subject:(speech processing) – OATD
You searched for subject:(speech processing). Showing records 1 – 28 of 28 total matches.
1. Nkadimeng, Calvin. Language identification using Gaussian mixture models.
ENGLISH ABSTRACT: The importance of Language Identification for African languages is seeing a dramatic increase… (more)
ENGLISH ABSTRACT: The importance of Language Identification for African languages is seeing a dramatic increase due to the development of telecommunication infrastructure and, as a result, an increase in volumes of data and speech traffic in public networks. By automatically processing the raw speech data the vital assistance given to people in distress can be speeded up, by referring their calls to a person knowledgeable in that language. To this effect a speech corpus was developed and various algorithms were implemented and tested on raw telephone speech data. These algorithms entailed data preparation, signal processing, and statistical analysis aimed at discriminating between languages. The statistical model of Gaussian Mixture Models (GMMs) were chosen for this research due to their ability to represent an entire language with a single stochastic model that does not require phonetic transcription. Language Identification for African languages using GMMs is feasible, although there are some few challenges like proper classification and accurate study into the relationship of langauges that need to be overcome. Other methods that make use of phonetically transcribed data need to be explored and tested with the new corpus for the research to be more rigorous.
AFRIKAANSE OPSOMMING: Die belang van die Taal identifiseer vir Afrika-tale is sien ’n dramatiese toename te danke aan die ontwikkeling van telekommunikasie-infrastruktuur en as gevolg ’n toename in volumes van data en spraak verkeer in die openbaar netwerke.Deur outomaties verwerking van die ruwe toespraak gegee die noodsaaklike hulp verleen aan mense in nood kan word vinniger-up ”, deur te verwys hul oproepe na ’n persoon ingelichte in daardie taal. Tot hierdie effek van ’n toespraak corpus het ontwikkel en die verskillende algoritmes is gemplementeer en getoets op die ruwe telefoon toespraak gegee.Hierdie algoritmes behels die data voorbereiding, seinverwerking, en statistiese analise wat gerig is op onderskei tussen tale.Die statistiese model van Gauss Mengsel Modelle (GGM) was gekies is vir hierdie navorsing as gevolg van hul vermo te verteenwoordig ’n hele taal met’ n enkele stogastiese model wat nodig nie fonetiese tanscription nie. Taal identifiseer vir die Afrikatale gebruik GGM haalbaar is, alhoewel daar enkele paar uitdagings soos behoorlike klassifikasie en akkurate ondersoek na die verhouding van TALE wat moet oorkom moet word.Ander metodes wat gebruik maak van foneties getranskribeerde data nodig om ondersoek te word en getoets word met die nuwe corpus vir die ondersoek te word strenger.
Advisors/Committee Members: Niesler, T. R., University of Stellenbosch. Faculty of Engineering. Dept. of Electrical and Electronic Engineering..
Subjects/Keywords: Electronic engineering; Speech processing systems
Nkadimeng, C. (2010). Language identification using Gaussian mixture models . (Thesis). Stellenbosch University. Retrieved from http://hdl.handle.net/10019.1/4170
Nkadimeng, Calvin. “Language identification using Gaussian mixture models.” 2010. Thesis, Stellenbosch University. Accessed October 20, 2019. http://hdl.handle.net/10019.1/4170.
Nkadimeng, Calvin. “Language identification using Gaussian mixture models.” 2010. Web. 20 Oct 2019.
Nkadimeng C. Language identification using Gaussian mixture models. [Internet] [Thesis]. Stellenbosch University; 2010. [cited 2019 Oct 20]. Available from: http://hdl.handle.net/10019.1/4170.
Nkadimeng C. Language identification using Gaussian mixture models. [Thesis]. Stellenbosch University; 2010. Available from: http://hdl.handle.net/10019.1/4170
2. Brummer, Niko. Measuring, refining and calibrating speaker and language information extracted from speech.
ENGLISH ABSTRACT: We propose a new methodology, based on proper scoring rules, for the evaluation… (more)
ENGLISH ABSTRACT: We propose a new methodology, based on proper scoring rules, for the evaluation of the goodness of pattern recognizers with probabilistic outputs. The recognizers of interest take an input, known to belong to one of a discrete set of classes, and output a calibrated likelihood for each class. This is a generalization of the traditional use of proper scoring rules to evaluate the goodness of probability distributions. A recognizer with outputs in well-calibrated probability distribution form can be applied to make cost-effective Bayes decisions over a range of applications, having di fferent cost functions. A recognizer with likelihood output can additionally be employed for a wide range of prior distributions for the to-be-recognized classes. We use automatic speaker recognition and automatic spoken language recognition as prototypes of this type of pattern recognizer. The traditional evaluation methods in these fields, as represented by the series of NIST Speaker and Language Recognition Evaluations, evaluate hard decisions made by the recognizers. This makes these recognizers cost-and-prior-dependent. The proposed methodology generalizes that of the NIST evaluations, allowing for the evaluation of recognizers which are intended to be usefully applied over a wide range of applications, having variable priors and costs. The proposal includes a family of evaluation criteria, where each member of the family is formed by a proper scoring rule. We emphasize two members of this family: (i) A non-strict scoring rule, directly representing error-rate at a given prior. (ii) The strict logarithmic scoring rule which represents information content, or which equivalently represents summarized error-rate, or expected cost, over a wide range of applications. We further show how to form a family of secondary evaluation criteria, which by contrasting with the primary criteria, form an analysis of the goodness of calibration of the recognizers likelihoods. Finally, we show how to use the logarithmic scoring rule as an objective function for the discriminative training of fusion and calibration of speaker and language recognizers.
AFRIKAANSE OPSOMMING: Ons wys hoe om die onsekerheid in die uittree van outomatiese sprekerherkenning- en taalherkenningstelsels voor te stel, te meet, te kalibreer en te optimeer. Dit maak die bestaande tegnologie akkurater, doeltre ender en meer algemeen toepasbaar.
Subjects/Keywords: Electronic engineering; Automatic speech recognition; Speech processing systems
Brummer, N. (2010). Measuring, refining and calibrating speaker and language information extracted from speech . (Thesis). Stellenbosch University. Retrieved from http://hdl.handle.net/10019.1/5139
Brummer, Niko. “Measuring, refining and calibrating speaker and language information extracted from speech.” 2010. Thesis, Stellenbosch University. Accessed October 20, 2019. http://hdl.handle.net/10019.1/5139.
Brummer, Niko. “Measuring, refining and calibrating speaker and language information extracted from speech.” 2010. Web. 20 Oct 2019.
Brummer N. Measuring, refining and calibrating speaker and language information extracted from speech. [Internet] [Thesis]. Stellenbosch University; 2010. [cited 2019 Oct 20]. Available from: http://hdl.handle.net/10019.1/5139.
Brummer N. Measuring, refining and calibrating speaker and language information extracted from speech. [Thesis]. Stellenbosch University; 2010. Available from: http://hdl.handle.net/10019.1/5139
3. Kruger, Johannes Petrus. Enkelsybanddemodulasie met behulp van syferseinverwerking.
The feasibility of modulation and demodulation of speech signals within a microprocessor is invertigated in the following study. Existing modulation… (more)
The feasibility of modulation and demodulation of speech signals within a microprocessor is invertigated in the following study. Existing modulation and demodulation techniques are investigated and new techniques. suitable for microprocessor implementation, described. Finally a single sideband demodulator was built using the TMS32010 microprocessor with results being better or comparable than existing analog techniques.
Subjects/Keywords: Modulation (Electronics); Demodulation (Electronics); Speech synthesis; Speech processing systems
Kruger, J. P. (2014). Enkelsybanddemodulasie met behulp van syferseinverwerking . (Thesis). University of Johannesburg. Retrieved from http://hdl.handle.net/10210/11224
Kruger, Johannes Petrus. “Enkelsybanddemodulasie met behulp van syferseinverwerking.” 2014. Thesis, University of Johannesburg. Accessed October 20, 2019. http://hdl.handle.net/10210/11224.
Kruger, Johannes Petrus. “Enkelsybanddemodulasie met behulp van syferseinverwerking.” 2014. Web. 20 Oct 2019.
Kruger JP. Enkelsybanddemodulasie met behulp van syferseinverwerking. [Internet] [Thesis]. University of Johannesburg; 2014. [cited 2019 Oct 20]. Available from: http://hdl.handle.net/10210/11224.
Kruger JP. Enkelsybanddemodulasie met behulp van syferseinverwerking. [Thesis]. University of Johannesburg; 2014. Available from: http://hdl.handle.net/10210/11224
4. Goussard, George Willem. Unsupervised clustering of audio data for acoustic modelling in automatic speech recognition systems.
