Celestin Apprentice 5/Apprentice-Release5.iso/Information/Specifications/Audio File Formats/Audio File Formats FAQ

Text File | 1996-06-23 | 115KB | 2,741 lines

Archive-name: audio-fmts
Submitted-by: Guido van Rossum <[e-mail]>
Version: 3.10
Last-modified: 2-Jan-1995

FAQ: Audio File Formats
=======================

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

Introduction
Device characteristics
Popular sampling rates
Compression schemes
Current hardware
File formats
File conversions
Playing audio files on UNIX
Playing audio files on micros
The Sound Site Newsletter
Posting sounds

Appendices (in part 2):

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


Introduction
------------

This is version 3 of this FAQ, which I started in November 1991 under
the name "The audio formats guide".  I bumped the major version number
again at the occasion of the split in two parts: part one is the main
text and part two consists of the collection of appendices.

I am posting this about once a fortnight, either unchanged (just to
inform new readers), or updated (if I learn more or when new hardware
or software becomes popular).  I post to alt.binaries.sounds.{misc,d}
and to comp.dsp, for maximal coverage of people interested in audio,
and to {news,comp}.answers, for easy reference.

The entire FAQ is also available by anonymous ftp from ftp.cwi.nl,
directory pub/audio, files AudioFormats.{part1,part2}.

BTW: All FAQs, including this one, are available for anonymous ftp on
the archive site rtfm.mit.edu in directory /pub/usenet/news.answers/.
The name under which a FAQ is archived appears in the "Archive-Name:"
line at the top of the article.  This FAQ is archived as
audio-fmts/part[12].

A companion posting with subject "Changes to: ..." is occasionally
posted listing the diffs between a new version and the last.  This is
not reposted, and it is suppressed when the diffs are bigger than the
new version.

Send updates, comments and questions to <[e-mail]>.  I'd like to
thank everyone who sent updates in the past.

--Guido van Rossum, CWI, Amsterdam <[e-mail]>


Device characteristics
----------------------

In this text, I will only use the term "sample" to refer to a single
output value from an A/D converter, i.e., a small integer number
(usually 8 or 16 bits).

Audio data is characterized by the following parameters, which
correspond to settings of the A/D converter when the data was
recorded.  Naturally, the same settings must be used to play the data.

- sampling rate (in samples per second), e.g. 8000 or 44100

- number of bits per sample, e.g. 8 or 16

- number of channels (1 for mono, 2 for stereo, etc.)

Approximate sampling rates are often quoted in Hz or kHz ([kilo-]
Hertz), however, the politically correct term is samples per second
(samples/sec).  Sampling rates are always measured per channel, so for
stereo data recorded at 8000 samples/sec, there are actually 16000
samples in a second.  I will sometimes write 8 k as a shorthand for
8000 samples/sec.

Multi-channel samples are generally interleaved on a frame-by-frame
basis: if there are N channels, the data is a sequence of frames,
where each frame contains N samples, one from each channel.  (Thus,
the sampling rate is really the number of *frames* per second.)  For
stereo, the left channel usually comes first.

The specification of the number of bits for U-LAW (pronounced mu-law
-- the u really stands for the Greek letter mu) samples is somewhat
problematic.  These samples are logarithmically encoded in 8 bits,
like a tiny floating point number; however, their dynamic range is
that of 12 bit linear data.  Source for converting to/from U-LAW
(written by Jef Poskanzer) is distributed as part of the SOX package
mentioned below; it can easily be ripped apart to serve in other
applications.  The official definition is the CCITT standard G.711.

There exists another encoding similar to U-LAW, called A-LAW, which
is used as a European telephony standard.  There is less support for
it in UNIX workstations.

(See the Appendix for some formulae describing U-LAW and A-LAW.)


Popular sampling rates
----------------------

Some sampling rates are more popular than others, for various reasons.
Some recording hardware is restricted to (approximations of) some of
these rates, some playback hardware has direct support for some.  The
popularity of divisors of common rates can be explained by the
simplicity of clock frequency dividing circuits :-).

Samples/sec     Description

5500            One fourth of the Mac sampling rate (rarely seen).

7333            One third of the Mac sampling rate (rarely seen).

8000            Exactly 8000 samples/sec is a telephony standard that
                goes together with U-LAW (and also A-LAW) encoding.
                Some systems use an slightly different rate; in
                particular, the NeXT workstation uses 8012.8210513,
                apparently the rate used by Telco CODECs.

11 k            Either 11025, a quarter of the CD sampling rate,
                or half the Mac sampling rate (perhaps the most
                popular rate on the Mac).

16000           Used by, e.g. the G.722 compression standard.

18.9 k          CD-ROM/XA standard.

22 k            Either 22050, half the CD sampling rate, or the Mac
                rate; the latter is precisely 22254.545454545454 but
                usually misquoted as 22000.  (Historical note:
                22254.5454... was the horizontal scan rate of the
                original 128k Mac.)

32000           Used in digital radio, NICAM (Nearly Instantaneous
                Compandable Audio Matrix [IBA/BREMA/BBC]) and other
                TV work, at least in the UK; also long play DAT and
                Japanese HDTV.

37.8 k          CD-ROM/XA standard for higher quality.

44056           This weird rate is used by professional audio
                equipment to fit an integral number of samples in a
                video frame.

44100           The CD sampling rate.  (DAT players recording
                digitally from CD also use this rate.)

48000           The DAT (Digital Audio Tape) sampling rate for
                domestic use.

Files samples on SoundBlaster hardware have sampling rates that are
divisors of 1000000.

While professinal musicians disagree, most people don't have a problem
if recorded sound is played at a slightly different rate, say, 1-2%.
On the other hand, if recorded data is being fed into a playback
device in real time (say, over a network), even the smallest
difference in sampling rate can frustrate the buffering scheme used...

There may be an emerging tendency to standardize on only a few
sampling rates and encoding styles, even if the file formats may
differ.  The suggested rates and styles are:

    rate (samp/sec) style mono/stereo

    8000 8-bit U-LAW mono
    22050 8-bit linear unsigned mono and stereo
    44100 16-bit linear signed mono and stereo


Compression schemes
-------------------

Strange though it seems, audio data is remarkably hard to compress
effectively.  For 8-bit data, a Huffman encoding of the deltas between
successive samples is relatively successful.  For 16-bit data,
companies like Sony and Philips have spent millions to develop
proprietary schemes.  Information about PASC (Philips' scheme) can be
found in Advanced Digital Audio by Ken C. Pohlmann.

Public standards for voice compression are slowly gaining popularity,
e.g. CCITT G.721 (ADPCM at 32 kbits/sec) and G.723 (ADPCM at 24 and 40
kbits/sec).  (ADPCM == Adaptive Delta Pulse Code Modulation.)  Sun
Microsystems has placed the source code of a portable implementation of
these algorithms (as well as G.711, which defines A-LAW and U-LAW) in
the public domain (needless to say, their proprietary implementation
distributed in binary form with Solaris is better :-).  One place to
ftp this source code from is ftp.cwi.nl:/pub/audio/ccitt-adpcm.tar.Z.
Source for another 32 kbits/sec ADPCM implementation, assumed to be
compatible with Intel's DVI audio format, can be ftp'ed from
ftp.cwi.nl:/pub/audio/adpcm.shar.  (** NOTE: if you are using v1.0,
you should get v1.1, released 17-Dec-1992, which fixes a serious bug
-- the quality of v1.1 is claimed to be better than U-LAW **)

GSM 06.10 is a speech encoding in use in Europe that compresses 160
13-bit samples into 260 bits (or 33 bytes), i.e. 1650 bytes/sec (at
8000 samples/sec).  A free implementation can be ftp'ed from
tub.cs.tu-berlin.de, file /pub/tubmik/gsm-1.0.tar.Z.

