HDTenEightyP/NetNews huggingface dataset:
From: rdippold@qualcom.qualcomm.com (Ron Dippold)
Newsgroups: comp.dsp,comp.compression.research
Subject: Re: Looking for telephone quality audio compression
Date: 9 Sep 92 22:31:35 GMT
Organization: Qualcomm, Inc., San Diego, CA

ja51359@uxa.cso.uiuc.edu (axelrod) writes:
>	I'm looking for a audio compression algorithm that will result
>in telephone quality reproduction.  I.E.  4Khz bandwidth, limited dynamic
>range, average S/N ratio.
>	I'm already familiar with using delta-fibonacci, delta-huffman
>techniques, but I'm looking for a more lossy algorithm that will give
>better compression results, more like on the order of 8:1 with 8-bit
>samples.
>	How is the quality of CELP compression?  I heard voices end up
>sounding robotic.  I'd like something that sounds natural.

Our version of CELP, QCELP, sounds quite decent.  If things aren't
tuned just right, voices can get a "sharpness" to them.  To my ear it
sounds superior to standard telephone, and those I've called have been
unable to tell when I'm calling with the desk phone and with the
cellular phone unless we introduced plenty of noise into the system
(at which time the voice starts sounding somewhat "bubbly" as the
noise overwhelms our error correction).

We output 192 bits per 20 millisecond frame which works out to 1200
bytes per second, or 4.3 megabytes per hour of speech.  In addition,
we do voice activity detection and can produce half, quarter, and
eighth rate frames.  The voice activity factor of standard speech
works out to about 0.6 with this method, which means that the
resulting data is only 60% of the size of that where we force it to
stay in full rate mode, which gives about 2.6 megs for an hour of
speech, or 720 bytes per second.

Given your sampling rate of 8000 Hz with 8 bit samples, that would be
28.8 megs for an hour of speech, so we're doing around 7:1 without
even bothering with voice activity, about 11:1 with it, including
error correction.

We're doing all this in an ASIC, but it demonstrates that it's
possible to get what you want with a version of CELP.  At least it
might be worth looking into.
-- 
I never made a mistake in my life.  I thought I did once, but I was wrong.
Newsgroups: comp.dsp,comp.compression.research
From: williamb@ee.ubc.ca (william burchill)
Subject: Re: Looking for telephone quality audio compression
Organization: University of BC, Electrical Engineering
Date: Wed, 9 Sep 1992 20:50:31 GMT

In article <BuBn7u.BFu@news.cso.uiuc.edu> ja51359@uxa.cso.uiuc.edu (axelrod) writes:
>	I'm looking for a audio compression algorithm that will result
>in telephone quality reproduction.  I.E.  4Khz bandwidth, limited dynamic
>range, average S/N ratio.
>	I'm already familiar with using delta-fibonacci, delta-huffman
>techniques, but I'm looking for a more lossy algorithm that will give
>better compression results, more like on the order of 8:1 with 8-bit
>samples.
>	How is the quality of CELP compression?  I heard voices end up
>sounding robotic.  I'd like something that sounds natural.
>
>Thanks,
>Jeff
>

CELP does not sound robotic above 4 kbits/sec, it's pretty much phone-like.
LPC, on the other hand, is best left to the Daleks on Dr. Who.
 
Newsgroups: comp.dsp
From: murali@iss.nus.sg (Murali)
Subject: Re: TMS320C25: Continuous speech problem
Reply-To: murali@iss.nus.sg
Organization: Institute Of Systems Science, NUS,Singapore.
Date: Wed, 9 Sep 1992 05:39:03 GMT

