HDTenEightyP/NetNews huggingface dataset:
From: someone@Texaco.com (Larry D. Pyeatt)
Subject: Re: Could S.Blaster recognize sounds??HELP
Organization: Texaco
Date: Fri, 20 Nov 1992 21:42:05 GMT

In article <1992Nov10.154616.6281@gw.wmich.edu>, x89olarte1@gw.wmich.edu writes:
|> I may be naive but that doesn't take my right to ask away  :")

That`s okay, we all start out naive.

|> 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 ??

Well..  It depends on your definition of too hard.  If you have spent the 
last four years learning everything you can about speech recognition systems,
then you would probably be able to do it after a few weeks of heavy programming.
If, however, you are totally new to speech recognition, it will take at least
6 months of reading and experimentation before you understand enough to begin
solving the problem. I did my MS research in speech recognition with neural
networks.  I have been thinking about trying to do what you are describing.
The main drawback that I see is the fact that the PC really does not have a
lot of computing power, and it is a pain to try and do multi-tasking under
MS-DOS.

If you are really interested, you should go to the library and start reading
the scientific journals on speech recognition.  You may want to look at using
a Hidden Markov Model ( HMM ), although it is really not that easy to  
understand at first.  You should also limit the vocabulary to about ten words
for your first attempt.  You should modularize your code as much as possible.
The major modules should perform the following:
1. signal aquisition,
2. preprocessing, and
3. recognition.

There are several different approaches to preprocessing.  The preprocessing
tecnique can make the difference between success and failure.  The result of 
preprocessing is to convert the raw signal into a series of "feature vectors"
which can be fed into the pattern recognizer.  You may want to use 
LPC cepstra, wavelet coeffecients, or any of a number of techniques.  Some
preprocessing techinques model the physical and neuronal processing which
takes place in the cochlea and aural pathways.  Other techniques ignore
biological systems and strike out in their own direction.

The recognizer takes a series of feature vectors and tries to relate them
to known patterns.  There are a lot of ways to solve this problem.  Some
approaches are good for single speaker, small vocabulary problems while 
others may be better suited to speaker independent, medium vocabulary 
problems.

-- 
Larry D. Pyeatt                 The views expressed here are not
Internet : pyeatt@texaco.com    those of my employer or of anyone
Voice    : (713) 975-4056       that I know of with the possible
                                exception of myself.
From: janin@cco.caltech.edu (Adam L. Janin)
Newsgroups: comp.speech
Subject: Looking for PD or low price voice recog for sparcs
Date: 20 Nov 1992 22:55:55 GMT
Organization: California Institute of Technology, Pasadena
NNTP-Posting-Host: punisher.caltech.edu

Sorry if this is a FAQ, but I'm looking for a free or low price voice
recognition system for the SPARC station.  Speaker dependent, non-continuous
and very small vocabulary (10-20 words) is sufficient.

	Thanks,

	Adam Janin
	janin@espresso.boeing.com
From: rid@ecs.soton.ac.uk (Bob Damper)
Newsgroups: comp.speech
Subject: Re: information about DECtalk
Date: 17 Nov 92 14:43:09 GMT

In <minow-101192102931@minow.apple.com> minow@apple.com (Martin Minow) writes:

>-- 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.

Try his 1987 ``Review of Text-to-Speech Conversion for English'' in
JASA (Vol. 82, pages 737 -- 792).  It's absolutely excellent (his
remarks on connectionism notwithstanding).  

>-- 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.

Cambridge University Press actually, in 1987.

>-- The letter-to-phoneme rules used in the original DECtalk were based on
>   Chomsky & Halle, Sound Pattern of English. 

Not quite.  The *formalism* (i.e.~context-dependent rewrite rules)
comes from Chomsky and Halle (1968) but not their specific content.
This formalism was first used for text-to-phoneme conversion by Bill
Ainsworth back in 1973.  Since then, it has become very popular.

>-- 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.

American Journal of Computational Linguistics, Microfiche 57, 1976,
pages 1 -- 72.  Its title is ``Phonological rules for text-to-speech
conversion''.

Bob Damper
rid@uk.ac.soton.ecs
Newsgroups: comp.speech
From: roy@ug.cs.dal.ca (David Roy)
Subject: ULAW information wanted.
Organization: Math, Stats & CS, Dalhousie University, Halifax, NS, Canada
Date: Sun, 15 Nov 1992 20:41:55 GMT


Hi, I'm in need for precise information on the ULAW format of sound
files.  I am attempting to create a basic speech recognition program
on a SUN, so I am using .au files which use ULAW encoding.  I am 
looking for a way to easily convert each sample to an integer that
will be a representation of the amplitute of the sample.

