Models of Speech Synthesis ROLF CARLSON Department of Speech Communication and Music Acoustics, Royal Institute of Technology, S-100 44 Stockholm, Sweden
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Speech Signal Basics
Speech Signal Basics Nimrod Peleg Updated: Feb. 2010 Course Objectives • To get familiar with: – Speech coding in general – Speech coding for communication (military, cellular) • Means: – Introduction the basics of speech processing – Presenting an overview of speech production and hearing systems – Focusing on speech coding: LPC codecs: • Basic principles • Discussing of related standards • Listening and reviewing different codecs. הנחיות לשימוש נכון בטלפון, פלשתינה-א"י, 1925 What is Speech ??? • Meaning: Text, Identity, Punctuation, Emotion –prosody: rhythm, pitch, intensity • Statistics: sampled speech is a string of numbers – Distribution (Gamma, Laplace) Quantization • Information Theory: statistical redundancy – ZIP, ARJ, compress, Shorten, Entropy coding… ...1001100010100110010... • Perceptual redundancy – Temporal masking , frequency masking (mp3, Dolby) • Speech production Model – Physical system: Linear Prediction The Speech Signal • Created at the Vocal cords, Travels through the Vocal tract, and Produced at speakers mouth • Gets to the listeners ear as a pressure wave • Non-Stationary, but can be divided to sound segments which have some common acoustic properties for a short time interval • Two Major classes: Vowels and Consonants Speech Production • A sound source excites a (vocal tract) filter – Voiced: Periodic source, created by vocal cords – UnVoiced: Aperiodic and noisy source •The Pitch is the fundamental frequency of the vocal cords vibration (also called F0) followed by 4-5 Formants (F1 -F5) at higher frequencies. -
The Role of Higher-Level Linguistic Features in HMM-Based Speech Synthesis
INTERSPEECH 2010 The role of higher-level linguistic features in HMM-based speech synthesis Oliver Watts, Junichi Yamagishi, Simon King Centre for Speech Technology Research, University of Edinburgh, UK [email protected] [email protected] [email protected] Abstract the annotation of a synthesiser’s training data on listeners’ re- action to the speech produced by that synthesiser is still unclear We analyse the contribution of higher-level elements of the lin- to us. In particular, we suspect that features such as pitch ac- guistic specification of a data-driven speech synthesiser to the cent type which might be useful in voice-building if labelled naturalness of the synthetic speech which it generates. The reliably, are under-used in conventional systems because of la- system is trained using various subsets of the full feature-set, belling/prediction errors. For building the systems to be evalu- in which features relating to syntactic category, intonational ated we therefore used data for which hand labelled annotation phrase boundary, pitch accent and boundary tones are selec- of ToBI events is available. This allows us to compare ideal sys- tively removed. Utterances synthesised by the different config- tems built from a corpus where these higher-level features are urations of the system are then compared in a subjective evalu- accurately annotated with more conventional systems that rely ation of their naturalness. exclusively on prediction of these features from text for annota- The work presented forms background analysis for an on- tion of their training data. going set of experiments in performing text-to-speech (TTS) conversion based on shallow features: features that can be triv- 2. -
Benchmarking Audio Signal Representation Techniques for Classification with Convolutional Neural Networks
sensors Article Benchmarking Audio Signal Representation Techniques for Classification with Convolutional Neural Networks Roneel V. Sharan * , Hao Xiong and Shlomo Berkovsky Australian Institute of Health Innovation, Macquarie University, Sydney, NSW 2109, Australia; [email protected] (H.X.); [email protected] (S.B.) * Correspondence: [email protected] Abstract: Audio signal classification finds various applications in detecting and monitoring health conditions in healthcare. Convolutional neural networks (CNN) have produced state-of-the-art results in image classification and are being increasingly used in other tasks, including signal classification. However, audio signal classification using CNN presents various challenges. In image classification tasks, raw images of equal dimensions can be used as a direct input to CNN. Raw time- domain signals, on the other hand, can be of varying dimensions. In addition, the temporal signal often has to be transformed to frequency-domain to reveal unique spectral characteristics, therefore requiring signal transformation. In this work, we overview and benchmark various audio signal representation techniques for classification using CNN, including approaches that deal with signals of different lengths and combine multiple representations to improve the classification accuracy. Hence, this work surfaces important empirical evidence that may guide future works deploying CNN for audio signal classification purposes. Citation: Sharan, R.V.; Xiong, H.; Berkovsky, S. Benchmarking Audio Keywords: convolutional neural networks; fusion; interpolation; machine learning; spectrogram; Signal Representation Techniques for time-frequency image Classification with Convolutional Neural Networks. Sensors 2021, 21, 3434. https://doi.org/10.3390/ s21103434 1. Introduction Sensing technologies find applications in detecting and monitoring health conditions. -
