The problem of this study is to compare the effectiveness of two methods of teaching beginning typewriting in the community college. The two methods are an audio-tutorial approach and the traditional textbook approach. Groups taught by the contrasting methods of instruction were compared on the basis of their production performance and their straight-copy skills after thirty-six class periods of instruction. A comparison was also made of the attrition rate of the two groups.

Cette thèse porte sur des analyses acoustiques et prosodiques du français à partir de grandes masses de données orales illustrant différents styles de parole (préparée et spontanée). Nous nous sommes intéressées aux attributs acoustiques et prosodiques qui pourraient caractériser la prononciation. En français, de nombreuses erreurs de reconnaissance automatique de la parole (RAP) sont dues à des mots fréquents homophones. Pour ces mots, la solution correcte dépend du modèle de langage. Une classification automatique (CA) a été effectuée pour discriminer deux paires homophones (‘et/est’ et ‘à/a’) par des propriétés acoustiques et prosodiques. Les résultats de la CA ont montré que le paire ‘et/est’ était plus dissociable. La CA par des attributs prosodiques et inter-segmentaux (15 attributs) s’est avérée aussi performante que celle utilisant la totalité des 62 attributs. Un test perceptif a été également effectué pour vérifier si les humains utilisaient eux aussi ces paramètres. Les résultats ont suggéré que des informations acoustiques et prosodiques pourraient être utiles pour effectuer un choix correct de mots dans des structures syntaxiquement ambigües. Ensuite, nous avons examiné des propriétés prosodiques globales aux niveaux du nom et du syntagme nominal. La comparaison entre mots lexicaux et grammaticaux a montré que la fréquence fondamentale (F0) montante et l’allongement vocalique de la dernière syllabe caractérisent les mots lexicaux, par opposition aux mots grammaticaux. Ainsi, le profil de F0 moyenne d’un syntagme nominal de longueur n pourrait être différent de celui du nom avec une valeur de F0 basse au début du syntagme. Les profils prosodiques peuvent être utiles pour localiser frontières de mots. Les résultats de ce travail pourront servir à localiser le focus et les entités-nommées par des classifieurs discriminants, et de manière plus générale à améliorer les techniques de localisation des frontières des mots pour la RAP. / This thesis focuses on acoustic and prosodic (fundamental frequency (F0), duration, intensity) analyses of French from large-scale audio corpora portraying different speaking styles: prepared and spontaneous speech. We are interested in particularities of segmental phonetics and prosody that may characterize pronunciation. In French, many errors caused by automatic speech recognition (ASR) systems arise from frequent homophone words, for which ASR systems depend on language model weights. Automatic classification (AC) was conducted to discriminate homophones by only acoustic and prosodic properties depending on their part-of-speech function or their position within prosodic words. Results from AC of two homophone pairs, et/est (and/is) and à/a (ton/has), revealed that the et/est pair was more discriminable. A selection of prosodic and inter-phoneme attributes, that is 15 attributes, performed as good results as with 62 attributes. Then corresponding perceptual tests have been conducted to verify if humans also use acoustico-prosodic parameters for the discrimination. Results suggested that acoustic and prosodic information might help in operating the correct choice in similar ambiguous syntactic structures. From the hypothesis that pronunciation variants were due to varying prosodic constraints, we examined overall prosodic properties of French on a lexical and phrase level. The comparison between lexical and grammatical words revealed F0 rise and lengthening at the end of final syllable on lexical words, while these phenomena were not observed for grammatical words. Analyses also revealed that the mean profile of a n length noun phrase could be different from that of a n length noun with a low F0 at the beginning of a noun phrase. The prosodic profiles can be helpful to locate word boundaries. Findings in this thesis will lead to localize focus and named-entity using discriminative classifiers, and to improve word boundary locations by an ASR post-processing step.

