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Dissertations / Theses on the topic 'Voice quality'
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Beck, J. M. "Organic variation and voice quality." Thesis, University of Edinburgh, 1988. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.382921.
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Laver, J. D. M. H. "Individual features in voice quality." Thesis, University of Edinburgh, 1987. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.376885.
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Tasyumruk, Lutfullah. "Analysis of voice quality problems of Voice Over Internet Protocol (VoIP)." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 2003. http://library.nps.navy.mil/uhtbin/hyperion-image/03sep%5FTasyumruk.pdf.
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Ho, Yuen-yan Eva. "Voice quality change using humming technique." Click to view the E-thesis via HKUTO, 1999. http://sunzi.lib.hku.hk/hkuto/record/B36209892.
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Thesis (B.Sc)--University of Hong Kong, 1999."A dissertation submitted in partial fulfilment of the requirements for the Bachelor of Science (Speech and Hearing Sciences), The University of Hong Kong, May 14, 1999." Also available in print.
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Jalalinajafabadi, Farideh. "Computerised GRBAS assessement of voice quality." Thesis, University of Manchester, 2016. https://www.research.manchester.ac.uk/portal/en/theses/computerised-grbas-assessement-of-voice-quality(7efd3263-b109-4137-87cf-b9559c61730b).html.
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Vocal cord vibration is the source of voiced phonemes in speech. Voice quality depends on the nature of this vibration. Vocal cords can be damaged by infection, neck or chest injury, tumours and more serious diseases such as laryngeal cancer. This kind of physical damage can cause loss of voice quality. To support the diagnosis of such conditions and also to monitor the effect of any treatment, voice quality assessment is required. Traditionally, this is done ‘subjectively’ by Speech and Language Therapists (SLTs) who, in Europe, use a well-known assessment approach called ‘GRBAS’. GRBAS is an acronym for a five dimensional scale of measurements of voice properties. The scale was originally devised and recommended by the Japanese Society of Logopeadics and Phoniatrics and several European research publications. The proper- ties are ‘Grade’, ‘Roughness’, ‘Breathiness’, ‘Asthenia’ and ‘Strain’. An SLT listens to and assesses a person’s voice while the person performs specific vocal maneuvers. The SLT is then required to record a discrete score for the voice quality in range of 0 to 3 for each GRBAS component. In requiring the services of trained SLTs, this subjective assessment makes the traditional GRBAS procedure expensive and time-consuming to administer. This thesis considers the possibility of using computer programs to perform objective assessments of voice quality conforming to the GRBAS scale. To do this, Digital Signal Processing (DSP) algorithms are required for measuring voice features that may indicate voice abnormality. The computer must be trained to convert DSP measurements to GRBAS scores and a ‘machine learning’ approach has been adopted to achieve this. This research was made possible by the development, by Manchester Royal Infirmary (MRI) Hospital Trust, of a ‘speech database’ with the participation of clinicians, SLT’s, patients and controls. The participation of five SLTs scorers allowed norms to be established for GRBAS scoring which provided ‘reference’ data for the machine learning approach.
To support the scoring procedure carried out at MRI, a software package, referred to as GRBAS Presentation and Scoring Package (GPSP), was developed for presenting voice recordings to each of the SLTs and recording their GRBAS scores. A means of assessing intra-scorer consistency was devised and built into this system. Also, the assessment of inter-scorer consistency was advanced by the invention of a new form of the ‘Fleiss Kappa’ which is applicable to ordinal as well as categorical scoring. The means of taking these assessments of scorer consistency into account when producing ‘reference’ GRBAS scores are presented in this thesis. Such reference scores are required for training the machine learning algorithms. The DSP algorithms required for feature measurements are generally well known and available as published or commercial software packages. However, an appraisal of these algorithms and the development of some DSP ‘thesis software’ was found to be necessary. Two ‘machine learning’ regression models have been developed for map- ping the measured voice features to GRBAS scores. These are K Nearest Neighbor Regression (KNNR) and Multiple Linear Regression (MLR). Our research is based on sets of features, sets of data and prediction models that are different from the approaches in the current literature. The performance of the computerised system is evaluated against reference scores using a Normalised Root Mean Squared Error (NRMSE) measure. The performances of MLR and KNNR for objective prediction of GRBAS scores are compared and analysed ‘with feature selection’ and ‘without feature selection’. It was found that MLR with feature selection was better than MLR without feature selection and KNNR with and without feature selection, for all five GRBAS components. It was also found that MLR with feature selection gives scores for ‘Asthenia’ and ‘Strain’ which are closer to the reference scores than the scores given by all five individual SLT scorers. The best objective score for ‘Roughness’ was closer than the scores given by two SLTs, roughly equal to the score of one SLT and worse than the other two SLT scores. The best objective scores for ‘Breathiness’ and ‘Grade’ were further from the reference scores than the scores produced by all five SLT scorers. However, the worst ‘MLR with feature selection’ result has normalised RMS error which is only about 3% worse than the worst SLT scoring. The results obtained indicate that objective GRBAS measurements have the potential for further development towards a commercial product that may at least be useful in augmenting the subjective assessments of SLT scorers.
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Ma, Pui-man Estella. "Assessing voice activity and participation implication of clinical management in voice disorders /." Click to view the E-thesis via HKUTO, 1999. http://sunzi.lib.hku.hk/hkuto/record/B36210031.
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Thesis (B.Sc)--University of Hong Kong, 1999."A dissertation submitted in partial fulfilment of the requirements for the Bachelor of Science (Speech and Hearing Sciences), The University of Hong Kong, April 30, 1999." Also available in print.
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Anskaitis, Aurimas. "Analysis of Quality of Coded Voice Signals." Doctoral thesis, Lithuanian Academic Libraries Network (LABT), 2010. http://vddb.laba.lt/obj/LT-eLABa-0001:E.02~2009~D_20100303_142141-66509.
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The dissertation investigates the problem of quality of coded voice. The main attention is paid to voice quality evaluation under packet loss conditions.
The aim of the work is to improve voice quality evaluation algorithms.
The tasks of the work are:
• construction of the means for measurement of voice quality of short voice signals;
• to define the concept of value of coded voice segment and to choose corresponding value metrics;
• to measure distributions of frame values in standard voice;
• to establish limits of distortions created by different codecs;
• to investigate inertia of wide spread codecs and establish the length of impact of one lost frame.
The dissertation consists of the introduction, 4 chapters, conclusions, list of literature.
Introduction presents the novelty and topicality of the work, tasks and aims of the work are formulated.
The first chapter is overview of voice quality evaluation methods, pros and cons of these methods are analyzed. PESQ algorithm and limits of its applicability are introduced in this chapter too. The lists of Lithuanian words for word intelligibility testing are created.
Chapter two presents the method of signal construction that allows to extend PESQ applicability to short signals. This chapter introduces the concept of frame value. Distributions of frame values are calculated.
Third chapter analyses distortions created by coding. It is shown that coding distortions... [to full text]Disertacijoje nagrin jama koduoto balso kokybės vertinimo problematika. Pagrindinis dėmesys skiriamas balso kokybės tyrimams, kai perduodama koduota šneka ir prarandami balso paketai. Darbo tikslas yra patobulinti koduoto balso kokybės vertinimo algoritmus. Darbo uždaviniai yra šie: • sukurti matavimo priemonę trumpų balso signalo atkarpų kokybei vertinti; • apibrėžti koduoto balso segmentų vertės sampratą ir parinkti vertės metrikas; • išmatuoti bendrinės šnekos balso segmentų verčių skirstinius; • nustatyti skirtingų koderių sukuriamų iškraipymų ribas; • ištirti paplitusių koderių inertiškumą, nustatyti kiek laiko pastebima prarastų paketų įtaka sekantiems segmentams. Disertaciją sudaro įvadas, keturi tiriamieji skyriai ir bendrosios išvados. Įvade pristatomas darbo naujumas, aktualumas, aptariamas autoriaus indėlis, formuluojami darbo tikslai. Pirmas skyrius yra apžvalginis – analizuojami balso kokybės vertinimo metodai, jų privalumai ir trūkumai. Kaip savarankiška dalis čia pristatyti autoriaus sudaryti sąrašai lietuviškų žodžių, skirtų šnekos suprantamumo tyrimams. Antrame skyriuje parodoma, kaip galima išplėsti kokybės vertinimo PESQ (angl. Perceptual Evaluation of Speech Quality) algoritmo taikymo ribas. Čia įvedama koduoto balso paketo vertės sąvoka, nustatomi statistiniai paketų vertės skirstiniai. Trečiame skyriuje nagrinėjami specifiniai koduotos šnekos iškraipymai ir kodavimo parametrų įtaka... [toliau žr. visą tekstą]
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Manka, David L. "Voice over Internet Protocol testbed design for non-intrusive, objective voice quality assessment." Thesis, Monterey, Calif. : Naval Postgraduate School, 2007. http://bosun.nps.edu/uhtbin/hyperion-image.exe/07Sep%5FManka.pdf.
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Thesis (M.S. in Electrical Engineering)--Naval Postgraduate School, September 2007.Thesis Advisor(s): Tummala, Murali ; McEachen, John C. "September 2007." Description based on title screen as viewed on October 23, 2007. Includes bibliographical references (p.91-94). Also available in print.
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Ast, Jered Daniel Pendse Ravindra. "The effect of dynamic voice codec selection for active calls on voice quality." Diss., A link to full text of this thesis in SOAR, 2007. http://soar.wichita.edu/dspace/handle/10057/1114.
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Thesis (M.S.)--Wichita State University, College of Engineering, Dept. of Electrical and Computer Engineering."May 2007." Title from PDF title page (viewed on December 14, 2007). Thesis adviser: Ravi Pendse. Includes bibliographic references (leaves 67-69).
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Kramer, Elena [Verfasser]. "Predicting perceptual voice quality from objective voice parameters in dysphonic patients / Elena Kramer." Lübeck : Zentrale Hochschulbibliothek Lübeck, 2013. http://d-nb.info/1029994641/34.
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Sulter, Arend Marten. "Variation of voice quality features and aspects of voice training in males and females." [S.l. : [Groningen] : s.n.] ; [University Library Groningen] [Host], 1996. http://irs.ub.rug.nl/ppn/152639853.
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Williams, Bonnie Blu. "An Investigation of Selected Female Singing- and Speaking-Voice Characteristics Through Comparison of a Group of Pre-Menarcheal Girls to a Group of Post-Menarcheal Girls." Thesis, University of North Texas, 1990. https://digital.library.unt.edu/ark:/67531/metadc330681/.
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The purpose of this study was to compare the speaking fundamental frequency, physiological vocal range, singing voice quality, and self-perceptions of the singing and speaking voice between two groups of girls ages 11 through 15 years, who were pre-menarcheal by 6 months and post-menarcheal by 10 months or more.
Subjects were volunteers who attended a North Texas public school system. Each subject was examined by an otolaryngologist. Age, height, weight, a hearing screening, and information on music classes and/or private music lessons were obtained.
The speaking fundamental frequency measure was obtained by having each subject speak for 30 seconds on a subject of choice and read a passage of approximately 100 syllables. The vocal range measure was obtained by having each subject begin at an arbitrary pitch and sing mah and moo up the scale as high as possible and mah and moo down the scale as low as possible. These four measures were repeated with the researcher giving visual gestures.
For singing-voice quality, each subject sang "America" in the key of her choice and again in the key of F major. Each subjects singing voice was rated according to breathiness. Data regarding self-perceptions of the singing and speaking voice were obtained through a rating assessment of 10 questions and a conversation with each subject.
There were no significant differences between the means of the pre-meanarcheal and post-menarcheal girls on speaking fundamental frequency, physiological vocal range, and singing-voice quality. But, more of the post-menarcheal girls exhibited lower speaking pitches, lower singing ranges, and increased breathiness in their singing voices than did the pre-menarcheal girls.