If anyone has information, or knows where it can be found, please e-mail
me ASAP - avoid posting I guess, this is probably a FAQ that just
happens not to be on the FAQ list!

Anyway, Thanks in advance.  Any help whatsoever would be greatly
appreciated.

-- 
    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: mfulsmb@uts.mcc.ac.uk (Martin Barry)
Newsgroups: comp.speech
Subject: Re: Help!! Looking for the most recent published books or papers
Date: 20 Nov 92 10:29:35 GMT
Organization: Manchester University, UK.

Another good introductory book, perhaps less technical than the ones Tony
Robinson mentions (hi, Tony) is:

Douglas O'Shaughnessy -- Speech Communication: Human and Machine
(Addison Wesley series in Electrical Engineering: Digital Signal Processing,
1987).

Martin Barry,
Department of Linguistics,
University of Manchester. (e-mail: M.C.Barry@Manchester.Ac.UK)
Newsgroups: comp.speech
From: alvin@netcom.com (Alvin H. White)
Subject: Re: Phonemic analyzer construction
Organization: Netcom - Online Communication Services  (408 241-9760 guest) 
Date: Tue, 24 Nov 1992 00:41:45 GMT

someone@Texaco.com (Larry D. Pyeatt) writes:

>You may want to look at HearSay II for inspiration.

Interesting title "HearSay II". It would seem fun to hear a description
or where it can be ftp'ed, etc.

alvin

Postal:  Alvin H. White, Gen. Sect.
         G.O.D.S.B.R.A.I.N.
         1505 De Rose Way #66 
         San Jose, CA 95126-4133 USA 

   Tel:  +1 408-286-2794
E-mail:  alvin@netcom.com

======================================================== 
Government Online Database Systems
Bureau for Resource Allocations to Information Networks 
[an idea waiting to happen]

120 West 38 North

Newsgroups: comp.speech
From: bst@well.sf.ca.us (Berkeley Speech Technologies)
Subject: Re: Speech synthesis on a SPARC?
Organization: Whole Earth 'Lectronic Link
Date: Mon, 16 Nov 1992 20:18:12 GMT

In article <62025@mimsy.umd.edu> furuta@cs.umd.edu (Richard Furuta) writes:
>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.

We offer a speech program for the SparcStation that accepts typed input
(you can also tell it to read files) and produces output using /dev/audio
Sorry, no source code available.  Currently available only in English, but
we have other languages on other platforms so you might ask when you call.
Contact Sara O'Malley at BST.

posted by Robert Bickford for BST
--
bst@well.sf.ca.us                                            (510)-841-5083
Berkeley Speech Technologies               The Full-Spectrum Speech Company
2246 Sixth Street  /  Berkeley  CA  94710
-- 
bst@well.sf.ca.us                                            (510)-841-5083
Berkeley Speech Technologies               The Full-Spectrum Speech Company
Newsgroups: comp.speech
From: someone@Texaco.com (Larry D. Pyeatt)
Subject: Re: Phonemic analyzer construction
Organization: Texaco
Date: Mon, 23 Nov 1992 16:02:24 GMT

In article <1992Nov23.133836.11680@hubcap.clemson.edu>, bdbryan@eng.clemson.edu (Ben Bryant) writes:
|> G'day Sirs,
|> I am thinking about building a connectionist phoneme analyzer, and am interested
|> in finding out some ideas about how to go about designing the "higher-level"
|> classifier which will discriminate among the outputs from several previously
|> trained "subclass instant" neural nets.
|> 
|> Basically, the way this would work is that a suitable NN architecture would
|> be chosen for the "lower-level" signal analysis stage, and instances of this
|> architecture would be trained using TIMIT or some other large database.
|> 
|> The way the training would take place is as follows:
|> 1) first the training tokens for each phonemic subclass would be extracted
|>    from the database.
|> 2) the phoneme tokens for each phonemic subclass extracted in step one would
|>    then be preprocessed with an appropriate feature representation technique.
|> 3) network instances would be trained using the chosen neural network architecture.
|>    A network instance will be trained for each phonemic subclass (i.e., voiced-stops,
|>    unvoiced-stops, diphthongs, vowels, etc.).
|> 4) after training all network instances, the outputs from the trained subnetworks
|>    would "somehow" be arbitrated to provide a decision of which phoneme was uttered
|>    within a given region of signal.
|> 
|> -The "somehow" in step 4) is what I really could use some help with. Any other
|> ideas for this system would be welcome as well. Thank you very much.
|> 
|> Sincerely,
|> -Benjamin Bryant 
|> 
|> 
|> 

