Speech Recognition Software and Vidispine
Total Page:16
File Type:pdf, Size:1020Kb
Load more
Recommended publications
-
Speech Recognition in Java Breandan Considine Jetbrains, Inc
Speech Recognition in Java Breandan Considine JetBrains, Inc. Automatic speech recognition in 2011 Automatic speech recognition in 2015 What happened? • Bigger data • Faster hardware • Smarter algorithms Traditional ASR • Requires lots of handmade feature engineering • Poor results: >25% WER for HMM architectures State of the art ASR • <10% average word error on large datasets • DNNs: DBNs, CNNs, RBMs, LSTM • Thousands of hours of transcribed speech • Rapidly evolving field • Takes time (days) and energy (kWh) to train • Difficult to customize without prior experience Free / open source • Deep learning libraries • C/C++: Caffe, Kaldi • Python: Theano, Caffe • Lua: Torch • Java: dl4j, H2O • Open source datasets • LibriSpeech – 1000 hours of LibriVox audiobooks • Experience is required Let’s think… • What if speech recognition were perfect? • Models are still black boxes • ASR is just a fancy input method • How can ASR improve user productivity? • What are the user’s expectations? • Behavior is predictable/deterministic • Control interface is simple/obvious • Recognition is fast and accurate Why offline? • Latency – many applications need fast local recognition • Mobility – users do not always have an internet connection • Privacy – data is recorded and analyzed completely offline • Flexibility – configurable API, language, vocabulary, grammar Introduction • What techniques do modern ASR systems use? • How do I build a speech recognition application? • Is speech recognition accessible for developers? • What libraries and frameworks exist -
General Subtitling FAQ's
General Subtitling FAQ’s What's the difference between open and closed captions? Open captions are sometimes referred to as ‘burnt-in’ or ‘in-vision’ subtitles. They are generally encoded as a permanent part of the video image. Closed captions is a generic term for subtitles that are viewer-selectable and is generally used for subtitles that are transmitted as a discrete data stream in the VBI then decoded and displayed as text by the TV receiver e.g. Line 21 Closed Captioning system in the USA, and Teletext subtitles, using line 335, in Europe. What's the difference between "live" and "offline" subtitling? Live subtitling is the real-time captioning of live programmes, typically news and sports. This is achieved by using either Speech Recognition systems or Stenographic (Steno) keyboards. Offline subtitling is created by viewing previously recorded material. Typically this produces better results because the subtitle author can ensure they produce An accurate translation/transcription with no spelling mistakes Subtitles are timed to coincide precisely with the dialogue Subtitles are positioned to avoid obscuring other important onscreen features Which Speech Recognition packages do Starfish support? Starfish supports Speech Recognition packages from IBM and Dragon and other suppliers who confirm to the Microsoft Speech API. The choice of the software is usually dictated by the availability of a specific language. Starfish Technologies Ltd FAQ General FAQ’s What's the difference between subtitling and captioning? The use of these terms varies in different parts of the world. Subtitling often refers to the technique of using open captions for language translation of foreign programmes. -
Speech Recognition Technology for Disabilities Education
J. EDUCATIONAL TECHNOLOGY SYSTEMS, Vol. 33(2) 173-184, 2004-2005 SPEECH RECOGNITION TECHNOLOGY FOR DISABILITIES EDUCATION K. WENDY TANG GILBERT ENG RIDHA KAMOUA WEI CHERN CHU VICTOR SUTAN GUOFENG HOU OMER FAROOQ Stony Brook University, New York ABSTRACT Speech recognition is an alternative to traditional methods of interacting with a computer, such as textual input through a keyboard. An effective system can replace or reduce the reliability on standard keyboard and mouse input. This can especially assist dyslexic students who have problems with character or word use and manipulation in a textual form; and students with physical disabilities that affect their data entry or ability to read, and therefore to check, what they have entered. In this article, we summarize the current state of available speech recognition technologies and describe a student project that integrates speech recognition technologies with Personal Digital Assistants to provide a cost-effective and portable health monitoring system for people with disabilities. We are hopeful that this project may inspire more student-led projects for disabilities education. 