Amiga Plus 1995 #2/Amiga Plus CD - 1995 - No. 2.iso/internet/faq/englisch/comp.speech

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Archive-name: comp-speech-faq/part1
Last-modified: 1995/01/11


              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.

   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 List -
   FAQ" in comp.misc, comp.archives.admin or news.answers.

   This FAQ is posted every 4 weeks to comp.speech, comp.answers &
   news.answers.

   It is also available for 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/news.answers/comp-speech-faq/*

   Or on the World Wide Web
     * http://www.speech.su.oz.au/comp.speech

   Or by sending email to [e-mail] with the following
   line in the body of the message:
     * send usenet/news.answers/comp-speech-faq/*

Admin

   Not much to report this month. Hopefully, February should see some
   major catch-up work.

FAQ Sections

   The FAQ is divided into the following sections:
     * FAQ Contents

     * List of Speech Technology Products and Software

     * FAQ Section 1: General Information on Speech Technology
     * FAQ Section 2: Signal Processing
     * FAQ Section 3: Speech Coding and Compression
     * FAQ Section 4: Natural Language Processing
     * FAQ Section 5: Speech Synthesis
     * FAQ Section 6: Speech Recognition

Comp.Speech FTP Site

   The comp.speech ftp site (which is described in Q1.2) contains the
   following:
     * Newsgroup Archives
     * Data Resources
     * General Information
     * Software

Acknowledgements

   Hundreds of people have made contributions to the comp.speech FAQ over
   the last two years; there are too many to name individually. Special
   thanks go to Tony Robinson and Joe Campbell who have been particularly
   helpful.

Maintainence

   The FAQ posting and the Comp.Speech WWW Site are maintained by

    Andrew Hunt
    ---
    Speech Technology Research Group
    Dept. of Electrical Engineering
    University of Sydney, NSW, 2006, Australia
    Ph: 61-2-351 4509
    Fax: [phone]
    email: [e-mail]


===========================================================================


                           COMP.SPEECH FAQ CONTENTS

Introduction

     * Overview
     * List of Packages

Section 1 : General Information on Speech Technology

     * Q1.1 What is comp.speech?
     * Q1.2 Where are the comp.speech archives?
     * Q1.3 Common abbreviations and jargon.
     * Q1.4 What are related newsgroups and mailing lists?
     * Q1.5 What are related journals and conferences?
     * Q1.6 What resources are available as handicap aids?
     * Q1.7 What speech data is available?
     * Q1.8 Speech File Formats, Conversion and Playing.
     * Q1.9 What "Speech Laboratory Environments" are available?
     * Q1.10 Miscelaneous Software and Other Resources.

Section 2 : Signal Processing for Speech

     * Q2.1 What sampling do I need for speech?
     * Q2.2 How do I find 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 What signal processing techniques are used in speech
       technology?
     * Q2.6 What speech sampling and signal processing hardware can I
       use?
     * Q2.7 How do I convert to/from mu-law format?

Section 3 : Speech Coding and Compression

     * Q3.1 Speech compression techniques.
     * Q3.2 What are some good references/books on coding/compression?
     * Q3.3 What software is available? (Includes CELP & G.7xx)

Section 4 : Natural Language Processing

     * Q4.1 What are some good references/books on NLP?
     * Q4.2 What NLP software is available?

Section 5 : Speech Synthesis

     * Q5.1 What is speech synthesis?
     * Q5.2 How can speech synthesis be performed?
     * Q5.3 What are some good references/books on synthesis?
     * Q5.4 What software/hardware is available?

Section 6 : Speech Recognition

     * Q6.1 What is speech recognition?
     * Q6.2 How can I build a very simple speech recogniser?
     * Q6.3 What does speaker dependent/adaptive/independent mean?
     * Q6.4 What does small/medium/large/very-large vocabulary mean?
     * Q6.5 What does continuous speech or isolated-word mean?
     * Q6.6 How is speech recognition done?
     * Q6.7 What are some good references/books on recognition?
     * Q6.8 What speech recognition packages are available?


===========================================================================


FAQ: List of Packages

    The comp.speech FAQ provides information on a range of software,
   hardware and resources.

Speech Data

     * Phonemic Samples
     * Linguistic Data Consortium (LDC)
     * Center for Spoken Language Understanding (CSLU)
     * PhonDat - A Large Database of Spoken German
     * Oxford Acoustic Phonetic Database

Speech Processing Environments

     * Entropic Signal Processing System (ESPS) and Waves
     * CSRE: Canadian Speech Research Environment
     * OGI Speech Tools
     * Matlab plus Signal Processing Toolbox
     * Signalyze 3.0 from InfoSignal
     * Kay Elemetrics CSL (Computer Speech Lab) 4300
     * MacSpeech Lab II (MSL II)
     * N!Power
     * Ptolemy
     * Khoros
     * SpeechViewer II

Other Resources

     * CMU Dictionary
     * Another Dictionary
     * BEEP dictionary
     * CUVOLAD dictionary
     * MRC database
     * Network Audio System
     * NEVOT (1.4v) from AT&T; BL
     * Human Audio Perception Document
     * Homophone List
     * Auditory Toolbox for Matlab
     * Auditory Modeller 1
     * Auditory Modeller 2

Audio I/O Hardware

     * Sun standard audio port (SPARC I & II)
     * Sun standard audio port (SPARC 10 & 20)
     * Ariel Signal Processors
     * IBM RS/6000 ACPA (Audio Capture and Playback Adapter)
     * Sound Galaxy NX , Aztech Systems
     * Sound Galaxy NX PRO, Aztech Systems
     * ATI Stereo F/X Sound Board
     * Various PC Sound Cards

Compression Software and Hardware

     * File format conversion
     * shorten - a lossless compressor for speech signals
     * 32 kbps ADPCM
     * GSM 06.10 Compression
     * G.721/722/723 Compression
     * G.728 Compression
     * G.728 LD-CELP vocoder
     * U.S.F.S. 1016 CELP vocoder for DSP56001
     * 8 Kbit/s CELP on the TMS320C5x family of DSP chips
     * CELP 3.2a & LPC

Natural Language Processing

     * Natural Language Software Registry (NLSR) - NLP Tools
     * Part of Speech Tagger

Speech Synthesis

     * Orator Text-to-Speech Synthesizer
     * Text to phoneme program (1)
     * Text to phoneme program (2)
     * Text to phoneme program (3)
     * Text to speech program
     * "Speak" - a Text to Speech Program
     * TheBigMouth - a Text to Speech Program
     * TextToSpeech Kit
     * SGI Developers Toolbox Synthesiser
     * rsynth
     * SENSYN speech synthesizer
     * spchsyn.exe
     * CSRE: Canadian Speech Research Environment
     * Eloquence (currently an alpha release)
     * JSRU
     * Klatt-style synthesiser
     * DECTalk
     * Speech Manager and PlainTalk
     * Various Mac Speech Output Applications
     * MacinTalk
     * Monologue by Creative Labs
     * Lernout & Hauspie Text-To-Speech SDK
     * Tinytalk
     * Narrator - narrator.device
     * Infovox Product Range
     * SIMTEL-20

Speech Recognition

     * HM2007 - Speech Recognition Chip
     * Voice Blaster Ver. 4.0
     * Votan
     * Entropic's HTK (HMM Toolkit)
     * DragonDictate version 3.0
     * DragonDictate for Windows
     * DragonVoiceTools
     * IBM Personal Dictation System
     * Osborne Personal Dictation System (in Australia)
     * VoiceServer for Windows
     * IN3 Voice Command for Windows
     * IN3 Voice Command
     * Phonetic Engine 400 (PE400) - Speech Systems, Inc.
     * SayIt
     * Kurzweil Voice for Windows 1.0
     * D6006 Voice Control Processor
     * Speech Commander - Listen for Windows
     * Voice-Trek 2.0
     * Visus SpeechKit
     * recnet
     * Lotec Speech Recognition Package
     * Myers' Hidden Markov Model software
     * Voice Command Line Interface
     * DATAVOX - French
     * PowerSecretary
     * ICSS system from IBM
     * Creative VoiceAssist


===========================================================================


FAQ SECTION 1 - General

  Q1.1: WHAT IS COMP.SPEECH?

   Comp.speech is a newsgroup for discussion of speech technology and
   speech science. It covers a wide range of issues from application of
   speech technology, to research, to products and lots more. By nature
   speech technology is an inter-disciplinary field and the newsgroup
   reflects this. However, computer application is the basic theme of the
   group.

   The following is a list of topics but does not cover all matters
   related to the field (no order of importance is implied).
     * 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: WHERE ARE THE COMP.SPEECH ARCHIVES?

   comp.speech is being archived for anonymous ftp.
     * ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/archive/

   comp.speech/archive contains the articles as they arrive. Batches of
   100 articles are grouped into a shar file, along with an associated
   file of Subject lines.