Designing Speech and Multimodal Interactions for Mobile, Wearable, and Pervasive Applications
Designing Speech and Multimodal Interactions for Mobile, Wearable, and Pervasive Applications Cosmin Munteanu Nikhil Sharma Abstract University of Toronto Google, Inc. Traditional interfaces are continuously being replaced Mississauga [email protected] by mobile, wearable, or pervasive interfaces. Yet when [email protected] Frank Rudzicz it comes to the input and output modalities enabling Pourang Irani Toronto Rehabilitation Institute our interactions, we have yet to fully embrace some of University of Manitoba University of Toronto the most natural forms of communication and [email protected] [email protected] information processing that humans possess: speech, Sharon Oviatt Randy Gomez language, gestures, thoughts. Very little HCI attention Incaa Designs Honda Research Institute has been dedicated to designing and developing spoken [email protected] [email protected] language and multimodal interaction techniques, Matthew Aylett Keisuke Nakamura especially for mobile and wearable devices. In addition CereProc Honda Research Institute to the enormous, recent engineering progress in [email protected] [email protected] processing such modalities, there is now sufficient Gerald Penn Kazuhiro Nakadai evidence that many real-life applications do not require University of Toronto Honda Research Institute 100% accuracy of processing multimodal input to be [email protected] [email protected] useful, particularly if such modalities complement each Shimei Pan other. This multidisciplinary, two-day workshop -
The RACAI Text-To-Speech Synthesis System
The RACAI Text-to-Speech Synthesis System Tiberiu Boroș, Radu Ion, Ștefan Daniel Dumitrescu Research Institute for Artificial Intelligence “Mihai Drăgănescu”, Romanian Academy (RACAI) [email protected], [email protected], [email protected] of standalone Natural Language Processing (NLP) Tools Abstract aimed at enabling text-to-speech (TTS) synthesis for less- This paper describes the RACAI Text-to-Speech (TTS) entry resourced languages. A good example is the case of for the Blizzard Challenge 2013. The development of the Romanian, a language which poses a lot of challenges for TTS RACAI TTS started during the Metanet4U project and the synthesis mainly because of its rich morphology and its system is currently part of the METASHARE platform. This reduced support in terms of freely available resources. Initially paper describes the work carried out for preparing the RACAI all the tools were standalone, but their design allowed their entry during the Blizzard Challenge 2013 and provides a integration into a single package and by adding a unit selection detailed description of our system and future development speech synthesis module based on the Pitch Synchronous directions. Overlap-Add (PSOLA) algorithm we were able to create a fully independent text-to-speech synthesis system that we refer Index Terms: speech synthesis, unit selection, concatenative to as RACAI TTS. 1. Introduction A considerable progress has been made since the start of the project, but our system still requires development and Text-to-speech (TTS) synthesis is a complex process that although considered premature, our participation in the addresses the task of converting arbitrary text into voice. -
Synthesis and Recognition of Speech Creating and Listening to Speech
ISSN 1883-1974 (Print) ISSN 1884-0787 (Online) National Institute of Informatics News NII Interview 51 A Combination of Speech Synthesis and Speech Oct. 2014 Recognition Creates an Affluent Society NII Special 1 “Statistical Speech Synthesis” Technology with a Rapidly Growing Application Area NII Special 2 Finding Practical Application for Speech Recognition Feature Synthesis and Recognition of Speech Creating and Listening to Speech A digital book version of “NII Today” is now available. http://www.nii.ac.jp/about/publication/today/ This English language edition NII Today corresponds to No. 65 of the Japanese edition [Advance Notice] Great news! NII Interview Yamagishi-sensei will create my voice! Yamagishi One result is a speech translation sys- sound. Bit (NII Character) A Combination of Speech Synthesis tem. This system recognizes speech and translates it Ohkawara I would like your comments on the fu- using machine translation to synthesize speech, also ture challenges. and Speech Recognition automatically translating it into every language to Ono The challenge for speech recognition is how speak. Moreover, the speech is created with a human close it will come to humans in the distant speech A Word from the Interviewer voice. In second language learning, you can under- case. If this study is advanced, it will be possible to Creates an Affluent Society stand how you should pronounce it with your own summarize the contents of a meeting and to automati- voice. If this is further advanced, the system could cally take the minutes. If a robot understands the con- have an actor in a movie speak in a different language tents of conversations by multiple people in a natural More and more people have begun to use smart- a smartphone, I use speech input more often. -
Commercial Tools in Speech Synthesis Technology
