QRZ! Ham Radio 1/QRZ Ham Radio Callsign Database - December 1993.iso/arrl/faqaudio.2

Text File | 1993-11-21 | 48KB | 1,205 lines

Newsgroups: alt.binaries.sounds.misc,alt.binaries.sounds.d,comp.dsp,news.answer
s,comp.answers
From: [e-mail] (Guido van Rossum)
Subject: FAQ: Audio File Formats (part 2 of 2)
Followup-To: alt.binaries.sounds.d,comp.dsp
Organization: CWI, Amsterdam
Date: Mon, 27 Sep 1993 09:25:17 GMT
Lines: 1190

Archive-name: audio-fmts/part2
Submitted-by: Guido van Rossum <[e-mail]>
Version: 3.05
Last-modified: 27-Sep-1993

Appendices
==========

Here are some more detailed pieces of info that I received by e-mail.
They are reproduced here virtually without much editing.

Table of contents
-----------------

FTP access for non-internet sites
AIFF Format (Audio IFF)
The NeXT/Sun audio file format
IFF/8SVX Format
Playing sound on a PC
The EA-IFF-85 documentation
US Federal Standard 1016 availability
Creative Voice (VOC) file format
RIFF WAVE (.WAV) file format
U-LAW and A-LAW definitions
AVR File Format
The Amiga MOD Format

------------------------------------------------------------------------
FTP access for non-internet sites
---------------------------------

>From the sci.space FAQ:

    Sites not connected to the Internet cannot use FTP directly, but
    there are a few automated FTP servers which operate via email.
    Send mail containing only the word HELP to [e-mail]
    or [e-mail], and the servers will send you
    instructions on how to make requests.  (The bitftp service is no
    longer available through UUCP gateways due to complaints about
    overuse :-( )

Also:

    FAQ lists are available by anonymous FTP from rftm.mit.edu
    and by email from [e-mail] (send a message
    containing "help" for instructions about the mail server).


------------------------------------------------------------------------
AIFF Format (Audio IFF) and AIFC
--------------------------------

This format was developed by Apple for storing high-quality sampled
sound and musical instrument info; it is also used by SGI and several
professional audio packages (sorry, I know no names).  An extension,
called AIFC or AIFF-C, supports compression (see the last item below).

I've made a BinHex'ed MacWrite version of the AIFF spec (no idea if
it's the same text as mentioned below) available by anonymous ftp from
ftp.cwi.nl [192.16.184.180]; the file is /pub/audio/AudioIFF1.2.hqx.
A newer version is also available: /pub/audio/AudioIFF1.3.hqx.
But you may be better off with the AIFF-C specs, see below.

Mike Brindley ([e-mail]) writes:

"The complete AIFF spec by Steve Milne, Matt Deatherage (Apple) is
available in 'AMIGA ROM Kernal Reference Manual: Devices (3rd Edition)'
1991 by Commodore-Amiga, Inc.; Addison-Wesley Publishing Co.;
ISBN 0-201-56775-X, starting on page 435 (this edition has a charcoal
grey cover).  It is available in most bookstores, and soon in many
good librairies."

According to Mark Callow ([e-mail]):

A PostScript version of the AIFF-C specification is available via
anonymous ftp on FTP.SGI.COM (192.48.153.1) as /sgi/aiff-c.9.26.91.ps.

Benjamin Denckla <[e-mail]> writes:

A piece of information that may be of some use to people who want to use
AIFF files with their Macintosh Think C programs:  AIFF data structures are
contained in the file AIFF.h in the "Apple #Includes" folder that comes
on the distribution disks.  I found this out a little too late: I had
already coded my own structures.  I assume that this header file comes
with Apple programming products like MPW [C|C++] as well.

An important file format for the Mac which is only mentioned once in the
FAQ is the Sound Designer II file format.  There is also an older Sound
Designer I format.  I have the SDII format in electronic form but I don't
think I'm at liberty to distribute it.  It can be obtained by applying to
become a 3rd Party Developer for Digidesign.  This process is simple
(1-page application) and free.  Call Digidesign at 415-688-0600 for
information.  The SDII file format is interesting in that all non-sample
data (sample rate, channels, etc.) is contained in the resource fork and
the data fork contains sample data only.

------------------------------------------------------------------------
The NeXT/Sun audio file format
------------------------------

Here's the complete story on the file format, from the NeXT
documentation.  (Note that the "magic" number is ((int)0x2e736e64),
which equals ".snd".)  Also, at the end, I've added a litte document
that someone posted to the net a couple of years ago, that describes
the format in a bit-by-bit fashion rather than from C.

I received this from Doug Keislar, NeXT Computer.  This is also the
Sun format, except that Sun doesn't recognize as many format codes.  I
added the numeric codes to the table of formats and sorted it.


SNDSoundStruct:  How a NeXT Computer Represents Sound

The NeXT sound software defines the SNDSoundStruct structure to
represent sound.  This structure defines the soundfile and Mach-O
sound segment formats and the sound pasteboard type.  It's also used
to describe sounds in Interface Builder.  In addition, each instance
of the Sound Kit's Sound class encapsulates a SNDSoundStruct and
provides methods to access and modify its attributes.

Basic sound operations, such as playing, recording, and cut-and-paste
editing, are most easily performed by a Sound object.  In many cases,
the Sound Kit obviates the need for in-depth understanding of the
SNDSoundStruct architecture.  For example, if you simply want to
incorporate sound effects into an application, or to provide a simple
graphic sound editor (such as the one in the Mail application), you
needn't be aware of the details of the SNDSoundStruct.  However, if
you want to closely examine or manipulate sound data you should be
familiar with this structure.

