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Author : Christoph Nieuwoudt
Publisher :
ISBN 13 :
Total Pages : 237 pages
Book Rating : 4.:/5 (484 download)
Download or read book Cross-language Acoustic Adaptation for Automatic Speech Recognition written by Christoph Nieuwoudt and published by . This book was released on 2000 with total page 237 pages. Available in PDF, EPUB and Kindle. Book excerpt:
Author : L. Ashok Kumar
Publisher : John Wiley & Sons
ISBN 13 : 1394214170
Total Pages : 428 pages
Book Rating : 4.3/5 (942 download)
Download or read book Automatic Speech Recognition and Translation for Low Resource Languages written by L. Ashok Kumar and published by John Wiley & Sons. This book was released on 2024-03-28 with total page 428 pages. Available in PDF, EPUB and Kindle. Book excerpt: AUTOMATIC SPEECH RECOGNITION and TRANSLATION for LOW-RESOURCE LANGUAGES This book is a comprehensive exploration into the cutting-edge research, methodologies, and advancements in addressing the unique challenges associated with ASR and translation for low-resource languages. Automatic Speech Recognition and Translation for Low Resource Languages contains groundbreaking research from experts and researchers sharing innovative solutions that address language challenges in low-resource environments. The book begins by delving into the fundamental concepts of ASR and translation, providing readers with a solid foundation for understanding the subsequent chapters. It then explores the intricacies of low-resource languages, analyzing the factors that contribute to their challenges and the significance of developing tailored solutions to overcome them. The chapters encompass a wide range of topics, ranging from both the theoretical and practical aspects of ASR and translation for low-resource languages. The book discusses data augmentation techniques, transfer learning, and multilingual training approaches that leverage the power of existing linguistic resources to improve accuracy and performance. Additionally, it investigates the possibilities offered by unsupervised and semi-supervised learning, as well as the benefits of active learning and crowdsourcing in enriching the training data. Throughout the book, emphasis is placed on the importance of considering the cultural and linguistic context of low-resource languages, recognizing the unique nuances and intricacies that influence accurate ASR and translation. Furthermore, the book explores the potential impact of these technologies in various domains, such as healthcare, education, and commerce, empowering individuals and communities by breaking down language barriers. Audience The book targets researchers and professionals in the fields of natural language processing, computational linguistics, and speech technology. It will also be of interest to engineers, linguists, and individuals in industries and organizations working on cross-lingual communication, accessibility, and global connectivity.
Author : Christoph Nieuwoudt
Publisher :
ISBN 13 :
Total Pages : pages
Book Rating : 4.:/5 (89 download)
Download or read book Cross-langauge acoustic adaptation for automatic speech recognition written by Christoph Nieuwoudt and published by . This book was released on 2001 with total page pages. Available in PDF, EPUB and Kindle. Book excerpt:
Author : Silke Goronzy
Publisher : Springer
ISBN 13 : 3540362908
Total Pages : 135 pages
Book Rating : 4.5/5 (43 download)
Download or read book Robust Adaptation to Non-Native Accents in Automatic Speech Recognition written by Silke Goronzy and published by Springer. This book was released on 2003-07-01 with total page 135 pages. Available in PDF, EPUB and Kindle. Book excerpt: Speech recognition technology is being increasingly employed in human-machine interfaces. A remaining problem however is the robustness of this technology to non-native accents, which still cause considerable difficulties for current systems. In this book, methods to overcome this problem are described. A speaker adaptation algorithm that is capable of adapting to the current speaker with just a few words of speaker-specific data based on the MLLR principle is developed and combined with confidence measures that focus on phone durations as well as on acoustic features. Furthermore, a specific pronunciation modelling technique that allows the automatic derivation of non-native pronunciations without using non-native data is described and combined with the previous techniques to produce a robust adaptation to non-native accents in an automatic speech recognition system.