International Journal of Research in Engineering, Science and Management 320 Volume-2, Issue-12, December-2019 www.ijresm.com | ISSN (Online): 2581-5792 Commercial Tools in Speech Synthesis Technology D. Nagaraju1, R. J. Ramasree2, K. Kishore3, K. Vamsi Krishna4, R. Sujana5 1Associate Professor, Dept. of Computer Science, Audisankara College of Engg. and Technology, Gudur, India 2Professor, Dept. of Computer Science, Rastriya Sanskrit VidyaPeet, Tirupati, India 3,4,5UG Student, Dept. of Computer Science, Audisankara College of Engg. and Technology, Gudur, India Abstract: This is a study paper planned to a new system phonetic and prosodic information. These two phases are emotional speech system for Telugu (ESST). The main objective of usually called as high- and low-level synthesis. The input text this paper is to map the situation of today's speech synthesis might be for example data from a word processor, standard technology and to focus on potential methods for the future. ASCII from e-mail, a mobile text-message, or scanned text Usually literature and articles in the area are focused on a single method or single synthesizer or the very limited range of the from a newspaper. The character string is then preprocessed and technology. In this paper the whole speech synthesis area with as analyzed into phonetic representation which is usually a string many methods, techniques, applications, and products as possible of phonemes with some additional information for correct is under investigation. Unfortunately, this leads to a situation intonation, duration, and stress. Speech sound is finally where in some cases very detailed information may not be given generated with the low-level synthesizer by the information here, but may be found in given references. -
The Role of Speech Processing in Human-Computer Intelligent Communication
THE ROLE OF SPEECH PROCESSING IN HUMAN-COMPUTER INTELLIGENT COMMUNICATION Candace Kamm, Marilyn Walker and Lawrence Rabiner Speech and Image Processing Services Research Laboratory AT&T Labs-Research, Florham Park, NJ 07932 Abstract: We are currently in the midst of a revolution in communications that promises to provide ubiquitous access to multimedia communication services. In order to succeed, this revolution demands seamless, easy-to-use, high quality interfaces to support broadband communication between people and machines. In this paper we argue that spoken language interfaces (SLIs) are essential to making this vision a reality. We discuss potential applications of SLIs, the technologies underlying them, the principles we have developed for designing them, and key areas for future research in both spoken language processing and human computer interfaces. 1. Introduction Around the turn of the twentieth century, it became clear to key people in the Bell System that the concept of Universal Service was rapidly becoming technologically feasible, i.e., the dream of automatically connecting any telephone user to any other telephone user, without the need for operator assistance, became the vision for the future of telecommunications. Of course a number of very hard technical problems had to be solved before the vision could become reality, but by the end of 1915 the first automatic transcontinental telephone call was successfully completed, and within a very few years the dream of Universal Service became a reality in the United States. We are now in the midst of another revolution in communications, one which holds the promise of providing ubiquitous service in multimedia communications. -
Introduction to Digital Speech Processing Schafer Rabiner and Ronald W
SIGv1n1-2.qxd 11/20/2007 3:02 PM Page 1 FnT SIG 1:1-2 Foundations and Trends® in Signal Processing Introduction to Digital Speech Processing Processing to Digital Speech Introduction R. Lawrence W. Rabiner and Ronald Schafer 1:1-2 (2007) Lawrence R. Rabiner and Ronald W. Schafer Introduction to Digital Speech Processing highlights the central role of DSP techniques in modern speech communication research and applications. It presents a comprehensive overview of digital speech processing that ranges from the basic nature of the speech signal, through a variety of methods of representing speech in digital form, to applications in voice communication and automatic synthesis and recognition of speech. Introduction to Digital Introduction to Digital Speech Processing provides the reader with a practical introduction to Speech Processing the wide range of important concepts that comprise the field of digital speech processing. It serves as an invaluable reference for students embarking on speech research as well as the experienced researcher already working in the field, who can utilize the book as a Lawrence R. Rabiner and Ronald W. Schafer reference guide. This book is originally published as Foundations and Trends® in Signal Processing Volume 1 Issue 1-2 (2007), ISSN: 1932-8346. now now the essence of knowledge Introduction to Digital Speech Processing Introduction to Digital Speech Processing Lawrence R. Rabiner Rutgers University and University of California Santa Barbara USA [email protected] Ronald W. Schafer Hewlett-Packard Laboratories Palo Alto, CA USA Boston – Delft Foundations and Trends R in Signal Processing Published, sold and distributed by: now Publishers Inc. -
Voice User Interface on the Web Human Computer Interaction Fulvio Corno, Luigi De Russis Academic Year 2019/2020 How to Create a VUI on the Web?