There are also two US federal standards, 1016 (Code excited linear
prediction (CELP), 4800 bits/s) and 1015 (LPC-10E, 2400 bits/s).  See
also the appendix for 1016.

Tony Robinson <[e-mail]> has written a good FAST loss-less
compression for lots of different audio formats (particularly good for
WAV and MOD files).  The software is available by anonymous ftp from
svr-ftp.eng.cam.ac.uk, directory misc, file shorten-1.08.tar.Z.

(Note that U-LAW and silence detection can also be considered
compression schemes.)

Here's a note about audio codings by Van Jacobson <[e-mail]>:
Several people used the words "LPC" and "CELP" interchangably.  They
are very different.  An LPC (Linear Predictive Coding) coder fits
speech to a simple, analytic model of the vocal tract, then throws
away the speech & ships the parameters of the best-fit model.  An LPC
decoder uses those parameters to generate synthetic speech that is
usually more-or-less similar to the original.  The result is
intelligible but sounds like a machine is talking.  A CELP (Code
Excited Linear Predictor) coder does the same LPC modeling but then
computes the errors between the original speech & the synthetic model
and transmits both model parameters and a very compressed
representation of the errors (the compressed representation is an
index into a 'code book' shared between coders & decoders -- this is
why it's called "Code Excited").  A CELP coder does much more work
than an LPC coder (usually about an order of magnitude more) but the
result is much higher quality speech: The FIPS-1016 CELP we're working
on is essentially the same quality as the 32Kb/s ADPCM coder but uses
only 4.8Kb/s (the same as the LPC coder).

The comp.compression FAQ has some text on the 6:1 audio compression
scheme used by MPEG (a video compression standard-to-be).  It's
interesting to note that video compression reaches much higher ratios
(like 26:1).  This FAQ is ftp'able from rtfm.mit.edu in directory
/pub/usenet/news.answers/compression-faq, files part1 and part2.

Comp.compression also carries a regular posting "How to uncompress
anything" by David Lemson <[e-mail]>, which (tersely) hints on
which program you need to uncompress a file whose name ends in .<foo>
for almost any conceivable <foo>.  Ftp'able from ftp.cso.uiuc.edu
in the directory /doc/pcnet as the file compression.

Documentation on a digital cellular telephone system by Qualcomm Inc.
can be ftp'ed from ftp.qualcomm.com:/pub/cdma; the vocoder is in
appendix A.

Apple has an Audio Compression/Expansion scheme called ACE (on the GS)
/ MACE (on the Macintosh).  It's a lossy scheme that attempts to
predict where the wave will go on the next sample. There's very little
quality change on 8:4 compression, somewhat more for 8:3.  It does
guarantee exactly 50% or 62.5% compression, though.  I believe MACE
uses larger ratios/more loss, but I'm unsure of the specific numbers.
(Marc Sira)


Current hardware
----------------

I am aware of the following computer systems that can play back and
(sometimes) record audio data, with their characteristics.  Note that
for most systems you can also buy "professional" sampling hardware,
which supports much better quality, e.g. >= 44.1 k 16 bits stereo.
The characteristics listed here are a rough estimate of the
capabilities of the basic hardware only (and even here I am on thin
ice, with systems becoming ever more powerful).

machine             bits            max sampling rate    #output channels

Mac (all types)     8               22k                  1
Mac (newer ones)    16              64k                  4(128)
Apple IIgs          8               32k / >70k           16(st)
PC/soundblaster pro 8               ?/(22k st, 44.1k mo) 1(st)
PC/soundblaster 16  16              44.1k                1(st)
PC/pas              8               44.1k st, 88.2k mo   1(st)
PC/pas-16           16              44.1k st, 88.2k mo   1(st)
PC/turtle beach multisound 16       44.1k                1(st)
PC/cards with aria chipset 16       44.1k                1(st)
PC/roland rap-10    16              44.1k                1(st)
PC/gravis ultrasound 8/16           44.1k                14-32(st)
Atari ST            8               22k                  1
Atari STE,TT        8               50k                  2
Atari Falcon 030    16              50k                  8(st)
Amiga               8               varies above 29k     4(st)
Sun Sparc           U-LAW           8k                   1
Sun Sparcst. 10     U-LAW,8,16      48k                  1(st)
NeXT                U-LAW,8,16      44.1k                1(st)
SGI Indigo          8,16            48k                  4(st)
SGI Indigo2,Indy    8,16            48k                  16(st,4-channel)
Acorn Archimedes    ~U-LAW          ~180k                8(st)
Sony NWS-3xxx       U,A,8,16        8-37.8k              1(st)
Sony NWS-5xxx       U,A,8,16        8-48k                1(st)
VAXstation 4000     U-LAW           8k                   1
DEC 3000            U-LAW           8k                   1
DEC 5000/20-25      U-LAW           8k                   1
Tandy 1000/*L*      8               >=44k                1
Tandy 2500          8               >=44k                1
HP9000/705,710,425e U,A-LAW,16      8k                   1
HP9000/715,725,735  U,A-LAW,16      48k                  1(st)
HP9000/755 option:  U,A-LAW,16      48k                  1(st)
NCD MCX terminal    U,A,8,16        52k                  1(st)

4(st) means "four voices, stereo"; sampling rates xx/yy are
different recording/playback rates; *L* is any type with 'L' in it.

All these machines can play back sound without additional hardware,
although the needed software is not always standard; also, some
machines need external hardware to record sound (or to record at
higher quality, like the NeXT, whose built-in sampling hardware only
does 8000 samples/sec in U-LAW).  Please don't send me details on
optional or 3rd party hardware, there is too much and it is really
beyond the scope of this FAQ.  In particular, there is a separate
newsgroup devoted to PC sound cards: comp.sys.ibm.pc.soundcard, which
includes FAQ of its own (also posted to comp.answers and news.answers).

The new VAXstation 4000 (VLC and model 60) series lets you PLAY audio
(.au) files, and the package DECsound will let you do the recording.
In fact, DECsound is given away free with Motif 1.1 and supports the
VAXstation, Sun SPARCstation, DECvoice, and DECaudio devices.  Sun
sound files work without change.  The Alpha systems also have DECsound
bundled with Motif.  Also, the DEC2000/300 (aka DECpc AXP 150) can use
a Microsoft Sound Card, with AudioFile (see below) for sound.

Notes for the DECstation 5000/20-25: You need either XMedia tools from
DEC ($$$$), or the AudioFile package (which works nicely) from
crl.dec.com (see below). The audio device is "/dev/bba", you cannot
send ".au" files directly to the device, the Xmedia/AF software
provide an "audioserver" which must be run to play/record sounds.

The SGI Personal IRIS 4D/30 and 4D/35 have the same capabilities as
the Indigo. The audio board was optional on the 4D/30.
The Indigo2 and Indy features are a superset of the Indigo features.