Two questions of the perceptions rating assessment were significant, with the post-menarcheal girls citing higher incidences of vocal inconsistencies than the pre-menarcheal girls. The findings of the qualitative data analysis indicated that more post-menarcheal girls had an adequate vocabulary to describe various aspects of their singing and speaking voices than did the pre-menarcheal girls.
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Li, Lai-ching Gina. "Voice quality of Cantonese tones an acoustic study /." Click to view the E-thesis via HKUTO, 1997. http://sunzi.lib.hku.hk/hkuto/record/B3620948X.
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Thesis (B.Sc)--University of Hong Kong, 1997."A dissertation submitted in partial fulfilment of the requirements for the Bachelor of Science (Speech and Hearing Sciences), The University of Hong Kong, April 30, 1997." Also available in print.
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Le, An Thanh. "Controlling and Monitoring Voice Quality in Internet Communication." Scholar Commons, 2017. http://scholarcommons.usf.edu/etd/6659.
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The Voice over Internet Protocol (VoIP) is on its way to surpassing toll quality. Although VoIP shares its transmission channel with other communication traffic, today internet has a wider bandwidth than the legacy Digital Loop Carrier and voice could be digitized higher than traditional 8 kbps, to say 16 kbps. Thus, VoIP should not be limited by the toll quality. However, VoIP quality could go down, as a result of unpredictable traffic congestion and network imperfections. These two situations cause delay jitter and packet loss of VoIP. To overcome these challenges, there are ongoing works for service providers including but not limited to optimizing routing and adding more bandwidth. There are also works by developers at the user’s end, which includes compressing voice packet size and processing playout delay adapted to the network condition.
While VoIP planning or off-line quality monitoring and control use overall quality measurements such as mean opinion score (MOS) or R-factor, the real-time quality supervision typically uses the network condition factors only. The control mechanism that is based on network quality could adjust the channel parameter by changing Codec and its parameters, and changing playout delay, etc. to minimize the loss of voice quality.
As bandwidth plays a prominent role in IP traffic congestion, compressing the packet header is a possible solution to minimize congestion. Replacing a completed packet header with a smaller header will significantly reduce the packet header size. For instance, with a context, a compressed header will not consist of RTP header and, thus, could reduce 16 bytes from each packet. However, the primary question is how to deal with delay jitter calculation without time stamping. In this research, a delay jitter calculation for VoIP packet without timestamp has been provided.
Compressing payload or using high compressing Codecs, is another major solution for preventing quality downgrade with limited bandwidth. The challenge with many Codec and the tradeoff between Codec quality and packet loss due to limited bandwidth has been addressed in this research with a summary of Codec quality evaluation and a bandwidth planning calculation.
Although the E-model and its R-factor has been proposed by the International Telecommunication Union (ITU) for VoIP quality measurement, with many network and Codec parameters, it could only be used for offline quality control. Since accessing a live traffic for monitoring live quality is somewhat impossible, at the client side, only packet loss and delay jitter matters. In this research, more in-depth investigation of adaptive playout delay based on jitter prediction has been carried out and recommended as the end user solution for quality improvement. An adaptive playout delay based on Markov model also has been developed in detail and tested with real VoIP network. This development has closed the gap between research and engineering. Therefore, the Markov model could be evaluated and implemented.
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Murrin, Paul. "Objective measurement of voice activity detectors." Thesis, University of York, 1999. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.325647.
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Sundin, Anton. "Audio quality perception of SCIP encrypted voice transmission over low quality radio links." Thesis, Uppsala universitet, Signaler och System, 2016. http://urn.kb.se/resolve?urn=urn:nbn:se:uu:diva-298312.
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Tactical radio communications used in military applications hasstrict requirements regarding security and has to be operable inrough environments in which there may be disturbances and disruptionson a radio link. The performance of the Secure CommunicationInteroperability Protocol (SCIP) operating in an asynchronouscommunication network with various levels of packet loss isinvestigated and found inadequate mainly due to problems withcryptographic synchronization between the transmitting and receivingunits. The introduction of additional counter data to each datapacket remedies this problem and allows the receiving units to fillthe holes left by packet losses with filler packets, maintainingsynchronization. The audio quality can then be measured using thePerceptual Evaluation of Speech Quality (PESQ) algorithm.Measurements are performed in an emulated radio link with aconfigurable packet loss ratio developed by Saab. The results showthat parts of SCIP can be used alongside the counter solution withoutimpacting the audio quality. The insertion of filler packets is shownto have a positive effect on the audio quality, while aggregation ofpackets to conserve transmission data rate is shown to have anegative effect.
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Chan, Mei-mei Rainy. "The effect of hydration and vocal rest on vocal quality and function after Karaoke singing among people." Click to view the E-thesis via HKUTO, 2000. http://sunzi.lib.hku.hk/hkuto/record/B36207457.
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Thesis (B.Sc)--University of Hong Kong, 2000."A dissertation submitted in partial fulfilment of the requirements for the Bachelor of Science (Speech and Hearing Sciences), The University of Hong Kong, May 10, 2000." Also available in print.
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Nakkhongkham, Saiyoot. "Measuring the quality of service of voice over IP." Thesis, University of Hawaii at Manoa, 2003. http://hdl.handle.net/10125/7007.
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This thesis addresses and proposes the methods to measure and classify problems encountered when transmitting voice over the Internet. These problems are delay, jittering, and lost of data. Transmitting voice over the Internet is commonly known as Voice over Internet Protocol (VoIP). The fact that the current Internet infrastructure was not designed to transmit real-time data makes these problems become eminent. Solutions to improve the quality of real-time data transmissions over the Internet are currently under research. This thesis presents a VoIP software-based application tool to help VoIP engineers to better measure and classify these problems. This software tool is called Voice Over Internet Protocol Quality of Service Measuring Tool (VoIP QoS MT). This tool works in a live mode and in a test traffic mode. Live mode allows users to transmit actual voice conversations between two computers over the Internet, using a few digitization and compression schemes. Quality of service can be observed in real-time speech. The test traffic mode works without voice conversation. Its payloads are programmed to carry information for the purpose of testing delay, jittering, and lost statistics. Detailed functionalities and related background information for each components used in the application are presented accordingly in this paperxii, 110 leaves
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Gustafsson, Erik, and Fredrik Larsson. "Översättning och validering av Voice-Related Quality of Life." Thesis, Linköpings universitet, Logopedi, 2012. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-77932.
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En röststörning kan sägas föreligga då rösten inte fungerar eller låter som den brukar så att det påverkar kommunikationen. Prevalensen för röststörningar uppskattas till omkring 6 % av den vuxna befolkningen. När rösten inte fungerar som den ska leder det till emotionella, sociala och funktionella svårigheter för individen och har negativ inverkan på livskvaliteten. Voice- Related Quality of Life (V-RQOL) är ett självskattningsformulär som mäter vilken inverkan en röststörning kan ha på individens livskvalitet. Detta instrument är internationellt välanvänt, men har inte funnits översatt till svenska. Syftet med föreliggande uppsats var att översätta och validera V-RQOL för en svensk population. Översättningen skedde genom så kallad back translation och den svenska versionen fick namnet Röstrelaterad livskvalitet (RRL). RRL och Rösthandikappindex (RHI) distribuerades i pappersformat till en röstpatientgrupp (n = 88) och en röstfrisk grupp (n = 110). Reliabiliteten av domänerna och samtliga tio påståenden på RRL var hög för patientgruppen med Cronbach’s alfa- värden från 0.82 till 0.90. Det fanns en tydlig relation mellan den självskattade röstkvaliteten och poängen på RRL och formuläret kunde även differentiera mellan röstpatienter och röstfriska. Dessa resultat indikerar att formulärets begreppsvaliditet är god. Pearson’s korrelationsanalys visade att det fanns en signifikant negativ korrelation mellan RRL:s och RHI:s domäner och totalpoäng. Detta starka samband mellan ”the gold standard”, RHI, och RRL innebär en god kriterierelaterad validitet för formuläret. Sammanfattningsvis visar resultaten att RRL har hög reliabilitet och god validitet, dessutom är formuläret kort vilket innebär en liten arbetsinsats. Formuläret anses därför vara ett pålitligt och värdefullt tillägg i den kliniska bedömningen av röststörningar.A voice disorder can be said to exist when the voice does not work or sound as it normally should in a manner so that it interferes with communication. The prevalence of voice disorders is estimated to be about 6 % of the adult population. When the voice does not work as it should it may lead to emotional, social and functional difficulties for the individual and with negative effects on the quality of life. Voice- Related Quality of Life (V-RQOL) is a self-reporting questionnaire which measures the effect of a voice disorder on the quality of life of an individual. The instrument is frequently used internationally, but no Swedish translation has existed. In the present study, the purpose was to translate and validate V-RQOL for a Swedish population. The method for translating the questionnaire was back translation. The translated questionnaire was given the Swedish name Röstrelaterad livskvalitet (RRL). RRL and the Swedish version of The Voice Handicap Index, Rösthandikappindex (RHI), was distributed to a group of voice patients (n = 88) and a group of non-voice patients (n = 110). The reliability of the domains and the combined items of RRL was high according to Cronbach’s alpha with alpha values ranging from 0.82 to 0.90. There was a strong relation between the self–estimated voice quality and the scores on RRL, and the RRL- questionnaire was shown to differentiate between voice patients and non-voice patients. These results indicate that the construct validity of the questionnaire is good. Pearson’s correlation analysis demonstrated a significant negative correlation between the domains and total scores of RRL and RHI. This strong relation between “the gold standard”, RHI, and RRL proved that the criterion validity of the questionnaire is good. In summary the results show that RRL has a high reliability and good validity, in addition to this, the questionnaire is short and requires minimal work. The questionnaire is therefore considered to be a reliable and valuable addition to the clinical assessment of voice disorders.
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Guršnys, Darius. "Voice quality evaluation methods and means for mobile communications." Doctoral thesis, Lithuanian Academic Libraries Network (LABT), 2008. http://vddb.library.lt/obj/LT-eLABa-0001:E.02~2008~D_20081119_135710-84655.
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Thesis presents the research of voice frame losses in the GSM. The possibility of evaluation of quality of short duration signals using PESQ algorithm is analysed. New method for preparation of short duration signals for PESQ measurements is proposed. Original method based on quality classes for voice quality evaluation in mobile communications is presented.Disertacijoje nagrinėjami GSM kanale vykstantys balso paketų praradimai. Nagrinėjama galimybė vertinti trumpų balso segmentų kokybę PESQ algoritmu. Pasiūlomas naujas, mažos trukmės balso signalų kokybės vertinimo PESQ algoritmu metodas. Pristatomas originalus balso kokybės vertinimo metodas mobiliojo ryšio sistemoms, paremtas balso kokybės klasėmis.
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Sun, Lingfen. "Speech quality prediction for voice over Internet protocol networks." Thesis, University of Plymouth, 2004. http://hdl.handle.net/10026.1/870.