In article 19631@cc.uow.edu.au, u8801763@cc.uow.edu.au (Radivoj Kouzan) writes:
>u9066624@cc.uow.edu.au (Tippu Hassan) writes:
>
>>Hello anyone,
>
>>My name is Tippu and I live in Australia. I am working with the TI
>>"SWDS" (Software Development Systems) board which is a support tool
>>for the TMS320C25 dsp chip. I am using the TMS320C25 serial port to
>>read in digitized speech samples and then storing them on external
>>RAM memory which is on the SWDS board. The SWDS only has 32K bytes
>>of onboard RAM avialable and therefore I can only store 2 seconds
>>worth of 16-bit digitized speech. So in order to record continous
>>speech I am using the cpu processor to upload these speech samples
>>from the SWDS board RAM and store them on files once the RAM is full. 
>>This is done via C applications written for the cpu. 
>>My problem is that in order for the cpu to access the SWDS RAM once 
>>it is full, it MUST place the TMS320C25 in halt mode, then upload 
>>the speech samples and then unhalt the TMS320C25 to resume reading 
>>in the next load of speech samples. But the cpu cannot execute this 
>>process quite enough, and TMS320C25 loses some speech samples while 
>>it is halted. I cannot uses the grep command on the TMS320C25, which
>>makes the external RAM global, because it requires external circuitry
>>for the TMS320C25 and it is wired onto the SWDS board. Can anyone offer
>>any ideas or help as it would be much appreciated.
>
>
>
>
>Sell it and buy yourself a DSP that works for Digital Signals.



I was once using  a TMS320C20 based add-on board to digitize and process
speech data. We used a double-buffering scheme wherein the DSP chip raise
a semaphore after filling one buffer and continues filling the second buffer,
while the CPU will start reading the data from the first buffer once the semaphore
is raised. This scheme worked pretty well for TMS320c20, which we were able to 
access upto register level from the PC side( not like the systems which hide the 
internals by giving you a set of function calls or a library ! ). I don't know if you've
tried this but this is one of the most commonly used method to get data without
halting the DSP chip.

Murali.

P.S : the semaphore was a software one and we used a memory location accessible 
       by both DSP and PC CPU for that purpose.  I suppose you have that kind of 
       dual-ported memory in your system.
Newsgroups: comp.dsp
From: rayw@ee.ubc.ca (raymond w m woo)
Subject: questions on designing DSP board based on DSP56000
Organization: University of BC, Electrical Engineering
Date: Sun, 6 Sep 1992 00:30:58 GMT

Hi, I am about to design a DSP board based on the Motorola DSP56000 chip.
I have never done any similar design and I am new to this DSP chip. 
Is there any book other than the DSP56000 User's Manual that I can use? 
Also, does anyone have any suggestion on the design (eg, which memory, A/D, D/A
chips to use, etc)?

Any advice would be greatly appreciated!

raymond  <rayw@ee.ubc.ca>
From: adarsh@sybase.com (Adarsh Pannu)
Newsgroups: comp.dsp
Subject: Seeking General Information !
Keywords: info
Date: 11 Sep 92 00:36:45 GMT
Followup-To: poster
Distribution: usa
Organization: Sybase, Inc.

Hi !

One of my friends is starting in the M.S. programme at one of the
universities in US. DSP happens to be one of the courses he could
choose from. While he does have some background on the subject, he
would like to know more about the general aspects like -

	How would a specialization in the subject help him
		professionally ?
	Is DSP more of a research subject, or does it have many
		real-world applications existing ?
	What is the scope of the subject in the future ?
	Other bits of information, not related to the technical
		aspects of the subject ... etc etc.

(Please pardon my ignorance :-))

I would really appreciate any responses on these queries. Email
would be perfect.

Thanks,

Adarsh
adarsh@sybase.com
From: heilerr@hpcc01.corp.hp.com (Richard Heiler)
Date: Wed, 9 Sep 1992 20:51:36 GMT
Subject: Re: Motorola HC16 finally came, now what :-)
Organization: the HP Corporate notes server
Newsgroups: comp.dsp

rick\@hpprsd1.mayfield.hp.com

Hello,

I have the same problem.  I would like to use my SE/30 and a pc simulator
instead of a pc.

I called Motorola and they referred me to the company who wrote the    
programming environment.  They said no way!  They wouldn't even give me
a description of how the i/o worked.  

Good Luck,


Rick Heiler
From: prandolini@qut.edu.au
Newsgroups: comp.dsp
Subject: MATLAB and Speech Processing
Date: 11 Sep 92 12:33:33 EST
Organization: Queensland University of Technology


I am currently evaluating MATLAB for
signal processing and speech processing
applications.  Does anyone have any
experience in using MATLAB for speech
processing?

