HDTenEightyP/NetNews huggingface dataset:
From: malcolm@Apple.COM (Malcolm Slaney)
Newsgroups: comp.dsp
Subject: Re: Pitch shifting
Date: 6 Jan 93 04:43:35 GMT
Organization: Apple Computer Inc, Cupertino, CA

cmh@eng.cam.ac.uk (C.M. Hicks) writes:
> - Signal Modelling approach. Model the signal (eg linear prediction) and
>   adjust the model parameters so that when the signal is reconstructed, it
>   is at the changed pitch.
>   Main problems: model choice, block boundaries, computational load.

Xavier Serra did some really excellent work on this approach when he was
at Stanford.  His thesis talk included some great examples of just the
pitchy (deterministic) and noisy components of a musical signal.  He also
played some examples that were pitch shifted.  They sounded perfect to
my ears.

Here's the complete reference from INSPEC.

							Malcolm Slaney
							Apple Perception Group
							and Stanford CCRMA

TITLE:         Spectral modeling synthesis: a sound analysis/synthesis system
                 based on a deterministic plus stochastic decomposition.
AUTHOR:        Serra, X. (Dept. of Music, Stanford Univ., CA, USA); Smith, J.,
                 III
PUBLICATION:   Computer Music Journal (Winter 1990) vol.14, no.4, p. 12-24. 21
                 refs.
DOCUMENT TYPE: Journal article
LANGUAGE:      English
ABSTRACT:  When generating musical sound on a digital computer, it is important
   to have a good model whose parameters provide a rich source of meaningful
   sound transformations. This paper addresses the synthesis technique of
   spectrum modeling. It describes a technique called spectral modeling
   synthesis (SMS) that models time-varying spectra as (1) a collection of
   sinusoids controlled through time by piecewise linear amplitude and
   frequency envelopes (the deterministic part), and (2) a time-varying
   filtered noise component (the stochastic part). The analysis procedure first
   extracts the sinusoidal trajectories by tracking peaks in a sequence of
   short-time Fourier transforms. These peaks are then removed by spectral
   subtraction. The remaining 'noise floor' is then modeled as white noise
   through a time-varying filter. A piecewise linear approximation to the upper
   spectral envelope of the noise is computed for each successive spectrum, and
   the stochastic part is synthesized by means of the overlap-add technique.
   The SMS technique has proved to give general, high quality transformations
   for a wide variety of musical signals.
From: tcreasy@fox.nstn.ns.ca (Tim Creasy)
Newsgroups: comp.dsp
Subject: Motorola 56001 Transistor Count
Date: 5 Jan 93 13:31:37 GMT
Organization: NSTN Network Operations Centre, Nova Scotia, Canada

Does anyone have an idea of the number of transistors
in a DSP56001 processor?   Ballpark would be OK.

Thanks,
TiM
Newsgroups: comp.dsp
From: mek@guinan.psu.edu (Mark E. Kotanchek)
Subject: Re: Imaginary signals
Organization: Penn State Center for Academic Computing
Date: Tue, 12 Jan 93 15:59:28 GMT

In article <TIM.93Jan10124435@ear-ache.mit.edu> tim@ear-ache.mit.edu (Tim  
Wilson) writes:
> 880840m@axe.acadiau.ca (MICHAEL ALEXANDER MCKAY) writes:
> 
>    Is there a time-domain filter which, given a real signal as input,  
will
>    output the corresponding imaginary signal?
>    I've tried to make a filter with unity gain and pi/2 phase shift for
>    all frequencies, but with no luck.
> 
> See Oppenheim and Shafer, _Digital Signal Processing_ (1st edition),
> Section 7.4:  "Hilbert Transform Relations for Complex Sequences."  In
> particular, see Section 7.4.1, "Design of Hiblert Transformers."
> I quote:  "...the ideal Hibert transformer or 90-degree phase shifter
> takes its place alongside the ideal lowpass filter and ideal
> bandlimited differentiator as valuable theoretical concepts which
> correspond to noncausal systems and for which the system function
> exists only in a restricted sense."
> 
> In the same section, O&S discuss finite-duration approximations of the
> ideal Hilbert transformer using frequency-sampling, windowing, and
> equiripple approximations.  Finally, they reference B. Gold, A. V.
> Oppenheim, and C. M. Rader, "Theory and Implementation of the Discrete
> Hibert Transform," _Proc. Symp. Computer Processing in
> Communications_, Vol. 19, Polytechnic Press, 1970, New York for a
> recursive implementation of a 90-degree phase-splitting system.
> 
> Good luck.
> --
> Tim Wilson
> Internet:  tim@ear-ache.mit.edu
> UUCP:  mit-eddie!mit-athena!tim

