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Dissertations / Theses on the topic 'MFCC'

Author: Grafiati
Published: 4 June 2021
Last updated: 25 July 2025

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1

Mukherjee, Rishiraj. "Speaker Recognition Using Shifted MFCC." Scholar Commons, 2012. http://scholarcommons.usf.edu/etd/4136.

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Speaker Recognition is the art of recognizing a speaker from a given database using speech as the only input. In this thesis we will be discussing a novel approach to detect speakers. Here we will introduce the concept of shifted MFCC to add improvement over the performance from previous work which has shown quite a decent amount of accuracy of about 95% at best. We will be talking about adding different parameters which also contributed in improving the efficiency of speaker recognition. Also we will be testing our algorithm on Text dependent speech data and Text Independent speech data. Our
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2

Tolunay, Atahan. "Text-Dependent Speaker Verification Implemented in Matlab Using MFCC and DTW." Thesis, Linköpings universitet, Informationskodning, 2010. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-60992.

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Even though speaker verification is a broad subject, the commercial and personal use implementations are rare. There are several problems that need to be solved before speaker verification can become more useful. The amount of pattern matching and feature extraction techniques is large and the decision on which ones to use is debatable. One of the main problems of speaker verification in general is the impact of noise. The very popular feature extraction technique MFCC is inherently sensitive to mismatch between training and verification conditions. MFCC is used in many speech recognition appl
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3

Krotký, Jan. "Dekodér pro systém detekce klíčových slov." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2009. http://www.nusl.cz/ntk/nusl-218176.

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The essay presents the basic characteristics of human speech recognition, describes systems for the detection of key words and further deals with the proposal of each decoder blocks divided into three chapters. The first one describes the operations that are performed before the signal distribution of the framework and the segmentation. The second chapter describes the calculation of short-term energy, the number of zero passes and self-correlative, prediction and Mel-frequency cepstral coefficients. The third chapter, which describes the design of the block decoder, describes the method of dy
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4

Mubarak, Omer Mohsin Electrical Engineering &amp Telecommunications Faculty of Engineering UNSW. "Speech and music discrimination using short-time features." Awarded by:University of New South Wales. Electrical Engineering & Telecommunications, 2006. http://handle.unsw.edu.au/1959.4/31954.

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This thesis addresses the problem of classifying an audio stream as either speech or music, an issue which is beginning to receive increasing attention due to its wide range of applications. Various techniques have been presented in last decade to discriminate between speech and music. However, their accuracy is still not sufficient since music can refer to a very broad class of signals due to the large number of musical instruments found in audio data. Performance can also be further compromised in noisy conditions, which are unavoidable in some practical situations. This thesis presents an a
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5

Pan, Linlin. "Research and simulation on speech recognition by Matlab." Thesis, Högskolan i Gävle, Avdelningen för elektronik, matematik och naturvetenskap, 2014. http://urn.kb.se/resolve?urn=urn:nbn:se:hig:diva-16950.

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With the development of multimedia technology, speech recognition technology has increasingly become a hotspot of research in recent years. It has a wide range of applications, which deals with recognizing the identity of the speakers that can be classified into speech identification and speech verification according to decision modes.The main work of this thesis is to study and research the techniques, algorithms of speech recognition, thus to create a feasible system to simulate the speech recognition. The research work and achievements are as following: First: The author has done a lot of i
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6

SIQUEIRA, JAN KRUEGER. "CONTINUOUS SPEECH RECOGNITION WITH MFCC, SSCH AND PNCC FEATURES, WAVELET DENOISING AND NEURAL NETWORKS." PONTIFÍCIA UNIVERSIDADE CATÓLICA DO RIO DE JANEIRO, 2011. http://www.maxwell.vrac.puc-rio.br/Busca_etds.php?strSecao=resultado&nrSeq=19143@1.

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CONSELHO NACIONAL DE DESENVOLVIMENTO CIENTÍFICO E TECNOLÓGICO<br>Um dos maiores desafios na área de reconhecimento de voz contínua é desenvolver sistemas robustos ao ruído aditivo. Para isso, este trabalho analisa e testa três técnicas. A primeira delas é a extração de atributos do sinal de voz usando os métodos MFCC, SSCH e PNCC. A segunda é a remoção de ruído do sinal de voz via wavelet denoising. A terceira e última é uma proposta original batizada de feature denoising, que busca melhorar os atributos extraídos usando um conjunto de redes neurais. Embora algumas dessas técnicas já sejam con
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7

Dobrovolskis, Martynas. "Šnekos atpažinimas." Master's thesis, Lithuanian Academic Libraries Network (LABT), 2005. http://vddb.library.lt/obj/LT-eLABa-0001:E.02~2005~D_20050614_154005-58155.