The SNDSoundStruct contains a header, information that describes the
attributes of a sound, followed by the data (usually samples) that
represents the sound.  The structure is defined (in
sound/soundstruct.h) as:

typedef struct {
    int magic;               /* magic number SND_MAGIC */
    int dataLocation;        /* offset or pointer to the data */
    int dataSize;            /* number of bytes of data */
    int dataFormat;          /* the data format code */
    int samplingRate;        /* the sampling rate */
    int channelCount;        /* the number of channels */
    char info[4];            /* optional text information */
} SNDSoundStruct;




SNDSoundStruct Fields



magic

magic is a magic number that's used to identify the structure as a
SNDSoundStruct.  Keep in mind that the structure also defines the
soundfile and Mach-O sound segment formats, so the magic number is
also used to identify these entities as containing a sound.





dataLocation

It was mentioned above that the SNDSoundStruct contains a header
followed by sound data.  In reality, the structure only contains the
header; the data itself is external to, although usually contiguous
with, the structure.  (Nonetheless, it's often useful to speak of the
SNDSoundStruct as the header and the data.)  dataLocation is used to
point to the data.  Usually, this value is an offset (in bytes) from
the beginning of the SNDSoundStruct to the first byte of sound data.
The data, in this case, immediately follows the structure, so
dataLocation can also be thought of as the size of the structure's
header.  The other use of dataLocation, as an address that locates
data that isn't contiguous with the structure, is described in
"Format Codes," below.





dataSize, dataFormat, samplingRate, and channelCount

These fields describe the sound data.

dataSize is its size in bytes (not including the size of the
SNDSoundStruct).

dataFormat is a code that identifies the type of sound.  For sampled
sounds, this is the quantization format.  However, the data can also
be instructions for synthesizing a sound on the DSP.  The codes are
listed and explained in "Format Codes," below.

samplingRate is the sampling rate (if the data is samples).  Three
sampling rates, represented as integer constants, are supported by
the hardware:

Constant        Sampling Rate (samples/sec)

SND_RATE_CODEC  8012.821        (CODEC input)
SND_RATE_LOW    22050.0 (low sampling rate output)
SND_RATE_HIGH   44100.0 (high sampling rate output)

channelCount is the number of channels of sampled sound.





info

info is a NULL-terminated string that you can supply to provide a
textual description of the sound.  The size of the info field is set
when the structure is created and thereafter can't be enlarged.  It's
at least four bytes long (even if it's unused).





Format Codes

A sound's format is represented as a positive 32-bit integer.  NeXT
reserves the integers 0 through 255; you can define your own format
and represent it with an integer greater than 255.  Most of the
formats defined by NeXT describe the amplitude quantization of
sampled sound data:

Value   Code    Format

0       SND_FORMAT_UNSPECIFIED  unspecified format
1       SND_FORMAT_MULAW_8      8-bit mu-law samples
2       SND_FORMAT_LINEAR_8     8-bit linear samples
3       SND_FORMAT_LINEAR_16    16-bit linear samples
4       SND_FORMAT_LINEAR_24    24-bit linear samples
5       SND_FORMAT_LINEAR_32    32-bit linear samples
6       SND_FORMAT_FLOAT        floating-point samples
7       SND_FORMAT_DOUBLE       double-precision float samples
8       SND_FORMAT_INDIRECT     fragmented sampled data
9       SND_FORMAT_NESTED       ?
10      SND_FORMAT_DSP_CORE     DSP program
11      SND_FORMAT_DSP_DATA_8   8-bit fixed-point samples
12      SND_FORMAT_DSP_DATA_16  16-bit fixed-point samples
13      SND_FORMAT_DSP_DATA_24  24-bit fixed-point samples
14      SND_FORMAT_DSP_DATA_32  32-bit fixed-point samples
15      ?
16      SND_FORMAT_DISPLAY      non-audio display data
17      SND_FORMAT_MULAW_SQUELCH        ?
18      SND_FORMAT_EMPHASIZED   16-bit linear with emphasis
19      SND_FORMAT_COMPRESSED   16-bit linear with compression
20      SND_FORMAT_COMPRESSED_EMPHASIZED        A combination of the two above
21      SND_FORMAT_DSP_COMMANDS Music Kit DSP commands
22      SND_FORMAT_DSP_COMMANDS_SAMPLES         ?
[Some new ones supported by Sun.  This is all I currently know. --GvR]
23      SND_FORMAT_ADPCM_G721
24      SND_FORMAT_ADPCM_G722
25      SND_FORMAT_ADPCM_G723_3
26      SND_FORMAT_ADPCM_G723_5
27      SND_FORMAT_ALAW_8


Most formats identify different sizes and types of
sampled data.  Some deserve special note:


--      SND_FORMAT_DSP_CORE format contains data that represents a
loadable DSP core program.  Sounds in this format are required by the
SNDBootDSP() and SNDRunDSP() functions.  You create a
SND_FORMAT_DSP_CORE sound by reading a DSP load file (extension
".lod") with the SNDReadDSPfile() function.

--      SND_FORMAT_DSP_COMMANDS is used to distinguish sounds that
contain DSP commands created by the Music Kit.  Sounds in this format
can only be created through the Music Kit's Orchestra class, but can
be played back through the SNDStartPlaying() function.

--      SND_FORMAT_DISPLAY format is used by the Sound Kit's
SoundView class.  Such sounds can't be played.


--      SND_FORMAT_INDIRECT indicates data that has become
fragmented, as described in a separate section, below.


--      SND_FORMAT_UNSPECIFIED is used for unrecognized formats.