Thesis (MScEng (Electrical and Electronic Engineering)) – University of Stellenbosch, 2011.
ENGLISH ABSTRACT: This thesis presents a system that is designed to replace the manual process… (more)
ENGLISH ABSTRACT: This thesis presents a system that is designed to replace the manual process of generating a pronunciation dictionary for use in automatic speech recognition. The proposed system has several stages. The first stage segments the audio into what will be known as the subword units, using a frequency domain method. In the second stage, dynamic time warping is used to determine the similarity between the segments of each possible pair of these acoustic segments. These similarities are used to cluster similar acoustic segments into acoustic clusters. The final stage derives a pronunciation dictionary from the orthography of the training data and corresponding sequence of acoustic clusters. This process begins with an initial mapping between words and their sequence of clusters, established by Viterbi alignment with the orthographic transcription. The dictionary is refined iteratively by pruning redundant mappings, hidden Markov model estimation and Viterbi re-alignment in each iteration. This approach is evaluated experimentally by applying it to two subsets of the TIMIT corpus. It is found that, when test words are repeated often in the training material, the approach leads to a system whose accuracy is almost as good as one trained using the phonetic transcriptions. When test words are not repeated often in the training set, the proposed approach leads to better results than those achieved using the phonetic transcriptions, although the recognition is poor overall in this case.
AFRIKAANSE OPSOMMING: Die doelwit van die tesis is om ’n stelsel te beskryf wat ontwerp is om die handgedrewe proses in die samestelling van ’n woordeboek, vir die gebruik in outomatiese spraakherkenningsstelsels, te vervang. Die voorgestelde stelsel bestaan uit ’n aantal stappe. Die eerste stap is die segmentering van die oudio in sogenaamde sub-woord eenhede deur gebruik te maak van ’n frekwensie gebied tegniek. Met die tweede stap word die dinamiese tydverplasingsalgoritme ingespan om die ooreenkoms tussen die segmente van elkeen van die moontlike pare van die akoestiese segmente bepaal. Die ooreenkomste word dan gebruik om die akoestiese segmente te groepeer in akoestiese groepe. Die laaste stap stel die woordeboek saam deur gebruik te maak van die ortografiese transkripsie van afrigtingsdata en die ooreenstemmende reeks akoestiese groepe. Die finale stap begin met ’n aanvanklike afbeelding vanaf woorde tot hul reeks groep identifiseerders, bewerkstellig deur Viterbi belyning en die ortografiese transkripsie. Die woordeboek word iteratief verfyn deur oortollige afbeeldings te snoei, verskuilde Markov modelle af te rig en deur Viterbi belyning te gebruik in elke iterasie. Die benadering is getoets deur dit eksperimenteel te evalueer op twee subversamelings data vanuit die TIMIT korpus. Daar is bevind dat, wanneer woorde herhaal word in die afrigtingsdata, die stelsel se benadering die…
Subjects/Keywords: Electronic engineering; Natural language processing (Computer science); Speech processing systems
Goussard, G. W. (2011). Unsupervised clustering of audio data for acoustic modelling in automatic speech recognition systems . (Thesis). Stellenbosch University. Retrieved from http://hdl.handle.net/10019.1/6686
Goussard, George Willem. “Unsupervised clustering of audio data for acoustic modelling in automatic speech recognition systems.” 2011. Thesis, Stellenbosch University. Accessed October 20, 2019. http://hdl.handle.net/10019.1/6686.
Goussard, George Willem. “Unsupervised clustering of audio data for acoustic modelling in automatic speech recognition systems.” 2011. Web. 20 Oct 2019.
Goussard GW. Unsupervised clustering of audio data for acoustic modelling in automatic speech recognition systems. [Internet] [Thesis]. Stellenbosch University; 2011. [cited 2019 Oct 20]. Available from: http://hdl.handle.net/10019.1/6686.
Goussard GW. Unsupervised clustering of audio data for acoustic modelling in automatic speech recognition systems. [Thesis]. Stellenbosch University; 2011. Available from: http://hdl.handle.net/10019.1/6686
5. Moonasar, Viresh. Hardware implementation of an automatic speaker recognition system using artificial neural networks.
Degree: 2002, Durban University of Technology
Submitted in fulfillment of the academic requirements for the degree of Master of Technology in Electrical Engineering in the Department of Electronic Engineering, Faculty of… (more)
Submitted in fulfillment of the academic requirements for the degree of Master of Technology in Electrical Engineering in the Department of Electronic Engineering, Faculty of Engineering, ML Sultan Technikon of Durban in South Africa, March 2002.
The use of speaker recognition technology in interactive voice response and electronic commerce systems has been limited. This is due to the lack of research attention and published results when compared to all the other areas of speech recognition technologies
Advisors/Committee Members: Venayagamoorthy, G. K..
Subjects/Keywords: Automatic speech recognition; Speech processing systems
Moonasar, V. (2002). Hardware implementation of an automatic speaker recognition system using artificial neural networks . (Thesis). Durban University of Technology. Retrieved from http://hdl.handle.net/10321/2714
Moonasar, Viresh. “Hardware implementation of an automatic speaker recognition system using artificial neural networks.” 2002. Thesis, Durban University of Technology. Accessed October 20, 2019. http://hdl.handle.net/10321/2714.
Moonasar, Viresh. “Hardware implementation of an automatic speaker recognition system using artificial neural networks.” 2002. Web. 20 Oct 2019.
Moonasar V. Hardware implementation of an automatic speaker recognition system using artificial neural networks. [Internet] [Thesis]. Durban University of Technology; 2002. [cited 2019 Oct 20]. Available from: http://hdl.handle.net/10321/2714.
Moonasar V. Hardware implementation of an automatic speaker recognition system using artificial neural networks. [Thesis]. Durban University of Technology; 2002. Available from: http://hdl.handle.net/10321/2714
6. Van der Walt, Craig. An investigation into the practical implementation of speech recognition for data capturing .
► A study into the practical implementation of Speech Recognition for the purposes of Data Capturing within Telkom SA. is described. As datacapturing is increasing in… (more)
▼ A study into the practical implementation of Speech Recognition for the purposes of Data Capturing within Telkom SA. is described. As datacapturing is increasing in demand a more efficient method of capturing is sought. The technology relating to Speech recognition is herein examined and practical gnidelines for selecting a Speech recognition system are described. These guidelines are used to show how commercially available systems can be evaluated. Specific tests on a selected speech recognition system are described, relating to the accuracy and adaptability of the system. The results obtained illustrate why at present speech recognition systems are not advisable for the purpose of Data capturing. The results also demonstrate how the selection of keywords words can affect system performance. Areas of further research are highlighted relating to recognition performance and vocabulary selection.
Van der Walt, C. (1993). An investigation into the practical implementation of speech recognition for data capturing . (Thesis). Cape Peninsula University of Technology. Retrieved from http://etd.cput.ac.za/handle/20.500.11838/1156
Van der Walt, Craig. “An investigation into the practical implementation of speech recognition for data capturing .” 1993. Thesis, Cape Peninsula University of Technology. Accessed October 20, 2019. http://etd.cput.ac.za/handle/20.500.11838/1156.
Van der Walt, Craig. “An investigation into the practical implementation of speech recognition for data capturing .” 1993. Web. 20 Oct 2019.
Van der Walt C. An investigation into the practical implementation of speech recognition for data capturing . [Internet] [Thesis]. Cape Peninsula University of Technology; 1993. [cited 2019 Oct 20]. Available from: http://etd.cput.ac.za/handle/20.500.11838/1156.
Van der Walt C. An investigation into the practical implementation of speech recognition for data capturing . [Thesis]. Cape Peninsula University of Technology; 1993. Available from: http://etd.cput.ac.za/handle/20.500.11838/1156
7. Nel, André Leon. A study of machine enabled communication for the severely disabled patient, with reference to image processing techniques.
This research aims at identifying technologies which could be used to produce a communication aid for the speechless severely disabled person. A… (more)
This research aims at identifying technologies which could be used to produce a communication aid for the speechless severely disabled person. A number of techniques are investigated which would enable the disabled to communicate by means of decoding eye movements. The tracking of the disabled's eye positions by means of two dimensional image processing and pattern recognition forms the basis of the input mechanism. A complete comparison between a number of candidate edge detectors is done to ensure that the system would function in as closeto real-time as is possible. Ancillary techniques such as a reduced grammar and a Markov model for letter posterior probabilities are developed and shown to improve the communication channel bandwidth. Simulated resultsshowwhat communication rate gains could be achieved. It appears that a possible communication aid could be built to perform at rates exceeding present day communication aids for the severely disabled at prices that could make such a device an economic possibility.