   Other useful information is also available in comp.speech/info.
     _________________________________________________________________

  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.
     * NLP - Natural Language Processing.
     * NN - Neural Network.
     * TI - Texas Instruments.
     * TIMIT - A large speech corpus from TI and MIT - see Q1.7
     * TTS - Text-To-Speech (i.e. synthesis).
     * VQ - Vector Quantisation.

     _________________________________________________________________

  Q1.4: WHAT ARE RELATED NEWSGROUPS AND MAILING LISTS?

      Newsgroups

   comp.ai - Artificial Intelligence newsgroup.
          Postings on general AI issues, language processing and AI
          techniques. Has a good FAQ including 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.

   comp.ai.nlang-know-rep - Natural Language Knowledge Representation
          Moderated group covering Natural Language.

   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.

   comp.compression - occasional articles on compression of speech.
          FAQ for comp.compression 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. Has a regular posting of a
          comprehensive list of Audio File Formats.

   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.

   alt.sci.physics.acoustics
          Some discussion of speech production & perception.

   alt.binaries.sounds.misc - posting of various sound samples

   alt.binaries.sounds.d - discussion about sound samples, recording
          and playback.

      Mailing Lists

   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:

          + [e-mail]

   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

          + [e-mail]
          + 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

          + [e-mail]
          + 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
          [e-mail] to subscribe.

     _________________________________________________________________

  Q1.5: WHAT ARE RELATED JOURNALS AND CONFERENCES?

   Try the following commercially oriented magazine:
     * Voice News - monthly industry newsletter
    Stoneridge Technical Services
    PO Box 1891, Rockville, MD, 20850, USA
    Phone: [phone]
     * Voice Technology News
     * Voice Processing Magazine (1-800-854-3112)
     * Speech Technology (no longer published)

   Try the following technical journals (some contact addresses below):-
     * IEEE Transactions on Speech and Audio Processing (from Jan 93)
     * IEEE Signal Processing Magazine (from Jan 93)
     * IEEE Transactions on Acoustics, Speech, and Signal Processing
       (ASSP) (now obsolete)
     * Computational Linguistics (COLING)
     * Computer Speech and Language
     * Journal of the Acoustical Society of America (JASA)
     * AVIOS Journal
     * ASR News

   Try the following conferences:-
     * ICASSP Intl. Conference on Acoustics Speech and Signal Processing
       (IEEE)
     * ICSLP Intl. Conference on Spoken Language Processing
     * EUROSPEECH European Conference on Speech Communication and
       Technology
     * AVIOS American Voice I/O Society Conference
     * SST Australian Speech Science and Technology Conference

   Here are a few contact addresses:-

   Publications:
          IEEE Transactions on Speech and Audio Processing (from Jan 93)
          IEEE Transactions on Acoustics, Speech, and Signal Processing
          (ASSP) - now obsolete.

   Organization:
          Institute of Electrical and Electronics Engineers (IEEE)

   Contact:
          IEEE Service Center
          445 Hoes Lane, PO Box 1331, Piscataway, NJ 08855, USA
          Phone: [phone]-IEEE or [phone]

   Publications:
          Computer Speech and Language

   Contact:
          Academic Press, Ltd.
          24-28 Oval Rd, London NW1, England

   Price:
          $136 (Institutions), $58 (Individuals)

   Publications:
          Association for Computational Linguistics

   Organization:
          Association for Computational Linguistics
          MIT Press Journals
          55 Hayward St, Cambridge, MA 02142, USA
          Phone: [phone]

     _________________________________________________________________

  Q1.6: WHAT RESOURCES ARE AVAILABLE AS HANDICAP AIDS?

   Can anyone provide information on speech technology aids for the deaf,
   blind, speech impaired, physically impaired and other groups who may
   benefit from speech technology?

    SpeechViewer II
     * Platform: IBM Machines from Mod 25 on.
     * Description: SpeechViewer II is a speech therapy tool. It
       provided 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
     * 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: The Psychological Corporation (TPC) [IBM Authorized
       Remarketer]
    Phone: [phone] or contact IBM on 1-800-426-4832.

     _________________________________________________________________

  Q1.7: WHAT SPEECH DATA IS AVAILABLE?

   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 appear to be available for a small
   cost. The databases normally require lots of storage space - do not
   expect to be able to ftp all the data you want.

    Phonemic Samples
     * First, 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

    Linguistic Data Consortium (LDC)
     * Briefly stated, the LDC has been 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.
       Here is list of some of the corpora:
          + The TIMIT and NTIMIT speech corpora
          + The Resource Management speech corpus (RM1, RM2)
          + The Air Travel Information System (ATIS0) speech corpus
          + The Association for Computational Linguistics - Data
            Collection Initiative text corpus (ACL-DCI)
          + The TI Connected Digits speech corpus (TIDIGITS)
          + The TI 46-word Isolated Word speech corpus (TI-46)
          + The Road Rally conversational speech corpora (including
            "Stonehenge" and "Waterloo" corpora)
          + The Tipster Information Retrieval Test Collection
          + The Switchboard speech corpus ("Credit Card" excerpts and
            portions of the complete Switchboard collection)
     * Further resources made available in the first year (or two):
          + The Machine-Readable Spoken English speech corpus (MARSEC)
          + The Edinburgh Map Task speech corpus
          + The Message Understanding Conference (MUC) text corpus of FBI
            terrorist reports
          + The Continuous Speech Recognition - Wall Street Journal
            speech corpus (WSJ-CSR)
          + The Penn Treebank parsed/tagged text corpus
          + The Multi-site ATIS speech corpus (ATIS2)
          + The Air Traffic Control (ATC) speech corpus
          + The Hansard English/French parallel text corpus
          + The European Corpus Initiative multi-language text corpus
            (ECI)
          + The Int'l Labor Organization/Int'l Trade Union multi-language
            text corpus (ILO/ITU)
          + Machine-readable dictionaries/lexical data bases (COMLEX,
            CELEX)
     * Detailed information about the Linguistic Data Consortium is
       available by anonymous from the address below. The files in the
       directory include more detailed information on the individual
       databases.
          + ftp://ftp.cis.upenn.edu/pub/ldc
     * For further information contact
    Linguistic Data Consortium
    441 Williams Hall, University of Pennsylvania
    Philadelphia, PA 19104-6305
    Phone: [phone]
    Fax: [phone]
    e-mail: [e-mail]

    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
       [e-mail]. (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 [e-mail] 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 by
       anonymous ftp:
          + ftp://speech.cse.ogi.edu/pub/releases
     * Contact: Mike Noel -
    email: [e-mail] Phone: [phone]

    PhonDat - A Large Database of Spoken German
     * The PhonDat continuous speech corpora are now available on CD-ROM
       media (ISO 9660 format).
          + PhonDat I (Diphone Corpus) : 6 CDs (1140.- DM)
          + PhonDat II (Train Enquiries Corpus): 1 CD ( 190.- DM)
     * PhonDat I comprises approx. 20.000, PhonDat II approx. 1500 signal
       files in high quality 16-bit 16 KHz recording. The corpora come
       with documentation containing the orthographic transcription and a
       citation form of the utterances, as well as a detailed file format
       description. A narrow phonetic transcription is available for
       selected files from corpus I and II.
     * For information and orders contact
    Barbara Eisen
    Institut fuer Phonetik
    Schellingstr. 3 / II
    D 80799 Munich 40
    Tel: +49 / 89 / 2180 -2454 or -2758
    Fax: +49 / 89 / 280 03 62

    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: [e-mail]

     _________________________________________________________________

  Q1.8: SPEECH FILE FORMATS, CONVERSION AND PLAYING.

   Section 2 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
     * ftp://ftp.cwi.nl/pub/audio/AudioFormats.part1,2

     _________________________________________________________________

  Q1.9: WHAT "SPEECH LABORATORY ENVIRONMENTS" ARE AVAILABLE?

   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 processing 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.