The Hilbert filter approach is the "correct" way to derive an analytic  
equivalent to a real-valued signal. However, if you can assume that your  
PASSBAND signal is also narrowband, there is an alternative. Assume that  
we have a passband centered at a frequency fc having a bandwidth BW and  
that the signal has been sampled at a frequency fS. Furthermore, the  
narrowband assumption implies that fc >> BW. Then, given the data stream  
s(k), we would:

1) Rotate the upper sideband to DC via 

                 y(k) = s(k)*exp(-j 2 pi fc k/fS)

2) Apply a low-pass filter to chop out the lower sideband (which is now  
centered on -2fc) and you have an basebanded analytic equivalent of your  
input signal. Make that low-pass filter of the decimating FIR type and you  
are probably picking up significant computational efficiencies.

This techniques is approximately a digital equivalent of the quadrature  
sampling approach and is valid whenever quadrature sampling would be  
appropriate. (However, these days a digital implementation takes up less  
space than the analog so for embedded systems, the digital is generally  
preferable.)

Hope this is of some use,

Mark.
--
Mark Kotanchek
Guidance & Control Dept - 363 ASB
Applied Research Lab/Penn State
P.O. Box 30
State College, PA 16804

Newsgroups: comp.dsp
From: tim@ear-ache.mit.edu (Tim Wilson)
Subject: Re: Imaginary signals
Organization: Massachusetts Institute of Technology
Date: Sun, 10 Jan 1993 17:44:35 GMT

880840m@axe.acadiau.ca (MICHAEL ALEXANDER MCKAY) writes:

   Is there a time-domain filter which, given a real signal as input, will
   output the corresponding imaginary signal?
   I've tried to make a filter with unity gain and pi/2 phase shift for
   all frequencies, but with no luck.

See Oppenheim and Shafer, _Digital Signal Processing_ (1st edition),
Section 7.4:  "Hilbert Transform Relations for Complex Sequences."  In
particular, see Section 7.4.1, "Design of Hiblert Transformers."
I quote:  "...the ideal Hibert transformer or 90-degree phase shifter
takes its place alongside the ideal lowpass filter and ideal
bandlimited differentiator as valuable theoretical concepts which
correspond to noncausal systems and for which the system function
exists only in a restricted sense."

In the same section, O&S discuss finite-duration approximations of the
ideal Hilbert transformer using frequency-sampling, windowing, and
equiripple approximations.  Finally, they reference B. Gold, A. V.
Oppenheim, and C. M. Rader, "Theory and Implementation of the Discrete
Hibert Transform," _Proc. Symp. Computer Processing in
Communications_, Vol. 19, Polytechnic Press, 1970, New York for a
recursive implementation of a 90-degree phase-splitting system.

Good luck.
--
Tim Wilson
Internet:  tim@ear-ache.mit.edu
UUCP:  mit-eddie!mit-athena!tim
From: cmh@eng.cam.ac.uk (C.M. Hicks)
Newsgroups: comp.dsp
Subject: Re: Pitch shifting
Date: 5 Jan 93 18:45:30 GMT
Organization: cam.eng

mikael@terapin.com (Mikael Andersson) writes:

>Was ther any results on the Pitch-shifting discussion in this conference?
>Are there any shortcuts or do i have to use FFT to change it? (And still
>keep the quality?)

You'll have problems keeping the quality with any method, since pitch-shift
is an awkward, very non-linear operation which can only really be
approximated to. There are three methods that I know of, the first of which
is the most common:

 - circular buffer, with different read-in and read-out rates. Try a buffer
   of 10 to 50ms in length.
   Main problems: discontinuities when the input and output pointers cross.