Subjects/Keywords: Speech disorders; Communication devices for people with disabilities; Image processing
Nel, A. L. (2014). A study of machine enabled communication for the severely disabled patient, with reference to image processing techniques . (Thesis). University of Johannesburg. Retrieved from http://hdl.handle.net/10210/9486
Nel, André Leon. “A study of machine enabled communication for the severely disabled patient, with reference to image processing techniques.” 2014. Thesis, University of Johannesburg. Accessed October 20, 2019. http://hdl.handle.net/10210/9486.
Nel, André Leon. “A study of machine enabled communication for the severely disabled patient, with reference to image processing techniques.” 2014. Web. 20 Oct 2019.
Nel AL. A study of machine enabled communication for the severely disabled patient, with reference to image processing techniques. [Internet] [Thesis]. University of Johannesburg; 2014. [cited 2019 Oct 20]. Available from: http://hdl.handle.net/10210/9486.
Nel AL. A study of machine enabled communication for the severely disabled patient, with reference to image processing techniques. [Thesis]. University of Johannesburg; 2014. Available from: http://hdl.handle.net/10210/9486
8. Basson, J. A. L. Adaptive estimation of speech parameters.
Degree: MEng, Electrical, 1994, Stellenbosch University
ENGLISH ABSTRACT: Linear predictive coding(LPC), and transformations of it, is currently the most popular way of analysing speech signals. Major limitations of using a frame-based… (more)
ENGLISH ABSTRACT: Linear predictive coding(LPC), and transformations of it, is currently the most popular way of analysing speech signals. Major limitations of using a frame-based technique are that each frame is analysed in isolation of the rest while assuming the excitation source to be a white, gaussian process. In order to reduce computation time, an all pole model is usually employed. In this project an adaptive algorithm is proposed for speech signal analysis. The algorithm is based on the recursive least mean squares method with a variable forgetting factor. A pole-zero model is used to: estimate the anti-formants present in certain sounds (i.e. nasals and nasalized vowels). This method offers better detection of poles and zeros in stationary environments and faster tracking of pole and zero frequencies in nonstationary signals than other sequential methods. An effective input estimation algorithm eliminates the influence of pitch on the parameter estimates by assuming the input to be a white noise process or a pulse sequence.
AFRIKAANSE OPSOMMING: Linieere voorspellings-kodering, en transformasies daarvan, is huidiglik die gewildste metode t.o.v. die analise van spraakseine. Blok-gebaseerde algoritmes het ernstige tekortkominge. Elke raam word byvoorbeeld in isolasie van die res geanaliseer terwyl daar aangeneem word dat die intree na die spraakkanaal 'n wit, gaussiese ruisproses is. Om berekeningstyd te beperk word 'n model met slegs pole gebruik. In hierdie projek word 'n aanpasbare algoritme (gebaseer op die rekursiewe kleinste kwadrate metode) met 'n varierende vergeetfaktor voorgestel. 'n Pool-zero model bied akkurater opsporing van pole en zeros in stasionere omgewings. Dit bied ook vinniger volging van pool en zero frekwensies in nie-stasionere seine as ander aanpasbare algoritmes. 'n Effektiewe intree-beramings algoritme skakel die invloed van die fundamentele frekwensie op die beraamde parameters uit. Dit word reggekry deur te aanvaar dat die intree 'n wit ruis-proses of 'n pols reeks kan wees.
Advisors/Committee Members: Du Preez, J. A., Stellenbosch University. Faculty of Engineering. Dept. of Electrical & Electronic Engineering..
Subjects/Keywords: Speech processing systems; Automatic speech recognition; Algorithms
Basson, J. A. L. (1994). Adaptive estimation of speech parameters . (Thesis). Stellenbosch University. Retrieved from http://hdl.handle.net/10019.1/58236
Basson, J A L. “Adaptive estimation of speech parameters.” 1994. Thesis, Stellenbosch University. Accessed October 20, 2019. http://hdl.handle.net/10019.1/58236.
Basson, J A L. “Adaptive estimation of speech parameters.” 1994. Web. 20 Oct 2019.
Basson JAL. Adaptive estimation of speech parameters. [Internet] [Thesis]. Stellenbosch University; 1994. [cited 2019 Oct 20]. Available from: http://hdl.handle.net/10019.1/58236.
Basson JAL. Adaptive estimation of speech parameters. [Thesis]. Stellenbosch University; 1994. Available from: http://hdl.handle.net/10019.1/58236
9. Nel, Pieter Willem. Automatic syllabification of untranscribed speech.
Degree: MScEng, Electrical and Electronic Engineering, 2005, Stellenbosch University
ENGLISH ABSTRACT: The syllable has been proposed as a unit of automatic speech recognition due to its strong links with human speech production and perception.… (more)
ENGLISH ABSTRACT: The syllable has been proposed as a unit of automatic speech recognition due to its strong links with human speech production and perception. Recently, it has been proved that incorporating information from syllable-length time-scales into automatic speech recognition improves results in large vocabulary recognition tasks. It was also shown to aid in various language recognition tasks and in foreign accent identification. Therefore, the ability to automatically segment speech into syllables is an important research tool. Where most previous studies employed knowledge-based methods, this study presents a purely statistical method for the automatic syllabification of speech. We introduce the concept of hierarchical hidden Markov model structures and show how these can be used to implement a purely acoustical syllable segmenter based, on general sonority theory, combined with some of the phonotactic constraints found in the English language. The accurate reporting of syllabification results is a problem in the existing literature. We present a well-defined dynamic time warping (DTW) distance measure used for reporting syllabification results. We achieve a token error rate of 20.3% with a 42ms average boundary error on a relatively large set of data. This compares well with previous knowledge-based and statistically- based methods.
AFRIKAANSE OPSOMMING: Die syllabe is voorheen voorgestel as 'n basiese eenheid vir automatiese spraakherkenning weens die sterk verwantwskap wat dit het met spraak produksie en persepsie. Onlangs is dit bewys dat die gebruik van informasie van syllabe-lengte tydskale die resultate verbeter in groot woordeskat herkennings take. Dit is ook bewys dat die gebruik van syllabes automatiese taalherkenning en vreemdetaal aksent herkenning vergemaklik. Dit is daarom belangrik om vir navorsingsdoeleindes syllabes automaties te kan segmenteer. Vorige studies het kennisgebaseerde metodes gebruik om hierdie segmentasie te bewerkstellig. Hierdie studie gebruik 'n suiwer statistiese metode vir die automatiese syllabifikasie van spraak. Ons gebruik die konsep van hierargiese verskuilde Markov model strukture en wys hoe dit gebruik kan word om 'n suiwer akoestiese syllabe segmenteerder te implementeer. Die model word gebou deur dit te baseer op die teorie van sonoriteit asook die fonotaktiese beperkinge teenwoordig in die Engelse taal. Die akkurate voorstelling van syllabifikasie resultate is problematies in die bestaande literatuur. Ons definieer volledig 'n DTW (Dynamic Time Warping) afstands funksie waarmee ons ons syllabifikasie resultate weergee. Ons behaal 'n TER (Token Error Rate) van 20.3% met 'n 42ms gemiddelde grens fout op 'n relatiewe groot stel data. Dit vergelyk goed met vorige kennis-gebaseerde en statisties-gebaseerde metodes.
Advisors/Committee Members: Du Preez, J. A., Stellenbosch University. Faculty of Engineering. Dept. of Electrical and Electronic Engineering..
Subjects/Keywords: Automatic speech recognition; Speech processing systems; Dissertations – Electronic engineering
Nel, P. W. (2005). Automatic syllabification of untranscribed speech . (Masters Thesis). Stellenbosch University. Retrieved from http://hdl.handle.net/10019.1/50285
Nel, Pieter Willem. “Automatic syllabification of untranscribed speech.” 2005. Masters Thesis, Stellenbosch University. Accessed October 20, 2019. http://hdl.handle.net/10019.1/50285.
Nel, Pieter Willem. “Automatic syllabification of untranscribed speech.” 2005. Web. 20 Oct 2019.
Nel PW. Automatic syllabification of untranscribed speech. [Internet] [Masters thesis]. Stellenbosch University; 2005. [cited 2019 Oct 20]. Available from: http://hdl.handle.net/10019.1/50285.
Nel PW. Automatic syllabification of untranscribed speech. [Masters Thesis]. Stellenbosch University; 2005. Available from: http://hdl.handle.net/10019.1/50285
10. Venter, Petrus Jacobus. Recording and automatic detection of African elephant (Loxodonta africana) infrasonic rumbles.
