HDTenEightyP/NetNews huggingface dataset:
From: mhall@occs.cs.oberlin.edu (Matthew Hall)
Subject: Very simple speech recognition Alg. wanted.
Organization: Oberlin College Computer Science
Date: Mon, 9 Nov 1992 20:24:32 GMT
Distribution: comp.speech
X-Posted-From: occs.cs.oberlin.edu
NNTP-Posting-Host: sol.ctr.columbia.edu

Hello-
	I asked this question before, and recieved no replies.
However I did recieve at least five requests to pass information on.
If you can help, please do.  Many people want to know.

Simply the question is this - How does one implement a speaker
dependant, discrete recognition system?  For my purposes, the
vocabulary can be very small (<100 commands), but others have shown
interest in larger vocabularies.

Specifically, what data should one store - what patterns are unique to
different words.  How does one search a "dictionary" for a specific
word, and how does one quickly and somewhat accurately match a word
spoken to it's saved (pattern?)  The sound, at least in my case, is
stored in a raw waveform.  I am using pascal on a Macintosh, but I am
pretty flexible.

If you can help me and the other querents out, either by source code
or pointers to information, please do. There seems to be a great
interest in this.

Thank you,
-matt hall
--
-------------------------------------------------------------------------------
Matt Hall.    mhall@occs.oberlin.edu  OR  SMH9666@OBERLIN.BITNET
              (216)-775-6613 (That's a Cleveland Area code. Lucky Me)

"Life's good, but not fair at all"  -Lou Reed

Xref: sparky comp.speech:232 alt.binaries.sounds.d:2044
From: simon@lia.di.epfl.ch (Simon Leinen)
Newsgroups: comp.speech,alt.binaries.sounds.d
Subject: Re: Looking for GSM 06.10 spec/implementation
Date: 8 Nov 92 20:45:11 GMT
Followup-To: comp.speech
Organization: DI-LIA -- Ecole Polytechnique Federale de Lausanne
In-reply-to: minakami@Xenon.Stanford.EDU's message of 8 Nov 92 19:43:21 GMT
X-Md4-Signature: d8f3c99e953539231960ae0e34f4fa90

minakami@Xenon.Stanford.EDU (Michael K. Minakami) writes:

   Hi, does anyone out there have the GSM 06.10 specification in
   softcopy, or alternatively some C code implementing it?

An implementation can be FTPed from
	tub.cs.tu-berlin.de:/pub/tubmik/gsm-1.0.tar.Z
                           +/pub/tubmik/gsm-1.0-patch1

or as a faster but not always up-to-date alternative:
	liasun3.epfl.ch:/pub/audio/gsm-1.0pl1.tar.Z

See the README file after the signature.
-- 
Simon Leinen.
Laboratoire d'Intelligence Artificielle
Ecole Polytechnique Federale de Lausanne

/*
 * Copyright 1992 by Jutta Degener and Carsten Bormann, Technische
 * Universitaet Berlin.  See the accompanying file "COPYRIGHT" for
 * details.  THERE IS ABSOLUTELY NO WARRANTY FOR THIS SOFTWARE.
 */
 
GSM 06.10 13 kbit/s RPE/LTP speech compression available
--------------------------------------------------------

The Communications and Operating Systems Research Group (KBS) at the
Technische Universitaet Berlin is currently working on a set of
UNIX-based tools for computer-mediated telecooperation that will be
made freely available.

As part of this effort we are publishing an implementation of the
European GSM 06.10 provisional standard for full-rate speech
transcoding, prI-ETS 300 036, which uses RPE/LTP (residual pulse
excitation/long term prediction) coding at 13 kbit/s.

GSM 06.10 compresses frames of 160 13-bit samples (8 kHz sampling
rate, i.e. a frame rate of 50 Hz) into 260 bits; for compatibility
with typical UNIX applications, our implementation turns frames of 160
16-bit linear samples into 33-byte frames (1650 Bytes/s).
The quality of the algorithm is good enough for reliable speaker
recognition; even music often survives transcoding in recognizable 
form (given the bandwidth limitations of 8 kHz sampling rate).