Fragmented Sound Data

Sound data is usually stored in a contiguous block of memory.
However, when sampled sound data is edited (such that a portion of
the sound is deleted or a portion inserted), the data may become
discontiguous, or fragmented.  Each fragment of data is given its own
SNDSoundStruct header; thus, each fragment becomes a separate
SNDSoundStruct structure.  The addresses of these new structures are
collected into a contiguous, NULL-terminated block; the dataLocation
field of the original SNDSoundStruct is set to the address of this
block, while the original format, sampling rate, and channel count
are copied into the new SNDSoundStructs.


Fragmentation serves one purpose:  It avoids the high cost of moving
data when the sound is edited.  Playback of a fragmented sound is
transparent-you never need to know whether the sound is fragmented
before playing it.  However, playback of a heavily fragmented sound
is less efficient than that of a contiguous sound.  The
SNDCompactSamples() C function can be used to compact fragmented
sound data.

Sampled sound data is naturally unfragmented.  A sound that's freshly
recorded or retrieved from a soundfile, the Mach-O segment, or the
pasteboard won't be fragmented.  Keep in mind that only sampled data
can become fragmented.



_________________________
>From mentor.cc.purdue.edu!purdue!decwrl!ucbvax!ziploc!eps Wed Apr  4
23:56:23 EST 1990
Article 5779 of comp.sys.next:
Path: mentor.cc.purdue.edu!purdue!decwrl!ucbvax!ziploc!eps
>From: [e-mail] (Eric P. Scott)
Newsgroups: comp.sys.next
Subject: Re: Format of NeXT sndfile headers?
Message-ID: <[e-mail]>
Date: 31 Mar 90 21:36:17 GMT
References: <[e-mail]>
Reply-To: [e-mail] (Eric P. Scott)
Organization: San Francisco State University
Lines: 42

In article <[e-mail]>
        [e-mail] (Brian Kendig) writes:
>I'd like to take a program I have that converts Macintosh sound
files
>to NeXT sndfiles and polish it up a bit to go the other direction as
>well.

Two people have already submitted programs that do this
(Christopher Lane and Robert Hood); check the various
NeXT archive sites.

>       Could someone please give me the format of a NeXT sndfile
>header?

"big-endian"
        0       1       2       3
        +-------+-------+-------+-------+
0       | 0x2e  | 0x73  | 0x6e  | 0x64  |       "magic" number
        +-------+-------+-------+-------+
4       |                               |       data location
        +-------+-------+-------+-------+
8       |                               |       data size
        +-------+-------+-------+-------+
12      |                               |       data format (enum)
        +-------+-------+-------+-------+
16      |                               |       sampling rate (int)
        +-------+-------+-------+-------+
20      |                               |       channel count
        +-------+-------+-------+-------+
24      |       |       |       |       |       (optional) info
string

28 = minimum value for data location

data format values can be found in /usr/include/sound/soundstruct.h

Most common combinations:

         sampling  channel    data
             rate    count  format
voice file   8012        1       1 =  8-bit mu-law
system beep 22050        2       3 = 16-bit linear
CD-quality  44100        2       3 = 16-bit linear

------------------------------------------------------------------------
IFF/8SVX Format
---------------

Newsgroups: alt.binaries.sounds.d,alt.sex.sounds
Subject: Format of the IFF header (Amiga sounds)
Message-ID: <[e-mail]>
From: [e-mail] (Joe Smith)
Date: 23 Oct 91 23:54:38 GMT
Followup-To: alt.binaries.sounds.d
Organization: BT North America (Tymnet)

The first 12 bytes of an IFF file are used to distinguish between an Amiga
picture (FORM-ILBM), an Amiga sound sample (FORM-8SVX), or other file
conforming to the IFF specification.  The middle 4 bytes is the count of
bytes that follow the "FORM" and byte count longwords.  (Numbers are stored
in M68000 form, high order byte first.)

                ------------------------------------------

FutureSound audio file, 15000 samples at 10.000KHz, file is 15048 bytes long.

0000: 464F524D 00003AC0 38535658 56484452    FORM..:.8SVXVHDR
      F O R M     15040 8 S V X  V H D R
0010: 00000014 00003A98 00000000 00000000    ......:.........
            20    15000        0        0
0020: 27100100 00010000 424F4459 00003A98    '.......BODY..:.
     10000 1 0    1.0   B O D Y     15000

0000000..03 = "FORM", identifies this as an IFF format file.
FORM+00..03 (ULONG) = number of bytes that follow.  (Unsigned long int.)
FORM+03..07 = "8SVX", identifies this as an 8-bit sampled voice.

????+00..03 = "VHDR", Voice8Header, describes the parameters for the BODY.
VHDR+00..03 (ULONG) = number of bytes to follow.
VHDR+04..07 (ULONG) = samples in the high octave 1-shot part.
VHDR+08..0B (ULONG) = samples in the high octave repeat part.
VHDR+0C..0F (ULONG) = samples per cycle in high octave (if repeating), else 0.
VHDR+10..11 (UWORD) = samples per second.  (Unsigned 16-bit quantity.)
VHDR+12     (UBYTE) = number of octaves of waveforms in sample.
VHDR+13     (UBYTE) = data compression (0=none, 1=Fibonacci-delta encoding).
VHDR+14..17 (FIXED) = volume.  (The number 65536 means 1.0 or full volume.)

????+00..03 = "BODY", identifies the start of the audio data.
BODY+00..03 (ULONG) = number of bytes to follow.
BODY+04..NNNNN      = Data, signed bytes, from -128 to +127.