1. INTRODUCTION Speech recognition allows “hands-free” control of various electronic devices. It is particularly advantageous to physically disabled persons. Speech recognition can be used in computers to control the system, launch applications, and create “print-ready” dictation and thus provide easy access to persons with hearing or 173 Ó 2004, Baywood Publishing Co., Inc. 174 / TANG ET AL. vision impairments. For example, a hearing impaired person can use a microphone to capture another’s speech and then use speech recognition technologies to convert the speech to text. -
Speech Recognition Systems: a Comparative Review
IOSR Journal of Computer Engineering (IOSR-JCE) e-ISSN: 2278-0661,p-ISSN: 2278-8727, Volume 19, Issue 5, Ver. IV (Sep.- Oct. 2017), PP 71-79 www.iosrjournals.org Speech Recognition Systems: A Comparative Review Rami Matarneh1, Svitlana Maksymova2, Vyacheslav V. Lyashenko3 , Nataliya V. Belova 3 1(Department of Computer Science, Prince Sattam Bin Abdulaziz University, Al-Kharj, Saudi Arabi) 2(Department of Computer-Integrated Technologies, Automation and Mechatronics, Kharkiv National University of RadioElectronics, Kharkiv, Ukraine) 3(Department of Informatics, Kharkiv National University of RadioElectronics, Kharkiv, Ukraine) Abstract: Creating voice control for robots is very important and difficult task. Therefore, we consider different systems of speech recognition. We divided them into two main classes: (1) open-source and (2) close-source code. As close-source software the following were selected: Dragon Mobile SDK, Google Speech Recognition API, Siri, Yandex SpeechKit and Microsoft Speech API. While the following were selected as open-source software: CMU Sphinx, Kaldi, Julius, HTK, iAtros, RWTH ASR and Simon. The comparison mainly based on accuracy, API, performance, speed in real-time, response time and compatibility. the variety of comparison axes allow us to make detailed description of the differences and similarities, which in turn enabled us to adopt a careful decision to choose the appropriate system depending on our need. Keywords: Robot, Speech recognition, Voice systems with closed source code, Voice systems with open source code --------------------------------------------------------------------------------------------------------------------------------------- Date of Submission: 13-10-2017 Date of acceptance: 27-10-2017 --------------------------------------------------------------------------------------------------------------------------------------- I. Introduction Voice control will make your application more convenient for user especially if a person works with it on the go or his hands are busy. -
Speech-To-Text System for Phonebook Automation
Speech-to-Text System for Phonebook Automation Thesis submitted in partial fulfillment of the requirements for the award of degree of Master of Engineering in Computer Science and Engineering Submitted By Nishant Allawadi (Roll No. 801032017) Under the supervision of: Mr. Parteek Bhatia Assistant Professor COMPUTER SCIENCE AND ENGINEERING DEPARTMENT THAPAR UNIVERSITY PATIALA – 147004 June 2012 i ii Abstract In the daily use of electronic devices, generally, the input is given by pressing keys or touching the screen. There are a lot of devices which give output in the form of speech but the way of giving input by speech is very new and it has been seen in a few of our latest devices. The speech input technology needs a Speech-to-Text system to convert input speech into its corresponding text and speech output technology needs a Text-to- Speech system to convert input text into its corresponding speech. In the work presented, a Speech-to-Text system has been developed with a Text-to- Speech module to enable speech as input as well as output. A phonebook application has also been developed to add new contacts, update previously saved contacts, delete saved contacts and call the saved contacts. This phonebook application has been integrated with Speech-to-Text system and Text-to-Speech module to convert the phonebook application into a complete speech interactive system. The first chapter, the basic introduction of the Speech-to-Text system has been presented. The second chapter discusses about the evolution of the Speech-to-Text technology along with the existing Speech-to-Text systems. -