► The value of studying elephant vocalizations lies in the abundant information that can be retrieved from it. Recordings of elephant rumbles can be used by… (more)
▼ The value of studying elephant vocalizations lies in the abundant information that can be retrieved from it. Recordings of elephant rumbles can be used by researchers to determine the size and composition of the herd, the sexual state, as well as the emotional condition of an elephant. It is a difficult task for researchers to obtain large volumes of continuous recordings of elephant vocalizations. Recordings are normally analysed manually to identify the location of rumbles via the tedious and time consuming methods of sped up listening and the visual evaluation of spectrograms. The application of speech processing on elephant vocalizations is a highly unexploited resource. The aim of this study was to contribute to the current body of knowledge and resources of elephant research by developing a tool for recording high volumes of continuous acoustic data in harsh natural conditions as well as examining the possibilities of applying human speech processing techniques to elephant rumbles to achieve automatic detection of these rumbles in recordings. The recording tool was designed and implemented as an elephant recording collar that has an onboard data storage capacity of 128 gigabytes, enough memory to record sound data continuously for a period of nine months. Data is stored in the wave file format and the device has the ability to navigate and control the FAT32 file system so that the files can be read and downloaded to a personal computer. The collar also has the ability to stamp sound files with the time and date, ambient temperature and GPS coordinates. Several different options for microphone placement and protection have been tested experimentally to find an acceptable solution. A relevant voice activity detection algorithm was chosen as a base for the automatic detection of infrasonic elephant rumbles. The chosen algorithm is based on a robust pitch determination algorithm that has been experimentally verified to function correctly under a signal-to-noise ratio as low as -8 dB when more than four harmonic structures exist in a sound. The algorithm was modified to be used for elephant rumbles and was tested with previously recorded elephant vocalization data. The results obtained suggest that the algorithm can accurately detect elephant rumbles from recordings. The number of false alarms and undetected calls increase when recordings are contaminated with unwanted noise that contains harmonic structures or when the harmonic nature of a rumble is lost. Data obtained from the recording collar is less prone to being contaminated than far field recordings and the automatic detection algorithm should provide an accurate tool for detecting any rumbles that appear in the recordings. Advisors/Committee Members: Prof J J Hanekom (advisor).
Subjects/Keywords: Voice activity detection; Pitch determination; Infrasonic rumbles; Elephant vocalizations; Bio acoustics; Speech processing; UCTD
Venter, P. (2008). Recording and automatic detection of African elephant (Loxodonta africana) infrasonic rumbles . (Masters Thesis). University of Pretoria. Retrieved from http://hdl.handle.net/2263/28329
Venter, Petrus. “Recording and automatic detection of African elephant (Loxodonta africana) infrasonic rumbles.” 2008. Masters Thesis, University of Pretoria. Accessed October 20, 2019. http://hdl.handle.net/2263/28329.
Venter, Petrus. “Recording and automatic detection of African elephant (Loxodonta africana) infrasonic rumbles.” 2008. Web. 20 Oct 2019.
Venter P. Recording and automatic detection of African elephant (Loxodonta africana) infrasonic rumbles. [Internet] [Masters thesis]. University of Pretoria; 2008. [cited 2019 Oct 20]. Available from: http://hdl.handle.net/2263/28329.
Venter P. Recording and automatic detection of African elephant (Loxodonta africana) infrasonic rumbles. [Masters Thesis]. University of Pretoria; 2008. Available from: http://hdl.handle.net/2263/28329
11. [No author]. Recording and automatic detection of African elephant (Loxodonta africana) infrasonic rumbles .
author], [. (2008). Recording and automatic detection of African elephant (Loxodonta africana) infrasonic rumbles . (Masters Thesis). University of Pretoria. Retrieved from http://upetd.up.ac.za/thesis/available/etd-10012008-161614/
author], [No. “Recording and automatic detection of African elephant (Loxodonta africana) infrasonic rumbles .” 2008. Masters Thesis, University of Pretoria. Accessed October 20, 2019. http://upetd.up.ac.za/thesis/available/etd-10012008-161614/.
author], [No. “Recording and automatic detection of African elephant (Loxodonta africana) infrasonic rumbles .” 2008. Web. 20 Oct 2019.
author] [. Recording and automatic detection of African elephant (Loxodonta africana) infrasonic rumbles . [Internet] [Masters thesis]. University of Pretoria; 2008. [cited 2019 Oct 20]. Available from: http://upetd.up.ac.za/thesis/available/etd-10012008-161614/.
author] [. Recording and automatic detection of African elephant (Loxodonta africana) infrasonic rumbles . [Masters Thesis]. University of Pretoria; 2008. Available from: http://upetd.up.ac.za/thesis/available/etd-10012008-161614/
12. Cilliers, Francois Dirk. Tree-based Gaussian mixture models for speaker verification.
Degree: Electrical and Electronic Engineering, 2005, Stellenbosch University
Thesis (MScEng (Electrical and Electronic Engineering)) – University of Stellenbosch, 2005.
The Gaussian mixture model (GMM) performs very effectively in applications such as speech and speaker… (more)
The Gaussian mixture model (GMM) performs very effectively in applications such as speech and speaker recognition. However, evaluation speed is greatly reduced when the GMM has a large number of mixture components. Various techniques improve the evaluation speed by reducing the number of required Gaussian evaluations.
Subjects/Keywords: Electronic engineering; Automatic speech recognition; Speech processing systems; Electrical and Electronic Engineering
Cilliers, F. D. (2005). Tree-based Gaussian mixture models for speaker verification . (Thesis). Stellenbosch University. Retrieved from http://hdl.handle.net/10019.1/1639
Cilliers, Francois Dirk. “Tree-based Gaussian mixture models for speaker verification.” 2005. Thesis, Stellenbosch University. Accessed October 20, 2019. http://hdl.handle.net/10019.1/1639.
Cilliers, Francois Dirk. “Tree-based Gaussian mixture models for speaker verification.” 2005. Web. 20 Oct 2019.
Cilliers FD. Tree-based Gaussian mixture models for speaker verification. [Internet] [Thesis]. Stellenbosch University; 2005. [cited 2019 Oct 20]. Available from: http://hdl.handle.net/10019.1/1639.
Cilliers FD. Tree-based Gaussian mixture models for speaker verification. [Thesis]. Stellenbosch University; 2005. Available from: http://hdl.handle.net/10019.1/1639
13. Pool, Jan. Investigation of the impact of high frequency transmitted speech on speaker recognition.
Degree: MScEng, Electrical and Electronic Engineering, 2002, Stellenbosch University
ENGLISH ABSTRACT: Speaker recognition systems have evolved to a point… (more)
ENGLISH ABSTRACT: Speaker recognition systems have evolved to a point where near perfect performance can be obtained under ideal conditions, even if the system must distinguish between a large number of speakers. Under adverse conditions, such as when high noise levels are present or when the transmission channel deforms the speech, the performance is often less than satisfying. This project investigated the performance of a popular speaker recognition system, that use Gaussian mixture models, on speech transmitted over a high frequency channel. Initial experiments demonstrated very unsatisfactory results for the base line system. We investigated a number of robust techniques. We implemented and applied some of them in an attempt to improve the performance of the speaker recognition systems. The techniques we tested showed only slight improvements. We also investigates the effects of a high frequency channel and single sideband modulation on the speech features of speech processing systems. The effects that can deform the features, and therefore reduce the performance of speech systems, were identified. One of the effects that can greatly affect the performance of a speech processing system is noise. We investigated some speech enhancement techniques and as a result we developed a new statistical based speech enhancement technique that employs hidden Markov models to represent the clean speech process.
AFRIKAANSE OPSOMMING: Sprekerherkenning-stelsels het 'n punt bereik waar nabyaan perfekte resultate verwag kan word onder ideale kondisies, selfs al moet die stelsel tussen 'n groot aantal sprekers onderskei. Wanneer nie-ideale kondisies, soos byvoorbeeld hoë ruisvlakke of 'n transmissie kanaal wat die spraak vervorm, teenwoordig is, is die resultate gewoonlik nie bevredigend nie. Die projek ondersoek die werksverrigting van 'n gewilde sprekerherkenning-stelsel, wat gebruik maak van Gaussiese mengselmodelle, op spraak wat oor 'n hoë frekwensie transmissie kanaal gestuur is. Aanvanklike eksperimente wat gebruik maak van 'n basiese stelsel het nie goeie resultate opgelewer nie. Ons het 'n aantal robuuste tegnieke ondersoek en 'n paar van hulle geïmplementeer en getoets in 'n poging om die resultate van die sprekerherkenning-stelsel te verbeter. Die tegnieke wat ons getoets het, het net geringe verbetering getoon. Die studie het ook die effekte wat die hoë-frekwensie kanaal en enkel-syband modulasie op spraak kenmerkvektore, ondersoek. Die effekte wat die spraak kenmerkvektore kan vervorm en dus die werkverrigting van spraak stelsels kan verlaag, is geïdentifiseer. Een van die effekte wat 'n groot invloed op die werkverrigting van spraakstelsels het, is ruis. Ons het spraak verbeterings metodes ondersoek en dit het gelei tot die ontwikkeling van 'n statisties gebaseerde spraak verbeteringstegniek wat gebruik maak van verskuilde Markov modelle om die skoon spraakproses voor te stel.