The new Apple Macs have more powerful audio hardware; the latest
models have built-in microphones.

Software exists for the PC that can play sound on its 1-bit speaker
using pulse width modulation (see appendix); the Soundblaster board
records at rates up to 13 k and plays back up to 22 k (weird
combination, but that's the way it is).

Here's some info about the newest Atari machine, the Falcon030.  This
machine has stereo 16 bit CODECs and a 32 MHz Motorola 56001 that can
handle 8 channels of 16 bit audio, up to 50 khz/channel with
simultaneous playback and record.  The Falcon DMA sound engine is also
compatible with the 8 bit stereo DMA used on the STe and TT. All of
these systems use signed data.

On the NeXT, the Motorola 56001 DSP chip is programmable and you can
(in principle) do what you want.  The SGI Indigo uses the same DSP chip but
it can't be programmed by users -- SGI prefers to offer it as a shared
system resource to multiple applications, thus enabling developers to
program audio with their Audio Library and avoid code modifications
for execution on future machines with different audio hardware, i.e. a
different DSP. For example, the Indigo2 and Indy do not have a DSP chip.

The Amiga also has a 6-bit volume, which can be used to produce
something like a 14-bit output for each voice.  The hardware can also
use one of each voice-pair to modulate the other in FM (period) or AM
(volume, 6-bits).

The Acorn Archimedes uses a variation on U-LAW with the bit order
reversed and the sign bit in bit 0.  Being a 'minority' architecture,
Arc owners are quite adept at converting sound/image formats from
other machines, and it is unlikely that you'll ever encounter sound in
one of the Arc's own formats (there are several).

Tandy notes (Jeffrey L. Hayes <[e-mail]>): The maximum
sampling rate for output is at least 44k.  (I don't know the maximum
rates; I have recorded at 22k and played at 44k.  Higher rates are
probably possible.)  There is one output channel, not three.  The
belief that there are 3 channels probably stems from the fact that
Music.pdm, bundled with these machines, can create 3- channel music
modules (analogous to Amiga .mod's).  Music.pdm probably does that
because it is designed to work with the Tandy's 3-voice tone generator
circuitry (compatible with the Texas Instruments SN76496 in the IBM
PC-Jr) if there is insufficient RAM to load sound samples.  The Tandy
chip is able to record at lower rates than it is able to play back, as
is the Soundblaster (i.e., the divider used to program the chip to
record is lower than that used to program the chip to play back).  The
Tandy DAC can go faster than the original Soundblaster, however.

The NCD MCX terminal has audio integrated with its X server.  The
NCDAudio server is an extension of the X server, working together with
it, with stress on the networking capability of sound transmission.
The NCDAudio API provides format handling (ULAW8, Linear Unsig 8,
Linear Sig 8, Linear Sig 16 MSB, Linear Unsig 16 MSB), flowing (to the
server, from the server, to the i/o, from the i/o), wave form
generators (Square, Sine, Saw, Constant) and the capability of area
broadcast using UDP.  Provision for manipulating data files
(SND, WAV, VOC & AU) is also provided.

CD-I machines form a special category.  The following formats are used:

     - PCM 44.1 kHz standard CD format
     - ADPCM - Addaptive Delta PCM
       - Level A 37.8 kHz 8-bit
       - Level B 37.8 kHz 4-bit
       - Level C 18.9 kHz 4-bit


File formats
------------

Historically, almost every type of machine used its own file format
for audio data, but some file formats are more generally applicable,
and in general it is possible to define conversions between almost any
pair of file formats -- sometimes losing information, however.

File formats are a separate issue from device characteristics.  There
are two types of file formats: self-describing formats, where the
device parameters and encoding are made explicit in some form of
header, and "raw" formats, where the device parameters and encoding
are fixed.

Self-describing file formats generally define a family of data
encodings, where a header fields indicates the particular encoding
variant used.  Headerless formats define a single encoding and usually
allows no variation in device parameters (except sometimes sampling
rate, which can be a pain to figure out other than by listening to the
sample).

The header of self-describing formats contains the parameters of the
sampling device and sometimes other information (e.g. a
human-readable description of the sound, or a copyright notice).  Most
headers begin with a simple "magic word".  (Some formats do not simply
define a header format, but may contain chunks of data intermingled
with chunks of encoding info.)  The data encoding defines how the
actual samples are stored in the file, e.g. signed or unsigned, as
bytes or short integers, in little-endian or big-endian byte order,
etc.  Strictly spoken, channel interleaving is also part of the
encoding, although so far I have seen little variation in this area.

Some file formats apply some kind of compression to the data, e.g.
Huffman encoding, or simple silence deletion.

Here's an overview of popular file formats.

        Self-describing file formats
        ----------------------------

extension, name   origin          variable parameters (fixed; comments)

.au or .snd       NeXT, Sun       rate, #channels, encoding, info string
.aif(f), AIFF     Apple, SGI      rate, #channels, sample width, lots of info
.aif(f), AIFC     Apple, SGI      same (extension of AIFF with compression)
.iff, IFF/8SVX    Amiga           rate, #channels, instrument info (8 bits)
.voc              Soundblaster    rate (8 bits/1 ch; can use silence deletion)
.wav, WAVE        Microsoft       rate, #channels, sample width, lots of info
.sf               IRCAM           rate, #channels, encoding, info
none, HCOM        Mac             rate (8 bits/1 ch; uses Huffman compression)
none, MIME        Internet        (see below)
none, NIST SPHERE DARPA speech community (see below)
.mod or .nst      Amiga           (see below)

Note that the filename extension ".snd" is ambiguous: it can be either
the self-describing NeXT format or the headerless Mac/PC format, or
even a headerless Amiga format.

I know nothing for sure about the origin of HCOM files, only that
there are a lot of them floating around on our system and probably at
FTP sites over the world.  The filenames usually don't have a ".hcom"
extension, but this is what SOX (see below) uses.  The file format
recognized by SOX includes a MacBinary header, where the file
type field is "FSSD".  The data fork begins with the magic word "HCOM"
and contains Huffman compressed data; after decompression it it is 8
bits unsigned data.

IFF/8SVX allows for amplitude contours for sounds (attack/decay/etc).
Compression is optional (and extensible); volume is variable; author,
notes and copyright properties; etc.

AIFF, AIFC and WAVE are similar in spirit but allow more freedom in
encoding style (other than 8 bit/sample), amongst others.

There are other sound formats in use on Amiga by digitizers and music
programs, such as IFF/SMUS.

Appendices describes the NeXT and VOC formats; pointers to more info
about AIFF, AIFC, 8SVX and WAVE (which are too complex to describe
here) are also in appendices.

DEC systems (e.g. DECstation 5000) use a variant of the NeXT format
that uses little-endian encoding and has a different magic number
(0x0064732E in little-endian encoding).

Standard file formats used in the CD-I world are IFF but on the disc
they're in realtime files.

An interesting "interchange format" for audio data is described in the
proposed Internet Standard "MIME", which describes a family of
transport encodings and structuring devices for electronic mail.  This
is an extensible format, and initially standardizes a type of audio
data dubbed "audio/basic", which is 8-bit U-LAW data sampled at 8000
samples/sec.