0030: 04030201 02030303 04050605 05060605
0040: 06080806 07060505 04020202 01FF0000
0050: 00000000 FF00FFFF FFFEFDFD FDFEFFFF
0060: FDFDFF00 00FFFFFF 00000000 00FFFF00
0070: 00000000 00FF0000 00FFFEFF 00000000
0080: 00010000 000101FF FF0000FE FEFFFFFE
0090: FDFDFEFD FDFFFFFC FDFEFDFD FEFFFEFE
00A0: FFFEFEFE FEFEFEFF FFFFFEFF 00FFFF01

This small section of the audio sample shows the number ranging from -5 (0xFD)
to +8 (0x08).  Warning: Do not assume that the BODY starts 48 bytes into the
file.  In addition to "VHDR", chunks labeled "NAME", "AUTH", "ANNO", or
"(c) " may be present, and may be in any order.  You will have to check the
byte count in each chunk to determine how many bytes to skip.

------------------------------------------------------------------------
Playing sound on a PC
---------------------

From: Eric A Rasmussen

Any turbo PC (8088 at 8 Mhz or greater)/286/386/486/etc. can produce a quality
playback of single channel 8 bit sounds on the internal (1 bit, 1 channel)
speaker by utilizing Pulse-Width-Modulation, which toggles the speaker faster
than it can physically move to simulate positions between fully on and fully
off.  There are several PD programs of this nature that I know of:

REMAC  - Plays MAC format sound files.  Files on the Macintosh, at least the
         sound files that I've ripped apart, seem to contain 3 parts.  The
         first two are info like what the file icon looks like and other
         header type info.  The third part contains the raw sample data, and
         it is this portion of the file which is saved to a seperate file,
         often named with the .snd extension by PC users.  Personally, I like
         to name the files .s1, .s2, .s3, or .s4 to indicate the sampling rate
         of the file. (-s# is how to specify the playback rate in REMAC.)
         REMAC provides playback rates of 5550hz, 7333hz, 11 khz, & 22 khz.
REMAC2 - Same as REMAC, but sounds better on higher speed machines.
REPLAY - Basically same as REMAC, but for playback of Atari ST sounds.
         Apparently, the Atari has two sound formats, one of which sounds like
         garbage if played by REMAC or REPLAY in the incorrect mode.  The
         other file format works fine with REMAC and so appears to be 'normal'
         unsigned 8-bit data.  REPLAY provides playback rates of 11.5 khz,
         12.5 khz, 14 khz, 16 khz, 18.5 khz, 22khz, & 27 khz.

These three programs are all by the same author, Richard E. Zobell who does
not have an internet mail address to my knowledge, but does have a GEnie email
address of R.ZOBELL.

Additionally, there are various stand-alone demos which use the internal
speaker, of which there is one called mushroom which plays a 30 second
advertising jingle for magic mushroom room deoderizers which is pretty
humerous.  I've used this player to playback samples that I ripped out of the
commercial game program Mean Streets, which uses something they call RealSound
(tm) to playback digital samples on the internal speaker. (Of course, I only do
this on my own system, and since I own the game, I see no problems with it.)

For owners of 8 Mhz 286's and above, the option to play 4 channel 8 bit sounds
(with decent quality) on the internal speaker is also a reality.  Quite a
number of PD programs exist to do this, including, but not limited to:

ModEdit, ModPlay, ScreamTracker, STM, Star Trekker, Tetra, and probably a few
more.

All these programs basically make use of various sound formats used by the
Amiga line of computers.  These include .stm files, .mod files
[a.k.a. mod. files], and .nst files [really the same hing].  Also,
these programs pretty much all have the option to playback the
sound to add-on hardware such as the SoundBlaster card, the Covox series of
devices, and also to direct the data to either one or two (for stereo)
parallel ports, which you could attach your own D/A's to.  (From what I have
seen, the Covox is basically an small amplified speaker with a D/A which plugs
into the parallel port.  This sounds very similiar to the Disney Sound System
(DSS) which people have been talking about recently.)

------------------------------------------------------------------------
The EA-IFF-85 documentation
---------------------------

From: [e-mail]

As promised, here's an ftp location for the EA-IFF-85 documentation.  It's
the November 1988 release as revised by Commodore (the last public release),
with specifications for IFF FORMs for graphics, sound, formatted text, and
more.  IFF FORMS now exist for other media, including structured drawing, and
new documentation is now available only from Commodore.

The documentation is at grind.isca.uiowa.edu [128.255.19.233], in the
directory /amiga/f1/ff185.  The complete file list is as follows:

DOCUMENTS.zoo
EXAMPLES.zoo
EXECUTABLE.zoo
INCLUDE.zoo
LINKER_INFO.zoo
OBJECT.zoo
SOURCE.zoo
TP_IFF_Specs.zoo

All files except DOCUMENTS.zoo are Amiga-specific, but may be used as a basis
for conversion to other platforms.  Well, I take that tentatively back.  I
don't know what TP_IFF_Specs.zoo contains, so it might be non-Amiga-specific.

------------------------------------------------------------------------
US Federal Standard 1016 availability
-------------------------------------

From: [e-mail] (Joe Campbell)

The U.S. DoD's Federal-Standard-1016 based 4800 bps code excited linear
prediction voice coder version 3.2 (CELP 3.2) Fortran and C simulation
source codes are available for worldwide distribution (on DOS
diskettes, but configured to compile on Sun SPARC stations) from NTIS
and DTIC.  Example input and processed speech files are included.  A
Technical Information Bulletin (TIB), "Details to Assist in
Implementation of Federal Standard 1016 CELP," and the official
standard, "Federal Standard 1016, Telecommunications:  Analog to
Digital Conversion of Radio Voice by 4,800 bit/second Code Excited
Linear Prediction (CELP)," are also available.