We are looking for a set of tools to
perform waveform editing, spectral
analysis, cepstrum, LPC, PARCOR, etc.

Obviously MATLAB is well suited to this
field of work (eg Signal Processing Toolbox),
but we do not want to reinvent the wheel.

If you have any advice, please e-mail me on
r.prandolini@qut.edu.au

Kind regards

Robert Prandolini
Centre for Signal Procesing Research
School of Electrical and Electronic Systems Engineering
Queensland University of Technology
Australia
Newsgroups: comp.dsp
From: muller@bismol.pa.dec.com (Eric Muller)
Subject: Re: Motorola HC16 finally came, now what :-)
Organization: DEC Systems Research Center
Date: Tue, 8 Sep 92 18:32:04 GMT

In article <1992Sep8.173831.29537@cas.org>, Vince Herried ask about
the protocol between the HC16 evalutation board and the PC used for
development.  I would be very interested to get this info as well.

Thanks,
Eric.
From: dak@kaa.informatik.rwth-aachen.de (David Kastrup)
Newsgroups: comp.dsp
Subject: Re: Hilbert Transform (?)
Date: 9 Sep 92 15:32:22 GMT
Organization: Rechnerbetrieb Informatik  /  RWTH Aachen

frandsen@seismo.CSS.GOV (Marvin Vaun Frandsen) writes:

>I too am interested in info on the theory and practice of Hilbert
>transforms.  I have seen contexts where the HT is used but would like
>to understand what it does, and why, better.

>--Marv Frandsen   frandsen@beno.css.gov

>-- 
>-------------------------------------------------------------------
>	  Marvin Vaun Frandsen       frandsen@beno.css.gov
>	    
It shifts any frequency components by 90 degree of phase.
So if you have a clean sine signal of arbitrary phase, and add the
square of it to the square of the Hilbert transform, you get a
constant term with the mean energy of your original wave.
Newsgroups: comp.dsp,comp.compression.research
From: indra@earthview.cms.udel.edu (Indra Jaya)
Subject: Wigner Distribution
Keywords: wigner distribution function, time-frequency distribution
Reply-To: indra@earthview.cms.udel.edu
Organization: Marine Studies, University of Delaware
Date: Sat, 12 Sep 1992 16:05:32 GMT

Hello,

Does anybody out there have some info [software or ftp site] on Wigner 
Distribution Function which generate time-frequency distribution for a
digital data set? Any help would be appreciated.

Indra Jaya
indra@earthview.cms.udel.edu


Newsgroups: comp.dsp
From: williamb@ee.ubc.ca (william burchill)
Subject: Re: Hilbert Transform (?)
Organization: University of BC, Electrical Engineering
Date: Wed, 9 Sep 1992 20:43:55 GMT

In article <dak.716052742@kaa> dak@kaa.informatik.rwth-aachen.de (David Kastrup) writes:
>frandsen@seismo.CSS.GOV (Marvin Vaun Frandsen) writes:
>
>>I too am interested in info on the theory and practice of Hilbert
>>transforms.  I have seen contexts where the HT is used but would like
>>to understand what it does, and why, better.
>
>>--Marv Frandsen   frandsen@beno.css.gov
>
>>-- 
>>-------------------------------------------------------------------
>>	  Marvin Vaun Frandsen       frandsen@beno.css.gov
>>	    
>It shifts any frequency components by 90 degree of phase.
>So if you have a clean sine signal of arbitrary phase, and add the
>square of it to the square of the Hilbert transform, you get a
>constant term with the mean energy of your original wave.

They are useful for generating SSB (single side band) signals and
also as a conceptual link between passband signals and their equiv-
alent complex baseband representation.  See "Communication Systems"
by Simon Haykin, readable and aimed at undergrads.