The "IRCAM" sound file system has now been superseded by the so-called
"BICSF" (for Berkeley/IRCAM/CARL Sound File system) software release.
More recently, there has been an effort at Princeton (Prof. Paul
Lansky) and Stanford (Stephen Travis Pope) to standardize several
extensions to BICSF.  A description of BICSF and the
Princeton/Stanford extensions is available by anonymous ftp from
ftp.cwi.nl, in directory /pub/audio/BICSF-info.  This file contains
further ftp pointers to software.

A sound file format popular in the DARPA speech community is the NIST
SPHERE standard.  The most recent version of the SPHERE package is
available via anonymous ftp from jaguar.ncsl.nist.gov in compressed
tar form as "sphere-v.tar.Z" (where "v" is the version code).  The
NIST SPHERE header is an object-oriented, 1024-byte blocked, ASCII
structure which is prepended to the waveform data.  The header is
composed of a fixed-format portion followed by an object-oriented
variable portion.  I have placed a short description of NIST SPHERE on
ftp.cwi.nl:/pub/audio/NIST-SPHERE.

Finally, a somewhat different but popular format are "MOD" files,
usually with extension ".mod" or ".nst" (they can also have a prefix
of "mod.").  This originated at the Amiga but players now exist for
many platforms.  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.  See the appendix "The Amiga MOD Format"
for a description of this file format (and pointers to ftp'able
players and example MOD files).

        Headerless file formats
        -----------------------

extension       origin          parameters
or name

.snd, .fssd     Mac, PC         variable rate, 1 channel, 8 bits unsigned
.ul             US telephony    8 k, 1 channel, 8 bit "U-LAW" encoding
.snd?           Amiga           variable rate, 1 channel, 8 bits signed

It is usually easy to distinguish 8-bit signed formats from unsigned
by looking at the beginning of the data with 'od -b <file | head';
since most sounds start with a little bit of silence containing small
amounts of background noise, the signed formats will have an abundance
of bytes with values 0376, 0377, 0, 1, 2, while the unsigned formats
will have 0176, 0177, 0200, 0201, 0202 instead.  (Using "od -c" will
also show any headers that are tacked in front of the file.)

The Apple IIgs records raw data in the same format as the Mac, but
uses a 0 byte as a terminator; samples with value 0 are replaced by 1.

        Sound formats and the Apple Macintosh
        -------------------------------------

(Thanks to Bill Houle, <[e-mail]>)

                             SOX/DOS   MAC
Sound Format           file ext  type  Mac program to convert to 'snd'
---------------------- --------  ----  -------------------------------
Mac snd                .snd      sfil  [n/a]
Amiga IFF/8SVX         .iff            AmigaSndConverter, BST
Amiga SoundTracker     .mod      STrk  ModVoicer
Audio IFF              .aiff     AIFF  SoundExtractor, Sample Editor,
                                       UUTool, BST, M5Mac
DSP Designer                     DSPs  SoundHack
IRCAM                  .sf       IRCM  SoundHack
MacMix                           MSND  SoundHack
RIFF WAVE              .wav            SoundExtractor, BST, Balthazar
SoundBlaster           .voc            SoundExtractor, BST
SoundDesigner/AudioMedia         Sd2f  SoundHack
Sound[Edit|Cap|Wave]   .hcom     FSSD  SoundExtractor, SoundEdit,
                                       Wavicle, BST
Sun uLaw/Next .snd     .au/.snd  NxTS  SoundExtractor, SoundHack,
                                       au<->snd, UUTool, BST


File conversions
----------------

        SOX (UNIX, PC, Amiga)
        ---------------------

The most versatile tool for converting between various audio formats
is SOX ("Sound Exchange").  It can read and write various types of
audio files, and optionally applies some special effects (e.g. echo,
channel averaging, or rate conversion).

SOX recognizes all filename extensions listed above except ".snd",
which would be ambiguous anyway, and ".wav" (but there's a patch, see
below).  Use type ".au" for NeXT ".snd" files.  Mac and PC ".snd"
files are completely described by these parameters:

        -t raw -b -u -r 11000

(or -r 22000 or -r 7333 or -r 5500; 11000 seems to be the most common
rate).

The source for SOX, version 6, platchlevel 8, was posted to
alt.sources, and should be widely archived.  (Patch 9 was posted later
and incporporates some important .wav fixes.)  To save you the trouble
of hunting it down, it can be gotten by anonymous ftp from
wuarchive.wustl.edu, in the directory usenet/alt.sources/articles,
files 7288.Z through 7295.Z.  (These files are compressed news
articles containing shar files, if you hadn't guessed.)  I am sure
many sites have similar archives, I'm just listing one that I know of
and which carries a lot of this kind of stuff.  (Also see the appendix
if you don't have Internet access.)

A compressed tar file containing the same version of SOX is available
by anonymous ftp from ftp.cwi.nl, in directory
/pub/audio/sox<version>.tar.Z.  You may be able to locate a nearer
version using archie!

Ports of SOX:

- The source as posted should compile on any UNIX and PC system.

- A PC version is available by ftp from ftp.cwi.nl (see above) as
  pub/audio/sox5dos.zip; also available from the garbo mail server.

- The latest Amiga SOX is available via anonymous ftp to
  wuarchive.wustl.edu, files systems/amiga/audio/utils/amisox*.  (See
  below for a non-SOX solution.)
  The final release of r6 will compile as distributed on the Amiga with
  SAS/C version 6.  Binaries (since many Amiga users do not own
  compilers) will continue to be available for FTP.

SOX usage hints:

- Often, the filename extension of sound files posted on the net is
  wrong.  Don't give up, try a few other possibilities using the
  "-t <type>" option.  Remember that the most common file type is
  unsigned bytes, which can be indicated with "-t ub".  You'll have to
  guess the proper sampling rate, but often it's 11k or 22k.

- In particular, with SOX version 4 (or earlier), you have to
  specify "-t 8svx" for files with an .iff extension.

- When converting linear samples to U-LAW using the .au type for the
  output file, you must specify "-U" for the output file, otherwise
  you will end up with a file containing a NeXT/Sun header but linear
  samples -- only the NeXT will play such files correctly.  Also, you
  must explicitly specify an output sampling rate with "-r 8000".
  (This may seem fixed for most cases in version 5, but it is still
  occasionally necessary, so I'm keeping this warning in.)

        Sun Sparc
        ---------

On Sun Sparcs, starting at SunOS 4.1, a program "raw2audio" is
provided by Sun (in /usr/demo/SOUND -- see below) which takes a raw
U-LAW file and turns it into a ".au" file by prefixing it with an
appropriate header.

        NeXT
        ----

On NeXTs, you can usually rename .au files to .snd and it'll work like
a charm, but some .au files lack header info that the NeXT needs.
This can be fixed by using sndconvert:

        sndconvert -c 1 -f 1 -s 8012.8210513 -o nextfile.snd sunfile.au

        SGI Indigo, Indigo2, Indy and Personal IRIS
        -------------------------------------------

SGI supports "soundfiler" (in /usr/sbin), a program similar in
spirit to SOX but with a GUI.  Soundfiler plays aiff, aifc, NeXT/Sun
and .wav formats.  It can do conversions between any of these formats
and to and from raw formats including mulaw.  It also does sample rate
conversions.