This is available through the National Technical Information Service:

NTIS
U.S. Department of Commerce
5285 Port Royal Road
Springfield, VA  22161
USA
[phone]

The "AD" ordering number for the CELP software is AD M000 118
(US$ 90.00) and for the TIB it's AD A256 629 (US$ 17.50).  The LPC-10
standard, described below, is FIPS Pub 137 (US$ 12.50).  There is a
$3.00 shipping charge on all U.S. orders.  The telephone number for
their automated system is 703-487-4650, or 703-487-4600 if you'd prefer
to talk with a real person.

(U.S. DoD personnel and contractors can receive the package from the
Defense Technical Information Center:  DTIC, Building 5, Cameron
Station, Alexandria, VA 22304-6145.  Their telephone number is
703-274-7633.)

The following articles describe the Federal-Standard-1016 4.8-kbps CELP
coder (it's unnecessary to read more than one):

Campbell, Joseph P. Jr., Thomas E. Tremain and Vanoy C. Welch,
"The Federal Standard 1016 4800 bps CELP Voice Coder," Digital Signal
Processing, Academic Press, 1991, Vol. 1, No. 3, p. 145-155.

Campbell, Joseph P. Jr., Thomas E. Tremain and Vanoy C. Welch,
"The DoD 4.8 kbps Standard (Proposed Federal Standard 1016),"
in Advances in Speech Coding, ed. Atal, Cuperman and Gersho,
Kluwer Academic Publishers, 1991, Chapter 12, p. 121-133.

Campbell, Joseph P. Jr., Thomas E. Tremain and Vanoy C. Welch, "The
Proposed Federal Standard 1016 4800 bps Voice Coder:  CELP," Speech
Technology Magazine, April/May 1990, p. 58-64.


The U.S. DoD's Federal-Standard-1015/NATO-STANAG-4198 based 2400 bps
linear prediction coder (LPC-10) was republished as a Federal
Information Processing Standards Publication 137 (FIPS Pub 137).
It is described in:

Thomas E. Tremain, "The Government Standard Linear Predictive Coding
Algorithm:  LPC-10," Speech Technology Magazine, April 1982, p. 40-49.

There is also a section about FS-1015 in the book:
Panos E. Papamichalis, Practical Approaches to Speech Coding,
Prentice-Hall, 1987.

The voicing classifier used in the enhanced LPC-10 (LPC-10e) is described in:
Campbell, Joseph P., Jr. and T. E. Tremain, "Voiced/Unvoiced Classification
of Speech with Applications to the U.S. Government LPC-10E Algorithm,"
Proceedings of the IEEE International Conference on Acoustics, Speech, and
Signal Processing, 1986, p. 473-6.

Copies of the official standard
"Federal Standard 1016, Telecommunications: Analog to Digital Conversion
of Radio Voice by 4,800 bit/second Code Excited Linear Prediction (CELP)"
are available for US$ 5.00 each from:

GSA Federal Supply Service Bureau
Specification Section, Suite 8100
470 E. L'Enfant Place, S.W.
Washington, DC  20407
[phone]

Realtime DSP code for FS-1015 and FS-1016 is sold by:

John DellaMorte
DSP Software Engineering
165 Middlesex Tpk, Suite 206
Bedford, MA  01730
USA
1-617-275-3733
1-617-275-4323 (fax)
[e-mail]

DSP Software Engineering's FS-1016 code can run on a DSP Research's Tiger 30
(a PC board with a TMS320C3x and analog interface suited to development work).

DSP Research
1095 E. Duane Ave.
Sunnyvale, CA  94086
USA
[phone]
[phone] (fax)

From: [e-mail] (Craig F. Reese)
Newsgroups: comp.speech,comp.dsp,comp.compression.research
Subject: CELP 3.2a release now available
Organization: Supercomputing Research Center (Bowie, MD)
Date: Tue, 3 Aug 1993 14:55:25 GMT

3 August 1993

CELP 3.2a Release

Dear CELPers,

We have placed an updated version of the FS-1016 CELP 3.2 code in the
anonymous FTP area on super.org (192.31.192.1).  It's in:

  /pub/celp_3.2a.tar.Z (please be sure to do the ftp in binary mode).

This is essentially the PC release that was on fumar, except that we
started directly from the PC disks.  The value added is that we have
made over 69 corrections and fixes.  Most of these were necessary
because of the 8 character file name limit on DOS, but there are some
others, as well.

The code (C, FORTRAN, diskio) all has been built and tested on a Sun4
under SunOS4.1.3.  If you want to run it somewhere else, then you may
have to do a bit of work.  (A Solaris 2.x-compatible release is
planned soon.)

[One note to PCers.  The files:
[
[       cbsearch.F celp.F csub.F mexcite.F psearch.F
[
[are meant to be passed through the C preprocessor (cpp).
[We gather that DOS (or whatever it's called) can't distinguish
[the .F from a .f.  Be careful!

Very limited support is available from the authors (Joe, et al.).
Please do not send questions or suggestions without first reading the
documentation (README files, the Technical Information Bulletin, etc.).
The authors would enjoy hearing from you, but they have limited time
for support and would like to use it as efficiently as possible.  They
welcome bug reports, but, again, please read the documentation first.
All users of FS-1016 CELP software are strongly encouraged to acquire
the latest release (version 3.2a as of this writing).

We do not know how long we will be able to leave the software on this
site, but it should be _at_least_ through 1 October 1993 (if you find
it missing, please drop me (Craig) a note).  Please try to get the
software during off hours (8 p.m. - 7 a.m. Eastern Standard time) or
folks here might complain and we'll have to get rid of the code (if
that happens, we'll try to pass it on to someone else, who can put it
on the net).  We would be more than happy for someone to copy it and
make it available elsewhere.