[pt] As tarefas de segmentação e classificação automáticas de áudio vêm se tornando cada vez mais importantes com o crescimento da produção e armazenamento de mídia digital. Este trabalho se baseia em características do padrão MPEG, que é considerado o padrão para acervos digitais, para gerir algoritmos de grande eficiência para realizar essas arefas. Ao passo que há muitos estudos trabalhando a partir do vídeo, o áudio ainda é pouco utilizado de forma eficiente para auxiliar nessas tarefas. Os algoritmos sugeridos partem da leitura apenas dos fatores de escala presentes no Layer 2 do áudio MPEG para ambas as tarefas. Com isso, é necessária a leitura da menor quantidade possível de informações, o que diminui significativamente o volume de dados manipulado durante a análise e torna seu desempenho excelente em termos de tempo de processamento. O algoritmo proposto para a classificação divide o áudio em quatro possíveis tipos: silêncio, fala, música e aplausos. Já o algoritmo de segmentação encontra as mudanças ignificativas de áudio, que são indícios de segmentos e mudanças de cena. Foram realizados testes com diferentes tipos de vídeos, e ambos os algoritmos mostraram bons resultados. / [en] With the growth of production and storing of digital media, audio segmentation and classification are becoming increasingly important. This work is based on characteristics of the MPEG standard, considered to be the standard for digital media storage and retrieval, to propose efficient algorithms to perform these tasks. While there are many studies based on video analysis, the audio information is still not widely used in an efficient way. The suggested algorithms for both tasks are based only on the scale factors present on layer 2 MPEG audio. That allows them to read the smallest amount of information possible, significantly diminishing the amount of data manipulated during the analysis and making their performance excellent in terms of processing time. The algorithm proposed for audio classification divides audio in four possible types: silent, speech, music and applause. The segmentation algorithm finds significant changes on the audio signal that represent clues of audio segments and scene changes. Tests were made with a wide range of types of video, and both algorithms show good results.

Made available in DSpace on 2016-04-26T18:18:19Z (GMT). No. of bitstreams: 1 Mauricio Nacib Pontuschka.pdf: 2745462 bytes, checksum: 95bf3c2456cd0f013c649839fefcf0bd (MD5) Previous issue date: 2009-11-10 / This thesis explores the audio field in current computerized systems, so as to contribute to the use of intrinsic characteristics of audio and its expressiveness in communicational digital environments. The issue under study is related to systematized articulation of audio, which allows the use of techniques that enable the randomness of sounds, thus broadening the conceptual range of expressiveness of cognitive and communicational processes in hypermedia environments and metaverses. The philosophical foundations of this work are on the area of Linguistics (Ferdinand de Saussure) and Semiotics (Charles Pierce), which justify the understanding of the role of audio in current communication processes. The studies of George Landow and Theodore Nelson on Hypertexts had the role of bringing into light the computerized systematization ability of communication processes, what has motivated us to explore audio, its peculiarities and computational specificities in digital environments. In this context, the random component is introduced in the systematization so that the elements of human perception detailed by Charles Pierce can be used individually. Afterwards we will discuss, some possibilities of application of random audio understood as sonic textures in communication processes and music. As main result, this work proposes that random audio can be employed in research projects of virtual environments (metaverses), culminating in a formulation of soundscapes (Murray Shafer) starting from the concept of hyperaudio / Esta tese explora o campo do áudio nos atuais sistemas computadorizados, de forma a contribuir para a utilização das características intrínsecas do áudio e de sua expressividade em ambientes digitais comunicacionais. A questão estudada volta-se para a articulação sistematizada do áudio, que permite a utilização de técnicas que viabilizam a aleatoriedade de sons, ampliando assim o leque das expressividades conceituais para os processos cognitivos e comunicacionais em ambientes hipermídias e metaversos. Colocando as bases filosóficas da área da Linguística (Ferdinand Saussure) e da Semiótica (Charles Pierce) fundamentamos o entendimento a respeito do papel do áudio nos processos atuais de comunicação. Os estudos de George Landow e Teodore Nelson a respeito de Hipertextos tiveram o papel de evidenciar a capacidade de sistematização computadorizada dos processos comunicacionais o que nos motivou a explorar o áudio em ambientes digitais, suas peculiaridades e especificidades computacionais. A partir deste contexto a componente aleatória é introduzida nesta sistematização de maneira que os elementos da percepção humana detalhados por Charles Pierce possam ser utilizados individualmente. Na sequência são discutidas possibilidades de aplicação do áudio aleatório, entendido como texturas sonoras, nos processos de comunicação e na música. Como principal resultado, este trabalho propõe que o áudio aleatório pode ser empregado em projetos de pesquisa de ambientes virtuais (metaversos), culminando em uma formulação de paisagens sonoras (Murray Shafer) a partir da ideia do conceito de hiperáudio