Three shell commands are also provided that give the same functionality:
"sfplay", "sfconvert", and "aifcresample" (all in /usr/sbin).

        Amiga
        -----

Mike Cramer's SoundZAP can do no effects except rate change and it
only does conversions to IFF, but it is generally much faster than
SOX.  (Ftp'able from the same directory as amisox above.)

Newer versions of OmniPlay (see below) will also convert to IFF.

        Tandy
        -----

The Tandy uses a proprietary format, which can use compression
(see appendix).  Jeffrey L. Hayes <[e-mail]> writes:

There is in fact a Windows 3.1 sound driver for the Tandy 2500-series
available from Radio Shack.  My informant says:  "Say that you have a
2500SX/33 and you lost your Windows Utilities/Drivers disk.  The cost is
$5.00."  (The driver will work on any 2500.)

Version 2.00 of Conv2snd by Kenneth Udut by Kenneth Udut is now on
Simtel.  It converts any 8-bit mono unsigned PCM file to Tandy
DeskMate .snd format.  The new version recognizes RIFF WAVE headers
and comes with a utility to convert .snd to .wav, Snd2wav.

In addition to the .snd format used by Sound.pdm, Tandy used an .sng format
with Music.pdm for song files.  .sng files are analogous to Amiga .mod
files, but they contain only the sequencing information.  The samples are
expected to be in .snd files in the current directory for Music.pdm.  It
should be possible to convert .sng to .mod - when I get around to it!

I have a collection of programs and information on the Tandy DAC on Simtel:
oak.oakland.edu:/pub/msdos/sound/tspak.zip.  A program to convert Tandy
.snd to .mod samples is included.

There are two Tandy .snd formats.  The old format was used on the 1000's;
the new format on the 2500's.  The 2500's can read the old format.

Tandy now includes Soundblaster support in its machines.  New Tandy's do
not have the proprietary Tandy DAC.


        Apple Macintosh
        ---------------

Bill Houle sent the following list:

Popular commercial apps are indicated with a [*].  All other programs
mentioned are shareware/freeware available from SUMEX and the various
mirror sites, or check archie for the nearest FTP location.

MAC SOUND CONVERSION PROGRAMS

SoundHack [Tom Erbe, [e-mail]]
  Can read/write Sound Designer II, Audio IFF, IRCAM, DSP Designer and NeXT
  .snd (or Sun .au); 8-bit uLaw, 8-bit linear, 32-bit floating point and 16-bit
  linear data encoding.  Can read (but not write) raw data files.  Implements
  soundfile convolution, a phase vocoder, a binaural filter and an amplitude
  analysis & gain change module.

SoundExtractor [Alberto Ricci, [e-mail]]
  Extracts 'snd' resources, AIFF, SoundEdit, VOC, and WAV data from
  practically anything, converting to 'snd' files.

Balthazar [Craig Marciniak, AOL:TemplarDev]
  Converts WAV files to 'snd'.

Brian's Sound Tool [Brian Scott, [e-mail]]
  Converts 'snd' or SoundEdit to WAV.  Can also convert WAV, VOC, AIFF, Amiga
  8SVX and uLaw to 'snd'.

AmigaSndConverter [Povl H. Pederson, [e-mail]]
  Converts Amiga IFF/8SVX to Mac 'snd'.

au<->Mac [Victor J. Heinz, vic:[e-mail]]
  Converts Sun uLaw to Mac 'snd'.

ULAW [Rod Kennedy, [e-mail]]
  Converts 'snd' to Sun uLaw.

UUTool [Bernie Wieser, [e-mail]]
  Primarily a uuencode/decode program, but in true Swiss Army Knife
  fashion can also read/write Sun uLaw, AIFF, and 'snd' files.

ModVoicer [Kip Walker, [e-mail]]
  Converts Amiga MOD voices into SoundEdit files or 'snd' resources.

Music 5 Mac [Simone Bettini, [e-mail]]
  Primarily a Music Synthesis system, but can also convert between 'snd', AIFF,
  and IBM .DAT(?).

See also the section on players -- some players also do conversions.


Playing audio files on UNIX
---------------------------

The commands needed to play an audio file depend on the file format
and the available hardware and software.  Most systems can only
directly play sound in their native format; use a conversion program
(see above) to play other formats.

        Sun Sparcstation running SunOS 4.x
        ----------------------------------

Raw U-LAW files can be played using "cat file >/dev/audio".

A whole package for dealing with ".au" files is provided by Sun on an
experimental basis, in /usr/demo/SOUND.  You may have to compile the
programs first.  (If you can't find this directory, either you are not
running SunOS 4.1 yet, or your system administrator hasn't installed
it -- go ask him for it, not me!)  The program "play" in this
directory recognizes all files in Sun/NeXT format, but a SS 1 or 2 can
play only those using U-LAW encoding at 8 k -- the SS 10 hardware
plays other encodings, too.

If you ca't find "play", you can also cat a ".au" file to /dev/audio,
if it uses U-LAW; the header will sound like a short burst of noise
but the rest of the data will sound OK (really, the only difference in
this case between raw U-LAW and ".au" files is the header; the U-LAW
data is exactly the same).

Finally, OpenWindows 3.0 has a full-fledged audio tool.  You can drop
audio file icons into it, edit them, etc.

        Sun Sparcstation running Solaris 2.0
        ------------------------------------

Under SVR4 (and hence Solaris 2.0), writing to /dev/audio from the
shell is a bad idea, because the device driver will flush its queue as
soon as the file is closed.  Use "audioplay" instead.  The supported
formats and sampling rates are the same as above.

        NeXT
        ----

On NeXT machines, the standard "sndplay" program can play all NeXT
format files (this include Sun ".au" files).  It supports at least
U-LAW at 8 k and 16 bits samples at 22 or 44.1 k.  It attempts
on-the-fly conversions for other formats.

Sound files are also played if you double-click on them in the file
browser.

        SGI Indigo, Indigo2, Indy and Personal IRIS
        -------------------------------------------

On SGI Indigo, Indigo2, Indy and the 4D/30 and /35 Personal IRIS workstations,
"WorkSpace" plays audio files in .aiff, .aifc, .au, and .wav formats if
you double click them and the sampling rate is one of 8000, 11025,
16000, 22050, 32000, 44100, or 48000.  On the Personal IRIS, you need
to have the audio board installed (check the output from hinv) and you
must run IRIX 3.3.2 or 4.0 or higher.  These files can also be played
with "soundfiler" and "sfplay".  ".aiff" and ".aifc" files at the above
sampling rates can also be played with playaifc.  (All in /usr/sbin)

There is no simple /dev/audio interface on these SGI machines.  (There
was one on 4D/25 machines, reading and writing signed linear 8-bit
samples at rates of 8, 16 and 32 k.)

A program "playulaw" was posted as part of the "radio 2.0" release
that I posted to several source groups; it plays raw U-LAW files on
the Indigo, Indigo2, Indy or Personal IRIS audio hardware.

        Sony NEWS
        ---------

The whole current Sony NEWS line (laptop, desktop, server) have
builtin sound capabilities.  You can buy an external board for the
older NEWS machines.  In the default mode (8k/8-bit mulaw), Sun .au
files are directly supported (you can 'cat' .au files to /dev/sb0 and
have them play.)  The /usr/sony/bin/sbplay command on NEWS-OS 6.0
also supports Sun .au files.