Advisors/Committee Members: Du Preez, J. A., Lourens, J. G., Stellenbosch University. Faculty of Engineering. Dept. of Electrical and Electronic Engineering..
Subjects/Keywords: Automatic speech recognition; Speech processing systems; Markov processes; Dissertations – Electronic engineering
Pool, J. (2002). Investigation of the impact of high frequency transmitted speech on speaker recognition . (Masters Thesis). Stellenbosch University. Retrieved from http://hdl.handle.net/10019.1/52895
Pool, Jan. “Investigation of the impact of high frequency transmitted speech on speaker recognition.” 2002. Masters Thesis, Stellenbosch University. Accessed October 20, 2019. http://hdl.handle.net/10019.1/52895.
Pool, Jan. “Investigation of the impact of high frequency transmitted speech on speaker recognition.” 2002. Web. 20 Oct 2019.
Pool J. Investigation of the impact of high frequency transmitted speech on speaker recognition. [Internet] [Masters thesis]. Stellenbosch University; 2002. [cited 2019 Oct 20]. Available from: http://hdl.handle.net/10019.1/52895.
Pool J. Investigation of the impact of high frequency transmitted speech on speaker recognition. [Masters Thesis]. Stellenbosch University; 2002. Available from: http://hdl.handle.net/10019.1/52895
14. Combrinck, Hendrik Petrus. A cost, complexity and performance comparison of two automatic language identification architectures.
► This dissertation investigates the cost-complexity-performance relationship between two automatic language identification systems. The first is a state-of-the-art archi¬tecture, trained on about three hours of phonetically… (more)
▼ This dissertation investigates the cost-complexity-performance relationship between two automatic language identification systems. The first is a state-of-the-art archi¬tecture, trained on about three hours of phonetically hand-labelled telephone speech obtained from the recognised OGLTS corpus. The second system, introduced by our¬selves, is a simpler design with a smaller, less complex parameter space. It is a vector quantisation-based approach which bears some resemblance to a system suggested by Sugiyama. Though trained on the same data, it has no need for any labels and is therefore less costly. A number of experiments are performed to find quasi-optimal parameters for the two systems. In further experiments the systems are evaluated and compared on a set of ten two-language tasks, spanning five languages. The more com¬plex system is shown to have a substantial performance advantage over the simpler design - 81% versus 65% on 40 seconds of speech. However, both results are well under reported state-of-the-art performance of 94% and would suggest that our systems can benefit from additional attention to implementation detail and optimisation of various parameters. Given the above, our suggested architecture may potentially provide an adequate solution where the high development cost associated with state-of-the-art technology and the necessary training corpora are prohibitive. Advisors/Committee Members: Prof E C Botha (advisor).
Subjects/Keywords: Speech processing systems; Pattern recognition systems; UCTD
Combrinck, H. (2006). A cost, complexity and performance comparison of two automatic language identification architectures . (Masters Thesis). University of Pretoria. Retrieved from http://hdl.handle.net/2263/30492
Combrinck, Hendrik. “A cost, complexity and performance comparison of two automatic language identification architectures.” 2006. Masters Thesis, University of Pretoria. Accessed October 20, 2019. http://hdl.handle.net/2263/30492.
Combrinck, Hendrik. “A cost, complexity and performance comparison of two automatic language identification architectures.” 2006. Web. 20 Oct 2019.
Combrinck H. A cost, complexity and performance comparison of two automatic language identification architectures. [Internet] [Masters thesis]. University of Pretoria; 2006. [cited 2019 Oct 20]. Available from: http://hdl.handle.net/2263/30492.
Combrinck H. A cost, complexity and performance comparison of two automatic language identification architectures. [Masters Thesis]. University of Pretoria; 2006. Available from: http://hdl.handle.net/2263/30492
15. [No author]. A cost, complexity and performance comparison of two automatic language identification architectures .
author], [. (2006). A cost, complexity and performance comparison of two automatic language identification architectures . (Masters Thesis). University of Pretoria. Retrieved from http://upetd.up.ac.za/thesis/available/etd-12212006-141335/
author], [No. “A cost, complexity and performance comparison of two automatic language identification architectures .” 2006. Masters Thesis, University of Pretoria. Accessed October 20, 2019. http://upetd.up.ac.za/thesis/available/etd-12212006-141335/.
author], [No. “A cost, complexity and performance comparison of two automatic language identification architectures .” 2006. Web. 20 Oct 2019.
author] [. A cost, complexity and performance comparison of two automatic language identification architectures . [Internet] [Masters thesis]. University of Pretoria; 2006. [cited 2019 Oct 20]. Available from: http://upetd.up.ac.za/thesis/available/etd-12212006-141335/.
author] [. A cost, complexity and performance comparison of two automatic language identification architectures . [Masters Thesis]. University of Pretoria; 2006. Available from: http://upetd.up.ac.za/thesis/available/etd-12212006-141335/
16. Kritzinger, Carl. Low bit rate speech coding.
Thesis (MScIng (Electrical and Electronic Engineering)) – University of Stellenbosch, 2006.
Despite enormous advances in digital communication, the voice is still the primary tool with which… (more)
Despite enormous advances in digital communication, the voice is still the primary tool with which people exchange ideas. However, uncompressed digital speech tends to require prohibitively high data rates (upward of 64kbps), making it impractical for many applications. Speech coding is the process of reducing the data rate of digital voice to manageable levels. Parametric speech coders or vocoders utilise a-priori information about the mechanism by which speech is produced in order to achieve extremely efficient compression of speech signals (as low as 1 kbps). The greater part of this thesis comprises an investigation into parametric speech coding. This consisted of a review of the mathematical and heuristic tools used in parametric speech coding, as well as the implementation of an accepted standard algorithm for parametric voice coding. In order to examine avenues of improvement for the existing vocoders, we examined some of the mathematical structure underlying parametric speech coding. Following on from this, we developed a novel approach to parametric speech coding which obtained promising results under both objective and subjective evaluation. An additional contribution by this thesis was the comparative subjective evaluation of the effect of parametric speech coding on English and Xhosa speech. We investigated the performance of two different encoding algorithms on the two languages.
Subjects/Keywords: Electronic engineering; Speech processing systems; Coding theory
Kritzinger, C. (2006). Low bit rate speech coding . (Thesis). Stellenbosch University. Retrieved from http://hdl.handle.net/10019.1/2078
Kritzinger, Carl. “Low bit rate speech coding.” 2006. Thesis, Stellenbosch University. Accessed October 20, 2019. http://hdl.handle.net/10019.1/2078.
Kritzinger, Carl. “Low bit rate speech coding.” 2006. Web. 20 Oct 2019.
Kritzinger C. Low bit rate speech coding. [Internet] [Thesis]. Stellenbosch University; 2006. [cited 2019 Oct 20]. Available from: http://hdl.handle.net/10019.1/2078.
Kritzinger C. Low bit rate speech coding. [Thesis]. Stellenbosch University; 2006. Available from: http://hdl.handle.net/10019.1/2078
17. Khan, Farhana. Auditory processing disorders: training curriculum for communication pathologists within the South African context.
► This study examined the nature of the undergraduate curricula for Auditory Processing Disorders (APD) for communication pathologists (speech-language therapists and audiologists) within the South African… (more)
▼ This study examined the nature of the undergraduate curricula for Auditory Processing Disorders (APD) for communication pathologists (speech-language therapists and audiologists) within the South African context. An exploratory descriptive survey design was utilised. The respondents were the authoritative voices in the area of APD, i.e. academics based at training institutions involved in the training of Speech-Language Therapists and Audiologists in the field of APD. They represented five of the six South African training institutions training speech-language therapists and audiologists. Information on the training programmes offered in APD was obtained with the use of a specifically designed questionnaire. This was further supplemented by copies of the course descriptors and / or study guides supplied by the respondents from the respective training institutions. A curriculum analysis framework was utilised to analyse the curricula (Jansen&Reddy, 1998). The findings of the study indicated that the curricula offered in APD at all training institutions compared well to current research and literature in the field of APD. However, information was not transparent on how the South African social and contextual issues were incorporated into training in APD. The researcher proposed that the curricula currently in use did not require major changes but appropriate amendments are required to be considered. The critical paradigm of inquiry was advocated to be used when training in the area of APD. Additionally, the researcher motivated for and recommended additions to the curricula on APD to address the South African situation as an essential part of the curriculum. The researcher’s principle guideline for amending the curricula was to incorporate these issues into the training based on the adoption of an ecological approach to assessment and remediation of APD. Advisors/Committee Members: Prof B Louw (advisor), Dr N Campbell (advisor).
