COMP.SPEECH FAQ POSTING - PART 1/3 [Note: this document has been automatically extracted from a WWW site: http://www.speech.su.oz.au/comp.speech/ This may introduce some formatting errors.] Comp.Speech Frequently Asked Questions The Frequently Asked Questions (FAQ) is a regular posting to comp.speech which attempts to answer some of the regular questions in the comp.speech newsgroup. It covers speech synthesis, speech recognition, speech coding and a range of related material. It contains lists of speech technology software and hardware, including commerical products, public domain and freeware software, plus it contains over 500 links to speech technology sites and software. The FAQ is not meant to discuss any topic exhaustively. It will hopefully provide readers with pointers on where to find useful information, especially material available on the Internet. If you have not already read the Usenet introductory material posted to news.announce.newusers, please do. For help with FTP (file transfer protocol) look for a regular posting of anonymous FTP FAQ in comp.misc, comp.archives.admin or news.answers. This FAQ is posted every 4 weeks to comp.speech, comp.answers and news.answers. It is also available on the World Wide Web: * Australia: http://www.speech.su.oz.au/comp.speech/ * Britain: http://svr-www.eng.cam.ac.uk/comp.speech/ * Japan: http://www.itl.atr.co.jp/comp.speech/ * USA: http://www.speech.cs.cmu.edu/comp.speech/ Or by anonymous ftp from the comp.speech archive site: * ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/FAQ-complete Or from the news.answers ftp site (and its mirrors): * ftp://rtfm.mit.edu/pub/usenet/comp.speech/* Or by sending email to mail-server@rtfm.mit.edu with the following line in the body of the message: * send usenet/news.answers/comp-speech-faq/* If you only have email access to the internet, then I suggest you obtain the Internet-by-email guide. Send email to mail-server@rtfm.mit.edu with the following line in the body of the message: * send usenet/news.answers/internet-services/access-via-email Admin Minor changes each month. Thanks to all the companies and individuals who send in information. Acknowledgements Hundreds of people and companies have made contributions to the comp.speech FAQ over the last few years - too many to name individually. Special thanks go to Tony Robinson and Kevin Lenzo who have provided a wide range of information and assistance. Tony Robinson also maintains the comp.speech ftp site which is an excellent resource for all people working with speech technology. I am grateful to the people at Sydney University, Cambridge University, ATR ITL and CMU for supporting the FAQ on their WWW sites. Disclaimer The comp.speech FAQ and WWW pages are provided as is without any express or implied warranties. While every effort has been taken to ensure the accuracy of the information presented here, the author assumes no responsibility for errors or omissions, or for damages resulting from the use of the information contained herein. The comp.speech FAQ and WWW pages should not be construed as representing the views or products of my employer, Sun Microsystems, Inc. Copyright and Reproduction Copyright (c) 1994-6 by Andrew Hunt, all rights reserved. The comp.speech FAQ posting may not be distributed for financial gain. The comp.speech FAQ posting may not be included in any collections or compilations without express permission from the author. The comp.speech FAQ posting may be posted to any USENET newsgroup, on-line service, or BBS as long as it is posted in its entirety with this copyright statement, and that a current version is always maintained. [Note: hyperlinks to the comp.speech WWW pages are encouraged.] Maintainer The FAQ posting and the Comp.Speech WWW Site are maintained on a volunteer basis by Andrew Hunt Speech Applications Group, Sun Microsystems Laboratories Two Elizabeth Drive, Chelmsford, MA, 01824-4195, USA Ph: (508) 442 2681 Fax: (508) 250 5067 andrew.hunt@east.sun.com ___________________________________________________________________________ comp.speech FAQ Table of Contents + SpeechLinks: Speech Technology Hyperlinks Pages * SpeechLinks: 500+ Speech Technology Links * SpeechLinks: General Speech Technology Links * SpeechLinks: Signal Processing for Speech * SpeechLinks: Speech Coding * SpeechLinks: Speech Synthesis * SpeechLinks: Speech Recognition + List Of Software/Hardware + Update Times + Availability + Odds 'n Ends + FAQ Section 1: General Information on Speech Technology * SpeechLinks: General * Q1.1: What is comp.speech? * Q1.2: comp.speech ftp site * Q1.3: Common abbreviations and jargon * Q1.4: Related newsgroups and mailing lists * Q1.5: Associations, publications and conferences * Q1.6: Handicap Aids * Q1.7: Speech Databases * Q1.8: Speech File Formats and Conversion * Q1.9: Speech Laboratory Environments and Audio Editors * Q1.10: Speech Research Sites * Q1.11: Miscellaneous Software and Resources + FAQ Section 2: Signal Processing * SpeechLinks: Signal Processing for Speech * Q2.1: What sampling do I need for speech? * Q2.2: Finding the pitch of a speech signal * Q2.3: How do I find the start and end points of a speech signal? * Q2.4: Where can I find FFT software? * Q2.5: Signal processing in speech technology * Q2.6: Speech sampling and signal processing hardware * Q2.7: How do I convert to/from mu-law format? * Q2.8: Signal Processing Software + FAQ Section 3: Speech Coding and Compression * SpeechLinks: Speech Coding * Q3.1: Speech compression techniques * Q3.2: Information on speech coding and compression * Q3.3: Speech Compression / Coding Software + FAQ Section 4: Natural Language Processing * Q4.1: NLP References and Books * Q4.2: NLP Software + FAQ Section 5: Speech Synthesis * SpeechLinks: Speech Synthesis * Q5.1: What is speech synthesis? * Q5.2: How can speech synthesis be performed? * Q5.3: References/Books on Synthesis * Q5.4: Speech Synthesis on the WWW * Q5.5: Speech Synthesis Software/Hardware + FAQ Section 6: Speech Recognition * SpeechLinks: Speech Recognition * Q6.1: What is speech recognition? * Q6.2: How is speech recognition performed? * Q6.3: How can I build a simple speech recogniser? * Q6.4: References & books on speech recognition * Q6.5: Speech Recognition Hardware/Software * Q6.6: Speaker Recognition (Verification and Identification) * Q6.7: Integrated Speech Products ___________________________________________________________________________ List of Software/Hardware/Information The comp.speech FAQ provides information on a range of software, hardware and resources. Q1.6: Handicap Aids * Man-Machine Interfacing * SpeechViewer II Q1.7: Speech Data * Bavarian Archive for Speech Signals * BUPT Spoken Digit Database (Chinese) * Center for Spoken Language Understanding (CSLU) * Examples of IPA Symbols * Linguistic Data Consortium (LDC) * NOISEX * Oxford Acoustic Phonetic Database * Phonemic Samples * RELATOR project * ShATR * University of Victoria Phonetic Database Q1.9: Speech Processing Environments * CSRE: Computerized Speech Research Environment * DADiSP from DSP Development Corporation * Entropic Signal Processing System (ESPS) and Waves * GoldWave * Kay Elemetrics Computer Speech Lab * Khoros * Matlab plus Signal Processing Toolbox * MacSpeech Lab II * N!Power * OGI Speech Tools * Ptolemy * Quadravox Speech Processing Products - Qbox * Speech Filing System (SFS) * Signalyze 3.0 from InfoSignal * SoundScope Q1.11: Miscelaneous Software and Resources Speech Application Interfaces * ASAPI: Advanced Speech API (AT&T) * SAPI: Microsoft Windows Speech API * SRAPI: Speech Recognition API * TAPI: Microsoft Windows Telephony API Network "Phone" Software * CUSeeMe * CyberPhone * DigiPhone * InterFACE from Hijinx * FAQ: How can I use the Internet as a telephone? * Nautilus: Secure Computer Telephony * NEVOT (1.4v) from AT&T BL * PGPfone * Speak Freely * Internet Phone from VocalTec * WebPhone * WebTalk Audio Processing Software * AF version AF3R1 * Voice E-Mail from Bonzi Software * MicNotePad Recording Software for Macs * MixViews * Network Audio System Release 1.1 * NIST Software - SPHERE and SCORE * Sound Processing Kit * TCPplay Human Audio Perception * Auditory Modeller 1 * Auditory Modeller 2 * Auditory Toolbox for Matlab * Human Audio Perception Document Dictionaries and other Lexical Tools * BEEP dictionary * CMU dictionary * CUVOLAD dictionary (Oxford Dictionary) * Comprehensive Word List * EAT: Edinburgh Associative Thesaurus * Homophone List * Moby Lexical Resources * MRC Psycholinguistic Database * WordNet * Dictionaries on the WWW Phonetic Fonts and Phonetic Samples * International Phonetic Alphabet * WWW: Phonetic Fonts and Examples Online * Summer Institute of Linguistics IPA Fonts * Phonetic Fonts for TeX and LaTeX * Yamada Language Center Very Miscellaneous Software * The vOICe * The Learning Company's Language Training * Wildfire - an Electronic Assistant Q2.6: Audio Hardware * Macintosh Audio Hardware * PC Audio Hardware * Unix Audio Hardware Q2.8: Signal Processing Software * SigLib from Numerix Ltd. Q3.3: Compression Software and Hardware * 32 kbps ADPCM * Castleton Network Systems - G.729 Voice Coder * CELP 3.2a & LPC-10 * 8 Kbit/s CELP on the TMS320C5x family of DSP chips * CyberVoice * Rockwell's DigiTalk * File format conversion * G.711/721/723 Compression * G.728 LD-CELP vocoder * G.728 Compression * GSM 06.10 Compression * Lernout & Hauspie Speech Coding (5 products) * Lernout & Hauspie Speech Coding SDK * MPEG Audio * shorten - a lossless compressor for speech signals * Sipro Lab Telecom Inc. Coding * Sonarc: Digital Audio Compression * StarAudio Compressor/Player * TrueSpeech from DSP Group * U.S.F.S. 1016 CELP vocoder for DSP56001 * ToolVox from Voxware Q4.2: Natural Language Processing * Natural Language Software Registry (NLSR) - NLP Tools * Part of Speech Tagger Q5.5: Speech Synthesis _Apple Macintosh_ * BeSTspeech from Berkeley Speech Technologies, Inc., (BST) * Infovox Product Range * Macintosh Speech Output Applications * Macintosh Speech Synthesis Manager * MacYack Pro * MBROLA: Free Speech Synthesis Project * ProVoice Developer's Speech Toolkit from First Byte * SENSYN speech synthesizer * Sound Bytes DeveloperUs Kit * Macintosh Speech Synthesis Manager _Windows (including 95, NT, 3.1)_ * AcuVoice * AT&T Watson Speech Synthesis * BeSTspeech from Berkeley Speech Technologies, Inc., (BST) * Creative TextAssist and TextAssist API * DECtalk: Text-to-Speech from Digital * ETI-Eloquence * HADIFIX * Infovox Product Range * IPOX: All Prosodic Speech Synthesis Architecture * Lernout and Hauspie Text-To-Speech Windows SDK * Listen2 Text Reader * MBROLA: Free Speech Synthesis Project * Monologue for Windows from First Byte * PAM - A Text-To-Speech Application * ProVerbe Speech Engine from ELAN Informatique * ProVoice Developer's Speech Toolkit from First Byte * SENSYN speech synthesizer * Sound Bytes DeveloperUs Kit * Tinytalk * TruVoice from Centigram * WinSpeech * ZMD Speech Synthesis _DOS_ * CSRE: Computerized Speech Research Environment * Infovox Product Range * MBROLA: Free Speech Synthesis Project * ProVoice Developer's Speech Toolkit from First Byte * SENSYN speech synthesizer * spchsyn.exe * Tinytalk * ZMD Speech Synthesis _OS/2_ * ProVerbe Speech Engine from ELAN Informatique * ProVoice Developer's Speech Toolkit from First Byte * Sound Bytes DeveloperUs Kit _Unix_ * AcuVoice * AsTeR * BeSTspeech from Berkeley Speech Technologies, Inc., (BST) * DECtalk: Text-to-Speech from Digital * ETI-Eloquence * Emacspeak - A Speech Output Subsystem For Emacs * Festival Speech Synthesis System * JSRU * Klatt-style synthesiser * KPE80 - A Klatt Synthesiser and Parameter Editor * "learph": Trainable text-to-phoneme software by Antonio Lucca * Lucent Technologies Bell Labs Text-to-Speech system * MBROLA: Free Speech Synthesis Project * Orator from Bellcore * ProVerbe Speech Engine from ELAN Informatique * rsynth * SENSYN speech synthesizer * SGI Developers Toolbox Synthesiser * Speak * TrueTalk * TruVoice from Centigram _Integrated Circuits and Dedicated Hardware_ * Eurovocs * Infovox Product Range * ProVerbe Speech Engine from ELAN Informatique * RC Systems V8600/V8601 Text to Speech synthesizers _Other Platforms_ * BeSTspeech from Berkeley Speech Technologies, Inc., (BST) * TheBigMouth (NeXT) * MBROLA: Free Speech Synthesis Project * Narrator Translator Library (Amiga) * Narrator (Amiga) * TextToSpeech Kit (NeXT) * Orator from Bellcore * SENSYN speech synthesizer * WreadFiles: File reader for Commodore Amiga _Unknown_ * Lernout and Hauspie Text-To-Speech (3 products) * SIMTEL * Text to Phoneme Program 1 * Text to phoneme program 2 * Text to phoneme program 3 Q6.5: Speech Recognition _Apple Macintosh_ * Digital Dreams Speech Recognition Plug-Ins * Dragon Dictation Products * Macintosh Speech Recognition Manager * PowerSecretary _Windows (including 95, NT, 3.1)_ * AT&T Watson Speech Recognition * Cambridge Voice for Windows * CustomVoice and CustomTelephone: A&G Graphics Interface Inc. * DragonDictate for Windows * Dragon Dictation Products * Dragon Developer Tools * Ficomp Interpreter 6000 * IBM VoiceType Dictation and Control * IN CUBE * Kurzweil Speech Recognition (2 products) * Lernout & Hauspie ASR SDK * Listen for Windows 2.0 from Verbex Voice Systems * Microsoft Speech Recognition * NCC Dictate * Phonetic Engine 500 (PE500) from Speech Systems, Inc. * Philips Speech Recognition (2 products) * ProNotes Voice Tools * PureSpeech * smARTspeak from Advanced Recognition Technologies, Inc. * Visual Voice from Stylus Innovation * VoiceAssist for Windows from Creative Labs, Inc. * VoiceServer for Windows * Whisper * WildCard Speech Products _DOS_ * DATAVOX - French * Dragon Developer Tools * Ficomp Interpreter 6000 * Jialong He's Speech Recognition Research Tool * smARTspeak from Advanced Recognition Technologies, Inc. * Votan VPC2100 Voice Card and VSP 1010 Speech Processor _OS/2_ * IBM VoiceType Dictation and Control _Unix_ * AbbotDemo * BBN Hark Telephony Recognizer * EARS: Single Word Recognition Package * Ficomp Interpreter 6000 * Hidden Markov Model Toolkit (HTK) from Entropic * IN CUBE * Jialong He's Speech Recognition Research Tool * Lotec Speech Recognition Package * Myers' Hidden Markov Model software * NICO Artificial Neural Network Toolkit * Nuance Speech Recognition System * PureSpeech * recnet _Integrated Circuits and Dedicated Hardware_ * HM2007 - Speech Recognition Chip * OKI VRP6679 - Speech Recognition Chip * Sensory Inc. Integrated Circuits * Speech Commander - Verbex Voice Systems * Voice Control Systems Recognition * VCS 2030 & 2060 Voice Dialer _Other Platforms_ * Simon Says (NeXT) * Voice Command Line Interface (Amiga) * Visus SpeechKit _Unknown_ * Berkeley Restaurant Project (BeRP) * Lernout & Hauspie ASR (3 products) * Voice-Trek 2.0 * Voicetek Corp. * Voice Processing Corporation Speech Recognition Product Line Q6.6: Speaker Verification and Identification * ImagineNation: Voice Activated UnLock Technology * Jialong He's Speaker Recognition (Identification) Tool * Keyware Biometric Security Products * SpeakerKey Voice Verifier from ITT * SpeakEZ Voice Print Speaker Verification * Voice Control Systems: Speaker Verification Technology Q6.7: Integrated Speech Products * SpeechWorksfrom Applied Language Technologies, Inc. * Nortel Speech Technology Products ___________________________________________________________________________ General Speech Technology comp.speech FAQ Section 1 * SpeechLinks: General * Q1.1: What is comp.speech? * Q1.2: comp.speech ftp site * Q1.3: Common abbreviations and jargon * Q1.4: Related newsgroups and mailing lists * Q1.5: Associations, publications and conferences * Q1.6: Handicap Aids * Q1.7: Speech Databases * Q1.8: Speech File Formats and Conversion * Q1.9: Speech Laboratory Environments and Audio Editors * Q1.10: Speech Research Sites * Q1.11: Miscellaneous Software and Resources Q1.1: What is comp.speech? Comp.speech is an unmoderated newsgroup for discussion of speech technology and speech science. It covers a wide range of issues from the application of speech technology, to research, to products and lots more. By its nature, speech technology is an inter-disciplinary field and the newsgroup reflects this. However, computer application is the basic theme of the group. Note: If you don't know what a newsgroup is, then talk to your local system administration about how to get access. A useful newsgroups for beginners is news.announce.newusers. You might also find the following documents useful. ftp://rtfm.mit.edu/pub/usenet/news.announce.newusers/What_is_Us enet? ftp://rtfm.mit.edu/pub/usenet/news.announce.newusers/Answers_to _Frequently_Asked_Questions_about_Usenet ftp://rtfm.mit.edu/pub/usenet/news.announce.newusers/Rules_for_ posting_to_Usenet ftp://rtfm.mit.edu/pub/usenet/news.announce.newusers/FAQs_about _FAQs The following is a list of some of the topics covered by comp.speech. * Speech Recognition - discussion of methodologies, training, techniques, results and applications. This should cover the application of techniques including HMMs, neural-nets and so on to the field. * Speech Synthesis - discussion concerning theoretical and practical issues associated with the design of speech synthesis systems. * Speech Coding and Compression - both research and application matters. * Phonetic/Linguistic Issues - coverage of linguistic and phonetic issues which are relevant to speech technology applications. Could cover parsing, natural language processing, phonology and prosodic work. * Speech System Design - issues relating to the application of speech technology to real-world problems. Includes the design of user interfaces, the building of real-time systems and so on. * Other matters - relevant conferences, jobs, books, software, hardware, and products. ___________________________________________________________________________ Q1.2: comp.speech ftp site Tony Robinson maintains the comp.speech ftp site. The ftp site is a comprehensive repository of software and information related to speech technology. The site is * ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/ Comp.speech Archives The comp.speech ftp site provides full archives of the comp.speech newsgroup dating back to the creation of the group in 1991. The postings are stored in the order in which they arrive. Batches of 1000 articles are grouped into gzip'ed tar file. Matching files listing the subjects are also provided. * ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/archive/ Software and Other Resources The comp.speech ftp site includes a wide range of useful software and resources. Tony has arranged it into a series of sub-directories: /analysis : Speech analysis software FFT code, a pitch tracker, RASTA code, and IEEE DSP code. /auditory : Auditory model software AIM, Auditory Toolbox and Lutear. /coding : Speech coding software ADPCM, CELP 3.2a, G711, G721, G723, GSM, LDCELP, LPC10, Shorten. /data : Repository for (small) speech-related databases BEEP, CMUDict, Homophone list, hVd database, Peterson Barney database /dictionaries : Phonetic dictionaries BEEP, CMUDict, CUVOALD, Homophone list, MRC database /info : Key postings to comp.speech archives by subject Lots of interesting info! /recognition : Speech recognition software AbbotDemo, Ears, Lotec, recnet, sound blaster recognition, whistle /simtel_sound : Mirror of the simtel/msdos/sound directory Range of useful software /simtel_voice : Mirror of the simtel/msdos/voice directory Another range of useful software /synthesis : Speech synthesis software Klatt synthesis software, Klatt parameter editor and rsynth. /tools : Miscelaneous tools Part-of-speech tagger, OGI speech tools, sox audio file format conversion, SPHERE software and more. ___________________________________________________________________________ Q1.3: Common abbreviations and jargon. * ANN - Artificial Neural Network. * ASR - Automatic Speech Recognition. * ASSP - Acoustics Speech and Signal Processing * AVIOS - American Voice I/O Society * CELP - Code-book Excited Linear Prediction. * COLING - COmputational LINGuistics * DTW - Dynamic Time Warping. * FAQ - Frequently Asked Questions. * HMM - Hidden Markov Model. * IEEE - Institute of Electrical and Electronics Engineers * JASA - Journal of the Acoustic Society of America * LPC - Linear Predictive Coding. * LVQ - Learned Vector Quantisation. * MFCC - Mel Frequency Cepstral Coefficients * NLP - Natural Language Processing. * NN - Neural Network. * TIMIT - A speech corpus with phoneme labels - see Q1.7 * TTS - Text-To-Speech (i.e. speech synthesis). * VQ - Vector Quantisation. ___________________________________________________________________________ Q1.4: Related newsgroups and mailing lists. Newsgroups comp.ai - Artificial Intelligence newsgroup. Postings on general AI issues, language processing and AI techniques. The comp.ai FAQ covers NLP, NN and other AI information. comp.ai.nat-lang - Natural Language Processing Group Postings regarding Natural Language Processing. Set up to cover a broard range of related issues and different viewpoints. A comp.ai.nat-lang FAQ posting is available. comp.ai.nlang-know-rep - Natural Language Knowledge Representation Moderated group. comp.ai.neural-nets - discussion of Neural Networks and related issues. There are often posting on speech related matters - phonetic recognition, connectionist grammars and so on. A comp.ai.neural-nets FAQ posting is available. comp.compression - occasional articles on compression of speech. The comp.compression FAQ has some info on audio compression standards. comp.dcom.telecom - Telecommunications newsgroup. Has occasional articles on voice products. comp.dsp - discussion of signal processing - hardware and algorithms and more. Has a good FAQ posting which is also available on the WWW and by ftp (addresses below). Has a regular posting of a comprehensive list of Audio File Formats. + http://www.bdti.com/faq/dsp_faq.htm + ftp://rtfm.mit.edu/pub/usenet/comp.dsp/ comp.multimedia - Multi-Media discussion group. Has occasional articles on voice I/O. sci.lang - Language. Discussion about phonetics, phonology, grammar, etymology and lots more. A sci.lang FAQ is available. alt.sci.physics.acoustics Some discussion of speech production & perception. alt.binaries.sounds.* - posting and discussion of sound samples. Mailing Lists Voice-Users Mailing List For discussion of any aspect of using voice recognition systems. + Using such systems safely, without muscle or voice strain + Techniques for improving recognition accuracy + How to set up the physical voice workstation + Tips for effective use of voice interfaces + Configuration of specific systems, troubleshooting, etc To subscribe fill out the web-based subscription form Posts to the list should go to: voice-users@voicerecognition.com Colibri News about language, speech, logic and information. Email: colibri@let.ruu.nl WWW: http://colibri.let.ruu.nl/ ECTL - Electronic Communal Temporal Lobe Founder & Moderator: David Leip. Moderated mailing list for researchers with interests in computer speech interfaces. This list serves a broad community including persons from signal processing, AI, linguistics and human factors. To subscribe, send your name, institute, department, daytime phone and email address to: + ectl-request@snowhite.cis.uoguelph.ca The ECTL archive site is ftp://snowhite.cis.uoguelph.ca/pub/ectl Prosody Mailing List Unmoderated mailing list for discussion of prosody. The aim is to facilitate the spread of information relating to the research of prosody by creating a network of researchers in the field. If you want to participate, send the following one-line message to + listserv@msu.edu + subscribe prosody Your Name foNETiks A moderated monthly newsletter distributed by e-mail. It carries job advertisements, notices of conferences, and other news of general interest to phoneticians, speech scientists and others. The editors are Linda Shockey and Gerry Docherty. To subscribe send the following 1 line message to + mailbase@mailbase.ac.uk + join fonetiks your_first_name your_second_name Digital Mobile Radio Covers lots of areas include some speech topics including speech coding and speech compression. Mail Peter Decker dec@dfv.rwth-aachen.de to subscribe. ___________________________________________________________________________ Q1.5: Associations, Journals and Conferences [Note: Also see the list provided in Shikano's WWW site on Speech and Acoustics: http://www.aist-nara.ac.jp/IS/Shikano-lab/database/internet-resource/e -www-site.html.] Associations Institute of Electrical and Electronics Engineers (IEEE) * Publications: include IEEE Transactions on Signal Processing, IEEE Transactions on Speech and Audio (from Jan 93), IEEE Transactions on Acoustics, Speech, and Signal Processing (now obsolete), IEEE Signal Processing Magazine. (More information on the WWW: http://www.ieee.org/sp/index.html). * Speech-Related Conferences: ICASSP - Intl. Conf. Acoustics, Speech, and Signal Processing. IEEE also runs speech technology related workshops and many other conferences. (Does anyone have a list?) * Contact: IEEE Service Center 445 Hoes Lane, PO Box 1331, Piscataway, NJ 08855, USA Phone: 1-800-678-IEEE or (201) 981-0060 * WWW: IEEE: http://www.ieee.org/ IEEE Signal Processing Society http://www.ieee.org/sp/index.html The Acoustical Society of America (ASA) * Publications: Journal of the Acoustical Society of America (JASA) * Conferences: ASA holds four meetings a year. Information is available on the WWW: http://asa.aip.org/meetings.html. * Contact: ASA Office Manager, 500 Sunnyside Blvd, Woodbury, NY 11797-2999, USA Ph: (516) 576-2360, FAX (516) 576-2377 Email: asa@aip.org * WWW: http://asa.aip.org/ European Speech Communication Association (ESCA) * Publications: Speech Communications * Conferences: EUROSPEECH is held every two years. E'97 will take place in Patras, Greece, in September 1997. ESCA organises regular speech-related workshops: see their WWW pages for details. * Contact: Secretariat ESCA ICP, Universite Stendhal, BP 25X, F38400 Grenoble Cedex 9, France Ph: (+33).76.82.43.36 Fax (+33).76.82.43.35 Email: esca@icp.grenet.fr * WWW: http://ophale.icp.grenet.fr/esca/esca.html Association for Computational Linguistics (ACL) * Publications: Computational Linguistics * SIGPHON: Special Interest Group for Computational Phonology. The home page is provided by the Centre for Cognitive Science at the University of Edinburgh. A special issue on Computational Phonology appeared in Vol 20, Num 3 of Computational Linguistics and included an Introduction to Computational Phonology by Steven Bird * Conferences: COLING is held bi-annually. ACL also organises a range of workshops. See the WWW pages for details. * Contact: P.O. Box 6090 Somerset, NJ 08875, USA Ph: (908) 873 3893 Email: acl@bellcore.com * WWW: http://www.cs.columbia.edu:80/~acl/ American Voice Input/Output Society (AVIOS) * Description: AVIOS is a not-for-profit organization, dedicated to disseminating information about applications using speech technology. It aims "to bridge the gap between emerging voice technology and its application, by providing an interactive forum for the technologists, students, system developers, business managers, and users actively involved in or with an interest in the field of voice processing." * Publications: International Journal of Speech Technology (with Kluwer Academic Publishers) The Journal of the American Voice Input/Output Society was published from 1984 to 1994. * Conferences: The International Voice Input/Output Applications Conference is held annually (since 1982): Sept 10-12, San Jose, CA. * Contact: 4010 Moorpark Avenue, Suite 105M, San Jose, CA 95117, USA Ph: +1-408-248-1353, Fax: +1-408-248-0251 Email: avios@pilot.net WWW: http://www.avios.com/ European Language Resources Association * Description: The European Language Resources Association was established in Luxembourg in February, 1995, with the goal of creating an organization to promote the creation, verification, and distribution of language resources in Europe. A non-profit organization, ELRA aims to serve as a central focal point for information related to language resources in Europe, It will help users and developers of European language resources, as well as government agencies and other interested parties, exploit language resources for a wide variety of uses. It will also oversee the distribution of language resources via CD-ROM and other means and promote standards for such resources. * More info: see the ELRA Home page for membership information, lists of resources etc. * Contact: K. Choukri, Executive Director ELRA 87, Avenue d'Italie, 75013 Paris, FRANCE Ph: +33 1 45 86 53 00, Fax: +33 1 45 86 44 88 Email: elra@calvanet.calvacom.fr WWW: http://www.icp.grenet.fr/ELRA/home.html ASSTA: Australian Speech Science and Technology Association * Conference: SST, the Australian conference on Speech Science and Technology, is held bi-annually. SST-96 will be held in Adelaide. * WWW: Home Page: http://cslab.anu.edu.au/~bruce/assta/ List of members: http://ciips.ee.uwa.edu.au/~roberto/assta-users/ SALT: UK Speech and Language Technology Club * WWW home page: http://salt.essex.ac.uk/salt/ Linguistic Associations * A comprehensive list of linguistic associations and linguistic WWW links is available at http://engserve.tamu.edu/files/linguistics/linguist/associations.h tml Industry Publications ASR News * Description: Monthly newsletter covering developments in the speech recognition and speech synthesis marketplace. * Note: Voice Information Associates also publish "Automatic Speech Recognition: A study of the world-wide market" (revised 1995) and "Text-to-Speech Technology Markets: 1995-2000" (revised 1995) * Contact: Voice Information Associates, Inc. 14 Glen Road South, P.O. Box 625, Lexington, MA 02173, USA Ph: +1-617-861-6680, Fax: +1-617-863-8790 Email: asrnews@tiac.net WWW: http://www.tiac.net/users/asrnews/ Voice News * Description: Monthly newsletter reporting on voice mail, voice response, speech recognition, speech synthesis, digital voice record/playback and related technologies, markets and company activities. Review copy available on request. * Contact: Stoneridge Technical Services P.O. Box 1891, Rockville, MD, 20849, USA Ph: +1-301-424-0114, Fax: +1-301-424-8971 Email: info@stoneridgetech.com WWW: http://www.stoneridgetech.com/ Speech Recognition Update * Description: Monthly news and analysis of speech recognition markets, applications and technology. A free sample copy is available by contacting TMA Associates. * Also: TMA Associates also publishes market studies, including The Advanced Speech Technology Market: Recognition, Synthesis and Compression (1996) and Voice ID (1996) . Contact: TMA Associates 6021 Wish Avenue, Encino, CA 91316, USA Ph: +1-818-708-0962, Fax: +1-818-345-2980 Email: 72162.3172@compuserve.com http://www.tmaa.com/ Voice Technology and Services News * Description: Follows integrated PC LAN messaging (voice, fax, mail, video) and speech technology. It follows the merging computer and telephone technologies, provides insights into business and marketing opportunities and offers executive timely information on industry trend analysis. * Contact: Phillips Business Information 1201 Seven Locks Rd., Potomac, Maryland, 20854, USA Ph: 1-800-777-5006 OR +1-301-340-1520 Subscription FAX: +1-301-309-3847 Editorial FAX: +1-424-4297 Telleconnect * Contact: +1-212-691-8215 Computer Telephony * Contact: +1-212-691-8215 Voice Processing Magazine * Contact: 1-800-854-3112 Speech Technology * Description: No longer published Technical and Research Publications Computer Speech and Language * Price: $US170 (Institutions), $US75 (Individuals), 4 issues per year. * Publisher: Academic Press Limited 24-28 Oval Road, London NW1, England WWW: http://www.apnet.com/ Speech Communication * Contact: ESCA (see above) * Publisher: Elsevier Science B.V. P.O. Box 521, 1000 AM Amsterdam, The Netherlands. WWW: http://www.elsevier.com/ IEEE Transactions on Speech and Audio Processing, IEEE Signal Processing Magazine, IEEE Transactions on Acoustics, Speech, and Signal Processing: OBSOLETE * Contact: IEEE (see above) Free Speech Journal * Description: A Web Journal dedicated to the state of the art in human language technology. Past volumes, editorial and submission information, and so on are * Contact: Editor-In-Chief: Ron Cole: cole@cse.ogi.edu WWW: http://www.cse.ogi.edu/CSLU/fsj/html/masthead.html Linguistics Abstracts Online * Description: online access to all abstracts published in Linguistics Abstracts since 1985, plus all current material as it becomes available. Over 250 publications are indexed. Free trial available. http://www.blackwellpublishers.co.uk/labs/ Computational Linguistics * Contact: Published by Computational Linguistics Assoc. (see above) Journal of the Acoustical Society of America (JASA) * Contact: Published by Acoustical Society of America (see above) International Journal of Speech Technology (was the AVIOS Journal) * Description: Focuses on speech technology and its applications, and promotes research and description of all aspects of speech input and output: applications, base technology, theory, approach, experiment, and testing. * Publisher: Kluwer Academic Publishers 101 Philip Drive, Norwell, MA 02061, USA Ph: +1-617-871-6300, Fax: +1-617-871-0449 * Submissions to: International Journal of Speech Technology Journals Editorial Office, Ms. Kelly Riddle Kluwer Academic Publishers (Address, phone, fax as above) Email: krkluwer@world.std.com Conferences ICSLP: Intl. Conference on Spoken Language Processing Next: 30 Nov to 4 Dec, 1998, Sydney, Australia Held in even years. ICASSP - Intl. Conf. Acoustics, Speech, and Signal Processing Eurospeech Computational Linguistics (COLING), held bi-annually International Voice Input/Output Applications Conference SST: Australian Speech Science and Technology Conference Also see the following lists on the WWW: Shikano's WWW site on Speech and Acoustics http://www.aist-nara.ac.jp/IS/Shikano-lab/database/internet-res ource/e-www-site.html Institute of Phonetic Sciences WWW list http://fonsg3.let.uva.nl/Other_pages.html#Meetings ___________________________________________________________________________ Q1.6: Handicap Aids The following are products and companies which support users who can benefit from the use of speech technology in a user interface. Please feel free to submit information on relevant products, names of companies and links to useful information on the Internet (especially WWW sites). [Of course, most of the products listed in Q5.5 and Q6.5 are useful.] * Man-Machine Interfacing * SpeechViewer II Man-Machine Interfacing * Description: Offers a service designed for people with physical challenges. Can successfully implement a computerized voice controlled system adapted to unique needs. They have developed a free-standing microphone and signal processing system to compensate for speech/articulation distortions, and background noise produced by electronic devices such as wheelchairs and respirators. * Contact: Man-Machine Interfacing P.O. Box 5371, Evanston, IL 60204 Ph: 1-888-425-2001, Fax : (847) 328-7975 Email: jwhite@mcs.com WWW: http://www.speechrec.com/ SpeechViewer II * Platform: IBM Machines from Mod 25 on. * Description: SpeechViewer II is a speech therapy tool. It provides graphical feedback of various speech features so that speech impaired individuals can improve their speech. It works with an audio bandwidth of 7.3 Khz and thus allows the therapist to work with sustained vowels and fricatives. A wide range of graphics are used to provide adequate variability to hold client interest. An extensive set of statistics are gathered which allows a therapist to do research or keep therapy records. The speech therapy modules are: + Awareness - Sound, Loudness, Pitch, Voicing Onset, Voicing + Skill Building - Pitch, Voicing, Phonology + Patterning - Pitch & Loudness - Waveform & Spectrogram, Spectra + Clinical Management - Profiles, Models, Client Data A multilingual option is available which provides support for 12 languages: Danish, Dutch, Finnish, French, German, Icelandic, Italian, Norwegian, Portuguese, Spanish, Swedish, and UK English. With the Multilingual Option, clinicians can use SpeechViewer II as a training tool for English as a second language and for foreign language training. * Hardware: Requires an IBM M-ACPA (Multimedia-Audio Capture Playback Adapter). It has a TI TMS320C25 DSP chip. The input sampling rate is 44.1 Khz stereo, 88.2 Khz mono. This is a 16 bit card. It has the following jacks: mic in, stereo line in, stereo line out, speaker out. Note: This card is being replaced by Mwave technology. For more info on Mwave contact Texas Instruments. * Price: + The software is $2130 list, $1491 educational, part number 92F2066. + The M-ACPA is $370 list, $222 educational, part number 92F3378. + The MicroChannel adapter part number is 92F3379 (same price). * Contact: IBM Special Needs Information 1000 N. W. 51st Street, Internal Zip 5432, Boca Raton, Florida 33431, USA Ph: 1-800-426-4832, TDD: 1-800-426-4833, Fax: 1-407-982-6059 Email: IBM_SPEC_NEEDS_INFO@vnet.ibm.com WWW: http://www.austin.ibm.com/pspinfo/snsspv2.html ___________________________________________________________________________ Q1.7: Speech databases A wide range of speech databases have been collected. These databases are primarily for the development of speech synthesis/recognition and for linguistic research. Some databases are free but most are not. The databases normally require lots of storage space (100's of MBytes is not unusual). Do not expect to be able to ftp large amounts of speech data. In addition to the descriptions of speech databases and speech database providers below, information can be obtained from LDC: Linguistic Data Consortium Provides a very wide range of speech and text data to research and commercial users: see below. COCOSDA Home Page: http://www.itl.atr.co.jp/cocosda/ The International Committee for the Co-ordination and Standardisation of Speech Databases and Assesment Techniques for Speech Input/Output. Shikano's WWW site on Speech and Acoustics http://www.aist-nara.ac.jp/IS/Shikano-lab/database/internet-res ource/e-www-site.html RELATOR Project European resource initiative: see below. The following speech data resources are described in the FAQ. * Bavarian Archive for Speech Signals * BUPT Spoken Digit Database (Chinese) * Center for Spoken Language Understanding (CSLU) * Examples of IPA Symbols * Linguistic Data Consortium (LDC) * NOISEX * Oxford Acoustic Phonetic Database * Phonemic Samples * RELATOR project * ShATR * University of Victoria Phonetic Database Bavarian Archive for Speech Signals * Description: The Bavarian Archive for Speech Signals (BAS) was founded in January 1995 as an initiative of the Institute of Phonetics at the University of Munich, Germany. The BAS will develop, validate, administrate and disseminate corpora of spoken German to the speech community as well as to speech engineering industry. Presently the following German speech corpora are available on ISO 9660 CDROM: Siemens 1000 - SI1000 5 CDROMs, newspaper corpus, read speech, 10 speakers x 1000 utterances Siemens 100 - SI100 7 CDROMs, read speech, 101 speakers x 100 sentences PhonDat 1 - PD1 6 CDROMs, new edition in preparation, read speech, 201 speakers x 450+ sentences PhonDat 2 - PD2 1 CDROM, read speech, 2nd edition, 16 speakers x 200 sentences, various labelled information Verbmobil Spontaneous speech recorded in a dialog task (appointment scheduling). More information on the VERBMOBIL project: http://www.dfki.uni-sb.de/verbmobil/ Corpora in Preparation PhonDat I - PD1: 2nd extended edition (Jul 1995) Strange Corpora - SC Reference Corpora that reflect certain well known problems in speech processing, like accents, repair, breaks, hesitations, repetitions, extreme F0, backround noise, pathological speech, speaker adaptation. The first SC corpus (SC1 Accents) will be edited in Jul 1995. BAS Edition of Verbmobil Corpora - VM: 2nd extended edition Articulatory data - AD: EMA data of speakers of SI1000 corpus ERBA: 10000 utterances from a train inquiry task * Misc: BAS is currently developing tools for the automatic annotation and segmentation of very large speech corpora. This includes the automatic detection of variants of pronunciation, a statistical based alignment and a rule-based refinement of the outcome. The BAS seeks to cooperate with public institutions as well as with industrial partners to further develop new German speech databases. BAS can be a platform to re-distribute existing German speech. * Contact and More Information: The BAS is located at the University of Munich, Germany. BAS c/o Institut fuer Phonetik Schellingstr. 3/II 80799 Muenchen, Germany Ph: +49-89-21802758, Fax: +49-89-2800362 Email: bas@sun1.phonetik.uni-muenchen.de WWW: http://www.phonetik.uni-muenchen.de/BASSeng.html BUPT Spoken Digit Database (Chinese) * Vocabulary : {0, 1/yi/, 2, 3, 4, 5, 6, 7, 8, 9, 1/yao/, /dui/, /cuo/ }, 13 words in total. * Size: 1202 speakers in total, 789 Males and 413 Females. Each speaker utters each word 2 times. Total of 31252 utterances. * Format: 8000Hz 14bit sampling. One utterance per file. * Contact: GLuck Co. 195 Berlioz 1C, Nun's Island Verdun H3E 1C1, Canada e-mail: weigang@zaphod.math.mcgill.ca Center for Spoken Language Understanding (CSLU) * The ISOLET speech database of spoken letters of the English alphabet. The speech is high quality (16 kHz with a noise cancelling microphone). 