The interfaces offered are a front end modelled after compress(1), and
a library API.  Compression and decompression run faster than realtime
on most SPARCstations.  The implementation has been verified against the
ETSI standard test patterns.

Jutta Degener (jutta@cs.tu-berlin.de)
Carsten Bormann (cabo@cs.tu-berlin.de)

Communications and Operating Systems Research Group, TU Berlin
Fax: +49.30.31425156, Phone: +49.30.31424315
From: cholette@JSP.UMontreal.CA (Cholette Martin)
Newsgroups: comp.speech
Subject: ************** speech recognition HELP *************
Date: 8 Nov 92 22:10:39 GMT
Organization: Universite de Montreal

Hello everyone! I have a 486 with a SB Pro sound card. I have to choose
a project for my bachelors degree in comp. sci. I am in my last year
and I have one month until project submissions deadline, then I will
have another 3 months to finish it.

I have always been interested in speech recognition and i am debating
whether or not to use this as my project. I have just got my SB Pro 
card 2 weeks ago, so I am not familliar with it, and I have no prior
experience with "sound" projects or theory. Can anyone tell me whether
or not I should try this for my project, What difficulty level am I
facing if I wish to do so, Can I do with my sound blaster? What kind
of programming challenges await me? Where should I start to look up
such a subject, what "theory" is involved? In other words any insight
on the matter would be greatly appreciated.


Thanks a million!

Martin Cholette
cholette@jsp.umontreal.ca
Newsgroups: comp.speech
From: ulrike@gmd.de (Ulrike Bauer)
Subject: information about DECtalk
Organization: GMD, Sankt Augustin, Germany
Date: Tue, 10 Nov 1992 13:15:34 GMT

I am a student who is writing her masters thesis about `generalization
in multilayer networks using the NETtalk files for training and
testing'. 

In this work I also compare some other systems generating phonemes
and stresses out of text, i.e. NETspeak, NETZSPRECH and DECtalk.

But till now I haven't got any real information about the DECtalk
system, how it works and which kind of research was done with it. I
haven't found any papers or articles concerning DECtalk. 

Please, could ANYBODY, who knows something about it, email me which
articles exist about DECtalk, how I can receive them in a very short
time or much better, whether, where and which articles I can get by
ftp. If you have an article about DECtalk in
it-doesn't-matter-what-kind-of-format and can email it to my address,
this will also be appreciated.

I am quite in a hurry to receive this/these articles, as I have write
down everything before Xmas.

Thanx a lot to all who are helping me to get the desired information
and articles about DECtalk.

I'm looking forward to your numerous answers.

				Ulrike

--
Ulrike Bauer	ulrike.bauer@gmd.de	(+49 2241) 14-2437
German National Research Center for Computer Science (GMD)           -- o
Research Group for Adaptive Systems - FIT.KI.AS                    _`\<, _
Schloss Birlinghoven, P.O.Box 1316, D-5205 Sankt Augustin 1, FRG  (_)/  (_)
Newsgroups: comp.speech
From: andrewh@ee.su.OZ.AU (Andrew Hunt)
Subject: Re: File format - Beginner Wants Info.
Keywords: audio file format
Reply-To: andrewh@ee.su.OZ.AU (Andrew Hunt)
Organization: University of Sydney, Australia
Date: Mon, 9 Nov 1992 03:47:40 GMT


In article <BxBwst.7Lw@cs.dal.ca>, roy@ug.cs.dal.ca (David Roy) writes:
> 
> This may seem like a very stupid question to any speech-recognition gurus out
> there, but I was wondering if anyone would have, or know of, a sound-file
> format type that would be relatively easy to read numeric values from.  
> What I mean by this is that I want a file type that I can pass through a
> function to get an array of values to use as a pattern-matching set.

There are a multitude of standard file formats in use as well as lots of
non-standard formats.  Some are easy to program with and others can be quite
complex.  Some also have software libraries available which make life easier.

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 defintions,
format conversion, standards, programming hints and lots more.  

Try looking at that list first.  I think you'll find several formats which
meet your requirments.