The problem of this study was to obtain empirical evidence of the functional nature of the Audio-visual Identification Instruction (AVII) model for designing effective music instruction for young children. The method was to use materials prepared according to the model specifications in actual classroom conditions. The purpose of the study was to compare the achievement gain of second grade children of high, middle, or low musical aptitude levels, who were instructed by experienced music specialists, first year music specialists, student teacher music specialists, or experienced classroom teachers using AVII model materials, on three tasks in the area of pitch and three tasks in the area of timbre. Subject to the circumstances and limitations of this investigation, the results indicate that the AVII model is effective for instruction for musical naming and identification tasks for young children.

<p>This thesis describes the design and implementation of a Digital Audio Broadcasting (DAB) System developed using C++ Language and SystemC libraries. The main aspects covered within this report are the data structure of DAB system, and some interesting points of SystemC Library very useful for the implementation of the final system. </p><p>It starts with a introduction of DAB system and his principals advantages. Next it goes further into the definition of data structures of DAB, they are FIC, MSC, and DAB audio frame, explained with MPEG and PAD packets. Later on this chapter there is an explanation of the SystemC library with special attention on the features that I used to implement the system. This features are the events used in the communication between processes and the interfaces needed for sending and receiving the data.</p><p>With all these points covered is quite easy for a reader to understand the implementation of the system, despite this point is covered in the last chapter of the thesis. The implementation is here explained in two different steps. The first one explain how is formed the DAB audio frame by means of MPEG frames that are wrote in channel by producer interface, this frames are readed by consumer interface. For this purpose I have created some classes and structures that are explained in this part. The second part explain how I obtain the DAB transmission frame which is obtained creating MSC frames, that are big data structures formed by groups of DAB audio frames, therefore there are some functions that act like a buffer and add audio frames to the MSC data structure. Of independent way there is the FIC frame that is generated of random way and its added to the transmission frame.</p>

A questionnaire related to environmental awareness was administered to 244 students enrolled in NR 101, Introduction to Natural Resources, during the Spring Quarter of 1979 in an attempt to answer the following questions: (1) To what extent will viewing a slide-tape presentation alter student responses on a questionnaire? (2) To what extent will the responses of the students viewing a slide-tape presentation differ from a control treatments result in significant differences in responses when compared to a control group?Results of this research indicate the treatments did alter the students' responses on one factor. The responses were altered towards an environmentally aware position and there were significant differences between treatment group responses when compared to the control group.Ball State UniversityMuncie, IN 47306

This thesis describes the design and implementation of a Digital Audio Broadcasting (DAB) System developed using C++ Language and SystemC libraries. The main aspects covered within this report are the data structure of DAB system, and some interesting points of SystemC Library very useful for the implementation of the final system. It starts with a introduction of DAB system and his principals advantages. Next it goes further into the definition of data structures of DAB, they are FIC, MSC, and DAB audio frame, explained with MPEG and PAD packets. Later on this chapter there is an explanation of the SystemC library with special attention on the features that I used to implement the system. This features are the events used in the communication between processes and the interfaces needed for sending and receiving the data. With all these points covered is quite easy for a reader to understand the implementation of the system, despite this point is covered in the last chapter of the thesis. The implementation is here explained in two different steps. The first one explain how is formed the DAB audio frame by means of MPEG frames that are wrote in channel by producer interface, this frames are readed by consumer interface. For this purpose I have created some classes and structures that are explained in this part. The second part explain how I obtain the DAB transmission frame which is obtained creating MSC frames, that are big data structures formed by groups of DAB audio frames, therefore there are some functions that act like a buffer and add audio frames to the MSC data structure. Of independent way there is the FIC frame that is generated of random way and its added to the transmission frame.