Multipurpose Large Vocabulary Continuous Speech Recognition Engine Julius
Multipurpose Large Vocabulary Continuous Speech Recognition Engine Julius rev. 3.2 (2001/12/03) Copyright (c) 1997-2000 Information-Technology Promotion Agency, Japan Copyright (c) 1991-2001 Kyoto University Copyright (c) 2000-2001 Nara Institute of Science and Technology All rights reserved. Translated from the original Julius-3.2-book by Ian Lane – Kyoto University Julius Julius is a high performance continuous speech recognition software based on word N-grams. It is able to perform recognition at the sentence level with a vocabulary in the tens of thousands. Julius realizes high-speed speech recognition on a typical desktop PC. It performs at near real time and has a recognition rate of above 90% for a 20,000-word vocabulary dictation task. The best feature of the Julius system is that it is multipurpose. By recombining the pronunciation dictionary, language and acoustic models one is able to build various task specific systems. The Julius code also is open source so one should be able to recompile the system for other platforms or alter the code for one's specific needs. Platforms currently supported include Linux, Solaris and other versions of Unix, and Windows. There are two Windows versions, a Microsoft SAPI 5.0 compatible version and a Windows DLL version. This documentation relates to the Unix version of Julius. For documentation on the Windows versions see the Julius for SAPI README (CSRC CD-ROM) (Japanese), or the Julius for SAPI homepage (Kyoto University) (Japanese). 1 Contacts/Links Contacts/Links Home Page: http://winnie.kuis.kyoto-u.ac.jp/pub/julius/ (Japanese) E-mail: [email protected] Developers Original System/Unix version: Akinobu Lee ([email protected]) Windows Microsoft SAPI version: Takashi Sumiyoshi Windows (DLL version): Hideki Banno 2 System Structure and Features System Structure The structure of the Julius speech recognition system is shown in the diagram below: N-gram language models and a HMM acoustic model are used. -
Multimodal HALEF: an Open-Source Modular Web-Based Multimodal Dialog Framework
Multimodal HALEF: An Open-Source Modular Web-Based Multimodal Dialog Framework Zhou Yu‡†, Vikram Ramanarayanan†, Robert Mundkowsky†, Patrick Lange†, Alexei Ivanov†, Alan W Black‡ and David Suendermann-Oeft† Abstract We present an open-source web-based multimodal dialog framework, “Multimodal HALEF”, that integrates video conferencing and telephony abilities into the existing HALEF cloud-based dialog framework via the FreeSWITCH video telephony server. Due to its distributed and cloud-based architecture, Multimodal HALEF allows researchers to collect video and speech data from participants in- teracting with the dialog system outside of traditional lab settings, therefore largely reducing cost and labor incurred during the traditional audio-visual data collection process. The framework is equipped with a set of tools including a web-based user survey template, a speech transcription, annotation and rating portal, a web visual processing server that performs head tracking, and a database that logs full-call audio and video recordings as well as other call-specific information. We present observations from an initial data collection based on an job interview application. Finally we report on some future plans for development of the framework. 1 Introduction and Related Work Previously, many end-to-end spoken dialog systems (SDSs) used close-talk micro- phones or handheld telephones to gather speech input [3] [22] in order to improve automatic speech recognition (ASR) performance of the system. However, this lim- its the accessibility of the system. Recently, the performance of ASR systems has improved drastically even in noisy conditions [5]. In turn, spoken dialog systems are now becoming increasingly deployable in open spaces [2]. -
The Wavesurfer Automatic Speech Recognition Plugin