Andrew Hunt
andrewh@ee.su.oz.au
From: doug@rds.com (Doug Kaye)
Newsgroups: comp.speech
Subject: VSELP vs. CELP 3.2
Date: Tue, 10 Nov 92 15:19:27 PST
Distribution: world
Organization: Rational Data Systems, San Rafael, CA

We've had a very positive reaction to our voice annotation software
during the beta test period, but our customers have unanimously
requested three enhancements: compression, compression and compression.

We currently offer Intel's DVI 4-bit ADPCM, but people want more.  Joe
Campbell has been most helpful in getting us started on CELP 3.2 (4800
bps) which sounds pretty good.  Although the basic algorithms are
public, it seems that the implementations tend to be proprietary.

Motorola's VSELP sounds a bit better than 3.2 CELP, but you'd expect
that given its 7950bps rate.  Unlike DoD's CELP, Motorola has kept
VSELP proprietary and won't be making a decision on how to market the
algorithm until January '93 at the earliest.

In addition to DSP-assisted compression, we'd like to offer Intel-based
(ie, PC) decompression, so my question is, does anyone have any
experience with decompressing CELP or VSELP on a '386 or '486 PC? 
Real-time would be helpful, but not absolutely required, given that
annotation is not typically a real-time application.

Finally, I'm looking for opinions on the suitability of CELP and VSELP
(or other algorithms) for use in voice annotation applications.

    ...doug

==================================================
Doug Kaye   Rational Data Systems   San Rafael, CA
doug@rds.com   {moon|uunet}!rds!doug  MHS:doug@rds
==================================================
From: npopp@waddel.next.com (Nicholas Popp)
Newsgroups: comp.speech
Subject: Dynamic Time Warping Algorithms Wanted !!!!!!
Date: 14 Nov 92 01:12:54 GMT

Hi, netters,

Does anybody have the C code for the most common dynamic time warping  
schemes (and be ready to share it!!!). I am also looking for the code to  
compute the cepstal (cepstum, whatever...) coefficients. A big thanks in  
advance to the nice fellow who will save me a few hours of development.

- Nicolas
Newsgroups: comp.speech
From: roy@ug.cs.dal.ca (David Roy)
Subject: File format - Beginner Wants Info.
Organization: Math, Stats & CS, Dalhousie University, Halifax, NS, Canada
Date: Sat, 7 Nov 1992 04:36:29 GMT


This may seem like a very stupid question to any speech-recognition gurus out
there, but I was wondering if anyone would have, or know of, a sound-file
format type that would be relatively easy to read numeric values from.  
What I mean by this is that I want a file type that I can pass through a
function to get an array of values to use as a pattern-matching set.

This article seems confusing to me, and I KNOW what I'm talking about! 

Please, if you have any insight, e-mail me - or post to this group.

Thanks in advance.

David.

-- 
    David Roy    - Dalhousie Computing Science Student                  
     ///\\\      - Avid Skydiver                                        
    ///  \\\     - A.K.A. ->    daroy@ac.dal.ca, 01roy@ac.dal.ca
 (902) 492-8153                         roy@ug.cs.dal.ca
From: furuta@cs.umd.edu (Richard Furuta)
Newsgroups: comp.speech
Subject: Speech synthesis on a SPARC?
Date: 13 Nov 92 16:44:47 GMT
Organization: U of Maryland, Dept. of Computer Science, Coll. Pk., MD 20742

I would appreciate pointers to programs that produce synthesized speech on
Sun Sparcstations.  I'm sorry for the FAQ-like nature of this question (I
tried scanning back in this group as far as I could on two different systems
without finding any leads).  Thanks in advance.

				--Rick
				  furuta@cs.umd.edu
From: simons@tuntu.think.com (Joshua Simons)
Newsgroups: comp.speech
Subject: Re: information about DECtalk
Followup-To: comp.speech
Date: 10 Nov 92 15:58:24
Organization: Thinking Machines Corporation, Cambridge MA, USA
NNTP-Posting-Host: tuntu.think.com



Quoted from the DECTalk Owner's manual:

Fairbanks, G. Voice and Articulation Drillbook, Second Edition, Harper
and Row, 1960.