Voice User Interface On The Web Human Computer Interaction Fulvio Corno, Luigi De Russis Academic Year 2019/2020 How to create a VUI on the Web? § Three (main) steps, typically: o Speech Recognition o Text manipulation (e.g., Natural Language Processing) o Speech Synthesis § We are going to start from a simple application to reach a quite complex scenario o by using HTML5, JS, and PHP § Reminder: we are interested in creating an interactive prototype, at the end 2 Human Computer Interaction Weather Web App A VUI for "chatting" about the weather Base implementation at https://github.com/polito-hci-2019/vui-example 3 Human Computer Interaction Speech Recognition and Synthesis § Web Speech API o currently a draft, experimental, unofficial HTML5 API (!) o https://wicg.github.io/speech-api/ § Covers both speech recognition and synthesis o different degrees of support by browsers 4 Human Computer Interaction Web Speech API: Speech Recognition § Accessed via the SpeechRecognition interface o provides the ability to recogniZe voice from an audio input o normally via the device's default speech recognition service § Generally, the interface's constructor is used to create a new SpeechRecognition object § The SpeechGrammar interface can be used to represent a particular set of grammar that your app should recogniZe o Grammar is defined using JSpeech Grammar Format (JSGF) 5 Human Computer Interaction Speech Recognition: A Minimal Example const recognition = new window.SpeechRecognition(); recognition.onresult = (event) => { const speechToText = event.results[0][0].transcript; -
A Multimodal User Interface for an Assistive Robotic Shopping Cart
electronics Article A Multimodal User Interface for an Assistive Robotic Shopping Cart Dmitry Ryumin 1 , Ildar Kagirov 1,* , Alexandr Axyonov 1, Nikita Pavlyuk 1, Anton Saveliev 1, Irina Kipyatkova 1, Milos Zelezny 2, Iosif Mporas 3 and Alexey Karpov 1,* 1 St. Petersburg Institute for Informatics and Automation of the Russian Academy of Sciences (SPIIRAS), St. Petersburg Federal Research Center of the Russian Academy of Sciences (SPC RAS), 199178 St. Petersburg, Russia; [email protected] (D.R.); [email protected] (A.A.); [email protected] (N.P.); [email protected] (A.S.); [email protected] (I.K.) 2 Department of Cybernetics, Faculty of Applied Sciences, University of West Bohemia, 301 00 Pilsen, Czech Republic; [email protected] 3 School of Engineering and Computer Science, University of Hertfordshire, Hatfield, Herts AL10 9AB, UK; [email protected] * Correspondence: [email protected] (I.K.); [email protected] (A.K.) Received: 2 November 2020; Accepted: 5 December 2020; Published: 8 December 2020 Abstract: This paper presents the research and development of the prototype of the assistive mobile information robot (AMIR). The main features of the presented prototype are voice and gesture-based interfaces with Russian speech and sign language recognition and synthesis techniques and a high degree of robot autonomy. AMIR prototype’s aim is to be used as a robotic cart for shopping in grocery stores and/or supermarkets. Among the main topics covered in this paper are the presentation of the interface (three modalities), the single-handed gesture recognition system (based on a collected database of Russian sign language elements), as well as the technical description of the robotic platform (architecture, navigation algorithm). -
Audio Processing Laboratory
AC 2010-1594: A GRADUATE LEVEL COURSE: AUDIO PROCESSING LABORATORY Buket Barkana, University of Bridgeport Page 15.35.1 Page © American Society for Engineering Education, 2010 A Graduate Level Course: Audio Processing Laboratory Abstract Audio signal processing is a part of the digital signal processing (DSP) field in science and engineering that has developed rapidly over the past years. Expertise in audio signal processing - including speech signal processing- is becoming increasingly important for working engineers from diverse disciplines. Audio signals are stored, processed, and transmitted using digital techniques. Creating these technologies requires engineers that understand real-time application issues in the related areas. With this motivation, we designed a graduate level laboratory course which is Audio Processing Laboratory in the electrical engineering department in our school two years ago. This paper presents the status of the audio processing laboratory course in our school. We report the challenges that we have faced during the development of this course in our school and discuss the instructor’s and students’ assessments and recommendations in this real-time signal-processing laboratory course. 1. Introduction Many DSP laboratory courses are developed for undergraduate and graduate level engineering education. These courses mostly focus on the general signal processing techniques such as quantization, filter design, Fast Fourier Transformation (FFT), and spectral analysis [1-3]. Because of the increasing popularity of Web-based education and the advancements in streaming media applications, several web-based DSP Laboratory courses have been designed for distance education [4-8]. An internet-based signal processing laboratory that provides hands-on learning experiences in distributed learning environments has been developed by Spanias et.al[6] .