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Voice recognition technologies appeared in the period of general device miniaturization, when all technologies were commonly integrated into one lust. There is no space for buttons and displays anymore. To have a good system of Lithuanian language recognition, a number of throughout researches must be implemented. Only after selecting the most efficient speech recognition scheme, we can proceed to the development of software adapted to the contemporary time. The aim of this paper is to determine, how efficient speech recognition is possible using neuron networks. MFCC and LPC coefficients were
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8

Julien, Eric. "Alignement du chant par rapport à une référence audio en temps réel." Mémoire, Université de Sherbrooke, 2013. http://hdl.handle.net/11143/6184.

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Dans l'optique de créer un système de karaoké qui modifie une interprétation chantée à capella en temps réel, il est nécessaire de pouvoir localiser l'interprète par rapport à une référence afin de pouvoir déterminer quelle serait la cible d'un algorithme de modification de la voix. Pour qu'un tel système fonctionne bien, il est nécessaire que l'algorithme d'alignement exploite au maximum les spécificités de la voix, qu'il utilise l'information liée au texte prononcé plutôt qu'aux aspects artistiques du chant, qu'il soit à temps réel et qu'il offr la plus faible latence possible. Afin d'attein
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9

Martins, Ana Caroline Vasconcelos. "GluA2 - Glutamatergic Receptor Study: A Molecular Approach." reponame:Repositório Institucional da UFC, 2017. http://www.repositorio.ufc.br/handle/riufc/28258.

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Submitted by José Orlando Soares de Oliveira (orlando.soares@bol.com.br) on 2017-11-30T12:23:47Z No. of bitstreams: 1 2017_tese_acvmartins.pdf: 10270409 bytes, checksum: f2b0eb40db54875e0e40a6d040ce7336 (MD5)<br>Rejected by Weslayne Nunes de Sales (weslaynesales@ufc.br), reason: A aluna optou por publicar apenas os elementos pré-textuais. on 2017-12-01T12:36:51Z (GMT)<br>Submitted by José Orlando Soares de Oliveira (orlando.soares@bol.com.br) on 2017-12-01T13:50:35Z No. of bitstreams: 1 Tese corrigida - elementos pretextuais.pdf: 159585 bytes, checksum: 9531b29bc8c5a46f5ed5753442df383f (M
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Vrba, Václav. "Robustní detekce klíčových slov v řečovém signálu." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2014. http://www.nusl.cz/ntk/nusl-220670.

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The master thesis is divided into two parts theoretical and practical. The theoretical part is focused on methods of analysis and detection of speech signals. In the practical part the system for isolated word recognition was created in Matlab. The system is speaker independent separately for men and women. Also two speech databases were created for further use in the aircraft cockpit. Tests and evaluations were performed even with added noise.
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11

SILVA, HARRY ARNOLD ANACLETO. "INDEPENDENT TEXT ROBUST SPEAKER RECOGNITION IN THE PRESENCE OF NOISE USING PAC-MFCC AND SUB BAND CLASSIFIERS." PONTIFÍCIA UNIVERSIDADE CATÓLICA DO RIO DE JANEIRO, 2011. http://www.maxwell.vrac.puc-rio.br/Busca_etds.php?strSecao=resultado&nrSeq=18212@1.

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COORDENAÇÃO DE APERFEIÇOAMENTO DO PESSOAL DE ENSINO SUPERIOR<br>O presente trabalho é proposto o atributo PAC-MFCC operando com Classificadores em Sub-Bandas para a tarefa de identificação de locutor independente do texto em ruído. O sistema proposto é comparado com os atributos MFCC (Coeficientes Cepestrais de Frequência Mel), PAC- MFCC (Fase Autocorrelação-MFCC ) sem uso de classificadores em sub-bandas, SSCH(Histogramas de Centróides de Sub-Bandas Espectrais) e TECC (Coeficientes Cepestrais da Energia Teager). Nesta tarefa de reconhecimento, utilizou-se a base TIMIT a qual é composta de 630
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12

Anifowose, Olakunle. "DESIGN OF A KEYWORD SPOTTING SYSTEM USING MODIFIED CROSS-CORRELATION IN THE TIME AND THE MFCC DOMAIN." Master's thesis, Temple University Libraries, 2012. http://cdm16002.contentdm.oclc.org/cdm/ref/collection/p245801coll10/id/205117.

