Página 1 dos resultados de 1106 itens digitais encontrados em 0.014 segundos

Características fonoarticulatórias na doença de Parkinson de início na meia idade e tardio; Speech and voice characteristics in middle age and late onset Parkinson's disease

Dias, Alice Estevo
Fonte: Biblioteca Digitais de Teses e Dissertações da USP Publicador: Biblioteca Digitais de Teses e Dissertações da USP
Tipo: Tese de Doutorado Formato: application/pdf
Publicado em 15/08/2006 Português
Relevância na Pesquisa
36.35%
Alterações fonoarticulatórias caracterizam a disartria hipocinética e podem ocorrer ao longo da evolução da doença de Parkinson (DP). No entanto, não existem estudos que evidenciem a influência da idade nessas alterações. Objetivo: Comparar e correlacionar selecionadas características fonoarticulatórias em pacientes com DP de início na meia idade e tardio. Método: Participaram 50 pacientes que constituíram dois grupos. O Grupo I foi composto por 30 (60%) pacientes com idade de início da DP entre 40 e 55 anos e o Grupo II, por 20 (40%) pacientes com início da doença após os 65 anos, ambos com a duração da doença variando de 2 a 18 anos. Todos foram submetidos à avaliação neurológica a partir da Parte III da Escala Unificada para a Doença de Parkinson (UPDRS) e Escala Modificada de Hoehn & Yahr e, fonoaudiológica, realizada por meio de análise perceptivo-auditiva (velocidade, inteligibilidade e tipo articulatório da fala e qualidade da voz) e acústica computadorizada (freqüência fundamental e intensidade da voz). Resultados: Não houve diferença estatisticamente significativa entre os dois grupos no que diz respeito ao estágio da doença, aos escores da escala UPDRS e às análises fonoaudiológicas. As análises de correlação não mostraram diferença estatisticamente significativa entre a qualidade...

Estatística multivariada aplicada no correlacionamento da qualidade de serviços em chamadas VOIP e a qualidade da fala aferida pela recomendação ITU-T G.107.; Multivariate analysis applied in correlating quality of services in VoIP calls and speech quality by ITU-T G.107 recommendation.

Alencar, Sérgio Costa Martins de
Fonte: Biblioteca Digitais de Teses e Dissertações da USP Publicador: Biblioteca Digitais de Teses e Dissertações da USP
Tipo: Dissertação de Mestrado Formato: application/pdf
Publicado em 06/10/2011 Português
Relevância na Pesquisa
66.31%
Vivemos atualmente uma era de convergência de tecnologias, motivada por questões tanto econômicas como de caráter operacional, na qual os serviços de dados, voz e vídeo estão migrando rapidamente para uma plataforma IP. Particularmente considerando o paradigma da telefonia IP neste processo de convergência, ocorrem desafios tecnológicos, pois temos de um lado os usuários finais que já possuem uma referência sobre a qualidade da fala, fruto das décadas de uso do sistema telefônico tradicional, e na outra extremidade as operadoras de telecomunicações que em sua última milha dependem de redes estatísticas, sem mecanismos adequados para a garantia de QoS. Assim, se torna vital para o sucesso da operação a devida identificação das relações entre os diversos componentes existentes entre terminais e sua contribuição para a qualidade de fala, percebida pelo assinante, de forma a entregar um serviço com qualidade similar ao Sistema Telefônico Fixo Comutado. Neste contexto, este trabalho busca identificar por meio de técnicas de estatística multivariada uma correlação entre métricas objetivas de Qualidade de Serviços aplicáveis em redes IP e a qualidade subjetiva da fala predita pelo algoritmo Modelo-E definido na recomendação ITU-T G.107. Um método de coleta e análise estatística de informações foi desenvolvido para atingir o objetivo proposto. Para sua validação um ambiente de testes foi criado...