Newsgroups: comp.dsp
From: Bob Zigon <bobz@truevision.com>
Subject: Debuggers/ICE's for AT&T 3210
Organization: Truevision, Inc., Indianapolis, IN
X-Useragent: Nuntius v1.1
Date: Thu, 10 Sep 1992 20:46:28 GMT


   I am looking for 3rd party debuggers and/or In Circuit Emulators for
the AT&T 3210.
Although AT&T has a debugger, it only works under VCOS. I want to
initially
bring up a 3210 system with a tiny kernel (i.e. No VCOS) that manipulates
the registers on the peripherals. Does anyone know of the existence of
such tools?

Bob Zigon
bobz@truevision.com
Newsgroups: comp.dsp
From: choup@bionette.cgrb.orst.edu (Ping Chou)
Subject: Gabor Transform
Organization: Biological Computing Consortium, OSU, Corvallis, OR.
Distribution: usa
Date: Sat, 5 Sep 1992 02:24:11 GMT

        There is an article in Instrumentation Newsletter (vol.4, no.2,
summer 1992, by National Instruments) talking about Gabor transform and its
usefulness in DSP. The STFT is also mentioned and compared and almost beaten
to dead.

        After reading several published papers on Gabor transform, I still
have no clear clue as how to write a program to do the transform. I figure
that Gabor transform must have been used in DSP for years because it was first
proposed in as early as 1946, and sounds good.

        Any pointer to where I can find a such program is appreciated. And I
would be happy to reinvent the wheel if there is reference on how to implement
Gabor transform. Thanks.

Ping
Newsgroups: comp.dsp,comp.compression.research
From: rdippold@qualcom.qualcomm.com (Ron Dippold)
Subject: Re: Looking for telephone quality audio compression
Organization: Qualcomm, Inc., San Diego, CA
Date: Thu, 10 Sep 1992 00:21:21 GMT

I've had some comments from our vocoder team on the actual figures
involved - I was actually underspecifying the product (so much for my
marketing career)!

First, I mentioned that the frames were 192 bits each including error
correction.  Without any of the error correction, etc. crud it's 160
bits, or 1000 bytes per second, so that's 80% of my earlier figure.

Second, I claimed we could cut 40% of the size due to the variable
rates.  It's actually 50%.

That gives 8:1 compression at full rate, 16:1 at variable rate for
standard conversations.

Finally, apparently "voice activity detection" has nasty connotations
of those horrible off/on systems which always seem to be chopping off
parts of your words.  Under our version, we're always doing something,
if only at low rate, which sounds much nicer.  That really has nothing
to do with the actual output, but I didn't want to impugn the great
job our vocoder people have done.  I'll never be in marketing (thank
god!).



-- 
I dress this way to annoy you.
Xref: sparky comp.dsp:2128 comp.compression.research:153
From: ja51359@uxa.cso.uiuc.edu (axelrod)
Newsgroups: comp.dsp,comp.compression.research
Subject: Looking for telephone quality audio compression
Date: 9 Sep 92 17:28:41 GMT
Organization: University of Illinois at Urbana


	I'm looking for a audio compression algorithm that will result
in telephone quality reproduction.  I.E.  4Khz bandwidth, limited dynamic
range, average S/N ratio.
	I'm already familiar with using delta-fibonacci, delta-huffman
techniques, but I'm looking for a more lossy algorithm that will give
better compression results, more like on the order of 8:1 with 8-bit
samples.
	How is the quality of CELP compression?  I heard voices end up
sounding robotic.  I'd like something that sounds natural.

Thanks,
Jeff
From: adrian@csqx.cs.rhbnc.ac.uk (A Johnstone)
Newsgroups: comp.dsp
Subject: What's on the new Microsoft sound board
Date: 15 Sep 92 17:29:29 GMT
Reply-To: adrian@csqx.cs.rhbnc.ac.uk (A Johnstone)
Organization: Dept of Comp Sci, Royal Holloway & Bedford New College Uni London

I hear that Microsoft are about to start selling a sound board 
which I guess might become some kind of standard if they base a whole
load of Windows stuff round it. The note I read said that it has
an Analog Devices chip on it. I'm a big fan of (and user of) the ADSP
2100 series. Does anybody know what this board has on it?