I would suggest using a BlackBoard system to do the arbitration.
You may want to look at HearSay II for inspiration.  Alternatively,
you could try a connectionist approach to the arbitration problem.
I do not know of anyone who has done previous work with the 
connectionist approach, but would like to hear any ideas.  I have
been thinking about doing something similar to what you have
suggested for my PhD research, which I hope to start next fall.

-- 
Larry D. Pyeatt                 The views expressed here are not
Internet : pyeatt@texaco.com    those of my employer or of anyone
Voice    : (713) 975-4056       that I know of with the possible
                                exception of myself.
Newsgroups: comp.speech
From: george@beta.ee.ufl.edu (George L)
Subject: Re: Looking for speech synthesis chips
Keywords: speech, synthesis
Organization: EE Dept at UF
Date: Sun, 22 Nov 92 07:03:55 GMT

In article <greg.722092872@coombs> greg@coombs.anu.edu.au (Greg Price) writes:
>eeltw@marlin.jcu.edu.au (Lloyd Walker) writes:
>
>>minimum order.  I have little info on Microchip SP0264 & Sp0256A;
>                                                          ^^^^^^^
>I think Tandy were/are selling the sp0256 and its companion chip the
>cts256a-al2. I just looked up my rod irving catalog and they sell both.
>sp0256 is $21.95 and the cts256a-al2 is $45.50. Rod irving has stores
>in melb and sydney, but they also do mail order. I know the sp0256's 
>output is a bit raspy, but anyway, the cts256a-al2 is a text to phoneme

Actually, the SP0256 received a lot of bad press in its most documented mode 
of operation (in the literature which accompanied the Radio Shack packaging).
Ever wonder what those extra address pins were for? The chip was capable of 
being fed with parameters which took much better advantage of its internal 
structure (vocal tract filter, excitation generators, etc...).

When run in this mode, the output was extremely higher in quality. Much like 
a high quality LPC recording. Intonations and inflections were plainly 
intelligible (but then again, I've been known to watch a snowy TV picture 
if its due to really good skip).

Of course, the bit rate was higher, but using English text input to the 
CTS256 as a comparison is not really comparing apples to oranges.

I can't find my literature on all this, but I think there were some other chips
made to support this family (memory expansion, etc...). I think it was all 
made by General Instrument. I tried to track them down a few years ago, but 
they had been sold off to some other company, and I have been told that the 
new company discontinued the product line.

Its a shame, I am sure many people experimented with this chip without 
discovering its true potential. So now, I have a few questions for the net:

1) Is this chip still manufactured by anyone?

2) Anyone have the official specifications for the extended modes?

3) Any previous General Instruments (I think) people out there that worked on 
this chip?

Thanks,
george@alpha.ee.ufl.edu
Newsgroups: comp.speech
From: mmcohen@cats.ucsc.edu (Michael M Cohen)
Newsgroups: comp.speech,comp.dsp,sci.lang
Subject: Re: Speech lab questions
Date: 21 Nov 1992 01:47:55 GMT
Organization: University of California; Santa Cruz
NNTP-Posting-Host: si.ucsc.edu


In article <1992Nov19.204229.12631@liberty.uc.wlu.edu> urbacht@liberty.uc.wlu.edu (Tom Urbach) writes:
>I'm looking into setting up a microcomputer or workstation based
>speech lab here at my uni and would like to get suggestions and
>recommendations. References to relevant FAQ archives would also be a
>help.