Comparative Analysis of Objective Distortion Measures for Speech Signals Degraded by Noise

Vieira Filho, Jozué ; Faria, Giovani César de
Fonte: Institute of Electrical and Electronics Engineers (IEEE) Publicador: Institute of Electrical and Electronics Engineers (IEEE)
Tipo: Conferência ou Objeto de Conferência Formato: 978-983
Português
Relevância na Pesquisa
46.29%
Speech signals degraded by additive noise can affects different applications in telecommunication. The noise may degrades the intelligibility of the speech signals and its waveforms as well. In some applications such as speech coding, both intelligibility and waveform quality are important but only intelligibility has been focused lastly. So, modern speech quality measurement techniques such as PESQ (Perceptual Evaluation of Speech Quality) have been used and classical distortion measurement techniques such as Cepstral Distance are becoming unused. In this paper it is shown that some classical distortion measures are still important in applications where speech corrupted by additive noise has to be evaluated.

An algorithm that improves speech intelligibility in noise for normal-hearing listeners

Kim, Gibak; Lu, Yang; Hu, Yi; Loizou, Philipos C.
Fonte: Acoustical Society of America Publicador: Acoustical Society of America
Tipo: Artigo de Revista Científica
Publicado em /09/2009 Português
Relevância na Pesquisa
46.09%
Traditional noise-suppression algorithms have been shown to improve speech quality, but not speech intelligibility. Motivated by prior intelligibility studies of speech synthesized using the ideal binary mask, an algorithm is proposed that decomposes the input signal into time-frequency (T-F) units and makes binary decisions, based on a Bayesian classifier, as to whether each T-F unit is dominated by the target or the masker. Speech corrupted at low signal-to-noise ratio (SNR) levels (−5 and 0 dB) using different types of maskers is synthesized by this algorithm and presented to normal-hearing listeners for identification. Results indicated substantial improvements in intelligibility (over 60% points in −5 dB babble) over that attained by human listeners with unprocessed stimuli. The findings from this study suggest that algorithms that can estimate reliably the SNR in each T-F unit can improve speech intelligibility.

Adaptive Redundant Speech Transmission over Wireless Multimedia Sensor Networks Based on Estimation of Perceived Speech Quality

Kang, Jin Ah; Kim, Hong Kook
Fonte: Molecular Diversity Preservation International (MDPI) Publicador: Molecular Diversity Preservation International (MDPI)
Tipo: Artigo de Revista Científica
Publicado em 31/08/2011 Português
Relevância na Pesquisa
46.36%
An adaptive redundant speech transmission (ARST) approach to improve the perceived speech quality (PSQ) of speech streaming applications over wireless multimedia sensor networks (WMSNs) is proposed in this paper. The proposed approach estimates the PSQ as well as the packet loss rate (PLR) from the received speech data. Subsequently, it decides whether the transmission of redundant speech data (RSD) is required in order to assist a speech decoder to reconstruct lost speech signals for high PLRs. According to the decision, the proposed ARST approach controls the RSD transmission, then it optimizes the bitrate of speech coding to encode the current speech data (CSD) and RSD bitstream in order to maintain the speech quality under packet loss conditions. The effectiveness of the proposed ARST approach is then demonstrated using the adaptive multirate-narrowband (AMR-NB) speech codec and ITU-T Recommendation P.563 as a scalable speech codec and the PSQ estimation, respectively. It is shown from the experiments that a speech streaming application employing the proposed ARST approach significantly improves speech quality under packet loss conditions in WMSNs.

Nonlinear Frequency Compression: Effects on Sound Quality Ratings of Speech and Music

Parsa, Vijay; Scollie, Susan; Glista, Danielle; Seelisch, Andreas
Fonte: SAGE Publications Publicador: SAGE Publications
Tipo: Artigo de Revista Científica
Publicado em /03/2013 Português
Relevância na Pesquisa
36.31%
Frequency lowering technologies offer an alternative amplification solution for severe to profound high frequency hearing losses. While frequency lowering technologies may improve audibility of high frequency sounds, the very nature of this processing can affect the perceived sound quality. This article reports the results from two studies that investigated the impact of a nonlinear frequency compression (NFC) algorithm on perceived sound quality. In the first study, the cutoff frequency and compression ratio parameters of the NFC algorithm were varied, and their effect on the speech quality was measured subjectively with 12 normal hearing adults, 12 normal hearing children, 13 hearing impaired adults, and 9 hearing impaired children. In the second study, 12 normal hearing and 8 hearing impaired adult listeners rated the quality of speech in quiet, speech in noise, and music after processing with a different set of NFC parameters. Results showed that the cutoff frequency parameter had more impact on sound quality ratings than the compression ratio, and that the hearing impaired adults were more tolerant to increased frequency compression than normal hearing adults. No statistically significant differences were found in the sound quality ratings of speech-in-noise and music stimuli processed through various NFC settings by hearing impaired listeners. These findings suggest that there may be an acceptable range of NFC settings for hearing impaired individuals where sound quality is not adversely affected. These results may assist an Audiologist in clinical NFC hearing aid fittings for achieving a balance between high frequency audibility and sound quality.