    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.
       Entropic also distributes HTK (the Hidden Markov Model Toolkit).
       HTK is described in Section 6 of this FAQ.
     * Cost: On request.
     * Contact:
    Entropic Research Laboratory, Washington Research Laboratory
    600 Pennsylvania Ave, S.E. Suite 202, Washington, D.C. 20003
    [phone]
    email - [e-mail]

    CSRE: Canadian Speech Research Environment
     * Platform: IBM/AT-compatibles
     * Description: CSRE is a microcomputer-based system designed to
       support speech research. CSRE provides a low-cost facility in
       support of speech research, using mass-produced and
       widely-available hardware. The project is non-profit, and relies
       on the cooperation of researchers at a number of institutions and
       fees generated when the software is distributed. Functions include
       speech capture, editing, and replay; several alternative spectral
       analysis procedures, with color and surface/3D displays; parameter
       extraction/ tracking and tools to automate measurement and support
       data logging; alternative pitch-extraction systems; parametric
       speech (KLATT80) and non-speech acoustic synthesis, with a variety
       of supporting productivity tools; and an experiment generator, to
       support behavioral testing using a variety of common testing
       protocols. A paper about the whole package can be found in:
          + Jamieson D.G. et al, "CSRE: A Speech Research Environment",
            Proc. of the Second Intl. Conf. on Spoken Language
            Processing, Edmonton: University of Alberta, pp. 1127-1130.
     * Hardware: Can use a range of data aqcuisition/DSP hardware
     * Cost: Distributed on a cost recovery basis.
     * Availability: For more information on availability contact
    Krystyna Marciniak
    email [e-mail]
    Tel [phone] Fax [phone].
   For technical information
    email [e-mail]
     * Note: Also included in Q5.4 on speech synthesis packages.

    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.
     * Availability: By anonymous ftp from
          + ftp://speech.cse.ogi.edu/pub/tools/
     * Contact: Try [e-mail]

    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: [phone]
    Email: [e-mail]
     * FTP: ftp://ftp.mathworks.com
     * WWW: http://www.mathworks.com/

    Signalyze 3.0 from InfoSignal
     * Platform: Macintosh
     * Description: Signalyze's basic conception revolves around up to
       100 signals, displayed synchronously in HyperCard fashion on
       "cards". The program offers a complement of signal editing
       features, quite a few spectral analysis tools, manual scoring
       tools, pitch extraction routines, a good set of signal
       manipulation tools, and extensive input-output capacity.

       Handles multiple file formats: Signalyze, MacSpeech Lab,
       AudioMedia, SoundDesigner II, SoundEdit/MacRecorder, SoundWave,
       three sound resource formats, and ASCII-text. Sound I/O: Direct
       sound input from MacRecorder and similar devices, AudioMedia,
       AudioMedia II and AD IN, some MacADIOS boards and devices, Apple
       sound input (built-in microphone). Sound output via Macintosh
       internal sound, via SoundManager 3.0, some MacADIOS boards and
       devices as well as via the Digidesign 16-bit boards.

       It has a range of capabilities for creating, editing and
       manipulating label files with flexibility in labelling format.
     * Compatibility: MacPlus and higher (including II, IIx, IIcx,
       IIci, IIfx, IIvx, IIvi, Portable, all PowerBooks, Centris and
       Quadras). Takes advantage of large and multiple screens and 16/256
       color/grayscales. System 7.0 compatible. Runs in background with
       adjustable priority.
     * Misc: A demo available upon request. Manuals and tutorial
       included. It is available in English, French, and German. An
       UPDATER to version 2.48 is now available in:
          + - The UNIL Gopher server (see last page of InfoSignal News 8)
          + - The LAIP FTP server. Address: MACFL4082.unil.ch, machine
            no. 130.223.104.31
   Also available are a demo program, and current questions and answers.
     * Cost: Individual licence US$350, site license US$500, plus
       shipping. Upgrades from version 2.0 are available.
     * Contact:
    North America - Network Technology Corporation
    91 Baldwin St., Charlestown MA 02129
    Fax: [phone] Phone: [phone]
   Elsewhere contact
    InfoSignal Inc.
    C.P. 73, 1015 LAUSANNE, Switzerland,
    FAX: [phone],
    Email: [e-mail].

    Kay Elemetrics CSL (Computer Speech Lab) 4300
     * Platform: Minimum IBM PC-AT compatible with extended memory (min
       2MB) with at least VGA graphics. Optimal would be 386 or 486
       machine with more RAM for handling larger amounts of data.
     * 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. Probably not
       much use without the extra LPC program, which also allows
       manipulation of pitch, formant and bandwidth parameters.

       Hardware includes an internal DSP board for the PC (requires ISA
       slot), and an external module containing signal processing chips
       which does A/D and D/A conversion.
     * Misc: A programmers kit is available for programming signal
       processing chips (experts only). A speaker and microphone are
       supplied. Manuals are included.
     * Cost: Recently approx 6000 pounds sterling.
     * Contact:
    UK distributors are Wessex Electronics,
    114-116 North Street, Downend, Bristol, B16 5SE
    Tel: [phone].
   In the USA contact:
    Kay Elemetrics Corp,
    12 Maple Avenue, PO Box 2025, Pine Brook, NJ 07058-9798
    Tel:[phone]

    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. The software an trade off accuracy and speech.
     * 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.
     * Cost: $4990 (in May '92 price list; no MSL soft/hardware package
       listed in January '93).
     * Contact:
    GW Instruments
    35 Medford Street, Somerville, MA 02143
    Phone: [phone] Fax: [phone]

    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: [phone] FAX: [phone]
    email: [e-mail]

    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

    Khoros
     * Description: Public domain image processing package with a basic
       DSP library. Not particularly applicable to speech, but not bad
       for the price.
     * Cost: Free
     * Availability: By anonymous ftp from ftp://pprg.eece.unm.edu

    SpeechViewer II
     * Description: Speech Therapy Tool. See the detailed description
       in the handicap section - Q1.6.

     _________________________________________________________________

  Q1.10: MISCELANEOUS SOFTWARE AND OTHER RESOURCES.

    CMU dictionary
     * Description: Phonemic transcriptions of 100,000 words with
       American English pronunciation.
     * Availability: 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

    Dictionary
     * 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: By anonymous ftp from
          + 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.

    BEEP dictionary
     * Description: Phonemic transcriptions of 100,000 English words.
       (British English pronunciations)
     * Availability: By anonymous ftp from the file
          + svr-ftp.eng.cam.ac.uk/comp.speech/data/beep-0.3.tar.Z

    CUVOLAD dictionary
     * Description: Computer Usable Version of the Oxford Advanced
       Learner's Dictionary Has British English pronunciations and parts
       of speech
     * Availability: By anonymous ftp from the directory
          + ftp://black.ox.ac.uk/ota/dicts/710

    MRC database
     * Description: The Medical Research Council Psycholinguistic
       Database Has British English pronunciations, parts of speech, word
       frequency and lots of other information.
     * Availability: By anonymous ftp from the directory
          + ftp://black.ox.ac.uk/ota/dicts/1054

    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.

    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
     * Contact: [e-mail]
          + http://www.research.digital.com/CRL/projects/AF/home.html

    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
   Source is available.
     * Availability: by anonymous ftp from
          + ftp://gaia.cs.umass.edu/pub/hgschulz/nevot
     * Contact: Henning Schulzrinne ([e-mail])

    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/HumanAudioPercept
            ion
     * Contact:
    Argiris A. Kranidiotis
    University Of Athens, Informatics Department
    email: [e-mail]

    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/data/homophones-1.01.t
            xt

    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: By anonymous FTP from the following site:
          + ftp://ftp.apple.com/pub/malcolm
   The following files are available:
          + 419487 AuditoryToolbox.mif.Z
          + 1372976 AuditoryToolbox.psc.Z
          + 573215 AuditoryToolbox.sea.hqx
          + 92160 AuditoryToolbox.tar
          + 36405 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. We do
       provide 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: [e-mail]

    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

     _________________________________________________________________




Andrew Hunt
  ---
Speech Technology Research Group        Ph:  61-2-351 4509
Dept. of Electrical Engineering            Fax: [phone]
University of Sydney, NSW, 2006, Australia    email: [e-mail]
Archive-name: comp-speech-faq/part2
Last-modified: 1995/01/19


              COMP.SPEECH FAQ POSTING - PART 2/3


[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.]



===========================================================================


FAQ SECTION 2 - Signal Processing for Speech

  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: HOW DO I FIND 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.
     _________________________________________________________________

  Q2.3: HOW DO I FIND THE START AND END POINTS OF A SPEECH SIGNAL?

   A large number of papers have been presented on this task. Try the
   following papers:
     * 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 pp153-159.

     _________________________________________________________________

  Q2.4: WHERE CAN I FIND FFT SOFTWARE?

   Try the following file available by anonymous ftp. It contains a
   series of optimised fft routines, including mixed-radix algorithms.
   The .gz suffix indicates GNU zip format.
     * ftp://usc.edu/pub/C-numanal/fft-stuff.tar.gz

     _________________________________________________________________

  Q2.5: WHAT SIGNAL PROCESSING TECHNIQUES ARE USED IN SPEECH TECHNOLOGY?

   This question is far to big to be answered in a FAQ posting.
   Fortunately there are many good books which answer the question. Some
   good introductory books include
     * 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
     * Douglas O'Shaughnessy -- Speech Communication: Human and Machine
       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: WHAT SPEECH SAMPLING AND SIGNAL PROCESSING HARDWARE CAN I USE?