For a good survey of such systems see
Read, C., Buder, E.H., & Kent, R.D. (1992)
Speech Analysis Systems: An Evaluation.
J. of Speech and Hearing Research, APril 1992
p314-332


-- 

======================================================================
=   Dr. Michael M. Cohen                   mmcohen@dewi.ucsc.edu     = 
=   Program in Experimental Psychology     mmcohen@fuzzy.ucsc.edu    =
=   433 Clark Kerr Hall                    408-459-2655 LAB          = 
=   University of California - Santa Cruz  408-459-2700 MSGS         = 
=   Santa Cruz, CA 95064                   408-459-3519 FAX          = 
======================================================================
Newsgroups: comp.speech
From: bdbryan@eng.clemson.edu (Ben Bryant)
Subject: Phonemic analyzer construction
Reply-To: bdbryan@eng.clemson.edu
Organization: College of Engineering, Clemson Univ.
Date: Mon, 23 Nov 1992 13:38:36 GMT

G'day Sirs,
I am thinking about building a connectionist phoneme analyzer, and am interested
in finding out some ideas about how to go about designing the "higher-level"
classifier which will discriminate among the outputs from several previously
trained "subclass instant" neural nets.

Basically, the way this would work is that a suitable NN architecture would
be chosen for the "lower-level" signal analysis stage, and instances of this
architecture would be trained using TIMIT or some other large database.

The way the training would take place is as follows:
1) first the training tokens for each phonemic subclass would be extracted
   from the database.
2) the phoneme tokens for each phonemic subclass extracted in step one would
   then be preprocessed with an appropriate feature representation technique.
3) network instances would be trained using the chosen neural network architecture.
   A network instance will be trained for each phonemic subclass (i.e., voiced-stops,
   unvoiced-stops, diphthongs, vowels, etc.).
4) after training all network instances, the outputs from the trained subnetworks
   would "somehow" be arbitrated to provide a decision of which phoneme was uttered
   within a given region of signal.

-The "somehow" in step 4) is what I really could use some help with. Any other
ideas for this system would be welcome as well. Thank you very much.

Sincerely,
-Benjamin Bryant
Newsgroups: comp.speech
Subject: Software for speech proc. with DT2821
From: nnguyen@fapse.unige.ch (NGUYEN Noel)
Date: 19 Nov 92 10:01:12 +0200
Organization: University of Geneva, Switzerland
Keywords: speech processing, spectral analysis, DT2821


Hi users.


  I am looking for a speech processing software working with the Data Translation
DT2821 I/O card. I'd like to have a program allowing to edit one or many acoustic
speech signals, to make spectral analyses such as an FFT or an LPC using various
windows (Hamming, Kaiser...), to compute spectrograms and so on. Does someone know
whether such a program exists or not? Any information on that point
would be very helpful.

Noel Nguyen
laboratory of psycholinguistics
Faculty of Psychology
University of Geneva, Switzerland.
From: greg@coombs.anu.edu.au (Greg Price)
Newsgroups: comp.speech
Subject: Re: Looking for speech synthesis chips
Date: 18 Nov 92 13:21:12 GMT
Organization: Australian National University
NNTP-Posting-Host: 150.203.76.2
Keywords: speech, synthesis

eeltw@marlin.jcu.edu.au (Lloyd Walker) writes:

>minimum order.  I have little info on Microchip SP0264 & Sp0256A;
                                                          ^^^^^^^
I think Tandy were/are selling the sp0256 and its companion chip the
cts256a-al2. I just looked up my rod irving catalog and they sell both.
sp0256 is $21.95 and the cts256a-al2 is $45.50. Rod irving has stores
in melb and sydney, but they also do mail order. I know the sp0256's 
output is a bit raspy, but anyway, the cts256a-al2 is a text to phoneme
converter which can be directly attached to the sp0256. There were some
kits using both a while back. You could probably get the data for them.

>	All I want to do is to store up to 8 messages that can be
>replayed at the press of a button from the portable unit.  I'm very open
>to suggestions as it is a major detraction from the final product for
>the client.

Jaycar sell a digital voice recorder kit for $99.50 (KC-5072) which is
the kit from the article in "Silicon Chip" Dec. 89. It can store
4 x 30 sec. messages, or they can be ganged together. Apparently there
are triggered inputs, or you can hook it up to your conputer. I don't know 
how easy it would be to convert it to 8 messages. I haven't used one, or
seen the schematic for it.

Good luck,
Greg

>	God Bless,  Lloyd.