 - FFT, scale frequency, IFFT.
   Main problems: discontinuities at block boundaries, computational load.

 - Signal Modelling approach. Model the signal (eg linear prediction) and
   adjust the model parameters so that when the signal is reconstructed, it
   is at the changed pitch.
   Main problems: model choice, block boundaries, computational load.

If you want to try this, I imagine you will get best results with the first
method, though the second might be worth a go. I have no personal experience
with the third, but I believe it has been used successfully on less than 
telephone-quality speech. I doubt it would be practical for music.

> 
>Mikael

Christopher Hicks
--
 ==============================================================================
 Christopher Hicks    |    If it doesn't fit...
 cmh@uk.ac.cam.eng    |                        ...you need a bigger hammer.
 ==============================================================================
From: dpkemp@afterlife.ncsc.mil (David P. Kemp)
Newsgroups: comp.dsp
Subject: Re: Format of .AU Sound files
Date: 5 Jan 93 16:49:19 GMT
Organization: The Great Beyond

In article <1993Jan4.181259.27617@linus.mitre.org> m22257@mitre.org (Jeff Correia) writes:

>Furthermore, in the sounds given in the /usr/demo/SOUND/sounds directory...
>1.  At what sampling rate were these sounds sampled?
>2.  Is there any header information in these files and if there is, what is
>it's format?

Please do at least a *little* RTFM'ing before posting questions like
this.

HINT: look in /usr/demo/SOUND/man.
-- 
   Dave Kemp   dpkemp@afterlife.ncsc.mil 
-----------------------------------------------------------------------------
"Washington [D.C.] is a city of Southern efficiency and Northern charm"
     -- John F. Kennedy
Newsgroups: comp.dsp
From: ieecjk@ccunix.ccu.edu.tw (Dr. C.J. Kuo)
Subject: EE faculty position in Taiwan
Organization: National Chung Cheng University, Taiwan, R.O.C.
X-Newsreader: TIN [version 1.1 PL6]
Date: Mon, 11 Jan 1993 10:41:19 GMT


Department of Electrical Engineering
National Chung Cheng University
Chiayi, Taiwan, R.O.C.


    The Department of Electrical Engineering, National Chung Cheng
University, Chiayi, Taiwan, ROC invites applications for faculty
positions at the Associate Professor and Professor levels.  Appointment
will be in the following areas: 

    1. Electrical machinery
    2. IC design (prefer experiences with communication ICs)
    3. Signal processing
    4. Computer architecture
    5. Intergrated optics
    6. Medical imaging
    7. Artificial intelligence
    8. Communication networks (prefer experiences with ATM)


    Only four vacancies are open and will be filled with four strongest
candidates submitting applications.  The candidate must have a Ph.D.
degree in a relevant discipline from a accredited university and must
have demonstrated academic potentials.  Faculty at NCCU are expected to
teach both undergraduate and graduate courses and conduct research.
NCCU is located in the southern part of Taiwan and have a spacious
(total area: 133 hectares) and scenic campus.  We are young and growing.
Currently we offer B.S. and M.S. degrees.  Starting with Fall Semester
1993, we will also offer Ph.D. programs. 


    Both accomplished scholar and new Ph.D. are welcome to apply.
Please send your resume, a 500 word autobiography including your goals
at NCCU, indications of two subjects, one each for undergraduate and
graduate curricula, that you wish to teach, and three references to 

    Dr. Thomas T. Fang, Head
    Department of Electrical Engineering
    National Chung Cheng University
    Chiayi, Taiwan 62107
    R.O.C.