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Electrical Engineering<br>M.S.E.E.<br>Abstract A Keyword Spotting System (KWS) is a system that recognizes predefined keywords in spoken utterances or written documents. The objective is to obtain the highest possible keyword detection rate without increasing the number of false detections in a system. The common approach to keyword spotting is the use of a Hidden Markov Model (HMM). These are usually complex systems which require training speech data. The Typical HMM approach uses garbage templates or HMM models to match non-keyword speech and non-speech sounds. The purpose of this research i
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13

GORDILLO, CHRISTIAN DAYAN ARCOS. "CONTINUOUS SPEECH RECOGNITION BY COMBINING MFCC AND PNCC ATTRIBUTES WITH SS, WD, MAP AND FRN METHODS OF ROBUSTNESS." PONTIFÍCIA UNIVERSIDADE CATÓLICA DO RIO DE JANEIRO, 2013. http://www.maxwell.vrac.puc-rio.br/Busca_etds.php?strSecao=resultado&nrSeq=23090@1.

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PONTIFÍCIA UNIVERSIDADE CATÓLICA DO RIO DE JANEIRO<br>COORDENAÇÃO DE APERFEIÇOAMENTO DO PESSOAL DE ENSINO SUPERIOR<br>PROGRAMA DE EXCELENCIA ACADEMICA<br>O crescente interesse por imitar o modelo que rege o processo cotidiano de comunicação humana através de maquinas tem se convertido em uma das áreas do conhecimento mais pesquisadas e de grande importância nas ultimas décadas. Esta área da tecnologia, conhecida como reconhecimento de voz, em como principal desafio desenvolver sistemas robustos que diminuam o ruído aditivo dos ambientes de onde o sinal de voz é adquirido, antes de que se esse
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14

Al-Ali, Ahmed Kamil Hasan. "Forensic speaker recognition under adverse conditions." Thesis, Queensland University of Technology, 2019. https://eprints.qut.edu.au/130783/1/Ahmed%20Kamil%20Hasan_Al-Ali_Thesis.pdf.

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The performance of forensic speaker recognition systems degrades significantly in the presence of environmental noise and reverberant conditions. This research developed new techniques to improve forensic speaker recognition performance under these conditions using fusion feature extraction techniques and speech enhancement based on the independent component analysis algorithm. A range of forensic speaker recognition applications will benefit from the research outcomes including criminal investigations and law enforcement agencies.
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15

Viana, Hesdras Oliveira. "Descritor de voz invariante ao ruído." Universidade Federal de Pernambuco, 2013. https://repositorio.ufpe.br/handle/123456789/11842.

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Submitted by João Arthur Martins (joao.arthur@ufpe.br) on 2015-03-10T19:07:24Z No. of bitstreams: 2 Dissertaçao Hesdras Viana.pdf: 2998238 bytes, checksum: de42b675472ac4632a3a3c04688a77d5 (MD5) license_rdf: 1232 bytes, checksum: 66e71c371cc565284e70f40736c94386 (MD5)<br>Approved for entry into archive by Daniella Sodre (daniella.sodre@ufpe.br) on 2015-03-10T19:43:06Z (GMT) No. of bitstreams: 2 Dissertaçao Hesdras Viana.pdf: 2998238 bytes, checksum: de42b675472ac4632a3a3c04688a77d5 (MD5) license_rdf: 1232 bytes, checksum: 66e71c371cc565284e70f40736c94386 (MD5)<br>Made available in DSpace on 20
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Erokyar, Hasan. "Age and Gender Recognition for Speech Applications based on Support Vector Machines." Scholar Commons, 2014. https://scholarcommons.usf.edu/etd/5356.

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Automatic age and gender recognition for speech applications is very important for a number of reasons. One of the reasons is that it can improve human-machine interaction. For example, the advertisements can be specialized based on the age and the gender of the person on the phone. It also can help identify suspects in criminal cases or at least it can minimize the number of suspects. Some other uses of this system can be applied for adaptation of waiting queue music where a different type of music can be played according to the person's age and gender. And also using this age and gender reco
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17

Barbosa, Emmanuel Duarte. "Descri??o bioqu?mica qu?ntica do bols?o de intera??o do ?ON Zn2+ na enzima ALAD humana." PROGRAMA DE P?S-GRADUA??O EM BIOQU?MICA, 2016. https://repositorio.ufrn.br/jspui/handle/123456789/21908.