Author : Mohamed Elmahdy
Publisher : Springer Science & Business Media
ISBN 13 : 1461419069
Total Pages : 120 pages
Book Rating : 4.4/5 (614 download)
Download or read book Novel Techniques for Dialectal Arabic Speech Recognition written by Mohamed Elmahdy and published by Springer Science & Business Media. This book was released on 2012-02-10 with total page 120 pages. Available in PDF, EPUB and Kindle. Book excerpt: Novel Techniques for Dialectal Arabic Speech describes approaches to improve automatic speech recognition for dialectal Arabic. Since speech resources for dialectal Arabic speech recognition are very sparse, the authors describe how existing Modern Standard Arabic (MSA) speech data can be applied to dialectal Arabic speech recognition, while assuming that MSA is always a second language for all Arabic speakers. In this book, Egyptian Colloquial Arabic (ECA) has been chosen as a typical Arabic dialect. ECA is the first ranked Arabic dialect in terms of number of speakers, and a high quality ECA speech corpus with accurate phonetic transcription has been collected. MSA acoustic models were trained using news broadcast speech. In order to cross-lingually use MSA in dialectal Arabic speech recognition, the authors have normalized the phoneme sets for MSA and ECA. After this normalization, they have applied state-of-the-art acoustic model adaptation techniques like Maximum Likelihood Linear Regression (MLLR) and Maximum A-Posteriori (MAP) to adapt existing phonemic MSA acoustic models with a small amount of dialectal ECA speech data. Speech recognition results indicate a significant increase in recognition accuracy compared to a baseline model trained with only ECA data.
Author : Aanchan Mohan
Publisher :
ISBN 13 :
Total Pages : pages
Book Rating : 4.:/5 (953 download)
Download or read book Knowledge Transfer by Sharing Acoustic-model Parameters for Automatic Speech Recognition written by Aanchan Mohan and published by . This book was released on 2016 with total page pages. Available in PDF, EPUB and Kindle. Book excerpt: "The objective of this thesis is to develop efficient methods for the transfer of knowledge between languages and speakers by sharing acoustic model parameters for automatic speech recognition (ASR). Knowledge transfer between languages is often useful when only a limited amount of transcribed data is available for ASR system development in a target language. Additionally, boot-strapping acoustic phonetic knowledge is also seen to improve ASR performance when adequate training data is available. These scenarios are used as examples to study issues in acoustic-phonetic knowledge-transfer for ASR. Furthermore, the parameters that characterize speaker variability could often be thought to lie in a low-dimensional subspace or a manifold. Parameters for a new test speaker are often estimated with knowledge transfer from training speaker information that is parametrized as a set of subspace vectors or low-dimensional embeddings on a manifold. The technical contributions in this thesis are as follows. First, acoustic mismatch due to different recording instruments and background conditions poses a problem when training a single multi-lingual statistical model on data from multiple languages. The subspace Gaussian mixture model (SGMM), which allows for natural sharing of model parameters between acoustic-phonetic units of different languages is used in this study. A two-stage procedure is proposed to compensate for speaker variability and environmental variability, prior to multi-lingual acoustic model training. As a result of this compensation procedure, ASR performance improvements are observed for all languages used in multi-lingual acoustic model training. Experimental results are presented on Hindi and Marathi speech data on a small-vocabulary agricultural commodities task. With only one hour of available Hindi data, multi-lingual acoustic model training with Marathi is seen to improve Hindi language ASR performance significantly compared to mono-lingual training. Second, to reduce the number of context-dependent errors in Hindi, an algorithm for borrowing state-level SGMM parameters from Marathi in the multi-lingual SGMM acoustic model is proposed. A statistically significant improvement is observed in Hindi language ASR. Furthermore, in order to reduce the number of parameters in the Hindi-Marathi multi-lingual acoustic model, the use of semi-tied covariance (STC) instead of full-covariance matrices is proposed. With a reduction of a factor of five relative to full-covariance parameters, similar ASR accuracy is maintained through the use of STCs. Third, the use of multi-task training for multi-lingual neural network acoustic models is studied. The use of multi-task training provides state of the art results on a well-known large vocabulary read speech task. Experiments on cross-language adaptation when only a limited amount of target language data is available are also presented. To reduce space and time-complexity to train these networks the impact of low-rank matrix factorization of the weight matrix in the final layer is presented. Finally, parameters that model speaker variability in Linear Input Network (LIN) based speaker adaptation for deep neural networks are assumed to lie on a manifold. Obtaining speaker specific parameters is treated as a task in a multi-task learning problem. Task parameters and their low-dimensional projections are assumed to lie on a manifold. A manifold constraint as a regularization term is introduced into the cost function for estimating LIN speaker parameters during test time. Experimental results are presented to evaluate this approach." --