Perceptual and acoustic impacts of aberrant properties of electrolaryngeal speech

Meltzner, Geoffrey S. (Geoffrey Seth), 1973-
Fonte: Massachusetts Institute of Technology Publicador: Massachusetts Institute of Technology
Tipo: Tese de Doutorado Formato: 171 p.; 12840980 bytes; 12840788 bytes; application/pdf; application/pdf
Português
Relevância na Pesquisa
36.42%
Advanced laryngeal cancer is often treated by surgical removal of the larynx (laryngectomy) thus rendering patients unable to produce normal voice and speech. Laryngectomy patients must rely on an alternative means of producing voice and speech, with the most common method being the use of an electrolarynx (EL). The EL is a small, hand-held, electromechanical device that acoustically excites the vocal tract when held against the neck or at the lips. While the EL provides a serviceable means of communication, the resulting speech has several shortcomings in terms of both intelligibility and speech quality. Previous studies have identified and tried to correct different single selected acoustic properties associated with the abnormal quality of EL speech, but with only limited success. There remains uncertainty about: 1) which components of the EL speech acoustic signal are contributing most to its abnormal quality and 2) what kinds of acoustic enhancements would be most effective in improving the quality of EL speech. Using a combination of listening experiments, acoustic analysis and acoustic modeling, this thesis investigated the perceptual and acoustic impacts of several aberrant properties of EL speech, with the overall goal of using the results to direct future EL speech improvement efforts. Perceptual experiments conducted by having 10 listeners judge the naturalness of differently enhanced versions of EL speech demonstrated that adding pitch information would produce the most benefit. Removing the EL self-noise and correcting for a lack of low frequency energy would also improve EL speech...

New single-ended objective measure for non-intrusive speech quality evaluation

Mahdi, Abdulhussain E; Picovici, Dorel
Fonte: Springer Publicador: Springer
Tipo: info:eu-repo/semantics/article; all_ul_research; ul_published_reviewed
Português
Relevância na Pesquisa
66.38%
peer-reviewed; This article proposes a new output-based method for non-intrusive assessment of speech quality of voice communication systems and evaluates its performance. The method requires access to the processed (degraded) speech only, and is based on measuring perception-motivated objective auditory distances between the voiced parts of the output speech to appropriately matching references extracted from a pre-formulated codebook. The codebook is formed by optimally clustering a large number of parametric speech vectors extracted from a database of clean speech records. The auditory distances are then mapped into objective Mean Opinion listening quality scores. An efficient data-mining tool known as the self-organizing map (SOM) achieves the required clustering and mapping/reference matching processes. In order to obtain a perception-based, speaker-independent parametric representation of the speech, three domain transformation techniques have been investigated. The first technique is based on a perceptual linear prediction (PLP) model, the second utilises a bark spectrum (BS) analysis and the third utilises mel-frequency cepstrum coefficients (MFCC). Reported evaluation results show that the proposed method provides high correlation with subjective listening quality scores...