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Submitted by Automa??o e Estat?stica (sst@bczm.ufrn.br) on 2017-02-02T13:30:50Z No. of bitstreams: 1 EmmanuelDuarteBarbosa_DISSERT.pdf: 9706329 bytes, checksum: cf979f942793c968afbd04719854d7f0 (MD5)<br>Approved for entry into archive by Arlan Eloi Leite Silva (eloihistoriador@yahoo.com.br) on 2017-02-08T19:26:36Z (GMT) No. of bitstreams: 1 EmmanuelDuarteBarbosa_DISSERT.pdf: 9706329 bytes, checksum: cf979f942793c968afbd04719854d7f0 (MD5)<br>Made available in DSpace on 2017-02-08T19:26:36Z (GMT). No. of bitstreams: 1 EmmanuelDuarteBarbosa_DISSERT.pdf: 9706329 bytes, checksum: cf979f942793c9
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Manso, Dalila Nascimento. "An?lise molecular da muta??o HIS275TIR isolada na Neuraminidase do H1N1 resistente ao oseltamivir." PROGRAMA DE P?S-GRADUA??O EM CI?NCIAS BIOL?GICAS, 2017. https://repositorio.ufrn.br/jspui/handle/123456789/24058.

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Submitted by Automa??o e Estat?stica (sst@bczm.ufrn.br) on 2017-10-04T22:23:59Z No. of bitstreams: 1 DalilaNascimentoManso_DISSERT.pdf: 1914411 bytes, checksum: 966fc442e252d656c3946bff697a75f5 (MD5)<br>Approved for entry into archive by Arlan Eloi Leite Silva (eloihistoriador@yahoo.com.br) on 2017-10-13T21:33:08Z (GMT) No. of bitstreams: 1 DalilaNascimentoManso_DISSERT.pdf: 1914411 bytes, checksum: 966fc442e252d656c3946bff697a75f5 (MD5)<br>Made available in DSpace on 2017-10-13T21:33:08Z (GMT). No. of bitstreams: 1 DalilaNascimentoManso_DISSERT.pdf: 1914411 bytes, checksum: 966fc442e252d6
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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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Matos, Adriano Nogueira. "Extração de características do sinal de voz utilizando análise fatorial verdadeira." Universidade Federal do Amazonas, 2008. http://tede.ufam.edu.br/handle/tede/2959.

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Made available in DSpace on 2015-04-11T14:03:17Z (GMT). No. of bitstreams: 1 DISSERTACAO ADRIANO NOGUEIRA.pdf: 382280 bytes, checksum: fc1f9e0caac3d97ff74a893e97298a71 (MD5) Previous issue date: 2008-12-17<br>Coordenação de Aperfeiçoamento de Pessoal de Nível Superior<br>Digital processing of speech signal is applied in several computer applications, which the major ones are the following: Recognition, synthesis and coding of speech. All these applications require the amount of data in the acoustic signal to be reduced, in order to allow processing by a computer device. The feature extract
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Abraham, Aby. "Continous Speech Recognition Using Long Term Memory Cells." Ohio University / OhioLINK, 2013. http://rave.ohiolink.edu/etdc/view?acc_num=ohiou1377777011.

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22

Li, Yi. "Speaker Diarization System for Call-center data." Thesis, KTH, Skolan för elektroteknik och datavetenskap (EECS), 2020. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-286677.

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To answer the question who spoke when, speaker diarization (SD) is a critical step for many speech applications in practice. The task of our project is building a MFCC-vector based speaker diarization system on top of a speaker verification system (SV), which is an existing Call-centers application to check the customer’s identity from a phone call. Our speaker diarization system uses 13-Dimensional MFCCs as Features, performs Voice Active Detection (VAD), segmentation, Linear Clustering and the Hierarchical Clustering based on GMM and the BIC score. By applying it, we decrease the Equal Error
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23

Čermák, Jan. "Rozpoznávání emočních stavů na základě analýzy řečového signálu." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2009. http://www.nusl.cz/ntk/nusl-218162.