   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).

   Can anyone provide information on Mac, SGI, NeXT and other hardware?

    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 standard 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)

    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).

    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: [phone] DSP BBS: 908-249-2124

    IBM RS/6000 ACPA (Audio Capture and Playback Adapter)
     * Description: The card supports PCM, Mu-Law, A-Law and ADPCM at
       44.1kHz (& 22.05, 11.025, 8kHz) with 16-bits of resolution in
       stereo. The card has a built-in DSP (don't know which one). The
       device also supports various formats for the output data, like
       big-endian, twos complement, etc. Good noise immunity.

       The card is used for IBM's VoiceServer (they use the DSP for
       speech recognition). Apparently, the IBM voiceserver has a
       speaker-independent vocabulary of over 20,000 words and each ACPA
       can support two independent sessions at once.
     * Cost: $US495
     * Contact: ?

    Sound Galaxy NX , Aztech Systems
     * Platform: PC - DOS,Windows 3.1
     * Cost: ?
     * Input: 8bit linear, 4-22 kHz.
     * Output: 8bit linear, 4-44.1 kHz
     * Misc: 11-voice FM Music Synthesizer YM3812; Built-in power
       amplifier; DSP signal processing support - ST70019SB Hardware
       ADPCM decompression (2:1,3:1,4:1) "AdLib" and "Sound Blaster"
       compatbility. Software includes a simple Text-to-Speech program
       "Monologue".

    Sound Galaxy NX PRO, Aztech Systems
     * Platform: PC - DOS,Windows 3.1
     * Cost: ?
     * Input: 2 * 8bit linear, 4-22.05 kHz(stereo), 4-44.1 KHz(mono).
     * Output: 2 * 8bit linear, 4-44.1 kHz(stereo/mono)
     * Misc: 20-voice FM Music Synthesizer; Built-in power amplifier;
       Stereo Digital/Analog Mixer; Configuration in EEPROM. Hardware
       ADPCM decompression (2:1,3:1,4:1). Includes DSP signal processing
       support. "AdLib" and "Sound Blaster Pro II" compatybility.
       Software includes a simple Text-to-Speech program "Monologue" and
       Sampling laboratory for Windows 3.1: WinDAT.
     * Contact: USA (510)6238988

    ATI Stereo F/X Sound Board
     * Platform: PC XT or AT - DOS, Windows 3.0, 3.1
     * Cost: $120 Canadian
     * Description: Input - 8 bit ADC, 44.1 kHz mono, 22.05 kHz Stereo.
       Output - Dynamic range = 48 dB, 32 anti-aliasing filters. Adds
       Stereo effect to existing mono Adlib or Sound Blaster apps.
       11-voice YAMAHA FM Music Synthesizer. Built-in 8 watt power
       amplifier, 4 watts per channel. Volume ctrl on rear. 2 Joystick
       input, software setup (no switches), software included. "AdLib"
       and "Sound Blaster" compatibility. DMA support for high speed
       digital audio. ADPCM decomp @ 4:1, 3:1, 2:1. Will play .WAV files.
       Optional MIDI I/O port $79. (MIDI IN, OUT, THRU, and sequencer).
     * Contact:
    ATI Technologies Inc.
    3761 Victoria Park Avenue, Scarborough, Ontario
    CANADA, M1W 3S2
    Ph: [phone] Fax: [phone]
    BBS: [phone] (9600 baud N.8.1)

    Other PC Sound Cards
============================================================================
sound          stereo/mono              compatible     included   voices
card           & sample rate            with           ports
============================================================================
Adlib Gold     stereo: 8-bit 44.1khz    Adlib ?        audio      20 (opl3)
1000                  16-bit 44.1khz                   in/out,    +2 digital
               mono: 8-bit 44.1khz                     mic in,    channels
                    16-bit 44.1khz                     joystick,
                                                       MIDI

Sound Blaster  mono: 8-bit 22.1khz      Adlib          audio       11 synth.
               FM synth with                           in/out,
               2 operators                             joystick,

Sound Blaster  stereo: 8-bit 22.05khz   Adlib          audio       22
Pro Basic      mono: 8-bit 44.1khz      Sound Blaster  in/out,
                                                       joystick,

Sound Blaster  stereo: 8-bit 22.05khz   Adlib          audio       11
Pro            mono: 8-bit 44.1khz      Sound Blaster  in/out
                                                       joystick,
                                                       MIDI, SCSI

Sound Blaster  stereo: 8-bit 4-44.1khz  Sound Blaster  audio       20
16 ASP         stereo: 16-bit 4-44.1khz                in/out,
                                                       joystick,
                                                       MIDI

Audio Port     mono: 8-bit 22.05khz     Adlib          audio       11
                                        Sound Blaster  in/out,
                                                       joystick

Pro Audio      stereo: 8-bit 44.1khz    Adlib          audio,      20
Spectrum +                              Pro Audio      in/out,
                                        Spectrum       joystick

Pro Audio      stereo: 16-bit 44.1khz   Adlib          audio       20
Spectrum 16                             Pro Audio      in/out,
                                        Spectrum       joystick,
                                        Sound Blaster  MIDI, SCSI

Thunder Board  stereo: 8-bit 22khz      Adlib          audio       11
                                        Sound Blaster  in/out,
                                                       joystick

Gravis         stereo: 8-bit 44.1khz    Adlib,         audio line  32 sampled
Ultrasound     mono: 8-bit 44.1khz      Sound Blaster  in/out,     32 synth.
                                                       amplified
                                                       out,
               (w/16-bit daughtercard)                 mic in, CD
               stereo: 16-bit 44.1khz                  audio in,
               mono: 16-bit 44.1khz                    daughterboard
                                                       ports (for
                                                       SCSI and
                                                       16-bit)

MultiSound     stereo: 16-bit 44.1kHz   Nothing        audio       32 sampled
               64x oversampling                        in/out,
                                                       joystick,
                                                       MIDI

=============================================================================

     _________________________________________________________________

  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);
}

     _________________________________________________________________


===========================================================================


FAQ SECTION 3 - Speech Coding and Compression

  Q3.1: SPEECH COMPRESSION TECHNIQUES.

   Can anyone provide a 1-2 page summary on speech compression?

   Note: the FAQ for comp.compression includes a few questions and
   answers on the compression of speech.
     _________________________________________________________________

  Q3.2: WHAT ARE SOME GOOD REFERENCES/BOOKS ON 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.
     * Makhoul, J. "Linear Prediction: A Tutorial Review." Proc. of the
       IEEE 63 (1975): 561 - 580.

     _________________________________________________________________

  Q3.3: WHAT SPEECH COMPRESSION/CODING SOFTWARE IS AVAILABLE?

   Note: there are two types of speech compression technique referred to
   below. Lossless technqiues preserve the speech through a
   compression-decompression phase. Lossy techniques do not preserve the
   speech prefectly. As a general rule, the more you compress speech, the
   more the quality degardes.

    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

    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 are in
          +
            ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/sources/shorten-1.
            14.tar.Z
          +
            ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/sources/shn114.zip

    32 kbps ADPCM
     * Platform: SGI and Sun Sparcs
     * Description: 32 kbps ADPCM C-source code (G.721 compatibility is
       uncertain)
     * Contact: Jack Jansen
     * Availablity: Anoymous ftp
          + ftp://ftp.cwi.nl/pub/adpcm.shar

    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
       [e-mail], [e-mail]
     * 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-1.0.5.
            tar.gz
          + MS-DOS compression from Germany:
            ftp://ftp.cs.tu-berlin.de/pub/local/kbs/tubmik/gsm/gsm-105.zi
            p
          + 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

    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 [e-mail], with
                GET ITU-3022
   as the *only* line in the body of the message. This is also available
       by anonymous ftp from:
          +
            ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/sources/G711_G721_
            G723.tar.Z

    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

    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:[phone] FAX:[phone]
    Internet: [e-mail]

    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:[phone] FAX:[phone]
    Email: [e-mail]

    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: [phone] Fax: [phone]
    Email: [e-mail]

    CELP 3.2a & LPC
     * Platform: Sun (the makefiles & source can be modified for other
       platforms)
     * Description: CELP is lossy compression technqiue. The U.S. DoD'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. Available for worldwide distribution (on
       DOS diskettes, but configured to compile on Sun SPARC stations)
       from NTIS and DTIC. Example input and processed speech files are
       included. A Technical Information Bulletin (TIB), "Details to
       Assist in Implementation of Federal Standard 1016 CELP," and the
       official standard, "Federal Standard 1016, Telecommunications:
       Analog to Digital Conversion of Radio Voice by 4,800 bit/second
       Code Excited Linear Prediction (CELP)," are also available.
     * Availability 1: Through the National Technical Information
       Service:
    NTIS
    U.S. Department of Commerce
    5285 Port Royal Road, Springfield, VA 22161, USA

       The "AD" ordering number for the CELP software is AD M000 118 (US$
       90.00) and for the TIB it's AD A256 629 (US$ 17.50). The LPC-10
       standard, described below, is FIPS Pub 137 (US$ 12.50). There is a
       $3.00 shipping charge on all U.S. orders. The telephone number for
       their automated system is 703-487-4650, or 703-487-4600 if you'd
       prefer to talk with a real person.