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IP networks are on a steep slope of innovation that will make them the long-term carrier of all types of traffic, including voice. However, such networks are not designed to support real-time voice communication because their variable characteristics (e.g. due to delay, delay variation and packet loss) lead to a deterioration in voice quality. A major challenge in such networks is how to measure or predict voice quality accurately and efficiently for QoS monitoring and/or control purposes to ensure that technical and commercial requirements are met. Voice quality can be measured using either subjective or objective methods. Subjective measurement (e.g. MOS) is the benchmark for objective methods, but it is slow, time consuming and expensive. Objective measurement can be intrusive or non-intrusive. Intrusive methods (e.g. ITU PESQ) are more accurate, but normally are unsuitable for monitoring live traffic because of the need for a reference data and to utilise the network. This makes non-intrusive methods(e.g. ITU E-model) more attractive for monitoring voice quality from IP network impairments. However, current non-intrusive methods rely on subjective tests to derive model parameters and as a result are limited and do not meet new and emerging applications. The main goal of the project is to develop novel and efficient models for non-intrusive speech quality prediction to overcome the disadvantages of current subjective-based methods and to demonstrate their usefulness in new and emerging VoIP applications. The main contributions of the thesis are fourfold: (1) a detailed understanding of the relationships between voice quality, IP network impairments (e.g. packet loss, jitter and delay) and relevant parameters associated with speech (e.g. codec type, gender and language) is provided. An understanding of the perceptual effects of these key parameters on voice quality is important as it provides a basis for the development of non-intrusive voice quality prediction models. A fundamental investigation of the impact of the parameters on perceived voice quality was carried out using the latest ITU algorithm for perceptual evaluation of speech quality, PESQ, and by exploiting the ITU E-model to obtain an objective measure of voice quality. (2) a new methodology to predict voice quality non-intrusively was developed. The method exploits the intrusive algorithm, PESQ, and a combined PESQ/E-model structure to provide a perceptually accurate prediction of both listening and conversational voice quality non-intrusively. This avoids time-consuming subjective tests and so removes one of the major obstacles in the development of models for voice quality prediction. The method is generic and as such has wide applicability in multimedia applications. Efficient regression-based models and robust artificial neural network-based learning models were developed for predicting voice quality non-intrusively for VoIP applications. (3) three applications of the new models were investigated: voice quality monitoring/prediction for real Internet VoIP traces, perceived quality driven playout buffer optimization and perceived quality driven QoS control. The neural network and regression models were both used to predict voice quality for real Internet VoIP traces based on international links. A new adaptive playout buffer and a perceptual optimization playout buffer algorithms are presented. A QoS control scheme that combines the strengths of rate-adaptive and priority marking control schemes to provide a superior QoS control in terms of measured perceived voice quality is also provided. (4) a new methodology for Internet-based subjective speech quality measurement which allows rapid assessment of voice quality for VoIP applications is proposed and assessed using both objective and traditional MOS test methods.
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Wun, Mo-yan Brenda. "A cross-cultural comparison on the impact of voice disorder on quality of life." Click to view the E-thesis via HKU Scholars Hub, 2003. http://lookup.lib.hku.hk/lookup/bib/B38890975.
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Thesis (B.Sc.)--University of Hong Kong, 2003."A dissertation submitted in partial fulfilment of the requirements for the Bachelor of Science (Speech and Hearing Sciences), The University of Hong Kong, April 30, 2003." Includes bibliographical references (p. 28-30) Also available in print.
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Andersson, Martin. "Parametric Prediction Model for Perceived Voice Quality in Secure VoIP." Thesis, Linköpings universitet, Informationskodning, 2016. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-127402.
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More and more sensitive information is communicated digitally and with thatcomes the demand for security and privacy on the services being used. An accurateQoS metric for these services are of interest both for the customer and theservice provider. This thesis has investigated the impact of different parameterson the perceived voice quality for encrypted VoIP using a PESQ score as referencevalue. Based on this investigation a parametric prediction model has been developedwhich outputs a R-value, comparable to that of the widely used E-modelfrom ITU. This thesis can further be seen as a template for how to construct modelsof other equipments or codecs than those evaluated here since they effect theresult but are hard to parametrise. The results of the investigation are consistent with previous studies regarding theimpact of packet loss, the impact of jitter is shown to be significant over 40 ms.The results from three different packetizers are presented which illustrates theneed to take such aspects into consideration when constructing a model to predictvoice quality. The model derived from the investigation performs well withno mean error and a standard deviation of the error of a mere 1:45 R-value unitswhen validated in conditions to be expected in GSM networks. When validatedagainst an emulated 3G network the standard deviation is even lower.v
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Ho, Elaine Mandy. "Effects of cultural and linguistic backgrounds on perceptual voice quality rating." Click to view the E-thesis via HKU Scholors Hub, 2005. http://lookup.lib.hku.hk/lookup/bib/B38279204.
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Thesis (B.Sc)--University of Hong Kong, 2005."A dissertation submitted in partial fulfilment of the requirements for the Bachelor of Science (Speech and Hearing Sciences), The University of Hong Kong, June 30, 2005." Also available in print.
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Huber, Stefan. "Voice Conversion by modelling and transformation of extended voice characteristics." Electronic Thesis or Diss., Paris 6, 2015. https://accesdistant.sorbonne-universite.fr/login?url=https://theses-intra.sorbonne-universite.fr/2015PA066750.pdf.
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La Conversion de la Voix (VC) vise à transformer les caractéristiques de la voix d’un locuteur source de manière qu’il sera perçu comme étant prononcé par un locuteur cible. Le principe de la VC est de définir des fonctions du transposition pour la conversion de la voix de l’un locuteur source à la voix de l’un locuteur cible. Les fonctions de transformation de VC systèmes "State-Of-The-Art" (START) adapte instantanément aux caractéristiques de la voix source. Cependant, la qualité est pas encore suffisant. Des améliorations considérables sont nécessaires que les techniques VC peuvent être utilisés dans un environnement industriel professionnel. L’objectif de cette thèse est d’augmenter la qualité de la conversion de la voix pour faciliter son applicabilité industrielle dans une mesure raisonnable. Les propriétés de base de différentes START algorithmes de la conversion de la voix sont discutés sur leurs avantages intrinsèques et ses déficits. Basé sur des évaluations expérimentales avec un GMM VC système la conclusion est que la plupart des systèmes VC START qui reposent sur des modèles statistiques sont, en raison de l’effet en moyenne de la régression linéaire, moins appropriées pour atteindre un score du similitude assez élevé avec le haut-parleur cible requise pour l’utilisation industrielle. Les contributions établies pendant de ce travail de thèse se trouvent dans les moyens étendus à a) modéliser l’excitation du source glottique, b) modéliser des descripteurs de la voix en utilisant un nouveau système de parole basée sur un modèle élargie de source-filtre, et c) avancer une nouveau système VC de l’Ircam en le combinant avec les contributions de a) et b)Voice Conversion (VC) aims at transforming the characteristics of a source speaker’s voice in such a way that it will be perceived as being uttered by a target speaker. The principle of VC is to define mapping functions for the conversion from one source speaker’s voice to one target speaker’s voice. The transformation functions of common State-Of-The-Art (START) VC system adapt instantaneously to the characteristics of the source voice. While recent VC systems have made considerable progress over the conversion quality of initial approaches, the quality is nevertheless not yet sufficient. Considerable improvements are required before VC techniques can be used in an professional industrial environment. The objective of this thesis is to augment the quality of Voice Conversion to facilitate its industrial applicability to a reasonable extent. The basic properties of different START algorithms for Voice Conversion are discussed on their intrinsic advantages and shortcomings. Based on experimental evaluations of one GMM-based State-Of-The-Art VC approach the conclusion is that most VC systems which rely on statistical models are, due to averaging effect of the linear regression, less appropriate to achieve a high enough similarity score to the target speaker required for industrial usage. The contributions established throughout this thesis work lie in the extended means to a) model the glottal excitation source, b) model a voice descriptor set using a novel speech system based on an extended source-filter model, and c) to further advance IRCAM’s novel VC system by combining it with the contributions of a) and b)
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Fung, Yam-cheung Kelvin. "The use of internal versus external standards in perceptual evaluation of voice quality." Click to view the E-thesis via HKUTO, 1994. http://sunzi.lib.hku.hk/hkuto/record/B36208899.
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Thesis (B.Sc)--University of Hong Kong, 1994."A dissertation submitted in partial fulfilment of the requirements for the Bachelor of Science (Speech and Hearing Sciences), The University of Hong Kong, April 29, 1994." Also available in print.
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Yu, Tik-yin Grace. "Perception of voice use and problems in female singers and broadcasters an impairment, activity limitation and participation restriction perspective /." Click to view the E-thesis via HKUTO, 2001. http://sunzi.lib.hku.hk/hkuto/record/B36208115.
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Thesis (B.Sc)--University of Hong Kong, 2001."A dissertation submitted in partial fulfilment of the requirements for the Bachelor of Science (Speech and Hearing Sciences), The University of Hong Kong, May 4, 2001." Also available in print.
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Pereira, Patricia Moraes. "Análise comparativa da voz em jovens mulheres antes e depois da prova de fala contínua." Universidade de São Paulo, 2015. http://www.teses.usp.br/teses/disponiveis/17/17151/tde-20072016-143744/.
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OBJETIVO: Comparar a voz de mulheres antes e depois de 60 e 90 minutos de prova de fala contínua, e após repouso de 15 minutos. MÉTODOS: Trinta e uma mulheres com idade entre 18 e 25 anos, foram submetidas à tarefa de resistência fonatória, utilizando-se da leitura de um texto padrão por 90 minutos, repetido até que o tempo se esgotasse. Antes da tarefa de fala contínua, após 60 minutos, após 90 minutos, e depois de 15 minutos de repouso vocal absoluto, aplicou-se questionário para conhecimento do bem estar vocal, registrou-se a emissão prolongada da vogal \"a\", para posterior extração das medidas acústicas e da análise perceptivo-auditiva com o uso da escala GIRBAS. A seguir, fez-se a mensuração das medidas do sistema aéreo fonatório, empregando os protocolos: capacidade vital pulmonar (CVP), tempo máximo de fonação (TMF) e a eficiência vocal (EV). A intensidade vocal foi registrada com decibelímetro e a auto-avaliação da percepção auditiva, tátil e cinestésica da voz com o uso de uma escala visual analógica de 100mm. RESULTADOS: Após 60 minutos de fala, aumentou a frequência fundamental (f0) de 215,4 para 220,2Hz (p<0,01), a ATRI (p=0,04) e a NHR (p=0,03), e com 90 minutos, a f0 variou de 215,4 para 223,6Hz (p<0,01), aumentando também a Fhi (p= 0,04) e a Flo (p= 0,02), e diminuindo a APQ (p=0,01) e a VTi (p=0,04). Comparando as medidas observadas na pré-prova e após o repouso, aumentaram f0 (p<0,01), Fhi (p=0,02) e Flo (p=0,03). Entre os tempos 60 minutos e após o repouso, houve aumento da PPQ (p=0,04), da ATRI (p=0,06) e da NHR (p=0,02). Para 90 minutos e repouso, a PPQ (p=0,03) e a Fatr (p=0,04) aumentaram. Vinte e sete participantes apresentaram grau geral da disfonia 1 tanto para 60 minutos, quanto para 90 minutos, e quatro passaram a apresentar grau 2 em 90 minutos (p=0,04). O parâmetro instabilidade alterou de grau 1, com 60 minutos, para grau 2, com 90 minutos de fala contínua (p=<0,01). A intensidade habitual aumentou (p<0,01) de 61,4 para 63,4dB após 90 minutos. Após o repouso, houve diminuição da intensidade (p=0,01), em relação ao pré-prova. Nas medidas observadas pelo sistema aéreo fonatório, o fluxo de ar expiratório diminuiu após 90 minutos de fala (p=0,04), aumentando depois do repouso (p=0,04). Após 90 minutos de fala a f0(Hz) aumentou 211,85 para 221,54 (p<0,01). A resistência aerodinâmica, impedância acústica e eficiência aerodinâmica aumentaram após 60 e após 90 minutos de fala. A auto avaliação perceptivo-auditiva e tátil-cinestésica da voz, observou que após 90 minutos de fala contínua todos os sintomas pioram, exceto a rouquidão e a voz grave. CONCLUSÃO: Houveram alterações das medidas acústicas após tarefa de fala contínua. O grau geral da disfonia e a instabilidade vocal aumentaram após 90 minutos de fala contínua. As medidas aerodinâmicas se comportaram de forma divergente entre os protocolos utilizados e os tempos de avaliação. A intensidade vocal habitual aumentou após 90 minutos de fala contínua e os sintomas perceptuais auditivos e tátil-cinestésicos aumentaram após a tarefa de fala contínuaPURPOSE: To compare the woman\'s voice before and after 60 and 90 minutes of continuous speech test, and after 15 break minutes. METHODS: Thirty-one women aged between 18 and 25 years, were submitted to phonation endurance task, using the reading of a standard text for 90 minutes repeating until the time was over. Before the continuous speech task after 60 minutes, after 90 minutes, and after 15 minutes of absolute voice rest, it was applied a questionnaire to knowledge of well-being vocal and was recorded the prolonged vowel \"a\", to take in a posterior time the acoustic measurements and perceptual analysis using the GIRBAS scale. Then, it was done the mensuration of measures phonation air system, using the protocols: lung vital capacity (LVC), maximum phonation time (MPT) and vocal efficiency (VE). The vocal intensity was recorded with a decibelimeter and self-assessment of auditory, tactile and kinesthetic perception of voice using a visual analog scale of 100mm. RESULTS: After 60 minutes of speech, increased the fundamental frequency (f0) of 215.4 for 220,2Hz (p <0.01), the ATRI (p = 0.04) and NHR (p = 0.03) with 90 minutes f0 ranged from 215.4 to 223,6Hz (p <0.01), also increasing FHI (p = 0.04), and Flo (p = 0.02) and decreasing APQ (p = 0.01) and VTi (p = 0.04). Comparing the measures observed in pre-test and after the break, increased f0 (p <0.01), FHI (p = 0.02) and Flo (p = 0.03). Between time of 60 minutes and after the break, was observed an increase in PPQ (p = 0.04) of ATRI (p = 0.06) and NHR (p = 0.02). For 90 minutes rest, PPQ (p = 0.03) and Fatr (p = 0.04) increased. Twentyseven subjects had overall grade of dysphonia 1 for both 60 minutes and for 90 minutes and began to show four grade 2 in 90 minutes (p = 0.04). The parameter of instability changed of step 1, with 60 minutes to 2 degree, with 90 minutes of continuous speech (p = <0.01). The usual intensity increased (p <0.01) 61.4 to 63,4dB after 90 minutes. After the break, there was a decrease in the intensity (p = 0.01), compared to the pre-test. In the measurements observed by phonation air system the flow of expiratory air decreased after 90 minutes of speech (p = 0.04) and raised after the rest (p = 0.04). After 90 minutes of speech f0 (Hz) to 221.54 211.85 increased (p <0.01). The aerodynamic resistance, acoustic impedance and aerodynamic efficiency increased after 60 and after 90 minutes of speech. Self perceptual assessment and tactile-kinesthetic voice, noted that after 90 minutes of continuous talk all the symptoms get worse, except for hoarseness and a deep voice. CONCLUSION: There were changes of acoustic measurements after continuous speech task. The overall degree of dysphonia and vocal instability increased after 90 minutes of continuous speech. The aerodynamic measures worked in different ways about the protocols used and the time evaluation. The usual voice intensity increased after 90 minutes of continuous speech and perceptual symptoms auditory and tactilekinesthetic increased after continuous speech task
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Nocito, Carlos Daniel. "A Network Conditions Estimator for Voice Over IP Objective Quality Assessment." Scholarly Repository, 2011. http://scholarlyrepository.miami.edu/oa_theses/292.