Good Luck,

Craig F. Reese  ([e-mail])
IDA/Supercomputing Research Center

Joe Campbell ([e-mail])
Department of Defense

P.S.  Just so you all know, I (Craig) am not actually involved in
CELP work.  I mainly got with Joe to help make the software available
on the Internet.  In the course of doing so, I cleaned up much of it,
but I am not, by any stretch, a CELP expert and will most likely
be unable to answer any technical questions concerning it. ;^)

From: [e-mail] (Richard Tobias)

For U.S. FED-STD-1016 (4800 bps CELP) _realtime_ DSP code and
information about products using this code using the AT&T DSP32C and
AT&T DSP3210, contact:

White Eagle Systems Technology, Inc.
1123 Queensbridge Way
San Jose, CA 95120
[phone]
[phone] (fax)
[e-mail]

From: Cole Erskine <[e-mail]>

[paraphrased]

Analogical Systems has a _real-time_ multirate implementation of U.S.
Federal Standard 1016 CELP operating at bit rates of 4800, 7200, and
9600 bps on a single 27MHz Motorola DSP56001. Source and object code
is available for a one-time license fee.

FREE, _real-time_ demonstration software for the Ariel PC-56D is
available for those who already have such a board by contacting
Analogical Systems.  The demo software allows you to record and
playback CELP files to and from the PC's hard disk.

Analogical Systems
2916 Ramona Street
Palo Alto, CA 94306
Tel: [phone]
FAX: [phone]

------------------------------------------------------------------------
Creative Voice (VOC) file format
--------------------------------

From: [e-mail]

(byte numbers are hex!)

    HEADER (bytes 00-19)
    Series of DATA BLOCKS (bytes 1A+) [Must end w/ Terminator Block]

- ---------------------------------------------------------------

HEADER:
=======
     byte #     Description
     ------     ------------------------------------------
     00-12      "Creative Voice File"
     13         1A (eof to abort printing of file)
     14-15      Offset of first datablock in .voc file (std 1A 00
                in Intel Notation)
     16-17      Version number (minor,major) (VOC-HDR puts 0A 01)
     18-19      2's Comp of Ver. # + 1234h (VOC-HDR puts 29 11)

- ---------------------------------------------------------------

DATA BLOCK:
===========

   Data Block:  TYPE(1-byte), SIZE(3-bytes), INFO(0+ bytes)
   NOTE: Terminator Block is an exception -- it has only the TYPE byte.

      TYPE   Description     Size (3-byte int)   Info
      ----   -----------     -----------------   -----------------------
      00     Terminator      (NONE)              (NONE)
      01     Sound data      2+length of data    *
      02     Sound continue  length of data      Voice Data
      03     Silence         3                   **
      04     Marker          2                   Marker# (2 bytes)
      05     ASCII           length of string    null terminated string
      06     Repeat          2                   Count# (2 bytes)
      07     End repeat      0                   (NONE)
      08     Extended        4                   ***

      *Sound Info Format:       **Silence Info Format:
       ---------------------      ----------------------------
       00   Sample Rate           00-01  Length of silence - 1
       01   Compression Type      02     Sample Rate
       02+  Voice Data

    ***Extended Info Format:
       ---------------------
       00-01  Time Constant: Mono: 65536 - (256000000/sample_rate)
                             Stereo: 65536 - (25600000/(2*sample_rate))
       02     Pack
       03     Mode: 0 = mono
                    1 = stereo


  Marker#           -- Driver keeps the most recent marker in a status byte
  Count#            -- Number of repetitions + 1
                         Count# may be 1 to FFFE for 0 - FFFD repetitions
                         or FFFF for endless repetitions
  Sample Rate       -- SR byte = 256-(1000000/sample_rate)
  Length of silence -- in units of sampling cycle
  Compression Type  -- of voice data
                         8-bits    = 0
                         4-bits    = 1
                         2.6-bits  = 2
                         2-bits    = 3
                         Multi DAC = 3+(# of channels) [interesting--
                                       this isn't in the developer's manual]

------------------------------------------------------------------------
RIFF WAVE (.WAV) file format
----------------------------

RIFF is a format by Microsoft and IBM which is similar in spirit and
functionality as EA-IFF-85, but not compatible (and it's in
little-endian byte order, of course :-).  WAVE is RIFF's equivalent of
AIFF, and its inclusion in Microsoft Windows 3.1 has suddenly made it
important to know about.

Rob Ryan was kind enough to send me a description of the RIFF format.
Unfortunately, it is too big to include here (27 k), but I've made it
available for anonymous ftp as ftp.cwi.nl:/pub/audio/RIFF-format.

And here's a pointer to the official description from Matt Saettler,
Microsoft Multimedia:

"The complete definition of the WAVE file format as defined by
IBM/Microsoft is available for anon. FTP from ftp.uu.net in the
vendor/microsoft/multimedia directory."

(Rob Ryan's version may actually be an extract from one of the files
stored there.)

------------------------------------------------------------------------
U-LAW and A-LAW definitions
---------------------------

[Adapted from information provided by [e-mail] (Rick
Duggan) and [e-mail] (David Perry)]

u-LAW (really mu-LAW) is

          sgn(m)   (     |m |)       |m |
   y=    ------- ln( 1+ u|--|)       |--| =< 1
         ln(1+u)   (     |mp|)       |mp|

A-LAW is

     |     A    (m )                 |m |    1
     |  ------- (--)                 |--| =< -
     |  1+ln A  (mp)                 |mp|    A
   y=|
     | sgn(m) (        |m |)    1    |m |
     | ------ ( 1+ ln A|--|)    - =< |--| =< 1
     | 1+ln A (        |mp|)    A    |mp|

Values of u=100 and 255, A=87.6, mp is the Peak message value, m is
the current quantised message value.  (The formulae get simpler if you
substitute x for m/mp and sgn(x) for sgn(m); then -1 <= x <= 1.)