       (U.S. DoD personnel and contractors can receive the package from
       the Defense Technical Information Center: DTIC, Building 5,
       Cameron Station, Alexandria, VA 22304-6145. Their telephone number
       is 703-274-7633.)
     * Availability 2: By anonymous ftp from:
          + ftp://ftp.super.org(192.31.192.1)/pub/celp_3.2a.tar.Z
          + OR
            ftp://svr-ftp.eng.cam.ac.uk/comp.speech/sources/celp_3.2a.tar
            .Z
     * Misc: The following articles describe the Federal-Standard-1016
       4.8-kbps CELP coder (it's unnecessary to read more than one):
          + 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.
          + Campbell, Joseph P. Jr., Thomas E. Tremain and Vanoy C.
            Welch, "The Proposed Federal Standard 1016 4800 bps Voice
            Coder: CELP," Speech Technology Magazine, April/May 1990, p.
            58-64.

       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
            International Conf. on Acoustics, Speech, and Signal
            Processing, 1986, p. 473-6.
   Copies of the official standard, "Federal Standard 1016, Tele-
       communications: Analog to Digital Conversion of Radio Voice by
       4,800 bit/second Code Excited Linear Prediction (CELP)" are
       available for US$ 5.00 each from:
    GSA Federal Supply Service Bureau
    Specification Section, Suite 8100
    470 E. L'Enfant Place, S.W.
    Washington, DC 20407
    [phone]
   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: [phone]
    [e-mail]
   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: [phone] Fax: [phone]

     _________________________________________________________________


===========================================================================


FAQ SECTION 4 - Natural Language Processing

   There is now a newsgroup specifically for Natural Language Processing.
   It is called comp.ai.nat-lang.

   There is also a lot of useful information on Natural Language
   Processing in the FAQ for comp.ai. That FAQ lists available software
   and useful references. It includes a substantial list of software,
   documentation and other info available by ftp.
     _________________________________________________________________

  Q4.1: WHAT ARE SOME GOOD REFERENCES/BOOKS ON NLP?

   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.

    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 conferences of the field are
     * ACL (held every year)
     * and COLING (held every two years). Most AI conferences have a NLP
       track; AAAI, ECAI, IJCAI and the Cognitive Science Society
       conferences usually are the most interesting for NLP. CUNY is an
       important psycholinguistic conference. There are lots of
       linguistic conferences: the most important seem to be NELS, the
       conference of the Chicago Linguistic Society (CLS), WCCFL, LSA,
       the Amsterdam Colloquium, and SALT.

     _________________________________________________________________

  Q4.2: WHAT NLP SOFTWARE IS AVAILABLE?

   Check the comments at the start of this section for information on
   other newsgroups and sources of information on NLP.

    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: [e-mail]
     * post:
    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 pf speech tagger developed by
       Eric Brill. For a detailed description of the tagger see chapter 6
       of his thesis.
     * Availability: The tagger and description are available by
       anonymous ftp from
          + ftp://lightning.lcs.mit.edu/pub/BRILL/Programs & Papers

     _________________________________________________________________




Andrew Hunt
  ---
Speech Technology Research Group        Ph:  61-2-351 4509
Dept. of Electrical Engineering            Fax: [phone]
University of Sydney, NSW, 2006, Australia    email: [e-mail]
Archive-name: comp-speech-faq/part3
Last-modified: 1995/01/19


              COMP.SPEECH FAQ POSTING - PART 3/3


[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.]



===========================================================================


FAQ SECTION 5 - Speech Synthesis

  Q5.1: WHAT IS SPEECH SYNTHESIS?

   Speech synthesis is the task of transforming written input to spoken
   output. The input can either be provided in a graphemic/orthographic
   or a phonemic script, depending on its source.
     _________________________________________________________________

  Q5.2: HOW CAN SPEECH SYNTHESIS BE PERFORMED?

   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.
     _________________________________________________________________

  Q5.3: WHAT ARE SOME GOOD REFERENCES/BOOKS ON SYNTHESIS?

   The following are good introductory books/articles.
     * Douglas O'Shaughnessy -- Speech Communication: Human and Machine
       Addison Wesley series in Electrical Engineering: Digital Signal
       Processing, 1987.
     * D. H. Klatt, "Review of Text-To-Speech Conversion for English",
       Jnl. of the Acoustic Society of America (JASA), v82, Sept. 1987,
       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).
     * John Allen, Sharon Hunnicut and Dennis H. Klatt, "From Text to
       Speech: The MITalk System", Cambridge University Press, 1987.

     _________________________________________________________________

  Q5.4: WHAT SPEECH SYNTHESIS SOFTWARE/HARDWARE IS AVAILABLE?

   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.

    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 very good
       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.
     * Availability and Pricing: Contact Bellcore's Licensing Office
       (1-800-527-1080) or email Anthony Lindsey [e-mail]

    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://wuarchive.wustl.edu/mirrors/unix-c/utils/phoneme.c

    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/sources/english2ph
            oneme.shar

    Text to speech program
     * Description: A implementation of the Klatt phoneme to waveform
       speech synthesiser.
     * Availability: By anonymous ftp
          +
            ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/sources/klatt-0.02
            .tar.Z

    "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

    TheBigMouth - a Text to Speech Program
     * Platform: NeXT
     * Description: Text to speech program based on concatenation of
       pre-recorded speech segments. NeXT equivalent of "Speak" for Suns.
     * Availability: try NeXT archive sites such as
       sonata.cc.purdue.edu.

    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: [phone] Fax: [phone]
    Order Desk: 1-800-L-ORATOR (US and Canada only)
    Email: [e-mail]

    SGI Developers Toolbox Synthesiser
     * Platform: SGI
     * Description: The SGI Developer Toolbox 4.0 CDROM contains a
       basic public 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

    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
       and now utilises stress marks in the dictionary in synthesising
       intonation.
     * Price: Free
     * Availability: by anonymous ftp from
          +
            ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/sources/rsynth-2.0
            .tar.Z
          +
            ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/sources/rsynth-2.0
            .tar.gz

    SENSYN speech synthesizer
     * Platform: PC, Mac, 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
    64 Sidney Street, Cambridge MA 02139.
    Fax: [phone]; Tel: [phone].
    Email: [e-mail]

    spchsyn.exe
     * Platform: PC?
     * Availability: By anonymous ftp as a self extracting DOS archive.
          +
            ftp://evans.ee.adfa.oz.au/mirrors/tibbs/applications/spchsyn.e
            xe
     * Requirements: May require special TI product(s), but all source
       is there.

    CSRE: Canadian Speech Research Environment
     * Platform: PC
     * Cost: Distributed on a cost recovery basis.
     * Description: CSRE is a software system which includes in
       addition to the Klatt speech synthesizer, SPEECH ANALYSIS and
       EXPERIMENT CONTROL SYSTEM. A paper about the whole package can be
       found in:
          + Jamieson D.G. et al, "CSRE: A Speech Research Environment",
            Proc. of the Second Intl. Conf. on Spoken Language
            Processing, Edmonton: University of Alberta, pp. 1127-1130.
     * Hardware: Can use a range of data aqcuisition/DSP hardware.
     * Availability: For more information contact
    Krystyna Marciniak
    email [e-mail]
    Tel [phone] Fax [phone].
   For technical information email [e-mail]
     * Note: A more detailed description is given in Section 1.9 on
       speech environments.

    Eloquence (currently an alpha release)
     * Platform: Windows and Solaris
     * Description: Software based text-to-speech package. 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.

       Uses high-level linguistic parsing, which obviates the need for a
       huge dictionary. Handles numbers, acronyms, currency, etc.
       Includes a set of annotation symbols, for placing stress on
       particular words, expressing excitement/boredom, etc. Also allows
       phonetic input. The final version, including support for Windows
       DDE and OLE and UNIX Sockets, will be released by the end of 1994.