Accuracy analysis on call quality assessments in voice over IP

Han, Yi; Fitzpatrick, John; Murphy, Liam; Dunne, Jonathan
Fonte: IEEE Computer Society Publicador: IEEE Computer Society
Tipo: info:eu-repo/semantics/conferenceObject; all_ul_research; ul_published_reviewed
Português
Relevância na Pesquisa
45.97%
peer-reviewed; Voice over IP (VoIP) now has tremendous influence on the telecommunication market with its flexibility and price advantage. Users of VoIP expect call quality to be as good as, if not better than the traditional Public Switched Telephone Network (PSTN). However in VoIP, factors that are related to the IP transport network such as packet loss, delay, bandwidth, jitter, and voice encoding (codec) all affect call quality. Call quality assessment in VoIP systems is mainly conducted with off-line tests using the Perceptual Evaluation of Speech Quality (PESQ) [1] methodology. Another method that can be utilised is an on-line approach using the E-Model, which can be used in real time. However, these two methods have limits and inaccuracy, and often do not give the same results. Call quality assessment is often used to adjust system and codec parameters. Therefore, given inaccurate results, the system would decrease the adjustment efficiency or even inadvertently decrease call quality. The primary contribution of this paper is a comparison between the accuracy of PESQ and the E-Model investigated by conducting an extensive set of experiments in a real enterprise network using a widely deployed Voice over IP (VoIP) product. Experiments were conducted under varying controlled network conditions. The results show that under various conditions...

Effects of the wireless channel, signal compression and network architecture on speech quality in VOIP networks

Nikolaos, Tiantioukas
Fonte: Monterey California. Naval Postgraduate School Publicador: Monterey California. Naval Postgraduate School
Tipo: Tese de Doutorado
Português
Relevância na Pesquisa
46.28%
Voice over Internet Protocol (VoIP) telephony is an emerging technology slowly finding its way into military applications. It provides several advantages over PSTN but comes short on performance, quality of service and availability. The purpose of this thesis is to measure the quality of voice in VoIP communications. More specifically it investigates the effects of wireless channel conditions as well as channel coding and compression on the received speech quality. Both simulation and experimentation are conducted using Matlab code and Speex software and across commercial VoIP networks. Simulation shows that fading channel parameters can heavily affect the quality of received speech. Speech compression results in bit rate gain, but, on the other hand, the signal becomes more sensitive to errors. The performance of an outdoor wireless network is better than that of an indoor network. The VoIP network architecture can affect the received speech quality on a long-distance connection.

A Comparison of Open-Source Segmentation Architectures for Dealing with Imperfect Data from the Media in Speech Synthesis

Gallardo-Antolín, Ascensión; Montero, Juan Manuel; King, Simon
Fonte: International Speech Communication Association Publicador: International Speech Communication Association
Tipo: info:eu-repo/semantics/publishedVersion; info:eu-repo/semantics/bookPart; info:eu-repo/semantics/conferenceObject
Publicado em //2014 Português
Relevância na Pesquisa
46.2%
Traditional Text-To-Speech (TTS) systems have been developed using especially-designed non-expressive scripted recordings. In order to develop a new generation of expressive TTS systems in the Simple4All project, real recordings from the media should be used for training new voices with a whole new range of speaking styles. However, for processing this more spontaneous material, the new systems must be able to deal with imperfect data (multi-speaker recordings, background and fore-ground music and noise), filtering out low-quality audio segments and creating mono-speaker clusters. In this paper we compare several architectures for combining speaker diarization and music and noise detection which improve the precision and overall quality of the segmentation.; This work has been carried out during the research stay of A. Gallardo-Antolín and J. M. Montero at the Centre for Speech Technology Research (CSTR), University of Edinburgh, supported by the Spanish Ministry of Education, Culture and Sports under the National Program of Human Resources Mobility from the I+D+i 2008-2011 National Program, extended by agreement of the Council of Ministers in October 7th, 2011. The work leading to these results has received funding from the European Union under grant agreement No 287678. It has also been supported by EPSRC Programme Grant grant...