        Others
        ------

Most other UNIX boxes don't have audio hardware and thus can't play
audio data.  This is actually rapidly changing and most new hardware
that hits the market has some form of audio support.  Unfortunately
there is no single portable interface for audio that comes near the
acceptance and functionality (let alone code size :-) of X11 for
graphics.  There are at least two network-transparent packages, both
in some way based on the X11 architecture, that attempt to fillo the
gap:

DEC CRL's AudioFile supports Digital RISC systems running Ultrix,
Digital Alpha AXP systems running OSF/1, Sun Sparcs, and SGI
AL-capable systems (e.g., Indigo, Indy).  The source kit is located at
ftp site crl.dec.com in /pub/DEC/AF.

NCD's NetAudio supports NCD's MCX line of X terminals as well as
Sparcs running either SunOS 4.1.3 or Solaris 2.2, using the /dev/audio
interface (they claim it should be easy to port).  The source it
located at ftp.x.org in contrib/netaudio.  It is also ported to SGI
(tested on IRIX 5.x), and there are unconfirmed rumors that it is
being ported to SCI and Linux.


Playing audio files on the Vaxstation 4000 (VMS)
------------------------------------------------

1) Without DECsound

".au" files can be played by COPYING them to device "SOA0:".  This
device is set up by enabling the driver SODRIVER.  You can use the
following command file:

$!---------------- cut here -------------------------------
$! sound_setup.com    enable SOUND driver
$ run sys$system:sysgen
connect soa0 /adapter=0 /csr=%x0e00 /vector=%o304 /driver=sodriver
exit
$ exit
$!----------------- cut here ------------------------------------

2) With DECsound (bundled with motif)

Just start DECsound by selecting it from the session manager in the
applications menu. (Not there use "@vue$library:sound$vue_startup").
Make sure settings; device type (vaxstation 4000) and play settings
(headphone jack) are selected.  To play files from the DCL prompt
(handy if you want to play sounds on a remote workstation) set a
symbol up as follows;
PLAY == "$DECSOUND -VOLUME 50 -PLAY"
usage;
DCL> play sound.au

3) Audio port

The external audio port comes with a telephone-jack-like port.   For
starters, you can plug a telephone RECEIVER right into this port to
hear your first sound files.   After that, you can use the adapter
(that came with the VaxStation), and plug in a small set of stereo
speakers or headphones (the kind you'd plug into a WALKMAN, for
example), for more volume.  The adapter also has a microphone plug so
that you can record sounds if DECsound is installed.


Playing audio files on micros
-----------------------------

Most micros have at least a speaker built in, so theoretically all you
need is the right software.  Unfortunately most systems don't come
bundled with sound-playing software, so there are many public domain
or shareware software packages, each with their own bugs and features.
Most separate sound recording hardware also comes with playing
software, most of which can play sound (in the file format used by
that hardware) even on machines that don't have that hardware
installed.

        PC or compatible
        ----------------

Chris S. Craig announces the following software for PCs:

ScopeTrax       This is a complete PC sound player/editor package.  Sounds
                can be played back at ANY rate between 1kHz to 65kHz through
                the PC speaker or the Sound Blaster.  It supports several
                file formats including VOC, IFF/8SVX, raw signed and raw
                unsigned.  A separate executable is provided to convert
                .au and mu-law to raw format.  ScopeTrax requires EGA/VGA
                graphics for editing and displaying sounds on a REALTIME
                oscilloscope.  The package also includes:
                      * An expanded memory player which can play sounds
                        larger than 640K in size.
                      * Basic (rough) sound compression/uncompression
                        utilities.
                      * Complete documentation.
                The package is FREEWARE!  It is available on SIMTEL in the
                PD1:[MSDOS.SOUND] directory.

One of the appendices below contains a list of more programs to play
sound on the PC.

        Atari
        -----

For sounds on Atari STs - programs are in the atari/sound/players
directory on atari.archive.umich.edu.

        Tandy
        -----

On a Tandy 1000 or 2500, sounds can be played and recorded with DeskMate
Sound (SOUND.PDM), or if they are not stored in compressed format, they can
also be played by a program called PLAYSND.  Playsnd also plays .voc, .wav,
.iff, .mod samples, and headerless 8-bit PCM (signed or unsigned).  The
author, John Ball ([e-mail]) has decided to place the program
and source code in the public domain.  Playsnd will also play on the PC
speaker.  Also, Tspak (see above) contains programs to record and play
.wav files.

        Amiga
        -----

On the Amiga, OmniPlay by David Champion <[e-mail]>
plays and converts IFF-8SVX, AIFF, WAV, VOC, .au, .snd, and 8 bit raw
(signed, unsigned, u-law) samples.  As of version 1.23, OmniPlay will
also convert any playable sample to 8SVX.  Files: wuarchive.wustl.edu
in /systems/amiga/audio/sampleplayers/oplay123.lha (?)
amiga.physik.unizh.ch in mus/play/oplay123.lha

        Apple Macintosh
        ---------------

Malcolm Slaney from Apple writes:

 "We do have tools to play sound back on most of our Unix hosts.  We wrote
 a program called TcpPlay that lets us read a sound file on a Unix host,
 open a TCP/IP connection to the Mac on my desk, and plays the file.  We
 think of it as X windows for sound (at least a step in that direction.)

 This software is available for anonymous FTP from ftp.apple.com.
 Look for  ~ftp/pub/TcpPlay/TcpPlay.sit.hqx.

 Finally, there are MANY tools for working with sound on the Macintosh. Three
 applications that come to mind immediately are SoundEdit (formerly by
 Farralon and now by MacroMind/Paracomp), Alchemy and Eric Keller's Signalyze.
 There are lots of other tools available for sound editing (including some
 of the QuickTime Movie tools.)"

Bill Houle sent the following lists:

Popular commercial apps are indicated with a [*].  All other programs
mentioned are shareware/freeware available from SUMEX and the various
mirror sites, or check archie for the nearest FTP location.

MAC SOUND EDITORS

Sample Editor [Garrick McFarlane, [e-mail]]
  Plays AIFF and 'snd' sounds.  Can convert between AIFF and 'snd'.
  Can record from built-in mic.  Can add effects such as fade,
  normalize, delay, etc.

Wavicle [Lee Fyock]
  Plays SoundEdit files.  Can convert to 'snd'.  Can record from built-in mic.
  Can add effects such as fade, filter, reverb, etc.

[*]SoundEdit/SoundEdit Pro [Farallon/MacroMind*Paracomp]
  Plays SoundEdit and 'snd' sounds.  Can read/write SoundEdit files and 'snd'
  sounds.  Can record from built-in mic.  Can add effects such as
  echo, filter, reverb, etc.


MAC SOUND PLAYERS

Sound-Tracker  [Frank Seide]
  Plays Amiga SoundTracker files in foreground or background.

Macintosh Tracker [Thomas R. Lawrance, [e-mail]]
  Plays Amiga SoundTracker files in foreground or background. A port of Marc
  Espie's Unix Tracker version with Frank Seide's core player thrown in for
  good measure.

The Player [Antoine Rosset & Mike Venturi]
  Plays AIFF, SoundEdit, MOD, and 'snd' files.