150 speakers x 26 letters of the English alphabet twice in random order. The ISOLET data base can be purchased for $100 by sending an email request to vincew@cse.ogi.edu. (This covers handling, shipping and medium costs). The data base comes with a technical report describing the data. * CSLU has a telephone speech corpus of 1000 English alphabets. Callers recite the alphabet with brief pauses between letters. This database is available to not-for-profit institutions for $100. The data base is described in the proceedings of the International Conference on Spoken Language Processing. + Contact vincew@cse.ogi.edu if interested. * CSLU has released for universities its Continuous English Speech Corpus. The corpus contains recorded speech from 690 different speakers, with label files at various levels - including word level and phonetic labels. The data were collected as part of the OGI Multi-language telephone corpus. CSLU provides speech corpora to all universities without charge. To order a corpus, print the license agreement/order form, complete it, and fax it to the CSLU. A description of the corpora and an order form are available: http://www.cse.ogi.edu/CSLU/ ftp://speech.cse.ogi.edu/pub/releases * Contact: Mike Noel: noel@cse.ogi.edu Examples of IPA Symbols UCLA Sounds of the World's Languages * Description: The UCLA Sounds of the World's Languages are available for Macintosh users (no DOS based system currently available). The sounds are stored in a Hypercard database developed at the UCLA Phonetics Laboratory. The aim is to illustrate and teach about the range of sounds used in human languages with material on more than 80 languages. The set demonstrates particular highlights of the sound systems focusing especially on rarer sounds that students may not otherwise have a chance to hear from a native speaker. The recordings are based on the archives of recordings collected at UCLA, with additional contributions from outside collaborators. All the languages can be accessed from the list of language names, or by clicking on the language name in a set of maps. Support for part of this work was provided by NSF. The database currently includes examples of languages from Agul and Akan to Zulu. * Availability: 15 DSDD disks, requiring about 35 meg of disk space when expanded. Available for $50 individual $100 institutions. Prepayment in US dollars (checks or international money orders payable to "UC Regents") must accompany all orders. * Contact: The UCLA Phonetics Laboratory Linguistics Department, UCLA, Los Angeles, CA 90095 1543 Tel: (310) 825-1254 E-mail: oldfogey@ucla.edu John Eslings "IPA Labels" * Description: A HyperCard stack which is available for free or a nominal fee. * Contact: John Esling can be reached by email: pdb@uvvm.uvic.ca. Linguistic Data Consortium (LDC) The LDC was established to broaden the collection and distribution of speech and natural language data bases for the purposes of research and technology development in automatic speech recognition, natural language processing and other areas where large amounts of linguistic data are needed. Detailed information on the LDC is now available on the WWW: http://www.ldc.upenn.edu/. The LDC WWW server provides information on membership agreements, license agreements, and summaries of speech and text corpora available. Speech Corpora * TIMIT Acoustic-Phonetic Continuous Speech Corpora and NYNEX Telephone Version of TIMIT Corpus (NTIMIT) * Resource Management Corpora * Air Travel Information System (ATIS) Corpora (multiple) * ARPA Continuous Speech Recognition Corpora (WSJ etc) * Switchboard Corpus of Recorded Telephone Conversations and Switchboard Corpus Excerpts (Credit Card Conversations) * Texas Instruments 46-Word Speaker-Dependent Isolated Word Corpus (TI46) * Texas Instruments Speaker-Independent Connected-Digit Corpus (TIDIGITS) * Road Rally Conversational Speech Corpus * HCRC Map Task Corpus * Air Traffic Control Corpus (ATC0) * SPIDRE Speaker Identification Corpus * YOHO Speaker Verification Corpus * OGI Multi-Language Corpus and OGI Spelled and Spoken Telephone Corpus * BRAMSHILL * MACROPHONE * King Corpus for Speaker Verification Research * WSJCAM0: Cambridge Read News Corpus * TRAINS Spoken dialog corpus * NYNEX PhoneBook Database * Frontiers in Speech Processing Text Corpora * Association for Computational Linguistics Data Collection Initiative (ACL/DCI) * The Penn Treebank Project - Release 2 * TIPSTER Information Retrieval Text Research Collection * United Nations Parallel Text Corpus (English, French, Spanish) * Japanese Language Financial New * European Corpus Initiative-1 Lexical Databases * CELEX Lexical Database * COMLEX : COMmon LEXical Database of English (English syntax and pronunciation) Contact information: Linguistic Data Consortium 3615 Market Street, Suite 200, Philadelphia, PA, 19104-2608, USA. Phone: +1 (215) 898-0464 Fax: +1 (215) 573-2175 e-mail: ldc@ldc.upenn.edu WWW: http://www.ldc.upenn.edu/ NOISEX-92 * Description: Database of recording of various noises available on 2 CDROMs. Some material from the same source is available by anonymous ftp in the IEEE's Signal Processing Information Base. The samples include + Voice babble + Factory noise + HF radio channel noise, pink noise, white noise + Various military noises; fighter jets (Buccaneer, F16), destroyer noises (engine room, operations room), tank noise (Leopard, M109), machine gun + Volvo 340 * Availability 1: The cost of this database is 135 Pounds Sterling for the set of two CD-ROMs. Send payment with order to: The Speech Research Unit, Ex1, DRA Malvern, St.Andrew's Road, Malvern, Worcestershire, WR14 3PS, UK Tel +44-684-894074 Fax +44-684-894384 Note: The supply of CD-ROMs is limited so please check that they are still available before placing an order. The only acceptable methods of payment are cheques (from the UK only) or bank drafts in Pounds Sterling drawn on a UK bank. They should be made payable to:- Public Sub Account HMG 4768. * Availability 2: Information on how to obtain a copy of the NATO RSG.10 NOISE-ROM-0 can be obtained from the DRA Speech Research Unit (address above) or from: Dr. Herman Steeneken, TNO Institute for Perception, P.O. Box 23, 3769 ZG Soesterberg, The Netherlands. * Availability 3 (WWW): Examples of the NOISEX database are available on the Rice University Digital Signal Processing (DSP) group home page. (Note the files are large (>20MB). http://spib.rice.edu/spib/select_noise.html Oxford Acoustic Phonetic Database * Available on compact disc, from J. Pickering and B. Rosner. It contains data on vowel-consonant and consonant-vowel combinations in both stressed and unstressed locations. The language covered include French, German, Hungarian, Italian, Japanese, British English, Spanish and English. For further information write to Electronic Publishing, Oxford University Press, Walton Street, Oxford OX2 6DP, UK. The ISBN is 0-19-268086-2 * Contact: Prof. B. Rosner Dept. of Experimental Psychology South Parks Rd, Oxford, OX1 3UD, UK email: burton.rosner@wolfson.ox.ac.uk Phonemic Samples * Some basic data. The following ftp sites have samples of English phonemes (American accent I believe) in Sun audio format files. See Question 1.8 for information on audio file formats. ftp://sounds.sdsu.edu/.1/phonemes: This ftp site appears to be obsolete. Does anyone know a new address? ftp://phloem.uoregon.edu/pub/Sun4/lib/phonemes: There appears to be some config problem with this ftp server. ftp://sunsite.unc.edu/pub/multimedia/sun-sounds/phonemes The RELATOR project * Description: RELATOR is a European-wide consortium of researchers who, with the support of the European Commission, are striving to establish a European repository of linguistic resources. Linguistic resources comprise a variety of spoken and written language materials, including lexicons, grammars, corpora, and spoken language databases. RELATOR will ensure that the requirements of the European language processing community receive attention. The RELATOR WWW pages provide information on the consortium, The languages currently covered by the RELATOR consortium include Danish, Dutch, English, French, German, Greek, Italian, Portuguese, Spanish plus multilingual resources. The resources include both text and speech. * WWW: http://cristal.icp.grenet.fr/Relator/homepage.html ShATR * Description: Multi-simultaneous-speaker corpus available on one CDROM. This specialised corpus is primarily intended to provide acoustic material for studies in auditory scene analysis. However many researchers in the speech sciences, ranging from acoustics to discourse analysis may find it a valuable source of information. The corpus has been transcribed and aligned at four different levels of analysis. An overlap analysis between the individual speaker channels and word counts are available. There is also a general tool for accessing concurrent events in transcribed multi-sound-source databases. * Cost: 30 Pounds Sterling for one CD-ROM. Availability, licensing and ordering information is provided on ShATR's home page. * Examples: Samples of the ShATR database are available on ShATR's home page and by anonymous ftp ftp://ftp.dcs.shef.ac.uk/share/spandh/ShATR/ * Contact: Speech and Hearing Research Group Department of Computer Science, University of Sheffield Regents Court, 211 Portobello Street, Sheffield S1 4DP, U.K. WWW: http://www.dcs.shef.ac.uk/research/groups/spandh/pr/ShATR/ShATR.ht ml University of Victoria Phonetic Database * Platform: Computerized Speech Lab CSL4300, MultiSpeech on Winxx or Win95 with any multimedia card, or a SoundBlaster16 option with support from the PDBAUDIO program. * Description: Phonetic database consisting of proprietary format digitized speech samples from 45 world languages on CDROM. The CDROM is supported by hardcopy documentation containing the phonetic inventory of each language, transcriptions and orthography of each digitized speech sample. The PDB depicts and compares the the sounds, symbols and conventions of transcription used by these languages. More information is available from the STR web site. * Contact: Speech Technology Research Ltd., Suite B - 1623 McKenzie Avenue, Victoria, B.C. V8N 1A6, Canada Ph: +1-250-477-0544 Email: products@speechtech.com WWW: http://www.speechtech.com/home/speechtech/ ___________________________________________________________________________ Q1.8: Speech File Formats and Conversion Q2.7 of this FAQ has information on mu-law coding. A very good and very comprehensive list of audio file formats is prepared by Guido van Rossum. The list is posted regularly to comp.dsp and alt.binaries.sounds.misc, amongst others. It includes information on sampling rates, hardware, compression techniques, file format definitions, format conversion, standards, programming hints and lots more. It is also available by ftp from WWW: ftp://ftp.cwi.nl/pub/audio/index.html Text: ftp://ftp.cwi.nl/pub/audio/AudioFormats.part1,2 A useful source of software (Sox, ulaw conversion, SoundKit etc) is: http://peace.wit.com/sounds/SoundConversion/ ___________________________________________________________________________ Q1.9: Speech Laboratory Environments and Audio Editors First, what is a Speech Laboratory Environment? A speech lab is a software package which provides the capability of recording, playing, analysing, processing, displaying and storing speech. Your computer will require audio input/output capability. The different packages vary greatly in features and capability - best to know what you want before you start looking around. Most general purpose audio editing packages will be able to process speech but do not necessarily have some specialised capabilities for speech (e.g. formant analysis). The following article provides a good survey. * Read, C., Buder, E., & Kent, R. "Speech Analysis Systems: An Evaluation" Journal of Speech and Hearing Research, pp 314-332, April 1992. The following is a list of the speech labs described in the FAQ. * CSRE: Computerized Speech Research Environment * DADiSP from DSP Development Corporation * Entropic Signal Processing System (ESPS) and Waves * GoldWave * Kay Elemetrics Computer Speech Lab * Khoros * Matlab plus Signal Processing Toolbox * MacSpeech Lab II * N!Power * OGI Speech Tools * Ptolemy * Quadravox Speech Processing Products - Qbox * Speech Filing System (SFS) * Signalyze 3.0 from InfoSignal * SoundScope CSRE: Computerized Speech Research Environment * Platform: DOS * Description: CSRE (pronounced "Caesar") is a speech processing system for the PC. It provides + Signal recording and playback + Signal editing + Pitch and spectral analysis and formant analysis + Speech synthesis with an implementation of the Klatt-1980 parametric speech synthesizer * Requirements: PC compatible (80486DX), 1 Meg RAM (recommend 4M), DOS 3.2 (recommend 6.22), VGA graphics (640x480; 16 colors) 30 Meg of hard disk space (5 Meg for CSRE plus space for audio recordings), and a supported audio card . * Cost: See AVAAZ WWW Pages * Contact: AVAAZ Innovations Inc. P.O.Box 8040, 1225 Wonderland Rd. N, London, Ontario, CANADA, N6G 2B0 Ph: +1-519-472-7944, Fax: +1-519-472-7814 Email: info@avaaz.com WWW: http://www.icis.on.ca/homepages/avaaz/ * Note: See also the CSRE entry in Q5.5 on speech synthesisers. DADiSP from DSP Development Corporation * Platform: Windows and various Unix * Description: DADiSP is designed for scientists and engineers to collect, analyze, and display scientific and technical data. Packages available include AdvDSP, Controls, DADiMP, Filters, GPIBLab, NeuralNet, and Stats. A description of the application of DADiSP to speech processing is provided on the DSP Development Corporation WWW site. Detailed product information is available on the DSP Development Corporation WWW site and by filling out a WWW form. * Cost: Unknown * Availability: See the DSP Development Corporation WWW site A free, fully featured demo of DADiSP 4.0 is available from the DSP Development Corporation WWW site and can be mailed on floppy disk. A special Student Edition of DADiSP is available for free. * Contact: DSP Development Corporation One Kendall Square, Cambridge, MA 02139, USA Ph: (617) 577-1133 Fax: (617) 577-8211 EMail: info@dadisp.com WWW: http://www.dadisp.com/ Entropic Signal Processing System (ESPS) and Waves * Platform: Range of Unix platforms. * Description: ESPS is a comprehensive set of speech analysis/processing tools for the UNIX environment. The package includes UNIX commands, and a comprehensive C library (which can be accessed from other languages). Waves is a graphical front-end for speech processing. Speech waveforms, spectrograms, pitch traces etc can be displayed, edited and processed in X windows and Openwindows (versions 2 & 3). Waves also includes a signal labelling utility which provides multiple feature labelling and useful features for fast labelling of large speech databases. Other Entropic products are HTK (see Q6.5) and TrueTalk (see Q5.5). * Misc: A more detailed description is provided on the Entropic WWW pages (http://www.entropic.com/esps.html). * Cost: On request. * Contact: Entropic Research Laboratory, Washington Research Laboratory 600 Pennsylvania Ave, S.E. Suite 202, Washington, D.C. 20003 (202) 547-1420 email: info@entropic.com WWW: http://www.entropic.com/ GoldWave * Platform: Windows * Description: GoldWave is a digital audio editor for Microsoft Windows. It features realtime amplitude/spectrum oscilloscopes, large file editing, effects, and support for a wide variety of sound formats. + Editing of multiple waveforms and large waveforms + Realtime amplitude/spectrum oscilloscopes + Resizable device controls window for accessing audio devices + Realtime fast forward and rewind playback + Effects: distortion, Doppler, echo, filter, mechanize, offset, pan, volume shaping, invert, resample, transpose, etc + Multiple file formats and conversions: .WAV, .AU, .IFF, .VOC, .SND, .MAT, .AIFF, and raw data + CD-ROM controls window More information is available on the GoldWave home page. * Cost: Shareware * Availability: Through the GoldWave home page: http://web.cs.mun.ca/~chris3/goldwave/goldwave.html * Contact: Chris Craig: chris3@cs.mun.ca Kay Elemetrics CSL (Computer Speech Lab) 4300 * Platform: Minimum IBM PC-AT compatible with extended memory (min 2MB) with at least VGA graphics. More powerful machines preferable. * Description: Speech analysis package, with optional separate LPC program for analysis/synthesis. Uses its own file format for data, but has some ability to export data as ascii. The main editing/analysis prog (but not the LPC part) has its own macro language, making it easy to perform repetitive tasks. Options - more information on the Kay Elemetrics Corp. WWW site: + Multi-Dimensional Voice Program (MDVP) + Voice Range Profile (Phonetograph) + Real-Time Spectrogram + Sona-Match + Palatometer Database + IPA Transcription Tutorial + Delayed Auditory Feedback (DAF) + Disordered Voice Database + Auditory Perception Program and Database + Motor Speech Profile Program + CSL-Pitch + Real-Time EGG Processing + Signal Enhancement in Noise Program + Synthesis Program + DAT Interface and Four Channel Input + Phonetic Database + Direct-to-Disk Program + Programmers Kit + Condenser Microphone + Multi-Speech * Cost: Contact Kay Elemetrics Corp. * Contact: Kay Elemetrics Corp. 2 Bridgewater Lane, Lincoln Park, NJ 07035, USA Ph: +1-201-628-6200, Fax: +1-201-628-6363 Toll free tel. 1-800-289-5297 [WWW: http://www.kayelemetrics.com/ - available soon] Khoros * Platform: Any Unix - source code available. * Description: Khoros is a technical computing environment for image and signal processing, visual programming and software development. * Price: On request. * Availability: Khoral Research Inc. 6001 Indian School Rd. NE Suite 200, Albuquerque, NM 87110, USA Ph: (505)837-6500, Fax: (505) 881-3842 Email: info@khoral.com ftp: ftp://ftp.khoral.com/ WWW: http://www.khoral.com/ Matlab plus Signal Processing Toolbox * Platform: Wide range * Description: Matlab (MATrix LABoratory) is a technical computing environment for numerical computation and visualization based on a matrix oriented, interpreted programming language. The programming environment provides support for the development of customized operations, along with debugging facilities and a graphical user interface toolkit. Audio output is provided. A specialised Signal Processing Toolbox is available which provides many functions which are useful for speech analysis. It includes filter design, spectral estimation, statistical signal processing, waveform generation, and signal and spectrogram display. A specialised Auditory Toolbox is available which contains functions useful to people interested in auditory/cochlear models. A more detailed description is given in Q1.10. * Price: On request. * Contact: The Math Works Inc. 24 Prime Park Way, Natick, MA 01760-1500 USA Ph: 1-508-653 1415 Fax: 1-508-653 6284 Email: info@mathworks.com ftp: ftp://ftp.mathworks.com WWW: http://www.mathworks.com/ MacSpeech Lab II (MSL II) * Platform: Macintosh * Description: A sound analysis and acquisition for Macs. MSL II delivers the most common functions for speech analysis (FFTs, LPCs, f0 extraction, etc.) & produces grayscale spectrographic displays. Can be used for various speech technology and phonetic training tasks. * Hardware: Requires MacADIOS ("Macintosh Analog/Digital Input/Output System") hardware for speech I/O at 12/16 bits. * Misc: Software no longer updated by GW Instruments; MSL soft/hardware will not perform input/output on Quadras, for example, though analysis seems fine. Known to operate properly on systems as high as IIcx & II fx. * Availability: MSL has been replaced by SoundScope; see the SoundScope entry for more detail. * Contact: GW Instruments 35 Medford Street, Somerville, MA 02143, USA Phone: (617) 625-4096 Fax: (617) 625-1322 N!Power * Platform: SUN, DEC and HP workstations. * Description: An object-oriented software package with a MOTIF GUI interface and a range of functionality for data analysis/editing, signal analysis, speech processing, real-time A/D and D/A, and 2D/3D interactive graphics. N!Power replaces ILS. N!Power can provide a Block Diagram user interface, menus, pop-ups, and a high-level IEEE standard symbolic scripting language. You can customize the blocks, menus and pop-ups with mouse point-and-click operations. * Contact: Signal Technology, Inc. 104 W. Anapamu, Suite J, Santa Barbara, CA 93101-3126 Phone: +1-805-899-8300, Fax: +1-805-899-4344 Email: stisales@signal.com WWW: http://www.silcom.com/~stilarry/ OGI Speech Tools * Developers from the Center for Spoken Language Understanding (CSLU) at the Oregon Graduate Institute of Science and Technology (Portland Oregon) * Platform: Unix * Description: The OGI Speech tools include : + An X windows display tool (LYRE) for displaying data in a time synchronous fashion for a. the speech signal b. spectrograms c. phoneme labels, and other information. + A Neural Network (NOPT) training package. + An set of C library routines (LIBNSPEECH) for the manipulation of speech data, including: a. PLP Analysis, b. Rasta PLP Analysis, c. Linear Predictive Coding, d. Mel Cepstrum Coding, e. Fast Fourier Transform + A set of utilities for converting file formats such as ADC, NIST, mu-law, binary files, and ascii. Includes filtering. + A database utility (find_phone) to automate speech database related enquiries. It allows the user to specify a particular label or set of labels in a given context, display all occurrences of the label, and relabel the occurrences if desired. + A Vector-Quantizer based on the Linde Buzo and Gray (LBG) algorithm. + A set of PERL Scripts which have been used mainly to automate the use of the OGI Speech Tools. + MAN Pages for all routines and programs developed, as well as a User manual in both in postscript and tex format. * Misc: Software is written in ANSI C. * Contact: Email: tools@cse.ogi.edu WWW: http://www.cse.ogi.edu/CSLU/ ftp: ftp://speech.cse.ogi.edu/pub/tools/ Ptolemy * Platform: Sun SPARC, DecStation (MIPS), HP (hppa). * Description: Ptolemy provides a highly flexible foundation for the specification, simulation, and rapid prototyping of systems. It is an object oriented framework within which diverse models of computation can co-exist and interact. Ptolemy can be used to model entire systems. Ptolemy has been used for a broad range of applications including signal processing, telecomunications, parallel processing, wireless communications, network design, radio astronomy, real time systems, and hardware/software co-design. Ptolemy has also been used as a lab for signal processing and communications courses. Ptolemy has been developed at UC Berkeley over the past 3 years. Further information, including papers and the complete release notes, is available from the FTP site. * Cost: Free * Availability: The source code, binaries, and documentation are available by anonymous ftp from ftp://ptolemy.berkeley.edu/pub/README Quadravox Speech Processing Products - Qbox * Platform: Windows 3.1, Windows 95 * Description: Qbox comprises a Windows-based LPC-12 analysis and editing sytem and a parallel-port driven programmer for one-time-programmable TI TSP50P11 synthesis chips. The analysis software utilizes standard 11025Hz, 16bit monaural .wav files for input and allows graphical editing of the coded pitch, gain, and reflection coefficients. It can also be used to define concatenation sequences of individual phrases. Data rates depend on the original sound, but are typically below 2000bits/sec. The processed data can then be merged with synthesis and control routines and programmed into the TI synthesizer. The Quadravox-developed synthesis routine accepts run-time modifications of pitch and frame-length (speed), as well as externally defined concatenation sequences. The synthesis chip interface can be defined as a matrixed-keyboard drive, a simple parallel control, or a serial bus control supporting up to 31 individually addressed devices and modules. * Cost: $90-$150 depending on options selected. * Contact: Quadravox, Inc. 1701 N. Greenville Ave., Suite 608, Richardson, TX, 75081 USA Ph: 214-669-4002 Email: info@quadravox.com WWW: http://www.quadravox.com/ Speech Filing System (SFS) * Platform: Unix and DOS * Description: SFS provides a computing environment for conducting speech research. It comprises software tools, file and data formats, subroutine libraries, graphics, standards and special programming languages. It performs standard operations such as recording, replay, waveform editing and labelling, spectrographic and formant analysis and fundamental frequency estimation. For more information, see ftp://pitch.phon.ucl.ac.uk/pub/sfs/README * Misc: SFS is copyrighted University College London, but is currently supplied free of charge to research establishments for non-profit use. * Availability: SFS source code is available by anonymous FTP from: ftp://pitch.phon.ucl.ac.uk/pub/sfs/ * Contact: Mark Huckvale University College London, Gower Street, London WC1E 6BT, UK Email: SFS@phonetics.ucl.ac.uk ftp: ftp://pitch.phon.ucl.ac.uk/pub/sfs/ Signalyze 3.0 from InfoSignal * Platform: Macintosh * Description: Signalyze is an interactive program for the analysis of speech and other acoustic material. Signalyze's basic concept revolves around the display of up 100 signals in HyperCard fashion. The program offers a range of signal editing features, spectral analysis tools, manual scoring tools, pitch extraction routines, signal manipulation tools, and extensive input-output capacity. It also has a range of capabilities for creating, editing and manipulating label files with flexibility in labelling format. Signalyze handles the following file formats: Signalyze, MacSpeech Lab, AudioMedia, SoundDesigner II, SoundEdit/MacRecorder, SoundWave, sound resource formats, and ASCII-text. Sound I/O: Direct sound input from Apple 8- or 16-bit sound input Sound output via Macintosh 8- or 16-bit sound. * Compatibility: MacPlus and higher. Takes advantage of large screens, multiple screens and 16/256 color/grayscales. System 7.0 compatible. Runs in background with adjustable priority. * Misc: Manuals and tutorials included (250 pp.). Program is switchable to English, French, and German. For more information and demo: WWW: http://www.agoralang.com:2410/pubdirsoftware.html WWW: http://www.agoralang.com:2410/signalyze.html Gopher: gopher://uldns1.unil.ch:70/11/unilgophers/gopher_lett/LAIP * Cost: Individual licence US$450, departmental license US$750, organisational license US$1250, plus shipping. Upgrades from version 2.0 are available. * Contact: The Americas: Network Technology Corporation 91 Baldwin St., Charlestown, MA 02129, USA Phone: +1-617-241-9205, Fax: +1-617-241-5064 --- Elsewhere: InfoSignal Inc. C.P. 73, 1015 LAUSANNE, Switzerland, Fax: +41 21 691-1372, Email: 76357.1213@COMPUSERVE.COM SoundScope * Platform: Macintosh: 68K and PowerPC native * Description: The SoundScope product family is used primarily in speech teaching & research, with some applications in animal sounds, forensics, and general acoustic analysis. It can record, view, analyze, play, copy, paste, store and print sound waveforms. Analysis functions include spectrogram, fundamental frequency (Fo), Linear Predictive Coding (LPC) including formant tracking, LPC residual, jitter (pitch perturbation), shimmer (amplitude perturbation), HNR, frequency spectrum, spectral slice, envelope, energy and zero crossing. Includes limited built-in filtering, runs any filter created with WLFDAP. An integrated text editor stores notes and calculation results. SoundScope lets you design your own custom "instrument" screen, tasks (macros) and menus. Supplied instruments include 1 channel analyser (dual snap, dual time, spectrogram, spectrum), 2 channel analyser, segment analyser, multi-channel recorder, etc. * Note: Supercedes MacSpeech Lab II. * Price: $490 to $4990, less educational discount * Availability: In North America, directly from GW Instruments. Contact the company for international distributors. * Contact: GW Instruments 35 Medford Street, Somerville, MA 02143, USA Ph: +1-617-625-4096, Fax: +1-617-625-1322 Email: info@gwinst.com ___________________________________________________________________________ Q1.10: Speech Research Sites Rather than try to list the places round the world which perform speech research this FAQ lists sites on the WWW where other comprehensive lists are maintained. Try the following: Shikano's WWW site on Speech and Acoustics http://www.aist-nara.ac.jp/IS/Shikano-lab/database/internet-res ource/e-www-site.html Lists of speech research sites by country. Currently includes around 100 sites. The list of Japanese sites is particularly comprehensive. Mambo Speech Research List http://mambo.ucsc.edu/psl/speech.html Lists about 50 speech research sites and related information sources. Very nice presentation! ESCA: European Speech Communication Association http://ophale.icp.grenet.fr/esca/labos.html Links to around 15 European speech research sites and around 15 related sources of information. Institute for Perception Research: Speech on the Web http://www.tue.nl/ipo/hearing/webspeak.htm Jan Roelof de Pijper at the Institute for Perception Research has a long list of research sites plus links to lots of other speech material on the WWW. Russ Wilcox's list of Commercial Speech Recognition http://www.tiac.net/users/rwilcox/speech.html Links to information on speech technology vendors, speech research labs, speech resources, on-line demos and more. Speech Groups List: Leeds University Cognitive Psychology Research Group http://lethe.leeds.ac.uk/research/cogn/speechlab/other.html List of about 25 research sites. Institute of Phonetic Sciences, Amsterdam http://fonsg3.let.uva.nl/Other_pages.html#Phonetics Good list of European sites. Speech and Hearing Research Group, University of Sheffield, UK http://www.dcs.shef.ac.uk/research/groups/spandh/world/misclink s.html Links to sites in the UK, USA, Europe and the rest of the world. Duncan M. Forrest's Speech Recognition Resource List http://www.skye.co.za/dmf/speech/ Most speech research sites have links to other speech research sites somewhere in their WWW pages. ___________________________________________________________________________ Q1.11: Miscellaneous Software and Resources. Speech Interface Standards: APIs etc * ASAPI: Advanced Speech API (AT&T) * SAPI: Microsoft Windows Speech API * SRAPI: Speech Recognition API * TAPI: Microsoft Windows Telephony API Network "Phone" Software * CUSeeMe * CyberPhone * DigiPhone * InterFACE from Hijinx * FAQ: How can I use the Internet as a telephone? * Nautilus: Secure Computer Telephony * NEVOT (1.4v) from AT&T BL * PGPfone * Speak Freely * Internet Phone from VocalTec * WebPhone * WebTalk Audio Processing Software * AF version AF3R1 * Voice E-Mail from Bonzi Software * MicNotePad Recording Software for Macs * MixViews * Network Audio System Release 1.1 * NIST Software - SPHERE and SCORE * Sound Processing Kit * TCPplay Human Audio Perception Other useful information on Auditory Modeling can be found in Malcolm Slaney's home page http://www.interval.com/~malcolm/ Martin Cooke's home page Speech and Hearing Research Group, Dept of Computer Science, University of Sheffield, UK. http://www.dcs.shef.ac.uk/~martin/ * Auditory Modeller 1 * Auditory Modeller 2 * Auditory Toolbox for Matlab * Human Audio Perception Document Dictionaries and other Lexical Tools * BEEP dictionary * CMU dictionary * CUVOLAD dictionary (Oxford Dictionary) * Comprehensive Word List * EAT: Edinburgh Associative Thesaurus * Homophone List * Moby Lexical Resources * MRC Psycholinguistic Database * WordNet * Dictionaries on the WWW Phonetic Fonts and Phonetic Samples * International Phonetic Alphabet * WWW: Phonetic Fonts and Examples Online * Summer Institute of Linguistics IPA Fonts * Phonetic Fonts for TeX and LaTeX * Yamada Language Center Subjective Evaluation of Speech Quality Dynastat, Inc. Speech Intelligibility Testing with Diagnostic Rhyme Test (DRT), Modified Rhyme Test (MRT), Phonetically Balanced Word Lists (PB), Diagnostic Medial Consonant Test (DMCT), Diagnostic Alliteration Test (DALT), ICAO Spelling Alphabet Test (SpAT) Speech Quality (Acceptability) Evaluation with Diagnostic Acceptability Measure (DAM), Mean Opinion Score (MOS), Degredation Mean Opinion Score (DMOS) Contact: Dynastat, Inc. 2704 Rio Grande, Suite 4, Austin, TX 78705, USA Ph: +1-512-476-4797, Fax: 512/472-2883 Email: sharpley@dynastat.com WWW: http://www.bga.com/dynastat/ ANSI S3.2-1989: American National Standard for Measuring the Intelligibility of Speech Over Connunication Systems Available from American National Standards Institute (ANSI) Ph: +1-212-642-4900, Fax: +1-212-398-0023 WWW: http://www.ansi.org/ Louis Pols' List of References on Synthesis Development And Assessment 700 references: http://www.itl.atr.co.jp/cocosda/output/synth.refs Very Miscellaneous * The vOICe * The Learning Company's Language Training * Wildfire - an Electronic Assistant ASAPI: Advanced Speech API (AT&T) * Description: The AT&T ASAPI Specification is a open, cross-platform, easy-to-use speech API that can support speech engines from AT&T and other vendors. ASAPI does not replace the Microsoft Speech API, but it provides extensions and enhancements to the Microsoft SAPI Specification including support for SAPI-compatible applications. The ASAPI Specification defines two types of interfaces. The "ASAPI Extensions" interface which provides extensions to the MS-SAPI interface as well as C++ class encapsulation of SAPI functionality. The "Visual ASAPI" interface provides an even higher-level abstraction of SAPI/ASAPI low-level functionality such that application developers can quickly and easily embed speech technology into existing or new applications. Special Purpose Recognizers are examples of Visual ASAPI interfaces which integrate lower-level functionality that an application developer can access via a simple interface. * More information: Contact Jose Garcia at AT&T on (908) 957-5457 or by email: jrg@att.com. For more information on the WATSON Speech Engine which supports ASAPI and news about ASAPI please visit the AT&T Advanced Speech Products Group home page or call 1-800-5-WATSON. SAPI: Microsoft Windows Speech API * Platform: Windows 95 and Windows NT 3.51 * Description: The Microsoft Speech API provides applications with the ability to incoporate speech recognition (command & control or dictation) or text-to-speech, using either C/C++ or Visual Basic. SAPI follows the OLE Component Object Model (COM) architecture. It is supported by many major speech technology vendors. The major interfaces are + Voice Commands: high level speech recognition API for command and control. + Voice Text: simple high level text-to-speech API. + Speech Recognition: provides detailed control of a speech recognition engine for both command-and-control and dictation. + Text-to-Speech: provides detailed interface to a text-to-speech engine for control of playback, speaking style, voice quality etc. + Multimedia Audio Objects: audio I/O for microphones, headphones, speakers, telephone lines, files etc. * Availability: Download Microsoft's latest speech technology, including the Microsoft Speech SDK, command and control recognition, the Microsoft dictation research demonstration and text-to-speech. * More information: Email: MSSpeech@Microsoft.Com WWW: The Microsoft Speech API WWW: An Overview of the Microsoft Speech API Documentation included with the Microsoft SDK. * See also: TAPI: Microsoft Telephone API SRAPI: Speech Recognition API * Platform: Various * Description: The SRAPI provides support for speech recognition, text-to-speech and other media playback. The SRAPI Committee is a nonprofit Utah corporation with the goal of providing solutions for interaction of speech technology with applications. Core members include: Novell, Inc., Dragon Systems, IBM, Kurzweil AI, Intel, and Philips Dictation Systems. Additional contributing members include Articulate Systems, DEC, Kolvox Communications, Lernout and Hauspie, Syracuse Language Systems, Voice Control Systems, Corel, Verbex and Voice Processing Corporation. * More information: WWW: http://www.srapi.com/ Email: For more information on the SRAPI Developer CD, send email to srapi@srapi.com with Subject "SRAPI CD Info". TAPI: Microsoft Windows Telephony API * Description: TAPI allows applications to support telephone communication. TAPI facilitates include: + Connecting directly to a telephone network. + Automatic phone dialing. + Transmission of data (files, faxes, electronic mail). + Access to data (news, information services). + Conference calling. + Voice mail. + Caller identification. + Control of a remote computer. + Collaborative computing over telephone lines. Windows 95 comes with a telephony application, DIALER.EXE, that can dial voice calls, act as a proxy for applications making simple telephony requests, and maintain a call log. * More information: The Win32 Software Development Kit (SDK) contains documentation, tools, and sample code for TAPI including the Microsoft Telephony Programmer's Reference and the Microsoft Telephony Service Provider Interface (TSPI) for Telephony. WWW: Tapping in TAPI, TAPI White Paper * See also: SAPI: Microsoft Speech API CUSeeMe * Platform: Macintosh and Windows * Description: Cornell University software for audio and video conferencing over the Internet. * Requirments: Macintosh to RECEIVE video: + Macintosh platform with a 68020 processor or higher + System 7 or higher operating system + Minimum 16-level-grayscale (e.g. color) + IP network connection and MacTCP + Apple's QuickTime, to receive slides with SlideWindow Macintosh to SEND video: + All the above plus + Quicktime installed + video digitizer (with vdig software) and Camera For Windows: + Video receive only 386SX, Video send & receive 386DX, Video receive w/Audio 486SX, Video send & receive w/Audio 486DX + Windows 3.1 or higher running in Enhanced Mode. + Winsock + 256 color (8 bit) video driver + Video camera and a video capture board that supports Microsoft Video For Windows + For audio: Windows Sound board that conforms to the Windows MultiMedia Specification, speakers and a microphone * Availability: Mac: http://cu-seeme.cornell.edu/get_cuseeme.html Windows: http://cu-seeme.cornell.edu/PC.CU-SeeMeCurrent.html * More information: http://cu-seeme.cornell.edu/ CyberPhone * Platform: Sun Workstations running Solaris 2.x (SunOS 5.x) * Description: Provides voice communications over the internet. Has a graphical user interface and requires no additional hardware. An optional centralized server system is available to make finding and connecting to other users easier. * Availability: a free demonstration is available by anonymous ftp ftp://magenta.com/pub/cyberphone * Contact: Email: cyberphone@magenta.com. More information is available on the WWW: http://magenta.com/cyberphone/. DigiPhone * Platform: Macintosh, Windows 3.1 and Windows 95 * Description: DigiPhone provides two-way phone conversations by dialing direct and over the Internet. Includes encryption for privacy, caller ID, call screening, call timer, adjustable sound and compression quality, messaging, and access to the Global Directory providing a database of DigiPhone users. + DigiPhone v1.03: provides the standard features listed above. [ More information]. + DigiPhone Deluxe: provides the standard features of DigiPhone v1.03 and adds conference calling, mute, speed dial, call recording and playback, voice effects, customizations, and internet tools. [ More information]. + DigiPhone for Mac: provides the standard features listed above, plus cross-platform compatibility and mute. [ More information]. * Requirements: DigiPhone v1.03 requires 386DX/33 or faster, 4MB RAM, 9,600 bps modem, Sound Blaster 16 card (or any compatible half or full duplex card), and a local internet connection with SLIP or PPP. [Recommend 486DX/33 and 14,400 bps modem] DigiPhone Deluxe has the same requirements on v1.03 but requires 486DX/33 or faster. DigiPhone for Mac requires a 68030 33Mhz, 68040 25Mhz or Power PC, 4 MB RAM, System 7.x, 14,400 bps modem or better, Sound Manager 3.x for System 7, microphone and speakers, MacTCP or Open Transport and a local internet connection with SLIP or PPP. * Price and Availability: Contact Third Planet Publishing for pricing. Trial software is available from Third Planet Publishing. Orders and Upgrades can be made on the Web. Also available through many retailers. * Contact: Third Planet Publishing, Inc. 17770 Preston Rd, Dallas, Texas 75252, USA Ph: +1-972-733-3005, Fax: +1-972-380-8712 Email: 3pp@planeteers.com WWW: http://www.planeteers.com/ InterFACE from Hijinx * Platform: Windows * Description: InterFACE provides voice communication on the Internet through IRC (Internet Relay Chat) services. * Requirments: Recommend a 486DX, 8meg Ram, Windows, VGA Monitor and a 16 bit sound card. * Availability: Available on CD Only for $60.00 US, which includes, postage and handling. Demo versions available from the HiJiNX WWW site. * Contact: HiJiNX, Brisbane, Australia Email: jester@hijinx.com.au WWW: http://www.hijinx.com.au/ FAQ: How can I use the Internet as a telephone? * Description: Kevin M. Savetz and Andrew Sears have prepared an FAQ document titled _FAQ: How can I use the Internet as a telephone?