                  Adrian
Newsgroups: comp.dsp
From: rainer@spot.Colorado.EDU (Rainer Malzbender)
Subject: Want to talk to someone using C40's
Organization: University of Colorado, Boulder
Date: Mon, 14 Sep 1992 18:44:10 GMT

I'm considering using C40 TIM-40 modules for parallel supercomputing
and would appreciate hearing from someone who has some experience with
them, particularly relating to ease of use and performance relative to
tranputers or i860's. Thanks.

-- 
Rainer Malzbender
Post-doc, Physics
U. of Colorado, Boulder (303)492-1366           rainer@boulder.colorado.edu
From: xfeng@fees.vut.edu.au (Mr. Xue-Feng Leng)
Newsgroups: comp.dsp
Subject: Re: questions on designing DSP board based on DSP56000
Date: 8 Sep 92 12:36:13 GMT
Organization: Victoria University of Technology, Melbourne, Australia

>Hi, I am about to design a DSP board based on the Motorola DSP56000 chip.
>I have never done any similar design and I am new to this DSP chip. 
>Is there any book other than the DSP56000 User's Manual that I can use? 
>Also, does anyone have any suggestion on the design (eg, which memory, A/D, D/A
>chips to use, etc)?I recently made two channel A/D borad for my DSP56000 board. The original one only has one A/D channel. The A/D chip I used is DSP56ADC16 you can get from market. This is 16 bits resolution and maximum sampling rate is 100kHz. If you want higher sampling rate up to 400Hz you only can get 12bits resolutions. This chip is very simple to use it doesn't need any glue circuits.
Hope this helps. 
Newsgroups: comp.dsp
From: vjh21@cas.org (Vince Herried ext 2877)
Subject: Motorola HC16 finally came, now what :-)
Organization: Chemical Abstracts Service
Date: Tue, 8 Sep 1992 17:38:31 GMT

The MCU chip has a DS0, DS1, and some kind of clock line that goes to a PAL
chip.  The PAL converts the serial data into a parallel I/O line.
The book says ,plug in yer 25 wire cable between yer IBM pc and
the HC16 Evaluation board and run....

I don't have an IBM pee cee.  I've got a Commodore Amiga which uses
a MOTOROLA 68030 micro.  I think I can come up with an assembler, or
at least get my files assembled.  I would like to talk to the evaluation
board via my Amiga.

Here is the question......
Does anyone know the parallel port protocol used?
The PAL appears to convert the 17 bit command words as documented by the
CPU16 manual into a 'centronics' parallel port format.  The PAL connects
to 7 of the 8 data lines (The high order line is Not Connected?) the
Ack line, and the busy line.  The question is,  
are all 7 bits used, taking turns sending and receiving?
Are some used to send and others used to receive?
How is the handshaking done?  Do I gotta hook a logic probe just to
find out how the data gets to and fro?

I'm ready to start writting code on my Amiga but need some protocol first.

Email would be fine.
-- 
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
				 |     Vincent Herried  (614) 447-3600 x 2877
				 |     Chemical Abstracts Service
 _____________________           |     P.O. Box 3012
(_)________________   \          |     Columbus, OH 43210-0012
  ________________|\   \         |
 (_)______________\_\   \        |     Current thought: People who cannot be
   ______________________\       |     understood are geniuses, so management
  (_)____________________|       |     assigns them the toughest projects.
				 |
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
From: cew6@po.CWRU.Edu (Carlin E. Wiegner)
Newsgroups: comp.dsp
Subject: DSP for use in teleconferencing?
Date: 14 Sep 1992 21:09:57 GMT
Organization: Case Western Reserve University, Cleveland, Ohio (USA)
NNTP-Posting-Host: thor.ins.cwru.edu


I'm thinking about writing a program that lets you voice 
communicate between two equipped unix boxes and I was wondering
if DSPs make good compressors? I don't know much about them but
I'm trying to decrease the bandwidth needed to transmit the
voice (eventually the picture too). I'm thinking about using
a NeXT and was wondering if the Motorola DSP56001 is a good
one to learn assembler for? Thanks for the replies via email...
CW
From: lels@unpcs1.cs.unp.ac.za (Leonard Els)
Newsgroups: comp.dsp
Subject: Speech Recognition on a PC
Date: Tue, 8 Sep 1992 07:48:12 GMT
Organization: Dept. Computer Science, UNP
Distribution: world
NNTP-Posting-Host: pc013.cc.unp.ac.za
Keywords: speech recogntion


Hi,

Is there any thesis, or likes, which decsribes the code used for
speech recognition on a PC using a soundcard of some sort.