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Objective quality evaluation is a key element for the success of the emerging Voice over IP (VoIP) technologies. Although there are extensive economic incentives for the convergence of voice, data, and video networks, packet networks such as the Internet have inherent incompatibilities with the transport of real time services. Under this paradigm, network planners and administrators are interested in ongoing mechanisms to measure and ensure the quality of these real time services. Objective quality assessment algorithms can be broadly divided into a) intrusive (methods that require a reference signal), and b) non intrusive (methods that do not require a known reference signal). The latter group, typically requires knowledge of the network conditions (level of delay, jitter, packet loss, etc.), and that has been a very active area of research in the past decade. The state of the art methods for objective non-intrusive quality assessment provide high correlations with the subjective tests. Although good correlations have been achieved already for objective non-intrusive quality assessment, the current large voice transport networks are in a hybrid state, where the necessary network parameters cannot easily be observed from the packet traffic between nodes. This thesis proposes a new process, the Network Conditions Estimator (NCE), which can serve as bridge element to real-world hybrid networks. Two classifications systems, an artificial neural network and a C4.5 decision tree, were developed using speech from a database collected from experiments under controlled network conditions. The database was composed of a group of four female speakers and three male speakers, who conducted unscripted conversations without knowledge about the details of the experiment. Using mel frequency cepstral coefficients (MFCCs) as the feature-set, an accuracy of about 70% was achieved in detecting the presence of jitter or packet loss on the channel. This resulting classifier can be incorporated as an input to the E-Model, in order to properly estimate the QoS of a network in real time. Additionally, rather than just providing an estimation of subjective quality of service provided, the NCE provides an insight into the cause for low performance.
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Turajlic, Emir. "A novel framework for high-quality voice source analysis and synthesis." Thesis, Brunel University, 2006. http://bura.brunel.ac.uk/handle/2438/7300.
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The analysis, parameterization and modeling of voice source estimates obtained via inverse filtering of recorded speech are some of the most challenging areas of speech processing owing to the fact humans produce a wide range of voice source realizations and that the voice source estimates commonly contain artifacts due to the non-linear time-varying source-filter coupling. Currently, the most widely adopted representation of voice source signal is Liljencrants-Fant's (LF) model which was developed in late 1985. Due to the overly simplistic interpretation of voice source dynamics, LF model can not represent the fine temporal structure of glottal flow derivative realizations nor can it carry the sufficient spectral richness to facilitate a truly natural sounding speech synthesis. In this thesis we have introduced Characteristic Glottal Pulse Waveform Parameterization and Modeling (CGPWPM) which constitutes an entirely novel framework for voice source analysis, parameterization and reconstruction. In comparative evaluation of CGPWPM and LF model we have demonstrated that the proposed method is able to preserve higher levels of speaker dependant information from the voice source estimates and realize a more natural sounding speech synthesis. In general, we have shown that CGPWPM-based speech synthesis rates highly on the scale of absolute perceptual acceptability and that speech signals are faithfully reconstructed on consistent basis, across speakers, gender. We have applied CGPWPM to voice quality profiling and text-independent voice quality conversion method. The proposed voice conversion method is able to achieve the desired perceptual effects and the modified speech remained as natural sounding and intelligible as natural speech. In this thesis, we have also developed an optimal wavelet thresholding strategy for voice source signals which is able to suppress aspiration noise and still retain both the slow and the rapid variations in the voice source estimate.
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Lusoli, Wainer. "Voice and e-quality : the state of electronic democracy in Britain." Thesis, London School of Economics and Political Science (University of London), 2006. http://etheses.lse.ac.uk/1884/.
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This dissertation is broadly concerned with the issue of electronic democracy, i.e. whether, under what conditions and how does the Internet strengthen democracy in advanced industrial polities. Specifically, this work applies the theory of participation to recent British data on online political engagement in order to understand: whether and how the Internet modifies the existing structure of political inequality; whether and how the Internet alters the context of traditional political action; whether the Internet holds a democratising potential and what is its nature. Data collected and analysed include a survey of British citizens' online political behaviour, and three smaller, in-depth surveys of citizens' online political activities within limited settings: a national online consultation forum, routine politics by young party activists and charity work by an elderly activist network. More generally, the dissertation contributes towards clarifying the ongoing debate on electronic democracy, by examining the discourse surrounding the evolution of the issue. It reviews a large portion of the existing literature on online political engagement, organised in three main approaches. It presents and analyses seminal data on British online political engagement to assess the state of electronic democracy in Britain. Importantly, it advances a theoretical framework for the understanding of the 'real' digital divide, drawing on the theory of participation. The theory is an ideal explanatory base from which to depart in order to find the factors shaping the structure of online political opportunities and the way in which preferences are voiced, and heard, through the Internet. This dissertation speaks directly to the electronic democracy debate by setting the agenda on the notion of democratic equality and by focusing on the structure of voice in the information polity.
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IRWIN, LINDSAY K. "PERCEPTUAL EVALUATION OF VOICE QUALITY OF INDIVIDUALS WITH DYSPHAGIA AND DYSPHONIA." University of Cincinnati / OhioLINK, 2006. http://rave.ohiolink.edu/etdc/view?acc_num=ucin1148416348.
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DE, GREGORIS GREGORIO. "VOICE QUALITY AND TV INTERPRETING: A PROPOSAL FOR A GESTALTIC EVALUATION." Doctoral thesis, Università degli Studi di Trieste, 2016. http://hdl.handle.net/11368/2908100.
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RESUMEN. La presente tesis doctoral es un estudio de interpretación basado en corpus y consiste en una propuesta de evaluación subjetiva de tipo gestáltico de la interpretación simultánea transmitida por televisión. El objetivo principal del estudio ha sido la construcción de un modelo de evaluación de la calidad basado en la percepción gestáltica del habla y del sonido-imagen percibido a través del medio auiovisual. El modelo de percepción gestáltica adoptado está formado por voz-sílaba-prosodia-sentido-contexto-conocimiento (lingϋístico) del mundo, propuesto en “Il volto fonico delle parole” (Albano Leoni 2009), que es una reelaboración del modelo basado en melodía-ritmo-palabras-oraciones, propuesto por Karl Bϋhler en su “Teoría del lenguaje” (1934).
Se construyó un corpus temático formado por las interpretaciones en italiano (2) y en español (2 – España y Estados Unidos) de los Debates Presidenciales de Estados Unidos de 2012: el corpus ORenesit (Obama-Romney English español italiano) se incluye en el corpus de referencia CorIT (Corpus Italiano de Interpretación Televisiva).
El modelo de evaluación fue ensayado en una encuesta piloto basada en cuestionario, que incluye 3 extractos vídeo de la interpretación en italiano del Tercer Debate Presidencial de EE.UU. de 2008, entre Obama y McCain, debido a que el corpus ORenesit todavía no se había terminado. Uno de los tres vídeos fue modificado por fines experimentales: la voz del intérprete original se sustituyó por la de un actor doblador profesional que imitó en estudio la interpretación original leyendo la transcripción y escuchando al orador. Esta decisión respondía a dos necesidades, relacionadas sobre todo a la validez ecológica del experimento: a) ensayar el efecto de una voz telegénica; b) utilizar la expresión natural y personal del sujeto. El cuestionario se construyó sobre categorías extraídas de “La vive voix” (Fónagy 1983) e “L’Audio-Vision” (Chion 1990). Los datos obtenidos del cuestionario se trataron estadísticamente. Los resultados del estudio cuali-cuantitativo parecen confirmar una percepción gestáltica de la interpretación simultánea percibida a través del medio audio-visual formada por las componentes: sonido-imagen, sílaba-melodía(-voz-personalidad), palabras-oraciones. Lor resultados parecen poner en duda la efectividad del enfoque cuantitativo para el análisis de la percepción del habla.ABSTRACT. The present thesis is a corpus-based Interpreting study consisting of a proposal for a gestaltic subjective evaluation of quality in television broadcast simultaneous interpreting. The main objective of the research was to build and test a model of quality assessment based on the gestaltic perception both of speech and the sound-image perceived through the audiovisual medium. The model of gestaltic perception adopted is the one formed by voice-syllable-prosody-sense-context-(linguistic) knowledge of the world, proposed in “Il volto fonico delle parole” (Albano Leoni 2009), which is a re-elaborated version of the model based on melody-rhythm-words-sentences, proposed by Karl Bϋhler in his “Theory of Language” (1934). A thematic corpus was built consisting of 2 Italian and 2 Spanish (Spain and United States) interpretations of the 2012 US Presidential Debates: the corpus ORenesit (Obama-Romney English español italiano) is included in the reference corpus CorIT (Italian Television Interpreting Corpus). The assessment model was tested in a questionnaire-based pilot survey including 3 video excerpts from the Italian interpretations of the 2008 Third Presidential Debate (Obama vs. McCain), since the corpus ORenesit had not been completed yet. One of the 3 video excerpts was modified for experimental purpose: the interpreter’s voice was replaced with the voice of a professional actor and dubber, who imitated in studio the original interpretation while reading the transcript and listening to the speaker. This choice was made to fulfill two needs, mainly related to the ecological validity of the experiment: i) to test the effect of a telegenic voice; and ii) to use a natural and personal expression of the subject. The questionnaire was built on categories extracted from the “La vive voix” (Fónagy 1983) and “L’Audio-Vision” (Chion 1990). The data obtained were treated statistically. Results of the qualitative and quantitative research seem to confirm a gestaltic perception of interpreting speech received through audio-vision and formed by the following components: sound-image; syllable-melody(-voice-personality), words-sentences. Results seem to raise doubts on the effectiveness of the quantitative approach to the analysis of speech perception.