Speech Denoising Using Non-Negative Matrix Factorization with Kullback-Leibler Divergence and Sparseness Constraints

Ludeña-Choez, Jimmy; Gallardo-Antolín, Ascensión
Fonte: Springer Publicador: Springer
Tipo: info:eu-repo/semantics/acceptedVersion; info:eu-repo/semantics/bookPart; info:eu-repo/semantics/conferenceObject
Publicado em //2012 Português
Relevância na Pesquisa
46.11%
A speech denoising method based on Non-Negative Matrix Factorization (NMF) is presented in this paper. With respect to previous related works, this paper makes two contributions. First, our method does not assume a priori knowledge about the nature of the noise. Second, it combines the use of the Kullback-Leibler divergence with sparseness constraints on the activation matrix, improving the performance of similar techniques that minimize the Euclidean distance and/or do not consider any sparsification. We evaluate the proposed method for both, speech enhancement and automatic speech recognitions tasks, and compare it to conventional spectral subtraction, showing improvements in speech quality and recognition accuracy, respectively, for different noisy conditions.; This work has been partially supported by the Spanish Government grants TSI-020110-2009-103 and TEC2011-26807.; Proceedings of: IberSPEECH 2012 Conference, Madrid, Spain, November 21-23, 2012.

Single-Microphone Speech Dereverberation: Modulation Domain Processing and Quality Assessment

ZHENG, CHENXI
Fonte: Quens University Publicador: Quens University
Tipo: Tese de Doutorado
Português
Relevância na Pesquisa
46.34%
In a reverberant enclosure, acoustic speech signals are degraded by reflections from walls, ceilings, and objects. Restoring speech quality and intelligibility from reverberated speech has received increasing interest over the past few years. Although multiple channel dereverberation methods provide some improvements in speech quality/ intelligibility, single-channel dereverberation remains an open challenge. Two types of advanced single-channel dereverberation methods, namely acoustic domain spectral subtraction and modulation domain filtering, provide small improvement in speech quality and intelligibility. In this thesis, we study single-channel dereverberation algorithms. Firstly, an upper bound of time-frequency masking (TFM) performance for dereverberation is obtained using ideal time-frequency masking (ITFM). ITFM has access to both the clean and reverberated speech signals in estimating the binary-mask matrix. ITFM implements binary masking in the short time Fourier transform (STFT) domain, preserving only those spectral components less corrupted by reverberation. The experiment results show that single-channel ITFM outperforms four existing multi-channel dereverberation methods and suggest that large potential improvements could be obtained using TFM for speech dereverberation. Secondly...

Blind Estimation of Perceptual Quality for Modern Speech Communications

Falk, Tiago
Fonte: Quens University Publicador: Quens University
Tipo: Tese de Doutorado Formato: 1412501 bytes; application/pdf
Português
Relevância na Pesquisa
46.46%
Modern speech communication technologies expose users to perceptual quality degradations that were not experienced earlier with conventional telephone systems. Since perceived speech quality is a major contributor to the end user's perception of quality of service, speech quality estimation has become an important research field. In this dissertation, perceptual quality estimators are proposed for several emerging speech communication applications, in particular for i) wireless communications with noise suppression capabilities, ii) wireless-VoIP communications, iii) far-field hands-free speech communications, and iv) text-to-speech systems. First, a general-purpose speech quality estimator is proposed based on statistical models of normative speech behaviour and on innovative techniques to detect multiple signal distortions. The estimators do not depend on a clean reference signal hence are termed ``blind." Quality meters are then distributed along the network chain to allow for both quality degradations and quality enhancements to be handled. In order to improve estimation performance for wireless communications, statistical models of noise-suppressed speech are also incorporated. Next, a hybrid signal-and-link-parametric quality estimation paradigm is proposed for emerging wireless-VoIP communications. The algorithm uses VoIP connection parameters to estimate a base quality representative of the packet switching network. Signal-based distortions are then detected and quantified in order to adjust the base quality accordingly. The proposed hybrid methodology is shown to overcome the limitations of existing pure signal-based and pure link parametric algorithms. Temporal dynamics information is then investigated for quality diagnosis for hands-free speech communications. A spectro-temporal signal representation...