Subjects/Keywords: Curricula; Training institutions; Auditory processing disorder; Audiologist; Speech-language therapist; UCTD
Khan, F. (2007). Auditory processing disorders: training curriculum for communication pathologists within the South African context . (Masters Thesis). University of Pretoria. Retrieved from http://hdl.handle.net/2263/23990
Khan, Farhana. “Auditory processing disorders: training curriculum for communication pathologists within the South African context.” 2007. Masters Thesis, University of Pretoria. Accessed October 20, 2019. http://hdl.handle.net/2263/23990.
Khan, Farhana. “Auditory processing disorders: training curriculum for communication pathologists within the South African context.” 2007. Web. 20 Oct 2019.
Khan F. Auditory processing disorders: training curriculum for communication pathologists within the South African context. [Internet] [Masters thesis]. University of Pretoria; 2007. [cited 2019 Oct 20]. Available from: http://hdl.handle.net/2263/23990.
Khan F. Auditory processing disorders: training curriculum for communication pathologists within the South African context. [Masters Thesis]. University of Pretoria; 2007. Available from: http://hdl.handle.net/2263/23990
18. [No author]. Auditory processing disorders: training curriculum for communication pathologists within the South African context .
author], [. (2007). Auditory processing disorders: training curriculum for communication pathologists within the South African context . (Masters Thesis). University of Pretoria. Retrieved from http://upetd.up.ac.za/thesis/available/etd-04172007-153701/
author], [No. “Auditory processing disorders: training curriculum for communication pathologists within the South African context .” 2007. Masters Thesis, University of Pretoria. Accessed October 20, 2019. http://upetd.up.ac.za/thesis/available/etd-04172007-153701/.
author], [No. “Auditory processing disorders: training curriculum for communication pathologists within the South African context .” 2007. Web. 20 Oct 2019.
author] [. Auditory processing disorders: training curriculum for communication pathologists within the South African context . [Internet] [Masters thesis]. University of Pretoria; 2007. [cited 2019 Oct 20]. Available from: http://upetd.up.ac.za/thesis/available/etd-04172007-153701/.
author] [. Auditory processing disorders: training curriculum for communication pathologists within the South African context . [Masters Thesis]. University of Pretoria; 2007. Available from: http://upetd.up.ac.za/thesis/available/etd-04172007-153701/
19. Pilon, Suléne. Outomatiese Afrikaanse woordsoortetikettering / deur Suléne Pilon .
► Any community that wants to be part of technological progress has to ensure that the language(s) of that community has/have the necessary human language technology… (more)
▼ Any community that wants to be part of technological progress has to ensure that the language(s) of that community has/have the necessary human language technology resources. Part of these resources are so-called "core technologies", including part-of-speech taggers. The first part-of-speech tagger for Afrikaans is developed in this research project. It is indicated that three resources (a tag set, a twig algorithm and annotated training data) are necessary for the development of such a part-of-speech tagger. Since none of these resources exist for Afrikaans, three objectives are formulated for this project, i.e. (a) to develop a linpsticdy accurate tag set for Afrikaans; (b) to deter- mine which algorithm is the most effective one to use; and (c) to find an effective method for generating annotated Afrikaans training data. To reach the first objective, a unique and language-specific tag set was developed for Afrikaans. The resulting tag set is relatively big and consists of 139 tags. The level of specificity of the tag set can easily be adjusted to make the tag set smaller and less specific. After the development of the tag set, research is done on different approaches to, and techniques that can be used in, the development of a part-of-speech tagger. The available algorithms are evaluated by means of prerequisites that were set and in doing so, the most effective algorithm for the purposes of this project, TnT, is identified. Bootstrapping is then used to generate training data with the help of the TnT algorithm. This process results in 20,000 correctly annotated words, and thus annotated training data, the hard resource which is necessary for the development of a part-of-speech tagger, is developed. The tagger that is trained with 20,000 words reaches an accuracy of 85.87% when evaluated. The tag set is then simplified to thirteen tags in order to determine the effect that the size of the tag set has on the accuracy of the tagger. The tagger is 93.69% accurate when using the diminished tag set. The main conclusion of this study is that training data of 20,000 words is not enough for the Afrikaans TnT tagger to compete with other state-of-the-art taggers. The tagger and the data that is developed in this project can be used to generate even more training data in order to develop an optimally accurate Afrikaans TnT tagger. Different techniques might also lead to better results; therefore other algorithms should be tested.
Subjects/Keywords: Afrikaans; Part-of-speech tagging; Syntax; Morphology; Natural language processing; Computasional linguistics
Pilon, S. (2005). Outomatiese Afrikaanse woordsoortetikettering / deur Suléne Pilon . (Thesis). North-West University. Retrieved from http://hdl.handle.net/10394/726
Pilon, Suléne. “Outomatiese Afrikaanse woordsoortetikettering / deur Suléne Pilon .” 2005. Thesis, North-West University. Accessed October 20, 2019. http://hdl.handle.net/10394/726.
Pilon, Suléne. “Outomatiese Afrikaanse woordsoortetikettering / deur Suléne Pilon .” 2005. Web. 20 Oct 2019.
Pilon S. Outomatiese Afrikaanse woordsoortetikettering / deur Suléne Pilon . [Internet] [Thesis]. North-West University; 2005. [cited 2019 Oct 20]. Available from: http://hdl.handle.net/10394/726.
Pilon S. Outomatiese Afrikaanse woordsoortetikettering / deur Suléne Pilon . [Thesis]. North-West University; 2005. Available from: http://hdl.handle.net/10394/726
20. Muller, J. J. USB telephony interface device for speech recognition applications.
Automatic speech recognition (ASR) systems are an attractive means for companies to deliver value added… (more)
Automatic speech recognition (ASR) systems are an attractive means for companies to deliver value added services with which to improve customer satisfaction. Such ASR systems require a telephony interface to connect the speech recognition application to the telephone system. Commercially available telephony interfaces are usually operating system specific, and therefore hardware device driver issues complicate the development of software applications for different platforms that require telephony access. The drivers and application programming interface (API) for telephony interfaces are often available only for the Microsoft Windows operating systems. This poses a problem, as many of the software tools used for speech recognition research and development operate only on Linux-based computers. These interfaces are also typically in PCI/ISA card format, which hinders physical portability of the device to another computer. A simple, cheaper and easier to use USB telephony interface device, offering cross-platform portability, was developed and presented, together with the necessary API.
Subjects/Keywords: Electronic engineering; Telephone systems; Speech processing systems; Automatic speech recognition; Electrical and Electronic Engineering
Muller, J. J. (2005). USB telephony interface device for speech recognition applications . (Thesis). Stellenbosch University. Retrieved from http://hdl.handle.net/10019.1/2757
Muller, J J. “USB telephony interface device for speech recognition applications.” 2005. Thesis, Stellenbosch University. Accessed October 20, 2019. http://hdl.handle.net/10019.1/2757.
Muller, J J. “USB telephony interface device for speech recognition applications.” 2005. Web. 20 Oct 2019.
Muller JJ. USB telephony interface device for speech recognition applications. [Internet] [Thesis]. Stellenbosch University; 2005. [cited 2019 Oct 20]. Available from: http://hdl.handle.net/10019.1/2757.
Muller JJ. USB telephony interface device for speech recognition applications. [Thesis]. Stellenbosch University; 2005. Available from: http://hdl.handle.net/10019.1/2757
21. Hodgkinson, Warren. Interactive speech-driven facial animation.
One of the fastest developing areas in the entertainment industry is digital animation. Television programmes and movies frequently use 3D animations to enhance or replace… (more)
One of the fastest developing areas in the entertainment industry is digital animation. Television programmes and movies frequently use 3D animations to enhance or replace actors and scenery. With the increase in computing power, research is also being done to apply these animations in an interactive manner. Two of the biggest obstacles to the success of these undertakings are control (manipulating the models) and realism. This text describes many of the ways to improve control and realism aspects, in such a way that interactive animation becomes possible. Specifically, lip-synchronisation (driven by human speech), and various modeling and rendering techniques are discussed. A prototype that shows that interactive animation is feasible, is also described.