_ The current document has the following sections: + Can I use the Internet as a telephone? + What do I need to call others on the Internet? + How does it work? + How do I make calls using a modem? + Is the sound quality as good as a regular telephone? + Is there a noticeable delay in hearing the other user? + What is the difference between full duplex and half duplex? + What is multicasting? + Can I talk to users of other phone software? + What software is available? The section on available software covers the following: + Mac: Maven, NetPhone, CU-Seeme, PGPfone + Windows: Speak Freely, CU-Seeme, Internet Phone, Digiphone, Internet Voice Chat, Internet Global Phone, Web Phone + UNIX: Speak Freely, nevot, vat, mtalk, ztalk * Availability: By Email Mail voice-faq-request@northcoast.com with "Subject: archive" and "Body: send voice-faq" FTP ftp://rtfm.mit.edu/pub/usenet/alt.internet.services/FAQ:_ How_can_I_use_the_Internet_as_a_telephone? WWW: http://rpcp.mit.edu/~asears/voice-faq.html * Contact: Andrew Sears: asears@mit.edu Kevin Savetz: savetz@northcoast.com Nautilus: Secure Computer Telephony * Platform: DOS, Linux, SunOS, Solaris. * Description: Nautilus is software which allows two users to hold a secure conversation with either over ordinary phone lines or over a computer network. Nautilus uses your computer's audio hardware to digitize and play back your speech using speech compression algorithms built into the program. It encrypts the compressed speech using your choice of the Blowfish, Triple DES, or IDEA block ciphers, and transmits the encrypted packets over the internet or your modem to another computer. At the other end, the process is reversed. Nautilus operates in half duplex mode like a speakerphone -- only one person can talk at a time. Either user can hit a key to switch between talking and listening. Audio quality ranges from fair to very good depending on which of the four speech coders is selected. The Nautilus WWW page provides more detailed information. * Requirements: Nautilus runs on IBM PC-compatible computers (386DX25 or faster) under MSDOS or Linux as well as audio-capable Sun workstations running SunOS or Solaris. The MSDOS version of Nautilus requires a Soundblaster compatible sound card and currently only runs over ordinary phone lines with a modem. To use Nautilus over ordinary telephone lines, a modem capable of connecting at 4800 bps or faster is required. * Availability: Nautilus is available in three different formats. As a DOS executable, it is available as an archive in zip format along with it's associated documentation. In source format, it is available as either a zip-ed archive, or a gzip-compressed tar archive. Nautilus is distributed freely (subject to US export restrictions) with full source code. This insures that its security can be independently examined and verified. Follow the instructions in the following README files to obtain Nautilus. + ftp://ftp.csn.org/mpj/README + ftp://ripem.msu.edu/pub/crypt/README * More information: WWW: http://www.lila.com/nautilus/ * Contacts: The Nautilus development team includes Bill Dorsey, Paul Rubin, Andy Fingerhut, Paul Kronenwetter, Bill Soley, and Pat Mullarky. To contact the developers, send email to nautilus@lila.com. NEVOT (1.4v) from AT&T BL * Platforms: Sun Sparc Station (SunOS 4.1.x) and Silicon Graphics * Description: Audio-conferencing tool which supports both point-to-point and broadcasting of audio using multicast IP. Audio encoding: + PCM 64kb/s 8-bits u-law encoded 8KHz PCM (G.711) + ADPCM 32 kb/s [Sun only] (G.721) + DVI ADPCM 32 kb/s + ADPCM 24 kb/s [Sun only] (G.723) + CELP 4.8 kb/s + LPC 2.4 kb/s * Availability: by anonymous ftp from ftp://gaia.cs.umass.edu/pub/hgschulz/nevot * Contact: Henning Schulzrinne (hgs@researh.att.com) PGPfone * Platform: Macintosh and Windows * Description: Pretty Good Privacy Phone is free secure audio connection software for the internet. It uses speech compression and strong cryptography protocols to give you the ability to have a real-time secure telephone conversation via a modem-to-modem connection. * Requirements (Mac): Fast modem: at least 14.4 Kbps V.32bis (28.8 Kbps V.34 recommended). An Apple Macintosh with at least a 25MHz 68LC040 processor (PowerPC recommended), running System 7.1 or above, Thread Manager 2.0.1, ThreadsLib 2.1.2, and Sound Manager 3.0. (These are available from Apple's FTP sites.) * Requirements (Windows): Fast modem: at least 14.4 Kbps V.32bis (28.8 Kbps V.34 recommended). A multimedia PC running Windows 95 or NT, with at least a 66 MHz 486 CPU (Pentium recommended), sound card, microphone, and speakers or headphones. * Contact: Jeffrey I. Schiller Email: jis@mit.edu WWW: http://web.mit.edu/network/pgpfone/ Speak Freely * Platform: Windows and Unix * Description: Free "Internet Phone" software supporting voice mail, multicasting, encryption and several coding methods. Includes 4 forms of data compression and encryption with DES, IDEA and PGP. The Windows and Unix versions are compatible. You can designate a bitmap file to be sent to users who connect so they can see who they're talking to. The Unix version does not have the graphical user interface of the Windows edition, but supports all its compression and encryption modes. * More information: http://www.fourmilab.ch/netfone/windows/speak_freely.html Internet Phone from VocalTec * Platforms: IBM Compatible * Description: Supports real-time conversations with Internet users by compressing speech. Voice-activation feature and interactive display. Features an graphical interface and on-line help. Up to date listing of all on-line users running Internet Phone. Join or create topics for conversation with people from all over the globe. Supports private topics for private conversations with family or with business associates. * Requirements: 486SX PC - 25 MHZ, 8MB RAM (recommended) An Internet Winsock 1.1 compatible TCP\IP connection (minimum connection: a 14,400 baud modem SLIP\PPP connection) Windows 3.1 Windows-compatible sound card * Cost: $US59 + shipping. You can order on the internet: http://www.vocaltec.com/order.html * More Information: WWW: http://www.vocaltec.com/ * Availability: Demo version: ftp://ftp.vocaltec.com/pub/iphone09.exe * Contact: VocalTec Inc. 157 Veterans Drive, Northvale, NJ 07647 Tel: 201-768-9400 Fax: 201-768-8893 E-mail: info@vocaltec.com WebPhone * Platform: Windows * Description: WebPhone provides telephone quality, real-time, full duplex, encrypted, point-to-point voice communication over the Internet and other TCP/IP based networks. (More detail provided on the NetSpeak WWW pages). * Requirements: 80486DX-33 MHz running Windows 3.1 or higher, 4 MB of RAM, MCI compliant sound card, Winsock 1.1 compliant stack, 14.4Kbps modem, VGA card capable of displaying 256 colors. Full duplex audio card required for full duplex. * Price: $49.95 (US) * Availability: via the WWW: http://www.netspeak.com/getphone.html * Contact: NetSpeak Corporation 902 Clint Moore Rd., Boca Raton, Fl. 33487, USA Ph: +1-407-997-4001, Fax: +1-407-997-2401 Email: info@netspeak.com WWW: http://www.netspeak.com/ WebTalk * Platform: Windows 3.1/95 * Description: Full-duplex or half duplex, telephone-quality voice, supports many commercial web browsers. * Contact: Quarterdeck Corporation 13160 Mindanao Way, 3rd Floor, Marina Del Rey, CA 90292-9705, USA Ph: +1-310-309-3700, Fax: +1-310-309-4217 Email: info@quarterdeck.com WWW: http://www.quarterdeck.com/ AF version AF3R1 * Platforms: DEC workstations (Alpha and MIPS), SparcStation, SGI * Description: The AF System is a device-independent network-transparent system including client applications and audio servers. With AF, multiple audio applications can run simultaneously, sharing access to the actual audio hardware. The AF3R1 distribution of AF includes server support for Digital RISC systems running Ultrix, Digital Alpha AXP systems running OSF/1, SGI Indigo running IRIX 4.0.5, Sun Microsystems SPARCstations running SunOS 4.1.3, and Sun Microsystems SPARCstations running Solaris 2.3. The servers support audio hardware ranging from the built-in CODEC audio on SPARCstations and Personal DECstations to 48 KHz stereo audio using the DECaudio TURBOchannel module or the SPARCstation DBRI interface * Availability: The source kit is distributed by anonymous ftp from ftp://crl.dec.com/pub/DEC/AF WWW: http://www.research.digital.com/CRL/projects/AF/home.html * Contact: af-request@crl.dec.com Voice E-Mail from Bonzi Software * Description: Voice E-Mail is an extension to regular e-mail which allows recorded voice messages to be transmitted in the same way as normal text messages. Voice E-Mail is available in several forms: Voice E-Mail 3.0 for WinCIM, Voice E-Mail 3.0 for America Online, Voice E-Mail 3.0 for Eudora, and Voice E-Mail 3.0 for Netscape. Voice E-Mail uses digital audio and image compression technology to compress messages before transferring them through CompuServe, America Online, and the Internet. * Availability: Go to the Bonzi home page - http://www.bonzi.com/ - and follow the links to the Internet Shopping Network's "Downloadable Software Division." * Further Information: Bonzi Software WWW: http://www.bonzi.com/ Email: info@bonzi.com Fax 805-238-5798 MicNotePad Recording Software for Macs * Platforms: Macintosh * Description: MicNotePad is audio recording tool designed to improve dictation (a digital replacement for the old-style mechnical tape systems used by typists). It uses the built-in microphone or sound input port and the hard disk to record conversations or speech of arbitrary length. Speech compression techniques are used to reduce the disk-space. Once it is recorded, single keystrokes control playback while you type in your word processor. * Contact: Nirvana Research WWW: http://moof.com/nirvana/ Email: nirvana@got.net MixViews * Description: A Unix/X sound editor. Does waveform play/record, and cut/splice. Has various filters, handles native file formats, FFT, LPC and more * Availability: by anonymous ftp including SunOS 4 and IRIX 5 binaries. ftp://foxtrot.ccmrc.ucsb.edu/pub/MixViews Network Audio System Release 1.1 * Platforms: Various (includes SunOS, Solaris, SGI) * Description: A device-independent mechanism for transferring, playing and recording audio signals over a network. Has a range of features suited to networks. * Cost: Free * Availability: By anonymous ftp from ftp://ftp.x.org:/contrib/audio/nas/netaudio-1.2.tar.gz Also available in the same directory are document files and some sample sounds. NIST SPeech HEader REsources Package (SPHERE) * Description: Standard speech header software from the National Institute of Standards & Technology (NIST). SPHERE headers represent information about sample frequency, sample format, etc. * Availability: By anonymous ftp from Readme File ftp://jaguar.ncsl.nist.gov/pub/sphere.README Source Code ftp://jaguar.ncsl.nist.gov/pub/sphere_2.5.tar.Z NIST Speech Recognition Scoring Package (SCORE) * Description: Software for scoring results of speech recognition systems from the National Institute of Standards & Technology (NIST) . * Availability: By anonymous ftp from README File ftp://jaguar.ncsl.nist.gov/pub/score.README Source Code ftp://jaguar.ncsl.nist.gov/pub/score_3.6.2.tar.Z Sound Processing Kit * Platforms: UNIX * Description: Sound Processing Kit (SPKit) is an object-oriented class library for audio signal processing. SPKit includes classes for various signal processing tasks and a way of implementing sound processing algorithms in a simple object-oriented manner. Sound Processing Kit is implemented in C++ and is designed to be portable. The current version requires a bare-bones C++ 2.0 compatible compiler (templates and exceptions are not needed). ANSI C standard libraries are required. SPKit includes classes for + Sound input and output + Basic signal processing + Dynamics processing (compressor, gating etc) + Filtering + Delay and reverberation + Distortion + Signal routing * Availability: Full documentation on the WWW: http://www.music.helsinki.fi/research/spkit/documentation /SPKit.html Software distribution: http://www.music.helsinki.fi/research/spkit/distribution/ spkit.tar.Z * Contact: Kai Lassfolk University of Helsinki Music Research Laboratory Email: spkit@elisir.helsinki.fi TCPplay * Description: TCPPlay lets you use your mac as an audio server for your Unix box. Provided with source code. Written by Bill Stafford, Rich Tsoi and Malcolm Slaney. * Availability: Anonymous ftp from ftp://ftp.apple.com/pub/malcolm/TcpPlay.sit.hqx ftp://worldserver.com/pub/malcolm/TcpPlay.sit.hqx Auditory Modeller 1 * Description: John Holdsworth's implementation of a gammatone filter bank and Roy Patterson's spiral model, in C (with X-window display). * Availability: By anonymous ftp from ftp://ftp.mrc-apu.cam.ac.uk/pub/aim Auditory Modeller 2 * Description:Lowel O'Mard's implementation of peripheral filtering, Ray Meddis's hair cell model and other stuff in C (as a library of routines). * Availability: By anonymous ftp from ftp://suna.lut.ac.uk/public/hulpo/lutear Auditory Toolbox for Matlab * Description: This toolbox provides extensions to Matlab which are useful to people interested in auditory/cochlear modeling. [Matlab is described is the previous section.] This toolbox has been tested on both Macintosh and Unix computers. It includes the following major models: + Lyon's Passive Long Wave Cochlear Model (our conventional model) + Patterson-Holdsworth ERB Filter bank with Meddis Hair cell + Seneff's Auditory Model (Stages I and II) + MFCC (Mel-scale frequency cepstral coefficients from the ASR world) + Spectrogram + Correlogram generation and pitch modeling + Simple vowel synthesis * Availability: From Malcolm Slaney home page and by anonymous FTP: ftp://ftp.apple.com/pub/malcolm The following files are available: + AuditoryToolbox.mif.Z + AuditoryToolbox.psc.Z + AuditoryToolbox.sea.hqx + AuditoryToolbox.tar + AuditoryToolbox.tar.Z The ".mif.Z" file is a Unix compressed version of the FrameMaker documentation. The ".psc.Z" file is a Unix compressed version of the Postscript documentation. The ".tar" and ".tar.Z" files are Unix TAR archives containing all of the m-functions and C-MEX source code. Finally, the ".sea.hqx" file is a Macintosh self-extracting archive that has been encoded using BinHex. There is precompiled version of the three MEX function for the Macintosh. * Misc: Our lawyers ask you to remind you that there is no warranty. We've done some testing but we undoubtably missed things. * Contact: Malcolm Slaney, Interval Resarch. Email: malcolm@interval.com WWW: http://www.interval.com/~malcolm/ Human Audio Perception Document * Description: Document prepared by Argiris Kranidiotis on the human audio perception system. It lists a number of references, gives plenty of numbers and some equations. * Availability: by anonymous ftp from the comp.speech archive site ftp://svr-ftp.eng.cam.ac.uk/comp.speech/info/HumanAudioPe rception * Contact: Argiris A. Kranidiotis University Of Athens, Informatics Department email: akra@zeus.di.uoa.ariadne-t.gr BEEP dictionary * Description: Phonemic transcriptions of over 250,000 English words. (British English pronunciations) * Availability: By anonymous ftp: BEEP dictionary README file svr-ftp.eng.cam.ac.uk/comp.speech/dictionaries/beep-0.7.R EADME BEEP Dictionary (1.1M) svr-ftp.eng.cam.ac.uk/comp.speech/dictionaries/beep.tar.g z CMU dictionary * Description: Phonemic transcriptions of 100,000 words with American English pronunciation. * Availability - WWW: http://www.speech.cs.cmu.edu/cgi-bin/cmudict * Availability - ftp: By anonymous ftp from the directory ftp://ftp.cs.cmu.edu/project/fgdata/dict/ with the files README, cmudict.0.2.Z, cmulex.0.1.Z, phoneset.0.1 CUVOLAD dictionary (Oxford Dictionary) * Description: Computer Usable Version of the Oxford Advanced Learner's Dictionary containing 70,000+ entries. Has British English pronunciations and parts of speech. * Availability: Anonymous ftp ftp://ota.ox.ac.uk/pub/ota/public/dicts/710/ Documentation: ftp://ota.ox.ac.uk/pub/ota/public/dicts/710/text710.doc Comprehensive Word List * Description: A comprehensive word list which should contain most common American words, abbreviations, hyphenations, and even incorrect spellings. The word lists were compiled from a number of sources: commercial news services, UseNet news postings, existing dictionaries, name lists, company lists, UNIX man pages, project Gutenberg's E-texts, project Wordnet, received mailings, etc. The current size is 460,000 words. * Availability: anonymous ftp ftp://wocket.vantage.gte.com/pub/standard_dictionary Note 1: There seems to be some sort of network problem reaching the server. Note 2: There is a README file which explains the file formats. EAT: Edinburgh Associative Thesaurus * Description: A set of word association norms showing the counts of word association as collected from subjects. * Availability: Source and WWW interactive versions Interactive version Provided by Computing and Information Systems Department (CISD) of Rutherford Appleton Laboratory, UK http://www.cis.rl.ac.uk/proj/psych/eat.html Set of word association norms ftp directory. 6 MB http://www.cis.rl.ac.uk/proj/psych/eat/eat/ Homophone List * A list of homophones in General American English is available by anonymous FTP from the comp.speech archive site: ftp://svr-ftp.eng.cam.ac.uk/comp.speech/dictionaries/homo phones-1.01.txt Moby Lexical Resources * Description: A set of lexical resources compiled by Grady Ward. 3449 Martha Ct., Arcata, CA 95521-4884, USA Email: grady@netcom.com OR grady@northcoast.com * Availability: Mirrored by Malcolm Crawford (m.crawford@dcs.shef.ac.uk) at the Institute for Language Speech and Hearing, the University of Sheffield. WWW: http://www.dcs.shef.ac.uk/research/ilash/Moby/ FTP: ftp://ftp.dcs.shef.ac.uk/share/ilash/Moby/ * Contents: Moby Hyphenator: mhyph.tar.Z 185,000 entries fully hyphenated. 980kB. Moby Language: mlang.tar.Z Word lists in five major languages. 2.3MB. Moby Part-of-Speech: mpos.tar.Z 230,000 entries with part(s) of speech listed in priority order. 1.2MB. Moby Pronunciator: mpron.tar.Z 175,000 entries fully International Phonetic Alphabet coded. 3.1MB. Moby Shakespeare: mshak.tar.Z The complete unabridged works of Shakespeare. 2.3.MB. Moby Thesaurus: mthes.tar.Z 30,000 root words, 2.5 million synonyms and related words. 12MB. Moby Words: mwords.tar.Z 610,000+ words and phrases. 4.0MB. MRC Psycholinguistic Database * Description: A machine usable dictionary containing over 150000 words with up to 26 linguistic and psycholinguistic attributes for each (e.g. pronunciation, part of speech, word frequency). Psycholinguistic Database was the basis for the "Oxford Psycholinguistic Database" available for Apple Macs from Oxford University Press. * Availability: Several versions with different formats: Interactive Version of MRC Psycholinguistic Database Produces lists of words meeting user-definable selection criteria. Provided by the Dept. of Psychology, University of Western Australia. http://www.psy.uwa.edu.au/uwa_mrc.htm ftp'able MRC Psycholinguistic Database Approximately 12M of data. ftp://ota.ox.ac.uk/pub/ota/public/dicts/1054/ README: ftp://ota.ox.ac.uk/pub/ota/public/dicts/1054/readme. Information: ftp://ota.ox.ac.uk/pub/ota/public/dicts/info WordNet * Description: WordNet is an on-line lexical reference system in which English nouns, verbs, adjectives and adverbs are organized into synonym sets, each representing one underlying lexical concept. Different relations link the synonym sets. WordNet was developed in the Cognitive Science Laboratory at Princeton University under the direction of Professor George Miller. * Availability: WWW Interface http://www.cogsci.princeton.edu/~wn/w3wn.html Source Distributions Unix (9.1MB), PC (5.8MB), Macintosh (7.5MB), Prolog (database only, 4.2MB). ftp://clarity.princeton.edu/pub/wordnet/ Extended interfaces developed by WordNet users (for X, Lisp etc) are listed in the WordNet home page. * Further information: Email: wordnet@princeton.edu WWW: WordNet home page: http://www.cogsci.princeton.edu/~wn/ README: ftp://clarity.princeton.edu/pub/wordnet/README Publications: ftp://clarity.princeton.edu/pub/wordnet/5papers.ps Dictionaries on the WWW For a while, there was a range of dictionaries and other lexical resources on the WWW and elsewhere on the Internet. However, due to copyright reasons, fewer sites are publishing dictionary information. When last checked, the following sites provide dictionaries or links to dictionaries on the net: CMU Dictionary http://www.speech.cs.cmu.edu/cgi-bin/cmudict Institute of Phonetic Sciences, Amsterdam Electronic dictionaries, including French, Norwegian Swahili and English. http://fonsg3.let.uva.nl/Other_pages.html 1913 Webster's Revised Unabridged Dictionary Available as a searchable HTML form at the University of Chicago ARTFL project site, and as a tagged working file and downloadable version (45MB) of the HTML at Project Gutenberg. Martin Ramsch's Englisch-Worterbucher aller Art Lists of on-line dictionaries, translation dictionaries, technical dictionaries, etc. http://www.uni-passau.de/forwiss/mitarbeiter/freie/ramsch/engli sch.html Galaxy's list of dictionaries etc. A comprehensive list of dictionaries, acronym lists, translation resources, and a Thesaurus. http://galaxy.einet.net/galaxy/Reference-and-Interdisciplinary- Information/Dictionaries-etc.html Webster's dictionary online http://c.gp.cs.cmu.edu:5103/prog/webster International Phonetic Alphabet * Description: The International Phonetic Association (http://www.arts.gla.ac.uk/IPA/ipa.html) defines the International Phonetic Alphabet. It is a standard set of symbols for transcribing the sounds of spoken languages. The full chart of IPA symbols is published on the International Phonetic Association WWW site. Also provided are charts for consonants, vowels, tones and accents, suprasegmentals, diacritics and other symbols. A cassette of sounds is available: see http://www.phon.ucl.ac.uk/home/wells/cassette.htm WWW: Phonetic Fonts and Examples Online George L. Dillon's list of phonetic resources [http://weber.u.washington.edu/~dillon/PhonResources.html] Vowel sounds of American English Examples of standard American vowels along with the IPA phonetic symbols and links to recordings. http://weber.u.washington.edu/~dillon/vowels.html Consonant sounds of English Examples of consonants along with the IPA phonetic symbols and links to recordings. http://weber.u.washington.edu/~dillon/consonants.html Vowel Quadrilaterals for American and British English Charts and audio. http://weber.u.washington.edu/~dillon/newstart.html IPA-ASCII A scheme for representing IPA transcriptions in ASCII for use in Usenet articles and email. http://weber.u.washington.edu/~dillon/ipaascii.html Some things about studying Speech Information on speech physiology, acoustic phonetics, speech perception, speech recognition and voice recognition. http://www.ccp.uchicago.edu/grad/Francis_Alex/speech.html Summer Institute of Linguistics IPA Fonts * Platform: Apple Macintosh and Mircosoft Windows * Description: International Phonetic Alphabet (IPA) fonts are available as freeware from the Summer Institute of Linguistics (SIL). The SIL Encore IPA Fonts are a set of scalable IPA fonts containing the full International Phonetic Alphabet with 1990 Kiel revisions. Three typefaces are included: SIL Doulos (similar to Times), SIL Sophia (similar to Helvetica), and SIL Manuscript (monowidth). Each font contains all the standard IPA discrete characters and non-spacing diacritics as well as some suprasegmental and punctuation marks. Each font comes in both PostScript Type 1 and TrueType formats. * Availability: Via the WWW and Gopher: + WWW: http://www.sil.org/ + Gopher: gopher://gopher.sil.org/11/gopher_root/computing/software/fon ts/ + Ftp for Windows: ftp://ftp.sil.org/fonts/win/silip12a.exe + Ftp for Mac: ftp://ftp.sil.org/fonts/mac/silipa12.sea_hqx Also available through the SIL email server. Send either of the following commands to MAILSERV@sil.org. Windows: SEND/MODE=BLOCK/ENCODING=UUENCODE [FTP.FONTS.WIN]SILIP12A.EXE Mac: SEND [FTP.FONTS.MAC]SILIPA12.SEA_HQX Finally, they are available on diskette from the address below. $US5 to cover the cost of shipping. * Contact: International Academic Bookstore Summer Institute of Linguistics 7500 W. Camp Wisdom Road, Dallas, TX 75236 U.S.A. Ph: 214-709-2404, Fax: 214-709-2433 e-mail: academic.books@sil.org WWW: http://www.sil.org/ Phonetic Fonts for TeX and LaTeX Linguistics/Tex mailing list ling-tex@ifi.uio.no Subscription method unknown. TIPA Created by Rei Fukui: fkr@tooyoo1.l.u-tokyo.ac.jp. Source: ftp://tooyoo.L.u-tokyo.ac.jp/pub/TeX/tipa/ Postscript manual: ftp://tooyoo.L.u-tokyo.ac.jp/pub/TeX/tipa/tipaman.ps Compressed postscript manual: ftp://tooyoo.L.u-tokyo.ac.jp/pub/TeX/tipa/tipaman.ps WSUIPA: Washington State University International Phonetic Alphabet fonts A basic WSUIPA font contains 128 phonetic characters and/or diacritics in five different point sizes (8, 9, 10, 11 and 12) and in three typefaces (roman, slanted and bold extended). Each size and typeface includes a TFM (TeX Font Metric) file and its related GF, PK or PXL file. A macro package and manual are provided. Apparently LaTeX 2.09 compatible - not LaTeX 2e compliant. Available from ftp://ftp.wustl.edu/packages/TeX/fonts/wsuipa/ OR from CTAN-ftp-archives: e.g. ftp://ftp.digital.com/pub/text/TeX/fonts/wsuipa/ Yamada Language Center * Platform: Apple Macintosh and Mircosoft Windows * Description: The Yamada Language Center maintains an archive of fonts to assist users who wish to display or type non-English fonts on their computers. Their WWW and ftp sites include five International Phonetic Alphabet fonts (or near IPA). They also have fonts for over 40 languages (American Sign Language, Arabic, Armenian, Bengali, Burmese, Celtic, Cherokee......). * Availability: : WWW Font List http://babel.uoregon.edu/yamada/fonts.html Windows Fonts http://babel.uoregon.edu/yamada/winfonts.html IPA Fonts http://babel.uoregon.edu/yamada/fonts/phonetic.html ftp site ftp://yftp@www-vms.uoregon.edu/fonts/ * Contact: Yamada Language Center, University of Oregon The vOICe * Description: Peter Meijer's Java applet/application for sound analysis and synthesis. + Platform: All (where Java VM available) + Interactive spectrographic synthesis: draw your own sound + Image sonification + Mathematical function sonification + Spectrographic sound analysis (Fourier, spectrogram) + Vision substitution research * Contact: Peter Meijer The Learning Company's Language Training * Platform: Windows and Macintosh * Description: Foreign-language training software for Spanish, French, German, Italian, Japanese, and English. In the Windows version for English, speech-recognition technology is used to help users improve accents. * Contact: The Learning Company Ph: (800) 852-2255 Email: webmaster@learningco.com WWW: http://www.learningco.Inter.net/foreign.html Wildfire - an Electronic Assistant * Platform: ? * Description: Wildfire is a phone-based electronic assistant. Functions include: + Screens, routes, and announces incoming calls. + Contact list with voicedialing. + Schedules and reminders for follow-up calls and action items. + Messaging and advanced voicemail features. * Contact: Wildfire Communications, Inc. 20 Maguire Road, Lexington, MA 02173 USA Ph: +1-617-674-1500, Fax: 617-674-1501 Demo line: 1-800-WILDFIRE Email: info@wildfire.com WWW: http://www.wildfire.com/ COMP.SPEECH FAQ POSTING - PART 2/3 * SpeechLinks: Signal Processing for Speech * Q2.1: What sampling do I need for speech? * Q2.2: Finding the pitch of a speech signal * Q2.3: How do I find the start and end points of a speech signal? * Q2.4: Where can I find FFT software? * Q2.5: Signal processing in speech technology * Q2.6: Speech sampling and signal processing hardware * Q2.7: How do I convert to/from mu-law format? * Q2.8: Signal Processing Software ___________________________________________________________________________ Q2.1: What sampling do I need for speech? For recorded speech to be understood by humans you need an 8kHz sampling rate or more and at least 8 bit sampling. This produces poor quality speech - but in can be understood. Improvements can be achieved by increasing the number of bits in sampling to 12bits or 16bits, or by using a non-linear encoding technique such as mu-law or A-law (see Q2.7). This improves the "signal-to-noise" ratio. Increasing the sampling rate above 8kHz, say to 10kHz, 16kHz or 20Khz, improves the frequency response: the higher the sampling frequency the better the high frequency content will be. A 16kHz sampling rate is a reasonable target for high quality speech recording and playback. When doing speech recognition you need to remember that the your computer is not as good as your ear so it will have trouble with poor quality sounds. The choice of an appropriate sampling setup depends very much on the speech recognition task and the amount of computer power available. ___________________________________________________________________________ Q2.2: Finding the pitch of a speech signal This topic comes up regularly in the comp.dsp newsgroup. Question 2.5 of the FAQ posting for comp.dsp gives a comprehensive list of references on the definition, perception and processing of pitch. The comp.dsp FAQ posting is posted regularly to the comp.dsp newsgroup, and is also available by ftp and on the WWW: * http://www.bdti.com/faq/dsp_faq.htm * ftp://rtfm.mit.edu/pub/usenet/comp.dsp/ The following provide pitch tracking software: * Most of the speech processing environments listed in Q1.9 including CSRE, ESPS, Kay Elemetrics Computer Speech Lab, OGI Speech Tools, Speech Filing System, Signalyze, Soundscope. ___________________________________________________________________________ Q2.3: Finding start and end points of a speech signal End-point detection algorithms identify sections in an incoming audio signal that contain speech. Accurate end-pointing is a non-trivial task, however, reasonable behaviour can be obtained for inputs which contain only speech surrounded by silence (no other noises). Typical algorithms look at the energy or amplitude of the incoming signal and at the rate of "zero-crossings". A zero-crossing is where the audio signal changes from positive to negative or visa versa. When the energy and zero-crossings are at certain levels, it is reasonable to guess that there is speech. More detailed descriptions are provided in the papers cited below and in the documentation for the following software. End-point detection software is available from: * ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/tools/ep.1.0.tar.gz * ftp://ftp.isip.msstate.edu/pub/software/signal_detector/sigd_v2.2.t ar.gz Plenty of research papers have been presented on end-pointing. Try the following: * Rabiner LR, Sambur MR, "An Algorithm for Determining the Endpoints of Isolated Utterances", Bell System Technical Journal, Vol 54, No. 2, pp 297-315, 1975. * Drago, P.G. et al. "Digital Dynamic Speech Detectors." IEEE Trans on Communications, Vol 26, No 1, Jan 78, pp. 140-145. * Newman, W.C. "Detecting Speech with an Adapative Neural Network." Electronic Design. 22 March 1990. * Taboada. J et al "Explicit Estimation of Speech Boundaries" IEE Proc. Sci. Meas. Technol., Vol 141, No.3, May 1994, pp 153-159. ___________________________________________________________________________ Q2.4: FFT Software * Comprehensive list of FFT software Links to over 65 different pieces of one-dimensional FFT code. http://tjev.tel.etf.hr/josip/DSP/fft.html * FFT Software including optimised fft routines and mixed-radix algorithms ftp://usc.edu/pub/C-numanal/fft-stuff.tar.gz OR, ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/analysis/fft-stuff. tar.gz * mixfft03.zip: C-source for a very fast arbitrary N FFT routine The C-source is ShareWare: read the text file included in the package before using the FFT routine commercially. Jens J. Nielsen: jnielsen@internet.dk Available from ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/analysis/mixfft03.z ip OR ftp://ftp.coast.net/simtel/msdos/c/mixfft03.zip * FFTW FFTW is a C subroutine library for computing the FFT in one or more dimensions. It is not limited to sizes that are powers of two, and includes real-complex and parallel transforms. Also on the FFTW web site are benchmarks comparing the performance and accuracy of many public-domain FFT implementations on a variety of platforms, as well as links to other sources of FFT code and information. Available from http://theory.lcs.mit.edu/~fftw Developed by Matteo Frigo and Steven G. Johnson: fftw@theory.lcs.mit.edu ___________________________________________________________________________ Q2.5: Signal processing in speech technology This question is far to big to be answered in a FAQ posting. Here are some WWW resources and books which cover the area well. Tony Robinson's Course Notes Dr. Tony Robinson of the Engineering Dept of Cambridge University has put his Speech Analysis course notes on the web. The base page is http://svr-www.eng.cam.ac.uk/~ajr/SA95/. There is information on the following: * Sampling theory * Filter bank analysis * Short-term fourier analysis * Linear prediction analysis * Formant analysis and voicing analysis * Speech coding * and more.... Joseph Picone's Course Notes Joseph Picone of the Institute for Signal and Information Processing (ISIP) at Mississippi State University has put two sets of course notes on the web: EE 4773/6773: Digital Signal Processing The course covers sampling, frequency analysis, z-transforms, filter design and more. The WWW site provides the syllabus, assignments, some source code data, exams, homework and solutions, lecture notes and more. EE 8993: Fundamentals of Speech Recognition The course covers background probability and phonetics/acoustics, speech signal analysis, dynamic programming, dynamic time warping, hidden Markov modelling, language modelling, neural networks, etc. The WWW sites provides the syllabus and lecture notes. Signal Processing Home page The Signal Processing Home page has information on a range of DSP issues. It includes references to a range of software and much more. http://tjev.tel.etf.hr/josip/DSP/sigproc.html Books and other References There are many good books which discuss signal processing for speech: * Digital processing of speech signals; L. R. Rabiner, R. W. Schafer. Englewood Cliffs; London: Prentice-Hall, 1978 * Voice and Speech Processing; T. W. Parsons. New York; McGraw Hill 1986 * Computer Speech Processing; ed Frank Fallside, William A. Woods Englewood Cliffs: Prentice-Hall, c1985 * Digital speech processing : speech coding, synthesis, and recognition edited by A. Nejat Ince; Kluwer Academic Publishers, Boston, c1992 * Speech science and technology; edited by Shuzo Saito pub. Ohmsha, Tokyo, c1992 * Speech analysis; edited by Ronald W. Schafer, John D. Markel, New York, IEEE Press, c1979 * Applied Speech Technology Edited by: Ann Syrdal (AT&T Bell Labs, Holmdel, New Jersey), Raymond Bennett (Ameritech, Hoffman Estates, Illinois) and Steven Greenspan (AT&T Bell Labs, Murray Hill, New Jersey). Publisher: CRC Press. * Speech Communication: Human and Machine Douglas O'Shaughnessy, Addison Wesley series in Electrical Engineering: Digital Signal Processing, 1987. * Discrete-time processing of speech signals; John R Deller, John G Proakis, John H L Hansen; Macmillan 1993. * Signal processing of speech; F J Owens; Macmillan 1993. ___________________________________________________________________________ Q2.6: Speech sampling and signal processing hardware In addition to the following information, have a look at the Audio File format document prepared by Guido van Rossum (see details in Section 1.8). Information is included on hardware for the following systems: * Macintosh Audio Hardware * PC Audio Hardware * Unix Audio Hardware Can anyone provide information for SGI, NeXT, other UNIX hardware and any other PC soundcards? Macintosh Audio Hardware - an overview * Description: ALL Macintosh computers come with the ability to play back sounds at any sample rate (sample rate conversion is done in software.) Older machines have 8 bit stereo output (hardware runs at 22254 samples/second). The newer machines have 16 bit stereo hardare running at 44100 samples/second. Most of the recent Macintosh computers come with sound input hardware. There are probably exceptions to this, but the older and some of the current low-end machines have 8 bit (linear) mono hardware running at 22254.54 samples/second. All of the PowerPC, AV, and the 500 series notebook computers come with 16 bit 44kHz stereo sampling hardware. They can also record at 22050 samples/second. The sound manager implements an AGC (Automatic Gain Control) function for the 8 bit hardware. The drivers have a switch to turn off the AGC. There are a number of DSP vendors that support high quality audio. Generally this means quieter analog sections, and more IO formats (AES/IBU, for example). Try DigiDesign and Spectral Innovations. The software drivers for sound are described in "Inside Macintosh: Sound". If you want to see some sample code check out the sources for the Matlab "Sound and Image Toolbox". They can be found at ftp://ftp.apple.com/pub/malcolm/SoundAndImageToolbox.cpt. hqx Routines that play and record sounds using the toolbox are included (and interfaced to Matlab). PC Audio Hardware Note: new soundcards are becoming available all the time - the information below is definitely not up to date. Check out the following newsgroups for up-to-date information. * comp.sys.ibm.pc.soundcard * comp.sys.ibm.pc.soundcard.GUS * comp.sys.ibm.pc.soundcard.advocacy * comp.sys.ibm.pc.soundcard.games * comp.sys.ibm.pc.soundcard.misc * comp.sys.ibm.pc.soundcard.music * comp.sys.ibm.pc.soundcard.tech The Soundcard WWW Site is an excellent source of information: * http://www.wi.leidenuniv.nl/audio/ An good source of programs and information for soundcards is SimTel: * http://www.acs.oakland.edu/oak/SimTel/win3/sound.html Additional information on PC soundcards is provided by the FAQ postings for the comp.sys.ibm.pc.soundcard.misc newsgroup. These are available by anonymous ftp from: ftp://rtfm.mit.edu/pub/usenet/comp.sys.ibm.pc.soundcard.misc/ * Aria Soundcard FAQ * Aria Soundcard Support List * MIDI files software archives on the Internet * Turtle Beach sound cards FAQ Unix Audio Hardware Could someone please provide information on the audio capabilities of other Unix platforms? Sun standard audio port: SPARC I & II * Input and Output: 1 channel, 8 bit mu-law encoded, 8kHz sample rate. This provides telephone quality sampling. Sun DBRI audio port (SPARC 10 & 20) * Input and Output: Stereo (2 channels). 