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Nguyen, Duy Huy. "Enhancing and improving voice transmission quality over LTE network : challenges and solutions." Electronic Thesis or Diss., Evry, Institut national des télécommunications, 2017. http://www.theses.fr/2017TELE0002.
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LTE (Long Term Evolution) a été développé et normalisé par le 3GPP (3rd Generation Partnership Project). C’est un réseau à commutation de paquets. Cela signifie que la voix sur LTE (VoLTE) est un service de VoIP avec les exigences de qualité de service garantis au lieu de transmettre dans un réseau à commutation de circuits tels que les systèmes existants (2G/3G). VoLTE est déployé dans un réseau entièrement IP combinée avec IMS (IP Sous-système Multimédia). De ce fait, le déploiement de VoLTE est assez complexe et comment assurer la qualité de transmission de la voix sur les réseaux LTE est un très grand défi. Ainsi, il faut plusieurs solutions différentes pour renforcer et améliorer la qualité de transmission de la voix sur les réseaux LTE. Dans cette thèse, nous présentons des solutions en vue d’améliorer la qualité de transmission de la voix sur les réseaux LTE pour les services audio à bandes étroites et larges. Pour cela, il nous faudra différents facteurs complets en solutions. L’un d’eux est QoE (Qualité de l’Expérience) qui est une nouvelle tendance. Et afin de déterminer la perception des utilisateurs pour le service en temps réel tel que VoLTE, nous utilisons le E-model étendu et le WB (large bande) E-model pour des services audio à bandes étroites et larges respectivement. Les solutions proposées ici portent principalement sur des éléments clés dans les réseaux LTE, tels que le codage par chaine, MAC (Contrôle d’Accès Moyen) des systèmes de planification et la qualité de voix du moniteur décrits comme suit. Tout d’abord, des algorithmes améliorés pour renforcer le codec de la chaine LTE (codeur et décodeur) ont été proposés. Pour améliorer le codeur de chaine LTE, un algorithme d’adaptation conjointe a été déployé. Le but de cet algorithme est de minimiser la redondance générée par codage en chaine avec une légère réduction de la perception de l’utilisateur. Ensuite, afin d’améliorer le décodeur par chaine LTE, un algorithme amélioré Log-MAP a été présenté. Cet algorithme vise à obtenir la performance BER (Bit Error Rate) qui est le plus proche du Log-MAP avec une complexité de calcul réduite par rapport à l’état de l’art. Deuxièmement, la chaine et les systèmes QoS de planification améliorés de la perception de l’utilisateur et du mode de priorité VoIP ont été proposés. Ces planificateurs sont déployés à la fois pour les utilisateurs d’audio à larges et à étroites bandes. Les résultats numériques montrent qu’ils surpassent plusieurs planificateurs en vedette tels que FLS, M-LWDF et EXP/PF en termes de retard, de taux de perte de paquets, de débit cellulaire, d’indice et de l’équité et d’efficacité spectrale dans presque tous les cas. Enfin, pour assurer la qualité vocale de transmission sur le réseau LTE, la prédiction de la satisfaction des utilisateurs est essentielle. Pour cette raison, nous présentons deux modèles non intrusifs pour mesurer la qualité de la voix sur les réseaux LTE. Ces modèles sont utilisés pour les utilisateurs d’audio à bandes étroites et larges bandes. Les modèles proposés ne se réfèrent pas au signal original. Par conséquent, ils sont très appropriés pour prédire la qualité de l’appel vocal sur les réseaux LTELTE (Long Term Evolution) has been developed and standardized by 3GPP (3rd Generation Partnership Project). It is a packet-switched network. This means voice over LTE (VoLTE) is a VoIP service with the guaranteed QoS requirements instead of transmitted in a circuit-switched network such as the legacy system (2G/3G). Since VoLTE is deployed in an All-IP network combined with IMS (IP Multimedia Subsytem), thus, the VoLTE deployment is quite complex and how to ensure voice transmission quality over LTE networks is a very big challenge. Therefore, there needs to be many different solutions to enhance and improve voice transmission quality over LTE networks. In this dissertation, we present solutions to enhance and improve voice transmission quality over LTE networks for both narrowband and wideband audio services. In order to do that, there needs to be many different factors complemented in solutions. One of them is QoE (Quality of Experience) which is a new trend. And in order to determine user perception for real-time service such as VoLTE, we use extended E-model and WB (Wideband) E-model for narrowband and wideband audio services, respectively. The proposed solutions in this thesis mainly focus on key elements in LTE networks such as channel coding, MAC (Medium Access Control) scheduling schemes and monitor voice quality described as follows. First, enhanced/improved algorithms for enhancing LTE channel codec (coder and decoder) have been proposed. In order to enhance LTE channel coder, a joint source-channel code rate adaption algorithm has been deployed. The goal of this algorithm is to minimize redundancy generated by channel coding with a slight reduction of user perception. Next, in order to enhance LTE channel decoder, an improved Log-MAP algorithm has been presented. This algorithm aims at obtaining BER performance that is closest to the LOP-MAP with the computational complexity reduced in comparison with state-of-the-art. Second, channel- and QoS-Aware scheduling schemes with the enhancement of user perception and VoIP priority mode have been proposed. These schedulers are deployed for both narrowband and wideband audio users. The numerical results show that they outperform several featured schedulers such as FLS, M-LWDF, and EXP/PF in terms of delay, packet loss rate, cell throughput, fairness index, and spectral efficiency in almost cases. Last, in order to ensure voice transmission quality over LTE network, prediction of user satisfaction is essential. For this reason, we present two object non-intrusive models for measuring voice quality in LTE networks. These models are used for narrowband and wideband audio users. The proposed models do not refer to the original signal, thus, they are very suitable for predicting voice call quality in LTE networks
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Malandraki, Georgia. "Persisting Effects of Aspiration and Penetration on Voice Quality and Vocal Pitch." Ohio University / OhioLINK, 2004. http://rave.ohiolink.edu/etdc/view?acc_num=ohiou1103140461.
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Nguyen, Duy Huy. "Enhancing and improving voice transmission quality over LTE network : challenges and solutions." Thesis, Evry, Institut national des télécommunications, 2017. http://www.theses.fr/2017TELE0002/document.
Full textAbstract:
LTE (Long Term Evolution) a été développé et normalisé par le 3GPP (3rd Generation Partnership Project). C’est un réseau à commutation de paquets. Cela signifie que la voix sur LTE (VoLTE) est un service de VoIP avec les exigences de qualité de service garantis au lieu de transmettre dans un réseau à commutation de circuits tels que les systèmes existants (2G/3G). VoLTE est déployé dans un réseau entièrement IP combinée avec IMS (IP Sous-système Multimédia). De ce fait, le déploiement de VoLTE est assez complexe et comment assurer la qualité de transmission de la voix sur les réseaux LTE est un très grand défi. Ainsi, il faut plusieurs solutions différentes pour renforcer et améliorer la qualité de transmission de la voix sur les réseaux LTE. Dans cette thèse, nous présentons des solutions en vue d’améliorer la qualité de transmission de la voix sur les réseaux LTE pour les services audio à bandes étroites et larges. Pour cela, il nous faudra différents facteurs complets en solutions. L’un d’eux est QoE (Qualité de l’Expérience) qui est une nouvelle tendance. Et afin de déterminer la perception des utilisateurs pour le service en temps réel tel que VoLTE, nous utilisons le E-model étendu et le WB (large bande) E-model pour des services audio à bandes étroites et larges respectivement. Les solutions proposées ici portent principalement sur des éléments clés dans les réseaux LTE, tels que le codage par chaine, MAC (Contrôle d’Accès Moyen) des systèmes de planification et la qualité de voix du moniteur décrits comme suit. Tout d’abord, des algorithmes améliorés pour renforcer le codec de la chaine LTE (codeur et décodeur) ont été proposés. Pour améliorer le codeur de chaine LTE, un algorithme d’adaptation conjointe a été déployé. Le but de cet algorithme est de minimiser la redondance générée par codage en chaine avec une légère réduction de la perception de l’utilisateur. Ensuite, afin d’améliorer le décodeur par chaine LTE, un algorithme amélioré Log-MAP a été présenté. Cet algorithme vise à obtenir la performance BER (Bit Error Rate) qui est le plus proche du Log-MAP avec une complexité de calcul réduite par rapport à l’état de l’art. Deuxièmement, la chaine et les systèmes QoS de planification améliorés de la perception de l’utilisateur et du mode de priorité VoIP ont été proposés. Ces planificateurs sont déployés à la fois pour les utilisateurs d’audio à larges et à étroites bandes. Les résultats numériques montrent qu’ils surpassent plusieurs planificateurs en vedette tels que FLS, M-LWDF et EXP/PF en termes de retard, de taux de perte de paquets, de débit cellulaire, d’indice et de l’équité et d’efficacité spectrale dans presque tous les cas. Enfin, pour assurer la qualité vocale de transmission sur le réseau LTE, la prédiction de la satisfaction des utilisateurs est essentielle. Pour cette raison, nous présentons deux modèles non intrusifs pour mesurer la qualité de la voix sur les réseaux LTE. Ces modèles sont utilisés pour les utilisateurs d’audio à bandes étroites et larges bandes. Les modèles proposés ne se réfèrent pas au signal original. Par conséquent, ils sont très appropriés pour prédire la qualité de l’appel vocal sur les réseaux LTELTE (Long Term Evolution) has been developed and standardized by 3GPP (3rd Generation Partnership Project). It is a packet-switched network. This means voice over LTE (VoLTE) is a VoIP service with the guaranteed QoS requirements instead of transmitted in a circuit-switched network such as the legacy system (2G/3G). Since VoLTE is deployed in an All-IP network combined with IMS (IP Multimedia Subsytem), thus, the VoLTE deployment is quite complex and how to ensure voice transmission quality over LTE networks is a very big challenge. Therefore, there needs to be many different solutions to enhance and improve voice transmission quality over LTE networks. In this dissertation, we present solutions to enhance and improve voice transmission quality over LTE networks for both narrowband and wideband audio services. In order to do that, there needs to be many different factors complemented in solutions. One of them is QoE (Quality of Experience) which is a new trend. And in order to determine user perception for real-time service such as VoLTE, we use extended E-model and WB (Wideband) E-model for narrowband and wideband audio services, respectively. The proposed solutions in this thesis mainly focus on key elements in LTE networks such as channel coding, MAC (Medium Access Control) scheduling schemes and monitor voice quality described as follows. First, enhanced/improved algorithms for enhancing LTE channel codec (coder and decoder) have been proposed. In order to enhance LTE channel coder, a joint source-channel code rate adaption algorithm has been deployed. The goal of this algorithm is to minimize redundancy generated by channel coding with a slight reduction of user perception. Next, in order to enhance LTE channel decoder, an improved Log-MAP algorithm has been presented. This algorithm aims at obtaining BER performance that is closest to the LOP-MAP with the computational complexity reduced in comparison with state-of-the-art. Second, channel- and QoS-Aware scheduling schemes with the enhancement of user perception and VoIP priority mode have been proposed. These schedulers are deployed for both narrowband and wideband audio users. The numerical results show that they outperform several featured schedulers such as FLS, M-LWDF, and EXP/PF in terms of delay, packet loss rate, cell throughput, fairness index, and spectral efficiency in almost cases. Last, in order to ensure voice transmission quality over LTE network, prediction of user satisfaction is essential. For this reason, we present two object non-intrusive models for measuring voice quality in LTE networks. These models are used for narrowband and wideband audio users. The proposed models do not refer to the original signal, thus, they are very suitable for predicting voice call quality in LTE networks
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Brackin, Margueritte Patricia Dodd. "Translating the voice of the customer into preliminary design specifications." Diss., Georgia Institute of Technology, 1997. http://hdl.handle.net/1853/17936.