Author : Jean-Claude Junqua
Publisher : Springer Science & Business Media
ISBN 13 : 9401597197
Total Pages : 277 pages
Book Rating : 4.4/5 (15 download)
Download or read book Robustness in Language and Speech Technology written by Jean-Claude Junqua and published by Springer Science & Business Media. This book was released on 2013-03-09 with total page 277 pages. Available in PDF, EPUB and Kindle. Book excerpt: In this book we address robustness issues at the speech recognition and natural language parsing levels, with a focus on feature extraction and noise robust recognition, adaptive systems, language modeling, parsing, and natural language understanding. This book attempts to give a clear overview of the main technologies used in language and speech processing, along with an extensive bibliography to enable topics of interest to be pursued further. It also brings together speech and language technologies often considered separately. Robustness in Language and Speech Technology serves as a valuable reference and although not intended as a formal university textbook, contains some material that can be used for a course at the graduate or undergraduate level.
Author : Tanja Schultz
Publisher : Elsevier
ISBN 13 : 0080457622
Total Pages : 540 pages
Book Rating : 4.0/5 (84 download)
Download or read book Multilingual Speech Processing written by Tanja Schultz and published by Elsevier. This book was released on 2006-06-12 with total page 540 pages. Available in PDF, EPUB and Kindle. Book excerpt: Tanja Schultz and Katrin Kirchhoff have compiled a comprehensive overview of speech processing from a multilingual perspective. By taking this all-inclusive approach to speech processing, the editors have included theories, algorithms, and techniques that are required to support spoken input and output in a large variety of languages. Multilingual Speech Processing presents a comprehensive introduction to research problems and solutions, both from a theoretical as well as a practical perspective, and highlights technology that incorporates the increasing necessity for multilingual applications in our global community. Current challenges of speech processing and the feasibility of sharing data and system components across different languages guide contributors in their discussions of trends, prognoses and open research issues. This includes automatic speech recognition and speech synthesis, but also speech-to-speech translation, dialog systems, automatic language identification, and handling non-native speech. The book is complemented by an overview of multilingual resources, important research trends, and actual speech processing systems that are being deployed in multilingual human-human and human-machine interfaces. Researchers and developers in industry and academia with different backgrounds but a common interest in multilingual speech processing will find an excellent overview of research problems and solutions detailed from theoretical and practical perspectives. State-of-the-art research with a global perspective by authors from the USA, Asia, Europe, and South Africa The only comprehensive introduction to multilingual speech processing currently available Detailed presentation of technological advances integral to security, financial, cellular and commercial applications
Author : Matthias Woelfel
Publisher : John Wiley & Sons
ISBN 13 : 0470714077
Total Pages : 600 pages
Book Rating : 4.4/5 (77 download)
Download or read book Distant Speech Recognition written by Matthias Woelfel and published by John Wiley & Sons. This book was released on 2009-04-20 with total page 600 pages. Available in PDF, EPUB and Kindle. Book excerpt: A complete overview of distant automatic speech recognition The performance of conventional Automatic Speech Recognition (ASR) systems degrades dramatically as soon as the microphone is moved away from the mouth of the speaker. This is due to a broad variety of effects such as background noise, overlapping speech from other speakers, and reverberation. While traditional ASR systems underperform for speech captured with far-field sensors, there are a number of novel techniques within the recognition system as well as techniques developed in other areas of signal processing that can mitigate the deleterious effects of noise and reverberation, as well as separating speech from overlapping speakers. Distant Speech Recognitionpresents a contemporary and comprehensive description of both theoretic abstraction and practical issues inherent in the distant ASR problem. Key Features: Covers the entire topic of distant ASR and offers practical solutions to overcome the problems related to it Provides documentation and sample scripts to enable readers to construct state-of-the-art distant speech recognition systems Gives relevant background information in acoustics and filter techniques, Explains the extraction and enhancement of classification relevant speech features Describes maximum likelihood as well as discriminative parameter estimation, and maximum likelihood normalization techniques Discusses the use of multi-microphone configurations for speaker tracking and channel combination Presents several applications of the methods and technologies described in this book Accompanying website with open source software and tools to construct state-of-the-art distant speech recognition systems This reference will be an invaluable resource for researchers, developers, engineers and other professionals, as well as advanced students in speech technology, signal processing, acoustics, statistics and artificial intelligence fields.