The WaveSurfer Automatic Speech Recognition Plugin Giampiero Salvi and Niklas Vanhainen KTH, School of Computer Science and Communication, Department of Speech Music and Hearing, Stockholm, Sweden fgiampi, [email protected] Abstract This paper presents a plugin that adds automatic speech recognition (ASR) functionality to the WaveSurfer sound manipulation and visualisation program. The plugin allows the user to run continuous speech recognition on spoken utterances, or to align an already available orthographic transcription to the spoken material. The plugin is distributed as free software and is based on free resources, namely the Julius speech recognition engine and a number of freely available ASR resources for different languages. Among these are the acoustic and language models we have created for Swedish using the NST database. Keywords: Automatic Speech Recognition, Free Software, WaveSurfer 1. Introduction Automatic Speech Recognition (ASR) is becoming an im- portant part of our lives, both as a viable alternative for humans-computer interaction, but also as a tool for linguis- tics and speech research. In many cases, however, it is trou- blesome, even in the language and speech communities, to have easy access to ASR resources. On the one hand, com- mercial systems are often too expensive and not flexible enough for researchers. On the other hand, free ASR soft- ware often lacks high quality resources such as acoustic and language models for the specific languages and requires ex- pertise that linguists and speech researchers cannot afford. In this paper we describe a plugin for the popular sound manipulation and visualisation program WaveSurfer1 (Sjolander¨ and Beskow, 2000) that attempts to solve the above problems. -
Systematic Review on Speech Recognition Tools and Techniques Needed for Speech Application Development
INTERNATIONAL JOURNAL OF SCIENTIFIC & TECHNOLOGY RESEARCH VOLUME 9, ISSUE 03, MARCH 2020 ISSN 2277-8616 Systematic Review On Speech Recognition Tools And Techniques Needed For Speech Application Development Lydia K. Ajayi, Ambrose A. Azeta, Isaac. A. Odun-Ayo, Felix.C. Chidozie, Aeeigbe. E. Azeta Abstract: Speech has been widely known as the primary mode of communication among individuals and computers. The existence of technology brought about human computer interface to allow human computer interaction. Existence of speech recognition research has been ongoing for the past 60 years which has been embraced as an alternative access method for individuals with disabilities, learning shortcomings, Navigation problems etc. and at the same time allow computer to identify human languages by converting these languages into strings of words or commands. The ongoing problem in speech recognition especially for under-resourced languages has been identified to be background noise, speed, recognition accuracy, etc. In this paper, in order to achieve a higher level of speed, recognition accuracy (word or sentence accuracy), manage background noise, etc. we have to be able to identify and apply the right speech recognition algorithms/techniques/parameters/tools needed ultimately to develop any speech recognition system. The primary objective of this paper is to make summarization and comparison of some of the well-known methods/algorithms/techniques/parameters identifying their steps, strengths, weaknesses, and application areas in various stages of speech recognition systems as well as for the effectiveness of future services and applications. Index Terms: Speech recognition, Speech recognition tools, Speech recognition techniques, Isolated speech recognition, Continuous speech recognition, Under-resourced languages, High resourced languages. -
Automated Regression Testing Approach to Expansion and Refinement of Speech Recognition Grammars
University of Central Florida STARS Electronic Theses and Dissertations, 2004-2019 2008 Automated Regression Testing Approach To Expansion And Refinement Of Speech Recognition Grammars Raul Dookhoo University of Central Florida Part of the Computer Sciences Commons, and the Engineering Commons Find similar works at: https://stars.library.ucf.edu/etd University of Central Florida Libraries http://library.ucf.edu This Masters Thesis (Open Access) is brought to you for free and open access by STARS. It has been accepted for inclusion in Electronic Theses and Dissertations, 2004-2019 by an authorized administrator of STARS. For more information, please contact [email protected]. STARS Citation Dookhoo, Raul, "Automated Regression Testing Approach To Expansion And Refinement Of Speech Recognition Grammars" (2008). Electronic Theses and Dissertations, 2004-2019. 