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Li, Hong. "Quality-of-service routing for Voice-over-IP in service overlay networks." Thesis, McGill University, 2010. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=86808.
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Voice-over-IP (VoIP) becomes more and more popular with the development of service convergence in the next generation network. This thesis focuses on improving VoIP quality with application-layer routing in service overlay networks.Internet end-to-end delay is one of the most important impairments on VoIP quality. We therefore analyze, model and simulate it to better understand it and thus to discover potential advantages of routing in service overlay networks. Based on the investigation of the Internet end-to-end delay, we find that VoIP quality on a pair of diverse paths is better and more stable than that on a single path. We therefore propose a novel centralized data fusion approach to search for the best pair of diverse paths. This method jointly optimizes source routing with adaptive play-out scheduling at the receiver. It requires transmitting the delay distributions of all the overlay links for estimating the delay distributions of diverse paths. We propose to transmit only the model parameters of the link delay distributions to reduce the communication overhead. It is shown that the best pair of diverse paths can be estimated with a small error.
Nonetheless, the centralized approach is computationally expensive. We therefore propose an online diverse routing method, which uses distributed learning automata to actively probe path delays and to determine the best pair of diverse paths for VoIP based on the state of the learning automata. We have demonstrated the scalability and the optimality of the approach by simulations, and proven the optimality of the approach using Kushner's weak convergence method. VoIP quality has been shown to improve from unsatisfactory levels to satisfactory levels. In addition, we propose a method to detect and recover from link failures based on the state of the learning automata. Considerable improvement in link failure recovery time has been achieved.
In sum, this work demonstrates that the proposed centralized diverse routing approach is effective to improve VoIP quality in terms of R-factor for small overlay networks, and that the proposed distributive diverse routing approach together with the link failure detection scheme provides a scalable, effective and robust solution to VoIP routing for large overlay networks.
Voix sur IP (VoIP) est un service dont la popularité crôıt avec le développement de la convergence entre les services dans les réseaux dits de nouvelle génération. Dans cette thèse, nous nous appliquons à améliorer la qualité des services de VoIP grâce au routage au niveau la couche application en utilisant des réseaux dédiés. Dans cette thèse, nous procédons à l'étude des délais de bout en bout du réseau Internet, qui sont le facteur impactant le plus sur la qualité de la VoIP. Nous analysons, modelons et synthétisons des traces de délais de bout en bout, afin de découvrir un potentiel intérêt relatif à leur utilisation dans le cadre du routage au niveau le la couche application utilisant des réseaux dédiés.
En nous appuyant sur l'étude des traces de délais de bout en bout, nous montrons que la qualité de la VoIP peut être améliorée et stable en utilisant un couple de routes diverses, au lieu d'une seule route. Nous donc proposon un centre de fusion de données qui utilise notre approche pour trouver le meilleur couple de routes diverses. Cette méthode optimise le routage source conjointement avec adaptation play-out au niveau du récepteur. Il faut transmettre des délais des-dites distributions de tous les liens au niveau le la couche application pour estimer des distributions de tous les couples de routes possibles. Nous proposons de transmettre uniquement les paramètres du modèle de la distribution des délais, afin de réduire des côuts de communication. Nous prouvons que cette méthode peut trouver le meilleur couple de routes à une faible erreur.
Comme la technique centralisée requiert une grande puissance de calcul, nous proposons une solution de routage divers extensible en ligne, qui utilise l'apprentissage distribué au- tomata activement sonde des délais de bout en bout et détermine la meilleure paire de diverses voies de VoIP basé sur l'état de l'apprentissage d'automates. Nous avons dé- montré l'extensibilité et l'optimalité de cette approche par les simulations, et démontré l'optimalité de l'approche par l'utilisation de la méthode de convergence faible de Kushner. Nous montrons que la qualité de la VoIP est ameliorée, passant d'une qualité inacceptable à une qualité acceptable. De plus, nous proposons une méthode pour détecter les défaillances du lien et de sa récupération sur la base de paramètres de l'apprentissage d'automates, qui permettent une réduction considérable du temps de récupération à la suite de la défaillance d'un lien.
En somme, cette thèse démontre que la proposition de la diversité de routage centralisé approche est efficace pour améliorer la qualité de la VoIP en termes de R-facteur pour les petits réseaux de la couche application, et que l'apprentissage de un couple de routes diverses avec des méthodes de détection de défaillance offre un extensible, efficace et robuste solution pour services de VoIP grâce au grands réseaux de la couche application.
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Qiao, Zizhi. "Enhancement of perceived quality of service for Voice over Internet Protocol systems." Thesis, University of Plymouth, 2008. http://hdl.handle.net/10026.1/329.
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Voice over Internet Protocol (VoIP) applications are becoming more and more popular in the telecommunication market. Packet switched VoIP systems have many technical advantages over conventional Public Switched Telephone Network (PSTN), including its efficient and flexible use of the bandwidth, lower cost and enhanced security. However, due to the IP network's 'Best Effort' nature, voice quality are not naturally guaranteed in the VoIP services. In fact, most current VoIP services can not provide as good a voice quality as PSTN. IP Network impairments such as packet loss, delay and jitter affect perceived speech quality as do application layer impairment factors, such as codec rate and audio features. Current perceived Quality of Service (QoS) methods are mainly designed to be used in a PSTN/TDM environment and their performance in VoIP environment is unknown. It is a challenge to measure perceived speech quality correctly in VoIP system and to enhance user perceived speech quality for VoIP system. The main goal of this project is to evaluate the accuracy of the existing ITU-T speech quality measurement method (Perceptual Evaluation of Speech Quality - PESQ) in mobile wireless systems in the context of VoIP, and to develop novel and efficient methods to enhance the user perceived speech quality for emerging VoIP services especially in mobile VoIP environment. The main contributions of the thesis are threefold: (1) A new discovery of PESQ errors in mobile VoIP environment. A detailed investigation of PESQ performance in mobile VoIP environment was undertaken and included setting up a PESQ performance evaluation platform and testing over 1800 mobile-to-mobile and mobile-to- PSTN calls over a period of three months. The accuracy issues of PESQ algorithm was investigated and main problems causing inaccurate PESQ score (improper time-alignment in the PESQ algorithm) were discovered. Calibration issues for a safe and proper PESQ testing in mobile environment were also discussed in the thesis. (2) A new, simple-to-use, VoIP jitter buffer algorithm. This was developed and implemented in a commercial mobile handset. The algorithm, called 'Play Late Algorithm' adaptively alters the playout delay inside a speech talkspurt without introducing unnecessary extra end-to-end delay. It can be used as a front-end to conventional static or adaptive jitter buffer algorithms to provide improved performance. Results show that the proposed algorithm can increase user perceived quality without consuming too much processing power when tested in live wireless VoIP networks. (3) A new QoS enhancement scheme. The new scheme combines the strengths of adaptive codec bit rate (i.e. AMR 8-modes bit rate) and speech priority marking (i.e. giving high priority for the beginning of a voiced segment). The results gathered on a simulation and emulation test platform shows that the combined method provides a better user perceived speech quality than separate adaptive sender bit rate or packet priority marking methods.
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Manickam, Kathiresan. "Objective voice quality modelling and analysis of vocal fold functionality in radiotherapy." Thesis, Liverpool John Moores University, 2004. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.421413.
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Härkönen, P. (Pauli). "The Acoustic Voice Quality Index äänihuulihalvauspotilaiden ja spasmodisten dysfoniapotilaiden äänen laadun mittarina." Master's thesis, University of Oulu, 2018. http://urn.fi/URN:NBN:fi:oulu-201802081193.
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Tämän pro gradu -tutkielman tarkoituksena oli selvittää tietokonepohjaisen objektiivisen The Acoustic Voice Quality Index -mittarin (AVQI) soveltuvuutta suomenkielisten puhujien äänen laadun mittaukseen. Tutkimuksessa kysyttiin, pystyykö mittari erottamaan terveet äänet äänihäiriöistä. Tutkitut häiriöryhmät olivat äänihuulihalvaus ja spasmodinen dysfonia. Myös äänihäiriöryhmien väliltä etsittiin eroja AVQI-mittauksilla.
Tähän tutkimukseen osallistui yhteensä 95 tutkittavaa: äänihuulihalvauksesta kärsivät (n = 20), spasmodisesta dysfoniasta kärsivät (n = 19) ja terveääniset puhujat (n = 56). Kaikki olivat äidinkieleltään suomea puhuvia aikuisia. Äänihäiriöistä kärsivät puhujat olivat Tampereen yliopistollisen sairaalan foniatrian poliklinikan potilaita, joilla oli oltava diagnosoituna oikea äänihäiriö. Tutkimuksen aineistona käytetyt äänitykset kerättiin terveäänisten osalta Tampereen yliopistolla ja äänihäiriöstä kärsivien osalta TAYS:n foniatrisella poliklinikalla, joissa kummassakin oli käytössä sama äänityslaitteisto. Kultakin puhujalta kerättiin kolme pidennetyn vokaalin näytettä sekä luentanäyte. Äänitiedostot analysoitiin AVQI-skriptillä Praat-puheanalyysiohjelmassa. Tulokseksi saadut AVQI-arvot muodostuivat painotetusti kuudesta eri äänenlaatua mittaavasta akustisesta parametrista.
Tässä tutkimuksessa löydettiin tilastollisesti erittäin merkitsevää eroa verrattaessa terveäänisten puhujien AVQI-arvojen keskiarvoa kumman tahansa äänihäiriöryhmän keskiarvoon. Myös kaikkien paitsi yhden AVQI:n osaparametrin keskiarvojen väliltä löydettiin tilastollisesti erittäin merkitsevät erot verrattaessa terveäänisten ryhmää äänihäiriöryhmiin. Ainoastaan piirteessä tilt of trendline through spectrum (spektrin regressiosuoran kulma) ei tilastollista merkitsevyyttä löytynyt kontrolliryhmän ja spasmodisesta dysfoniasta kärsivien väliltä. Piirre kuitenkin erotti ainoana piirteenä äänihuulihalvauksen ja spasmodisen dysfonian toisistaan tilastollisesti merkitsevästi.
Tutkimus antaa näyttöä siitä, että AVQI on tehokas erottamaan häiriintyneen äänen häiriöttömästä suomenkielisilläkin puhujilla. Tuloksia voidaan pitää suuntaa antavina, eivätkä ne ole automaattisesti yleistettävissä kaikkiin äänihäiriöryhmiin. Varmuutta ei saatu siitä, erottaako AVQI äänihäiriöitä laadullisesti toisistaan.
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Alves, Maxine. "The effect of hydration of voice quality in adults : a systematic review." Diss., University of Pretoria, 2017. http://hdl.handle.net/2263/65554.