       Produces male and female voices for General American English.
       Dialects under development include Alabama, Brooklyn, and Boston.
     * Price: $5000 (unconfirmed)
     * Availability:
    Eloquent Technology, Inc.
    2389 North Triphammer Road
    Ithaca, NY 14850
    Ph: (607) 607-266-7025 Fax: [phone]
    Email: [e-mail]

    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 [e-mail])

    Klatt-style synthesiser
     * Platform: Unix
     * Cost: Free
     * Description: Software posted to comp.speech in late 1992.
     * Availability: By anonymous ftp from the comp.speech archives
          +
            ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/sources/klatt-0.02
            .tar.Z

    DECTalk
     * Description: Speech synthesis hardware and software. Detailed
       information on DECtalk and other DEC products is available on a
       World-Wide Web site.
          + http://www.digital.com/info.html
   For specific information on DECtalk, check out this www url:
          +
            http://www.digital.com/archive/pub/Digital/info/Customer-Updat
            e/940620005.txt

    Speech Manager and PlainTalk
     * Platform: Macintosh
     * Cost: Free
     * Description: Apple's new text-to-speech system extension(s) that
       enable applications (listed below) 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.apple.com/dts/mac/sys.soft/speech
   There are 3 files in this directory:

        6273632 Aug 14 22:51 macintalk-pro.hqx
                PlainTalk Text-To-Speech 1.0 speech synthesizer extension
                (includes Female Voice, Compressed); TTS Female Voice;
                TTS Male Voice; and TTS Male Voice, Compressed. Requires
                68020 or better!

        370108 Aug 13 04:30 speech-manager-docs.hqx
                Apple DocViewer format (Inside Macintosh style, no
                installation instructions - just drag everything onto
                your closed System Folder).

        262569 Aug 7 07:01 speech-manager.hqx
                Speech Manager 1.1.1 (includes Marvin's voice) and
                MacInTalk Voices 1.1.1 (9 more voices). Runs most Macs.

    Various Mac Speech Output Applications
     * Platform: Macintosh
     * Cost: Free (except for At Ease)
     * Description: Some of the Speech Manager aware text-to-speech
       (TTS) applications, etc. are listed below (there are more on the
       Apple Developer CD-ROMs).
Application, etc.  Source            Comments
_________________ ________  _________________________________________________
AddressSpeech     info-mac  4D talking address book (from Speech Pack 2.0)
At Ease 2.0    MacWarehouse Friendly desktop that speaks file names
At Ease 2.0 WG MacWarehouse Friendly desktop that speaks file names
Eliza 3.1            AOL    Talking Eliza (Rogerian psych therapist)
FB speech      Inside Basic Mag, volume 3, no. 6. FutureBasic demo
FB Speech demo Inside Basic Mag, volume 3, no. 7. FutureBasic demo
Fortune 1.1       info-mac  Like a talking UNIX fortune command - slick
Homer 0.92d9  zaphod.ee.pitt.edu  GUI IRC client, assign nicks voices - slick
MacMessage 1.0  FirstClassBBS  Share talking messages/customizable startup
Say               info-mac  MPW Tool which converts standard input to speech
ScriptTools 1.2   info-mac  Write AppleScript scripts to say text messages
Siege Watch 1.01f info-mac  Wryly political speaking clock
SoToSpeak1.0.0b10 info-mac  Two voice conversation (also see Fortune's About)
Speak It!         info-mac  Type in a message and have it spoken
Speaker 1.11      info-mac  Simple text file editor, speaks on CR, macros
Speecher 1.2.1    info-mac  Customizable word pronunciation/substitution
SpeechManagerdemo info-mac  Command line interface, C source, aka -explorer
Speech Pack 2.0   info-mac  4th Dimension external, add speech to database
SpeechUnitEx      info-mac  Pascal source code for speech in Lab 7
speek-02b         info-mac  Speech XCMD for HyperCard
TalkingClockPro2.0info-mac  AppleScriptable talking clock extension (2.0b0)
TeachText 7.2      AV Mac   Apple's talking TeachText (simple editor w/QT)
Tex-Edit 1.9         AOL    Talking word processor, McSink like, modeming
VoiceDemo 1.0.1   info-mac  Bare bones phrase talker
Welcome!v1.3.1    info-mac  A talking Welcome to Macintosh startup
?                     ?     Talking Plug-In-Module for MS Word 5,
                               experimental, unsupported, buggy, beware!
Speech Rhythms       AOL    A cool text file for one of the above apps
_____
     * Sources:
          + AOL = America Online
          + info-mac = {ftp sumex-aim.stanford.edu, ftp
            wuarchive.wustl.edu, et al.}
          + MacWarehouse = [phone]
     * Misc: Apple's work in spoken language technologies and systems
       is described in:
          + Lee, Kai-Fu. "The Conversational Computer: An Apple
            Perspective." (Keynote Speech) In Proc. Eurospeech in Berlin,
            September, 1993.

    MacinTalk
     * Platform: Macintosh
     * Cost: Free
     * Description: Formant based speech synthesis. There is also a
       program called "tex-edit" which apparently can pronounce English
       sentences reasonably using Macintalk.
     * Note: MacinTalk doesn't run reliably on Macintosh's with new
       sound hardware under the lastest OS (System 7.1 w/HUD 2.0). More
       recent software is listed above.
     * Availability: By anonymous ftp from many archive sites (have a
       look on archie if you can). tex-edit is on many of the same sites.
       Try
          +
            ftp://wuarchive.wustl.edu/mirrors2/info-mac/Old/card/macintalk
            .hqx
          +
            ftp://wuarchive.wustl.edu/mirrors2/info-mac/Old/card/macintalk
            -stack.hqx
          +
            ftp://wuarchive.wustl.edu/mirrors2/info-mac/app/tex-edit-15.hq
            x

    Monologue by Creative Labs
     * Platform: PC Windows plus SoundBlaster 16
     * Cost: $99.00 or free with some MultiMedia packages
     * Description: Phoneme based speech synthesis software which
       provides output on Sound Blaster compatible audio cards. It
       includes a dictionary of words that are "exceptions" together with
       a a dictionary manager for modifying those words. It can be used
       as a stand alone program with Windows' Clipboard or as a DDE
       server dynamically linked (DLL) to a program you write.
     * Cost: $99.00 or free with some MultiMedia packages
     * Contact:
    Creative Labs Inc.
    1901 McCarthy Boul, Milpitas, CA 95035, USA
    Tel: [phone] Fax: [phone] BBS: 408-428-6660
    OR Creative Technology Ltd.
    67 Ayer Rajah Crescent #03-18, Singapore 0513
    Tel: [phone] Fax: [phone] BBS: 65-776-2423

    Lernout & Hauspie Text-To-Speech SDK
     * Platform: IBM-Compatible
     * 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(33Mhz) or higher, 8Mb
       RAM, MS DOS 5.0(or higher), MS Windows 3.1 (or higher), Compiler
       and linker: Microsoft(R) Visual C++ or Borland C++, Windows(TM)
       3.1 compatible sound card, preferably 16 bit e.g. Soundblaster,
       Windows Sounds System, Pro Audio Spectrum
     * Price: Unconfirmed $1,999 per copy, and $499 per each additional
       language (American English, French, German, or Spanish).
     * Contact: USA [phone]

    Tinytalk
     * Platform: PC
     * Description: Shareware package is a speech 'screen reader' which
       is used by many blind users.
     * Availability: By anonymous ftp
          + ftp://handicap.shel.isc-br.com/speech
   Get the files ttexe166.zip and ttdoc166.zip.

    Narrator - narrator.device
     * 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

    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: for IBM PC, XT, AT, PS/2 model 30 or compatible
     * 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, Interface: 9-pole D-SUB (RS 232-C) 300-9600
            Baud
     * 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: any
     * 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, Interface: 9 pole D-SUB (RS 232-C) 300-9600
            Baud
     * 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: any
     * Misc: Infovox multi-lingual Text-to-Speech Technologies can
       interface with Apple's PlainTalk System. It enables Apple Third
       party developers to write application software with synthetic
       speech output using their usual Apple Plain Talk Text-to-Speech
       interface. Software already written for the English speaking
       market using Apple Plain Talk can be now distributed worldwide,
       provided message strings are translated.
     * Contact:
    Telia Promotor Infovox AB
    TTS Sales Division
    P.O. Box 2069
    S-171 02 Solna, Sweden
    Ph: [phone] Fax: [phone]
    email: [e-mail]

    SIMTEL-20
     * The following is a list of speech related software available from
       SIMTEL-20 and its mirror sites for PCs.
     * The SIMTEL internet address is WSMR-SIMTEL20.Army.Mil
       [192.88.110.20] Try looking at your nearest archive site first.
       [Note: problems have been reported in accessing this site - does
       anyone know a new address?]
Directory PD1: MSDOS.VOICE
 Filename   Type Length   Date   Description
 ==============================================
 AUTOTALK.ARC  B   23618  881216  Digitized speech for the PC
 CVOICE.ARC    B   21335  891113  Tells time via voice response on PC
 HEARTYPE.ARC  B   10112  880422  Hear what you are typing, crude voice synth.
 HELPME2.ARC   B    8031  871130  Voice cries out 'Help Me!' from PC speaker
 SAY.ARC       B   20224  860330  Computer Speech - using phonemes
 SPEECH98.ZIP  B   41003  910628  Build speech (voice) on PC using 98 phonemes
 TALK.ARC      B    8576  861109  BASIC program to demo talking on a PC speaker
 TRAN.ARC      B   39766  890715  Repeats typed text in digital voice
 VDIGIT.ZIP    B  196284  901223  Toolkit: Add digitized voice to your programs
 VGREET.ARC    B   45281  900117  Voice says good morning/afternoon/evening

     _________________________________________________________________


===========================================================================


FAQ SECTION 6 - Speech Recognition

  Q6.1: WHAT IS 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.
     _________________________________________________________________

  Q6.2: HOW CAN I BUILD A VERY SIMPLE SPEECH RECOGNISER?