16-bit linear sampling. Multiple sample rates (48000, 44100, 37800, 32000, 22050, 18900, 16000, 11025, 9600, 8000 Hz) Silicon Graphics Audio The Silicon Graphics audio Frequently Asked Questions (FAQ) is the best place to get information on SGI audio capabilities and programming. It provides information on connecting the audio output, using the DSP capabilities, controlling the audio output, programming, useful software and more. It is available from: * WWW: http://www-viz.tamu.edu/~sgi-faq/faq/html/audio/ * News: comp.sys.sgi.misc * Ftp: ftp://viz.tamu.edu/pub/sgi/faq/ Ariel Signal Processors * Platform: Various * Description: A range of signal I/O, A/D, D/A and DSP products are available. There are too many to list. * Contact: Ariel Corp. 433 River Road, Highland Park, NJ 08904. Ph: 908-249-2900 Fax: 908-249-2123 DSP BBS: 908-249-2124 ___________________________________________________________________________ Q2.7: How do I convert to/from mu-law format? Mu-law coding is a form of compression for audio signals including speech. It is widely used in the telecommunications field because it improves the signal-to-noise ratio without increasing the amount of data. Typically, mu-law compressed speech is carried in 8-bit samples. It is a companding technqiue. That means that carries more information about the smaller signals than about larger signals. On SUN Sparc systems have a look in the directory /usr/demo/SOUND. Included are table lookup macros for ulaw conversions. [Note however that not all systems will have /usr/demo/SOUND installed as it is optional - see your system admin if it is missing.] OR, here is some sample conversion code in C. /** ** Signal conversion routines for use with Sun4/60 audio chip **/ #include stdio.h unsigned char linear2ulaw(/* int */); int ulaw2linear(/* unsigned char */); /* ** This routine converts from linear to ulaw ** ** Craig Reese: IDA/Supercomputing Research Center ** Joe Campbell: Department of Defense ** 29 September 1989 ** ** References: ** 1) CCITT Recommendation G.711 (very difficult to follow) ** 2) "A New Digital Technique for Implementation of Any ** Continuous PCM Companding Law," Villeret, Michel, ** et al. 1973 IEEE Int. Conf. on Communications, Vol 1, ** 1973, pg. 11.12-11.17 ** 3) MIL-STD-188-113,"Interoperability and Performance Standards ** for Analog-to_Digital Conversion Techniques," ** 17 February 1987 ** ** Input: Signed 16 bit linear sample ** Output: 8 bit ulaw sample */ #define ZEROTRAP /* turn on the trap as per the MIL-STD */ #define BIAS 0x84 /* define the add-in bias for 16 bit samples */ #define CLIP 32635 unsigned char linear2ulaw(sample) int sample; { static int exp_lut[256] = {0,0,1,1,2,2,2,2,3,3,3,3,3,3,3,3, 4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4, 5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5, 5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5, 6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6, 6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6, 6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6, 6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6, 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7, 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7, 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7, 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7, 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7, 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7, 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7, 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7}; int sign, exponent, mantissa; unsigned char ulawbyte; /* Get the sample into sign-magnitude. */ sign = (sample >> 8) & 0x80; /* set aside the sign */ if (sign != 0) sample = -sample; /* get magnitude */ if (sample > CLIP) sample = CLIP; /* clip the magnitude */ /* Convert from 16 bit linear to ulaw. */ sample = sample + BIAS; exponent = exp_lut[(sample >> 7) & 0xFF]; mantissa = (sample >> (exponent + 3)) & 0x0F; ulawbyte = ~(sign | (exponent << 4) | mantissa); #ifdef ZEROTRAP if (ulawbyte == 0) ulawbyte = 0x02; /* optional CCITT trap */ #endif return(ulawbyte); } /* ** This routine converts from ulaw to 16 bit linear. ** ** Craig Reese: IDA/Supercomputing Research Center ** 29 September 1989 ** ** References: ** 1) CCITT Recommendation G.711 (very difficult to follow) ** 2) MIL-STD-188-113,"Interoperability and Performance Standards ** for Analog-to_Digital Conversion Techniques," ** 17 February 1987 ** ** Input: 8 bit ulaw sample ** Output: signed 16 bit linear sample */ int ulaw2linear(ulawbyte) unsigned char ulawbyte; { static int exp_lut[8] = {0,132,396,924,1980,4092,8316,16764}; int sign, exponent, mantissa, sample; ulawbyte = ~ulawbyte; sign = (ulawbyte & 0x80); exponent = (ulawbyte >> 4) & 0x07; mantissa = ulawbyte & 0x0F; sample = exp_lut[exponent] + (mantissa << (exponent + 3)); if (sign != 0) sample = -sample; return(sample); } ___________________________________________________________________________ Q2.8: Signal Processing Software [Note: Question 1.9 lists speech laboratory environments and audio editors, many of which provide basic and advanced signal processing capabilities.] Signal Processing Products * SigLib from Numerix Ltd. On the Web The following sites provide lists of useful DSP software. Not all the software is directly applicable to speech processing. comp.dsp FAQ http://www.bdti.com/faq/dsp_faq.htm DSP Internet Resources http://www.eg3.com/ http://www.eg3.com/dsp.htm Poynton's Digital Signal Processing Resource List http://www.inforamp.net/~poynton/Poynton-dsp.html WWW Pages Relating to Sound Computation http://datura.cerl.uiuc.edu/netstuff/sigsoundLinks.html Yahoo - Signal and Image Processing http://www.yahoo.com/Science/Engineering/Electrical_Engineering /Signal_and_Image_Processing/ Sound Related Resources http://pscinfo.psc.edu/~geigel/menus/sound.html SPLIB: Signal Processing url LIBrary http://jazz.rice.edu/splib/ Wavelet's Home Page http://www.mat.sbg.ac.at/~uhl/wav.html SigLib from Numerix Ltd. * Platform: Windows, Unix and all major DSPs * Description: SigLib is an ANSI C Source DSP Library and includes functions for the following areas : spectrum analysis, windowing, filtering (fixed and adaptive coefficient), convolution, correlation, covariance, signal generation, statistical analysis, regression analysis, communications and modulation, digital effects, vectors processing, control, graphics and file I/O. Detailed product information and a description of the application of SigLib to speech processing is provided on the Numerix Ltd. WWW site. * Availability: A free demonstration of SigLib V2.0 is available from the Numerix Ltd. WWW site. Educational discount is available for SigLib. * Contact: Numerix Ltd., 157 Sileby Road, Barrow-on-Soar, Leics, LE12 8LW, UK. Phone/Fax : +44 (0)1509 413195 Email: numerix@numerix.co.uk WWW: http://www.numerix.co.uk/ ___________________________________________________________________________ Speech Coding and Compression comp.speech FAQ Section 3 * SpeechLinks: Speech Coding * Q3.1: Speech compression techniques * Q3.2: Information on speech coding and compression * Q3.3: Speech Compression / Coding Software ___________________________________________________________________________ Q3.1: Speech compression techniques Provided by Tony Robinson: The aim of speech compression is to produce a compact representation of speech sounds such that when reconstructed it is perceived to be close to the original. The two main measures of closeness are intelligibility and naturalness. The standard reference point is toll quality speech, this is the same as what would be expected over a telephone line, for example, speech coded at 8 kHz using 8 bit ulaw coding and a maximum frequency of about 3.3 kHz. This is a bit rate of 64 kbps, and as such represents a compressed form over (say) 16 bit, 16 kHz speech which is the standard in speech recognition work. ulaw coding does not exploit the (normally large) sample to sample correlations found in speech. ADPCM is the next family of speech coding techniques, and does exploit this redundancy by using a simple linear filter to predict the next sample of speech. The resulting prediction error is typically quantised to 4 bits thus giving a bit rate of 32 kbps (see, for example, the software in Q3.3: 32 kbps ADPCM, G.711/721/723 Compression, shorten). The advantages of ADPCM are that is simple to implement and has very low delay. To obtain more compression specific properties of the speech signal must be modelling. The main assumption is known as the source filter model of speech production. This assumes that a source (voicing or fricative excitation) is passed through a filter (the vocal tract response) to produce the speech. The simplest implementation of this is known as a LPC synthesiser (e.g. LPC10e). At every frame the speech is analysed to compute the filter coefficients, the energy of the excitation, a voicing decision, and a pitch value if voiced. At the decoder a regular set of pulses for voiced speech or white noise for unvoiced speech is passed through the linear filter and multiplied by the gain to produce the speech. This is a very efficient system and typically produces speech coded at 1200-2400bps. With clever acoustic vector prediction this can be reduced to 300-600bps. The disadvantages are a loss of naturalness over most of the speech and occasionally a loss of intelligibility. The CELP family of coders compensates for the lack of quality of the simple LPC model by using more information in the excitation. Each of a set of codebook of excitation vectors is tried and the index of the one that best matches the original speech is transmitted. This results in an increase in the bit rate to typically 4800-9600bps. Most speech coding research is currently directed towards CELP coders. (See, for example, CELP 3.2a, a TMS implementation, a G.728 LD-CELP vocoder, and the L&H implementation. ___________________________________________________________________________ Q3.2: Information on speech coding and compression Reference Books The following books cover speech coding/compression. * Douglas O'Shaughnessy, Speech Communication: Human and Machine, Addison Wesley series in Electrical Engineering: Digital Signal Processing, 1987. * Bishnu Atal in ed. Fallside, F. and W. Woods, ed. Computer Speech Processing. London: Prentice/Hall International, 1985. N. S. Jayant and P. Noll, Digital Coding of Waveforms, Prentice Hall, ISBN 0-13-211913-7 01, 1984. * W.B. Kleijn and K.K. Paliwal (Eds.), Speech Coding and Synthesis, Elsevier, Amsterdam, 1995. Contents, preface etc on the WWW: http://www.elsevier.nl/section/engtech/scs/menu.htm * Thomas P. Barnwell, Kambiz Nayebi and Craig H Richardson, Speech Coding: A Computer Laboratory Textbook, John Wiley and Sons Inc, 1996. * Schuyler R Quackenbush, Tom P Barnwell III, Mark A Clements, Objective Measures of Speech Quality, Prentice-Hall, 1988. And the are good tutorial articles. * Makhoul, J. "Linear Prediction: A Tutorial Review." Proc. of the IEEE 63 (1975): 561 - 580. On the WWW comp.compression FAQ Includes a few questions and answers on the compression of speech. ftp://rtfm.mit.edu/pub/usenet/comp.compression/ Tony Robinson's Speech Analysis Course A complete course on speech analysis, including some stuff on speech coding. http://svr-www.eng.cam.ac.uk/~ajr/SA95/ http://svr-www.eng.cam.ac.uk/~ajr/SA95/node78.html ITU Coding Standards Members of the ITU (International Telecommunications Union) can obtain copies of the Series G Recommendations (including G.711/721/723/728) from the ITU WWW site (http://www.itu.ch/) and from http://www.itu.ch/itudoc/itu-t/rec/g/g700-799.html. Jason Woodard's Speech Coding Page Introduction to speech coding plus information on a series of speech coding standards. http://www-mobile.ecs.soton.ac.uk/speech_codecs/index.html WWW searchable online-bibiliography for Phonetics and Speech Technology Over 8000 entries provided by Institut fur Phonetik at Johann Wolfgang Goethe-Universitat Frankfurt. http://www.uni-frankfurt.de/~ifb/bib_engl.html Ciaran McElroy's Speech Coding Page Introduction to many types of speech coding. http://wwwdsp.ucd.ie/speech/tutorial/speech_coding/speech_tut.h tml Examples of speech coding Nam Phamdo's Speech Coding Demonstration Examples of ADPCM, LD-CELP, CELP, LPC10 and CELP coding and coding over a noisy channel. http://admii.arl.mil/~fsbrn/phamdo/speech_demo.html Phil Karn's Digital/Analog Voice Demo Examples of several speech coding systems. http://www.qualcomm.com/people/pkarn/voicedemo/ ___________________________________________________________________________ Q3.3: Speech Compression / Coding Software The following speech compression software is described in the FAQ. * 32 kbps ADPCM * Castleton Network Systems - G.729 Voice Coder * CELP 3.2a & LPC-10 * 8 Kbit/s CELP on the TMS320C5x family of DSP chips * CyberVoice * Rockwell's DigiTalk * File format conversion * G.711/721/723 Compression * G.728 LD-CELP vocoder * G.728 Compression * GSM 06.10 Compression * Lernout & Hauspie Speech Coding (5 products) * Lernout & Hauspie Speech Coding SDK * MPEG Audio * shorten - a lossless compressor for speech signals * Sipro Lab Telecom Inc. Coding * Sonarc: Digital Audio Compression * StarAudio Compressor/Player * TrueSpeech from DSP Group * U.S.F.S. 1016 CELP vocoder for DSP56001 * ToolVox from Voxware 32 kbps ADPCM * Platform: SGI and Sun Sparcs * Description: 32 kbps ADPCM C-source code (G.721 compatibility is uncertain) * Contact: Jack Jansen * Availablity: http://www.cwi.nl/ftp/audio/adpcm.shar Castleton Network Systems - G.729 Voice Coder * Platform: TI TMS320C5x DSP * Description: G.729, also called CS-ACELP (Conjugate-Structure Algebraic Code Excited Linear Prediction), is a state-of-the-art voice compression ITU (International Telecommunications Union) standard that can be used in a wide range of applications including wireless communications, digital satellite systems, packetized speech and digital leased lines. G.729 provides 8000 bits/s bandwidth for compressed speech at toll quality (equivalent to G.726 32 kbit/s ADPCM under clean channel condition). Also, G.729 has lower complexity and lower bit rate than G.728. The Castleton G.729 implementation provides a bit-exact implementation of the ITU standard on a single TI TMS320C5x DSP. The software is C callable and fully re-entrant, which allows easy interfacing and multi-channel capability. The encoder and decoder are fully independent, therefore, a DSP device can run a number of full-duplex or half-duplex channels. The coder and the decoder are able to operate under a real-time task switching kernel. * Cost and Availablity: Contact Castleton Network Systems. * Contact: Castleton Network Systems Corporation 350 Terry Fox Drive, Kanata, Ontario, Canada K2K 2W5 Ph: 613-591-8786, Fax: 613-591-8783 Email: inquire@castleton.com WWW: http://www.castleton.com/ CELP 3.2a & LPC-10 * Platform: Sun (the makefiles and source can be modified for other platforms) * Description: CELP is lossy compression technqiue. The US Department of Defences's Federal-Standard-1016 based 4800 bps code excited linear prediction voice coder version 3.2a (CELP 3.2a). Fortran and C simulation source codes. * Availability: By anonymous ftp from: ftp://ftp.super.org/pub/speech/celp_3.2a.tar.Z Or from the comp.speech ftp server ftp://svr-ftp.eng.cam.ac.uk/comp.speech/coding/celp_3.2a.tar.Z ftp://svr-ftp.eng.cam.ac.uk/comp.speech/coding/celp_3.2a.tar.gz LPC-10 Fortran source code is also available: ftp://ftp.super.org/pub/speech/lpc10-1.0.tar.gz Here is a modified LPC-10 release that includes ANSI C source: http://www.arl.wustl.edu/~jaf/lpc/ * Documentation: The following articles describe the Federal-Standard-1016 4.8-kbps CELP coder: + Campbell, Joseph P. Jr., Thomas E. Tremain and Vanoy C. Welch, "The Federal Standard 1016 4800 bps CELP Voice Coder," Digital Signal Processing, Academic Press, 1991, Vol. 1, No. 3, p. 145-155. + Campbell, Joseph P. Jr., Thomas E. Tremain and Vanoy C. Welch, "The DoD 4.8 kbps Standard (Proposed Federal Standard 1016)," in Advances in Speech Coding, ed. Atal, Cuperman and Gersho, Kluwer Academic Publishers, 1991, Chapter 12, p. 121-133. The U.S. DoD's Federal-Standard-1015/NATO-STANAG-4198 based 2400 bps linear prediction coder (LPC-10) was republished as a Federal Information Processing Standards Publication 137 (FIPS Pub 137). It is described in: + Thomas E. Tremain, "The Government Standard Linear Predictive Coding Algorithm: LPC-10," Speech Technology Magazine, April 1982, p. 40-49. There is also a section about FS-1015 in the book: + Panos E. Papamichalis, Practical Approaches to Speech Coding, Prentice-Hall, 1987. The voicing classifier used in the enhanced LPC-10 (LPC-10e) is described in: + Campbell, Joseph P., Jr. and T. E. Tremain, "Voiced/Unvoiced Classification of Speech with Applications to the U.S. Government LPC-10E Algorithm," Proceedings of the IEEE Intl. Conf. on Acoustics, Speech, and Signal Processing, 1986, p. 473-6. * Vendors: Realtime DSP code for FS-1015 and FS-1016 is sold by: + John DellaMorte, DSP Software Engineering 165 Middlesex Tpk, Suite 206, Bedford, MA 01730, USA Ph: 1-617-275-3733 Fax: 1-617-275-4323 Email: dspse.bedford@channel1.com DSP Software Engineering's FS-1016 code can run on a DSP Research's Tiger 30 (a PC board with a TMS320C3x and analog interface suited to development work). + DSP Research 1095 E. Duane Ave, Sunnyvale, CA 94086, USA Ph: (408)773-1042 Fax: (408)736-3451 8 Kbit/s CELP on the TMS320C5x family of DSP chips * Description: For low bandwidth transmission of voice, compact voice storage for archival purposes, low-cost digital answering machines and efficient storage for voice mail. Features : + near toll quality at 8 Kb/s. + Variable rate option with 1 Kb/s silence encoding. + Implemented on a fixed-point processor for lower system cost. + Attractive licensing scheme. + Future availability of 4 Kb/s. + Custom rates possible. Capacity : + Two half-duplex or one full duplex channels on the 20 MIPS 'C5x (at 95% and 55% CPU utilization respectively). + Two full duplex channels on the 28.6 MIPS 'C5x (at 77% CPU utilization). + Requires 9 K-words program memory and 3 K-words data memory. + Decoding in real-time on a 486 class CPU. * Contact: CVI Inc. 443 Vienna Cres. North Vancouver, BC, Canada V7N 3B3 Tel: (604) 987 1719 Fax: (604) 986 8139 Email: cvi@extropia.wimsey.com CyberVoice * Description: Cybernetics InfoTech, Inc. offers the following products + Telephone voice compression at 1.2, 2.4, 4.8 and 6.0 kbit/s with good-communications-quality to near-toll-quality coded voice; + Wideband voice (7-kHz bandwidth) compression at 16 kbit/s with near-original-quality coded voice; + Internet Voice E-mail software with voice editing, high-quality low-data-rate voice compression, fast/slow voice playback, and more. * Availablity: C code and Windows .DLL for telephone voice compression and wideband voice compression are available for licensing. Real-time DSP codes are under development. Voice E-mail software is available for purchase and download from the CyberVoice home page. * Contact: Cybernetics InfoTech, Inc. 2 Professional Dr., #228, Gaithersburg, MD 20879 WWW: http://www.cybit.com/ E-mail: info@cybit.com Fax: 301-590-0359 Rockwell's DigiTalk * Description: The DigiTalk coder operates at a sampling rate of 8KHz and transmits 223 bits of coded speech every 26ms, giving an overall bit rate of 8.577Kbps. The algorithm is based on analysis-by-synthesis predictive coding with vector-coded excitation, in which the excitation signal is optimized by minimizing the perceptually weighted error between the original and synthesized speech. More information and results of perceptual tests are available on the WWW. * Availablity: See the WWW page: http://www.nb.rockwell.com/ref/digitalk/ File format conversion * Platform: SUN OS? * Description: Conversion utility able to encode and decode between the the following formats: G.723, G.721, A-law, u-law and linear. * Availability: By anonymous ftp from ftp://ftp.cwi.nl/pub/audio/ccitt-adpcm.tar.Z G.711/721/723 Compression * Description: + G.711 : CCITT u-law and A-law compression + G.721 : CCITT 32 kbps ADPCM coder + G.723 : CCITT 24 kbps and 40 kbps ADPCM coders * Availability: By email to itudoc@itu.ch, with GET ITU-3022 as the *only* line in the body of the message. It is also available by anonymous ftp from: ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/coding/G711_G 721_G723.tar.Z G.728 LD-CELP vocoder * Platform: Analog Devices ADSP-2171 * Description: Real-time, full-duplex G.728 LD-CELP vocoder that runs on a single Analog Devices ADSP-2171. Source and object code available for a one-time license fee. * Contact: Cole Erskine Analogical Systems 299 California Avenue, Suite 120 Palo Alto, CA 94306, USA Tel:(415) 323-3232 FAX:(415) 323-4222 email: cole@analogical.com G.728 Compression * Description: G.728 low delay celp package written by Alex Zatsman of Analog Devices, Inc. * Availability: By anonymous ftp from ftp://dspsun.eas.asu.edu/pub/speech/ldcelp.tgz GSM 06.10 Compression * Platform: Unix; faster than real time on most Sun SPARCstations * Description: GSM 06.10 is a standardized lossy speech compression employed by most European wireless telephones. It uses RPE/LTP (residual pulse excitation/long term prediction) coding to compress frames of 160 13-bit samples (8 kHz sampling rate, i.e. a frame rate of 50 Hz) into 260 bits. * Contact: GSM 06.10 support and implementation _jutta@cs.tu-berlin.de_, cabo@cs.tu-berlin.de * Availability: The following configurations are available be anonymous ftp: gzip compression from Germany: ftp://ftp.cs.tu-berlin.de/pub/local/kbs/tubmik/gsm/gsm-1. 0.7.tar.gz MS-DOS compression from Germany: ftp://ftp.cs.tu-berlin.de/pub/local/kbs/tubmik/gsm/ddj/gs m-107.zip MS-DOS compression from USA: ftp://ftp.mv.com/pub/ddj/1194.12/gsm-105.zip * Misc: The WWW site is http://www.cs.tu-berlin.de/~jutta/toast.html Lernout & Hauspie Speech and Music Coding Product Range * Product name: L&H.smc650: 32kbps ADPCM Speech coding + Implementation of ADPCM 32 kbps based on CCITT G721 standard. + Estimated quality: 4.1 MOS (Mean Opinion Score) + Hardware Example: Analog Devices ADSP2101 + Input / Output signal: A-Law or mu-Law PCM (64 kbps); Linear signal with up to 16 bits per sample; 8 kHz sampling rate * Product name: L&H.smc550: LD-CELP 16 kbps speech coding + Proprietary implementation of LD-CELP 16 kbps based on CCITT G728 standard. + Estimated quality: 4.0 MOS (Mean Opinion Score) + Hardware Example: Motorola 5600X + Input / Output signal: A-Law or mu-Law PCM (64 kbps); Linear signal with up to 16 bits per sample; 8 kHz sampling rate * Product name: L&H.smc450: 16-17.5 kbps speech coding + Estimated Quality: 3.9 MOS (Mean Opinion Score) + Hardware Examples: Analog Devices ADSP2101, Intel 486 DX2/66 MHz + Input / Output Signal: A-Law or mu-Law PCM (64 kbps); Linear signal with up to 16 bits per sample; 8 kHz sampling rate. * Product name: L&H.smc350: 4.8-9.6 kbps speech coding + Proprietary CELP based software for compression rates of 4.8 kbps to 9.6 kbps + Estimated Quality: 3.5 MOS (Mean Opinion Score) + Hardware Examples: AT&T DSP32C + Input / Output signal: A-Law or mu-Law PCM (64 kbps); Linear signal with up to 16 bits per sample; 8 kHz or 11.025kHz sampling rate. * Product name: L&H.smc250: 2.4 kbps speech coding + Combination of multi band excitation and code book excited linear prediction. + Estimated Quality: 3.0 MOS (Mean Opinion Score). + Hardware Examples: Intel 486 DX2/66 MHz, Analog Devices ADSP2101 + Input signal: A-Law or mu-Law PCM (64 kbps); Linear signal with 12-15 bits per sample; 8 kHz sampling rate. + Output signal: A-Law or mu-Law PCM (64 kbps); Linear signal with 12-15 bits per sample; 8 kHz sampling rate. * See also: L&H Speech Coding SDK * More Information: On the WWW: http://www.lhs.com/coding.html * Cost: Unknown * Contact: Lernout and Hauspie Speech Products 20 Mall Road, 4th Floor Burlington, MA 01803, USA Ph: +1-617-238-0960, Fax: +1-617-238-0986 Email: sales@lhs.com WWW: http://www.lhs.com/ Lernout & Hauspie Speech Coding SDK * Description: Windows based software development kit for integrating speech coding technology with Windows based PC applications. * Requirements: IBM-compatible 486 DX/33 MHz + 2MB RAM + MS DOS 5.0 + MS Windows 3.1 (or higher) + Sound Blaster compatible sound board. * See also: L&H Speech Coding Products * More Information: On the WWW: http://www.lhs.com/coding.html * Cost: Unknown * Contact: Lernout and Hauspie Speech Products 20 Mall Road, 4th Floor Burlington, MA 01803, USA Ph: +1-617-238-0960, Fax: +1-617-238-0986 Email: sales@lhs.com WWW: http://www.lhs.com/ MPEG Audio MPEG (Moving Pictures Experts Group) is a standard methods for compression and transmission of digital video and audio. Detailed FAQs and WWW sites are available for MPEG: MPEG Pointers and Resources http://www.mpeg.org/ FAQ by Luigi: http://www.crs4.it/~luigi/MPEG/mpegfaq.html FAQ by Frank Gadegast http://www.powerweb.de/mpeg/mpegfaq/ FAQ by by Chad Fogg http://www-plateau.cs.berkeley.edu/mpegfaq/MPEG-2-FAQ.html How to Install an MPEG Audio Player for your Web Navigator http://www.mpeg.org/index.html/MPEG-audio-player.html MPEG Audio Software on the WWW Audio and Music Applications for Silicon Graphics Systems Lists 4 MPEG audio players for SGI machines. http://reality.sgi.com/employees/cook/audio.apps/public.html MPEG-1 Audio Layer 3 encoder, decoder and FAQ From the Fraunhofer Institute http://www.iis.fhg.de/departs/amm/layer3/index.html MPEG-2 Audio FAQ from Philips http://www.keymodules.philips.com/MD/mpeg/faqmpeg2.htm MPEG-1 and MPEG-2 audio software Universitaet Hannover ftp://ftp.tnt.uni-hannover.de/pub/MPEG/audio/ MPEG-1 Audio Layer 1 &2 encoder - decoder Internet Underground Music Archive (IUMA) ftp://ftp.iuma.com/audio_utils/converters/source/ Buddy Software Library: MPEG-1 Audio Layer 3 encoder and player http://www.buddy.org/softlib.html MPEG-1 Audio Layer 1 & 2 decoder and verifier at CCETT ftp://ftp.ccett.fr/pub/mpeg/audio_new/ MPEG-2 Audio encoder and decoder at CCETT ftp://ftp.ccett.fr/pub/mpeg/mpeg2/ MPEG Audio - MetaSound * Platform: MS Windows/3.1 and Windows/95 * Description: MetaSound is a partial MPEG-1 software decoder which is designed to work with hardware video decoders. It can reduce the hardware cost by eliminating the need for a hardware audio decoder. Currently, MetaSound has been successfully incorporated to work with three hardware video decoders. Features + Performance: For 486 DX4-100 machines or above, MetaSound can deliver FM quality (22 KHz) sound. For Pentium-90 or above machines, MetaSound requires 40% CPU bandwidth to deliver CD quality (44.1 KHz) sound. + Portability: it can take less than one month to port to new hardware video decoders. + CD standard supports including Video CD 1.0, Video CD 2.0, and CDI. + User interface with full set of functions: volume control, stop, pause, forward, backward, mute, resume, select the previous/next program track (Video CD 2.0), randomly select a program track (Video CD 2.0). + Error Recovery: can automatically skip error bitstreams. * Contact: Meta Media, Inc. F8, #10-1, Ho-Ping East Rd. Sec. 1, Taipei, Taiwan, R.O.C. Ph: 011-886-2-369-3330, Fax: 011-886-2-369-3331 Email: mmedia@ms4.hinet.net.tw shorten - a lossless compressor for speech signals * Platform: UNIX/DOS * Description: A fast waveform coder suitable for a speech and music signals in a wide variety of file formats. The degree of compression is adjustable from lossless to three bits a sample. 16bit 16kHz speech generally attains 50% lossless compression and 16:3 compression of CDROM quality speech is obtainable with only minor audiable degredation. * Availability: Anonymous ftp - UNIX and DOS versions ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/coding/shorte n.tar.gz ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/coding/shorte n.tar.Z ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/coding/shorte n.zip Sipro Lab Telecom Inc. Coding * Platform: Various processors * Description: Coding software for several International Standards plus two Proprietary standards. International Standards 1. PCS 1900 (a 13 kbps codec, established as a North American PCS standard) 2. Enhanced GSM (a 13 kbps codec) 3. G.723 (8 kbps codec established as a multi-purpose international standard) 4. G.729 (a dual-rate codec for the video phone market) 5. G.729 Annex A (8 kbps codec made for Digital Simultaneous Voice & Data transmission in the modem industry). Proprietary Standards 1. ACELP 8 v2.0 codec (flexible dual rate codec equipped with a VAD) 2. ACELP 4.8 codec * Contact: Sipro Lab Telecom Inc. 770, Chemin Lucerne, Ville Mont-Royal (Quebec), H3R 2H6 CANADA Ph: (514) 737-5874, Fax: (514) 737-2327 E-mail: sales@sipro.com WWW: http://www.sipro.com/ Sonarc: Digital Audio Compression * Platform: DOS and Windows * Description: Sonarc provides reversable, variable-rate compression of audio signals. Obtains compression ratio which averages about 2:1. Supports monaural and stereo files, 8-bit and 16-bit files, and WAVE and VOC formats. * Availablity: Shareware by Richard P. Sprague Speech Compression P.O. Box 1785, Wilsonville, OR, 97070-1785, USA Ph: (503) 263-3102 Email: 76635.3652@compuserve.com StarAudio Compressor/Player * Platform: Win95 * Description: Using a time-domain process delivers lossless decompressed data. Processes any source of .wav file format, high quality 16-bit audio data at any sampling rate. Requires no special hardware and decompression speed is real-time on most 486's and on any Pentium. The higher the sampling rate the higher the compression ratio; minimum compression of 4:1 for 11k data, and usually exceeding 7:1 for 44k data. Full bandwidth of signal is preserved with default compression options. Compression options allow increase of compression ratio further with a slight trade off in the reduction of the output quality. A decompression library is available for application development. * Demo: Download the shareware version of the program from the STR WWW site. * Misc: A technical paper is available in Word 6.0 format: ftp://ftp.speechtech.com/pub/speechtech/docs/audocw60.exe * Contact: Speech Technology Research Ltd., Suite B - 1623 McKenzie Avenue, Victoria, B.C. V8N 1A6, Canada Ph: +1-250-477-0544 Email: products@speechtech.com WWW: http://www.speechtech.com/home/speechtech/ TrueSpeech from DSP Group * Description: TrueSpeech is a family of speech compression and decompression algorithms and software. It is designed for personal computers and personal communications devices. With the high compression ratios ranging from 15:1 to 27:1, TrueSpeech improves the storage and communications transmission of digital voice information and can be used in the integration of personal computers and telephones. TrueSpeech can be utilized in many products and applications such as: + Multimedia PCs + Sound cards and modems + Computer/telephony and teleconferencing + Voice mail systems and PBX systems + Wireless/cellular applications + Personal digital assistants + Games, Education + Video/cable and on-line services The TrueSpeech encoder is available for free in the Sound System of Windows 95 and Windows NT. The DSPG WWW pages have information on how to add TrueSpeech capability to your WWW pages. * Contact: DSP Group, Inc. 3120 Scott Boulevard, Santa Clara, CA 95054-3317, USA Phone: (408) 986-4300 Fax: (408) 986-4323 Email: Webster@dspg.com WWW: http://www.dspg.com/index.html U.S.F.S. 1016 CELP vocoder for DSP56001 * Platform: DSP56001 * Description: Real-time U.S.F.S. 1016 CELP vocoder that runs on a single 27MHz Motorola DSP56001. Free demo software available for PC-56 and PC-56D. Source and object code available for a one-time license fee. * Contact: Cole Erskine Analogical Systems 299 California Avenue, Suite 120 Palo Alto, CA 94306, USA Tel:(415) 323-3232 FAX:(415) 323-4222 Email: cole@analogical.com ToolVox from Voxware * Platform: Windows and soon available on Mac (in Beta now) and Unix * Description: ToolVox is a proprietary frequency domain speech coder. 11 KHz speech is coded to an average rate of between 5,000 bits per second and 9,000 bps. Real-time compression algorithms available for 2,400 bps. 22 KHz playback, as well as a ultra low bit rate 8 KHz codec are coming soon. On playback, the time scale can be changed by a 5x factor, pitch can be modified over a 3 octave range, and vocal personality can be modified using a tranformation function called VoiceFonts(tm). * Misc 1: A SDK for Windows is available. * Misc 2: Demo software is available from the Voxware Inc WWW page: http://www.voxware.com/ * Price: Basic toolkit is $895 US. OEM and mass distribution licenses are separate. Ordering information is provided on the Voxware WWW server. * Contact: Voxware, Inc. Ph: (609) 497-1212 Fax: (609) 497-2490 Sale information: sales@voxware.com WWW: http://www.voxware.com/ ___________________________________________________________________________ Natural Language Processing comp.speech FAQ Section 4 There is now a newsgroup specifically for Natural Language Processing; comp.ai.nat-lang. A FAQ posting is available for the group: ftp://rtfm.mit.edu/pub/usenet/comp.ai.nat-lang/Natural_Language _Processing_FAQ There is also a lot of useful information on Natural Language Processing in the comp.ai FAQ. That FAQ lists available software and useful references. It includes a substantial list of software, documentation and other info available by ftp. The FAQ has information on the following: * Q4.1: NLP References and Books * Q4.2: NLP Software ___________________________________________________________________________ Q4.1: NLP References and Books Take a look at the FAQ for the "comp.ai" newsgroup as it also includes some useful references. * James Allen: Natural Language Understanding, (Benjamin/Cummings Series in Computer Science) Menlo Park: Benjamin/Cummings Publishing Company, 1987. + This book consists of four parts: syntactic processing, semantic interpretation, context and world knowledge, and response generation. * G. Gazdar and C. Mellish, Natural Language Processing in Prolog, Addison Wesley, 1989 * G. Gazdar and C. Mellish, Natural Language Processing in Lisp, Addison Wesley, 1989 * G. Gazdar and C. Mellish, Natural Language Processing in Pop11, Addison Wesley, 1989 + Emphasis on parsing, especially unification-based parsing, lots of details on the lexicon, feature propagation, etc. Fair coverage of semantic interpretation, inference in natural language processing, and pragmatics; much less extensive than in Allen's book, but more formal. There are three versions, one for each programming language listed above, with complete code. * Shapiro, Stuart C.: Encyclopedia of Artificial Intelligence Vol.1 and 2. New York: John Wiley & Sons, 1990. + There are articles on the different areas of natural language processing which also give additional references. * Paris, Ce'cile L.; Swartout, William R.; Mann, William C.: Natural Language Generation in Artificial Intelligence and Computational Linguistics. Boston: Kluwer Academic Publishers, 1991. + The book describes the most current research developments in natural language generation and all aspects of the generation process are discussed. The book is comprised of three sections: one on text planning, one on lexical choice, and one on grammar. * Readings in Natural Language Processing, ed by B. Grosz, K. Sparck Jones and B. Webber, Morgan Kaufmann, 1986 + A collection of classic papers on Natural Language Processing. Fairly complete at the time the book came out (1986) but now seriously out of date. Still useful for ATN's, etc. * Klaus K. Obermeier, Natural Language Processing Technologies in Artificial Intelligence: The Science and Industry Perspective, Ellis Horwood Ltd, John Wiley & Sons, Chichester, England, 1989. The following are extensive bibliographies related to NLP: * Computational Parsing : Syntactic Analysis, Semantic Analysis, Semantic Interpretation, Parsing Algorithms, Parsing Strategies : BIBLIOGRAPHY, by Conrad F. Sabourin 1994, 2 volumes, 1029p, ISBN 2-921173-02-6, INFOLINGUA inc., P.O. Box 187 Snowdon, Montreal, H3X 3T4, Canada. * Computational Text Understanding : Natural Language Programming, Argument Analysis : BIBLIOGRAPHY, by Conrad F. Sabourin 1994, 657p, ISBN 2-921173-06-9, INFOLINGUA inc., P.O. Box 187 Snowdon, Montreal, H3X 3T4, Canada. See also: http://gomer.mlink.net/infolingua.html * Computational Text Generation : Generation from data or Linguistic Structure, Text Planning, Sentence Generation, Explanation Generation : BIBLIOGRAPHY, by Conrad F. Sabourin with a survey article by Mark T. Maybury 1994, 649p, ISBN 2-921173-07-7, INFOLINGUA inc., P.O. Box 187 Snowdon, Montreal, H3X 3T4, Canada. See also: http://gomer.mlink.net/infolingua.html * Natural Language Processing : Interfaces to Databases, to Expert Systems, to Robots, to Operating Systems, and to Question-Answering Systems : BIBLIOGRAPHY, by Conrad F. Sabourin, 1994, 2 volumes, 847p, ISBN 2-921173-08-5 INFOLINGUA inc., P.O. Box 187 Snowdon, Montreal, H3X 3T4, Canada See also: http://gomer.mlink.net/infolingua.html Journals The major journals of the field are * Computational Linguistics and _Cognitive Science_ for the artificial intelligence aspects, * Cognition for the psychological aspects, * Language and _Linguistics and Philosophy_ and Linguistic Inquiry for the linguistic aspects. * Artificial Intelligence occasionally has papers on natural language processing. Conferences The major NLP conferences are * ACL: held annually * COLING: held biannually Most AI conferences have a NLP track; AAAI, ECAI, IJCAI and the Cognitive Science Society conferences usually interesting for NLP. CUNY is an important psycholinguistic conference. Other conferences include NELS, the conference of the Chicago Linguistic Society (CLS), WCCFL, LSA, the Amsterdam Colloquium, and SALT. ___________________________________________________________________________ Q4.2: NLP Software Natural Language Software Registry (NLSR) - NLP Tools * The Natural Language Software Registry is available from the German Research Institute for Artificial Intelligence (DFKI) in Saarbrucken. Its purpose is to facilitate the exchange and evaluation of natural language processing software within the research community. To this end, the NLSR is cataloging natural language software projects, both commercial and non- commercial. The new updated and enlarged version contains more than 100 descriptions of natural processing software. Registry listings include: + speech signal processors, such as the Computerized Speech Lab (Kay Elemetrics) + morphological analyzers, such as PC-KIMMO (Summer Institute for Linguistics) + parsers, such as Alveytools (University of Edinburgh) + semantic and pragmatic analyzer, such as NLL (University of the Saarland, Germany) + generation programs, such as FUF (Ben Gurion University of the Negev) + knowledge representation systems, such as Rhet (University of Rochester) + multicomponent systems, such as ELU (ISSCO), PENMAN (ISI), Pundit (UNISYS), SNePS (SUNY Buffalo), + NLP-Tools, such as GULP (University of Georgia) or Linguist (Kansai Research Laboratory) + applications programs (misc.) * If you have developed a piece of software for natural language processing that other researchers might find useful, you can include it by returning the questionnaire available from the sources below. * ftp://ftp.dfki.uni-sb.de/pub/registry * e-mail: registry@dfki.uni-sb.de * Natural Language Software Registry Deutsches Forschungsinstitut fuer Kuenstliche Intelligenz (DFKI) Stuhlsatzenhausweg 3 D-66123 Saarbruecken Germany * Other ftp sites are ftp://crlftp.nmsu.edu/pub/non-lexical/NL_Software_Registy ftp://dri.cornell.edu/pub/Natural_Language_Software_Registry Part of Speech Tagger * Description: A rule-based part of speech tagger developed by Eric Brill. * Availability: The tagger software, about 10 descriptive papers and related data are available by anonymous ftp from ftp://ftp.cs.jhu.edu/pub/brill/ COMP.SPEECH FAQ POSTING - PART 3/3 * SpeechLinks: Speech Synthesis * Q5.1: What is speech synthesis? * Q5.2: How can speech synthesis be performed? * Q5.3: References/Books on Synthesis * Q5.4: Speech Synthesis on the WWW * Q5.5: Speech Synthesis Software/Hardware ___________________________________________________________________________ Q5.1: What is speech synthesis? Speech synthesis programs convert written input to spoken output by automatically generating synthetic speech. Speech synthesis is often referred to a "Text-to-Speech" conversion (TTS). ___________________________________________________________________________ Q5.2: Performing speech synthesis There are several algorithms. The choice depends on the task they're used for. The easiest way is to just record the voice of a person speaking the desired phrases. This is useful if only a restricted volume of phrases and sentences is used, e.g. messages in a train station, or schedule information via phone. The quality depends on the way recording is done. More sophisticated but worse in quality are algorithms which split the speech into smaller pieces. The smaller those units are, the less are they in number, but the quality also decreases. An often used unit is the phoneme, the smallest linguistic unit. Depending on the language used there are about 35-50 phonemes in western European languages, i.e. there are 35-50 single recordings. The problem is combining them as fluent speech requires fluent transitions between the elements. The intellegibility is therefore lower, but the memory required is small. A solution to this dilemma is using diphones. Instead of splitting at the transitions, the cut is done at the center of the phonemes, leaving the transitions themselves intact. This gives about 400 elements (20*20) and the quality increases. The longer the units become, the more elements are there, but the quality increases along with the memory required. Other units which are widely used are half-syllables, syllables, words, or combinations of them, e.g. word stems and inflectional endings. The Museum of Speech Analysis and Synthesis has pictures of artificial speech systems going back over 150 years: worth a visit. ( http://mambo.ucsc.edu/psl/smus/smus.html) ___________________________________________________________________________ Q5.3: References/Books on Synthesis Books and Papers * Thierry Dutoit, An Introduction to Text-to-Speech Synthesis, Kluwer Academic Publishers (Dordrecht), 1997, ISBN 0-7923-4498-7, 312 pages. Volume 3 in the series on Text, Speech and Language Technology. * Douglas O'Shaughnessy, Speech Communication: Human and Machine Addison Wesley series in Electrical Engineering: Digital Signal Processing, 1987. * T.V. Raman, Auditory User Interfaces --Toward The Speaking Computer Kluwer Academic Publishers, Boston, ISBN 0-7923-9984-6, August 1997, 168 pp. * D. H. Klatt, "Review of Text-To-Speech Conversion for English", Jnl. of the Acoustic Society of America (JASA), Vol 82, pp 737-793. * "Talking Machines, Theories, Models and Designs" Eds, G. Bailly & C. Benoit (Elsevier: North Holland) * I. H. Witten. Principles of Computer Speech, London: Academic Press, Inc., 1982. * W.B. Kleijn and K.K. Paliwal (Eds.), Speech Coding and Synthesis, Elsevier, Amsterdam, 1995. Contents, preface etc on the WWW: http://www.elsevier.nl/section/engtech/scs/menu.htm * John Allen, Sharon Hunnicut and Dennis H. Klatt, "From Text to Speech: The MITalk System", Cambridge University Press, 1987. * J.P.H. van Santen, R. W. Sproat, J. P. Olive, and J. Hirschberg, "Progress in Speech Synthesis", Springer, 1996. On the WWW * Survey of the State of the Art in Human Language Technology Report edited by Ronald A. Cole et. al. with a section on Text-to-Speech Technologies. http://www.cse.ogi.edu/CSLU/HLTsurvey/ch5node1.html Bibliographies and Reference Lists * WWW searchable online-bibiliography for Phonetics and Speech Technology with more than 8000 entries. Provided by Institut fur Phonetik at Johann Wolfgang Goethe-Universitat Frankfurt. http://www.uni-frankfurt.de/~ifb/bib_engl.html * Computational Speech Processing: Speech Analysis, Recognition, Understanding, Compression, Transmission, Coding, Synthesis ; Text to Speech Systems, Speech to Tactile Displays, Speaker Identification, Prosody Processing : BIBLIOGRAPHY, by Conrad F. Sabourin, 1994, 2 volumes, 1187p, ISBN 2-921173-21-2, INFOLINGUA inc., P.O. Box 187 Snowdon, Montreal, H3X 3T4, Canada. See also: http://gomer.mlink.net/infolingua.html ___________________________________________________________________________ Q5.4: Speech Synthesis on the WWW Most of the following are links to WWW pages with demonstrations of speech synthesis. Plenty more links are included in the detailed list of speech synthesis software/hardware in Q5.5. Speech Synthesis "Museum" URL: http://www.cs.bham.ac.uk/~jpi/synth/museum.html Maintained by Jon Iles (j.p.iles@cs.bham.ac.uk) at the University of Birmingham. Information and speech samples for + YorkTalk + Loughborough Sound Images + University of Birmingham - FDFS + Eurovocs + DECtalk + AT&T Bell Labs Synthesiser + S.W.A.Ll.C. - Welsh Synthesis from CSTR + All-Prosodic Speech Synthesis - IPOX + Orator from Bellcore The Festival Speech Synthesis System http://www.cstr.ed.ac.uk/projects/festival.html Pre-synthesized examples in English, Welsh and Spanish, and online demo of English. Pavarobotti http://www.shc.uiowa.edu/fun/pavarobotti/pavarobotti.html WWW demo of the Pavarobotti synthesis technology developed at the National Center for Voice and Speech (http://www.shc.uiowa.edu/ncvs_home.html). Say... http://wwwtios.cs.utwente.nl/say WWW demo of the rsynth speech synthesis software. The WWW capability was implemented by Axel Belinfante. Musee sonore de la synthese de la Parole en francais http://www.icp.grenet.fr/exemples_synthese/ex.html Speech synthesis examples from a series of French language speech synthesisers plus links to other speech synthesis demo pages. + ICP-Grenoble + CNET-Lannion (with TD-PSOLA) + KTH-Stockholm + Universite-Mons - several versions Lucent Technologies Bell Labs Text-to-Speech http://www.bell-labs.com/project/tts/ Demos and samples of the latest Lucent Technologies Bell Labs Text-to-Speech system. WATSON FlexTalk from AT&T Advanced Speech Products Group http://www.att.com/aspg/demo.html WWW interface to the WATSON FlexTalk speech synthesis demonstration. AT&T Bell Laboratories Voices http://www.research.att.com/cgi-bin/cgiwrap/mjm/voices.cgi WWW interface to the AT&T Bell Laboratories text to speech (TTS) synthesizer Laureate from British Telecom http://www.labs.bt.com/innovate/speech/laureate/ Demo of the Laureate speech synthesis