Or better still, are there any ftp sites with such code.
I am looking for a simple digit recognition system

Any pointers?


Leonard Els
E-mail : lels@unpcs1.cs.unp.ac.za
         els@unpsun1.cc.unp.ac.za
Newsgroups: comp.dsp
From: wiegand@rtsg.mot.com (Robert Wiegand)
Subject: Re: DSP56000 memory
Reply-To: motcid!wiegand@uunet.uu.net
Organization: Motorola Inc., Cellular Infrastructure Group
Date: Fri, 4 Sep 1992 20:41:22 GMT

stettbac@isibee5 (Juerg Stettbacher) writes:


>Hello world

>The DSP56001 has 3 memory spaces (X, Y and P).
>X and Y are data memory spaces. Their sizes depend on
>the particular application. P holds the program.

>Question: What's a reasonable size for program memory P.
>(Common DSP applications with  monitor for stand alone.
>Software development in C and assembler).

>Any suggestions?


>Thanks
>juerg at ethz switzerland

The program size will also depend on your application.
I would suggest 8K x 24 as a minimum size for program memory.
8K x 8 chips are common and not too expensive. 3 parts will give you
the 8K x 24 program memory.
I also believe Motorola makes a 8K x 24 part.

If your application will be large you could go to 32K x 24 by using
three 32K x 8 parts. This is somewhat more expensive, but still not
too bad.

It is also possible to use the same parts for X, Y and P memory.
You need a fast address decoder to do this, but it saves on memory
chips.


-- 
------------------------------------------------------------------------------
Robert Wiegand - Motorola Inc.
uunet!motcid!wiegand
Disclamer: I didn't do it - I was somewhere else at the time.
Xref: sparky comp.dsp:2150 comp.compression.research:169
From: antsu@polaris.utu.fi (Antti-Pekka Virtanen)
Subject: Re: 56000 C compiler
Organization: University of Turku, Computing Centre
Date: 9 Sep 92 14:23:00


>   Annotate is a Motorola program that reads the debug directives in the 
>   assembly language file, and uses them to merge the C source into the 
>   assembly language file as comments. It isn't free software.
>
>	   Sorry,
>
>	   Jeff Enderwick
>	   jeff@dsp.sps.mot.com

Does someone have the GDB sources (which didn't come with the Gcc sources)
for DSP56000 ?

(BTW the Motorola Gcc source is available in nic.funet.fi 
~pub/ham/dsp/dsp56k-tools/dsp56k-gcc.tar.Z if someone is interested )

Regards, Antti-Pekka	<antsu@nic.funet.fi>

--
 **************************************************************
 * Mail: Antti-Pekka Virtanen  E-Mail:   antsu@utu.fi         *
 *       Arholankuja 3 as 21             antsu@mea.utu.fi     *
 *       SF-21100 Naantali               antsu@nic.funet.fi   *
 *       Finland                         antsu@sandra.pp.fi   *
 **************************************************************
Newsgroups: comp.dsp
From: sippola@tnclus.tele.nokia.fi
Subject: Companding with DSP, info needed
Organization: Nokia Telecommunications.
Date: Mon, 14 Sep 1992 09:43:00 GMT


        Hello,

        I'm looking for material about digitally implemented 
        compressor/expandor. Could anybody help me?

        At  cellular radio, dynamics of the signal is compressed 2:1 before
        sending  it  to  radio,  and  after  received  from  radio signal is
        expanded  1:2.  Nowadays we have analog implementation, but I'm
        making studies to change implementation suitable for TMS320C16.
   ------------------------------------------------------------------------
       In Real Life: Jaakko Sippola    Phone: + 358 31 2407442       
       sippola@tnclus.tele.nokia.fi    Fax:   + 358 31 2407222
                Organization: Nokia Telecommunications      
   ------------------------------------------------------------------------

        
Newsgroups: comp.dsp