.ncl.edu.tw/handle/33299797950806290067.

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碩士<br>中興大學<br>統計學研究所<br>99<br>The aim of this paper is to discuss the 337 mandarin consonant word recognition under the vowel is correct. The feature of Mel-frequency cepstrum coefficient (MFCC) and k-nearest neighbor (KNN) method are used for the recognition. Four experimental factors are considered in the word. That is, the length of frame, the dimension of MACC, the swing of frame, the duration of the consonant. We can fine the optimal parameters to the four experimental factors and the highest recognition is 95.84%.

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Chu, Feng-Seng, and 朱峰森. "Improved Approaches of Processing Perceptual Linear Prediction（PLP）and Mel Frequency Cepstrum Coefficient（MFCC）Parameters for Robust Speech Recognition." Thesis, 2005. http://ndltd.ncl.edu.tw/handle/26578739886453071884.

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Jhong, Jing-Jyue, and 鍾靖爵. "Using the Method of Common Vector to Recognize Isolated Mandarin Word for Speaker-dependent System with Optimal Mel-frequency Cepstrum Coefficient." Thesis, 2011. http://ndltd.ncl.edu.tw/handle/zwc24k.

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碩士<br>國立中興大學<br>統計學研究所<br>99<br>The paper aims to investigate the recognition of 340 isolated mandarin words by using the feature of Mel-frequency cepstrum coefficient, since it is the most well-known feature which can be less susceptible to noise interference. The speech model is constructed by the method of common vector. Moreover, the two-stage method will also be used to improve the recognition for similar consonants. The speech database in this experiment are recorded by twelve different speakers. Each isolated mandarin word is recorded ten times. We will find the optimal parameter set in

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Wu, Jhong-Da, and 吳忠達. "Using K-Nearest Neighbor Method and the Optimal Mel-Frequency Cepstrum Coefficient Feature to Recognize Isolated Mandarin Word for Speaker-Dependent System." Thesis, 2011. http://ndltd.ncl.edu.tw/handle/47366663775116977709.

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碩士<br>中興大學<br>統計學研究所<br>99<br>This paper is mainly to discuss the speech recognition of 337 isolation mandarin words for speaker dependent. The feature is Mel-frequency cepstrum coefficient(Mfcc), and the method is k-nearest neighbor(knn), for the recognition, we try to find out the optimal parameters to obtain high performance recognition. Six experimental factors(the length of frame, the dimension of Mfcc, the number of frame, the weight of consonant and vowel, the swing of frame and the duration of consonant) we considered in the work. We find that the best average rate of recognition in d

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