Applications received by Mar. 1, 1993 will be screened by Mar. 5 for
the first list of finalists.  Applications received after Mar. 1 will
be considered if not enough strong candidates have filed applications by
then.
From: gints@prophet.esd.sgi.com (Gints Klimanis)
Newsgroups: comp.dsp
Subject: Re: is ZOLA still around?
Date: 11 Jan 93 19:52:50 GMT
Organization: Silicon Graphics, Inc.
NNTP-Posting-Host: prophet.esd.sgi.com

In article <1993Jan11.032158.3772@cactus.org>, thompson@cactus.org
(Charles Thompson) writes:
|> 
|> 
|> Does anyone know if ZOLA is still around? They made a DSP
|> filter/design program for the MAC that runs under MPW.
|> Do they make anything for the PC? it would be wise....

Yes.  Lester Langley is still cranking on DSP Designer.  

ZOLA@AppleLink.Apple.COM (Zola Tech, Lester A Longley,PAS)
Newsgroups: comp.dsp
From: kambiz@thecube.tcs.com (Kambiz Homayounfar)
Subject: Jobs Offered
Keywords: Speech Coding, DSP
Organization: Teknekron Communications Inc.
Date: Thu, 7 Jan 1993 19:27:45 GMT

Positions Available 

Teknekron Communications Systems Inc. (Berkeley, CA) is a technology  
development and licensing company with a staff of 180 most of whom are  
engineers and scientists.  We work on a variety of  R&D projects in areas  
of wireless communications (cellular, cordles, PCS) and telecommunications  
software (network management systems, operational contol).  
 
Senior Scientist, Speech Coding
Requires MS or Ph.D in electrical engineering with minimum of 10 years  
experience in design and implementation of speech compression algorithms  
for telecommunications and consumer electronics products. 
Responsibilities include:
* Formulation of novel speech compression algorithms at bit rates below 16  
kbps
* High-Level (using C-language) and CAD-based (COSSAP, WAVES) simulation  
of speech coding algorithms
* Objective and subjective evaluation of low bit-rate speech coders
* Real-time implementation of speech coding algorithms on general-purpose  
DSP chips (TI TMS family, AT&T DSP1600) 
* Design of processor architectures for custom implementation of speech  
compression algorithms
* Analysis of standard (CCITT, TIA, ETSI) speech coding algorithms for  
low-power implementation on custom DSP architectures

DSP Application Engineer
Requires BS/MS in Electrical Engineering with minimum of 5 years of  
product development experience in wireless or consumer electronics  
industries. Sales engineering experience and knowledge of Japanese  
language highly desirable. 

Responsibilities include:
* Assembly-level DSP software development
* Design and construction of DSP-based hardware prototypes
* Consult with customers' engineers to assure compliance with customers's  
needs and specifications

Contact: 
Kambiz Homayounfar
Senior Manager, Wireless Communications
Teknekron Communications Systems Inc.
2121 Allston Way
Berkeley, CA 94704
Fax # (510) 848 8851
Internet: kambiz@tcsi.com


Xref: sparky comp.dsp:2934 comp.speech:409 rec.audio.pro:2705
Organization: University of Illinois at Chicago
Date: Thu, 7 Jan 1993 03:38:01 CST
From: <U28709@uicvm.uic.edu>
Newsgroups: comp.dsp,comp.speech,rec.audio.pro
Subject: NEW TRUE digital audio capture

I have developed a PC-card that captures audio data comforming to the
AES/EBU and S/PDIF format. This output is available from home and
professional CD and DAT players.
I am thinking about marketing this card (I call it DEar), but I am not
sure there is enough interest to justify the effort.  If you are
interested, please send me E-Mail with the following information:

1) How would you use it?
2) Software you already own you would like the card to work with, and
why.
3) Hardware you already own you would like the card to work with, and
why.
4) Software you would like to accompany the card and why (include
software you might have written)
5) How would you use an optional digital output port?
6) Which would be the best way to advertise and distribute this card?
7) How many people do you know you think would be interested in the same
way you are?