Author : Shinji Watanabe
Publisher : Springer
ISBN 13 : 331964680X
Total Pages : 433 pages
Book Rating : 4.3/5 (196 download)
Download or read book New Era for Robust Speech Recognition written by Shinji Watanabe and published by Springer. This book was released on 2017-10-30 with total page 433 pages. Available in PDF, EPUB and Kindle. Book excerpt: This book covers the state-of-the-art in deep neural-network-based methods for noise robustness in distant speech recognition applications. It provides insights and detailed descriptions of some of the new concepts and key technologies in the field, including novel architectures for speech enhancement, microphone arrays, robust features, acoustic model adaptation, training data augmentation, and training criteria. The contributed chapters also include descriptions of real-world applications, benchmark tools and datasets widely used in the field. This book is intended for researchers and practitioners working in the field of speech processing and recognition who are interested in the latest deep learning techniques for noise robustness. It will also be of interest to graduate students in electrical engineering or computer science, who will find it a useful guide to this field of research.
Author : Jayren Jugpal Sooful
Publisher :
ISBN 13 :
Total Pages : pages
Book Rating : 4.:/5 (956 download)
Download or read book Automated Phoneme Mapping for Cross-language Speech Recognition written by Jayren Jugpal Sooful and published by . This book was released on 2013 with total page pages. Available in PDF, EPUB and Kindle. Book excerpt: This dissertation explores a unique automated approach to map one phoneme set to another, based on the acoustic distances between the individual phonemes. Although the focus of this investigation is on cross-language applications, this automated approach can be extended to same-language but different-database applications as well. The main goal of this investigation is to be able to use the data of a source language, to train the initial acoustic models of a target language for which very little speech data may be available. To do this, an automatic technique for mapping the phonemes of the two data sets must be found. Using this technique, it would be possible to accelerate the development of a speech recognition system for a new language. The current research in the cross-language speech recognition field has focused on manual methods to map phonemes. This investigation has considered an English-to-Afrikaans phoneme mapping, as well as an Afrikaans-to-English phoneme mapping. This has been previously applied to these language instances, but utilising manual phoneme mapping methods. To determine the best phoneme mapping, different acoustic distance measures are compared. The distance measures that are considered are the Kullback-Leibler measure, the Bhattacharyya distance metric, the Mahalanobis measure, the Euclidean measure, the L2 metric and the Jeffreys-Matusita distance. The distance measures are tested by comparing the cross-database recognition results obtained on phoneme models created from the TIMIT speech corpus and a locally-compiled South African SUN Speech database. By selecting the most appropriate distance measure, an automated procedure to map phonemes from the source language to the target language can be done. The best distance measure for the mapping gives recognition rates comparable to a manual mapping process undertaken by a phonetic expert. This study also investigates the effect of the number of Gaussian mixture components on the mapping and on the speech recognition system's performance. The results indicate that the recogniser's performance increases up to a limit as the number of mixtures increase. In addition, this study has explored the effect of excluding the Mel Frequency delta and acceleration cepstral coefficients. It is found that the inclusion of these temporal features help improve the mapping and the recognition system's phoneme recognition rate. Experiments are also carried out to determine the impact of the number of HMM recogniser states. It is found that single-state HMMs deliver the optimum cross-language phoneme recognition results. After having done the mapping, speaker adaptation strategies are applied on the recognisers to improve their target-language performance. The models of a fully trained speech recogniser in a source language are adapted to target-language models using Maximum Likelihood Linear Regression (MLLR) followed by Maximum A Posteriori (MAP) techniques. Embedded Baum-Welch re-estimation is used to further adapt the models to the target language. These techniques result in a considerable improvement in the phoneme recognition rate. Although a combination of MLLR and MAP techniques have been used previously in speech adaptation studies, the combination of MLLR, MAP and EBWR in cross-language speech recognition is a unique contribution of this study. Finally, a data pooling technique is applied to build a new recogniser using the automatically mapped phonemes from the target language as well as the source language phonemes. This new recogniser demonstrates moderate bilingual phoneme recognition capabilities. The bilingual recogniser is then further adapted to the target language using MAP and embedded Baum-Welch re-estimation techniques. This combination of adaptation techniques together with the data pooling strategy is uniquely applied in the field of cross-language recognition. The results obtained using this technique outperform all other techniques tested in terms of phoneme recognition rates, although it requires a considerably more time consuming training process. It displays only slightly poorer phoneme recognition than the recognisers trained and tested on the same language database.