3503. https://stars.library.ucf.edu/etd/3503 AUTOMATED REGRESSION TESTING APPROACH TO EXPANSION AND REFINEMENT OF SPEECH RECOGNITION GRAMMARS by RAUL AVINASH DOOKHOO B.S. University of Guyana, 2004 A thesis submitted in partial fulfillment of the requirements for the degree of Master of Science in the School of Electrical Engineering and Computer Science in the College of Engineering and Computer Science at the University of Central Florida Orlando, Florida Fall Term 2008 © 2008 Raul Avinash Dookhoo ii ABSTRACT This thesis describes an approach to automated regression testing for speech recognition grammars. A prototype Audio Regression Tester called ART has been developed using Microsoft’s Speech API and C#. ART allows a user to perform any of three tasks: automatically generate a new XML-based grammar file from standardized SQL database entries, record and cross-reference audio files for use by an underlying speech recognition engine, and perform regression tests with the aid of an oracle grammar. -
Speech@Home: an Exploratory Study
Speech@Home: An Exploratory Study A.J. Brush Paul Johns Abstract Microsoft Research Microsoft Research To understand how people might use a speech dialog 1 Microsoft Way 1 Microsoft Way system in the public areas of their homes, we Redmond, WA 98052 USA Redmond, WA 98052 USA conducted an exploratory field study in six households. [email protected] [email protected] For two weeks each household used a system that logged motion and usage data, recorded speech diary Kori Inkpen Brian Meyers entries and used Experience Sampling Methodology Microsoft Research Microsoft Research (ESM) to prompt participants for additional examples of 1 Microsoft Way 1 Microsoft Way speech commands. The results demonstrated our Redmond, WA 98052 USA Redmond, WA 98052 USA participants’ interest in speech interaction at home, in [email protected] [email protected] particular for web browsing, calendaring and email tasks, although there are still many technical challenges that need to be overcome. More generally, our study suggests the value of using speech to enable a wide range of interactions. Keywords Speech, home technology, diary study, domestic ACM Classification Keywords H.5.2 User Interfaces: Voice I/O. General Terms Human Factors, Experimentation Copyright is held by the author/owner(s). Introduction CHI 2011, May 7–12, 2011, Vancouver, BC, Canada. The potential for using speech in home environments ACM 978-1-4503-0268-5/11/05. has long been recognized. Smart home research has suggested using speech dialog systems for controlling . home infrastructure (e.g. [5, 8, 10, 12]), and a variety We conducted a two week exploratory speech diary of novel home applications use speech interfaces, such study in six households to understand whether or not as cooking support systems [2] and the Nursebot speech interaction was desirable, and, if so, what types personal robotic assistant for the elderly [14]. -
Speech Phonetization Alignment and Syllabification (SPPAS): a Tool for the Automatic Analysis of Speech Prosody Brigitte Bigi, Daniel Hirst
SPeech Phonetization Alignment and Syllabification (SPPAS): a tool for the automatic analysis of speech prosody Brigitte Bigi, Daniel Hirst To cite this version: Brigitte Bigi, Daniel Hirst. SPeech Phonetization Alignment and Syllabification (SPPAS): a tool for the automatic analysis of speech prosody. Speech Prosody, May 2012, Shanghai, China. pp.19-22. hal-00983699 HAL Id: hal-00983699 https://hal.archives-ouvertes.fr/hal-00983699 Submitted on 25 Apr 2014 HAL is a multi-disciplinary open access L’archive ouverte pluridisciplinaire HAL, est archive for the deposit and dissemination of sci- destinée au dépôt et à la diffusion de documents entific research documents, whether they are pub- scientifiques de niveau recherche, publiés ou non, lished or not. The documents may come from émanant des établissements d’enseignement et de teaching and research institutions in France or recherche français ou étrangers, des laboratoires abroad, or from public or private research centers. publics ou privés. SPeech Phonetization Alignment and Syllabification (SPPAS): a tool for the automatic analysis of speech prosody. Brigitte Bigi, Daniel Hirst Laboratoire Parole et Langage, CNRS & Aix-Marseille Université [email protected], [email protected] Abstract length, pitch and loudness of the individual sounds which make up an utterance”. [7] SPASS, SPeech Phonetization Alignment and Syllabifi- cation, is a tool to automatically produce annotations Linguists need tools for the automatic analysis of at which include utterance, word, syllable and phoneme seg- least these three prosodic features of speech sounds. In mentations from a recorded speech sound and its tran- this paper we concentrate on the tool for the automatic scription.