Objective Assessment of Dysarthric Speech Intelligibility

HUMMEL, RICHARD
Fonte: Quens University Publicador: Quens University
Tipo: Tese de Doutorado
Português
Relevância na Pesquisa
46.12%
The de-facto standard for dysarthric intelligibility assessment is a subjective intelligibility test, performed by an expert. Subjective tests are often costly, biased and inconsistent because of their perceptual nature. Automatic objective assessment methods, in contrast, are repeatable and relatively cheap. Objective methods can be broken down into two subcategories: reference-free, and reference based. Reference-free methods employ estimation procedures that do not require information about the target speech material. This potentially makes the problem more difficult, and consequently, there is a deficit of research into reference-free dysarthric intelligibility estimation. In this thesis, we focus on the reference-free intelligibility estimation approach. To make the problem more tractable, we focus on the dysarthrias of cerebral palsy (CP). First, a popular standard for blind speech quality estimation, the ITU-T P.563 standard, is examined for possible application to dysarthric intelligibility estimation. The internal structure of the standard is discussed, along with the relevance of its internal features to intelligibility estimation. Afterwards, several novel features expected to relate to some of the acoustic properties of dysarthric speech are proposed. Proposed features are based on the high-order statistics of parameters derived from linear prediction (LP) analysis...

Speech enhancement using empirical mode decomposition and the Teager–Kaiser energy operator; Rehaussement du signal de parole par EMD et opérateur de Teager-Kaiser

KHALDI, Kais; BOUDRAA, Abdelouahab; KOMATY, Ali
Fonte: AIP Publicador: AIP
Português
Relevância na Pesquisa
46.14%
The authors would like to thank Professor Mohamed Bahoura from Universite de Quebec a Rimouski for fruitful discussions on time adaptive thresholding; In this paper a speech denoising strategy based on time adaptive thresholding of intrinsic modes functions (IMFs) of the signal, extracted by empirical mode decomposition (EMD), is introduced. The denoised signal is reconstructed by the superposition of its adaptive thresholded IMFs. Adaptive thresholds are estimated using the Teager–Kaiser energy operator (TKEO) of signal IMFs. More precisely, TKEO identifies the type of frame by expanding differences between speech and non-speech frames in each IMF. Based on the EMD, the proposed speech denoising scheme is a fully data-driven approach. The method is tested on speech signals with different noise levels and the results are compared to EMD-shrinkage and wavelet transform (WT) coupled with TKEO. Speech enhancement performance is evaluated using output signal to noise ratio (SNR) and perceptual evaluation of speech quality (PESQ) measure. Based on the analyzed speech signals, the proposed enhancement scheme performs better than WT-TKEO and EMD-shrinkage approaches in terms of output SNR and PESQ. The noise is greatly reduced using time-adaptive thresholding than universal thresholding. The study is limited to signals corrupted by additive white Gaussian noise.

Voiced speech enhancement based on adaptive filtering of selected intrinsic mode functions; Rehaussement du signal de parole voisé par filtrage adaptatif des modes intrinsèques empiriques

KHALDI, kais; TURKI, Monia; BOUDRAA, Abdel-Ouahab
Fonte: World Scientific Publishing Company Publicador: World Scientific Publishing Company
Português
Relevância na Pesquisa
46.18%
In this paper a new method for voiced speech enhancement combining the Empirical Mode Decomposition (EMD) and the Adaptive Center Weighted Average (ACWA) filter is introduced. Noisy signal is decomposed adaptively into intrinsic oscillatory components called Intrinsic Mode Functions (IMFs). Since voiced speech structure is mostly distributed on both medium and low frequencies, the shorter scale IMFs of the noisy signal are beneath noise, however the longer scale ones are less noisy. Therefore, the main idea of the proposed approach is to only filter the shorter scale IMFs, and to keep the longer scale ones unchanged. In fact, the filtering of longer scale IMFs will introduce distortion rather than reducing noise. The denoising method is applied to several voiced speech signals with different noise levels and the results are compared with wavelet approach, ACWA filter and EMD–ACWA (filtering of all IMFs using ACWA filter). Relying on exhaustive simulations, we show the efficiency of the proposed method for reducing noise and its superiority over other denoising methods, i.e., to improve Signal-to-Noise Ratio (SNR), and to offer better listening quality based on a Perceptual Evaluation of Speech Quality (PESQ). The present study is limited to signals corrupted by additive white Gaussian noise.