This book is intended for speech therapists and includes and introduction
to phonetics and acoustics. One of its diagnostic tests, the amplifier
passage (p. 114), was used to adjust DECtalk's speech rate parameters.

Bruckert, E., Minow, M. and Tetschner, W. "Three-Tiered Software and
VLSI Aid Development System to Read Text Aloud," Electronics, Vol 56,
No. 8, April 21, 1983, pp. 133-138.

The inner workings of DECtalk are described by its implementors.

Klatt, D.H. "Software for a Cascade/Paralleled Formant Synthesizer,"
Journal of the Acoustical Society of America, Vol 67, pp 971-995.

Describes the speech synthesizer that is the model for the one used in
DECtalk.

Klatt, D.H. "Review of the Science and Technology of Speech Synthesis," in
Overviews of Emerging Research Techniques in Hearing, Bioacoustics, and
Biomechanics, Proceedings of the 1981 Meeting, National Academy Press,
Washington, D.C., pp 15-38. (See also Speech Group Working Papers, MIT
Research Laboratory of Electronics, Vol III.)

Describes several speech synthesis technologies.

Klatt, D.H. "The Klattalk Text-to-Speech System," Proceedings of the 
International Conference of Acoustics, Speech and Signal Processing,
1982, IEEE Catalog No. 82CH1746-7, pp 1589-1592.

The linguist responsible for DECtalk describes its operation.



--
simons@think.com
thinking machines corporation
245 first street					tel: 617.234.2844
cambridge, ma  02142-1264				fax: 617.234.4444
Newsgroups: comp.speech
From: tiejun@utdallas.edu (Tiejun Zhang)
Subject: Test
Keywords: Test
Reply-To: $user@utdallas.edu
Organization: The University of Texas at Dallas, ACC
Date: Fri, 13 Nov 1992 22:34:48 GMT

This is only a test. Please ignore it. Thanx.


Newsgroups: comp.speech
From: wb9omc@dynamo.ecn.purdue.edu (Duane P Mantick)
Subject: SP1000/VRS1000 help!!!!!!
Keywords: hosed
Organization: Purdue University Engineering Computer Network
Date: 10 Nov 92 16:01:16 GMT


A friend of mine has some troubles trying to use this speech chip set.
He is using it for his EET Senior Design Project here at Purdue, and
cannot simply change to another item.

If there are any of you out there who can help him, please email him
at schlutt@harbor.ecn.purdue.edu.

Specifically, he'd like to get his hands on the VRS1000 manual or at
least some info from it.

Duane
wb9omc@harbor.ecn.purdue.edu
From: doug@rds.com (Doug Kaye)
Newsgroups: comp.speech
Subject: Compression Divergence
Date: Wed, 11 Nov 92 15:54:13 PST
Distribution: world
Organization: Rational Data Systems, San Rafael, CA

Normally (ie, in a real-time enviornment) audio is compressed then
decompressed and that's it.  However in certain non-real-time
applications audio may be compressed and decompressed repeatably.  In
such applications it is important that the compression/decompression
algorithms be "convergent".  By this, I mean that within a small number
of compress/decompress cycles the distortion ceases to increase.

We are currently using Intel's DVI ADPCM algorithm because (among its
other advantages) it converges after the very first such cycle. 
Successive compress/decompress cycles result in identical data.

Has anyone had experience with regard to this convergence pheonomenon
with CELP?  VSELP?  MPEG?  Other audio compression algorithms?

	...doug

==================================================
Doug Kaye   Rational Data Systems   San Rafael, CA
doug@rds.com   {moon|uunet}!rds!doug  MHS:doug@rds
==================================================
Newsgroups: comp.speech
From: joel@ecse.rpi.edu (Joel Glickman)
Subject: Speech recognition (Set of learned commands)
Date: Thu, 5 Nov 1992 14:40:39 GMT

This is probably in some FAQ of some kind .. but .. where would I find more
information about developing a limited-vocabulary speech recognition system.
I want to keep it simple .. The user would "teach" the system a small set of
verbal commands (20 or so). The micro/DSP/ADC would then listen to a verbal
instruction and compare the spectrum/signature to the two knowns to determine
which one it might be..