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Objectives. To critically appraise scientific, peer-reviewed articles, published in the past 10 years on the effects of different levels of hydration on voice quality in adults. Study design. Systematic review. Method. Five databases were searched using the key words “vocal fold hydration/dehydration”, “voice quality”, “and “hygienic voice therapy”. The PRISMA-P guidelines were followed. The included studies were scored based on ASHA’s levels of evidence and quality indicators, as well as, the Cochrane Collaboration’s risk of bias tool. Results. Systemic dehydration as a result of fasting and not ingesting fluids had a significant, negative effect on the parameters of NHR, shimmer, jitter, frequency and the s/z ratio. Water ingestion led to significant improvements in shimmer, jitter, frequency and MPT values. Caffeine does not appear to negatively affect voice production. Laryngeal desiccation challenges by oral breathing led to surface dehydration which negatively affected jitter, shimmer, NHR, PTP and PPE. Steam inhalation significantly improved NHR, shimmer and jitter. Only nebulization of sterile water, isotonic solution and saline solution improved PTP, throat and mouth dryness and fundamental frequency respectively. An indication of a potential positive effect of nebulization substances was observed. Treatments in high humidity environments prove to be effective and adaptations of low humidity environments should be encouraged. Conclusion. Recent literature regarding vocal hydration is high quality evidence. Systemic hydration is the easiest and most cost effective solution to improve voice quality. Surface hydration using steam inhalation and nebulization as well as environmental modification can be suggested for professional voice users. Recent evidence therefore supports the inclusion of hydration in a vocal hygiene program.Dissertation (MCommunication Pathology)--University of Pretoria, 2017.
Speech-Language Pathology and Audiology
MCommunication Pathology
Unrestricted
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Malandraki, Georgia A. "Persisting effects of aspiration and penetration on voice quality and vocal pitch." Ohio : Ohio University, 2004. http://www.ohiolink.edu/etd/view.cgi?ohiou1103140461.
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Majed, Najmeddine. "Measuring and improving the quality of experience of mobile voice over IP." Thesis, Ecole nationale supérieure Mines-Télécom Atlantique Bretagne Pays de la Loire, 2018. http://www.theses.fr/2018IMTA0099/document.
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Les réseaux mobiles 4G basés sur la norme LTE (Long Term Evolution), sont des réseaux tout IP. Les différents problèmes de transport IP comme le retard, la gigue et la perte despaquets peuvent fortement dégrader la qualité des communications temps réel telles que la téléphonie. Les opérateurs ont mis en oeuvre des mécanismes d’optimisation du transport de la voix dans le réseau afin d'améliorer la qualité perçue. Cependant, les algorithmes propriétaires de gestion de la qualité dans les terminaux ne sont pas spécifiés dans les standards. Dans ce contexte, nous nous intéressons aux mécanismes d'adaptation de média, intégrés dans les terminaux afin d'améliorer la qualité d’expérience (QoE). En particulier, nous évaluons de manière expérimentale des métriques QoE de la voix sur LTE (VoLTE) en utilisant une méthode de test standardisée. Nous proposons d’améliorer la méthode de test et discutons la manière dont cette méthode peut être étendue pour évaluer les performances du buffer de gigue. Nous évaluons également de manière expérimentale la qualité de WebRTC dans différentes conditions radios en utilisant un réseau réel. Nous évaluons l'impact du buffer de gigue et de la variation du débit sur la qualité mesurée. Pour améliorer la robustesse des codecs contre la perte de paquets, nous proposons d’utiliser une redondance simple au niveau applicatif. Nous implémentons cette redondance pour le codec EVS (Enhanced Voice Service) et nous évaluons ses performances. Enfin, nous proposons un protocole de signalisation qui permet d’envoyer des requêtes de redondance au cours d’une communication afin d’activer ou désactiver celle-ci dynamiquementFourth-generation mobile networks, based on the Long Term Evolution (LTE) standard, are all- IP networks. Thus, mobile telephony providers are facing new types of quality degradations related to the voice packet transport over IP network such as delay, jitter and packet loss. These factors can heavily degrade voice communications quality. The real-time constraint of such services makes them highly sensitive to delay and loss. Network providers have implemented several network optimizations for voice transport to enhance perceived quality. However, the proprietary quality management algorithms implemented in terminals are left unspecified in the standards. In this context, we are interested in media adaptation mechanisms integrated in terminals to enhance the overall Quality of Experience (QoE). In particular, we experimentally evaluate Voice over LTE (VoLTE) QoE metrics such as delay and Mean Opinion Score (MOS) sing a standardized test method. We propose some enhancements to the actual test method and discuss how this method can be extended to evaluate de-jitter buffer performance. We also experimentally evaluate WebRTC voice quality in different radio conditions using a realLTE test network. We evaluate the impact of jitter buffer and bit rate variations on the measured quality. To enhance voice codec robustness against packet loss, we propose a simple application layer redundancy. We implemented it for the Enhanced Voice Service (EVS) codec and evaluate it. Finally, we propose a signaling protocol that allows sending redundancy requests during a call to dynamically activate or deactivate the redundancy mechanism
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Tucker, Beatrice M. "The student voice: Using student feedback to inform quality in higher education." Thesis, Curtin University, 2015. http://hdl.handle.net/20.500.11937/2158.
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This thesis presents a scholarly synthesis of a series of original published works providing evidence that student feedback, collected and analysed using valid and defensible methods, is effective in improving the quality of teaching and learning in higher education. The key factors underpinning the effective use of student feedback for quality improvement of teaching and learning in higher education are revealed. Students’ perceptions of their experience in achieving learning outcomes are reported.
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El, Malki Karim. "A novel approach to high quality voice using echo cancellation and silence detection." Thesis, University of Sheffield, 1998. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.286579.
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Knight, Elizabeth Johnson. "The Effect of Head Flexion/extension on Acoustic Measures of Singing Voice Quality." Thesis, University of North Texas, 2013. https://digital.library.unt.edu/ark:/67531/metadc500127/.
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A study was undertaken to identify the effect of head flexion/extension on singing voice quality. The amplitude of the fundamental frequency (F0), and the singing power ratio (SPR), an indirect measure of singer’s formant activity, were measured. F0 and SPR scores at four experimental head positions were compared with the subjects’ scores at their habitual positions. Three vowels and three pitch levels were tested. F0 amplitudes and low frequency partials in general were greater with more extended head positions, while SPR increased with neck flexion. No effect of pitch or vowel was found. Gains in SPR appear to be the result of damping low frequency partials rather than amplifying those in the singer’s formant region. Raising the amplitude of F0 is an important resonance tool for female voices in the high range, and may be of benefit to other voice types in resonance, loudness, and laryngeal function.
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Santos, Aline Oliveira. "Parâmetros acústicos e perceptivo-auditivos da voz de adultos e idosos." Universidade de São Paulo, 2012. http://www.teses.usp.br/teses/disponiveis/25/25143/tde-12062012-161204/.
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Estudos revelam que homens e mulheres apresentam modificações vocais em decorrência do envelhecimento, entretanto, a maioria compara indivíduos jovens e idosos, agrupando-os em grandes intervalos etários. Estudar indivíduos da faixa etária próximas à terceira idade e compará-los em menores intervalos etários pode ser mais sensível para evidenciar características importantes. Objetivou-se verificar: quais as diferenças dos parâmetros acústicos e perceptivo-auditivos da voz de homens e mulheres de diversas décadas etárias; quais características vocais modificam com o avanço da idade, e determinar a relação entre as características perceptivo-auditivas e acústicas nessa população. Participaram do estudo 125 homens e 140 mulheres, com idades entre 30 e 79 anos, agrupados por décadas etárias. Por meio de uma escala analógica visual, foram avaliados, por três juízes, o grau geral do desvio vocal (G), rugosidade (R) e soprosidade (S) de fala encadeada e vogal sustentada. Foram analisados por meio do programa Mult Dimension Voice Program (KayPentax) os parâmetros frequência fundamental (F0), desvio-padrão da F0 (dp F0), jitter (%), shimmer (%), proporção ruído-harmonico (NHR), índice de turbulência vocal (VTI) e índice de fonação suave (SPI). A comparação entre os grupos foi realizada por meio de ANOVA e Tukey, as correlações, por meio do teste de Pearson, (significância de 5%). Na fala, homens e mulheres de 30-49 anos apresentaram menor G e R que os de idade superior a 50 anos (p<0,000) e mulheres com idade entre 50-59 anos apresentaram maior S que as de 60-79 (p=0,026). Em ambos os gêneros, à medida que a idade aumentou, maiores foram G e R durante a fala, enquanto que S reduziu durante a vogal de mulheres (p=0,005). A análise acústica mostrou que VTI foi maior em sujeitos de 70-79 anos em relação aos de 40-49 (p<0,040). O SPI dos sujeitos de 40-49 anos foi o maior (p<0,000). Houve correlação positiva entre o avanço da idade e dp F0 e NHR nos homens (p<0,000 e 0,023), e negativa para SPI nas mulheres (p=0,025). Quanto mais elevada a F0 da voz masculina, maior S (p=0,043); quanto mais reduzida a F0 da voz feminina, maior R (p=0,006). Conclui-se que é importante estudar sujeitos da faixa de transição entre a fase adulta e idosa, visto as diferenças de qualidade vocal em sujeitos maiores de 50 anos em relação aos mais jovens. Agrupar os sujeitos entre décadas etárias colabora para a compreensão do envelhecimento vocal, haja vista os sujeitos da sétima década que apresentaram maior VTI que os da quarta, enquanto estes últimos apresentam maior SPI que os das demais faixas etárias, além de ter evidenciado as diferenças relacionadas à soprosidade, que foi maior nas mulheres de meia idade que em idosas. Para homens e mulheres, quanto maior o grau geral e a rugosidade, maiores são os valores relacionados à instabilidade de frequência, perturbação de frequência e intensidade e medidas de ruído. Já para a soprosidade, a correlação se deu apenas para a instabilidade de frequência, perturbação de frequência e intensidade e SPI. A F0 correlacionou-se com a qualidade vocal de forma distinta entre homens e mulheres da faixa etária estudada.A number of studies have found that men and women can present vocal changes as a result of aging; meanwhile, most of the studies compare young and elderly people, grouping them in large age ranges. Reducing the subjects to adult and seniors age groups and comparing them in smaller age ranges can be more sensible to evidence significant characteristics. The purpose of this study was to verify differences on acoustic measures and perceptual analysis of the voice of adults and seniors, which of them are modified by aging and set the relation between perceptual analysis and acoustic measures on this population. Two hundred and sixty-five, men (n=125) and women (n=140) from 30 to 79 years-old, grouped into decade age ranges had their voice evaluated by 3 judges. Speech samples and sustained vowels were submitted to perceptive analysis consisted of the assessment of grade of overall deviation (G), roughness (R) and breathiness (B), using a visual-analog scale. Acoustic measures of speaking fundamental frequency (F0) and its standard deviation (sdF0), jitter (%),shimmer (%), noise-harmonic ratio (NHR), voice turbulence index (VTI) and soft phonation index (SPI) were assessed by Multi-Dimensional Voice Program (Kay Pentax). The comparison among the groups was held by ANOVA and Tukey and the correlations by Pearson\'s test (5% significance). During speech, men and women from 30-49 years-old have presented less G and R than the subjects 50-older (p<0,000) and women aged 50-59 had a greater B than women of 60-79 years-old (p=0,026). The parameters G and R increased with aging for men and women at the speech task, and B reduced in women at the sustained vowel task (p=0,005). About the acoustic measures, VTI was greater in subjects of 70-79 year-old than 40-49 ones (p<0,040). SPI of subjects from 40-49 years-old was the greatest. Positive correlation was found between aging, sdF0 and NHR in men (p<0,000 e 0,023), and negative for SPI in women (p=0,025). The higher F0 of mens voice, the greater is B (p=0,043); the more reduced F0 on women voice, the greater is R (p=0,006). Its relevant to study subjects on transition from adult to senior ages, since the differences on voice quality in subjects 50 or older are greater than in young people. Grouping the subjects by decade contributed to better understand of vocal aging. For instance, the 70 or older group have shown a greater VTI than people in their 40s, while this last group have shown a bigger SPI than others age ranges, in addition to evidenced differences related to breathiness that washigher in middle age women than in elderly. For both genders the bigger the general voice deviation, and the roughness, the bigger are parameters related to instability of frequency, its disturbance, intensity and noise ratios. With regard to breathiness, the correlation happens only to frequency instability, its disturbance, intensity and SPI. The correlation between F0 and vocal quality was different to men and women of the studied age ranges.
