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 mail-server@rtfm.mit.edu 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: 61-2-351 3847
email: andrewh@speech.su.oz.au
===========================================================================
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:
+ ectl-request@snowhite.cis.uoguelph.ca
The ECTL archive site is
+ ftp://snowhite.cis.uoguelph.ca/pub/ectl
Prosody Mailing List
Unmoderated mailing list for discussion of prosody. The aim is
to facilitate the spread of information relating to the
research of prosody by creating a network of researchers in the
field. If you want to participate, send the following one-line
message to
+ listserv@msu.edu
+ subscribe prosody Your Name
foNETiks
A moderated monthly newsletter distributed by e-mail. It
carries job advertisements, notices of conferences, and other
news of general interest to phoneticians, speech scientists and
others The editors are Linda Shockey and Gerry Docherty. To
subscribe send the following 1 line message to
+ mailbase@mailbase.ac.uk
+ join fonetiks your_first_name your_second_name
Digital Mobile Radio
Covers lots of areas include some speech topics including
speech coding and speech compression. Mail Peter Decker
dec@dfv.rwth-aachen.de to subscribe.
_________________________________________________________________
Q1.5: 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: (301) 424-0114
* 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: 1-800-678-IEEE or (201)981-0060
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: (617)253-2889
_________________________________________________________________
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: 1-800-228-0752 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: +1 (215) 898-0464
Fax: +1 (215) 573-2175
e-mail: ldc@unagi.cis.upenn.edu
Center for Spoken Language Understanding (CSLU)
* The ISOLET speech database of spoken letters of the English
alphabet. The speech is high quality (16 kHz with a noise
cancelling microphone). 150 speakers x 26 letters of the English
alphabet twice in random order. The ISOLET data base can be
purchased for $100 by sending an email request to
vincew@cse.ogi.edu. (This covers handling, shipping and medium
costs). The data base comes with a technical report describing the
data.
* CSLU has a telephone speech corpus of 1000 English alphabets.
Callers recite the alphabet with brief pauses between letters.
This database is available to not-for-profit institutions for
$100. The data base is described in the proceedings of the
International Conference on Spoken Language Processing.
+ Contact vincew@cse.ogi.edu if interested.
* CSLU has released for universities its Continuous English Speech
Corpus. The corpus contains recorded speech from 690 different
speakers, with label files at various levels - including word
level and phonetic labels. The data were collected as part of the
OGI Multi-language telephone corpus. CSLU provides speech corpora
to all universities without charge. To order a corpus, print the
license agreement/order form, complete it, and fax it to the CSLU.
A description of the corpora and an order form are available by
anonymous ftp:
+ ftp://speech.cse.ogi.edu/pub/releases
* Contact: Mike Noel -
email: noel@cse.ogi.edu Phone: (503) 690-1309
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: burton.rosner@wolfson.ox.ac.uk
_________________________________________________________________
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
(202) 547-1420
email - info@entropic.com
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 march@uwovax.uwo.ca
Tel (519) 661-3901 Fax (519) 661-3805.
For technical information
email ramji@uwovax.uwo.ca
* 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 tools@cse.ogi.edu
Matlab plus Signal Processing Toolbox
* Platform: Wide range
* Description: Matlab (MATrix LABoratory) is a technical computing
environment for numerical computation and visualization based on a
matrix oriented, interpreted programming language. The programming
environment provides support for the development of customized
operations, along with debugging facilities and a graphical user
interface toolkit. Audio output is provided.
A specialised Signal Processing Toolbox is available which
provides many functions which are useful for speech analysis. It
includes filter design, spectral estimation, statistical signal
processing, waveform generation, and signal and spectrogram
display.
A specialised Auditory Toolbox is available which contains
functions useful to people interested in auditory/cochlear models.
A more detailed description is given in Q1.10.
* Price: On request.
* Contact: The Math Works Inc.
24 Prime Park Way, Natick, MA 01760-1500 USA
Ph: 1-508-653 1415 Fax: 1-508-653 6284
Email: info@mathworks.com
* FTP: ftp://ftp.mathworks.com
* WWW: http://www.mathworks.com/
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: 617-241-5064 Phone: 617-241-9205
Elsewhere contact
InfoSignal Inc.
C.P. 73, 1015 LAUSANNE, Switzerland,
FAX: +41 21 691-1372,
Email: 76357.1213@COMPUSERVE.COM.
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: 0272 571404.
In the USA contact:
Kay Elemetrics Corp,
12 Maple Avenue, PO Box 2025, Pine Brook, NJ 07058-9798
Tel:(201) 227-7760
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: (617) 625-4096 Fax: (617) 625-1322
N!Power
* Platform: SUN, DEC and HP workstations.
* Description: An object-oriented software package with a MOTIF
GUI interface and a range of functionality for data
analysis/editing, signal analysis, speech processing, real-time
A/D and D/A, and 2D/3D interactive graphics. N!Power replaces ILS.
N!Power can provide a Block Diagram user interface, menus,
pop-ups, and a high-level IEEE standard symbolic scripting
language. You can customize the blocks, menus and pop-ups with
mouse point-and-click operations.
* Contact:
Signal Technology, Inc.
104 W. Anapamu, Suite J, Santa Barbara, CA 93101-3126
Phone: 805-899-8300 FAX: 805-899-4344
email: larry@signal.com
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.x.org:/contrib/audio/nas/netaudio-1.2.tar.gz">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: af-request@crl.dec.com
+ 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 (hgs@researh.att.com)
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: akra@zeus.di.uoa.ariadne-t.gr
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: malcolm@interval.com
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: 61-2-351 3847
University of Sydney, NSW, 2006, Australia email: andrewh@speech.su.oz.au