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The thesis is focused on the emotional states classification in the Matlab program, using neural networks and the classifier which is based on a combination of Gaussian density functions. It deals with the speech signal processing; the prosodic and spectral signs and the MFCC coefficients were extracted from the signal. The work also deals with the quality evaluation of individual signs of which the most suitable were chosen in order to provide the correct classification of emotional states. In order to identify the emotional states, two different methods were used. The first method of classif
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Vianna, J?ssica de F?tima. "Bioqu?mica qu?ntica da capreomicina e da estreptomicina em complexo com o ribossomo bacteriano." PROGRAMA DE P?S-GRADUA??O EM CI?NCIAS BIOL?GICAS, 2017. https://repositorio.ufrn.br/jspui/handle/123456789/22614.

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Submitted by Automa??o e Estat?stica (sst@bczm.ufrn.br) on 2017-04-03T22:31:54Z No. of bitstreams: 1 JessicaDeFatimaVianna_DISSERT.pdf: 3724208 bytes, checksum: f7d62fbcd54bf6b212f2003b461810c5 (MD5)<br>Approved for entry into archive by Arlan Eloi Leite Silva (eloihistoriador@yahoo.com.br) on 2017-04-11T18:14:29Z (GMT) No. of bitstreams: 1 JessicaDeFatimaVianna_DISSERT.pdf: 3724208 bytes, checksum: f7d62fbcd54bf6b212f2003b461810c5 (MD5)<br>Made available in DSpace on 2017-04-11T18:14:29Z (GMT). No. of bitstreams: 1 JessicaDeFatimaVianna_DISSERT.pdf: 3724208 bytes, checksum: f7d62fbcd54bf6
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Káčerová, Erika. "Odhad formantových kmitočtů pomocí strojového učení." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2019. http://www.nusl.cz/ntk/nusl-400852.

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This Master's thesis deals with the issue of formant extraction. A system of scripts in Matlab interface is created to generate values of the first three formant frequencies from speech recordings with the use of Praat and Snack(WaveSurfer). Mel Frequency Cepstral Coefficients and Linear Predictive Coefficients are extracted from the audio files in order to be added to the database. This database is then used to train a neural network. Finally, the designed neural network is tested.
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Dobrotka, Matúš. "Detekce Akustického Prostředí z Řeči." Master's thesis, Vysoké učení technické v Brně. Fakulta informačních technologií, 2018. http://www.nusl.cz/ntk/nusl-385945.

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The topic of this thesis is an audio recording classification with 15 different acoustic scene classes that represent common scenes and places where people are situated on a regular basis. The thesis describes 2 approaches based on GMM and i-vectors and a fusion of the both approaches. The score of the best GMM system which was evaluated on the evaluation dataset of the DCASE Challenge is 60.4%. The best i-vector system's score is 68.4%. The fusion of the GMM system and the best i-vector system achieves score of 69.3%, which would lead to the 20th place in the all systems ranking of the DCASE
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Lima, Neto Jos? Xavier de. "Bioqu?mica qu?ntica na diferencia??o dos n?veis de ativa??o de receptores AMPA por agonistas parciais Wilardina." Universidade Federal do Rio Grande do Norte, 2015. http://repositorio.ufrn.br/handle/123456789/19861.

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Bastas, Selin A. "Nocturnal Bird Call Recognition System for Wind Farm Applications." University of Toledo / OhioLINK, 2012. http://rave.ohiolink.edu/etdc/view?acc_num=toledo1325803309.

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Duarte, Dami Doria Narayana. "Um estudo da relevância da dinâmica espectral na classificação de sons domésticos." Universidade Federal de Sergipe, 2016. https://ri.ufs.br/handle/riufs/5021.

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Conselho Nacional de Pesquisa e Desenvolvimento Científico e Tecnológico - CNPq<br>This work presents a study of the spectral dynamics characteristics of audio signals. More specifically, we aim at detecting regularities that can be modeled in typical domestic sounds, in order to classify them. Our starting point is the work of Sehili et al. [2], in which a household sounds classification system based on GMM is proposed. The Sehili system is reproduced in this work as a baseline system. Following the same protocol of experiments, a 73 % recognition rate is achieved. Afterwards, three sets
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Duarte, Dami Doria Narayana. "Um estudo da relevância da dinâmica espectral na classificação de sons doméstic." Universidade Federal de Sergipe, 2016. http://ri.ufs.br:8080/xmlui/handle/123456789/5021.