system - not yet commercially available. ORATOR from Bellcore Online demo of the ORATOR system developed at Bellcore. http://www.bellcore.com/ORATOR/ SVOX from TIK, ETH in Zurich http://www.tik.ee.ethz.ch/cgi-bin/w3svox Demo of German speech synthesis from Institut fur Technische Informatik und Kommunikationsnetze. Speech Synthesis Research at OGI http://www.cse.ogi.edu/CSLU/research/TTS Examples of diphone speech corpora and algorithms developed at OGI for synthesis of American English and Mexican Spanish using the Festival framework. Lyricos http://www.cse.ogi.edu/CSLU/research/TTS/research/sing.html Demos of the Lyricos singing voice synthesis system. Concatenation-based synthesis of singing voice from MIDI input. Multi-Lingual TTS from Gerhard-Mercator University, Duisburg http://www.fb9-ti.uni-duisburg.de/demos/speech.html Synthesis in German, English or Japanese. TMH: Institutionen for Taloverforing och Musikakustik, Kungliga Tekniska Hogskolan http://www.speech.kth.se/info/software.html Synthesis in Swedish, Finish, Norwegian, Icelandic, Danish, British and American English, French, German, Italian, Spanish, LA Spanish and Greek. Haskins Laboratory WWW Site http://www.haskins.yale.edu/Haskins/MISC/special.html Examples of several types of speech synthesis. Articulatory Synthesis by HyperASY. SineWave Synthesis. Gestural Computational Model. Pattern Playback system of the 1940's! BeSTspeech from Berkeley Speech Technologies, Inc., (BST) http://www.bestspeech.com/weblang.html Eurovocs Multilingual Speech Synthesis http://www.elis.rug.ac.be/ELISgroups/speech/research/eurovocs.h tml Based on Lernout and Hauspie technology. HADIFIX German Speech Synthesis http://asl1.ikp.uni-bonn.de/~tpo/Hadiq.en.html Provided by the Instituts fur Kommunikationsforschung und Phonetik, Universitat Bonn. Centigram's TruVoice Demo http://www.centigram.com/centigram/TruVoice/index.html Allows control of speech rate, pitch and other prosodic charateristics. MBROLA: Free Speech Synthesis Project http://tcts.fpms.ac.be/synthesis/modelcmp.html WWW demo of MBROLA which compares the quality of PSOLA, MBR-PSOLA, LPC, and Hybrid Harmonic/Stochastic concatenative synthesizers. Provided by the TCTS Lab, Faculti Polytechnique de Mons, Belgium Institute of Phonetic Sciences http://fonsg3.let.uva.nl/IFA-Features.html Links to lots of on-line speech synthesis demonstrations provided by the Institute of Phonetic Sciences of the Faculty of Arts of the University of Amsterdam. Yahoo page on speech generation http://www.yahoo.com/Science/Computer_Science/Artificial_Intell igence/Natural_Language_Processing/Speech_Generation/ ___________________________________________________________________________ Q5.5: Speech Synthesis Software/Hardware Please email any updates, corrections or additions to the following list. The range of commercially available synthesis software is growing rapidly so any help in keeping up to date will be appreciated. Other lists of speech synthesis software on the WWW include: Kevin Lenzo's list of Macintosh Speech Resources and Apps http://www.cs.cmu.edu/~lenzo/mac_speech_apps.html Speech Toys Speech Synthesis Information http://www.speechtoys.com/spchtoys/spsyn.html In the FAQ... The following speech recognition software/hardware is described in the comp.speech FAQ. _Apple Macintosh_ * BeSTspeech from Berkeley Speech Technologies, Inc., (BST) * Infovox Product Range * Macintosh Speech Output Applications * Macintosh Speech Synthesis Manager * MacYack Pro * MBROLA: Free Speech Synthesis Project * ProVoice Developer's Speech Toolkit from First Byte * SENSYN speech synthesizer * Sound Bytes DeveloperUs Kit * Macintosh Speech Synthesis Manager _Windows (including 95, NT, 3.1)_ * AcuVoice * AT&T Watson Speech Synthesis * BeSTspeech from Berkeley Speech Technologies, Inc., (BST) * Creative TextAssist and TextAssist API * DECtalk: Text-to-Speech from Digital * ETI-Eloquence * HADIFIX * Infovox Product Range * IPOX: All Prosodic Speech Synthesis Architecture * Lernout and Hauspie Text-To-Speech Windows SDK * Listen2 Text Reader * MBROLA: Free Speech Synthesis Project * Monologue for Windows from First Byte * PAM - A Text-To-Speech Application * ProVerbe Speech Engine from ELAN Informatique * ProVoice Developer's Speech Toolkit from First Byte * SENSYN speech synthesizer * Sound Bytes DeveloperUs Kit * Tinytalk * TruVoice from Centigram * WinSpeech * ZMD Speech Synthesis _DOS_ * CSRE: Computerized Speech Research Environment * Infovox Product Range * MBROLA: Free Speech Synthesis Project * ProVoice Developer's Speech Toolkit from First Byte * SENSYN speech synthesizer * spchsyn.exe * Tinytalk * ZMD Speech Synthesis _OS/2_ * ProVerbe Speech Engine from ELAN Informatique * ProVoice Developer's Speech Toolkit from First Byte * Sound Bytes DeveloperUs Kit _Unix_ * AcuVoice * AsTeR * BeSTspeech from Berkeley Speech Technologies, Inc., (BST) * DECtalk: Text-to-Speech from Digital * ETI-Eloquence * Emacspeak - A Speech Output Subsystem For Emacs * Festival Speech Synthesis System * JSRU * Klatt-style synthesiser * KPE80 - A Klatt Synthesiser and Parameter Editor * "learph": Trainable text-to-phoneme software by Antonio Lucca * Lucent Technologies Bell Labs Text-to-Speech system * MBROLA: Free Speech Synthesis Project * Orator from Bellcore * ProVerbe Speech Engine from ELAN Informatique * rsynth * SENSYN speech synthesizer * SGI Developers Toolbox Synthesiser * Speak * TrueTalk * TruVoice from Centigram _Integrated Circuits and Dedicated Hardware_ * Eurovocs * Infovox Product Range * ProVerbe Speech Engine from ELAN Informatique * RC Systems V8600/V8601 Text to Speech synthesizers _Other Platforms_ * BeSTspeech from Berkeley Speech Technologies, Inc., (BST) * TheBigMouth (NeXT) * MBROLA: Free Speech Synthesis Project * Narrator Translator Library (Amiga) * Narrator (Amiga) * TextToSpeech Kit (NeXT) * Orator from Bellcore * SENSYN speech synthesizer * WreadFiles: File reader for Commodore Amiga _Unknown_ * Lernout and Hauspie Text-To-Speech (3 products) * SIMTEL * Text to Phoneme Program 1 * Text to phoneme program 2 * Text to phoneme program 3 AcuVoice * Platform: Windows, Solaris * Description: AcuVoice is a natural sounding text-to-speech system built using a concatenative approach. Currently it is available for an American English Male Voice. Software Developer Kits are available for the Windows Platform (32-Bit) and also for the Solaris Platform. More information and samples are available on the Acuvoice web site. * Contact: AcuVoice, Inc. 84 W. Santa Clara Street, Suite 720, San Jose, CA 95113-1810 Ph: 1(408)289-1661, Fax: 1(408)289-1201 Demo: 1(408)289-1177 Email: AcuVoice1@AOL.COM WWW: http://www.acuvoice.com/ AsTeR * Platform: UNIX * Description: TTS front-end program which encodes structural information about documents in speech synthesis. For more information check out: http://www.research.digital.com/CRL/personal/raman/aster/ aster-toplevel.html * Operation requirements: Lisp: Lucid, clisp * Contact: T. V. Raman WWW: http://www.research.digital.com/CRL/personal/raman/raman.html Email: raman@adobe.com AT&T Watson Speech Synthesis * Platform: Windows 95/NT on a Pentium 75 Mhz or higher * Description: Watson is a software implementation of AT&T Bell Laboratories voice processing technology. Watson includes BLASR Speech Recognition (see Q6.6) and FlexTalk speech synthesis. It requires no special hardware to run other than a standard sound card and/or phone card. Technical details for the FlexTalk speech synthesis include: + Compliant with MS Speech API. + Male and Female Voices available + 8 KHz and 11 KHz output + SoundBlaster compatible sound card and drivers required + Context sensitive abbreviation expansion + Accurate pronunciation of most proper names + Adjustable vocal tract size, speed, volume, pitch, etc. + American English only - other languages in development The AT&T Advanced Speech Products Group home page provides more detailed information including a Frequently Asked Questions list, information for application developers on the Independent Software Vendor (ISV) Program (including info on the SDK, licensing, and the training program). * Requirements: Uses 2 MB RAM, 10 MB Disk. Requires a Pentium 75 MHz or higher (uses * Cost and Availability: WATSON is a software-based speech platform with a Software Developers Kit (SDK) that allows application developers to use voice processing in their applications. It is not available as a stand-alone product. Licensing information (inc. price) is provided in the AT&T Advanced Speech Products Group home page * See also: Watson BLASR speech recognition in Q6.5, Microsoft Speech API, and Advanced Speech API. * Contact: AT&T Advanced Speech Products Group Suite 700, 44 East Mifflin Street, Madison, WI 53703, USA Ph: 1-800-5-WATSON, Fax: 1-608-259-2269 Email: aspg@attmail.com WWW: http://www.att.com/aspg/ BeSTspeech from Berkeley Speech Technologies, Inc., (BST) * Platform: available for Macintosh, Sun, Silicon Graphics, Windows PC and IBM RS/6000 platforms, and can be ported to others. * Description: BeSTspeech reads ASCII text no vocabulary limits. Available for Dutch, English (male and female), French, German, Italian, Portuguese, Spanish, Arabic, Cantonese, Japanese, Korean, Malay, Mandarin and Russian. * Availability: Berkeley Speech Technologies, Inc does not sell end user toolkits or products. * Contact: Berkeley Speech Technologies, Inc. 2246 Sixth Street, Berkeley, California 94710, USA Ph: (510) 841-5083, Fax: (510) 841-5093 Email: webmaster@bst.com WWW: http://www.bestspeech.com/index.html TheBigMouth - a Text to Speech Program * Platform: NeXT * Description: Text to speech program based on concatenation of pre-recorded speech segments. * Availability: ftp://ftp.cs.keio.ac.jp/pub/NeXT/source/TheBigMouth1.0.tar.Z Creative TextAssist * Platform: Windows * Description: Based on DECtalk speech synthesis. A detailed description of TextAssist is provided on the Creative WWW pages. TextAssist TextReader provides a convenient Windows user interface for text reading. * Availability: Creative TextAssist is bundled with most (all?) Creative Sound Blaster audio cards. TextAssist preview software is available from the Creative Labs TextAssist home page. * Contact: Creative Labs, Inc. Address, phone, email etc unknown WWW: http://www.creaf.com/ : http://www.creaf.com/wwwnew/tech/devcnr/tassist.html Creative TextAssist API * Platform: Windows * Description: The TextAssist API (TAAPI) is created for Microsoft Windows 3.1x and Windows 95 developers who intend to develop 16-bit Text-to-Speech software applications using Creative's TextAssist speech engine. It supports direct control of speech output characteristics, concurrent playback of text-to-speech and wave files, foreign language support, speech synchronization, exception dictionaries. It also includes a voice editing tool for creating new custom voices, a Visual Basic Custom Control for high-level support in Visual Basic and other languages * Availability: The TextAssist API is released to registered developers at no cost. * Contact: WWW: http://www.creaf.com/ FAQ: http://www.creaf.com/wwwnew/tech/devcnr/tassfaq.html CSRE: Computerized Speech Research Environment * Platform: DOS * Description: CSRE is a software system which includes in an implementation of the Klatt speech synthesizer. See the CSRE entry in Q1.9 and the AVAAZ WWW pages for more detail. * Contact: AVAAZ Innovations Inc. P.O.Box 8040, 1225 Wonderland Rd. N, London, Ontario, CANADA, N6G 2B0 Ph: +1-519-472-7944 , Fax: +1-519-472-7814 Email: info@avaaz.com WWW: http://www.icis.on.ca/homepages/avaaz/ DECtalk Speech Synthesis * Platform: Windows NT, Alpha with Digital UNIX and RS232 ports * Description: Converts ordinary text into natural-sounding, intelligible speech. Provides personalized voices, and extensive user controls. DECtalk technology is available for the following packaging options. + DECtalk PC card option: An industry-standard ISA/EISA bus card implementation that can be integrated with any Intel 486 processor-based system running DOS or Windows. Applications can be interfaced to the bus via a DOS Terminate and Stay Resident (TSR) driver or a Windows Dynamic Link Library (DLL). This option is available with an external speaker with volume control and headphone jack. + DECtalk Express external package: An external, portable package that you can plug in to any PC or serial port. The external package includes a built-in speaker and headphone jack, plus combined on/off and volume controls and a rechargeable battery pack. + DECtalk Software solution: Software-only text to speech for Alpha or Intel systems running Windows NT or Alpha systems running Digital UNIX. Provides complete speech synthesis capabilities so developers can enhance applications with DECtalk technology. DECtalk Software output can be directed to audio devices, into WAVE files, or into memory buffers. * Pricing: ://www.systems.digital.com/DIcatalog/html/DECtalk-Speech-Synthesis -oi.html * More Information: Digital Equipment Corporation WWW pages: http://www.digital.com/ DECtalk page: http://www.systems.digital.com/DIcatalog/html/DECtalk-Software.htm l Ph: 1-800-DIGITAL DECtalk Software * Platform: Digital UNIX and Windows NT * Description: DECtalk converts standard ASCII text into natural, intelligible speech. Speech output through any audio device is supported by Microsoft Video for Windows or Multimedia Services for Digital UNIX. An API gives developers direct access to text-to-speech functions. Provides nine voice personalities (4 female, 4 male, 1 child). Provides punctuation and tonal control, supports customized pronunciation of trade jargon and acronyms. Common programming interface works with both Alpha and Intel platforms. * More Information: Digital Equipment Corporation WWW pages: http://www.digital.com/ DECtalk Software page: http://www.systems.digital.com/DIcatalog/html/DECtalk-Software.htm l WWW: http://www.systems.digital.com/DIcatalog/html/DECtalk-Speech-Synth esis.html Ph: 1-800-DIGITAL ETI-Eloquence * Platform: MS Windows (Win95,NT,3.1), Solaris, SunOS, SGI, RS/6000 * Description: ETI-Eloquence is a software based text-to-speech system. It generates waveforms completely algorithmically instead of by concatenating waveforms, for maximum flexibility and naturalism. For instance, when the user requests a deeper voice, the software simulates a larger vocal tract, instead of simply pitch-shifting samples. It uses high-level linguistic parsing, which obviates the need for a huge dictionary. It handles numbers, acronyms, currency, etc. It includes a set of annotation symbols, for placing stress on particular words, expressing excitement/boredom, etc. Also allows phonetic input. Supports MS SAPI. Produces male and female voices for General American English. Dialects under development include Alabama and Brooklyn. * Price: Flexible license agreements on application. * Availability:Eloquent Technology, Inc. 2389 North Triphammer Road, Ithaca, NY 14850 , USA Ph: (607) 266-7025, Fax: (607) 266-7030 Email: info@eloq.com WWW: http://www.eloq.com/ Emacspeak - A Speech Output Subsystem For Emacs * Platform: UNIX, Emacs * Description: Emacspeak is a speech output system that will allow someone who cannot see to work directly on a UNIX system. Emacspeak is built on top of Emacs. With emacspeak loaded, Emacs provides spoken feedback for everything you do. Emacspeak currently supports the new Dectalk Express speech synthesizer, as well as older versions of the Dectalk e.g. the MultiVoice. See the Emacspeak WWW page, the Emacspeak FAQ or the Emacspeak distribution for additional details. * Requirements: Requires GNU FSF Emacs 19 (version 19.23 or later) and TCLX 7.3B (Extended TCL) to run Emacspeak. * Availability: Emacspeak WWW page http://www.research.digital.com/CRL/personal/raman/emacsp eak/emacspeak.html Emacspeak source http://www.research.digital.com/CRL/personal/raman/emacsp eak/emacspeak.tar.gz * Contact: T. V. Raman, raman@adobe.com Eurovocs * Platform: Various - RS232 hardware connection * Description: Eurovocs is a stand-alone text-to-speech synthesizer which uses the text-to-speech technology of Lernout and Hauspie Speech Products. Available for Dutch, French, German and American English with other languages planned for release soon. One Eurovocs device can support two different languages. Eurovocs can be connected to any computer via a standard serial interface (RS232). It supports personal dictionaries, generation of DTMF tones, and pronunciation of special character sequences such as digit strings, telephone-numbers, date and time indications, abbreviations, alphanumeric strings etc. * Contact: Technologie & Revalidatie Postbus 128, B-9000 Gent, Belgium Ph: +32-9-264 33 97, Fax: +32-9-264 35 94 E-mail: noe@elis.rug.ac.be WWW: http://www.elis.rug.ac.be/ELISgroups/speech/research/eurovocs.html Festival Speech Synthesis System * Platform: General Unix (including Solaris (2.4,2.5), SunOS, HPUX, SGIs, Linux, Dec Alpha, FreeBSD) * Description: Festival is a general multi-lingual speech synthesis system developed at CSTR, University of Edinburgh. It offers a full text to speech system with various APIs, as well an environment for development and research of speech synthesis techniques. It is written in C++ with a Scheme-based command interpreter for general control. Festival's home page offers demos, the full manual and access to the download page. The distribution includes full source and documentation, and lexicons and speech databases for British English text to speech. * Price: Free for non-commercial use * Availability: by anonymous ftp: WWW: http://www.cstr.ed.ac.uk/projects/festival/download.html ftp: ftp://ftp.cstr.ed.ac.uk/pub/festival/ HADIFIX * Platform: Windows * Description: German speech synthesis system developed at the Institute for Communications Research and Phonetics , University of Bonn. Provides conversion of input text to phonemes, automatic prediction of stress, phrasing and pitch, and speech generation by concatenation of small units of natural speech. Demisyllables and similar units are used; they comprise all consonants before the vowel and the beginning of the vowel (initial demisyllable) or the end of the vowel and the following consonants (final demisyllable). For example, the word 'Strolch' is formed by concatenating 'Stro' and 'olch'. * Demo: Windows demo software available. Limited to synthesis of one short text (text.txt) at a time. Speech format limitations too. 1.3MB file. ftp://asl1.ikp.uni-bonn.de/pub/hadifix/hadidemo.zip A 1993 version is available with unlimited synthesis from a string of phonemic symbols and accent markers. 6MB file. ftp://asl1.ikp.uni-bonn.de/pub/hadifix/hadi25.lzh * WWW: http://asl1.ikp.uni-bonn.de/~tpo/Hadifix.en.html * On-line demo: http://asl1.ikp.uni-bonn.de/~tpo/Hadiq.en.html Infovox Product Range * Description: Multilingual Text-to-speech systems, languages available: American English, British English, German, French, Spanish, Italian, Swedish, Norwegian, Icelandic, Danish and Finnish. * Product name:INFOVOX 500, PC BOARD + Product description: Half length expansion board for IBM PC, XT, AT, PS/2 model 30 or compatible personal computers. The board can also be connected via the serial port. Language and control program for downloading into RAM or mounted on EPROMs + Platform: DOS/Windows with IBM PC, XT, AT, PS/2 model 30 or compatible + Delivered standard interface: MS DOS I/O driver * Product name: INFOVOX 600, OEM BOARD + Product description: OEM board built with CMOS IC's. Language and control program are stored in on-board fixed memory. + Platform: any, hardware interface: 9-pole D-SUB (RS 232-C) 300-9600 Baud. + Delivered standard interfaces: MS DOS I/O driver and interface to Apple Speech manager. * Product name: INFOVOX 700, DESKTOP UNIT + Product description: Desktop unit with built in Infovox 600 to be connected to any computer or terminal via an RS 232-C serial interface. Built in loudspeaker and rechargable battery for 4 hours use, and control knobs for continuous control of speech volume and speed. + Platform: various through hardware interface + Delivered standard interfaces: MS DOS I/O driver and interface to Apple Speech manager * Product name: INFOVOX 650, OEM BOARD + Product description: OEM-board built with CMOS IC's. Language and control program are stored in on-board memory. + Platform: any, hardware interface: 9 pole D-SUB (RS 232-C) 300-9600 Baud + Delivered standard interfaces: MS DOS I/O driver and interface to Apple Speech manager * Product name: INFOVOX 750, DESKTOP UNIT + Product description: Desktop unit with built in Infovox 650 to be connected to any computer or terminal via an RS 232-C serial interface. Built in loudspeaker and rechargable battery for 5 hours use, and a control knob for continuous control of speech volume. + Platform: various through hardware interface. Delivered standard interfaces include MS DOS I/O driver and interface to Apple Speech manager * Product name: Infovox 210, software for Apple Macintosh + Product description: Software based text-to-speech conversion. Produces 16 bit and 8 bit sound. Delivered on 3.5" diskettes with user lexicon and a complete documentation. + Platform: Apple Macintosh with minimum 68030, 33 MHz microprocessor. + Delivered standard interfaces: Standard interface to Apple Speech manager * Product name: Infovox 220, software for Microsoft Windows. + Product description: Software based text-to-speech conversion. Produces 16 bit sound and conforms to Microsoft Windows multimedia standard MCI. Delivered on 3.5" diskettes with user lexicon and a complete documentation. + Platform: Windows on IBM compatible PC with minimum 486/25MHz microprocessor. + Delivered standard interfaces: Standard interface to Microsoft Windows 3.1 and sound boards supporting Microsoft Windows multimedia driver for audio. * Contact: Telia Promotor Infovox AB TTS Sales Division P.O. Box 2069, S-171 02 Solna, Sweden Ph: +46 8 764 35 00, Fax: +46 8 735 78 76 Email: tts-sales@infovox.se WWW: http://www.promotor.telia.se/NYA/cc/t-s/index.html IPOX: All Prosodic Speech Synthesis Architecture * Platform: Windows * Description: IPOX is an experimental, all-prosodic speech synthesizer, developed by Arthur Dirksen and John Coleman. IPOX is freely available (after registration) for evaluation and non-profit research purposes. * Requirements: PC (preferably a fast 486) running Windows 3.1 or higher. Sound output requires a 16-bit Windows-compatible sound card * Availability: By WWW from http://www.tue.nl/ipo/people/adirksen/ipox/ipox.htm JSRU * Platform: UNIX and PC * Cost: 100 pounds sterling (from academic institutions and industry) * Description: A C version of the JSRU system, Version 2.3 is available. It's written in Turbo C but runs on most Unix systems with very little modification. A Form of Agreement must be signed to say that the software is required for research and development only. * Contact: Dr. E.Lewis _eric.lewis@bristol.ac.uk)_ Klatt-style synthesiser * Platform: Unix * Cost: Free * Description: Software posted to comp.speech in late 1992. * Availability: By ftp from the comp.speech ftp site + ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/synthesis/klatt.3. 04.tar.gz + ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/synthesis/klatt.3. 04.tar.Z * See also: KPE80 - A Klatt Synthesiser and Parameter Editor. KPE80 - A Klatt Synthesiser and Parameter Editor * Platform: Unix * Description: The KPE80 program provides a graphical interface for the implementation of the Klatt 1980 formant synthesiser written by Jon Iles and Nick Ing-Simmons. It was inspired by IGE, a piece of code written by Rob Fletcher ( http://www.york.ac.uk/~rpf1/IGE.html). * Technical Desc.: It is comprised of an X-Window interface and version 3.03 of the synthesiser code. The interface allows users to display and edit Klatt parameters using a graphical display which includes the time-amplitude waveform of both the original speech and its synthetic copy, and some signal analysis facilities. Most of the work in choosing the parameter values to produce the synthetic copy has to be done by the user. KPE will estimate the fundamental frequency contour from an original token; this estimate will need to be amended where errors occur. It is possible to specify the formant trajectories with some precision by overlaying the appropriate formant frequency parameter tracks on the spectrogram of the target waveform. A number of facilities exist to help in the refinement of parameter values: original and synthetic waveforms can be compared aurally, spectrally, and spectrographically using built-in speech analysis facilities. * File formats: KPE will read RIFF (.wav) files and SFS files. (SFS is a suite of speech-signal processing programs available free from Phonetics and Linguistics, UCL.) * Availability: KPE for SunOs 4.1.3 (statically compiled libraries) ftp://pitch.phon.ucl.ac.uk/pub/kpe/kpe80.sun413.tar.Z KPE for Linux (statically compiled libraries) ftp://pitch.phon.ucl.ac.uk/pub/kpe/kpe80.linux.tar.Z The source code (needs gcc and SUIT to compile) ftp://pitch.phon.ucl.ac.uk/pub/kpe/kpe80.src.tar.Z A postscript overview of KPE ftp://pitch.phon.ucl.ac.uk/pub/kpe/OVERVIEW.ps The SFS distribution ftp://pitch.phon.ucl.ac.uk/pub/sfs/ * See also: Public domain Klatt-style speech synthesis code. * Contact: Andrew Simpson Department of Phonetics and Linguistics, University College London Wolfson House, 4 Stephenson Way, London NW1 2HE Email: a.simpson@ucl.ac.uk WWW: http://www.phon.ucl.ac.uk/home/andrew/home.html "learph": Trainable text-to-phoneme software by Antonio Lucca * Platform: UNIX * Description: Experimental software which learns text to phoneme translation from examples using decision-tree-like data structures. It is based on the assumption that each letter can correspond to different phoneme strings depending on the context. * Availability: Examples and source are available on the WWW: http://www.silab.dsi.unimi.it/~al367212/ttsdoc.html * Contact: Antonio Lucca: toninlcc@tesi.dsi.unimi.it Lernout & Hauspie Text-to-Speech (3 products) Lernout & Hauspie have three TTS products. The functionality of the products is similar, however, they differ in hardware implementation and other details where described below. * L&H tts2000/T: TTS for the Telephony and Telecommunications Market * L&H tts2000/M: TTS for the Computer and Multimedia Market * L&H tts3000/C: TTS for the Buisness and Consumer Electronics Market * Description: Text to Speech (TTS) software based on parameterized segment concatenation (diphones, triphones and tetraphones) algorithms. Available for US English, German, Dutch, French, Spanish (Castilian), Italian and Korean. General features include: + The control of volume, speech rate and speech pitch. + The use of control sequences to customize TTS output (adding pauses, using phonetic input, etc.). + Switching between languages at run time. + A personal vocabulary editor is available for building exception dictionaries. + Readout modes: letter by letter, word by word or sentence by sentence. + Input formats: orthographic input, phonetic input, phonetic input with prosodic information. * tts2000/T + Output formats: 8 bit mu-law PCM, 8 bit A-law PCM, 16 bit linear PCM. + Sampling Frequency: 8kHz + Single channel platform examples: SHARP SH7000, ARM6/ARM7, Intel i960, TI TMS320C31, AT&T DSP3210 + Multi-channel platform examples: TI TMS320C31, AT&T DSP3210 * tts2000/M + Output formats: 8/16 bit wave format, 8 bit mu-law PCM, 8 bit A-law PCM, 16 bit linear PC. + Sampling Frequency: 8/10/11.025 kHz + Single processor platform examples: ARM6/ARM7, Intel 386/486/Pentium, Motorola 68040 + Two processor platform examples: {Intel 386/486/Pentium or Motorola 68030} and {ADI ADSP21XX or Motorola 5600X or TI TMS320C25/20C5X} * tts3000/C + Output formats: 8 bit mu-law PCM, 8 bit A-law PCM, 16 bit linear PCM. + Sampling Frequency: 10kHz + Single processor platform examples: SHARP SH7000, ARM6/ARM7, Intel i960, TI TMS320C31, AT&T DSP3210 + Two processors platform examples: { SHARP SH7000 or ARM6/ARM7 or Intel 386EX or Motorola 683XX} and {ADI ADSP21XX or Motorola 5600X or TI TMS320C25/C5X or TI TSP50C10} * See also: L&H Windows TTS SDK * More Information: on the Lernout & Hauspie WWW pages: http://www.lhs.com/tts.html * Price: Unknown * Contact: Lernout and Hauspie Speech Products 20 Mall Road, 4th Floor Burlington, MA 01803, USA Ph: +1-617-238-0960, Fax: +1-617-238-0986 Email: sales@lhs.com WWW: http://www.lhs.com/ Lernout & Hauspie Text-to-Speech Windows SDK * Platform: Windows * Description: The L&H Text-to-Speech software developers kit is able to integrate text-to-speech technology with your own or existing PC applications under Microsoft Windows 3.1. This software will allow conversion of written text into clear human sounding synthetic speech. * Requirements: IBM-compatible PC 386 DX/33 + 8Mb RAM + MS DOS 5.0 + MS Windows 3.1 (or higher) + SoundBlaster compatible sound board. * See also: L&H TTS Products * More Information: on the Lernout & Hauspie WWW pages: http://www.lhs.com/tts.html * Price: Unknown * Contact: Lernout and Hauspie Speech Products 20 Mall Road, 4th Floor Burlington, MA 01803, USA Ph: +1-617-238-0960, Fax: +1-617-238-0986 Email: sales@lhs.com WWW: http://www.lhs.com/ Listen2 Text Reader * Platform: Windows * Description: Listen2 is a multi-voice, multi-language text reader. Listen2 comes in two versions, English only that uses high quality male and female voices, and the International version that can speak up to 5 different languages: English, German, French, Spanish or Italian, all in male voices. The basic International program comes with built-in English and additional language fonts can be purchased separately. The English version comes complete. Both programs are dynamically switchable and configurable. This means that you can press a hot key to speed up the speech, make it louder or quieter, etc., as it is reading a file. You can also insert flags in text files to make it switch voices or switch languages, depending on what version you have. Listen2 has all the features of the JTS Reader shareware program plus a few more. It will voice your reminder messages or appointment list on start-up. It will also speak a reminder message on shutting down. * WWW: A more complete description is available on the Listen2 web page * Contact: Tom Slemko: e-mail: tslemko@islandnet.com, or, JTS Micro Consulting Ltd 10931 Lytton Road, RR#4, Ladysmith, B.C., Canada, V0R 2E0 WWW: http://www.islandnet.com/jts/ Lucent Technologies Bell Labs Text-to-Speech system * Platform: UNIX and Win-95/NT * Description:Lucent Technologies provides a web site with demos and samples of their latest speech synthesis technology. The site has interactive demos in American English, German, and Mandarin Chinese, and the capability to adjust voice parameters on the fly. Pre-synthesized demos for French, Italian, Russian, and Romanian are also provided. The site includes downloadable papers with detailed system descriptions. * WWW: http://www.bell-labs.com/project/tts/ Macintosh Speech Output Applications * Platform: Macintosh * Description: A comprehensive list of Macintosh Speech Applications is provided by Kevin Lenzo at CMU: http://www.cs.cmu.edu/~lenzo/mac_speech_apps.html The Apple Speech WWW Site also has some useful information: http://www.speech.apple.com/ Speech Manager and PlainTalk * Platform: Macintosh * Description: Apple's text-to-speech system extensions that enable applications to perform text-to-speech conversion. The Speech Manager runs on most Macs, but PlainTalk (and the high quality voices) requires a 68020 Mac or better. * Availability: By anonymous ftp from: ftp://ftp.support.apple.com/pub/apple_sw_updates/US/Macintosh/Syst em/PlainTalk 1.4.1/ This directory contains subdirectories for recent versions of PlainTalk. The current release (PlainTalk 1.4.1) contains the English Text-To-Speech with about a dozen voices (English_Text-to-Speech.hqx: 5.3 MByte), Mexican Spanish (Mexican_Spanish_TTS.hqx: 2.8 MByte), and the English Speech Recognition software (English_Speech_Recognition.hqx: 2.3MByte). * Cost: Free * WWW: The latest information is available from Apple's WWW page for speech recognition and synthesis: http://www.speech.apple.com/ * Note 1: Check out Kevin Lenzo's list of Macintosh Speech Applications. * Note 2: Joshua Baer (josh@skyweyr.com) runs a mailing list for Plaintalk. For subscription and other information visit the Plaintalk Discussion List Home page * Contact: Apple Computer, Inc. 1 Infinite Loop, Cupertino, CA 95014, USA WWW: http://www.speech.apple.com/ Email: PlainTalk@atg.apple.com MacYack Pro * Platform: Macintosh * Description: MacYack Pro is a commercial speech package for Macintosh that uses the PlainTalk Text-to-Speech synthesis software. Features include: + Add speech to any word processor. + Hear notification dialogs and other dialog boxes. + See and hear a customized message at startup or shutdown. + Hear calculations instantly. + Correct pronounciation errors. + Create custom double-clickable "speech files." + Have speaking alert sounds. + Add speech to HyperCard stacks. + Use AppleScript to add speech to other programs. * Price: $29.95 for a limited time, reduced from $49.95 regular price. 30 days money back guarantee. * Contact: Scantron Quality Computers 20200 Nine Mile Rd. St. Clair Shores, MI 48080 Ph: 1-800-777-3642, Fax: 810-774-2698 E-mail: sales@sqc.com WWW: http://www.sqc.com/ Product Info: http://www.lowtek.com/macyack/ MBROLA: Free Speech Synthesis Project * Platform: Sun4, Sun/SunOS5.4, HP, VAX/VMS, DEC Alpha/VMS, PS/DOS, PS/Windows 3.1, PS/Windows 95, PC/Solaris2.4, PC/Linux, SGI INDY/IRIX, NeXT, and soon for Macintosh. * Description: MBROLA is a high-quality, diphone-based speech synthesizer which is available for free. It is provided by the TCTS Lab of the Faculte Polytechnique de Mons (Belgium) which aims to obtain a set a speech synthesizers for as many languages as possible which will be free of use for non-commercial, non-military applications. MBROLA 2.00 takes a list of phonemes as input, together with prosodic information (duration of phonemes and a piecewise linear description of pitch), and produces 16bit speech samples at the sampling frequency of the diphone database (typically 16kHz). (It is therefore NOT a Text-To-Speech (TTS) synthesizer, since it does not accept raw text as input.) Databases are now being prepared for English, Spanish, Italian, Dutch, and Romanian. Collaborations are welcome. More information can be found at the MBROLA project homepage. * Demonstration: WWW demo of MBROLA which compares the quality of PSOLA, MBR-PSOLA, LPC, and Hybrid Harmonic/Stochastic concatenative synthesizers is available at http://tcts.fpms.ac.be/synthesis/modelcmp.html. * Contact: Dr Thierry Dutoit Faculte Polytechnique de Mons, TCTS Lab, 31, bvd Dolez, B-7000 Mons, Belgium. Ph: +32-65-374133, Fax: +32-65-374129 e-mail: mbrola@tcts.fpms.ac.be WWW: http://tcts.fpms.ac.be/synthesis/mbrola.html Monologue for Windows from First Byte * Platform: Windows * Description: Monologue is a software program that reads text from the clipboard in Windows 16 or 32 bit applications. It can be found as a bundled product with many sound cards and multimedia general purpose computer systems. Monologue can add the element of speech to virtually any text oriented application. Any pronounceable combination of letters and numbers will be spoken clearly. It can be applied to tasks such as eyes-free proofreading, data verification (e.g. spreadsheets), reading E-mail and more. User-changeable parameters provide control over the sound quality by allowing for changes in pitch, and the speed of speech. An exception dictionary saves preferred pronunciation of words and abbreviations. Monologue Win32 now includes support for the Microsoft SAPI. Monologue male "SpeechFonts" are available for US English, British English, German, French, Latin American Spanish, Italian. A US English Female SpeechFont is also available. For more detailed information and examples go to the First Byte WWW pages. * Availability: Currently bundled with many sound cards and multimedia general purpose computer systems. For pricing, licensing details, and release information see the First Byte WWW pages or email info@firstbyte.davd.com. * See also: ProVoice Developer's Speech Toolkit from First Byte * Contact: First Byte 19840 Pioneer Ave., Torrance, CA 90503 Ph: 310-793-0610 Fax: 310-793-0611 Email: info@firstbyte.davd.com WWW: http://www.firstbyte.davd.com/ Narrator Translator Library * Platform: Amiga * Description: A US English text to phoneme translator, implemented as a resident software library, for use with the Amiga Narrator Device. This software was supplied as a standard part of the Amiga operating system software up to O.S version 2.04. (Translator version 37.1, 1991) Approximately 700 translation rules are used to create the 'ARPAbet' phonemes. This software is functional on all current Amiga systems (O.S. 3.1). * Availability: limited to pre-owned system software disks and unsold O.S upgrade kits (Pre-O.S. 2.1). Replacement Library: Translator42 * Platform: Amiga * Description: an independent replacement for the Commodore-supplied "translator.library" which is a part of the Narrator speech synthesis package. It implements multi-lingual text-to-speech for an Amiga. The translation rules for each language are defined in a plain text 'Accent' file. There is a provision for the selection of unique languages for text segments by inserting in-line markup codes in the text: e.g. "Hello there! \french{Bonjour} \deutsch{gute morgen}". 'Accent' files for American English, British English, Swedish, Maori, Finnish, German, Icelandic, Klingon, Polish, Italian, and Welsh languages included in the archive. * Availability: Amiga The most current version, 42.4, of the library and source are available by anonymous ftp from Aminet: ftp://ftp.doc.ic.ac.uk/pub/aminet/util/libs/translator42.lha ftp://ftp.doc.ic.ac.uk/pub/aminet/dev/src/tran42src.lha Narrator * Platform: Amiga * Description: Formant based speech synthesis. Includes a Engish-to-phoneme translation library, and a SPEAK: pseudo-device for speech output. * Hardware: Standard Amiga hardware * Availability: Part of AmigaOS * See Also: The Narrator Translation library TextToSpeech Kit * Platform: NeXT Computers * Description: The TextToSpeech Kit does unrestricted conversion of English text to synthesized speech in real-time. The user has control over speaking rate, median pitch, stereo balance, volume, and intonation type. Text of any length can be spoken, and messages can be queued up, from multiple applications if desired. Real-time controls such as pause, continue, and erase are included. Pronunciations are derived primarily by dictionary look-up. The Main Dictionary has nearly 100,000 hand-edited pronunciations which can be supplemented or overridden with the User and Application dictionaries. A number parser handles numbers in any form. A letter-to-sound knowledge base provides pronunciations for words not in the Main or customized dictionaries. Dictionary search order is under user control. Special modes of text input are available for spelling and emphasis of words or phrases. The actual conversion of text to speech is done by the TextToSpeech Server. The Server runs as an independent task in the background, and can handle up to 50 client connections. * Misc: The TextToSpeech Kit comes in two packages: the Developer Kit and the User Kit. The Developer Kit enables developers to build and test applications which incorporate text-to-speech. It includes the TextToSpeech Server, the TextToSpeech Object, the pronunciation editor PrEditor, several example applications, phonetic fonts, example source code, and developer documentation. The User Kit provides support for applications which incorporate text-to-speech. It is a subset of the Developer Kit. * Hardware: Uses standard NeXT Computer hardware. * Cost: + TextToSpeech User Kit: $175 CDN ($145 US) + TextToSpeech Developer Kit: $350 CDN ($290 US) + Upgrade from User to Developer Kit: $175 CDN ($145 US) * Availability: Trillium Sound Research 1500, 112 - 4th Ave. S.W., Calgary, Alberta, Canada, T2P 0H3 Tel: (403) 284-9278 Fax: (403) 282-6778 Order Desk: 1-800-L-ORATOR (US and Canada only) Email: TTSInfo@trillium.ab.ca Orator Text-to-Speech Synthesizer * Platform: SUN SPARC, Decstation 5000. Written in C, and therefore portable to other UNIX platforms. Some successful ports: HP, RS-6000, PC-Unix [Linux]. * Description: Sophisticated speech synthesis package. Has text preprocessing (for abbreviations, numbers), acronym rules, and human-like spelling routines. Natural-sounding synthesis based on demisyllable concatenation. Has high accuracy for pronunciation of names of people, places and businesses in America; good accuracy for English text; rules for stress and intonation marking; various methods of user control and customization at most stages of processing. A new version of the ORATOR system is under development. Both ORATOR and this new "ORATOR II" system are capable of general text synthesis. The ORATOR II system has a more natural-sounding voice. * Hardware: Runs on common SPARC or Decstation workstations, using their internal audio output capability. Recommend at least 16M of memory. * WWW: More detailed information plus examples of ORATOR synthesis are available on the ORATOR WWW pages: http://www.bellcore.com/ORATOR/ * Misc 1: A free demo cassette is available. * Misc 2: Examples of Orator are also available on the University of Birmingham Speech Synthesis "Museum" WWW site (see Q5.4). * Availability and Pricing: Contact Bellcore's Licensing Office Tel: 1-800-521-CORE (521-2673) Fax: 1-908-336-2559 Email: Anthony Lindsey: alin1@panix.com WWW: http://www.bellcore.com/ORATOR/ PAM - A Text-To-Speech Application * Platform: Windows * Description: PAM is a talking personal assistant and text reader application. It uses the ProVoice TTS package. PAM will verbally advise about appointments and reminder messages at specified times during the day. It can read text files, clipboard text, and text sent in DDE messages. Using the full verbal interface, PAM can be used by visually challenged individuals. Shareware - thirty day free trial. * Requirements: Any Windows sound card, speakers or headphones. Min. memory - 4 megs, 8 megs recommended. * WWW: A more complete description is available on the JTS homepage: http://www.islandnet.com/~tslemko/ * Availability: The shareware can be downloaded by ftp from ftp://ftp.islandnet.com/jts/pam_en3c.zip. The file size is approx. 