   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_SpeechRecognit
       ion

    QUICKY RECOGNIZER sketch:

    Here 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.

   Q6.7 contains information on public domain speech recognition
   software: Lotec and Myers' Hidden Markov Model software.
     _________________________________________________________________

  Q6.3: 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 independent 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.
     _________________________________________________________________

  Q6.4: 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.

     _________________________________________________________________

  Q6.5: 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).
     _________________________________________________________________

  Q6.6: 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, MFCC, cochlea modelling
   and many, 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.
     _________________________________________________________________

  Q6.7: WHAT ARE SOME GOOD REFERENCES/BOOKS ON SPEECH RECOGNITION?

   Some reviews of speech recognition for personal computers:
     * "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 [phone].
     * 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 ?

   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
     * Douglas O'Shaughnessy -- Speech Communication: Human and Machine
       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

   More specific books/articles:
     * Hidden Markov models for speech recognition; X.D. Huang, Y. Ariki,
       M.A. Jack. Edinburgh: Edinburgh University Press, c1990
     * Automatic speech recognition: the development of the SPHINX
       system; by Kai-Fu Lee; Boston; London: Kluwer Academic, c1989
     * Prosody and speech recognition; Alex Waibel (Pitman: London)
       (Morgan Kaufmann: San Mateo, Calif) 1988
     * S. E. Levinson, L. R. Rabiner and M. M. Sondhi, "An Introduction
       to the Application of the Theory of Probabilistic Functions of a
       Markov Process to Automatic Speech Recognition" in Bell Syst.
       Tech. Jnl. v62(4), pp1035--1074, April 1983
     * R. P. Lippmann, "Review of Neural Networks for Speech
       Recognition", in Neural Computation, v1(1), pp 1-38, 1989.

     _________________________________________________________________

  Q6.8: WHAT SPEECH RECOGNITION PACKAGES ARE AVAILABLE?

   The following packages are presented in no particular order.

    HM2007 - Speech Recognition Chip
     * 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$30 for the HM2007 and US$100 for the demo board.
     * Warning: Several people have reported problems in obtaining
       small numbers of this chip (say less than 10). It appears that the
       distributors (include the one listed below) are only interested in
       large volumes. If you know of a good source please send it in for
       inclusion in the FAQ.
     * Contact:
    The Summa Group Limited
    One California Street, Suite #1940,
    San Francisco, CA 94111
    Ph: [phone]

    Voice Blaster Ver. 4.0
     * Platform: IBM AT or higher, DOS or Wndows 3.1
     * Description: Uses a Sound Blaster or compatible board. Contains
       a microphone headset and a connector for LPT1:. A printer can
       still be used on LPT1:. Will recognize 1024 words that are trained
       by the operator. Each word activates a macro that can enter an
       ascii word on the screen or into a word processor or invoke a
       batch file. An optional footswitch may be installed. Software to
       run under DOS or Windows 3.1 is included.
     * Cost: Around $150 Canadian.
     * Contact:
    COVOX Inc.
    675 Conger Street
    Eugene, Oregon, 97402, USA
    Ph: [phone] Fax: [phone]
    BBS: [phone]

    Votan
     * Platform: MS-DOS, SCO UNIX
     * Description: Isolated word and continuous speech modes, speaker
       dependant and (limited) speaker independent. Vocab size is 255
       words or up to a fixed memory limit - but it is possible to
       dynamically load different words for effectively unlimited number
       of words.
     * Rough Cost: Approx US $1,000-$1,500
     * Requirements: Cost includes one Votan Voice Recognition ISA-bus
       board for 386/486-based machines. A software development system is
       also available for DOS and Unix.
     * Misc: Up to 8 Votan boards may co-exist for 8 simultaneous voice
       users. A telephone interface is also available. There is also a
       4GL and a software development system. Apparently there is more
       than one version - more info required.
     * Contact: 800-877-4756, 510-426-5600

    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 1.8.
     * Misc: The availability of HTK changed in early 1993 when
       Entropic obtained exclusive marketing rights to HTK from the
       developers at Cambridge.
     * Cost: On request.
     * Contact:
    Entropic Research Laboratory,
    600 Pennsylvania Ave, S.E. Suite 202,
    Washington, D.C. 20003, USA
    Phone: [phone].
    email - [e-mail]

    DragonDictate version 3.0
     * Platform: PC
     * Description: Speaker-adaptive recognition system for discrete
       speech. Provides 110,000 word dictionary and also allows user to
       add words. Active vocabulary of 5,000, 30,000, or 60,000 words.
       Allows dictation into almost all DOS applications (word
       processors, spreadsheets, etc.) and hands-free operation of the
       PC.
     * Cost:Prices including audio board and high-quality headset
       microphone:
          + US$695 (5,000 word Starter Edition)
          + US$995 (30,000 word Classic Edition)
          + US$1,995 (60,000 word Power Edition)
     * Requirements: Minimum of 33 Mhz 486 with 8-16M memory and at
       least 29M disk space (depending on product), one 8-bit slot, DOS
       5.0 and up (also runs in a DOS box under Windows or OS/2).
     * Contact:
    Dragon Systems, Inc.
    320 Nevada Street
    Newton, MA 02160, USA
    Tel: [phone], Fax: [phone]

    DragonDictate for Windows
     * Platform: PC
     * Description: Speech-to-text dictation system. Discrete speech;
       speaker- adaptive. Also provides command/control and 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.
     * Rough Cost:Prices including software, documentation and
       microphone:
          + DragonDictate Starter Edition (5,000 words active) -- $395
          + DragonDictate Classic Edition (30,000 words active) -- $695
          + DragonDictate Power Edition (60,000 words active) -- $1,695
     * Requirements: 486/33, 7-10 MB dedicated RAM (depending on
       edition), Windows 3.1 or later. Supported sound boards: Media
       Vision Pro Audio Studio 16, Creative Labs Sound Blaster 16,
       Microsoft Windows Sound System, IBM Audio Capture/Playback
       Adapter.
     * Contact:
    Dragon Systems, Inc.
    320 Nevada Street
    Newton, MA 02160, USA
    Phone: [phone] Fax: [phone]

    DragonVoiceTools
     * Platform: PC
     * Description: Programmer's toolkit for developing speech-aware
       DOS or Windows applications. Recognizes continuously spoken digits
       and discretely spoken words or phrases. Up to 1,000 words can be
       active at one time. Use words from 110,000 word dictionary
       (included) and/or develop your own word models.
     * Cost:
          + US$1,995 (developer's kit)
          + US$595 (end-user system)
     * Requirements: Minimum of 20 Mhz 386 (larger vocabulary requires
       faster processor) with at least 5M memory and at least 19M disk
       space (depending on vocabulary size), DOS 5.0 and up, Windows 3.1
       and up, Borland C or C++ or Microsoft C or C++. Also requires IBM
       M-ACPA card available from IBM or Dragon Systems ($325).
     * Contact:
    Dragon Systems, Inc.
    320 Nevada Street,
    Newton, MA 02160, USA
    Tel: [phone], Fax: [phone]

    IBM VoiceType Dictation

    OR: Osborne Personal Dictation System (in Australia)
     * Platform: Intel I486 & IBM OS/2
     * Description: Independent Speaker, discrete speech dictation with
       navigation. Navigation does not require setup, most applications
       are automatically speech enabled by dynamic control analysis.
       Dictation averages 70WPM with 95% accuracy and uses statistical
       trigram modelling. The base system is 22K words, other
       vocabularies available for specific industries.
     * Requirements: 486SX or above, 16MB Ram, 30MB File space,
       Dictation Adapter
     * Cost: Software $495 (includes mic) / Hardware $495
     * Misc 1: A Windows version is now available.
     * Misc 2: Based on IBM Tangora Technology
     * Availability: US English. Other languages (UK, FR, GR, IT, and
       ES) available 3Q94.
     * Contact: US Contact 1-800-TALK-2-ME or 1-914-766-9252.