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Conselho Nacional de Pesquisa e Desenvolvimento Científico e Tecnológico - CNPq<br>This work presents a study of the spectral dynamics characteristics of audio signals. More specifically, we aim at detecting regularities that can be modeled in typical domestic sounds, in order to classify them. Our starting point is the work of Sehili et al. [2], in which a household sounds classification system based on GMM is proposed. The Sehili system is reproduced in this work as a baseline system. Following the same protocol of experiments, a 73 % recognition rate is achieved. Afterwards, three sets
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Ali, Ahmed Mohamed Abdel Maksoud. "Multi-dialect Arabic broadcast speech recognition." Thesis, University of Edinburgh, 2018. http://hdl.handle.net/1842/31224.

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Dialectal Arabic speech research suffers from the lack of labelled resources and standardised orthography. There are three main challenges in dialectal Arabic speech recognition: (i) finding labelled dialectal Arabic speech data, (ii) training robust dialectal speech recognition models from limited labelled data and (iii) evaluating speech recognition for dialects with no orthographic rules. This thesis is concerned with the following three contributions: Arabic Dialect Identification: We are mainly dealing with Arabic speech without prior knowledge of the spoken dialect. Arabic dialects could
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Kotulek, Milan. "Jednoduchý textově nezávislý hlasový zámek - Softwarový systém pro verifikaci mluvčích." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2015. http://www.nusl.cz/ntk/nusl-221256.

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A brief introduction into biometrics is described in this thesis leading to description and to design a solution of verification system using speech analysis. The designed system provides firstly basic signal processing, then vowel recognition in fluent Czech speech. For each found vowel, observed speech features are calculated. The created GUI application was tested on created speaker database and its efficiency is approximately 54 % for short testing utterances, and approx. 88 % for long testing utterances respectively.
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Costa, Roner Ferreira da. "BioquÃmica quÃntica das estatinas, aspirina e anti-hipertensivos." Universidade Federal do CearÃ, 2011. http://www.teses.ufc.br/tde_busca/arquivo.php?codArquivo=6234.

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Conselho Nacional de Desenvolvimento CientÃfico e TecnolÃgico<br>As doenÃas cardiovasculares (CVDs) compreendem um amplo espectro de doenÃas do coraÃÃo e vasos sanguÃneos (artÃrias e veias), entre as quais se incluem a doenÃa das artÃrias coronÃrias, o ataque cardÃaco, a angina, a sÃndrome coronariana aguda, o aneurisma da aorta, arritmias cardÃacas, a doenÃa cardÃaca congÃnita, a insuficiÃncia cardÃaca e a doenÃa cardÃaca reumÃtica. Entre os principias fÃrmacos que tratam as doenÃas cardiovasculares estÃo: (i) as estatinas, que atuam inibindo a 3-hidroxi-3-metilgluratil coenzima A (HMG-CoA) r
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Costa, Roner Ferreira da. "Bioquímica quântica das estatinas, aspirina e anti-hipertensivos." reponame:Repositório Institucional da UFC, 2011. http://www.repositorio.ufc.br/handle/riufc/12543.

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COSTA, Roner Ferreira da. Bioquímica quântica das estatinas, aspirina e anti-hipertensivos. 2011. 185 f. Tese (Doutorado em Física) - Programa de Pós-Graduação em Física, Departamento de Física, Centro de Ciências, Universidade Federal do Ceará, Fortaleza, 2011.<br>Submitted by Edvander Pires (edvanderpires@gmail.com) on 2015-05-29T22:17:20Z No. of bitstreams: 1 2011_tese_rfcosta.pdf: 5384677 bytes, checksum: b7096c8a3fe046f09eec5640166b7cba (MD5)<br>Approved for entry into archive by Edvander Pires(edvanderpires@gmail.com) on 2015-05-29T22:18:27Z (GMT) No. of bitstreams: 1 2011_tese_rfcosta.p
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Kryške, Lukáš. "Rozpoznávání řeči s pomocí nástroje Sphinx-4." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2014. http://www.nusl.cz/ntk/nusl-220655.