1 MByte. * Price: $US40 for the registered version. * Contact: Tom Slemko: e-mail: tslemko@islandnet.com, or, JTS Micro Consulting Ltd 10931 Lytton Road, RR#4, Ladysmith, B.C., Canada, V0R 2E0 ProVerbe Speech Engine from ELAN Informatique * Platform: Windows 3.x, NT, 95, OS/2, Unix Solaris, Unix SCO and hardware * Description: The ProVerbe Speech Engine from ELAN Informatique produces natural sounding speech from written text. Naturalness is achieved by using the TD-PSOLA process from the CNET (France telecom's research lab.) which is based on the concatenation of elementary speech units (including diphones). Supported languages are British English, American English, Russian, German, French and Spanish. For multi-channel applications Elan Informatique also provides hardware platforms. Elan Informatique provides a SDK reference document (sdken.doc: WinWord6 format). * Demo versions: Telephone demonstration: +33-561 17 67 01 Sample sound files and demonstration software available. A CD-ROM with all these demonstrations is available by registration. * Contact: Elan Informatique 4 rue Jean Rodier, 31400 TOULOUSE FRANCE Contact person: Pierre Delrat Phone: +33-561-36-0777 Fax: +33-61-36-0770 BBS: +33-561-36-0788 E-mail: sales@elan.fr ftp: ftp://ftp.elan.fr WWW: http://www.elan.fr/ ProVoice Developer's Speech Toolkit from First Byte * Platform: ProVoice Developer's Toolkits are available for DOS, Windows 3.1, Windows 95, Windows NT, OS/2, and Macintosh. * Description: ProVoice allows programmers to add synthesized speech to their applications. Your program passes text strings to the ProVoice speech engine that translates text into audible speech. Male and/or female "SpeechFonts" are available for many languages; English, French, German, UK British English, Italian, and Spanish. ProVoice converts text to speech in two phases using a set of phonetic translation and pronunciation rules. First, the software analyzes and translates text into "sound descriptors", a phonetic language with pitch, duration, and amplitude codes which are needed to produce stress patterns in phrases and sentences. Rules are used to analyze words, numbers, and punctuation. The second phase converts the intermediate phonetic language in speech signals; algorithms drive distinct speech signals into smooth flowing, continuous, clear speech. Real time synchronization of mouth movement and word boundaries allows animation of a graphical talking character, or highlighting of displayed text as it is spoken. Necessary tools and examples are provided for programmers to manipulate the ProVoice speech technology; including installation instructions, extensive samples programs, and complete documentation. In addition, sample code is provided on disk to illustrate speech programming techniques. * Note 1: First Byte will perform custom work for embedded systems. * Note 2: ProVoice Windows includes support for the Microsoft SAPI. It will speak through any Windows-supported wave audio device. * Note 3: Distribution of ProVoice for commercial use is subject to execution of a Commercial Product Distribution License Agreement. * WWW: For more detailed information and examples go to the First Byte WWW page: http://www.firstbyte.davd.com/ * See also: Monologue for Windows from First Byte * Price and Availability: Contact First Byte * Contact: First Byte 19840 Pioneer Ave., Torrance, CA 90503 Ph: 310-793-0610, Fax: 310-793-0611 Email: info@firstbyte.davd.com WWW: http://www.firstbyte.davd.com/ RC Systems V8600/V8601 Text to Speech synthesizers * Platform 1: IBM PC: ISA card. * Platform 2: Interface to PC/104 standard microcontrollers. * Platform 3: Standalone (or embedded) hardware thru RS232 or parallel printer port or processor bus. * Description: Converts plain ASCII text to speech. Programmable voices, pitch rate, volume, etc. Built-in DTMF and tone generators. * Price: $151-$299 US (qty 1) * Contact: RC Systems 1609 England Avenue, Everett, WA 98203, USA Ph: (206) 355-3800 Fax: (206) 355-1098 Europe: +44181 539-0285 rsynth * Platform: Various (including Solaris2.3, SunOS4.1.3, HPUX, SGI Irix4.x, Linux) * Description: Public domain text-to-speech systm assembled from a variety of sources. It supports CMU and BEEP format dictionaries (as described in Q1.10) and now utilises stress marks in the dictionary in synthesising intonation. * Price: Free * Misc: Axel Belinfante has implemented a WWW rsynth demo: http://wwwtios.cs.utwente.nl/say. * Availability: by anonymous ftp from ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/synthesis/rsy nth-2.0.tar.Z ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/synthesis/rsy nth-2.0.tar.gz SENSYN speech synthesizer * Platform: PC/DOS/Windows, Macintosh, Sun, and NeXT * Rough Cost: $300 * Description: This formant synthesizer produces speech waveform files based on the (Klatt) KLSYN88 synthesizer. It is intended for laboratory and research use. Note that this is NOT a text-to-speech synthesizer, but creates speech sounds based upon a large number of input variables (formant frequencies, bandwidths, glottal pulse characteristics, etc.) and would be used as part of a TTS system. Includes full source code. * Availability: Sensimetrics Corporation Sidney Street, Cambridge MA 02139. Fax: (617) 225-0470; Tel: (617) 225-2442. Email: sensimetrics@sens.com WWW: http://www.sens.com/ SGI Developers Toolbox Synthesiser * Platform: SGI * Description: The SGI Developer Toolbox 4.0 CDROM contains a basicpublic domain text-to-speech program in the publics/speak directory. The directory includes man pages and source. * Availability: on the SGI Developer Toolbox 4.0 CDROM SIMTEL A wide range of speech related software, sound-blaster software and signal processing software for PCs is available on SimTel and its mirror sites. It can be obtained by ftp from: ftp://ftp.coast.net/SimTel/msdos/voice/ and is now on the WWW: http://www.acs.oakland.edu/oak/SimTel/win3/sound.html Voicemaker The archives include the program Voicemaker which synthesises speech from phonemes using "concatenation" of phonemes recorded by the user. Voicemaker is a freeware program. It requires an IBM or compatible, 512KB RAM, sound blaster compatible sound card. ftp://ftp.coast.net/SimTel/msdos/voice/vm110.zip Sound Bytes DeveloperUs Kit * Platform: Subroutine library for Windows, OS/2 and Macintosh * Hardware: Windows - 16 MHz 80386 (minimum) running Windows 3.1; 4 Mb RAM with at least 1.4 Mb RAM free. Disk space 1.4 Mb. OS/2 - 16 MHz 80386 (minimum) running OS/2 2.0 or above; 8 Mb RAM with at least 1.4 Mb RAM free. Mac - Any Mac with at least 2.5 Mb of RAM running 6.0.4 or higher. Telephone compatible. Compatible with commonly used sound cards. * Description: SBDK is a software-only sentence-level synthesizer that converts unrestricted English text (ASCII) into synthesized voice through diphone concatenation. SBDK utlizes parsing to incorporate the intonational and rhythmic patterns of normal speech. The developerUs kit includes two voices, one female and one male. The product has a 55,000-word built-in dictionary and a tool for creating customized user dictionaries. It converts numbers, dates, dollars, phone numbers and times to words, and has a SoundOut facility that provides a choice of pronouncing unknown words phonetically or spelling them out. Developers can vary voice pitch (130-220 Hz) and rate (65-200 wpm), synchronize speech to other events, have multiple channels of speech to the same or different boards, etc. Speech sampling options: 8-bit linear; 8-bit companded at 11 kHz (Windows); 8-bit mu-law PCM at 8 or 11 kHz; 16-bit linear at 11 kHz. * Cost: Sound Bytes may be licensed for internal use or resale. Site license fee= $3750. Resale or Internal runtime fees= 2% of net sales price per runtime sold, OR $150 per telephone port, OR per unit pricing for internal use determined case-by-case. * Misc: Demo disks are available for Windows and the Mac. * Availability: Natural Speech Technologies, Inc. Ph: (619) 457-2526. spchsyn.exe * Platform: DOS * Availability: By anonymous ftp as a self extracting DOS archive. ftp://evans.ee.adfa.oz.au/mirrors/tibbs/applications/spchsyn.exe * Requirements: May require special TI product(s), but all source is there. "Speak" - a Text to Speech Program * Platform: Sun SPARC * Description: Text to speech program based on concatenation of pre-recorded speech segments. A function library can be used to integrate speech output into other code. * Hardware: SPARC audio I/O * Availability: by anonymous ftp ftp://wilma.cs.brown.edu/pub/speak.tar.Z Speech Manager and PlainTalk * Platform: Macintosh * Description: Apple's text-to-speech system extensions that enable applications to perform text-to-speech conversion. The Speech Manager runs on most Macs, but PlainTalk (and the high quality voices) requires a 68020 Mac or better. * Availability: By anonymous ftp from: ftp://ftp.support.apple.com/pub/apple_sw_updates/US/Macintosh/Syst em/PlainTalk 1.4.1/ This directory contains subdirectories for recent versions of PlainTalk. The current release (PlainTalk 1.4.1) contains the English Text-To-Speech with about a dozen voices (English_Text-to-Speech.hqx: 5.3 MByte), Mexican Spanish (Mexican_Spanish_TTS.hqx: 2.8 MByte), and the English Speech Recognition software (English_Speech_Recognition.hqx: 2.3MByte). * Cost: Free * WWW: The latest information is available from Apple's WWW page for speech recognition and synthesis: http://www.speech.apple.com/ * Note 1: Check out Kevin Lenzo's list of Macintosh Speech Applications. * Note 2: Joshua Baer (josh@skyweyr.com) runs a mailing list for Plaintalk. For subscription and other information visit the Plaintalk Discussion List Home page * Contact: Apple Computer, Inc. 1 Infinite Loop, Cupertino, CA 95014, USA WWW: http://www.speech.apple.com/ Email: PlainTalk@atg.apple.com Text to phoneme program (1) * Platform: unknown * Description: Text to phoneme program. Based on Naval Research Lab's set of text to phoneme rules. * Availability: by anonymous ftp ftp://shark.cse.fau.edu/pub/src/phon.tar.Z Text to phoneme program (2) * Platform: unknown * Description: Text to phoneme program. * Availability: by anonymous ftp ftp://ftp.doc.ic.ac.uk/packages/unix-c/utils/phoneme.c.gz Text to phoneme program (3) * Description: A public domain version of the same Naval Research Lab text to phoneme rules. * Availability: By anonymous ftp ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/synthesis/english2phon eme.tar.gz Tinytalk * Platform: DOS / Windows??? * Description: Shareware package is a speech 'screen reader' which is used by many blind users. * Price: Tinytalk is now $150. There are package deals on Tinytalk with various speech synthesizers. * Availability: Tinytalk is available by anonymous ftp from the following site Files: ttexe167.zip and ttdoc167.zip (executable and documenation) ftp://ftp.netcom.com/pub/eb/ebohlman/ (Note: it is a busy ftp server.) * Contact: Eric Bohlman OMS Development 610-B Forest Ave., Wilmette, IL 60091 Ph: (800)831-0272 Fax: 708-251-5793 Outside North America: (708)-251-5787 Email: ebohlman@netcom.com TrueTalk * Platform: Sun Sparcstation 1+/2/LX/5/10/20 with SunOS 4.1.3, or SGI Indy/Indigo/Indigo2 with IRIX 5.2. More platforms in development. * Description: Personal TrueTalk, by Entropic Research Laboratory, Inc., is an all-software Text-to-Speech (TTS) system designed to voice-enable UNIX X-Windows workstations. It combines a graphical interface with a powerful TTS engine based on technology developed by AT&T Bell Laboratories. Features include: + Intelligible, prosodically natural speech. + Text taken from file input, highlighted X selections, the interface scratch pad, other programs connected through a TCP/IP socket, or Tcl/Tk applications via the Tk "send" mechanism. + Stop, pause and resume while speech is in progress. + Visual indication of corresponding text position when paused. + Nine speaking voices, with Male and Female versions of each voice. + Adjustable speaking rate and volume. + Supports drop-in text filters; "email" and "lively" examples included. + Audio output through workstation headphones or speaker. + Complete on-line documentation, including mouse-activated help windows. * Misc: A more detailed description of TrueTalk is available on the Entropic WWW server: http://www.entropic.com/truetalk.com * Availability: You can obtain Personal TrueTalk through the Internet. For details, see ftp://ftp.entropic.com/pub/truetalk/README.ptt Personal TrueTalk is available free of charge for evaluation purposes. You can fully-enable your evaluation copy at any time by purchasing a license key from Entropic. * Requirements: 12MB disk space, 8MB process size (24MB system RAM recommended). * Cost: US$495; US$395 academic * Contact: Entropic Research Laboratory, Inc., Washington, D.C. Voice: 1-800-ENTROPIC (North America), (202) 547 1420 Fax: (202) 547-6648 Email: truetalk@entropic.com WWW: http://www.entropic.com/ TruVoice from Centigram * Platform: Windows-NT, Windows 95, Windows 3.1 (limited release), Sun Solaris 2.x * Description: TruVoice., an advanced text-to-speech converter, is available for multiple environments. TruVoice converts text into spoken language. TruVoice adds intelligible, natural-sounding speech to sound enabled platforms. + Small, 1.5MB, memory footprint + Advanced text pre-processing + No vocabulary restrictions + User-definable pronunciation dictionary + Accurately pronounces surnames and place names + Preprocessor provides e-mail and spreadsheet reading capabilities and expands abbreviations. + Multiple languages available: American English, Latin American Spanish, German, French, Italian + Flexible pitch, volume and speech rate + Intonation support for punctuation + Supports navigational capabilities such as, pause, resume and jump forward / jump back with sentence or word boundaries More detailed information is provided in the brochure page on the Centigram WWW site. A demonstration of TruVoice is available on the Centigram WWW pages. * Cost: + Windows versions are $495 for the SDK + Solaris versions are $995 + Contact Centigram for other pricing. * Contact: TruVoice Sales Centigram Communications Corporation 91 East Tasman Drive, San Jose, CA 95134 Ph: (408) 944-0250 Fax: (408) 428-3732 Demo: 800-746 1632 Email: webmaster@centigram.com WWW: http://www.centigram.com/ WinSpeech * Platform: Windows * Description: WinSpeech is a text-to-speech application that reads text and produces speech to the audio output. Features basic text editing tools, talk from editing window, DDE server allows other Windows applications to send text for talking, coach mode for providing audio instructions throughout the program, dictionary editing tools for customizing pronunciation. WSPLIB text-to-speech DLL is a speech functions library for developers. More information available by email. * Requirements: System requirements: IBM PC or compatible computer with Windows 3.1 or higher. Sound card is recommended but not required. * Availability: Freeware available through the PC WholeWare WWW page. * Contact: PC WholeWare 33 Justin Street, Lexington, MA 02173, U.S.A. Email: info@pcww.com WWW: http://www.pcww.com/index.html WreadFiles: File reader for Commodore Amiga * Platform: Commodore Amiga * Description: WreadFiles is a vocal text file reader program for use on the Commodore Amiga. The text is printed to the screen and spoken. Features include: + Text is read in sentences rather than lines. + Dynamic Speech Correction on over 4000 word or word fragments. + Pronunciations for many place names, personal names, foreign names, foreign expressions and abbreviations. + Run from Workbench or CLI. + Used with A1000 (OS 1.3), A3000 (OS 2.04-2.1), and A4000 (OS 3.0) * Requirements: Standard Amiga Translator.library and Narrator.device required. 2.04 versions recommended. 1 Meg or more ram recommended. External speakers required. * Availability: No fee requested for non-commercial use. From: + GEnie: Page 555,3 File Number 24627 + Aminet ftp://ftp.wustl.edu/pub/aminet/util/misc/WreadFiles47.lha * Contact: Written by Michael L. Barlow Email: M.Barlow1@GEnie.geis.com or mbarlow@pacific.telebyte.com or MikeB@cuix.pscu.com ZMD Speech Synthesis "Speaky" Speech Synthesis from ZMD * Platform: DSP solution for platform independent speech synthesis implementation * Description: "Speaky" provides German speech synthesis system in a DSP solution. It includes pre-processing of input ASCII text with unlimited vocabulary, both parametric and non-parametric speech synthesis algorithms, and prosody modelling. More detailed information and audio samples can be found at the ZMD WWW Site. * Contact: Zentrum Mikroelektronik Dresden GmbH Grenzstrasse 28, D-01109 Dresden, Germany Ph: +49-351-8822-306, Fax: +49-351-8822-337 Email: assp@zmd-gmbh.de WWW: http://www.zmd-gmbh.de/ ZMD PCMCIA Speech Synthesis Card * Platform: MS-DOS, Windows * Description: Complete text-to-speech synthesis system for the German language with unlimited vocabulary using VOICE Processor "Speaky". The required pre-processing of the input ASCII text is performed by a software programm that is downloaded automatically from the PCMCIA Speech Synthesis Card during the card's initialising routine. Headphone or active loudspeaker can be connected directly for signal output. More detailed information and audio samples can be found at the ZMD WWW Site. * Requirements: PC Card slot, Card & Socket Services Software * Contact: Zentrum Mikroelektronik Dresden GmbH Grenzstrasse 28, D-01109 Dresden, Germany Ph: +49-351-8822-306, Fax: +49-351-8822-337 Email: assp@zmd-gmbh.de WWW: http://www.zmd-gmbh.de/ ___________________________________________________________________________ Speech Recognition comp.speech FAQ Section 6 * SpeechLinks: Speech Recognition * Q6.1: What is speech recognition? * Q6.2: How is speech recognition performed? * Q6.3: How can I build a simple speech recogniser? * Q6.4: References & books on speech recognition * Q6.5: Speech Recognition Hardware/Software * Q6.6: Speaker Recognition (Verification and Identification) * Q6.7: Integrated Speech Products ___________________________________________________________________________ Q6.1: What is speech recognition? Automatic Speech Recognition Automatic speech recognition is the process by which a computer maps an acoustic speech signal to text. Automatic speech understanding is the process by which a computer maps an acoustic speech signal to some form of abstract meaning of the speech. What does speaker dependent / adaptive / independent mean? A speaker dependent system is developed to operate for a single speaker. These systems are usually easier to develop, cheaper to buy and more accurate, but not as flexible as speaker adaptive or speaker independent systems. A speaker independent system is developed to operate for any speaker of a particular type (e.g. American English). These systems are the most difficult to develop, most expensive and accuracy is lower than speaker dependent systems. However, they are more flexible. A speaker adaptive system is developed to adapt its operation to the characteristics of new speakers. It's difficulty lies somewhere between speaker independent and speaker dependent systems. What does small/medium/large/very-large vocabulary mean? The size of vocabulary of a speech recognition system affects the complexity, processing requirements and the accuracy of the system. Some applications only require a few words (e.g. numbers only), others require very large dictionaries (e.g. dictation machines). There are no established definitions, however, try * small vocabulary - tens of words * medium vocabulary - hundreds of words * large vocabulary - thousands of words * very-large vocabulary - tens of thousands of words. What does continuous speech or isolated-word mean? An isolated-word system operates on single words at a time - requiring a pause between saying each word. This is the simplest form of recognition to perform because the end points are easier to find and the pronunciation of a word tends not affect others. Thus, because the occurrences of words are more consistent they are easier to recognise. A continuous speech system operates on speech in which words are connected together, i.e. not separated by pauses. Continuous speech is more difficult to handle because of a variety of effects. First, it is difficult to find the start and end points of words. Another problem is "coarticulation". The production of each phoneme is affected by the production of surrounding phonemes, and similarly the the start and end of words are affected by the preceding and following words. The recognition of continuous speech is also affected by the rate of speech (fast speech tends to be harder). ___________________________________________________________________________ Q6.2: How is speech recognition performed? A wide variety of techniques are used to perform speech recognition. There are many types of speech recognition. There are many levels of speech recognition / analysis / understanding. Typically speech recognition starts with the digital sampling of speech. The next stage is acoustic signal processing. Most techniques include spectral analysis; e.g. LPC analysis (Linear Predictive Coding), MFCC (Mel Frequency Cepstral Coefficients), cochlea modelling and many more. The next stage is recognition of phonemes, groups of phonemes and words. This stage can be achieved by many processes such as DTW (Dynamic Time Warping), HMM (hidden Markov modelling), NNs (Neural Networks), expert systems and combinations of techniques. HMM-based systems are currently the most commonly used and most successful approach. Most systems utilise some knowledge of the language to aid the recognition process. Some systems try to "understand" speech. That is, they try to convert the words into a representation of what the speaker intended to mean or achieve by what they said. ___________________________________________________________________________ Q6.3: How can I build a simple speech recogniser? QUICKY RECOGNIZER sketch: Doug Danforth provides a detailed account in article 253 in the comp.speech archives. A summary is provided below. It is also available by anonymous ftp ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/info/DIY_SpeechReco gnition This is a simple recognizer that should give you 85%+ recognition accuracy. The accuracy is a function of the words you have in your vocabulary. Long distinct words are easy. Short similar words are hard. You can get 98+% on the digits with this recognizer. Overview: * Find the begining and end of the utterance. * Filter the raw signal into frequency bands. * Cut the utterance into a fixed number of segments. * Average data for each band in each segment. * Store this pattern with its name. * Collect training set of about 3 repetitions of each pattern (word). * Recognize unknown by comparing its pattern against all patterns in the training set and returning the name of the pattern closest to the unknown. Many variations upon the theme can be made to improve the performance. Try different filtering of the raw signal and different processing methods. Public Domain Recognition Software Q6.5 contains information on public domain speech recognition software including: Lotec and Myers' Hidden Markov Model software. Discrete Hidden Markov Model Demonstration Software Hidden Markov Models (HMMs) are widely used in speech recognition systems. Joe Picone has put together some demonstration software for basic discrete HMMs including Viterbi and Baum-Welch training and evaluation, random sequence generation (generating data from a model), and model updating (useful for incremental training). There is a simple demo program that supports all of these modes from command line arguments. This allows experiments to test the classic coin-toss examples commonly described in textbooks. The code closely parallels the following textbook: * J.R. Deller, Jr., J.G. Proakis, and J.H.L. Hansen, Discrete-Time Processing of Speech Signals, MacMillan, 1993, ISBN: 0-02-328301-7. The code is written in C++ and is intended to facilitate learning and understanding of the algorithms. The code is available on the ISIP web site: http://www.isip.msstate.edu/software/ Lecture notes corresponding to the examples are also available: http://www.isip.msstate.edu/publications/1996/speech_recognition_short _course ___________________________________________________________________________ Q6.4: References & books on speech recognition * Product Reviews and Comparisons * Using Speech Recognition: Health Issues * On the WWW * Technology: General and Introductory * Technical * Course Notes * Bibliographies and Reference Lists Product Reviews and Comparisons * "Talk Show", Wayne Rash Jr., PC Magazine (USA), Dec 20, 1994. * "Seybold Report on Desktop Publishing" published a nine-page, head-to-head comparison of Dragon's DOS software with IBM's OS/2 software. March 7, 1994; Volume 8, Number 7; Pages 3-11; ISSN:0889-9762; Seybold Publications, P.O. Box 644, Media, PA 19063 USA, phone (610) 565-2480. * McGraw-Hill Inc.'s "BYTE, the Magazine of Technology Integration," published a two-page review of IBM's Personal Dictation System software. May 1994; Volume ?, Number ?; Pages 145-146; ISSN:0360-5280; Editorial, Executive, and Circulation address: One Phoenix Mill Lane, Peterborough, NH 03458 USA, phone ? Using Speech Recognition: Health Issues * The National Center for Voice and Speech provides some basic information on preserving "Vocal Health" on their WWW site: http://www.shc.uiowa.edu/hygiene/home.html * Voice Users Mailing List: detail in Q1.4.html of the FAQ. * Typing Injury FAQ: http://www.cs.princeton.edu:80/~dwallach/tifaq/ has a range of information on Typing Injuries, avoiding them, alternatives and more. * Typing Injuries Page: http://alumni.caltech.edu/~dank/typing-archive.html has links to dozens of useful resources. * Voice Problems -- Prevention and Correction: advice on preserving your voice with specific hints for using speech recognition. ftp://ftp.csua.berkeley.edu/pub/typing-injury/voice-problems * " Talking to a PC May Be Hazard To Your Throat", by Julie Chao in the Wall Street Journal. * " Talking to Computers Has its Hazards", by Gordon Arnaut in The Globe and Mail On the WWW * Survey of the State of the Art in Human Language Technology: Report edited by Ronald A. Cole et. al. with a section on Spoken Input Technologies. http://www.cse.ogi.edu/CSLU/HLTsurvey/ch1node2.html Technology: General and Introductory Some general introduction books on speech recognition technology: * Fundamentals of Speech Recognition; Lawrence Rabiner & Biing-Hwang Juang Englewood Cliffs NJ: PTR Prentice Hall (Signal Processing Series), c1993, ISBN 0-13-015157-2 * Speech recognition by machine; W.A. Ainsworth London: Peregrinus for the Institution of Electrical Engineers, c1988 * Speech synthesis and recognition; J.N. Holmes Wokingham: Van Nostrand Reinhold, c1988 * Speech Communication: Human and Machine, Douglas O'Shaughnessy; Addison Wesley series in Electrical Engineering: Digital Signal Processing, 1987. * Electronic speech recognition: techniques, technology and applications, edited by Geoff Bristow, London: Collins, 1986 * Readings in Speech Recognition; edited by Alex Waibel & Kai-Fu Lee. San Mateo: Morgan Kaufmann, c1990 Technical * Hidden Markov models for speech recognition; X.D. Huang, Y. Ariki, M.A. Jack. Edinburgh: Edinburgh University Press, c1990 * Speech Recognition: The Complete Practical Reference Guide; T. Schalk, P. J. Foster: Telecom Library Inc, New York; ISBN O-9366648-39-2; 377 pages; paperback only. Covers speech recognition in a telephony environment and wish to use call processing hardware based in PCs. It is written using Dialogic hardware as the example for the hardware. * Automatic speech recognition: the development of the SPHINX system; by Kai-Fu Lee; Boston; London: Kluwer Academic, c1989 * An Introduction to the Application of the Theory of Probabilistic Functions of a Markov Process to Automatic Speech Recognition, S. E. Levinson, L. R. Rabiner and M. M. Sondhi; in Bell Syst. Tech. Jnl. v62(4), pp1035--1074, April 1983 * Review of Neural Networks for Speech Recognition, R. P. Lippmann; in Neural Computation, v1(1), pp 1-38, 1989. * Automatic Speech and Speaker Recognition: Advanced Topics, C.H. Lee, F.K. Soong and K.K. Paliwal (Eds.), Kluwer, Boston, 1996. Course Notes * Joseph Picone of the Institute for Signal and Information Processing (ISIP) at Mississippi State University has put the course notes for "Fundamentals of Speech Recognition" on the WWW. The course covers background probability and phonetics/acoustics, speech signal analysis, dynamic programming, dynamic time warping, hidden Markov modelling, language modelling, neural networks, etc. The WWW sites provides the syllabus and lecture notes. WWW: http://www.isip.msstate.edu/publications/1996/ee_8993/ Bibliographies and Reference Lists * WWW searchable online-bibiliography for Phonetics and Speech Technology with more than 8000 entries. Provided by Institut fur Phonetik at Johann Wolfgang Goethe-Universitat Frankfurt. http://www.uni-frankfurt.de/~ifb/bib_engl.html * Computational Speech Processing: Speech Analysis, Recognition, Understanding, Compression, Transmission, Coding, Synthesis ; Text to Speech Systems, Speech to Tactile Displays, Speaker Identification, Prosody Processing : BIBLIOGRAPHY, by Conrad F. Sabourin, 1994, 2 volumes, 1187p, ISBN 2-921173-21-2, INFOLINGUA inc., P.O. Box 187 Snowdon, Montreal, H3X 3T4, Canada. See also: http://gomer.mlink.net/infolingua.html ___________________________________________________________________________ Q6.5: Speech Recognition Hardware and Software The number of speech recognition packages, and the information about the software is changing rapidly. Any help with keeping this information up to date will be appreciated. * Products in the FAQ * Speech Recognition Processors (ICs) * Recognition Information on the WWW * Speech Recognition Resellers and Value-Add In the FAQ: The following speech recognition software/hardware is described in the comp.speech FAQ. _Apple Macintosh_ * Digital Dreams Speech Recognition Plug-Ins * Dragon Dictation Products * Macintosh Speech Recognition Manager * PowerSecretary _Windows (including 95, NT, 3.1)_ * AT&T Watson Speech Recognition * Cambridge Voice for Windows * CustomVoice and CustomTelephone: A&G Graphics Interface Inc. * DragonDictate for Windows * Dragon Dictation Products * Dragon Developer Tools * Ficomp Interpreter 6000 * IBM VoiceType Dictation and Control * IN CUBE * Kurzweil Speech Recognition (2 products) * Lernout & Hauspie ASR SDK * Listen for Windows 2.0 from Verbex Voice Systems * Microsoft Speech Recognition * NCC Dictate * Phonetic Engine 500 (PE500) from Speech Systems, Inc. * Philips Speech Recognition (2 products) * ProNotes Voice Tools * PureSpeech * smARTspeak from Advanced Recognition Technologies, Inc. * Visual Voice from Stylus Innovation * VoiceAssist for Windows from Creative Labs, Inc. * VoiceServer for Windows * Whisper * WildCard Speech Products _DOS_ * DATAVOX - French * Dragon Developer Tools * Ficomp Interpreter 6000 * Jialong He's Speech Recognition Research Tool * smARTspeak from Advanced Recognition Technologies, Inc. * Votan VPC2100 Voice Card and VSP 1010 Speech Processor _OS/2_ * IBM VoiceType Dictation and Control _Unix_ * AbbotDemo * BBN Hark Telephony Recognizer * EARS: Single Word Recognition Package * Ficomp Interpreter 6000 * Hidden Markov Model Toolkit (HTK) from Entropic * IN CUBE * Jialong He's Speech Recognition Research Tool * Lotec Speech Recognition Package * Myers' Hidden Markov Model software * NICO Artificial Neural Network Toolkit * Nuance Speech Recognition System * PureSpeech * recnet _Integrated Circuits and Dedicated Hardware_ * HM2007 - Speech Recognition Chip * OKI VRP6679 - Speech Recognition Chip * Sensory Inc. Integrated Circuits * Speech Commander - Verbex Voice Systems * Voice Control Systems Recognition * VCS 2030 & 2060 Voice Dialer _Other Platforms_ * Simon Says (NeXT) * Voice Command Line Interface (Amiga) * Visus SpeechKit _Unknown_ * Berkeley Restaurant Project (BeRP) * Lernout & Hauspie ASR (3 products) * Voice-Trek 2.0 * Voicetek Corp. * Voice Processing Corporation Speech Recognition Product Line Speech Recognition Processors (ICs) Jean-Pierre Lereboullet has put together a detailed list of Voice Recognition Processors which covers about 15 ICs and pieces of related hardware (including D6106, HM2007, MSM6679, RSC-164, TC8860F/64F/65F, 5A128). The document is available on the comp.speech ftp server: ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/info/VoiceRecognitionProce ssors Recognition Information on the WWW In addition to the entries on speech recognition in this FAQ, the following WWW sites provide information on speech recognition: Commercial Speech Recognition: Russ Wilcox of PureSpeech Inc. http://www.tiac.net/users/rwilcox/speech.html Macintosh Speech Resources and Apps http://www.cs.cmu.edu/~lenzo/mac_speech_apps.html Speech Recognition Information: 21st Century Eloquence http://www.voicerecognition.com/ Applied Speech Technology Laboratory of CLSI at Stanford http://csli-www.stanford.edu/users/bscott/SRTech.html Speech Toys Speech Recognition Page http://www.speechtoys.com/spchtoys/sprec.html Speech recognition product lists: postings to comp.speech ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/info/SpeechRecognit ionProducts Search Alta Vista for Speech Recognition Search Lycos for Speech Recognition Yahoo pages on Speech Recognition http://www.yahoo.com/business/corporations/computers/software/v oice_recognition/ http://www.yahoo.com/Science/Computer_Science/Artificial_Intell igence/Natural_Language_Processing/Speech_Recognition/ Speech Recognition Resellers and Value-Added Services 1stVoice 2470 El Camino Real, Suite 110, Palo Alto CA 94306-1701 Ph: 415-857-1320, Fax: 415-856-6996 WWW: http://www.1stvoice.com/ Email: mail@1stvoice.com Dragon Dictation Products 21st Century Eloquence 325-A Royal Poinciana Plaza, Palm Beach, Florida 33480, USA Ph: 800-245-2133, Fax: 407-835-4901 WWW: http://www.voicerecognition.com/ Kurzweil, IBM VoiceType, Dragon, Kolvox Auscript (Australia) Suite 2, Level 3, 60-70 Elizabeth St, Sydney, NSW 2000, Australia Ph: +61-2-238 6565, Fax: +61-2-238 6566 WWW: http://www.auscript.com.au/ Dragon Systems BRITE WWW: http://www.brite.com/ Computer Telephony Integration & Interactive Voice Response DAX Systems, Inc. 30 Chapin Road, Unit 1201, P.O. Box 778, Pine Brook, NJ/USA 07058 Ph: +1-201-227-8111, Fax: +1-201-227-8197 Email: info@daxsystems.com WWW: http://www.daxsystems.com/ Computer Telephony and Integrated Voice Response HealthCare Resources 1444 Aviation Blvd, #103, Redondo Beach, CA 90278, USA Ph: +1-310-937-5156, Fax: +1-310-937-5159 EMail: Scalif@AOL.COM Power Secretary & Dragon Dictate. Specializing in: Medical/Dental, Motion Picture Industry, Carpal Tunnel related and Disabled Persons. O'Brien Resources Ph: (540) 347-4988 (Address unknown) Email: obrien@crosslink.net WWW: http://www.crosslink.net/~obrien/ Kurzweil Voice Recognition Products SCI VoiceAutomated 215 1/2 Main Street, Huntington Beach, CA 92648, USA Ph: 800-597-6600, Ph: +1-714-969-7632, Fax: +1-714-969-0122 http://www.voiceautomated.com/ IBM VoiceType, Kurzweil Voice, DragonDictate and Philips speech. Synapse 3095 Kerner Blvd., Suite S, San Rafael, CA 94901, USA Ph: (415) 455-9700, Fax: (415) 455-9801 Email: SYNAPSE_ADAPTIVE@msn.com WWW: http://www.synapseadaptive.com/ Dragon Systems, Kurzweil and IBM products. Talk Technology Ph: 1-800-270-1672, Fax: 1-516-360-1213 Email: info@talktechnology.com http://www.talktechnology.com/ Talk Technology, Inc. Tel: +1-718-745-9199, Fax: +1-718-499-6480 Email: mnm@pipeline.com WWW: http://www.usbusiness.com/talk/ Dragon Dictate and portable (notebook) solutions ToppCopy Telecom Email: ffalzett@toppcopy.com WWW: http://www.toppcopy.com/ Philips Digital Dictation VoiceWare Systems 230 California Street, Suite 410, San Francisco, CA 94111 Ph: (415) 433-2001, Fax: (415) 433-6909 Email: info@talk2type.com WWW: http://www.talk2type.com/home.htm IBM, Dragon Systems, Kurzweil Applied Intelligence, WildCard Technologies WorkLink A.D.A. Solutions by WorkLink 2566-A Telegraph Avenue, Berkeley, California 94704 USA Ph: 510-848-8363, Fax:510-848-7322 WWW: http://www.worklink.net/ Email: wayne@worklink.net Dragon Dictation Products AbbotDemo * Platform: SunOS4, IRIX, Linux, HU-UX * Description: Large vocabulary, speaker independent, continuous automatic speech recognition system. Uses recurrent neural networks and hidden Markov models with a 5,000 word vocabulary upgradable) and a trigram word grammar. Includes a front end for waveform capture and display (including spectrogram) and a graphical display of the phoneme representation as well as a rewriting display of the best guess word sequence. * Requirements: UN*X, X, 8 Mbyte free RAM, 486DX or faster processor, 16 bit soundcard, reasonable quality microphone and a copy of the Wall Street Journal newspaper. * Price: Free for non-commercial use * Availability: By anonymous ftp from ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/recognition/AbbotDemo * Note 1: This is not a complete system for dictation. * Note 2: At present there are no sources with this distribution. For sources for an earlier version see the recnet entry. * Note 3: Not supported. * Contact: AbbotDemo@compute.demon.co.uk Tony Robinson Cambridge University Engineering Department Trumpington Street, Cambridge, CB2 1PZ, UK Tel: +44-1223-332815 Fax: +44-1223-332662 AT&T Watson Speech Recognition * Platform: Windows 95/NT on a Pentium 75 Mhz or higher * Description: Watson is a software implementation of AT&T Bell Laboratories voice processing technology. Watson includes BLASR Speech Recognition and FlexTalk speech synthesis (see Q5.5). It requires no special hardware to run other than a standard sound card and/or phone card. Technical details for BLASR Speech Recognition include: + Compliant with Microsoft Speech API and Telephone API + Speaker independent, continuous speech recognition + Fast, run-time vocabulary change + Open mic and telephone line environments + SoundBlaster compatible sound card and drivers required + Subword models and whole-word digit models + Background, silence, and filler/garbage models + 50 word name vocabulary or 100 word phrase real-time recognition with 95% accuracy + Rejection of out-of-vocabulary words + American English only - other languages in development + Barge-in speech begin/end notification - requires hardware echo cancellation The AT&T Advanced Speech Products Group home page provides more detailed information including a Frequently Asked Questions list, information for application developers on the Independent Software Vendor (ISV) Program (including info on the SDK, licensing, and the training program). * Requirements: Uses 2 MB RAM, 10 MB Disk. Requires a Pentium 75 MHz or higher CPU (uses * Cost and Availability: WATSON is a software-based speech platform with a Software Developers Kit (SDK) that allows application developers to use voice processing in their applications. It is not available as a stand-alone product. Licensing information (inc. price) is provided in the AT&T Advanced Speech Products Group home page * See also: Watson FlexTalk speech synthesis in Q5.5, Microsoft Speech API, and Advanced Speech API. * Contact: AT&T Advanced Speech Products Group Suite 700, 44 East Mifflin Street, Madison, WI 53703, USA Ph: 1-800-5-WATSON, Fax: 1-608-259-2269 Email: aspg@attmail.com WWW: http://www.att.com/aspg/ BBN Hark Telephony Recognizer * Platform: Available for Unix-based workstation and PC platforms including IBM RS6000/AIX and Pentium/SCO Unix. * Description: Large vocabulary (2,000+ words), speaker independent, continuous ASR software. Specifically designed for large scale telephony applications. Using a client/server architecture, all features and capabilities are integrated in one software product instead of on separate boards. Very memory efficient, the Hark Telephony Recognizer runs in as little as 2MB of physical memory. Multiple recognizers can be run on a single platform. Uses Hidden Markov Model and phoneme-based BBN recognition algorithms. An API is provided for integration with existing applications. A developer's toolkit is available. * Price and availability: Price varies depending on vocabulary size. Version 3.0 available immediately. * Misc: BBN Hark provides application design and human factors consulting services. Regular monthly training classes on developing speech-enabled applications are held at BBN Hark's Cambridge (Mass) headquarters. * WWW: For additional information see BBN Hark's home page. * Contact: BBN Hark Systems 70 Fawcett Street, Cambridge, MA 02138, USA Tel: 617-873-4636 Fax: 617-873-2473 WWW: http://www.bbn.com/bbn_hark/HarkHome.html Berkeley Restaurant Project (BeRP) * Description: BeRP is a test bed for a speech recognition system being developed by the International Computer Science Institute in Berkeley, CA. BeRP is a medium-vocabulary, speaker-independent spontaneous continuous speech understanding system. BeRP functions as a knowledge consultant whose domain is the restaurants in the city of Berkeley. The system serves as a testbed for several research projects, including robust feature extraction, connectionist phonetic likelihood estimation, automatic induction of multiple pronunciation lexicons, foreign accent detection and modeling, advanced language models, and lip-reading. * Note: As far