Converting from u-LAW to A-LAW is in a sense "lossy" since there are
quantizing errors introduced in the conversion.

"..the u-LAW used in North America and Japan, and the
A-LAW used in Europe and the rest of the world and
international routes.."

References:

Modern Digital and Analog Communication Systems, B.P.Lathi., 2nd ed.
ISBN 0-03-027933-X

Transmission Systems for Communications
Fifth Edition
by Members of the Technical Staff at Bell Telephone Laboratories
Bell Telephone Laboratories, Incorporated
Copyright 1959, 1964, 1970, 1982

A note on the resolution of U-LAW by Frank Klemm <[e-mail]>:

8 bit U-LAW has the same lowest  magnitude like 12 bit linear and 12 bit
U-LAW like 16 linear.

Device/Coding   Resolution              Resolution
                on maximal level        on low level
 8 bit linear    8                       8
 8 bit ulaw      6                      12      (used for digital telephone)
12 bit linear   12                      12
12 bit ulaw     10                      16      (used in DAT/Longplay)
16 bit linear   16                      16

estimated for some analoge technique:
tape recorder (HiFi DIN)
                 8                       9      (no Problem today)
tape recorder (semiprofessional)
                10.5                    13.5

------------------------------------------------------------------------
AVR File Format
---------------

From: [e-mail] (Howard Chu)

A lot of PD software exists to play Mac .snd files on the ST. One other
format that seems pretty popular (used by a number of commercial packages)
is the AVR format (from Audio Visual Research). This format has a 128 byte
header that looks like this:

        char magic[4]="2BIT";
        char name[8];           /* null-padded sample name */
        short mono;             /* 0 = mono, 0xffff = stereo */
        short rez;              /* 8 = 8 bit, 16 = 16 bit */
        short sign;             /* 0 = unsigned, 0xffff = signed */
        short loop;             /* 0 = no loop, 0xffff = looping sample */
        short midi;             /* 0xffff = no MIDI note assigned,
                                   0xffXX = single key note assignment
                                   0xLLHH = key split, low/hi note */
        long rate;              /* sample frequency in hertz */
        long size;              /* sample length in bytes or words (see rez) */
        long lbeg;              /* offset to start of loop in bytes or words.
                                   set to zero if unused. */
        long lend;              /* offset to end of loop in bytes or words.
                                   set to sample length if unused. */
        short res1;             /* Reserved, MIDI keyboard split */
        short res2;             /* Reserved, sample compression */
        short res3;             /* Reserved */
        char ext[20];           /* Additional filename space, used
                                   if (name[7] != 0) */
        char user[64];          /* User defined. Typically ASCII message. */

-----------------------------------------------------------------------
The Amiga MOD Format
--------------------

From: [e-mail] (Norman Lin)

MOD files are music files containing 2 parts:

(1) a bank of digitized samples
(2) sequencing information describing how and when to play the samples

MOD files originated on the Amiga, but because of their flexibility
and the extremely large number of MOD files available, MOD players
are now available for a variety of machines (IBM PC, Mac, Sparc
Station, etc.)

The samples in a MOD file are raw, 8 bit, signed, headerless, linear
digital data.  There may be up to 31 distinct samples in a MOD file,
each with a length of up to 128K (though most are much smaller; say,
10K - 60K).  An older MOD format only allowed for up to 15 samples in
a MOD file; you don't see many of these anymore.  There is no standard
sampling rate for these samples.  [But see below.]

The sequencing information in a MOD file contains 4 tracks of
information describing which, when, for how long, and at what frequency
samples should be played.  This means that a MOD file can have up
to 31 distinct (digitized) instrument sounds, with up to 4 playing
simultaneously at any given point.  This allows a wide variety
of orchestrational possibilities, including use of voice samples
or creation of one's own instruments (with appropriate sampling
hardware/software).  The ability to use one's own samples as instruments
is a flexibility that other music files/formats do not share, and
is one of the reasons MOD files are so popular, numerous, and diverse.

15 instrument MODs, as noted above, are somewhat older than 31
instrument MODs and are not (at least not by me) seen very often
anymore.  Their format is identical to that of 31 instrument MODs
except:

(1) Since there are only 15 samples, the information for the last (15th)
    sample starts at byte 440 and goes through byte 469.
(2) The songlength is at byte 470 (contrast with byte 950 in 31 instrument
    MOD)
(3) Byte 471 appears to be ignored, but has been observed to be 127.
    (Sorry, this is from observation only)
(4) Byte 472 begins the pattern sequence table (contrast with byte 952
    in a 31 instrument MOD)
(5) Patterns start at byte 600 (contrast with byte 1084 in 31 instrument MOD)

"ProTracker," an Amiga MOD file creator/editor, is available for ftp
everywhere as pt??.lzh.

From: Apollo Wong <[e-mail]>

From: [e-mail] (Mark Cox)
Newsgroups: alt.sb.programmer
Subject: Re: Format for MOD files...
Message-ID: <[e-mail]>
Date: 18 Mar 92 10:36:08 GMT
Organization: University of Bradford, UK

[e-mail] (Winthrop D Chan) writes:
>I'd like to know if anyone has a reference document on the format of the
>Amiga Sound/NoiseTracker (MOD) files. The author of Modplay said he was going
>to release such a document sometime last year, but he never did. If anyone

I found this one, which covers it better than I can explain it - if you
use this in conjunction with the documentation that comes with Norman
Lin's Modedit program it should pretty much cover it.