>/-----------------------------------------------------------------------\
>( Lloyd Walker,  PhD	       : Internet: L_Walker@trout.jcu.edu.au	)
>( Faculty of Engineering         :   or      eeltw@marlin.jcu.edu.au    )
>( James Cook University of North Qld: 				      	)
>( Douglas		         :			._   /\		)
>( Queensland                   :  		       /  \_/ *\	)	
>( AUSTRALIA     4811          :                       (         )     	)
>(  Ph  (077) 81 5410       : Fax +61 77	251 348        \_.--.__/        )
>(---------------------------------------------------          V         )
>(	      AUSTRALIA - Has to be seen to be believed!                )
>\-----------------------------------------------------------------------/
From: alissali@imag.fr (Mamoun ALISSALI)
Newsgroups: comp.speech
Subject: Re: information about DECtalk
Date: 16 Nov 92 11:04:39 GMT
Organization: IMAG Institute, University of Grenoble, France

In article <mathew.721513408@henry> mathew@elroy.Jpl.Nasa.Gov (Mathew Yeates) writes:
>But what is DecTalk? Where's the FAQ?

  I don't know about any FAQ for this groupe, and I'd like to see one.

  I'm trying to get more information about DECTalk, till then I'll tell the
  little I know.
  DECTalk is a text-to-speech system, I think it is a kind of peripheral with
  an access-function libraray. It's based on the Klattalk system developped by
  D. Klatt after his participation in the MITalk project.
  so here it is, I'd be grateful to hear more about the subject (TTS systems in
  general). If enough people are interrested I'd post more about work done in
  Europe in the field.

Mamoun.
Newsgroups: comp.speech
From: freed@europa.orion.adobe.com (Alex Freed)
Subject: Re: Looking for speech synthesis chips
Organization: Adobe Systems Inc
Date: Wed, 18 Nov 1992 22:54:29 GMT


I happen to have two SP0256A chips with full docs. The first $20 takes both.
--
 _______________________________________________________
| -Alex Freed (The opinions expressed are my own.   	|		    
|	        However everyone is entitled to them.)	|		    
| freed%adobe.com@uucp-gw-1.pa.dec.com			|
 -------------------------------------------------------
From: rid@ecs.soton.ac.uk (Bob Damper)
Newsgroups: comp.speech
Subject: Re: information about DECtalk
Date: 17 Nov 92 14:28:07 GMT

In <ulrike.721401334@gmd.de> ulrike@gmd.de (Ulrike Bauer) writes:

>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.

Beware -- it is not a simple matter to compare text-to-phoneme
systems.  Even if they use the same principles (e.g.~both systems are
based on back-propagation networks, like NETtalk and NETspeak), the
testing reported by the authors is usually quite different -- to the
extent that comparison is difficult.  When the systems use totally
different principles, it is not entirely clear how they should be
compared.  For a self-learning system like NETtalk, the tradition is to
test on unseen data -- not present in the training set.  But the
concepts of `unseen data' and `training set' aren't even applicable to
systems like DECtalk where rules are manually-derived by experts.

I have written something on these problems -- the manuscript is
nearing completion.  Email me if you want to know more.

>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. 

DECtalk is the commercial embodiment of the KLATTalk rule-based system
from MIT.  Major contributors were Dennis Klatt, Jonathan Allen and
Sheri Hunnicutt.  There's an enormous literature on text-to-phoneme
conversion by rule: the above names figure prominently.  Try the book
``From Text to Speech: the MITalk System'' by Allen, Hunnicutt and
Klatt (1987).

Bob Damper
Vision, Speech and Signal Processing Research Group
Department of Electronics and Computer Science
University of Southampton
Southampton S09 5NH
United Kingdom                Email: rid@uk.ac.soton.ecs
Xref: sparky comp.speech:270 comp.dsp:2597
From: alissali@imag.fr (Mamoun ALISSALI)
Newsgroups: comp.speech,comp.dsp
Subject: Re: disguizing identity of talker
Date: 16 Nov 92 11:11:34 GMT
Followup-To: comp.speech
Organization: IMAG Institute, University of Grenoble, France

In article <1992Nov12.145247.24996@mnemosyne.cs.du.edu> sliberm@nyx.cs.du.edu (Sergio Liberman) writes:
>
>What kind of algorithms exist that are capable of 
>disguizing the voice of a person?
>
[...]
 Are there other methods?

 If I got the question right, I'd suggest you to read about synthesis by
 analysis methods. One that is becoming more and more used is OLA (OverLap and
 Add) with its variants (PSOLA, TD-PSOLA). The idea is to analyse registered
 sound segments, extract some parameters, modify appropriate value (like pitch,
 which might answer your question), then reconstruct the signal with the new
 parameters.
 If this is what you want I could give some pointers to the subject.

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


Newsgroups: comp.speech