    VoiceServer for Windows
     * Platform: PC
     * Description: Speaker dependent, each with an independent
       directory. Isolated word. Upto 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

    IN3 Voice Command for Windows
     * Platform: PC with Windows 3.1
     * Description: IN3 is now available for MS-Windows. Users can call
       applications to the foreground with voice commands. Once the
       application is called, the user may enter commands and data with
       voice commands. Voice macros can reduce the strain of repetitive
       stress injuries (RSI) such as Carpel Tunnel Syndrome (CTS) by
       replacing heavy repetitive keyboard hammering with simple voice
       operations. Voice macros take complex operations and reduce them
       to simple verbal commands. Voice input can provide new facilities
       for tasks which could not easily have been otherwise performed
       without the multiple axis of input. IN3 is hardware-independent,
       users with any Windows-compatible audio add speech recognition to
       the desktop. IN3 works with either 8 bit or 16 bit Windows audio
       boards. IN3 is based on continuous word-spotting technology. A
       developer API is also available for creating voice-enabled
       applications.
     * Price: $179 U.S.
     * Requirements: PC with 80386 processor or better, Microsoft
       Windows 3.1, and Windows compatible audio system with microphone.
     * Misc: Fully functional demos are available on Compuserve in
       various Multimedia and CAD forums. Demos are also available from
       "America on Line", the comp.binaries.ms-windows archive sites, and
       various BBS systems. It is also available by anonymous ftp
          +
            ftp://ftp.wustl.edu/usenet/comp.binaries.ms-windows/v3/in3demo
            .zip
          + ftp://ftp.uwasa.fi/mirror/ultrasound/demo/in3demo.zip
   An equivilant Sun product is described below.
     * Contact:
    Brantley Kelly
    Email: [e-mail] CIS: 75120,431
    FAX: [phone] Phone: [phone]
    Command Corp. Inc, 3675 Crestwood Parkway, Duluth GA 30136, USA

    IN3 Voice Command
     * Platform: Sun SPARCstation
     * Description: IN3 provides a secure, robust, word spotting,
       continuous speech recognition facility for the Sun OS or Solaris
       operating systems. The recognition system is a secure operating
       system facility capable of working with various interfaces,
       microphones, and devices. The operating system interface works
       with native UNIX outside of X Windows as well as provides enhanced
       X Windows facilities including named window support. The user
       interface provides a means to quickly create commands on the fly
       for replacing long strings and complex operations with voice
       macros. [Voice macros can reduce the strain of repetitive stress
       injuries (RSI) such as Carpel Tunnel Syndrome (CTS) by replacing
       heavy repetitive keyboard hammering with simple voice operations.
       ] The IN3 user interface works with generic X servers and window
       managers. A developer API is also available for creating voice-
       enabled applications, interfacing with other audio sources, and
       providing extensive application control over the recognition
       facility.
     * Availability: SunSite archive at SunSITE.unc.edu as well as on
       Catalyst CDware as both a runable demo and unlockable software.
     * Hardware Required: Sun SPARCstation with audio input. Noise
       canceling microphone recommended but not required.
     * Software Required:
          + Sun OS 4.1.2 with OpenWindows 3.0
          + or, Sun OS 4.1.3
          + or, Solaris 2.1 or Solaris 2.2
     * Misc: An equivilant MS-Windows product is described above.
     * Price: $495 U.S.
     * Contact:
    Brantley Kelly
    Email: [e-mail] CIS: 75120,431
    FAX: [phone] Phone: [phone]
    Command Corp. Inc, 3675 Crestwood Parkway, Duluth GA 30136, USA

    Phonetic Engine 400 (PE400) - Speech Systems, Inc.
     * Platform: PC
     * Description: Speaker independent, large vocabulary, continuous
       speech recognition for MS Windows or DOS.
     * Rough Cost: $1195 US dollars. Includes board, microphone,
       developer kit, documentation, 2 days of technical training and 90
       days of technical support.
     * Requirements: IBM AT class machine or better plus 5M disk space.
       Most processing is performed on-board (4M standard or 16M
       upgrade).
     * Misc: Requires developer to provide a context-free grammar.
       Vocabulary size unknown (quotes from 500 - 2000 words per
       grammar), but dynamic grammar switching capabilities may increase
       the effective vocabulary size. Development system includes
       lower-level C,C++ library (VoiceLib), higher-level DLL (SPOT)
       callable from many languages, SPOT/VBX, a custom control for
       Visual Basic and Visual C++.
     * Contact:
    Speech Systems, Inc.
    2945 Center Green Court South
    Boulder, CO 80301-2275, USA
    Tel: [phone] Fax: [phone]

    SayIt
     * Platform: Sun SPARCstation
     * Description: Voice recognition and macro building package for
       Suns in the Openwindows 3.0 environment. Speaker dependent
       discrete speech recognition. Vocabularies can be associated to
       applications and the active vocabulary follows the application
       that has input focus. Macros can include mouse commands,
       keystrokes, Unix commands, sound, Openwindow actions and more. An
       evaluation copy is available by email.
     * Hardware: Microphone required (SunMicrophone is fine).
     * Cost: $US295
     * Contact:
    Phone: [phone]-UNIX or 1-415-572-0200
    Fax: [phone]
    Email: [e-mail]

    Kurzweil Voice for Windows
     * Platform: MS Windows 3.1
     * 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.
     * Cost: US $995
     * Hardware: 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:
    Phone: [phone]
    Email: [e-mail]

    D6006 Voice Control Processor
     * Platform: ?
     * Description: ?
     * Contact:
    DSP Telecommunications Inc.
    2855 Kifer Road, Suite 202, Santa Clara CA 95051, USA
    Tel:[phone]
    Fax:[phone]

    Speech Commander - Listen for Windows
     * Platform: ?
     * Description: ?
     * Contact:
    Verbex Voice Systems
    1090 King Georges Post Rd., Bldg 107,
    Edison NJ 08837, USA
    Tel:[phone]
    Fax:[phone]

    Voice-Trek 2.0
     * Platform: ?
     * Description: ?
     * Contact:
    Tardis Technology Inc., Voice Recognition Div.
    10321 Los Alamitos Blvd., Los Alamitos CA 90720
    Tel:[phone] Fax:[phone]

    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:
       [e-mail]

    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: [e-mail]
     * Availability: by anonymous ftp
          +
            ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/misc/recnet-1.3.ta
            r.Z

    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 perform 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: [e-mail]

    Myers' Hidden Markov Model software
     * 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 pit 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 three files in the directory
          + ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/sources
   The files are
          + hmm.README
          + hmm-1.0.tar.Z
          + OR, hmm-1.0.tar.gz
   (Note: hmm-1.0.tar.Z and hmm-1.0.tar.gz compressed and GNU compressed
       versions of the same files)
     * Contact: Richard Myers: email [e-mail]

    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.
     * Cost: Free?
     * 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 [e-mail]

    DATAVOX - French
     * Platform: PC
     * Description: Continuous speech - speaker independent or
       dependent.
     * Rough Cost: ?
     * 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
    Fax: [phone]
    Voice: 33 1 69 41 15 04

    PowerSecretary
     * Platform: Centris 650, 660AV. Quadra 650, 660AV, 700,800, 840AV,
       900, 950.
     * Description: Speaker dependent/adaptive system requiring words
       to be separated by short pauses.
     * Vocabulary: 30,000 at any one time, automatically selected from
       120,000-word dictionary.
     * Cost: US$2,495; non-AV machines need an audio board will cost
       about US$300.
     * Requirements: Minimum of 16M of ram and System 7.0.
     * Contact:
    Articulate Systems
    600 W. Cummings Park, Suite 4500
    Woburn, MA 01801
    Ph: [phone] Fax: [phone].

    ICSS system from IBM
     * Description: A large vocabulary, speaker independent, continuous
       speech system which runs under Windows, OS/2, and AIX.
     * Requirements: Soundboard (e.g. Soundblaster)
     * Price: $US319
     * Contact:
    A&G Graphics Interface
    ICSS Reseller
    51 Gore Street, Cambridge, MA, 02139, USA
    [phone]

    Custom Voice(TM) by A&G Graphics Interface
     * Description: 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: Visual Basic or any development platform that
       supports VBX.
     * Price: $US495 or $695 bundled with ICSS.
     * Contact:
    A&G Graphics Interface
    51 Gore Street, Cambridge, MA, 02139, USA
    [phone]

    Creative VoiceAssist
     * Platform: PC (?)
     * Price: $US99.95
     * Contact:
    Creative Labs
    Ph: 1-800-998-5227

     _________________________________________________________________




Andrew Hunt
  ---
Speech Technology Research Group        Ph:  61-2-351 4509
Dept. of Electrical Engineering            Fax: [phone]
University of Sydney, NSW, 2006, Australia    email: [e-mail]