Anyone know of any good references for this? I want to implement it using a
microcontroller and DSP processor..
-Joel
Newsgroups: comp.speech
From: mathew@elroy.Jpl.Nasa.Gov (Mathew Yeates)
Subject: Re: information about DECtalk
Organization: Image Analysis Systems Group, JPL
Date: Wed, 11 Nov 1992 20:23:28 GMT

But what is DecTalk? Where's the FAQ?

Mathew Yeates
From: mfulsmb@uts.mcc.ac.uk (Martin Barry)
Newsgroups: comp.speech
Subject: Re: Different accents
Date: 4 Nov 92 14:55:38 GMT
Organization: Manchester University, M13 9PL, UK.

In article <Bx6G4L.708@alsvid.une.edu.au> jack@alsvid.une.edu.au (Jack Claff) writes:
>I guess this is more of a linguistics question.  I want to get information on
>how to speak in different accents.  In America for example, there is a big
>difference between people from say New York, Texas and Alabama.  Can anyone
>provide me with a reference that shows how words are pronounced using
>different accents. 
>
>I'd like to get Irish, Scottish and a variety of English accents as well.

Accents of English, by J.C. Wells, Cambridge University Press, 1982.
In three volumes, of which vol. 2 covers the British Isles in detail, and
vol. 3 has a lengthy section (100 pp) on North America.

It's entirely atheoretical and rather anecdotal, but perhaps the nearest
thing to a comprehensive work of reference available. And highly readable,
if you're interested in the subject matter.

Martin Barry, Department of Linguistics, University of Manchester, UK.
Xref: sparky comp.speech:230 comp.ai:4196 comp.human-factors:2575 comp.sys.sun.wanted:1768 comp.multimedia:3284
From: janina@netcom.com (World Institute on Disability)
Newsgroups: comp.speech,comp.ai,comp.human-factors,comp.sys.sun.wanted,comp.multimedia,misc.
Subject: Re: voice computer control
Keywords: computers, voice input, 'keyboard' simulation
Date: 8 Nov 92 16:29:41 GMT
Organization: Netcom - Online Communication Services (408 241-9760 guest)

Speech to text systems certainly exist.  But they have trouble distinguishing
one word's end from another's beginning and, therefore, prefer you to pause
between words.  Makes you talk in a manner unto which you're probably not accustomed.  So, if you're
willing to "blurt" your thoughts out that way, you'll be OK.  Else, you may
have to wait.
 
				Janina Sajka
Newsgroups: comp.speech
From: x89olarte1@gw.wmich.edu
Newsgroups: comp.speech
Subject: Could S.Blaster recognize sounds??HELP
Date: 10 Nov 92 15:46:16 EST
Organization: Western Michigan University

I may be naive but that doesn't take my right to ask away  :")

The sound board Soundblaster Pro has a mic input.

would it be too hard to make a program that once you have in a disk say

100 words saved (the wavelength or whatever) . when you talk into the board 

w/the mic it compares it and if it's close enough it performs some action

corresponding to that word , and if no one matches do nothing or say

"repeat" or some like that ??


If you have an answer (either YES or No or MAYBE) please E-mail me.


Thanks a lot in advance.

Enrique
Newsgroups: comp.speech
From: danforth@riacs.edu (Douglas G. Danforth)
Subject: Re: Very simple speech recognition Alg. wanted.
Organization: RIACS, NASA Ames Research Center
Distribution: comp.speech
Date: Thu, 12 Nov 92 23:25:26 GMT


Addendum to QUICKY:

Amplitude normalization: 
  Forgot to mention that the amplitude in each segment should be normalized
otherwise loud sounds will look different from  soft sounds even though
the same word is spoken.  The Interstate Algorithm actually just
uses the amplitude difference (1 bit) from one lowpass filter to the
next higher frequency within a segment.  For 16 filters there are
15 differences.  An increase gets a 1 bit.  No change or decrease gets
a 0 bit.  The result is a 256 bit (pad low frequency with zeros) pattern.
You can try other schemes as well.