Author : Yongxin Zhang
Publisher :
ISBN 13 :
Total Pages : 282 pages
Book Rating : 4.:/5 (163 download)
Download or read book Acoustic Model and Pronunciation Adaptation in Automatic Speech Recognition written by Yongxin Zhang and published by . This book was released on 2006 with total page 282 pages. Available in PDF, EPUB and Kindle. Book excerpt:
Author :
Publisher :
ISBN 13 :
Total Pages : 9 pages
Book Rating : 4.:/5 (228 download)
Download or read book Towards Multilingual Interoperability in Automatic Speech Recognition written by and published by . This book was released on 2000 with total page 9 pages. Available in PDF, EPUB and Kindle. Book excerpt: In this communication, we address multilingual interoperability aspects in speech recognition. After giving a tentative definition of multilingual interoperability, we discuss speech recognition components and their language-specific aspects. We give a sample overview of past multilingual speech recognition research and development across different speaking styles (read, prepared and conversational). The problem of adaptation to new languages is addressed. Language-independent and cross- language techniques for acoustic modeling provide a means to port recognition systems to new languages without language specific acoustic data. Pronunciation lexical and text material appear to be the most crucial language-dependent resources for porting. Fast porting being a step towards multilingual interoperability the ongoing efforts of producing multilingual pronun ciation lexical and collecting multilingual text corpora should be extended to the largest possible number of written languages.
Author : Christian Hacker
Publisher : Logos Verlag Berlin GmbH
ISBN 13 : 3832522581
Total Pages : 272 pages
Book Rating : 4.8/5 (325 download)
Download or read book Automatic Assessment of Children Speech to Support Language Learning written by Christian Hacker and published by Logos Verlag Berlin GmbH. This book was released on 2009 with total page 272 pages. Available in PDF, EPUB and Kindle. Book excerpt: Focus of this work are pattern recognition related aspects of computer assisted pronunciation training (CAPT) for second language learning. An overview of commercial systems shows that pronunciation training is being addressed by the growing field of computer assisted language learning only to a small extend, although in the state-of-the-art section a number of such approaches for automatic assessment can already be presented. In the present thesis different approaches are extended and combined. In particular a large set of nearly 200 pronunciation and prosodic features is developed. By this approach pronunciation scoring is regarded as classification task in high-dimensional feature space. Automatic speech recognition is the basis of most pronunciation scoring algorithms. In this thesis a system is presented, which supports second language learning at school, i.e. the target users are children. For this reason a state-of-the-art speech recognition engine is adapted to children speech, since young speakers are only hardly recognised by automatic systems. Phonetically motivated rules for typical mispronunciation errors are integrated into the system to make it suitable for pronunciation scoring. Evaluating an algorithm for pronunciation assessment is more difficult than simply counting the correctly recognised mistakes, since there exists no objective ground truth. This can be shown by evaluating the annotations of 14 teachers. However, with different measures it can be verified that the accuracy of the system (in comparison with teachers) thoroughly reaches the agreement among teachers. The evaluation is conducted with native German speakers learning English.