Characterisation of noisy speech channels in 2G and 3G mobile networks

Leite, Bruno Daniel Moreira
Fonte: Instituto Politécnico do Porto Publicador: Instituto Politécnico do Porto
Tipo: Dissertação de Mestrado
Publicado em //2013 Português
Relevância na Pesquisa
56.44%
As the wireless cellular market reaches competitive levels never seen before, network operators need to focus on maintaining Quality of Service (QoS) a main priority if they wish to attract new subscribers while keeping existing customers satisfied. Speech Quality as perceived by the end user is one major example of a characteristic in constant need of maintenance and improvement. It is in this topic that this Master Thesis project fits in. Making use of an intrusive method of speech quality evaluation, as a means to further study and characterize the performance of speech codecs in second-generation (2G) and third-generation (3G) technologies. Trying to find further correlation between codecs with similar bit rates, along with the exploration of certain transmission parameters which may aid in the assessment of speech quality. Due to some limitations concerning the audio analyzer equipment that was to be employed, a different system for recording the test samples was sought out. Although the new designed system is not standard, after extensive testing and optimization of the system's parameters, final results were found reliable and satisfactory. Tests include a set of high and low bit rate codecs for both 2G and 3G, where values were compared and analysed...

The impact of speech disorders quality of life: a questionnaire proposal

Lúcio,Gialile de Sá; Perilo,Tatiana Vargas de Castro; Vicente,Laélia Cristina Caseiro; Friche,Amélia Augusta de Lima
Fonte: Sociedade Brasileira de Fonoaudiologia Publicador: Sociedade Brasileira de Fonoaudiologia
Tipo: Artigo de Revista Científica Formato: text/html
Publicado em 01/01/2013 Português
Relevância na Pesquisa
36.32%
PURPOSE: To develop a questionnaire to analyze the impact of speech disorders on quality of life and verify its reliability. METHODS: A literature review on instruments that assess the quality-of-life was performed, particularly those concerning communication disorders. The questionnaire was designed with 18 closed questions: one related to speech impairments, another about quality-of-life, and 16 questions covering the physical, emotional, and social domains. The questionnaire was applied to a population of 24 individuals of both sexes, aged between 12 and 50 years; 12 patients (case group) had phonetic speech disorders, and 12 (control group) had no impairment in oral communication. They were paired according to age and sex. To analyze the reliability of the instrument, the internal consistency of the items was assessed through Cronbach's Alpha coefficient. RESULTS: The internal consistency of the 16 questions concerning the domains was á=0.93; for the physical domain, á=0.71; for the emotional domain, á=0.77; and for the social domain, á=0.85. CONCLUSION: The questionnaire showed good reliability in identifying the impact of speech disorders on the individuals' quality of life.

Perceptual Evaluation Of Playout Buffer Algorithm For Enhancing Perceived Quality Of Voice Transmission Over Ip Network

Perwej, Yusuf; Parwej, Firoj
Fonte: Universidade Cornell Publicador: Universidade Cornell
Tipo: Artigo de Revista Científica
Publicado em 12/05/2012 Português
Relevância na Pesquisa
36.32%
Voice over Internet Protocol (VoIP) is a technology that allows you to make voice calls using a broadband Internet connection instead of a regular (or analog) phone line. Voice over Internet Protocol (VoIP) has led human speech to a new level, where conversation across continents can be much cheaper & faster. However, as IP networks are not designed for real-time applications, the network impairments such as packet loss, jitter and delay have a severe impact on speech quality. The playout buffer at the receiver side is used to compensate jitter at a trade-off of delay and loss. We found the characteristics of delay and loss are dependent on IP network and sudden variable delay (spike) often performs both regular and irregular characteristics. Different playout buffer algorithms can have different impacts on the achievement speech quality. It is important to design a playout buffer algorithm which can help achieve an optimum speech quality. In this paper, we investigate to the understanding how network impairments and existing adaptive buffer algorithms affect the speech quality and further to design a modified buffer algorithm to obtain an optimized voice quality. We conduct experiments to existing algorithms and compared their performance under different network conditions with high and low network delay variations. Preliminary results show that the new algorithm can enhance the perceived speech quality in most network conditions and it is more efficient and suitable for real buffer mechanism.; Comment: 19 pages...