Doug Danforth

Newsgroups: comp.speech
From: minow@apple.com (Martin Minow)
Subject: Re: information about DECtalk
Date: Tue, 10 Nov 1992 18:41:49 GMT
Organization: Macintosh Developer Services
Followup-To: comp.speech

In article <ulrike.721401334@gmd.de>, ulrike@gmd.de (Ulrike Bauer) wrote:
> 
> Please, could ANYBODY, who knows something about it, email me which
> articles exist about DECtalk, ...

The DECtalk documentation (your local Dec office might be able to find a
copy) has a bibliography and summary description. From memory, there are
several sources for more information.

-- Dennis Klatt, who invented the technology used in DECtalk, published
   extensively. Most of his work was published by JASA (Journal of the
   Acoustic Society of America). Any good university library would have
copies,
   and you can use bibliographies to trace the data further.

-- The MIT Speech group published a large volume describing MITalk, the
   research predecessor to DECtalk. Jon Allen is the primary author, and
   this, too, should be in a university library. I believe it was published
   by MIT Press.

-- Electronics Magazine published a non-technical description of DECtalk
   in early 1984. Although the authors were listed as Klatt, Bruckert, and
   Minow (but not necessarily in that order), the actual writing was done
   by a contracted journalist. (We did edit the result, of course.)

-- You might also look for reports from the MIT Research Laboratory of
   Electronics, where Dennis' work often appeared in pre-print or draft
form.

-- The letter-to-phoneme rules used in the original DECtalk were based on
   Chomsky & Halle, Sound Pattern of English. The software emulation was
   written by Sherri Hunnicutt and published in a microfiche series by
   a journal of computational linguistics whose precise name I can't
recall;
   IJCAI, perhaps.

Hope this helps.

Martin Minow
minow@apple.com
Newsgroups: comp.speech
From: polish@cs.columbia.edu (Nathaniel Polish)
Subject: Looking for AVIOS
Organization: Columbia University Department of Computer Science
Distribution: usa
Date: Tue, 10 Nov 1992 21:42:32 GMT


I am looking for the phone number of an organization called AVIOS. 
AIOS stands for: American Voice Input/Output Society.
Any information would be appreciated.

Thanks

Nat Polish
Newsgroups: comp.speech,alt.binaries.sounds.d
From: minakami@Xenon.Stanford.EDU (Michael K. Minakami)
Subject: Looking for GSM 06.10 spec/implementation
Originator: minakami@Xenon.Stanford.EDU
Organization: Computer Science Department, Stanford University.
Date: Sun, 8 Nov 1992 19:43:21 GMT

Hi, does anyone out there have the GSM 06.10 specification in softcopy,
or alternatively some C code implementing it?

Thanks,
Mike
-- 
-------------------------------------------------------------------------------
Michael Minakami            | These thoughts are my own, not that anyone's 
Computer Science/Psychology | tried claiming them.
Stanford University         |
From: barons@media-lab.mit.edu (Barry Arons)
Newsgroups: comp.speech
Subject: Re: Looking for AVIOS
Date: 11 Nov 92 19:24:10 GMT
Article-I.D.: news.1992Nov11.192410.16107
Distribution: usa
Organization: M.I.T. Media Laboratory

AVIOS stands for American Voice Input/Output Society.  AVIOS has one conference per year, usually in September.  It also publishes two journal issues per year.
Avios can be contacted at: 4010 Moorpark Ave, Suite 105M, San Jose CA, 95117.
(408) 248-1353.

Barry.
Newsgroups: comp.speech,comp.dsp
From: sliberm@nyx.cs.du.edu (Sergio Liberman)
Subject: disguizing identity of talker
Organization: University of Denver, Dept. of Math & Comp. Sci.
Date: Thu, 12 Nov 92 14:52:47 GMT


What kind of algorithms exist that are capable of 
disguizing the voice of a person?

I remeber reading once about the LPC speech compression
algorithm where if the bit rate was reduced enough, then it
was impossible to identify the talker after decompressing
the speech. Are there other methods?

Sergio
---
sliberm@nyx.cs.du.edu