as I know the BeRP software is in-house software - that is, it is not made available for distribution. * More information: http://www.icsi.berkeley.edu/real/berp.html Cambridge Voice for Windows * Platform: Windows * Description: Speaker-independent recognition of continuous speech in real time. Vocabularies can range from small to very large (more than 60,000 word forms). Support is planned for languages including English, Danish, Dutch, French, German, Italian, Norwegian, Spanish, Swedish, and Japanese. The engine complies with the Microsoft Speech API. * Contact: Cambridge Group Research, Ltd. Box 7290, Buffalo Grove, IL 60089 Ph: (708) 821-1040, Fax: (708) 821-1041 E-mail: 76061.3350@compuserve.com CustomVoice and CustomTelephone: A&G Graphics Interface Inc. * Platform: Windows * CustomVoice: Speech recognition custom control for Visual Basic, Visual C++, Borland C++, and other development platforms that support *.VBX. Provides an engine/proprietary independent development platform for speech recognition. Currently supports ICSS, but should soon support other platforms. Includes a grammar debugger and parser APIs to parse spoken speech into useful data types. Requirements: 486/DX or better PC, Windows 3.1 or Windows for Workgroups, 8Mb RAM (minimum), SoundBlaster 16, microphone, and mouse. Supports Visual Basic, Visual C++, Borland C++, and Delphi. * CustomTelephone: Windows-based developers tool that allows programmers to build speech enabled "telephony" applications via standard custom control properties (VBX). It supports IBM VoiceType Application Factory (VTAF), a continuous speech, speaker independent speech recognizer, and supports voice response boards such as Dialogic. Comes with a VB custom control, pre-built grammar sets for common data types, an interactive grammar debugger to identify valid speech patterns, and parser API functions that convert recognized speech into data types supported by VB, C++ and Delphi. Includes sample applications with source code, and VBX, VCL and DLLs. Bundled with speech recognition engines. Requirements: 486/DX or better, Windows 3.1 or Windows for Workgroups, 8Mb RAM (minimum), SoundBlaster or compatible sound card, Dialogic D2X or D4X board, and mouse. Microphone and speaker optional. Supports Visual Basic, Visual C++, Borland C++, and Delphi. * Contact: A&G Graphics Interface 51 Gore Street, Cambridge, MA 02141-1213 , USA Ph: +1-617-492-0120, Fax: +1-617-427-2133 Email: customvc@world.std.com CompuServe: 74774,273 CompuServe ( GO SPEECH ) WWW: http://www.customvoice.com/ DATAVOX - French * Platform: PC / DOS * Description: Continuous speech - speaker independent or dependent. * Requirements: 2 PC format boards (RdF1000 and TdS 96/25) and an A/D - D/A module (ASA116) * Misc: Application software may dialog with DATAVOX through 2 types of interfaces : + Keyboard overlay: The application software may be used with any PC compatible package. No specific adaptation is necessary, you only need to define your configuration with the application software. + C library: Allows a user-written program to drive the recognition system. DATAVOX is based on the AMADEUS speech recognition software developed at LIMSI. It provides + Continuous speech recognition with 500 words speaker dependent, 50 words speaker independent (custom-made vocabulary). + Grammar of the application language (syntax acquisition, verification and simplification software). + Large vocabulary : DATAVOX can recognize vocabularies of several thousand words as long as there are no more than 500 words in the active vocabulary at any given node. It takes less than 1 second to change syntax and vocabulary. + Training controlled by the system (use of co-articulation models). + Response time less than 500 ms for any phrase length. + Synthetis (ADPCM) can be heard simultaneously while recognition is being carried out. * Contact: VECSYS Le Chene rond, 91570 Bievres, France Voice: 33 1 69 41 15 04, Fax: 33 1 69 41 24 30 Digital Dreams Speech Recognition Plug-Ins * Platform: Apple Macintosh * Description (General): A suite of speech plug-ins for the interactive multimedia market which enable developers to quickly incorporate speech recognition into their titles without having to resort to a low-level programming language, such as C. Speech plug-ins bridge the gap between a speech recognition API, such as Apple's PlainTalk Speech Recognition technology, and authoring/development environments, such as Macromedia Director or HyperCard. Digital Dreams currently offers Macintosh speech plug-ins for Macromedia Director and HyperCard. Support for other environments, including AppleScript, Apple Media Tool, Authorware, and Windows is being developed. Currently available for North American Adult English. More information is available on the Digital Dreams WWW site. * ShockTalk: is a combination of Netscape, ShockWave and Speech Recognition technologies for the Power Macintosh and Quadra AVs that enables you to navigate web sites and hyperlinks using spoken commands as well as create shockwave movies that respond to spoken user interactions. * Requirements: Power Macintosh (PowerPC w/ MacOS) Microphone (PlainTalk compatible) PlainTalk Speech Synthesis and PlainTalk Speech Recognition Netscape Navigator * Contact: Digital Dreams 4308 Harbord Drive, Oakland, CA, 94618, USA Tel: (510) 547-6929 Fax: (510) 547-6799 email: dreams@surftalk.com WWW: http://www.surftalk.com/ FTP: ftp://ftp.surftalk.com/ DragonDictate for Windows * Platform: Windows * Description: Information moved to the page on Dragon Dictation products including DragonDictate for Windows Dragon Dictation Products * Dragon NaturallySpeaking * DragonDictate for Windows * Dragon PowerSecretary * General Information Dragon NaturallySpeaking * Platform: Windows * Description: General purpose, continuous speech dictation system. Personal Edition has a 30,000 word active vocabulary and comes with a 200,000+ word pronunciation dictionary; users can also add their own words or phrases. More information on Dragon's NaturallySpeaking web site. * Requirements: 133Mhz Pentium, 32 MB RAM (Windows 95) or 48 MB RAM (Windows NT 4.0), supported sound card. * Price: see Dragon's NaturallySpeaking web site. * Related products: see general information below * Contact: see general information below DragonDictate for Windows * Platform: Windows * Description: Speech-to-text dictation system. Discrete dictation; continuous command/control; speaker-adaptive. Also provides mouse movement for hands-free operation of Windows. Comes with a 120,000 word pronunciation dictionary; users can also add their own words or phrases. Dictate directly into any application. Available in US and UK English, French, Italian, German, Spanish, and Swedish. Add-on vocabularies for medicine, law, business and finance, computers and technology, journalism. Available as DragonDictate Singles Editions (10,000 words active), DragonDictate Personal Edition (10,000 words active), DragonDictate Classic Edition (30,000 words active), DragonDictate Power Edition (60,000 words active). Includes Office97 support. More information on the Dragon Systems web site. * Requirements: 486/66, 7-10 MB dedicated RAM (depending on edition), Windows 3.1x, NT 3.51, or 95. Supported sound boards: Creative Labs Sound Blaster 16, Microsoft Windows Sound System, IBM M-Audio Capture/Playback Adapter, many notebooks with built-in audio. See Dragon Systems Compatibility list for details. * Price: Check at the Dragon Systems web site. * Related products: see general information below * Contact: see general information below Dragon PowerSecretary * Platform: Apple Macintosh * Description: Speaker dependent/adaptive system requiring words to be separated by short pauses. Available as PowerSecretary Power Edition, Personal Edition, PowerSecretary MED for Healthcare Professionals. Vocabulary: 30,000 - 60,000 at any one time, automatically selected from 120,000-word dictionary. * Requirements: Power Macintosh 6100, 7100, 8100, Performa 6100 series, Powerbook 540, 68040 class Macintosh such as Quadra 660AV, 700, 800, 840AV, 900, 950, Centris 650 and 660AV. Hard Disk with at least 25Mb free. System 7.5 or greater (Some systems require add-on hardware) * More information: PowerSecretary home page * Related products: see general information below * Contact: see general information below General Information Dragon Dictation Products * Dragon NaturallySpeaking * DragonDictate for Windows * Dragon PowerSecretary * General Information Dragon Developer Products * Dragon PhoneQuery * DragonXTools * Dragon SpeechTool * Dragon VoiceTools Related Web Sites * Simon Crosby's FAQ for DragonDictate Contact: * Dragon Systems, Inc. 320 Nevada Street, Newton, MA 02160, USA Tel: 1-617-965-5200 or 1-800-TALK-TYP Fax: 1-617-527-0372 Email: info@dragonsys.com WWW: http://www.dragonsys.com/ CompuServe: GO DRAGON Dragon Developer Tools * Dragon PhoneQuery * DragonXTools * Dragon SpeechTool * Dragon VoiceTools Dragon PhoneQuery * Platform: Windows NT * Description: Software for building voice response systems. Callers are able to do the following: Ask for information using completely natural and continuous language. Have a spoken dialog to fine tune a request. Request information to be faxed, sent by electronic mail, or read over the phone, using text-to-speech. More information on the Dragon Systems telephony pages. * Requirements: Pentium or Pentium Pro PC running Windows NT 4.0. Telephone interconnect requirements vary by application. * Related products: see general information below * Contact: see general information below DragonXTools * Platform: Windows * Description: VBX and OCX controls that allow an application to control DragonDictate's capabilities, ranging from small vocabulary command and control to customized large vocabulary dictation. More information is available on the Dragon Developer pages * Related products: see general information below * Contact: see general information below Dragon SpeechTool * Platform: Windows * Description: Create small, optimized vocabularies for your speech-enabled applications, or supplement DragonDictate's extensive built-in vocabularies with specialized terms and names. More information is available on the Dragon Developer pages * Related products: see general information below * Contact: see general information below Dragon VoiceTools * Platform: Windows, DOS * Description: integrate small-vocabulary speech recognition directly into your DOS and Windows 3.1x applications. More information is available on the Dragon Developer pages * Related products: see general information below * Contact: see general information below General Information Dragon Dictation Products * Dragon NaturallySpeaking * DragonDictate for Windows * Dragon PowerSecretary * General Information Dragon Developer Products * Dragon PhoneQuery * DragonXTools * Dragon SpeechTool * Dragon VoiceTools Related Web Sites * Simon Crosby's FAQ for DragonDictate Contact: * Dragon Systems, Inc. 320 Nevada Street, Newton, MA 02160, USA Tel: 1-617-965-5200 or 1-800-TALK-TYP Fax: 1-617-527-0372 Email: info@dragonsys.com WWW: http://www.dragonsys.com/ CompuServe: GO DRAGON EARS: Single Word Recognition Package * Platform: Linux and Unixs with the Voxware sound driver * Description: Intended as a limited ready-to-use single word recognizer. However, its design aims at being a platform for various kinds of methods used in speech recognition (SR). EARS is designed to be a flexible environment for recognition system components; for example, take this feature extractor and that recognizing method, and this list of words. New methods for single word recognition can be integrated easily, as EARS uses C++ abstract base classes. You speak the words you want to be recognized later. Your utterances can be saved to RIFF WAV files for inspection, change or delete them before they are further processed to the pattern files on which the recognizer is finally trained. As of version 0.20, the feature extractors are: Rasta-PLP, PLP, LPC, Mel-Cepstrum. The implemented recognizers are: DTW and non-recurrent neural nets on fixed-size sound patterns. * Requirements: Soundcard with mic * Misc 1: The current version is an Alpha release. * Misc 2: For more information subscribe to the EARS mailing list. Send email to majordomo@phil.uni-sb.de with "subscribe ears-list" in the body. * Misc 3: Niels Thorwirth (thorwir@pi4.informatik.uni-mannheim.de) has made changes to Version 0.14 which support the AF audio server software (see Q1.11) and the OGI Speech Tools (see Q1.9) so that EARS is more portable to other UNIX platforms. Available by email to Niels. * Requirements: Soundcard with mic * Availability: Source and Linux binaries are available by anonymous ftp ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/recognition/ears-0.26. tar.gz ftp://sunsite.unc.edu/pub/Linux/apps/sound/speech/ears-0.26.tar.gz * Contact: Ralf W. Stephan: ralf@ark.franken.de Ficomp Interpreter 6000 * Platform: DOS, Windows 3.1, Win95, Win NT, UNIX * Description: Ficomp Systems, inc., is a systems integrator that has developed commercial speaker-dependent, continuous-speech recognition applications for use in high noise environments on several platforms. Applications are specialized in the finance industry for exchange floors, banks and brokerage firms. * Contact: Ficomp Systems, Inc. Ph: (732) 274-2600, Fax: (732) 274-2601 117 Docks Corner Road, Dayton, NJ 08810 E-Mail: fsisales1@aol.com WWW: http://www.ficompsystems.com/ HM2007 - Speech Recognition Chip * Platform: Intergrated circuit. * Description: HM2007 is a 48-pin single chip CMOS voice recognition LSI circuit with on-chip analog front end, voice analysis, recognition process and system control functions. A 40 word isolated-word voice recognition system can be composed of an external microphone, keyboard, SRAM and a few other components. When combined with a microprocessor, an intelligent recognition system can be built. A demo board for this chip is being distributed by The Summa Group. * Cost: Approx US$16 for the HM2007 and US$160 for the demo board. * Misc: Jean-Pierre Lereboullet's document on Voice Recognition Processors provides additional information on the HM2007. * Producer: HUALON Microelectronic Corp. USA Tel: (415) 288 0390 Fax: (415) 288-0399 * Distributor 1: Marywale Engineering Company Tel: (602) 247 4451 Fax: (602) 247 6167 Email: meco@indirect.com * Distributor 2: The Summa Group Limited One California Street, Suite #1940, San Francisco, CA 94111 Ph: (415) 288-0390 * Distributor 3: Images Company 39 Seneca Loop, Staten Island, NY 10314, USA Ph: +1-718-698-8305, Fax: +1-718-982-6145 Sells single piece quanities of HM2007 48Pin Dip Chip and HM2007 52 Pin PLCC style chip. Sells HM2007 Demo Kits unassembled $100.00 and assembled $135.00 (using 48 Pin dip chip) Entropic's HTK (HMM Toolkit) * Platform: Range of Unix platforms. * Description: HTK is a software toolkit for building continuous density HMM based speech recognisers. It consists of a number of library modules and a number of tools. Functions include speech analysis, training tools, recognition tools, results analysis, and an interactive tool for speech labelling. Many standard forms of continuous density HMM are possible. Can perform isolated word or connected word speech recognition. It van model whole words, sub- word units. Can perform speaker verification and other pattern recognition work using HMMs. HTK is now integerated with the ESPS/Waves speech research environment which is described in Section 1.9. * Misc 1: The availability of HTK changed in early 1993 when Entropic obtained exclusive marketing rights to HTK from the developers at Cambridge. * Misc 2: More detailed information on HTK is available from the Entropic WW server: http://www.entropic.com/htk.html * Cost: On request. * Contact: Entropic Research Laboratory, 600 Pennsylvania Ave, S.E. Suite 202, Washington, D.C. 20003, USA Phone: (202) 547-1420. email - info@entropic.com WWW: http://www.entropic.com/ IBM VoiceType Dictation * Platform: OS/2 and Windows * Description: IBM VoiceType Dictation supports speech input at 70-100 words a minute and can be used to control your desktop and applications. Isolated-word, speaker-dependent system using a speech adapter card. Available for U.S. English, U.K. English, French, German, Italian, Spanish and Arabic. Provided with a general office vocabulary and support for major OS/2 and Windows applications. Additional specialised vocabularies are available: + US: Legal, Emergency Medicine, Radiology and Journalism + UK: Legal + IT: Radiology * Requirements: See http://www.software.ibm.com/workgroup/voicetyp/vtprod13.html * Cost: See http://www.software.ibm.com/workgroup/voicetyp/vtordna.html * Misc: An IBM VoiceType Dictation FAQ is supported by UltraMedia Systems International (a distributor of IBM VoiceType): http://www.infi.net/~ums/ibmfaq.htm * Demo software: Available on the IBM WWW site: http://www.software.ibm.com/workgroup/voicetyp/vtcust1.html * Contact: US Ph: 1-800-TALK-2-ME or 1-914-766-1900. Email: talk2me@vnet.ibm.com WWW: http://www.software.ibm.com/workgroup/voicetyp/vtcust1.html IBM VoiceType Control (US Only) * Platform: OS/2 and Windows * Description: VoiceType Control is a speech recognition navigator that lets you control programs by speaking. VoiceType Control converts voice commands to keystroke macros. The program provides speaker independent, continuous speech recognition, so you do not have to train the program for your specific speech patterns. * Requirements: ? * Cost: ? * Demo software: http://www.software.ibm.com/workgroup/voicetyp/vtcust2.html * Contact: US Ph: 1-800-TALK-2-ME or 1-914-766-1900. Email: talk2me@vnet.ibm.com WWW: http://www.software.ibm.com/workgroup/voicetyp/vtcust2.html IN CUBE * Platform: Three versions for Windows 95, Windows NT and Sun SPARCstations * IN CUBE for Windows 95: Developed for general purpose Windows 95 users. It is packaged for online distribution with a full working demo and an option to register and unlock the full product. The system uses Command Corp's Mark II continuous speech recognition engine and handles changable lexicons of up to 75 commands. + Price: $49.95 US + Requirements: 386/25MHz processor or better, Microsoft Windows 3.1 or later, Windows compatible sound card or built-in audio, and microphone. + Availability: http://www.commandcorp.com/cci/win95.html Demo mode available. * IN CUBE Mark II Pro for Windows NT: IN CUBE is a continuous realtime speech recognition system developed to provide a fast and convenient means of window navigation and voice macro command input for command intensive applications like CAD and publishing. Speaker-dependent training and ability to add new commands and macros. + Price: $495 including the PRO 8 microphone. $540 including the MT 858 desk microphone. + Requirements: Windows NT, Windows NT-compatible audio board (16-bit audio recommended). + Availability: http://www.commandcorp.com/cci/pront.html Demo available. * IN CUBE Voice Command for Sun SPARCstations: Provides continuous realtime speech recognition system for window navigation and voice macro command input to the workstation. Speaker-dependent training and ability to add new commands and macros. An IN CUBE Application Programming Interface is available with a library of linkable object modules is available for developers. + Price: $495 per seat. The developer's API sells for $695. + Requirements: SUN OS 4.1.x or Solaris 2.x with OpenWindows and Motif. Works with all audio-equipped SPARCs and clones. Models range from SPARCStation 1s to SPARCStation 20s. + Availability: http://www.commandcorp.com/cci/in3sparc.html A free 5 day evaluation license is available. * Contact: Command Corp. Inc., 3761 Venture Drive, PO Box 956099, Duluth, Georgia, 30136, USA Ph: +1-770-813-8030 Email: in3@commandcorp.com WWW: http://www.commandcorp.com/incube_welcome.html Jialong He's Speech Recognition Research Tool * Platform: SUN SPARC (SunOS), PC (MSDOS) * Description: This is a speech recognition research tool. it contains a feature extraction program and three speech recognizers: a DTW recognizer, discrete didden Markov model (DHMM) based recognizer and Continuous density hidden Markov mode (CHMM) with Gaussian mixture functions based recognizer. The utilities are grouped as: + feature -- extract featue vectors from a speech signal (MFCC etc.) + dtwcmp -- dynamic time-wapping (DTW) comparision. + gensym -- turn vector sequences to discrete observation symbols. dhmm -- discrete HMM training program. dtest -- DHMM companion test program. + chmm -- continuous density HMM training program. viterbi -- CHMM companion test program. Note, this is a research tool not a complete speech recognition system. * Availability: By anonymous ftp: MSDOS Version UK: ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/recognition/s pchtool.zip Germany: ftp://ftp.informatik.uni-ulm.de/pub/NI/jialong/spchtool.z ip Sun SPARC version, compiled with GNU C UK: ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/recognition/s pch_sun_v1.tar.gz Germany: ftp://ftp.informatik.uni-ulm.de/pub/NI/jialong/speech_sun _v1.tar.gz * See also: Jialong He's Speaker Recognition (Identification) Tool * Contact: Jialong He email: jialong@neuro.informatik.uni-ulm.de Kurzweil Voice for Windows * Platform: Windows 3.1 or later * Description: Kurzweil Voice for Windows is a dictation product enabling the user to create text and enter data by speaking to Windows-based applications. System is adaptive but requires no initial training. Users can choose either 30,000 or 60,000 word active vocabulary. Application command translation templates for popular Windows application such as WordPerfect, 1-2-3, Organizer, Word (30+ applications are listed on the Kuzweil WWW pages). More detailed information is available on the Kurzweil WWW pages. * Requirements: 486DX/33 or higher, 8 or 16 MB dedicated memory (depends on vocabulary, 30 MBs dedicated disk space, VGA or higher, Kurzweil-supplied microphone and DSP board. * Contact: Kurzweil Applied Intelligence, Inc. 411 Waverley Oaks Road, Waltham, MA 02154 USA Phone: 1-800-380-1234 Email: info@kurzweil.com WWW: http://www.kurzweil.com/ Kurzweil Clinical Reporter * Platform: Windows 3.1 or later * Description: Kurzweil Clinical Reporter is a voice-activated clinical reporting system for computer-based patient records. The family of products includes: + VoiceEM for emergency medicine + VoiceEM/TR for triage reporting + VoiceRAD for diagnostic imaging and radiology + VoicePATH for surgical and anatomical pathology + VoiceMED for Primary Care for family medicine, internal medicine and pediatrics + VoiceORTHO for office-based orthopaedic surgery + VoiceCATH for invasive cardiology + VoiceReport for general reporting * More information: from the Kurzweil WWW pages: http://www.kurzweil.com/medical/ * Contact: Kurzweil Applied Intelligence, Inc. 411 Waverley Oaks Road, Waltham, MA 02154 USA Phone: 1-800-380-1234 Email: info@kurzweil.com WWW: http://www.kurzweil.com/ Lernout & Hauspie ASR 1000/T and 1000/M [Note: L&H asr200/A is described below.] * L&H asr1000/T: ASR for the Telephony and Telecommunications Market * L&H asr1000/M: TTS for the Computer and Multimedia Market * Description: Automatic speech recognition software providing continuous speech recognition, isolated word recognition, keyword spotting or continuous digits recognition. The engine is speaker independent, and phoneme-based with optimization for commonly used words. General features include: + Languages available: US English, German, French, Spanish (Castilian), Dutch. + Available vocabulary: >100,000 words. + Line adaptation. + Rejection of out of vocabulary/grammar words. + N-best alternatives for isolated word recognition and keyword spotting. + Push to talk. * asr1000/T + Single channel platform examples: Motorola 56156, TI TMS320C2X/C3X/C5X + Multi-channel platform examples: TI TMS320C3X/C5X, AT&T DSP32C/3210, Motorola 96000 + Input: 8 kHz telephone sampling * asr1000/M + Single processor platform examples: Intel 486/Pentium + Input: 8 kHz telephone or 11 kHz microphone sampling * See also: L&H ASR SDK for Windows * More Information: on the Lernout & Hauspie WWW pages: http://www.lhs.com/asr.html * Cost: Unknown * Contact: Lernout & Hauspie Speech Products 800 West Cummings Park, Suite 3100 Woburn, MA 01801, USA Tel: (617) 238 0960 Fax: (617) 238 0986 Email: sales@lhs.com WWW: http://www.lhs.com/ Lernout & Hauspie ASR 200/A for the Automotive and Industrial Market * Description: Automatic speech recognition software providing isolated word recognition, keyword spotting and alphabet recognition (optional). This engine is robust, speaker independent and word based. Other features: + Vocabulary: 100 words US English + Voice activation detection + Response time + Platform examples: Analog Devices ADSP2101/5 + Input: 8 kHz telephone or microphone sampling * See also: L&H ASR SDK for Windows * More Information: on the Lernout & Hauspie WWW pages: http://www.lhs.com/asr.html * Cost: Unknown * Contact: Lernout and Hauspie Speech Products 20 Mall Road, 4th Floor Burlington, MA 01803, USA Ph: +1-617-238-0960, Fax: +1-617-238-0986 Email: sales@lhs.com WWW: http://www.lhs.com/ Lernout & Hauspie ASR SDK * Platform: Windows * Description: Windows based Software Development Kits are available for integrating automatic speech recognition technology with Windows based PC applications. * Requirements: IBM-compatible 486 DX/33 MHz + 8 MB RAM + MS DOS 5.0 + MS Windows 3.1 (or higher) + Sound Blaster compatible sound board. * See also: L&H ASR Products * More Information: on the Lernout & Hauspie WWW pages: http://www.lhs.com/asr.html * Contact: Lernout and Hauspie Speech Products 20 Mall Road, 4th Floor Burlington, MA 01803, USA Ph: +1-617-238-0960, Fax: +1-617-238-0986 Email: sales@lhs.com WWW: http://www.lhs.com/ Listen for Windows 2.0 from Verbex Voice Systems * Platform: Windows * Description: Listen for Windows Version 2.0 is a Speaker Independent software product that provides continuous speech recognition for Windows applications. The product works with most industry standard sound cards and PCs with inbedded audio chips. Listen for Windows comes with over 16,000 commands in speech interfaces for over 40 software applications, such as MS Office, Lotus SmartSuite,Quicken, etc. The Listen Command Editor allows a user to change or add commands to existing speech interfaces or create new speech interfaces for most Windows applications. More detailed information is available on the Verbex Listen for Windows page. Verbex also sells Verbal Advantage Voice Browser for controlling a web browser, Verbal Advantage DeskTop for controlling desktop applications. * Requirements: 486/25SX PC or higher * Pricing and Availbility: See the Verbex ordering page for pricing. Verbex products are available over the web or can be shipped. Microphones available from Verbex. * Demo: A "Freeware" demo is available from the Verbex WWW site demo page. * Contact: Verbex Voice Systems 1090 King Georges Post Rd., Bldg 107, Edison NJ 08837, USA Ph: 1-800-ASK-VRBX, (908) 225-5225, Fax:(908) 225-7764 WWW: http://www.verbex.com/ Lotec Speech Recognition Package * Platform: Sun * Description: Public domain speech recognition software. Operates from input in Sun audio format (.au files) and outputs word hypotheses and time labelling data. The software includes programs to collect speech samples, a labeller, a "featurizer" which parameterises speech files, a word spotter and the recogniser. The software can real time recognition on a Sparc 10 for small vocabularies. * Requirements: Sun SPARC audio input and a "decent" microphone Sun multimedia demo software (in /usr/demo/SOUND) and X. * Availability: By anonymous ftp ftp://ftp.sanpo.t.u-tokyo.ac.jp/pub/nigel/lotec/lotec.tar.Z * Contact: Nigel Ward: _nigel@sanpo.t.u-tokyo.ac.jp _ Macintosh Speech Recognition Manager * Platform: Macintosh * Description: supports developers who wish to add speech recognition to existing Macintosh applications. Provides speaker independent recognition and robustness to noise. Apple's Speech home page provides developer information and the complete speech recognition and synthesis synthesis SDKs. The recognition SDK includes samples code, control panels, interfaces, documentation and the recognizer. * Availability: under licensing conditions from the Macintosh Speech Developer's page http://www.speech.apple.com/speech/dev/dev.html. * Requirements: Power Macintosh with 16-bit sound, System 7.5, and a PlainTalk Microphone or equivalent * Cost: Free * See also: Macintosh Plaintalk and Speech Manager (Q5.5). * Note: Check out Kevin Lenzo's list of Macintosh Speech Applications. * Contact: Apple Computer, Inc. 1 Infinite Loop, Cupertino, CA 95014, USA WWW: http://www.speech.apple.com/ Email: PlainTalk@atg.apple.com Microsoft Speech Recognition Microsoft Dictation Research Demonstration * Platform: Windows 95 or Windows NT 4.0 * Description: A free demonstration of research technology that enables a computer to transcribe what you speak into Windows applications such as email and word-processors. Features of the demo software include: + 60,000 word vocabulary with the ability to add new words + High recognition accuracy + Works with any Windows 5application + "Dictation Pad" provides enhanced dictation features + "IntelliSense" converts spoken numbers and times automatically + Compatible with the Microsoft Speech API * Requirements: Windows 95 or Windows NT 4.0, Pentium 90 or better (RISC builds are available), 16 megabytes of RAM on Windows 95, Sound card with 16 kHz 16 bit input signals, High quality close-talk microphone, Speakers. * Availability: Free demo software is available at: http://www.research.microsoft.com/research/srg/install.htm * More information: http://www.research.microsoft.com/research/srg/ Microsoft Command and Control Engine * Platform: Windows 95 * Description: Provides command and control speech recognition using SAPI (the Microsoft Speech API) and "Whisper", Microsoft's speech recognition technology. Features include: + Speaker independent, continuous, sub-word modeling, context free grammars + Has its own letter-to-sound rules means it can recognize any words in a grammar. + North American English + PC microphone and telephone speech recognition with high performance + Word spotting option + Results objects containing top-N choices, segmentation, and confidence + Written to SAPI, the Microsoft Speech API. * Requirements: Windows 95 or Windows NT 4.0, Pentium 60 or better. (RISC builds are available), 1.5 megabyte working set, 16 kHz or 8 kHz input signals, 6 megabytes on disk, Requires Microsoft Speech SDK to use. * Availability: Free demo software is available at: http://www.research.microsoft.com/research/srg/install.htm * More information: http://www.research.microsoft.com/research/srg/ Myers' Hidden Markov Model software * Platform: Unix * Description: Hidden Markov model software for automatic speech recognition. C++ code that implements a basic left-right hidden Markov model and corresponding Baum-Welch (ML) training algorithm. It is meant as an example of the HMM algorithms described by L.Rabiner and others. The code was built in order to learn how HMM systems work and we are now offering it to the net so that others can learn how to use HMMs for speech recognition. Keep in mind that ease of understanding was our primary concern, not efficiency. The code can be used to build an experimental speech recognition systems using "train_hmm" and "test_hmm", and can be used in conjunction with written tutorials on HMMs to understand how they work. * Availability: By anonymous ftp from the comp.speech archive site. There are two files in the directory + ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/recognition/ The files are + hmm.README + hmm-1.03.tar.gz * Contact: Richard Myers: rmyers@isx.edu NCC Dictate * Platform: Windows * Description: NCC Digital DictateTM is an add-on, enhanced interface for use with IBM's VoiceType(TM) Dictation for Windows and various Windows 3.1 applications (e.g. MS Word, WordPerfect). Digital DictateTM provides faster corrections and dictation rates and various other features. This version is not a stand alone product; it requires VoiceTypeTM Dictation to provide the speech recognition engine and the Windows application. Features include: + Direct dictation into Windows applications with access to all functions while dictating. + Versions for MS Word, WordPerfect, Ami Pro, and other Windows applications. + Speech enabled editing. + Capability to save speaker models and defer corrections. + Microphone "pause and restore" functions controlled with speech commands. + Add-on vocabularies for legal, medical, science and business. + SWITCH-ITTM foot pedal control or CardSwitchTM infrared wireless control available which switch between dictation and proofing/correction modes. * Requirements: IBM's VoiceTypeTM Dictation for Windows; a computer system meeting VoiceTypeTM Dictation for Windows requirements; VoiceTypeTM Dictation Adapter. * Availability: Through computer dealerships. * Price: $US295 * Contact: NCC Incorporated 5808 E. Turquoise, Scottsdale, AZ 85253 Ph: (602) 922-6236 Fax: (602) 596-9050 NICO Artificial Neural Network Toolkit * Platform: UNIX (ANSI C source code) * Description: The NICO Toolkit is an artificial neural network toolkit specifically designed and optimized for automatic speech recognition applications. Networks with both recurrent connections and time-delay windows are easily constructed. The network topology is flexible -- any number of layers is allowed and layers can be arbitrarily connected. Tools for extracting input-features from the speech signal are included as well as tools for computing target values from standard phonetic label-files. * Availability: Through the NICO homepage (http://www.speech.kth.se/NICO/index.html) or the download page. * Contact: Nikko Strom, nikko@speech.kth.se Nuance Speech Recognition System * Platform: UNIX-based workstations including Sun and SGI. * Description: The Nuance Recognizer features client-server architecture with multiple recognizers available on a single processing platform. Primarily developed for telephony-based applications, the system accepts speaker-independent, continuous speech and supports very large vocabularies. Included is a "template matching" natural language capability for identifying the meaning of speech. A toolkit is available for use in developing a wide variety of speech recognition applications. * Price and availability: Contact Nuance * Contact: Nuance Communications 1380 Willow Road, Menlo Park, CA 94025, USA Ph: +1-650-847-0000, Fax: +1-650-847-7979 WWW: http://www.nuance.com/ OKI VRP6679 - Voice Recognition Processor * Platform: Intergrated circuit. * Description: Speech recognition IC. 25 words max. Speaker independent recognition capability. Recognition rate quoted as 97% in a noisy environment (e.g. a car). * Misc: Alias MSM6679 * Misc 2: More information is provided in Jean-Pierre Lereboullet's document on Voice Recognition Processors. * Cost: Approx US$20. Demo board $876 * Availability: OKI Semiconductor and OKI Distributors Corporate Headquarters 785 North Mary Avenue, Sunnyvale, CA, 94086 2909 Tel: (408) 720 1900, Fax: (408) 720 1918 Phonetic Engine 500 (PE500) from Speech Systems, Inc. * Platform: Windows * Description: Speaker independent, 40,000 word vocabulary, continuous speech recognition for MS Windows. Grammars with high perplexity possible. Includes noise rejection. Uses proprietary DSP board. * Cost: Prices in US$ - quantity one. The PE500 SDK is $995.00 including board, microphone, and runtime software. Runtime only is $595.00. SpeechWizard(r) adds speech input to existing Windows applications, $295.00. Two-day training: $295.00 with purchase, $595.00 without. * Misc: The user defines the grammar of allowed utterances and must write software to invoke the board driver functions that control recognition. The user must also write software to collect/parse/interpret the ASCII text strings returned when recognition succeeds. * Misc 2: SSI now offers speech application development services. * Contact: Speech Systems, Inc. 2945 Center Green Court South Boulder, CO 80301-2275, USA Tel: 303.938.1110 Fax: 303.938.1874 http://www.speechsys.com Philips Speech Recognition (2 products) SpeechMagic: Dictation * Platform: Windows 3.1 and higher * Description: A continuous speech recognizer providing a 64,000 word vocabulary, speaker adaptation and multiple languages. SpeechMagic is currently available for English and German. SpeechMagic acts as a server application, processing speech input and providing text output. Uses an add-on ISA compatible recognition accelerator board. SpeechMagic provided a correction editor, editing and playback of recordings, and a vocabulary manager for entering new words, abbreviations, macros and special transcriptions (e.g. for foreign words). Windows DDE support and a native API are provided for integration. * Hardware Requirements: IBM compatible personal computer (486DX/ 66 MHz or higher), minimum 16 MB of RAM, hard disk capacity > 500 MB, and a Philips LFH 6210 Accelerator Board. * More Information: For more information visit the SpeechMagic WWW page or the Philips Speech home page. Speech Processing System 6000s (Europe only) * Description: Dictation of medical findings using continuous speech recognition. Designed for German speaking radiologists and encompasses the complete radiology vocabulary. The authors use dictation stations (PCs) which are fitted with microphones. The transcriptionists use editing stations (also PCs) which are additionally fitted with headphones and footswitches. The SP6000s has a single speech recognition unit serving all users, and it offers automatic data transfer as well as the advantages of digital dictation functions. For more information visit the Philips SP6000s WWW page. * More Information: For more information visit the Philips SP6000s WWW page or the Philips Speech home page. Dragon PowerSecretary * Platform: Apple * Description: Information moved to the page on Dragon Dictation products including Dragon PowerSecretary (Previously Articulate PowerSecretary.) ProNotes Voice Tools * Platform: Windows * Description: ProNotes Voice Tools are designed to bring the speech recognition capabilities of the IBM VoiceTypeTM Dictation System for Windows into any program without the need for the programmer to directly interface with the speech engine at the API level. There are five tools, as described below, which are all available in three forms: Visual Basic(TM) Custom Controls (known as VBXs), 16-bit OLE Custom Controls, and 32-bit OLE Custom Controls. The tools are intended for use by Windows(TM) developers working with Windows 3.1(TM), Windows for Workgroups 3.11(TM), Windows NT 3.51 Workstation(TM), and Windows 95(TM). The custom controls can be utilized with any application development environment which supports the use of such controls (e.g. Visual Basic and Visual C++). Playback and Record An object which allows developers to use the IBM Speech Engine to record and play back sound files. Can be used to add voice prompts and to allow end users to record and playback sound files. Voice Button An object having standard button properties and behavior, which can additionally be controlled by voice. The button can also be used as a label or a 3D panel. Dictation Window A text box that allows free dictation, voice macro utilization, and correction by voice. Each Dictation Window has access to global and context sensitive vocabularies for both command and dictation. There are three correction modes. Voice List Box Has standard list box properties and behavior, but can additionally be controlled by voice. A user can select items by pronouncing the entry's text or the entries can be numbered and selected accordingly. Voice Navigator Provides navigation by voice within an application developed with the Voice Tools, between voice-enabled objects described above, as well as some standard objects found within the application. * Requirements: Hardware: 80486/33 DX or higher, 60MB hard disk space for IBM VoiceType Dictation software, 10MB hard disk space for ProNotes Voice Tools, 3.5" floppy, VGA (or compatible), 16MB RAM, IBM VoiceType Dictation adapter, microphone, and speakers. Software: DOS version 6.0 or later, with SHARE.EXE running, Windows 3.1 or later, IBM VoiceType Dictation software, any programming environment or system compatible with Visual Basic or OLE Custom Controls. * Price: Unknown * Contact: Pronotes, Inc. 1546 Magee Avenue, Philadelphia, PA 19149, USA Ph: 800-70-NOTES or +1-215-533-8569, Fax: +1-215-533-1276 Email: proinfo@pronotes.com WWW: http://www.pronotes.com/ PureSpeech 2.0 Recognition Engine * Platform: Windows 3.1, Windows 95, Unix, Dialogic Antares DSP * Description: Speaker-independent, continuous speech, large active vocabulary speech recognition engine for American English, UK English, French, German and Spanish. Permits on-the-fly additions to the vocabulary using phonetic models and telephone or wideband microphone input. Flexible grammar, natural language processing, discourse models. Software only with a small RAM/CPU footprint. Can be used as a voice user interfaces (VUI's) for PC software applications. Can also be used for high-volume call center telephony, especially in banks, finance and other specialized applications. A toolkit for the Dialogic Antares is available. * Availability: PureSpeech is not available as a stand-alone product. It is available embedded in Windows-based software or as a toolkit. * Contact: PureSpeech, Inc 100 Cambridge Park Drive, Cambridge, MA 02140, USA Ph: (617) 441-0000 Fax: (617) 441-0001 Email: amy@speech.com WWW: http://www.speech.com/ recnet * Platform: UNIX * Description: Speech recognition for the speaker independent TIMIT and Resource Management tasks. It uses recurrent networks to estimate phone probabilities and Markov models to find the most probable sequence of phones or words. The system is a snapshot of evolving