I denna uppsats har filmljudet i krigsfilmerna Apocalypse Now och Saving Private Ryan undersökts. Detta har gjorts för att försöka bidra med ökad förståelse för filmljudets användningsområde och funktioner, främst för filmerna i fråga, men även för krigsfilm rent generellt. Filmljud i denna kontext omfattar allt det ljud som finns i film, men utesluter dock all ickediegetisk musik. Båda filmerna har undersökts genom en audio-visuell analys. En sådan analys görs genom att detaljgranska båda filmernas ljud- och bildinnehåll var för sig, för att slutligen undersöka samma filmsekvens som helhet då ljudet och bilden satts ihop igen. Den audio-visuella analysmetod som nyttjats i uppsatsen är Michel Chions metod, Masking. De 30 minuter film som analyserades placerades sedan i olika filmljudzoner, där respektive filmljudzons ljudinnehåll bland annat visade vilka främsta huvudfunktioner somfilmljudet hade i dessa filmer. Dessa funktioner är till för att bibehålla åskådarens fokus och intresse, att skapa närhet till rollkaraktärerna, samt att tillföra en hög känsla av realism och närvaro. Intentionerna med filmljudet verkade vara att flytta åskådaren in i filmens verklighet, att låta åskådaren bli ett med filmen. Att återspegla denna känsla av realism, närvaro, fokus samt intresse, visade sig också vara de intentioner som funnits redan i de båda filmernas förproduktionsstadier. Detta bevisar att de lyckats åstadkomma det de eftersträvat. Men om filmljudet använts på samma sätt eller innehar samma funktioner i krigsfilm rent genrellt går inte att säga.I have for this bachelor’s thesis examined the movie sound of the classic warfare movies Apocalypse Now and Saving Private Ryan. This is an attempt to contribute to a more profound comprehension of the appliance and importance of movie sound. In this context movie sound implies all kinds of sounds within the movies, accept from non-diegetic music. These two movies have been examined by an audio-visual analysis. It's done by auditing the sound and picture content separately, and then combined to audit the same sequence as a whole. Michel Chion, which is the founder of this analysis, calls this method Masking. The sound in this 30 minute sequence was then divided into different zones, where every zone represented a certain main function. These functions are provided to create a stronger connection to the characters, sustain the viewers interest and bring a sense of realism and presence. It seems though the intention with the movies sound is to bring the viewers to the scene in hand, and let it become their reality. To mirror this sense of realism, presence, focus and interest, proves to be the intention from an early stage of the production. This bachelor’s thesis demonstrates a success in their endeavours. Although it can’t confirm whether the movie sound have been utilized in the same manner or if they posess the same functions to warefare movies in general.

The audio signals can provide rich semantic cues for analyzing multimedia content, so audio information has been recently used for content-based multimedia indexing and retrieval. Due to growing amount of audio data, demand for efficient retrieval techniques is increasing. In this thesis work, we propose a complete, scalable and extensible audio based content management and retrieval system for news broadcasts. The proposed system considers classification, segmentation, analysis and retrieval of an audio stream. In the sound classification and segmentation stage, a sound stream is segmented by classifying each sub segment into silence, pure speech, music, environmental sound, speech over music, and speech over environmental sound in multiple steps. Support Vector Machines and Hidden Markov Models are employed for classification and these models are trained by using different sets of MPEG-7 features. In the analysis and retrieval stage, two alternatives exist for users to query audio data. The first of these isolates user from main acoustic classes by providing semantic domain based fuzzy classes. The latter offers users to query audio by giving an audio sample in order to find out the similar segments or by requesting expressive summary of the content directly. Additionally, a series of tests was conducted on audio tracks of TRECVID news broadcasts to evaluate the performance of the proposed solution.