Author : Albert Ali Salah
Publisher : Springer
ISBN 13 : 3030260615
Total Pages : 580 pages
Book Rating : 4.0/5 (32 download)
Download or read book Speech and Computer written by Albert Ali Salah and published by Springer. This book was released on 2019-08-09 with total page 580 pages. Available in PDF, EPUB and Kindle. Book excerpt: This book constitutes the proceedings of the 21st International Conference on Speech and Computer, SPECOM 2019, held in Istanbul, Turkey, in August 2019. The 57 papers presented were carefully reviewed and selected from 86 submissions. The papers present current research in the area of computer speech processing including audio signal processing, automatic speech recognition, speaker recognition, computational paralinguistics, speech synthesis, sign language and multimodal processing, and speech and language resources.
Author : Smile Markovski
Publisher : Springer Science & Business Media
ISBN 13 : 3642371698
Total Pages : 385 pages
Book Rating : 4.6/5 (423 download)
Download or read book ICT Innovations 2012 written by Smile Markovski and published by Springer Science & Business Media. This book was released on 2013-03-26 with total page 385 pages. Available in PDF, EPUB and Kindle. Book excerpt: The present stage of the human civilization is the e-society, which is build over the achievements obtained by the development of the information and communication technologies. It affects everyone, from ordinary mobile phone users to designers of high quality industrial products, and every human activity, from taking medical care to improving the state governing. The science community working in computer sciences and informatics is therefore under constant challenge; it has to solve the new appeared theoretical problem as well as to find new practical solutions. The fourth ICT Innovations Conference, held in September 2012 in Ohrid, Macedonia, was one of the several world-wide forums where academics, professionals and practitioners presented their last scientific results and development applications in the fields of high performance and parallel computing, bioinformatics, human computer interaction, security and cryptography, computer and mobile networks, neural networks, cloud computing, process verification, improving medical care, improving quality of services, web technologies, hardware implementations, cultural implication. In this book the best 37 ranked articles are presented.
Author : Jean-Claude Junqua
Publisher : Springer Science & Business Media
ISBN 13 : 0306470276
Total Pages : 193 pages
Book Rating : 4.3/5 (64 download)
Download or read book Robust Speech Recognition in Embedded Systems and PC Applications written by Jean-Claude Junqua and published by Springer Science & Business Media. This book was released on 2006-04-18 with total page 193 pages. Available in PDF, EPUB and Kindle. Book excerpt: Robust Speech Recognition in Embedded Systems and PC Applications provides a link between the technology and the application worlds. As speech recognition technology is now good enough for a number of applications and the core technology is well established around hidden Markov models many of the differences between systems found in the field are related to implementation variants. We distinguish between embedded systems and PC-based applications. Embedded applications are usually cost sensitive and require very simple and optimized methods to be viable. Robust Speech Recognition in Embedded Systems and PC Applications reviews the problems of robust speech recognition, summarizes the current state of the art of robust speech recognition while providing some perspectives, and goes over the complementary technologies that are necessary to build an application, such as dialog and user interface technologies. Robust Speech Recognition in Embedded Systems and PC Applications is divided into five chapters. The first one reviews the main difficulties encountered in automatic speech recognition when the type of communication is unknown. The second chapter focuses on environment-independent/adaptive speech recognition approaches and on the mainstream methods applicable to noise robust speech recognition. The third chapter discusses several critical technologies that contribute to making an application usable. It also provides some design recommendations on how to design prompts, generate user feedback and develop speech user interfaces. The fourth chapter reviews several techniques that are particularly useful for embedded systems or to decrease computational complexity. It also presents some case studies for embedded applications and PC-based systems. Finally, the fifth chapter provides a future outlook for robust speech recognition, emphasizing the areas that the author sees as the most promising for the future. Robust Speech Recognition in Embedded Systems and PC Applications serves as a valuable reference and although not intended as a formal University textbook, contains some material that can be used for a course at the graduate or undergraduate level. It is a good complement for the book entitled Robustness in Automatic Speech Recognition: Fundamentals and Applications co-authored by the same author.