research code. There is no documentation other than published research papers. The components are: + A preprocessor which implements many standard and many non- standard front end processing techniques. + A recurrent net recogniser and parameter files + Two Markov model based recognisers, one for phone recognition and one for word recognition + A dynamic programming scoring package. The complete system performs competatively. * Cost: Free * Requirements: TIMIT and Resource Management databases * Contact: Tony Robinson: _ajr@eng.cam.ac.uk_ * Availability: by anonymous ftp ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/recognition/r ecnet-1.3.tar.Z Sensory Inc. Integrated Circuits * Platform: Integrated circuits * Description: Sensory's low cost high quality Interactive Speech line of speech recognition IC's are designed for consumer telephony products, portable consumer electronics, and other consumer applications. Technologies available include speech recognition (speaker-independent and speaker-dependent), speaker verification, speech/music synthesis, digital record/playback, and general product control on one chip. Development tools and demonstration units are available. Detailed product information on the Interactive Speech chips is available from the Sensory Circuits WWW site. * Contact: Sensory, Inc. 521 E. Weddell Drive, Sunnyvale, CA 94089 Ph: +1-408-744-9000, Fax: +1-408-744-1299 Email: Sales@SensoryInc.com WWW: http://www.sensoryinc.com/ Simon Says (NeXT) * Platform: NeXT * Description: Provides the ability to link commands to spoken phrases. * Availability:By anonymous ftp. Simon Says demo ftp://ftp.informatik.uni-muenchen.de/pub/comp/platforms/next/Audio /audio-apps/SimonSaysDemo.1.5.1.N.b.tar.gz Readme file ftp://ftp.informatik.uni-muenchen.de/pub/comp/platforms/next/Audio /audio-apps/SimonSaysDemo.1.5.1.README * Contact: Metrosoft 710 13th Street, Suite 310 X, San Diego, California 92101 Ph: 619.488.9411 Fax: 619.488.3045 Email: info@metrosoft.com [NeXTmail welcome] smARTspeak from Advanced Recognition Technologies, Inc. * Platform: Windows, Windows 95, DOS, and General Magic It also works on the following Processors/Microcontollers: Intel's 80 x 86, Intel's 8031, 8051, Motorola's 68000, and Hitachi's SH1, SH3, SH8. * Description: smARTspeak is suited to voice command and control applications, such as voice dialing in cellular and desktop telephones, or voice command operation in computers and multimedia products. It uses a compact (10KB size on 16 bit machines), fast, user dependent recognition engine. smARTspeak can recognize any language in any accent. ART recently completed a Software Developer Kit (SDK) for smARTspeak, running under Windows 3.1 or higher which allows the voice recognition engine to be used within Windows Applications. More detailed information on smARTspeak and the SDK is available on the ART WWW pages. * Availability: Currently liscensed to other equipment manufacturers (OEMs), system integraters, software, and application developers, and value added resellers (VARs) who port are technology into their product. * Contact: Advanced Recognition Technologies, Inc. International Office: 43 Brodezky Street, POB 39918, 61398 Tel Aviv, lsrael Ph: 972-3-642-7242, Fax: 972-3-642-5887 Email: 100274.3223@Compuserve.com WWW: http://www.artcomp.com/ US Office: 9574 Topanga Canyon Blvd. Chatsworth, CA 91311, USA Ph: 818-678-3999, Fax: 8181-678-3994 WWW: http://www.artcomp.com/ Speech Commander - Verbex Voice Systems * Platform: Various: external hardware with serial port connection * Description: A hand-held (portable) device about the size of a paperback book which provides speaker-dependent continuous speech recognition. The active vocabulary is dependent on the model chosen and can vary from 300 to 10,000 active words. The device connects through a serial port, so it can be connected to a wide range of computers. It comes with a battery pack. * Contact: Verbex Voice Systems 1090 King Georges Post Rd., Bldg 107, Edison NJ 08837, USA Ph: (908) 225-5225, Fax: (908) 225-7764 Email: sales@listen.verbex.com WWW: http://www.verbex.com/ 'Speech Recognition Expert' Toolkit for Windows * Description: Provides an object-oriented development tool designed to rapidly build speech enabled applications without writting source code. Currently supports IBM's VoiceType Application Factory. Future versions to support other platforms. Includes BlackBox library and Custom Grammar Tools. * Requirements: Layout for Windows from Objects, Inc. * Price: $US349 + Shipping/Handling * Contact: Speech Technologies, Inc. P.O. Box 3905 Naperville, IL 60567-3905 CompuServe @102147,3521 Ph: (708)983-7634 Visual Voice from Stylus Innovation * Platform: Microsoft Windows * Description: Visual Voice is a toolkit for building Windows-based voice processing and telephony applications including interactive voice response (e.g. touch-tone banking), fax-on-demand, and voice mail. Visual Voice can be used to add voice recognition to your telephony applications. Voice Recognition (VR) Support for Visual Voice is exposed as a standard VBX control and provides one or more voice recognition "resources" to your application. Applications can dynamically assign resources across several voice lines. Voice recognition is either "discrete" or "continuous". Discrete recognition is slightly more accurate and requires the speaker to pause briefly between words. Continuous recognition provides a natural way to enter information by speaking without pauses. Three configurations are supported: Software-Only Solution The software only solution uses Telaccount's SpeechEasy technology for discrete recognition using your PC's CPU. A vocabulary is included with digits, basic command words and more. Hardware-Assisted Solution with Dialogic AEB boards Discrete voice recognition in over 25 languages using Dialogic D/41D voice boards and the Dialogic VR/40 board. Vocabularies are included with digits, basic command words, voice mail vocabulary and more. Hardware-Assisted Solution with Dialogic PEB boards. Use the VR control with any Dialogic PEB-based voice board, such as the D/12x or D/24x, to access voice recognition resources from your phone lines. This requires a Dialogic VRP board with either 1 to 4 VRM/40 modules (4 channel discrete voice recognition modules) and/or 1 to 4 VRM/2C modules (2 channel continuous voice recognition modules). You can have up to 4 modules on each VRP: 4 VRM/40s for 16 channels of discrete voice recognition; 4 VRM/2Cs for 8 channels of continuous recognition; or a combination. Over 25 languages supported. Includes vocabularies as described above. * Pricing: Unknown * Availability: From Stylus Innovations Inc. or from the distributors listed on the Stylus WWW pages. * Misc: More detailed technical information, slide show demonstration software is available on the Stylus home page. * Contact: Stylus Innovation Inc. One Kendall Square, Building 300, Cambridge, MA 02139 Ph: (617) 621 9545, Fax: (617) 621 7862 WWW: http://www.stylus.com/ Compuserve forum: GO STYLUS Email: info@stylus.com Voice Command Line Interface * Platform: Amiga * Description: VCLI will execute CLI commands, ARexx commands, or ARexx scripts by voice command through your audio digitizer. VCLI allows you to launch multiple applications or control any program with an ARexx capability entirely by spoken voice command. VCLI is fully multitasking and will run in the background, continuously listening for your voice commands even while other programs are running. Documentation is provided in AmigaGuide format. VCLI 6.0 runs under either Amiga DOS 2.0 or 3.0. * Requirements: Supports the DSS8, PerfectSound 3, Sound Master, Sound Magic, and Generic audio digitizers. * Availability: by ftp from wuarchive.wustl.edu in the file systems/amiga/incoming/audio/VCLI60.lha and from amiga.physik.unizh.ch as the file pub/aminet/util/misc/VCLI60.lha * Contact: Author's email is RHorne@cup.portal.com Voice Control Systems Continuous Speech Recognition * Description: Voice Control Systems (VCS) continuous speech recognition is a proprietary phonetic recognizer based on technology developed at VCS over the last 17 years. It is robust for applications such as the "hands-free" automotive environment or telephone networks, both wireless and wireline. VCS speech recognition is used by many developers and manufacturers in telecommunications. VCS technology is a software-based capability which VCS has currently developed for a limited number of processing environments. VCS offers "off-the-shelf" capabilities for the TI-C3X and C4X DSPs with other hardware platform support planned for the future. As a benchmark, today's VCS continuous technology requires about 1/2 of a 33Mhz TMS320C31. VCS continuous technology is available in cellular and wireline based libraries for continuous digit input in approximately 15 languages. VCS continuous recognition is a modified HMM decision strategy built upon the foundation of VCS phonetic "front end". * Availability: VCS continuous technology is available today in software form from VCS or implemented in hardware or speech systems from VCS distributors including Dialogic Corporation, Brite Voice, Intervoice, Periphonics, and Syntellect. * Cost: Software royalties are volume based and range from per unit costs of $500 per recognizer to less than $5 in large quantities. * See also: the VCS Phonetic Dictionary Recognizer and VCS Isolated Word Speech Recognition below, and the VCS 2030 & 2060 Voice Dialers. * Contact: Voice Control Systems, Inc. 14140 Midway Rd., Dallas, Tx. 75244, USA Ph: +1-214-386-0300, Fax: +1-214-386-5555 Email: sales@vcsi.com WWW: http://www.voicecontrol.com/ Voice Control Systems Phonetic Dictionary Recognizer * Description: This recognizer is based upon a HMM type recognition strategy coupled with the VCS "front end" (feature extraction software). The HMM modeling is based upon the basic phonetic building blocks in each language. In American English this is approximately 43 units. The recognition vocabulary is built up by combining these units into word models. By building the words in this way new recognition vocabularies may be constructed. The phonetic assembly can also be used for "word spotting" recognition libraries. * Platform: This VCS recognition software runs on the TI TMS320C30 DSP. Two recognizers can operate on a single 55mhz C30. Currently the software may be purchased as an Enhanced Technology from VCS to run on the Dialogic VR/160p speech recognizer board. The hardware is purchased from Dialogic, with the "Enhanced" software purchased from VCS. Up to four phonetic recognizers can run on a single 160; one per VRM2C (C30-33mhz DSP) daughtercard. * Note: This recognizer is in its late "beta" stage of development and is available for U.S. English vocabularies. Other languages are presently under development. * Price: VCS software is priced at $350 per recognizer for unit quantities with volume discounts available. * See also: VCS Continuous Recognition above, VCS Isolated Word Speech Recognition below, and the VCS 2030 & 2060 Voice Dialers. * Contact: Voice Control Systems, Inc. 14140 Midway Rd., Dallas, Tx. 75244, USA Ph: +1-214-386-0300, Fax: +1-214-386-5555 Email: sales@vcsi.com WWW: http://www.voicecontrol.com/ Voice Control Systems Isolated Word Speech Recognition * Description: Voice Control Systems (VCS) isolated word recognition using VCS phonetic recognizer technology. It is robust in demanding environments such as the "hands-free" automotive environment, telephone networks, wireless or wireline. Capabilities include speaker-independent, speaker-dependent and speaker-adaptive recognition. Libraries are available for 45+ languages and custom vocabulary development services are available. The technology is suited for many applications including: + Desktop computing: such as keyboard accelerators orinteractive multimedia. + Network telephony: such as automating operator functions or voice dialing. + Computer telephony: such as remote access to a personal computers. + Automotive accessory control: such as voice activated cellular phones or other automotive accessories. + Consumer electronics: such as voice controllers for video games or VCRs and televisions. * Platform: Include Intel-X86, TI-C5X, C3X, C4X and C2X, OKI 6679, and NEC-V20 and V30, and can operate on 16 bit microcontrollers. As a benchmark, 8 recognizers can run on an Intel 486-33 DX. * Availability: The technology is available under software licenses direct from VCS or by purchasing hardware from an OEM. VCS OEMs include: Dialogic, Oki Semiconductor, Intervoice, Periphonics, etc. * Cost: VCS isolated word recognition software is available under a volume pricing license agreement. Small quantity royalties are in the $500.00 per recognizer range while large (millions) quantity royalties are less than $1.00 per recognizer. * See also: VCS Continuous Speech Recognition and VCS Phonetic Dictionary Recognizer above, and the VCS 2030 & 2060 Voice Dialers. * Contact: Voice Control Systems, Inc. 14140 Midway Rd., Dallas, Tx. 75244, USA Ph: +1-214-386-0300, Fax: +1-214-386-5555 Email: sales@vcsi.com WWW: http://www.voicecontrol.com/ Visus SpeechKit * Platform: NeXT * Description: SpeechKit is based on SPHINX, a speaker-independent, 1000 word or so, continuous speech recognition system which allows you to incorporate speech recognition into your applications. You can design your vocabulary and grammars. * Contact: Visus - no address or phone provided. A possible contact is Robert Brennan at Carnegie Mellon University. email: Robert_Brennan@cmu.edu VCS 2060 Voice Dialer VCS 2030 Voice Dialer * Platform: Stand-alone hardware, TMS320C5X based with VCS phonetic speech recognition and CELP speech compression. * Description: The VCS 2060 is a telephone dialing system which recognizes 50 names - and speed dials the associated telephone number. The VCS 2030 has 20 memories. Users use speaker-independent recognition to select the "call", "program", or "list" menu, then place a call, enroll a new memory, or listen to playback of entries in the phonebook. Enrollment is simple and includes a "name tag" enrollment pass so that when one selects an entry to call, the selection is confirmed by repeating the memory's associated name tag, e.g. "calling Pete". The system uses both speaker-independent and speaker-dependent technology from Voice Control Systems, Inc. * Installation: The VCS 2060 can be installed in series (RJ-11) with one phone for single phone operation or installed in parallel (RJ-31) to provide voice dialing from every phone in a house. * Cost: Standard retail prices: + VCS 2030 Voice Dialer - $269.00 + VCS 2060 Voice Dialer - $299.00 * Availability: From catalogs or direct from Voice Control Systems. Voice Control Systems 14140 Midway Rd., Dallas, Tx. 75225, USA Ph: 800-VCS-7525, Fax: +1-214-386-5555 Email: sales@vcsi.com WWW: http://www.voicecontrol.com/ Voice-Trek 2.0 * Platform: Unknown. * Description: VoiceTrek is primarily used by the United States Postal Service to sort mail. Tardis Technology Inc. was created to develop and market applications that utilize speech recognition. They do consulting work as well as turnkey systems. * Contact: Tardis Technology Inc., Voice Recognition Div. 6444 E. Spring St., #286, Long Beach, CA 90815-1500, USA Phone: +1-310-497-0077, Fax: +1-310-497-0080 VoiceAssist for Windows from Creative Labs, Inc. * Platform: Windows * Description: Seeking a description. * Availability: VoiceAssist preview software is available from the Creative Labs VoiceAssist home page. * Contact: Creative Labs, Inc. Ph: 1-800-998-1000 (Sales) Ph: 1-800-998-5227 (Product info and dealer referrals) CompuServe: support forum: GO BLASTER WWW: http://www.creaf.com/ VoiceServer for Windows * Platform: Windows * Description: Speaker dependent, each with an independent directory. Isolated words. Up to 1000 words/user, 300 words/window. 1 word occupies 2Kb on hard disk. Can be used to control Windows applications by issuing voice commands instead of menu selection. * Rough Cost: 292 Pounds(UK) * Requirements: None * Misc: Price includes a half-sized AT voice card (including a DSP), software, documentation & a microphone (attachable to keyboard or speaker). A light-weight high-spec headset is an optional extra. * Contact: Mark Redwood Applied Voice Technologies 26 Danbury Street, Islington, London, UK, N1 8JU Ph: + 44 71 454 1224 : Fax: + 44 71 454 1225 Voicetek Corp. * Platform: Unknown. * Description:Voicetek Corporation provides voice processing solutions, training and consulting services and an object-oriented, graphical Generations Platform for development of integrated computer telephony systems. * Contact: Voicetek Corporation 19 Alpha Road, Chelmsford, MA 01824, USA Ph: +1-508-250-9393, Fax: +1-508-250-9378 WWW: http://www.voicetek.com/ Votan VPC2100 Voice Card and VSP 1010 Speech Processor * Platform: DOS * VPC2100 Voice Card: a hardware and software system based on the TMS320C10. providing continuous speech recognition. The VPC2100 consists of a circuit board, microphone, speaker, software, and documentation. It is designed to add voice I/O and telephone management capabilities to the PC/AT and compatibles. Features: + Voice store-and-forward at 4- to 16.4-Kb/s speed + Speaker-independent speech recognition (0-9, YES, NO) + Continuous speaker-dependent speech recognition + Telephone interface, pulse or tone dialing, call progress, and DTMF + Software for development, voice mail, telephone management, and VoiceKey + High-level applications-generator software * Votan VSP 1010 speech-processor board: can service a single voice channel, providing recognition, voice output, and telephone interfacing. Digital signal processing is performed by a TMS320 integrated circuit. * Costs: Unknown * WWW: http://www.ti.com/sc/docs/dsps/develop/3rdparty/vot.htm * Contact: Votan Division, MOSCOM Corporation 6920 Koll Center Parkway, Suite 214, Pleasanton, CA 94566, USA Ph: +1-510-426-5600, Fax: +1-510-426-6767 Voice Processing Corporation Speech Recognition Product Line * Platform: Unknown. * Description: Voice Processing Corporation (VPC) supplies automated speech recognition systems. VPC's products are used in the telecommunications, cellular and personal computer markets to enable computers to understand human speech. The company's VPro product line is sold to original equipment manufacturers (OEMs), value added resellers (VARs), system integrators and application developers. VPC's speech recognition systems are currently used in applications such as voice mail, voice activated dialing, interactive voice response, and command and control of personal computers. The following are descriptions of the Voice Processing Corporation's VPro Product Line: VProContinuous, VPro/XD, VPro/RT, VProCel, VProSpeller, VProPRL, VPro hardware platforms, and the application Osprey. More information is available on these products at the VPC WWW site: http://www.vpro.com/ * VProContinuous(TM) is a speaker-independent, continuous digit recognizer. It recognizes digit strings spoken in a continuous manner, by any caller, without unnatural beeps or pauses. VProContinuous uses out-of-vocabulary rejection and word spotting technologies to reject extraneous words and phrases often spoken by callers. The VProContinuous vocabulary consists of the words "zero" through "nine," "yes," "no," and "oh." The product is language-independent. American English, Australian English, Brazilian Portuguese, Canadian French, Castilian Spanish, French, German, Italian, Mexican Spanish, Portuguese, Swiss German and U.K. English versions are available. * VPro/XD(TM) is a discrete or multiword speech recognizer for extra-demanding applications and/or vocabularies. This robust discrete product recognizes isolated discrete utterances (words or very short phrases). VPro/XD utilizes proprietary out-of-vocabulary rejection and word-spotting technologies. VPro/XD is speaker-independent and includes Talkover capability allowing speech-interrupt over prompts. Pre-trained vocabulary libraries are available in American English, Australian English, Brazilian Portuguese, Canadian French, Castilian Spanish, Central American Spanish, German, Italian, Mandarin Chinese, Mexican Spanish, Portuguese, Swiss German and UK English. Pre-trained vocabularies consisting of voice mail words, voice dialing words, call control words, banking, and emergency words are available in American English (both cellular and land-line). * VPro/RT(TM) is a discrete speech recognizer for rapid training of vocabularies in the field. This robust discrete product recognizes isolated discrete utterances. Application designers and end-users define the vocabulary of their choice and train the system in real-time either prior to system start-up, or adapting on-the-fly while the system is running live. Vocabularies can be subset, and applications involving thousands of words can be developed quickly. VPro/RT, which also supports Talkover, is suited to speaker-dependent recognition tasks, such as the personal directory of names in a voice-activated dailing application. VPro/RT is also good for applications that require speaker-independent vocabularies to be developed quickly in the field or those that require many vocabularies. VPro/RT can also be used as a tool for quick prototyping of applications. * VProCel consists of speaker-independent VProContinuous, VPro/XD and speaker-dependent VPro/RT specifically tuned for the cellular environment. The speaker-dependent discrete feature of VProCel allows for a user-defined 20-word personal directory, with a one-pass enrollment whereby users need only speak their chosen commands once. In addition, cellular-ready VPro/XD vocabularies consisting of voice-activated dialing command words are also available. VProCel is suited to voice-activated dialing applications using either digit strings or a listing of words in a personal directory. * VProSpeller is a recognizer that can determine which name or word is being spelled by a caller. Users may spell a string of letters (up to 32 letters) in an uninterrupted manner (without prompts or beeps between each letter). VProSpeller can recognize confusable letters by conducting an automated search of a database of words maintained by the application for the best candidates to match. * VProPRL Designed for customers who wish to enable VPC speech recognition technologies on platforms other than those supported by VPro hardware, the VProPRL is a portable recognizer library of VProContinuous, VPro/XD and VPro/RT, which can be embedded into a wide variety of hardware platforms. It consists of a library of object modules which can be linked with a user application or task. * VPro Hardware Platforms: VPro-42, VPro-84, VPro-88 : The VPro platforms are ISA compliant PC/AT boards. Each supports four to eight Virtual Speech Processors (VSPs). Each VSP, depending on load factors, can handle multiple telephone lines. Application and host computers communicate with each of the VSPs as separate autonomous units. VPro platforms use Texas Instruments TMS320C31 microprocessors which provide up to 133 MFLOPS of compute power. The platforms can have up to 8 megabytes of memory shared among all processors. In addition, each processor has 512K bytes of local memory. Both the PEB and MVIP PCM audio buses are supported by all VPro platforms. * Osprey is a call management software application that performs the kinds of telephone related activities typically done by a personal assistant, such as answering the phone, screening callers, routing calls, and taking and delivering messages. It is an automated phone attendant. * Price and availability: Contact Voice Processing Corporation * Contact: Kelli V. Smith Voice Processing Corporation 1 Main Street, Cambridge, MA, 02142 USA Ph: (617)494-0100 Fax: (617)494-4970 e-mail: KSmith@vpro.com WWW: http://www.vpro.com/ Whisper See the new page for Microsoft speech recognition software. * Platform: Windows 95 and Windows NT 4.0 * Description: Command and control recognition. WildCard Speech Products * Platform: Windows 3.1 and Windows 95 * OfficeTalk for Windows: provides voice commands for dictation, navigation, command and control, and formatting for business uses of computers. Provides user voice access to a wide variety of software applications in office suites from Microsoft, Novell/WordPerfect, and Lotus. More information on the WildCard OfficeTalk page. * LawTalk for Windows: adds features and interfaces that meet the specific needs of legal users. More information on the WildCard LawTalk page. * VoiceCompanion for the Internet: Surf the net using voice commands. Controls browsers like Netscape and Microsoft Explorer. More information on the VoiceCompanion web page. * VoiceCompanion - RemoteAccess: Over the telephone remote access to your desktop PC, for voicemail, FAX forwarding and address book information. More information on the VoiceCompanion web page. * Availability: WildCard Technologies Inc. 180 West Beaver Creek Road, Richmond Hill, Ontario, Canada L4B 1B4 Phone: (905) 731-6444, Fax: (905) 731-7017 Email: sales@wildcardtech.com WWW: http://www.wildcardtech.com/ ___________________________________________________________________________ Q6.6: Speaker Recognition (Verification and Identification) * Introduction * In the FAQ * On the WWW Introduction Speaker recognition is the process of automatically recognizing who is speaking on the basis of individual information included in speech signals. It can be divided into Speaker Identification and Speaker Verification. Speaker identification determines which registered speaker provides a given utterance from amongst a set of known speakers. Speaker verification accepts or rejects the identity claim of a speaker - is the speaker the person they say they are? Speaker recognition technology makes it possible to a the speaker's voice to control access to restricted services, for example, phone access to banking, database services, shopping or voice mail, and access to secure equipment. Both technologies require users to "enroll" in the system, that is, to give examples of their speech to a system so that it can characterise (or learn) their voice patterns. In the FAQ: * ImagineNation: Voice Activated UnLock Technology * Jialong He's Speaker Recognition (Identification) Tool * Keyware Biometric Security Products * SpeakerKey Voice Verifier from ITT * SpeakEZ Voice Print Speaker Verification * Voice Control Systems: Speaker Verification Technology On the WWW Survey of the State of the Art in Human Language Technology Report edited by Ronald A. Cole et. al. with a section on Speaker Recognition. http://www.cse.ogi.edu/CSLU/HLTsurvey/ch1node47.html Speaker Identification And Verification: LIMSI Report A technical description. http://www.limsi.fr/Recherche/TLP/reco/2pg95-sv/2pg95-sv.html Long Index of References on Automatic Speaker Verification A list of more than 350 papers on speaker verification in text or BibTeX format. Provided by G.Matas. http://sig.enst.fr/~chollet/ForMehdi/SpRecV1.l_ind.html CAVE: Caller Verification in Banking and Telecommunications European consortium developing speaker recognition technologies. http://www.ptt-telecom.nl/cave/ Hangai Lab demonstrations of speaker verification and speaker identification. Do it yourself demonstrations: http://miya8f05.ee.kagu.sut.ac.jp/study/speech/speech1.html http://miya8f05.ee.kagu.sut.ac.jp/study/speech/speech2.html Voice Activated UnLock Technology (VAULT): ImagineNation * Description: Password-based voice verification technology using a card to store voice-print data. Introductory information and the VAULT FAQ are provided on the ImagineNation WWW pages. * Contact: Imagine PO Box 212, Swansea, MA 02777, USA Ph: +1-508-678-9563 Fax: 508-678-1470 Email: feedback@ImagineNation.com WWW: http://www.ImagineNation.com/ Jialong He's Speaker Recognition (Identification) Tool * Platform: SUN SPARC (SunOS), PC (MSDOS) * Description: This package contains a set of speaker recognition research utilities, including Gaussian mixture models, VQ codebook designing program and MLP network. They can also be used as general classifiers. The utilities are divided into the following categories: + Feature extraction and dimensional reduction cepstrum -- extract features from speech sigals (LPCC, MFCC, etc.). search -- select effective features (SFS, SBS method). randline -- randomize the a sequence, auxiliary utility. bin2asc -- binary to ASCII, auxiliary utility. + MLP network mlptrain -- MLP network training program. mlptest -- MLP network test program. + VQ codebook training and test programs lbglvq -- VQ codebook training program. nearest -- VQ codebook test program. + Gaussian mixture model (GMM) gmmtrain -- GMM training program. gmmtest -- GMM test program. Note: this is a research tool not a true speaker recognition system. * Availability: By anonymous ftp: MSDOS Version UK: ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/recognition/s pkrtool.zip Germany: ftp://ftp.informatik.uni-ulm.de/pub/NI/jialong/spkrtool.z ip Sun SPARC version, compiled with GNU C UK: ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/recognition/s pkr_sun_v1.tar.gz Germany: ftp://ftp.informatik.uni-ulm.de/pub/NI/jialong/speaker_su n_v1.tar.gz * See also: Jialong He's Speech Recognition Research Tool * Contact: Jialong He email: jialong@neuro.informatik.uni-ulm.de Keyware Biometric Security Products * Description: VoiceGuardian and S2 Security Server provide authentication and access control technologies. An online demo of Voice Guardian is available. * Contact: Keyware Technologies _USA_ Keyware Technologies 500 West Cummings Park, Suite 3600, Woburn, MA 01801, USA Ph: (617) 933 1311, Fax: (617) 933 1554 _Belgium_ Keyware Technologies Excelsiorlaan 28-30, 1930 Zaventem, Belgium Ph: 32 2 721 4574, Fax: 32 2 721 5015 _Email:_ sales@keywareusa.com _WWW:_ http://www.keywareusa.com/ SpeakerKey Voice Verifier from ITT * Platform: Windows/Pentium and Solaris/SPARC * Description: SpeakerKey provides over-the-phone voice verification. It is configurable for use in a wide range of applications. SpeakerKey provides a Speaker Verification API (SVAPI). SpeakerKey uses two technologies: (1) speaker-independent digit recognition using hidden Markov models, (2) speaker verification using "Nearest Neighbour Matching with Likelihood Ratio Scoring and cohort speakers." Dr. Joe Campbell maintains a SpeakerKey FAQ on the WWW. It provides a more detailed description of SpeakerKey and discusses several speaker verification issues: http://www.vitro.bloomington.in.us:8080/~BC/REPORTS/SpeakerKeyFAQ. html * Requirements: Minimum 60 MHz Pentium (with sound card) or SPARCstation 5, plus phone line interface devices. * Price: Evaluation kits available from $75. Developer's kits are $1500. Run-time licenses are priced from $600 to $10,000 depending upon the number of user and/or verifications per hour. Application customization is available. * Contact: ITT Industries Fort Wayne, IN, USA Ph: +1-219-487-6321, Fax: +1-219-487-6126 Email: speakerkey@itt.com SpeakEZ Voice Print Speaker Verification * Description: Designed to prevent cell phone theft and cloning fraud by comparing the cellular caller's statement of a pass-phrase to a stored digital "voice print" of the authorized subscriber. If the caller's voice patterns do not match the stored voice print, service will be denied or the caller will be referred to operator assistance for further validation processing. Features include: + Customer selected password. + Vocabulary and language independent. + No special hardware required by customer. + Multiple delivery options. * Contact: T-NETIX, Inc. 6675 South Kenton Street Englewood, CO 80111 USA Phone: (800) 352-8628, (303) 790-9111, Fax: (303) 790-9540 WWW: http://www.t-netix.com/ Voice Control Systems: Speaker Verification Technology * Description: SpeechPrint ID technology provides language independent speaker verification. Features: + Multiple speech input formats + Operates over various microphones or the telephone network + Can can be used in conjunction with discrete and continuous recognition + Robust against background noise and spurious telephone channel noise For more information on features, hardware and software requirements, pricing and availability, contact Voice Control Systems, Inc. or visit their the VCS WWW site or the SpeechPrint ID WWW page. * See also: VCS speech recognition products in Q6.5. * Contact: Voice Control Systems, Inc. 14140 Midway Rd., Dallas, Tx. 75244, USA Ph: +1-214-386-0300, Fax: +1-214-386-5555 Email: sales@vcsi.com WWW: http://www.voicecontrol.com/ ___________________________________________________________________________ Q6.7: Integrated Speech Products This section lists those products which integrate different speech technologies into a single user package. For example, speech recognition and speech synthesis can be combined to provide a dialog management system. Strictly speaking, this doesn't really belong under in Section 6 (Speech Recognition) but since these products all include speech recognition, it seems a reasonable place to put it for now! In the FAQ... * SpeechWorksfrom Applied Language Technologies, Inc. * Nortel Speech Technology Products SpeechWorksfrom Applied Language Technologies, Inc. * Description: SpeechWorks and companion products provide advanced speech recognition technology for the telephony market. SpeechWorks can be used by developers to "speech-enable" call center, messaging, enhanced services, and other types of applications. The three major system modules - SpeechWorks, DialogModules and SpeechBuilder - are described below. More detailed information is available from the Applied Language Technologies home page. ALTech develops and markets speech understanding software which provides large vocabulary, speaker-independent, phonetic speech recognition. ALTech's software contains a comprehensive set of features for speech-enabling telephone-based transactions and services. SpeechWorks is based on technology licensed from the Spoken Language Systems Group at the Massachusetts Institute of Technology. * SpeechWorks: provides the core speech recognition capabilities. Features include: + Phonetic segment-based, speaker-independent, large vocabulary, continuous speech recognition + Real-time vocabulary generation directly from text + Database integration + "Barge-in" capability + Adaptive channel normalization + "n-best" output and associated confidence scores + Support for multiple languages + Software-only or DSP-based implementations + Support for multiple platforms and operating systems (e.g., SCO UNIX, WindowsNT, etc.) * DialogModules: manage the "conversation" between the system and the caller within an application. They provide high-level application building blocks which enable developers to quickly and easily add speech interfaces to computer telephony applications. Each DialogModule accomplishes a particular task within an application, ranging from "simple" tasks such as capturing a yes/no response or a phone number, to more complex tasks such as capturing credit card information or name and address information. DialogModules provide "out-of-the-box" functionality. They contain pre-built grammars, user-interface design, internal call flow and error recovery routines, parameters for customization and a set of C++ class libraries and C APIs. * SpeechBuilder: provides tools for customizing the DialogModules and for developing and maintaining applications. A GUI-based Vocabulary Editor provides the ability to generate and maintain vocabulary or word lists. Pronunciations can be generated automatically using the built-in dictionary or can be automatically generated using a set of text-to-phoneme rules. * Product Bundles: are available which combine SpeechWorks and multiple DialogModules into application templates for a set of generic application categories. + SpeechForms SpeechForms provides an interactive method for entering data over the phone, such as ordering products, filling out surveys and completing registration forms. Typical applications include: order entry, reservations, catalog and literature requests, catalog shopping, subscriptions, change of service, claims, credit card activation, home banking, stock transactions, and warranty reservations. + SpeechQuery SpeechQuery is used to deliver information in response to voice requests over the phone, such as airline information, product delivery status and retirement benefit information. Typical applications include: order status, product information, account balance, flight status, movie listings, job listings, stock quotes, guide services,classified ads, claims status, dealer locator services, and technical support. + SpeechAgent SpeechAgent provides a set of modules for automating telephone-based voice messaging applications, such as integrated messaging, single-number services and voice-dialing. Typical applications include: voice messaging, voice dialing, auto attendant, address book access, email access, and scheduling. * Platform: Platforms and Operating systems: ALTech's software can be deployed on industry-standard hardware platforms and operating systems including: Sun SPARC-based systems running SunOS or Solaris, IBM RS/6000s running AIX, HP systems running HP-UX, and 486/Pentium-based PCs and servers running Windows, WindowsNT, SCO UNIX, or Solaris. ALTech's systems are designed to run all or some of the software on a digital signal processor. * Availability: contact ALTech for licensing information. * Contact: Applied Language Technologies, Inc. 215 First Street, Cambridge, MA 02142 Ph: 617-225-0012, Fax: 617-225-0322 Email: to Alisa Moyer: moyer@altech.com WWW: http://www.altech.com/ Nortel Speech Technology Products * Nortel's AudioGram Delivery Service (ADS): When a busy or no answer condition is encountered, an intercept message offers ADS, which provides a service to the calling party by taking a message automatically. ADS records the caller's message and attempts delivery repeatedly if needed until the message is delivered. ADS is comprised of four independent services: 0+, 1+ and Local, Intentional, and Millenium AudioGram. ADS services utilize Nortel's Flexible Voice Recognition (FVR) voice-processing capabilities. ADS features include: + Cost-saving common service platform (NAV) + Builds upon existing network investment in toll infrastructure capabilities of AABS (Automated Alternate Billing Service) + Leverages the capabilities of existing TOPS (Traffic Operator Position System) attendants. More information: is available on the Nortel Multimedia Network Applications WWW page for AudioGram Delivery Service. * Nortel's Voice-Activated Auto Attendant (VAAA): Replaces touch tone menu with easy-to-use voice interface. Geared to businesses and corporations to provide more effective management of incoming customer calls. Residing on the Network Applications Vehicle (NAV) platform, VAAA uses Flexible Vocabulary Recognition (speaker-independent) technology to recognize spoken words, and directs calls accordingly. Other features include: + Cost-saving common service platform (NAV) + Serves DTMF and rotary dial callers. + Handles incoming calls for all corporate users (Centrex, PBX, or key systems) More information: is available on the Nortel Multimedia Network Applications WWW page for Voice-Activated Auto Attendant. * Nortel's Voice-Activated Dialing (VAD): Phoneme-based speech dialing capabilities provided through speaker-trained and speaker-independent technologies. Residing on the Network Applications Vehicle (NAV) platform, VAD enables subscribers to dial using speech, as well as to create and customize personal telephone directories. Other features include: + Cost-saving common service platform (NAV) + Speech playback and Text-to-speech synthesis + Dual Language capability (optional) + Speech Recording + Canadian French speechware (optional, prompts and FVR) + Spanish speechware (optional, prompts and FVR) + 75-name VAD directory size + Word-spotting + DTMF tone detection + Directory sharing + Scalable service deployment + Talk-through More information: is available on the Nortel Multimedia Network Applications WWW page for Voice-Activated Dialing. * Nortel's Voice-Activated Premier Dialing (VAPD): Enables businesses to take advantage of the public network directories to stimulate customer calls. Residing on the Network Applications Vehicle (NAV) platform, VAPD uses Flexible Vocabulary Recognition (speaker-independent) technology to recognize business names, and routes calls to the appropriate business entity. VAPD promotes cost savings by utilizing a common service platform, the Network Applications Vehicle (NAV). It services DTMF callers as well as rotary dialers, and handles incoming calls for all corporate users: Centrex, PBX, and key systems. More information: is available on the Nortel Multimedia Network Applications WWW page for Voice-Activated Premier Dialing. * Platform: This speech-based service operates on the Network Applications Vehicle (NAV) platform. NAV is a multi-application, digital signal processing platform supporting both speech- and display-based applications. The NAV platform provides the speech recognition capabilities and application logic used by NAV features an open, modular hardware architecture and flexible software design. Other features include: + Scalable hardware - from 24 to over 2000 ports per NAV node; 1 to 24 independent application shelves per node + Powerful speech processing - speaker-independent and speaker-trained speech processing support + Reliability - N+1, N+M, and 2N redundancy + Central Management - access via graphical user interface to remote connections * See Also: Nortel Feature Planning Guide, reference number 50004.11; NAV Applications and Planning Guide, reference number 50118.16. Nortel's Multimedia web pages: http://www.nortel.com/entprods/multimedia/ * Contact: NORTEL Multimedia Communications Systems Division Multimedia Network Applications 1000 Park Forty Plaza Durham, NC 27713 USA Ph: 1-800-4NORTEL WWW: http://www.nortel.com/entprods/multimedia/