Thanks for your time.  Any additional comments are welcomed.
Andy Rosa.
Newsgroups: comp.dsp
From: m22257@mitre.org (Jeff Correia)
Subject: Re: Format of .AU Sound files
Organization: The MITRE Corporation
Date: Mon, 4 Jan 1993 18:12:59 GMT

In article <1992Dec30.201851.45991@kuhub.cc.ukans.edu>,
christos@kuhub.cc.ukans.edu writes:
> 
> Hello there,
> 
> 	I am using the Public Domain program Sound Tools on the Sun station to record
> speech. The speech samples are saved in a mu-law 8bits/sample encoding rule with a 
> standard ".au" header precceding the data. What I would like to know is what the 
> format of the speech samples is? In other words, I am reading the bytes that 
> correspond to 8-bit speech samples but I don't know whether they are in signed
> binary format(2's complement) or unsigned binary format or anyother format. I have
> tried both binary signed (2' complement) and unsigned but it doesn't work. Does
> anybody know how the samples are stored?
> 
> 	I would appreciate any relevant response.
> 
> 
Furthermore, in the sounds given in the /usr/demo/SOUND/sounds directory...
1.  At what sampling rate were these sounds sampled?
2.  Is there any header information in these files and if there is, what is
it's format?

Jeff
Newsgroups: comp.dsp
From: rcogl@catt.citri.edu.au (Geoff Lambourne)
Subject: C source code for LPC-10 analysis/synthsis
Reply-To: rcogl@catt.citri.edu.au
Organization: CATT Centre at CITRI
Date: Wed, 6 Jan 93 04:57:54 GMT


Is there any source code avaliable for an LPC-10 coding system.  If so could someone post or email me a copy.

I am aware of the avaliablity of CELP US-FS1016 C code, but I am specifically after the simpler LPC-10 coder.

Thanks in advance,
	
	Geoff Lambourne
	rcogl@topcat.catt.citri.edu.au
From: oscar@msen.com (Oscar Linares)
Newsgroups: comp.dsp
Subject: DSP digital image processing
Date: 6 Jan 1993 10:48:29 GMT
Organization: Msen, Inc. -- Ann Arbor, Michigan
NNTP-Posting-Host: garnet.msen.com
X-Newsreader: TIN [version 1.1 PL8]

Hi,

I am interested in using the Motorola DSP 56001 for some digital image processing work
(gray-scale).  I am wondering if anyone out there might know about sample code or packages or
a book or article on how best to start approaching this problem.  Advice would even be better.

Thanks,

Oscar A. Linares, MD, PhD
Laboratory of Computers in Biomedical Applications
(oscar@msen.com)
From: izumi@mindseye.berkeley.edu (Izumi Ohzawa)
Newsgroups: comp.dsp
Subject: Re: Imaginary signals
Date: 10 Jan 1993 02:21:01 GMT
Organization: /etc/organization
NNTP-Posting-Host: mindseye.berkeley.edu

In article <C0LvMG.4B4@ccu.umanitoba.ca> langi@ee.umanitoba.ca writes:
>In article 726611777@axe.acadiau.ca, 880840m@axe.acadiau.ca (MICHAEL ALEXANDER MCKAY) writes:
>>
>>Is there a time-domain filter which, given a real signal as input, will
>>output the corresponding imaginary signal?
>>I've tried to make a filter with unity gain and pi/2 phase shift for
>>all frequencies, but with no luck.
>
>I don't really understand what you mean by 'the corresponding imaginary signal'.


Perhaps, he meant "quadrature signal" ?


-- 
Izumi Ohzawa             [ $BBg_78^=;(B ]
USMail: University of California, 360 Minor Hall, Berkeley, CA 94720
Telephone: (510) 642-6440             Fax:  (510) 642-3323
Internet: izumi@pinoko.berkeley.edu   (NeXTmail OK)
From: abirenbo@rigel.cel.scg.hac.com (Aaron Birenboim)
Newsgroups: comp.dsp
Subject: CELP ?!?? algorithms... CASE tools???
Date: 11 Jan 93 22:07:24 GMT
Organization: Hughes Aircraft Colorado Engineering Labs


I spoke to a non-technical person who claimed that her company was looking
for a "digital signal processing" person to develop CELP altorithms.
Do you have any idea what a CELP algorithm is?  If I am not familiar
with the acronym... does that imply i am not capable of the job?
(I'm a fast learner ;-)  Could she have ment CASE tool development?
Whats a CASE tool.... i see it all over but i do not know what it is.