Mr. A. Hardy Prof. S. von Solms
Subjects/Keywords: Computer animation; Speech processing systems; Three dimensional imaging; Computer simulation; Signal processing (digital techniques)
Hodgkinson, W. (2008). Interactive speech-driven facial animation . (Thesis). University of Johannesburg. Retrieved from http://hdl.handle.net/10210/807
Hodgkinson, Warren. “Interactive speech-driven facial animation.” 2008. Thesis, University of Johannesburg. Accessed October 20, 2019. http://hdl.handle.net/10210/807.
Hodgkinson, Warren. “Interactive speech-driven facial animation.” 2008. Web. 20 Oct 2019.
Hodgkinson W. Interactive speech-driven facial animation. [Internet] [Thesis]. University of Johannesburg; 2008. [cited 2019 Oct 20]. Available from: http://hdl.handle.net/10210/807.
Hodgkinson W. Interactive speech-driven facial animation. [Thesis]. University of Johannesburg; 2008. Available from: http://hdl.handle.net/10210/807
22. Theron, Karin. Temporal aspects of speech production in bilingual speakers with neurogenic speech disorders.
Theron, Karin. “Temporal aspects of speech production in bilingual speakers with neurogenic speech disorders.” 2004. Masters Thesis, University of Pretoria. Accessed October 20, 2019. http://hdl.handle.net/2263/27081.
Theron, Karin. “Temporal aspects of speech production in bilingual speakers with neurogenic speech disorders.” 2004. Web. 20 Oct 2019.
Theron K. Temporal aspects of speech production in bilingual speakers with neurogenic speech disorders. [Internet] [Masters thesis]. University of Pretoria; 2004. [cited 2019 Oct 20]. Available from: http://hdl.handle.net/2263/27081.
23. [No author]. Temporal aspects of speech production in bilingual speakers with neurogenic speech disorders .
author], [No. “Temporal aspects of speech production in bilingual speakers with neurogenic speech disorders .” 2004. Masters Thesis, University of Pretoria. Accessed October 20, 2019. http://upetd.up.ac.za/thesis/available/etd-08072003-152242/.
author], [No. “Temporal aspects of speech production in bilingual speakers with neurogenic speech disorders .” 2004. Web. 20 Oct 2019.
author] [. Temporal aspects of speech production in bilingual speakers with neurogenic speech disorders . [Internet] [Masters thesis]. University of Pretoria; 2004. [cited 2019 Oct 20]. Available from: http://upetd.up.ac.za/thesis/available/etd-08072003-152242/.
24. Schlünz, Georg Isaac. Advanced natural language processing for improved prosody in text-to-speech synthesis / G. I. Schlünz .
► Text-to-speech synthesis enables the speech-impeded user of an augmentative and alternative communication system to partake in any conversation on any topic, because it can produce… (more)
▼ Text-to-speech synthesis enables the speech-impeded user of an augmentative and alternative communication system to partake in any conversation on any topic, because it can produce dynamic content. Current synthetic voices do not sound very natural, however, lacking in the areas of emphasis and emotion. These qualities are furthermore important to convey meaning and intent beyond that which can be achieved by the vocabulary of words only. Put differently, speech synthesis requires a more comprehensive analysis of its text input beyond the word level to infer the meaning and intent that elicit emphasis and emotion. The synthesised speech then needs to imitate the effects that these textual factors have on the acoustics of human speech. This research addresses these challenges by commencing with a literature study on the state of the art in the fields of natural language processing, text-to-speech synthesis and speech prosody. It is noted that the higher linguistic levels of discourse, information structure and affect are necessary for the text analysis to shape the prosody appropriately for more natural synthesised speech. Discourse and information structure account for meaning, intent and emphasis, and affect formalises the modelling of emotion. The OCC model is shown to be a suitable point of departure for a new model of affect that can leverage the higher linguistic levels. The audiobook is presented as a text and speech resource for the modelling of discourse, information structure and affect because its narrative structure is prosodically richer than the random constitution of a traditional text-to-speech corpus. A set of audiobooks are selected and phonetically aligned for subsequent investigation. The new model of discourse, information structure and affect, called e-motif, is developed to take advantage of the audiobook text. It is a subjective model that does not specify any particular belief system in order to appraise its emotions, but defines only anonymous affect states. Its cognitive and social features rely heavily on the coreference resolution of the text, but this process is found not to be accurate enough to produce usable features values. The research concludes with an experimental investigation of the influence of the e-motif features on human speech and synthesised speech. The aligned audiobook speech is inspected for prosodic correlates of the cognitive and social features, revealing that some activity occurs in the into national domain. However, when the aligned audiobook speech is used in the training of a synthetic voice, the e-motif effects are overshadowed by those of structural features that come standard in the voice building framework.
Subjects/Keywords: Natural language processing; Text-to-speech synthesis; Prosody; Discourse; Information structure; Affect; OCC model; E-motif
…Network DP Dynamic Programming DSP Digital Signal Processing DTW Dynamic Time Warping F0… …Toolkit HTS HMM-Based Speech Synthesis System IP Intonation Phrase JND Just Noticeable… …Difference MFCC Mel-Frequency Cepstral Coefficients NLP Natural Language Processing NSR… …Nuclear Stress Rule OCC Ortony, Clore and Collins POS Part-of-Speech PP Phonological… …TTS Text-to-Speech WDP Word DP Score XP Syntactic Phrase vii Chapter 1 Introduction…
Schlünz, G. I. (2014). Advanced natural language processing for improved prosody in text-to-speech synthesis / G. I. Schlünz . (Thesis). North-West University. Retrieved from http://hdl.handle.net/10394/10634
Schlünz, Georg Isaac. “Advanced natural language processing for improved prosody in text-to-speech synthesis / G. I. Schlünz .” 2014. Thesis, North-West University. Accessed October 20, 2019. http://hdl.handle.net/10394/10634.
Schlünz, Georg Isaac. “Advanced natural language processing for improved prosody in text-to-speech synthesis / G. I. Schlünz .” 2014. Web. 20 Oct 2019.
Schlünz GI. Advanced natural language processing for improved prosody in text-to-speech synthesis / G. I. Schlünz . [Internet] [Thesis]. North-West University; 2014. [cited 2019 Oct 20]. Available from: http://hdl.handle.net/10394/10634.
Schlünz GI. Advanced natural language processing for improved prosody in text-to-speech synthesis / G. I. Schlünz . [Thesis]. North-West University; 2014. Available from: http://hdl.handle.net/10394/10634
25. Visser, Christina Magdalena. Die toepaslikheid van 'n Afrikaanse vertaling van die Scan-C : test for auditory processing disorders in children-revised vir voorskoolse leerders (Afrikaans).
► Currently a need for auditory processing test material for the identification of auditory processing disorders in children exists in the South African context. The purpose… (more)
▼ Currently a need for auditory processing test material for the identification of auditory processing disorders in children exists in the South African context. The purpose of this study was the Afrikaans translation and cultural adaptation of the SCAN-C: Test for Auditory Processing Disorders in Children-Revised (Keith, 2000a) and to determine the appropriateness thereof for preschool children. To achieve the objective of this study the research was performed in three phases and quantitative-descriptive research was utilized, with different methods for each of the phases. In Phase I the SCAN-C (Keith, 2000a) was translated and culturally adapted into Afrikaans for learners in Grade R. In Phase II the Siftingstoets vir Ouditiewe Prosesseringsafwykings (STOPA) was reviewed by speech-language therapists and/or audiologists as well as Grade R teachers (n = 11) by means of a customized questionnaire to determine appropriateness of the STOPA. A pilot study was conducted on Grade R learners (n = 10) before the STOPA was finalized and recorded. In Phase III the STOPA was performed on Grade R learners identified with normal auditory processing (n = 16) as well as Grade R learners identified with an auditory processing disorder (n = 14) to determine the validity and accuracy of the STOPA. The results indicated that the STOPA is age and culturally appropriate but that the validity and accuracy of the STOPA could not yet be proved. Clinical implications for identification, diagnosis and intervention for children with auditory processing disorders were discussed in order to provide the audiologist and speech-language therapist with evidence-based principles for clinical practice. Implications for further research include refining the STOPA during which normative data should be obtained. This study can therefore be viewed as the first step to standardize the STOPA, as well as to develop test material for other language and culture groups for the early identification of auditory processing disorders in children. Advisors/Committee Members: Dr N G Campbell (advisor), Prof B Louw (advisor).