Mark J Cox

/***********************************************************************

Protracker 1.1B Song/Module Format:
-----------------------------------

Offset  Bytes  Description
------  -----  -----------
   0     20    Songname. Remember to put trailing null bytes at the end...

Information for sample 1-31:

Offset  Bytes  Description
------  -----  -----------
  20     22    Samplename for sample 1. Pad with null bytes.
  42      2    Samplelength for sample 1. Stored as number of words.
               Multiply by two to get real sample length in bytes.
  44      1    Lower four bits are the finetune value, stored as a signed
               four bit number. The upper four bits are not used, and
               should be set to zero.
               Value:  Finetune:
                 0        0
                 1       +1
                 2       +2
                 3       +3
                 4       +4
                 5       +5
                 6       +6
                 7       +7
                 8       -8
                 9       -7
                 A       -6
                 B       -5
                 C       -4
                 D       -3
                 E       -2
                 F       -1

  45      1    Volume for sample 1. Range is $00-$40, or 0-64 decimal.
  46      2    Repeat point for sample 1. Stored as number of words offset
               from start of sample. Multiply by two to get offset in bytes.
  48      2    Repeat Length for sample 1. Stored as number of words in
               loop. Multiply by two to get replen in bytes.

Information for the next 30 samples starts here. It's just like the info for
sample 1.

Offset  Bytes  Description
------  -----  -----------
  50     30    Sample 2...
  80     30    Sample 3...
   .
   .
   .
 890     30    Sample 30...
 920     30    Sample 31...

Offset  Bytes  Description
------  -----  -----------
 950      1    Songlength. Range is 1-128.
 951      1    Well... this little byte here is set to 127, so that old
               trackers will search through all patterns when loading.
               Noisetracker uses this byte for restart, but we don't.
 952    128    Song positions 0-127. Each hold a number from 0-63 that
               tells the tracker what pattern to play at that position.
1080      4    The four letters "M.K." - This is something Mahoney & Kaktus
               inserted when they increased the number of samples from
               15 to 31. If it's not there, the module/song uses 15 samples
               or the text has been removed to make the module harder to
               rip. Startrekker puts "FLT4" or "FLT8" there instead.

Offset  Bytes  Description
------  -----  -----------
1084    1024   Data for pattern 00.
   .
   .
   .
xxxx  Number of patterns stored is equal to the highest patternnumber
      in the song position table (at offset 952-1079).

Each note is stored as 4 bytes, and all four notes at each position in
the pattern are stored after each other.

00 -  chan1  chan2  chan3  chan4
01 -  chan1  chan2  chan3  chan4
02 -  chan1  chan2  chan3  chan4
etc.

Info for each note:

 _____byte 1_____   byte2_    _____byte 3_____   byte4_
/                \ /      \  /                \ /      \
0000          0000-00000000  0000          0000-00000000

Upper four    12 bits for    Lower four    Effect command.
bits of sam-  note period.   bits of sam-
ple number.                  ple number.

Periodtable for Tuning 0, Normal
  C-1 to B-1 : 856,808,762,720,678,640,604,570,538,508,480,453
  C-2 to B-2 : 428,404,381,360,339,320,302,285,269,254,240,226
  C-3 to B-3 : 214,202,190,180,170,160,151,143,135,127,120,113

To determine what note to show, scan through the table until you find
the same period as the one stored in byte 1-2. Use the index to look
up in a notenames table.

This is the data stored in a normal song. A packed song starts with the
four letters "PACK", but i don't know how the song is packed: You can
get the source code for the cruncher/decruncher from us if you need it,
but I don't understand it; I've just ripped it from another tracker...

In a module, all the samples are stored right after the patterndata.
To determine where a sample starts and stops, you use the sampleinfo
structures in the beginning of the file (from offset 20). Take a look
at the mt_init routine in the playroutine, and you'll see just how it
is done.

Lars "ZAP" Hamre/Amiga Freelancers

***********************************************************************/

--
Mark J Cox -----
Bradford, UK ---


PS: A file with even *much* more info on MOD files, compiled by Lars
Hamre, is available from ftp.cwi.nl:/pub/audio/MOD-info.  Enjoy!


FTP sites for MODs and MOD players
----------------------------------

Subject: MODS AND PLAYERS!! **READ** info/where to get them
From: [e-mail] (Christopher Johnson)
Newsgroups: alt.binaries.sounds.d
Message-ID: <[e-mail]>
Date: 21 Dec 92 00:19:43 GMT
Organization: The University of Western Australia

Hello world,

For all those asking, here is where to get those mod players and mods.

SNAKE.MCS.KENT.EDU is the best site for general stuff.  look in /pub/SB-Adlib

Simtel-20 or archie.au(simtel mirror) in <msdos.sound>

for windows players ftp.cica.indiana.edu in pub/pc/win3/sound

here is a short list of players

mp or modplay   BEST OVERALL                    mp219b.zip
        simtel and snake

wowii           best for vga/fast machines      wowii12b.zip
        simtel and snake

trakblaster     best for compatability          trak-something
        simtel and snake        two versions, old one for slow
        machines

ss              cute display(hifi)              have_sex.arj
        found on local BBS (western Australia White Ghost)

superpro player generally good                  ssp.zip or similar
        found on night owl 7 CD

player?         cute display(hifi)              player.zip or similar
        found on night owl 7 CD

WINDOWS

Winmod pro      does protracker                 wmp????.zip
        cica

winmod          more stable                     winmod12.zip or similar
        cica

Hope this helps, e-mail me if you find any more players and I will add them in
for the next time mod player requests get a
little out of hand.

for mods ftp to wuarchive.wustl.edu and go to the amiga music directory (pub/am
iga/music/ntsb ?????)  that should do you for
a while

see you soon

Chris.

-----------------------------------------------------------------------