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This diploma thesis is aimed to find an effective method for continuous speech recognition. To be more accurate, it uses speech-to-text recognition for a keyword spotting discipline. This solution is able to be applicable for phone calls analysis or for a similar application. Most of the diploma thesis describes and implements speech recognition framework Sphinx-4 which uses Hidden Markov models (HMM) to define a language acoustic models. It is explained how these models can be trained for a new language or for a new language dialect. Finally there is in detail described how to implement the k
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Karlsson, David. "Ljudklassificering med Tensorflow och IOT-enheter : En teknisk studie." Thesis, Mittuniversitetet, Institutionen för informationssystem och –teknologi, 2020. http://urn.kb.se/resolve?urn=urn:nbn:se:miun:diva-39331.

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Artificial Inteligens and machine learning has started to get established as reco- gnizable terms to the general masses in their daily lives. Applications such as voice recognicion and image recognicion are used widely in mobile phones and autonomous systems such as self-drivning cars. This study examines how one can utilize this technique to classify sound as a complement to videosurveillan- ce in different settings, for example a busstation or other areas that might need monitoring. To be able to do this a technique called Convolution Neural Ne- twork has been used since this is a popular ar
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Li, Ke. "Analysis of Energy losses of Microbial Fuel Cells (MFCs) and Design of an Innovative Constructed Wetlands-MFC." The Ohio State University, 2017. http://rave.ohiolink.edu/etdc/view?acc_num=osu1500604673955179.

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Campos, Victor de Abreu [UNESP]. "Arcabouço para reconhecimento de locutor baseado em aprendizado não supervisionado." Universidade Estadual Paulista (UNESP), 2017. http://hdl.handle.net/11449/151725.

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Submitted by Victor de Abreu Campos null (victorde.ac@gmail.com) on 2017-09-27T02:41:28Z No. of bitstreams: 1 dissertacao.pdf: 5473435 bytes, checksum: 1e76ecc15a4499dc141983740cc79e5a (MD5)<br>Approved for entry into archive by Monique Sasaki (sayumi_sasaki@hotmail.com) on 2017-09-28T13:43:21Z (GMT) No. of bitstreams: 1 campos_va_me_sjrp.pdf: 5473435 bytes, checksum: 1e76ecc15a4499dc141983740cc79e5a (MD5)<br>Made available in DSpace on 2017-09-28T13:43:21Z (GMT). No. of bitstreams: 1 campos_va_me_sjrp.pdf: 5473435 bytes, checksum: 1e76ecc15a4499dc141983740cc79e5a (MD5) Previous issue date:
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Urbiš, Oldřich. "Algoritmy rozpoznávání řeči na FPGA/DSP." Master's thesis, Vysoké učení technické v Brně. Fakulta informačních technologií, 2008. http://www.nusl.cz/ntk/nusl-235943.

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This master's thesis deals with design of speech recognition algorithms with consideration of target technology, which is platform combinating digital signal processing and field programmable gate array. Algorithms for speech recognition includes: feature extraction of Melfrequency cepstral coefficients, hidden Markov models and their evaluation by Viterbi algorithm.
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Židlík, Pavel. "Počítačová analýza sportovních zápasů." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2009. http://www.nusl.cz/ntk/nusl-218104.

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This work deals with the possibility of a fast football match analysis from audio part of record with the possibility of implementation of some methods for other than football matches as well. The first intention was concentrated on detection of whiz of the soccer whistle that has specific frequency in its specter, which is out of common speech frequency. After detection harmonic frequency , the attention was focused on the definition of whiz meaning. Referee was helpful with the issue as he informed me about the number of whiz styles and provided me with referential samples for whiz classific
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Pelikán, Pavel. "Určení výšky osob z řečového projevu." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2013. http://www.nusl.cz/ntk/nusl-220197.

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Diploma’s thesis is focused on determining person’s height from spoken utterance. First part of the work evaluates present situation and refers to the published studies. Knowledge gained in these studies was used in this thesis. Study with the best results according to estimated height of the speakers was chosen. The experiment realized in the chosen study was performed in this work. The system for the estimation of the height of the speakers based on the speech signal was created. This system was successfully tested by using several acoustic features on spoken utterances from TIMIT database.
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Almeida, Christiane Raulino. "Extratores de características acústicas inspirados no sistema periférico auditivo." Universidade Federal de Sergipe, 2014. http://ri.ufs.br:8080/xmlui/handle/123456789/5014.