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Silva, Daniel Pestana da. "Proposta de periodização do treinamento vocal com técnica de vibração sonorizada de língua." Universidade de São Paulo, 2016. http://www.teses.usp.br/teses/disponiveis/25/25143/tde-08082016-083746/.
Full textAbstract:
O treinamento físico é um processo repetitivo e sistemático composto de exercícios físicos progressivos que tem por objetivo aperfeiçoar o desempenho. Um dos princípios da fisiologia do exercício é a sobrecarga que se baseia em desenvolver o treinamento com treinos intervalados, aumento de intensidade e de volume. O objetivo deste estudo foi avaliar o efeito da Proposta de Periodização do Treinamento Vocal (PPTV) com técnica de vibração sonorizada de língua na qualidade vocal de mulheres sem queixas vocais e com vozes saudáveis. Estudo prospectivo, controlado e randomizado, desenvolvido após aprovação do Comitê de Ética em Pesquisa com Seres Humanos da FOB/USP (parecer 1.198.625). Participaram do estudo 30 mulheres entre 18 e 39 anos, vocalmente saudáveis (evidenciadas por protocolo de sintomas vocais/laríngeos, avaliação vocal e laringológica inicial), divididas em dois grupos de forma randomizada: grupo experimental (GE) com 15 mulheres que receberam seis sessões da PPTV; grupo controle (GC) com 15 mulheres que receberam seis sessões de treinamento vocal tradicional. Após assinatura do termo de consentimento livre e esclarecido, as voluntárias passaram pelas avaliações antes e após treinamento e, após 30 dias do treinamento. As avaliações realizadas foram: investigação das sensações na voz, laringe, respiração e articulação (sessão a sessão); mensuração da intensidade vocal habitual; mensuração das medidas fonatórias; gravação vocal para posterior análise perceptivo-auditiva e acústica da voz. Para ambos os grupos o treinamento consistiu de 12 minutos de execução da técnica de vibração sonorizada de língua (TVSL), em pitch habitual. O treinamento vocal do GE (PPTV) considerou o princípio da sobrecarga, com administração da intensidade vocal e com intervalos controlados de execução da TVSL (30 segundos) e repouso (30 segundos). As voluntárias do GC executaram a TVSL de forma tradicional, com período de descanso a cada três minutos, mas sem controle do tempo de repouso. Os testes estatísticos (nível de significância de 0,05) revelaram que as sensações após o treino, controladas sessão a sessão, foram positivas para ambos os grupos (Teste de Sinais). As sensações positivas na voz e na articulação foram as mais relatadas por ambos os grupos. O teste ANOVA, seguido de Tukey revelou que o GE apresentou aumento significante da intensidade vocal habitual após a PPTV e 30 dias após na emissão da vogal /a/, o que não ocorreu com o GC; não houve diferença significante nas medidas fonatórias após o treino vocal em ambos os grupos. A análise perceptivo-auditiva (teste de Sinais) revelou que o parâmetro de instabilidade melhorou significantemente no GE após a PPTV e 30 dias após, o que não ocorreu com o GC. Por outro lado, o GC apresentou piora significante 30 dias após o treinamento vocal tradicional do parâmetro tensão. Por meio do teste ANOVA e Tukey, a análise acústica revelou melhora significante nos valores de jitter e variação da frequência (Vf0) no GE, 30 dias após, o que não ocorreu com o GC. Entretanto, o GC apresentou melhora significante do índice de fonação suave após o treino vocal tradicional, mas que não se sustentou 30 dias após. Este estudo permitiu concluir que a PPTV, com uso da TVSL foi capaz de produzir efeitos na qualidade vocal, com melhora da instabilidade vocal, intensidade vocal habitual e medidas acústicas (Vf0 e jitter) quando comparados ao treinamento vocal tradicional, em mulheres vocalmente saudáveis. O treinamento vocal proposto não influenciou negativamente nos relatos de sensações na voz, laringe, respiração e articulação. Conclui-se que o treino com o princípio da sobrecarga, com intensidade e intervalo controlados, levou à adaptação do sistema vocal, em relação à instabilidade. Portanto, os achados deste estudo tornam necessária a reflexão sobre a prática e execução das técnicas e treinamentos vocais tradicionais, ressaltando a importância dos princípios da fisiologia do exercício nas práticas fonoaudiológicas na clínica vocal.Physical training is a repetitive and systematic process composed of physical progressive exercises which aims to improve the performance. One of the principles of exercise physiology is the overload that is based on develop the training with interval training. The aim of this study was to evaluate the effect of Proposal of Periodization of the Vocal Training (PPVT) with sonorous tongue vibration technique in the vocal quality of women without vocal complaints and healthy voices. Prospective, controlled and randomized study, developed after approval by Human Research Ethics Committees (HRECs) from FOB/USP (purport 1.198.625).The study included 30 women aged 18 to 39, vocally healthy (evidenced through protocol of symptoms vocal/laryngeal, vocal assessment and initial laryngological), divided into two groups randomly: experimental group (EG) with 15 women who received six sessions of PPVT; control group (CG) with 15 women who received six sessions of traditional vocal training. After signing of the informed consent, the volunteers through the evaluations before and after training and, after 30 days of training. The assessments performed were: research of the sensations in the voice, larynx, breathing and articulation (session to session); measurement of the usual vocal intensity; measurement of phonatory measures; vocal recording for later analysis-perceptive-auditory and acoustics of the voice. For both groups the training consisted of 12 minutes of execution of sonorous tongue vibration technique (STVT) in usual pitch. Vocal training from EG (PPVT) considered the principle of overload, with administration of vocal intensity and controlled execution intervals STVT (30 seconds) and rest (30 seconds). The volunteers of the CG performed the STVT in a traditional way, with rest period every three minutes, but without the rest time control. The statistical tests (significance level of 0.05) showed that the sensations after training, controlled session to session were positive for both groups (signal Test). The positive sensations in the voice and articulation were the most reported by both groups. The test ANOVA, followed by Tukey revealed that the EG presented significant increase of usual vocal intensity after PPVT 30 days after the issuance of the vowel /a/, which did not occur with the CG; There was no significant difference in the phonatory measures after the vocal training in both groups. Perceptive-auditory analysis (Signal test) revealed that the instability parameter has improved significantly in the EG after the PPVT and 30 days after, which did not occur with the GC. On the other hand, the GC presented significant worsening 30 days after the traditional vocal training of tension parameter. Through ANOVA and Tukey test, acoustic analysis revealed significant improvement in jitter values and frequency variation (Vf0) in EG, 30 days after, what did not happened with the CG. Though, the CG presented significant improvement of the soft phonation index after the traditional vocal training, but it did not hold 30 days after.This study made it possible to conclude that the PPVT, with the use of the STVT was able to produce effects on vocal quality, with improved vocal instability, usual vocal intensity and acoustic measures (Vf0 and jitter) when compared to the traditional vocal training, women healthy vocally. The proposed vocal training did not influence negatively in the accounts of sensations in the voice, larynx, breathing and articulation. It is concluded that the training with the principle of overload, with intensity and controlled interval, led to the adaptation of the vocal system in relation to the instability. Therefore, the findings of this study make it necessary the reflection on the practice and implementation of techniques and traditional vocal training, emphasizing the importance of the principles of exercise physiology in the speech therapy practices at the clinic.
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Masson, Maria Lúcia Vaz. "Aula, repouso, aquecimento e desaquecimento vocal em professores de uma escola pública de ensino médio de Salvador - BA /." Marília : [s.n.], 2009. http://hdl.handle.net/11449/102209.
Full textAbstract:
Orientador: Maria de Lourdes Morales HoriguelaBanca: Léslie Piccolotto Ferreira
Banca: Maria Suzana Behlau
Banca: Eliana Maria Gradim Fabron
Banca: Simone Aparecida Capellini
Resumo: Justificativa: professores compõem uma categoria profissional que tem a voz como um dos cruciais instrumentos de trabalho. Fatores de risco os expõem a alterações vocais que podem prejudicar o exercício profissional. Dentre os aspectos considerados no cuidado com a voz, o aquecimento e o desaquecimento vocal possibilitam a preparação da voz para o uso em sala de aula. Objetivo: analisar os efeitos da aula, do repouso e de um procedimento de aquecimento e desaquecimento vocal na qualidade da voz e no grau de desconforto auto-referido. Métodos: Dezoito professores de uma escola pública estadual do município de Salvador-BA, selecionada por conveniência, compuseram a amostra deste estudo. Dividiram-se as amostras entre grupos experimental (n=8) e controle (n=10) e analisaram-se os efeitos da aula e repouso (grupo controle), aquecimento e desaquecimento vocal (grupo experimental), considerando-se a avaliação de juízes por meio da escala GRBASI e ressonância, análise acústica e grau de desconforto auto-referido. O material analisado foi submetido a tratamento estatístico, sendo considerado nível de significância de 5%. Resultados: A aula provocou elevação estatisticamente significante da freqüência fundamental e aumento do grau de desconforto auto-referido. Não houve diferença significante nas variáveis analisadas para o repouso vocal. O aquecimento vocal proporcionou redução do grau geral de alteração vocal e diminuição do desconforto, especialmente nos aspectos relacionados ao corpo. O desaquecimento vocal proporcionou diminuição da freqüência fundamental e redução do grau de desconforto, especialmente relacionado à voz. A comparação entre os grupos experimental e controle não demonstrou diferença estatisticamente significante em nenhuma variável analisada. Tanto o desaquecimento quanto o repouso vocal proporcionaram o retorno da voz ao ajuste coloquial ...(Resumo completo, clicar acesso eletrônico abaixo)
Abstract:Justification: Voice is a main work tool for teachers and they are exposed to risk factors able to cause voice disorders. Vocal warm-up and cool-down procedures are used as preparation for the use of voice in classrooms. Goal: to analyze the effects of vocal warm-up and cool-down procedure, rest and lecture on voice quality, as well as on the degree of self-reported discomfort. Methods: the subjects of this research were 18 teachers of a state school in Salvador/BA, which were selected by convenience. They were divided in two groups, one experimental (n=8) and the control (n=10). Then they were subject to an analysis of the effects of lecture and rest (control group), vocal warmup and cool-down procedures (experimental group). The criteria for comparison between the voices were evaluation by judges, considering the GRBASI scale and resonance, analysis of acoustic parameters and the degree of self-related discomfort. The significance level for the statistical treatment was 5%. Results: lecture resulted in statistically relevant increase of the fundamental frequency and rising of self-related discomfort. Rest did not result on statistically significant difference on the variables. Vocal warm-up resulted on decreasing in degree of vocal alteration and discomfort, especially on body related aspects. Vocal cool-down resulted on decreasing the fundamental frequency and the degree of discomfort, especially on voice related aspects. Comparison between experimental and control groups did not show any statistical difference on the analyzed variables. Cool-down and rest showed equally positive on adjusting back to the colloquial voice. Conclusion: lecture has raised the vocal attrition, indicating the vocal loading to which teachers are exposed. The proposed procedure showed to be positive and able to be applied preventively. New studies, with a larger number of subjects, should take place, ...(Complete abstract click electronic address below)
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