Subjects/Keywords: Ouditiewe prosesseringsafwykings; Voorskoolse identifisering; Oudioloog en spraak-taalterapeut; Diagnostiese ouditiewe prosesseringstoetse; Preschool identification; Auditory processing disorders; Multidisciplinary approach; Audiologist and speech language therapist; Auditory processing screening tests; Diagnostic auditory processing tests; Ouditiewe prosesseringsiftingstoetse; Multidissiplinêre benadering; UCTD
Visser, C. (2007). Die toepaslikheid van 'n Afrikaanse vertaling van die Scan-C : test for auditory processing disorders in children-revised vir voorskoolse leerders (Afrikaans) . (Masters Thesis). University of Pretoria. Retrieved from http://hdl.handle.net/2263/24665
Visser, Christina. “Die toepaslikheid van 'n Afrikaanse vertaling van die Scan-C : test for auditory processing disorders in children-revised vir voorskoolse leerders (Afrikaans).” 2007. Masters Thesis, University of Pretoria. Accessed October 20, 2019. http://hdl.handle.net/2263/24665.
Visser, Christina. “Die toepaslikheid van 'n Afrikaanse vertaling van die Scan-C : test for auditory processing disorders in children-revised vir voorskoolse leerders (Afrikaans).” 2007. Web. 20 Oct 2019.
Visser C. Die toepaslikheid van 'n Afrikaanse vertaling van die Scan-C : test for auditory processing disorders in children-revised vir voorskoolse leerders (Afrikaans). [Internet] [Masters thesis]. University of Pretoria; 2007. [cited 2019 Oct 20]. Available from: http://hdl.handle.net/2263/24665.
Visser C. Die toepaslikheid van 'n Afrikaanse vertaling van die Scan-C : test for auditory processing disorders in children-revised vir voorskoolse leerders (Afrikaans). [Masters Thesis]. University of Pretoria; 2007. Available from: http://hdl.handle.net/2263/24665
26. [No author]. Die toepaslikheid van 'n Afrikaanse vertaling van die Scan-C : test for auditory processing disorders in children-revised vir voorskoolse leerders (Afrikaans) .
author], [. (2007). Die toepaslikheid van 'n Afrikaanse vertaling van die Scan-C : test for auditory processing disorders in children-revised vir voorskoolse leerders (Afrikaans) . (Masters Thesis). University of Pretoria. Retrieved from http://upetd.up.ac.za/thesis/available/etd-05152007-092134/
author], [No. “Die toepaslikheid van 'n Afrikaanse vertaling van die Scan-C : test for auditory processing disorders in children-revised vir voorskoolse leerders (Afrikaans) .” 2007. Masters Thesis, University of Pretoria. Accessed October 20, 2019. http://upetd.up.ac.za/thesis/available/etd-05152007-092134/.
author], [No. “Die toepaslikheid van 'n Afrikaanse vertaling van die Scan-C : test for auditory processing disorders in children-revised vir voorskoolse leerders (Afrikaans) .” 2007. Web. 20 Oct 2019.
author] [. Die toepaslikheid van 'n Afrikaanse vertaling van die Scan-C : test for auditory processing disorders in children-revised vir voorskoolse leerders (Afrikaans) . [Internet] [Masters thesis]. University of Pretoria; 2007. [cited 2019 Oct 20]. Available from: http://upetd.up.ac.za/thesis/available/etd-05152007-092134/.
author] [. Die toepaslikheid van 'n Afrikaanse vertaling van die Scan-C : test for auditory processing disorders in children-revised vir voorskoolse leerders (Afrikaans) . [Masters Thesis]. University of Pretoria; 2007. Available from: http://upetd.up.ac.za/thesis/available/etd-05152007-092134/
27. Oosthuizen, Ilze. Bilateral processing benefit in sequentially implanted adult cochlear implant users.
► Bilateral cochlear implantation is accepted medical practice since 2008 in clinically suitable adults and children to enhance bilateral processing benefits. Bilateral implantation may lead to… (more)
▼ Bilateral cochlear implantation is accepted medical practice since 2008 in clinically suitable adults and children to enhance bilateral processing benefits. Bilateral implantation may lead to the restoration of some bilateral hearing advantages, such as improved speech recognition in noise, localisation, head shadow effect, summation, and squelch. The majority of the advantages stated in literature, though, are characteristic of the simultaneously implanted cochlear implant population. Simultaneous implantation is not yet a reality in South Africa due to funding constraints, therefore determining the bilateral processing abilities in sequentially implanted adults is essential. Determining bilateral processing benefits achievable with sequential implantation could result in evidence-based recommendations in terms of candidacy considerations, surgery protocols, motivations for medical aid funding for simultaneous cochlear implantation, and relevant measures to determine the bilateral processing benefit attainable. Furthermore, it might enhance audiologists‟ insight regarding post-implantation performance of sequentially implanted patients and enable them to counsel prospective candidates realistically. The aaim of this study was to determine the bilateral benefit attained by sequentially implanted adults. A quantitative, cross-sectional research approach was followed in a one group post-test-only exploratory research design. A purposive convenient sampling method with specified selection criteria was used to select 11 adult clients of an established cochlear implant programme in Pretoria. Tests of sound localisation in the horizontal plane and speech perception in noise were performed. During the test of sound localisation, performance with only the first or only the second implant was found to be very similar. For the majority of participants the second cochlear implant (CI 2) was the superior performing implant during xviii speech perception in noise testing, in spatially separated speech and noise conditions where noise was directed to the first implant, as well as in spatially coincident speech and noise. A statistical significant bilateral benefit (p < 0.05) was attained by sequentially implanted adults for sound localisation. A bilateral benefit for speech perception in noise was observed when noise was directed to the first implant and in the diotic listening condition with average benefits of 1.69 dB and 0.78 dB, respectively. It was not statistically significant (p > 0.05), however, and was smaller than bilateral benefit values achieved by simultaneously implanted adults in previous studies. The head shadow effect at 180° was found to be the strongest and most robust bilateral spatial benefit. Squelch and summation benefit values ranged from negative values to 2 dB and 6 dB, respectively. This corresponded with values found in previous studies. The improvement in speech perception in spatially distinct speech and noise from adding the ear with a better SNR (signal to noise ratio) indicated that the… Advisors/Committee Members: Prof D C D Swanepoel (advisor).
Subjects/Keywords: Kogleêre inplanting met beste prestasie; Speech-perception-in-noise; Summation; Squelch; Superior performing cochlear implant; Sound localisation; Simultaneous implantation; Sequential implantation; Head shadow effect; Cochlear implant; Bilateral processing; Bilateral implantation; Bilateral benefit; Sommering; Spraakwaarneming in lawaai; Selektiewe onderdrukking; Klanklokalisering; Gelyktydige inplanting; Opeenvolgende inplantting; Kopskadu-effek; Bilaterale prosessering; Bilaterale voordeel; Kogleêre inplanting; Bilaterale inplanting; UCTD
Oosthuizen, I. (2012). Bilateral processing benefit in sequentially implanted adult cochlear implant users . (Masters Thesis). University of Pretoria. Retrieved from http://hdl.handle.net/2263/30304
Oosthuizen, Ilze. “Bilateral processing benefit in sequentially implanted adult cochlear implant users.” 2012. Masters Thesis, University of Pretoria. Accessed October 20, 2019. http://hdl.handle.net/2263/30304.
Oosthuizen, Ilze. “Bilateral processing benefit in sequentially implanted adult cochlear implant users.” 2012. Web. 20 Oct 2019.
Oosthuizen I. Bilateral processing benefit in sequentially implanted adult cochlear implant users. [Internet] [Masters thesis]. University of Pretoria; 2012. [cited 2019 Oct 20]. Available from: http://hdl.handle.net/2263/30304.
Oosthuizen I. Bilateral processing benefit in sequentially implanted adult cochlear implant users. [Masters Thesis]. University of Pretoria; 2012. Available from: http://hdl.handle.net/2263/30304
28. [No author]. Bilateral processing benefit in sequentially implanted adult cochlear implant users .
author], [. (2012). Bilateral processing benefit in sequentially implanted adult cochlear implant users . (Masters Thesis). University of Pretoria. Retrieved from http://upetd.up.ac.za/thesis/available/etd-12092011-164009/
author], [No. “Bilateral processing benefit in sequentially implanted adult cochlear implant users .” 2012. Masters Thesis, University of Pretoria. Accessed October 20, 2019. http://upetd.up.ac.za/thesis/available/etd-12092011-164009/.
author], [No. “Bilateral processing benefit in sequentially implanted adult cochlear implant users .” 2012. Web. 20 Oct 2019.
author] [. Bilateral processing benefit in sequentially implanted adult cochlear implant users . [Internet] [Masters thesis]. University of Pretoria; 2012. [cited 2019 Oct 20]. Available from: http://upetd.up.ac.za/thesis/available/etd-12092011-164009/.
author] [. Bilateral processing benefit in sequentially implanted adult cochlear implant users . [Masters Thesis]. University of Pretoria; 2012. Available from: http://upetd.up.ac.za/thesis/available/etd-12092011-164009/