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Extracting information from acoustic signals is a common task in signal processing and pattern recognition. Broadly speaking, the processing system has, as initial task, to obtain a low-dimensional representation of the acoustic signal, extracted trough computational methods called feature extractors. This representation aims to present the sound of speech in a more convenient form to extract the information contained in the signal. Considering the initial task of processing systems, this work presents a detailed study of three classic methods for features extracting, namely: the Mel - Frequen
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Ujihara, Rintaro. "Multi-objective optimization for model selection in music classification." Thesis, KTH, Optimeringslära och systemteori, 2021. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-298370.

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With the breakthrough of machine learning techniques, the research concerning music emotion classification has been getting notable progress combining various audio features and state-of-the-art machine learning models. Still, it is known that the way to preprocess music samples and to choose which machine classification algorithm to use depends on data sets and the objective of each project work. The collaborating company of this thesis, Ichigoichie AB, is currently developing a system to categorize music data into positive/negative classes. To enhance the accuracy of the existing system, thi
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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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Грушко, Ярослав Володимирович. "Система голосової біометрії, економна до обчислювальних ресурсів". Master's thesis, КПІ ім. Ігоря Сікорського, 2019. https://ela.kpi.ua/handle/123456789/32176.

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Мета даної роботи – створити економну до обчислювальних ресурсів систему голосової біометрії. Основною ціллю роботи стали побудова загальної схеми такої системи, визначення її компонент та оптимальних параметрів. Об’єктом дослідження даної магістерської дипломної роботи є розпізнавання голосу людини комп’ютером. Предмет дослідження – голосова біометрія, тобто голосове розпізнавання особи. Спроєктована система складається з трьох основних модулів. Перший модуль – це алгоритм отримання голосового відбитку MFCCs. Другий модуль – це класифікатор, який має навчатися голосовими відбитками отрима
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Odehnal, Jiří. "Řízení a měření sportovních drilů hlasem/zvuky." Master's thesis, Vysoké učení technické v Brně. Fakulta informačních technologií, 2019. http://www.nusl.cz/ntk/nusl-399705.

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This master's thesis deals with the design and development of mobile aplication for Android platform. The aim of the work is to implement a simple and user-friendly user interface that would support and assist the user in trainning and sport exercises. The thesis also include implementation of sound detection to support during exercises and voice instruction by application. In practice the application should help in making training exercises more comfortable without the user being forced to keep mobile device in hand.
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Hacine-Gharbi, Abdenour. "Sélection de paramètres acoustiques pertinents pour la reconnaissance de la parole." Phd thesis, Université d'Orléans, 2012. http://tel.archives-ouvertes.fr/tel-00843652.

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L'objectif de cette thèse est de proposer des solutions et améliorations de performance à certains problèmes de sélection des paramètres acoustiques pertinents dans le cadre de la reconnaissance de la parole. Ainsi, notre première contribution consiste à proposer une nouvelle méthode de sélection de paramètres pertinents fondée sur un développement exact de la redondance entre une caractéristique et les caractéristiques précédemment sélectionnées par un algorithme de recherche séquentielle ascendante. Le problème de l'estimation des densités de probabilités d'ordre supérieur est résolu par la
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Houdek, Miroslav. "Rozpoznání emočního stavu člověka z řeči." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2009. http://www.nusl.cz/ntk/nusl-218117.

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This master thesis concerns with emotional states and gender recognition on the basis of speech signal analysis. We used various prosodic and cepstral features for the description of the speech signal. In the text we describe non-invasive methods for glottal pulses estimation. The described features of speech were implemented in MATLAB. For their classification we used the GMM classifier, which uses the Gaussian probability distribution for modeling a feature space. Furthermore, we constructed a system for recognition of emotional states of the speaker and a system for gender recognition from
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Evelyn. "Mediator combined gaseous substrate for electricity generation in microbial fuel cells (MFCs) and potential integration of a MFC into an anaerobic biofiltration system." Thesis, University of Canterbury. Department of Chemical ad Process Engineering, 2013. http://hdl.handle.net/10092/10733.

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Microbial fuel cells (MFCs) are emerging energy production technology which converts the chemical energy stored in biologically degradable compounds to electricity at high efficiencies. Microbial fuel cells have some advantages such as use of an inexpensive catalyst, operate under mild reaction conditions (i.e. ambient temperature, normal pressure and neutral pH), and generate power from a wide range and cheap raw materials. These make microbial fuel cell as an attractive alternative over other electricity generating devices. However, so far the major prob
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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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