am i an idiot for not knowing these acronyms?  i've been out of school and DSP
for for a few years, but i want to go back to DSP real bad!

   thanks in advance for any advice you can offer,
--
Aaron Birenboim         | If a single yeast cell were given an optimal growth
birenb@hac2arpa.hac.com |  media, within two weeks it would cover the entire
W (303) 344-6486        |  surface of the earth with a depth of two feet.....
H (303) 871-8271        |     wouldn't that be nice.    -- Micah Millspaw
Newsgroups: comp.dsp
From: eea78555@twins (Hwan-Rei Lee)
Subject: Correlated test sequence.
Organization: EENCTU
X-Newsreader: Tin 1.1 PL3
Date: Wed, 6 Jan 1993 07:52:55 GMT

Dear Sirs:
	I am now working on an experiment that needs several test sequence
with different correlations between successive sample. My question is
how to randomly generate a sequence that has autocorrlation r(1)=1, 0.9,
0.8 ...? 
	Thanks for helping in advance.

Henry, eea78555@twins.ee.ncut.edu.tw
Lines: 11
Date: Tue,  5 Jan 93 08:17:14 MST
From: metin@gecko.ee.byu.edu (Metin Gunsay)
Newsgroups: comp.dsp
Distribution: world
Organization: Brigham Young University, Provo UT USA
Subject: Dual channel sapling card


I am setting up a DSP lab course and need a card to be placed in a pc or 
mac (preferrable the mac) to sample 2 channels of 16 bit audio simultaneously.
We have about 500 bucks to spend on such a device, no dsp chip is need, only
dual sampling capability. I've been looking around without any luck, so any
clues would greatly be aprecieated.

thanx metin.

-- 
Thanx,    metin@gecko.ee.byu.edu
Newsgroups: comp.dsp
From: DSPWorld@world.std.com (Amnon Aliphas)
Subject: DSP APPLICATIONS CALL FOR EXHIBITORS
Organization: The World Public Access UNIX, Brookline, MA
Date: Thu, 7 Jan 1993 22:10:58 GMT


                        CALL FOR DSP APPLICATIONS

                   TO BE EXHIBITED AT DSP WORLD EXPO.




    International Conference on Signal Processing Applications & Technology


                         featuring DSP World Expo.


         September 28, 1993 - October 1, 1993, San Francisco Bay Area


       APPLICATIONS WILL BE SELECTED TO BE PRESENTED AT DSP WORLD EXPO.

               Application Areas:  Automotive
                                   Communications
                                   Computer Architectures
                                   Consumer Products
                                   DSP Machines
                                   DSP Software
                                   DSP Technology
                                   Geophysics
                                   Image Processing
                                   Industrial Control
                                   Instrumentation and Testing
                                   Medical Electronics
                                   Military Electronics
                                   Multimedia
                                   Navigation
                                   Neural Networks
                                   Radar
                                   Telephony
                                   Underwater
                                   Video
                                   VLSI Architectures
                                   
                                   and other applications


 Mail, Fax, e_mail a 400 word abstract for review, no later than April 30,1993.

 to:             DSP Associates
                 18 Peregrine Road
                 Newton, MA 02159

                 Tel: (617) 964-3817 or (617) 964-3195
                 Fax: (617) 969-6689

                 e_mail:  DSPWorld@world.std.com

______________________________________________________________________________

   DSP Associates -- Electronic Engineering Times -- Electronic World News

______________________________________________________________________________
From: Wilfred.Hansen@CS.CMU.EDU
Newsgroups: comp.soft-sys.andrew
Subject: Re: Tables containing rasters
Date: 8 Jan 93 16:20:50 GMT
Distribution: world
Organization: The Internet

The "official" answer is that rasters don't print in tables.  

Sigh.

It is pleasant to learn that it does work under groff.

The inability of printing rasters in tables has long been a deficiency
of AUIS, but it has not been easy to see how to fix the problem within
the current printing technology.  Perhaps printing technology is 
something we should discuss at the annual meeting.

Fred Hansen
Andrew Consortium, Carnegie Mellon
(412) 268-6788   wjh+@andrew.cmu.edu