SoundMaster (aka [*]Kaboom!) [Bruce Tomlin]
  Associates SoundEdit files to MacOS events.

SndControl [Riccardo Ettore, [e-mail]]
  Associates 'snd' sounds to MacOS events.

Canon 2 [Glenn Anderson, [e-mail]; Jeff Home, [e-mail]]
  Plays AIFF or 'snd' files in foreground or background.

Another Mac play/convert program: "It's called SoundApp. I wrote it,
([e-mail]) and it's FreeWare. It will play: SoundCap,
SoundEdit, WAVE, VOC, MOD, Amiga IFF (8SVX), Sound Designer, AIFF, AU,
Mac Resource, and DVI ADPCM. It can convert all the above to System 7
sound resources (except MOD where just the samples are extracted.) And
it will double buffer."


The Sound Site Newsletter
-------------------------

An electronic publication with lots of info about digitised sound and
sound formats, albeit mostly on PCs, is "The Sound Site Newsletter",
maintained by David Komatsu <[e-mail]> (this is a
temporary account until January 1995).  Issue 20 appeared in September
1994.  The Sound Site Newsletter (once again!) has its own ftp site:
sound.usach.cl.

The Sound Newsletter is posted to: comp.sys.ibm.pc.soundcard
                                   comp.sys.ibm.pc.misc
                                   rec.games.misc
FTP: oak.oakland.edu (misc/sound)
     garbo.uwasa.fi (pc/sound)
     sound.usach.cl (pub/Sound/Newsltr) [Home Base]


Posting sounds
--------------

The newsgroup alt.binaries.sounds.misc is dedicated to postings
containing sound.  (Discussions related to such postings belong in
alt.binaries.sounds.d.)

There is no set standard for posting sounds; uuencoded files in most
popular formats are welcome, if split in parts under 50 kBytes.  To
accomodate automatic decoding software (such as the ":decode" command
of the nn newsreader), please place a part indicator of the form
(mm/nn) at the end of your subject meaning this is number mm of a
total of nn part.

It is recommended to post sounds in the format that was used for the
original recording; conversions to other formats often lose
information and would do people with identical hardware as the poster
no favor.  For instance, convering 8-bit linear sound to U-LAW loses
the lower few bits of the data, and rate changing conversions almost
always add noise.  Converting from U-LAW to linear requires expansion
to 16 bit samples if no information loss is allowed!

U-LAW data is best posted with a NeXT/Sun header.

If you have to post a file in a headerless format (usually 8-bit
linear, like ".snd"), please add a description giving at least the
sampling rate and whether the bytes are signed (zero at 0) or unsigned
(zero at 0200).  However, it is highly recommended to add a header
that indicates the sampling rate and encoding scheme; if necessary you
can use SOX to add a header of your choice to raw data.

Compression of sound files usually isn't worth it; the standard
"compress" algorithm doesn't save much when applied to sound data
(typically at most 10-20 percent), and compression algorithms
specifically designed for sound (e.g. NeXT's) are usually
proprietary.  (See also the section "Compression schemes" earlier.)


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
The Sample Vision Format
Some Miscellaneous Formats
Tandy Deskmate .snd Format Notes


------------------------------------------------------------------------
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; 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 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 assume that this header file comes with
Apple programming products like MPW [C|C++] as well.  I found this out a
little too late: I had already coded my own structures.  These structures
of mine, along with other useful code for AIFF-based DSP in C, are
available for ftp at ftp.cs.jhu.edu in pub/dsp.

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, 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.  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]

Detailed description of new data blocks (VOC files version 1.20 and above):

        (Source is fax from Barry Boone at Creative Labs, 405/742-6622)

BLOCK 8 - digitized sound attribute extension, must preceed block 1.
          Used to define stereo, 8 bit audio
        BYTE bBlockID;       // = 8
        BYTE nBlockLen[3];   // 3 byte length
        WORD wTimeConstant;  // time constant = same as block 1
        BYTE bPackMethod;    // same as in block 1
        BYTE bVoiceMode;     // 0-mono, 1-stereo

        Data is stored left, right

BLOCK 9 - data block that supersedes blocks 1 and 8.
          Used for stereo, 16 bit.

        BYTE bBlockID;          // = 9
        BYTE nBlockLen[3];      // length 12 plus length of sound
        DWORD dwSamplesPerSec;  // samples per second, not time const.
        BYTE bBitsPerSample;    // e.g., 8 or 16
        BYTE bChannels;         // 1 for mono, 2 for stereo
        WORD wFormat;           // see below
        BYTE reserved[4];       // pad to make block w/o data
                                // have a size of 16 bytes

        Valid values of wFormat are:

                0x0000  8-bit unsigned PCM
                0x0001  Creative 8-bit to 4-bit ADPCM
                0x0002  Creative 8-bit to 3-bit ADPCM
                0x0003  Creative 8-bit to 2-bit ADPCM
                0x0004  16-bit signed PCM
                0x0006  CCITT a-Law
                0x0007  CCITT u-Law
                0x02000 Creative 16-bit to 4-bit ADPCM

        Data is stored left, right

------------------------------------------------------------------------
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.

The complete definition of the WAVE file format as defined by IBM and
Microsoft is available for anonymous FTP from ftp.microsoft.com, in
directory developer/MSDN/CD8 as file RIFFNE.ZIP, which contains a MS
help file (riffne.hlp).

Mark Stout <[e-mail]> clarifies: RIFFNE.HLP,
Multimedia Standards Update: New Multimedia Data Types and Data
Techniques 2.1.0, has only extensions onto the original Multimedia
Programming Interface and Data Specifications 1.0, which Bob Ryan has
made an excerpt from.  Most people only need the original spec (Bob
Ryan's excerpt).  However, for information on most compressed audio
formats, they should obtain RIFFNE.HLP.

Conor Frederick Prischmann <[e-mail]> points to two more
sites:

(1)     Take a look at ftp site : teeri.ouli.fi
               in the directory : /pub/msdos/programming/*
        it has some sub dirs like specs, utils and most importantly
        gpe.  Take that file and you know everything.

(2)     ftp.ircam.fr:/pub/music

------------------------------------------------------------------------
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/amiga/music/ntsb ?????)  that should do you for
a while

see you soon

Chris.

-----------------------------------------------------------------------
The Sample Vision Format
------------------------

From: "[e-mail]" <[e-mail]>

First, Sample Vision is a program used by professional musicians to
send and receive samples via a MIDI interface to the PC. While on the
PC, you can edit several parameters including loop points, pitch, time
compression, normalize, sample rate, ect.  The list of supported
samplers include: AKAI {S700,X700,S900, S950,S612,S1000/1100},
Casio{FZ1,FZ10M,FZ20M}, Ensoniq{EPS,EPS16,ASR10,Mirage},
Emu{Emax,EmaxII}, Korg{DSS1,DSM1,T workstation}, Oberheim DPX-1,
Peavey DPM-3, Roland {S10,MKS100,S220,S50,S330,S550}, Sequential
Circuits Prophet 2000/2002, Sample Dump Standard devices, Yamaha
TX16W.

The .smp format breaks down like this:

Offset     Size        Description
000        18          'SOUND SAMPLE DATA ' ASCII FILE ID
0018       04          '2.1 '   ASCII FILE VERSION
0022       60          USER COMMENTS    60 ASCII CHARACTERS
0082       30          SAMPLE NAME    LEFT JUSTIFIED 30 ASCII CHARACTERS
0112       04          SAMPLE SIZE   SAMPLE DATA COUNT IN WORDS
0116       ??          SAMPLE DATA  1 WORD PER SAMPLE, LEAST SIGNIFICANT BYTE
                                    FIRST, LSW FIRST; SIGNED 16 BIT INTEGERS

??         02(DW)      RESERVED
??         04(DD)      LOOP 1 START  USE SAMPLE COUNT NOT BYTE COUNT
??         04(DD)      LOOP 1 END
??         01(DB)      LOOP 1 TYPE   0=LOOP OFF,1=FORWARD,2=FORWARD/BACKWARD
??         02(DW)      LOOP 1 COUNT  TIMES TO EXECUTE LOOP BEFORE NEXT LOOP

THERE ARE SEVEN MORE IDENTICAL LOOP STRUCTURES FOR A TOTAL OF 8

??         10          MARKER 1 NAME ASCII MARKER NAME
??         04(DD)      MARKER 1 POSITION  FFFF MEANS UNUSED

THER ARE SEVEN MORE IDENTICAL MARKER STRUCTURES FOR A TOTAL OF 8

??         01(DB)       MIDI UNITY PLAYBACK NOTE         MIDI NOTE TO PLAY
                                                         THE SAMPLE AT ITS
                                                         ORIGINAL PITCH
??         04(DD)       SAMPLE RATE IN HERTZ
??         04(DD)       SMPTE OFFSET IN SUBFRAMES
??         04(DD)       CYCLE SIZE         SAMPLE COUNT IN ONE CYCLE OF
                                           THE SAMPLED SOUND. -1 IF UNKNOWN

(DD) 4 BYTES, LS BYTE FIRST, LS WORD FIRST
(DW) 2 BYTES, LS BYTE FIRST
(DB) 1 BYTE

That's about it. One thing I have noticed is that Sample Vision only
writes seven loop structures to file as opposed to the eight
structures it claims are written.
-----------------------------------------------------------------------
Some Miscellaneous Formats
--------------------------

From: [e-mail] (Bill Schottstaedt)

I thought you might find some of this information amusing -- a few
header formats I didn't find in your great audio file formats
documentation.  Some taken from the AFsp sources, or sox, or
local ancient documentation.  I also have short descriptions
of BICSF, NeXT/Sun, AIFF, RIFF, SMP, VOC, and so on, plus
full descriptions of the 2 Sound Designer formats, if you're
interested.


/* ------------------------------------ NIST ---------------------------------

 *

 *   0: "NIST_1A"
 *   8: data_location as ASCII representation of integer
 *      (apparently always "   1024")
 *  16: start of complicated header -- full details available upon request
 *
 *  here's an example:
 *
 *  NIST_1A
 *     1024
 *  database_id -s5 TIMIT
 *  database_version -s3 1.0
 *  utterance_id -s8 aks0_sa1
 *  channel_count -i 1
 *  sample_count -i 63488
 *  sample_rate -i 16000
 *  sample_min -i -6967
 *  sample_max -i 7710
 *  sample_n_bytes -i 2
 *  sample_byte_format -s2 01
 *  sample_sig_bits -i 16
 *  end_head
 */
/* ------------------------------------ SNDT ---------------------------------
 *
 * this taken from sndrtool.c (sox-10):
 *   0: "SOUND"
 *   6: 0x1a
 *   8-11: 0
 *  12-15: nsamples
 *  16-19: 0
 *  20-23: nsamples
 *  24-25: srate
 *  26-27: 0
 *  28-29: 10
 *  30-31: 4
 *  32-> : <filename> "- File created by Sound Exchange"
 *  .->95: 0
 */
/* ------------------------------------ ESPS ---------------------------------

 *
 *   16: 0x00006a1a or 0x1a6a0000
 *  136: if not 0, chans + format = 32-bit float
 *  144: if not 0, chans + format = 16-bit linear
 *

 *   from AFgetInfoES.c:
 *

 *       Bytes     Type    Contents
 *      8 -> 11    --     Header size (bytes)
 *     12 -> 15    int    Sampled data record size
 *     16 -> 19    int    File identifier
 *     40 -> 65    char   File creation date
 *    124 -> 127   int    Number of samples (may indicate zero)
 *    132 -> 135   int    Number of doubles in a data record
 *    136 -> 139   int    Number of floats in a data record
 *    140 -> 143   int    Number of longs in a data record
 *    144 -> 147   int    Number of shorts in a data record
 *    148 -> 151   int    Number of chars in a data record
 *    160 -> 167   char   User name
 *    333 -> H-1   --     Generic header items, including "record_freq"
 *                        {followed by a "double8"}
 *      H -> ...   --     Audio data
 */
/* ------------------------------------ INRS ---------------------------------

 *

 *   from AFgetInfoIN.c:
 *

 *    INRS-Telecommunications audio file:
 *       Bytes     Type    Contents
 *      0 ->  3    float  Sampling Frequency (VAX float format)
 *      6 -> 25    char   Creation time (e.g. Jun 12 16:52:50 1990)
 *     26 -> 29    int    Number of speech samples in the file
 *   The data in an INRS-Telecommunications audio file is in 16-bit integer
 *   format.
 *

 */
/* old Mus10, SAM formats, just for completeness
 *
 * These were used for sound data on the PDP-10s at SAIL and CCRMA in the
 * 70's and 80's.
 * The word length was 36-bits.
 *
 * "New" format as used by nearly all CCRMA software pre-1990:
 *
 *  WD 0 - '525252525252
 *  WD 1 - Clock rate in Hz (PDP-10 36-bit floating point)
 *  WD 2 - #samples per word,,pack-code
 *      (has # samples per word in LH, pack-code in RH)
 *      0 for 12-bit fixed point
 *      1 for 18-bit fixed point
 *      2 for  9-bit floating point incremental
 *      3 for 36-bit floating point
 *      4 for 16-bit sambox fixed point, right justified
 *      5 for 20-bit sambox fixed point
 *      6 for 20-bit right-adjusted fixed point (sambox SAT format)
 *      7 for 16-bit fixed point, left justified
 *      N>9 for N bit bytes in ILDB format
 *  WD 3 - # channels
 *      1 for MONO
 *      2 for STEREO
 *      4 for QUAD
 *  WD 4 - Maximum amplitude (if known)
 *      is a floating point number
 *      is zero if not known
 *      is maximum magnitude (abs value) of signal
 *  WD 5        number of Sambox ticks per pass
 *              (inverse of Sambox clock rate, sort of)
 *  WD 6 - Total #samples in file.
 *         If 0 then #wds_in_file*#samps_per_wd assumed.
 *  WD 7 - Block size (if any). 0 means sound is not blocked.
 *  WDs '10-'77 Reserved for EDSND usage
 *  WDs '100-'177 Text description of file (in ASCIZ format)
 *
 *
 * "Old" format
 *
 *  WD 0 - '525252525252
 *  WD 1 - Clock rate
 *      has code in LH, actual INTEGER rate in RH
 *      code=0 for 6.4Kc (or anything else)
 *          =1 for 12.8Kc, =2 for 25.6Kc, =3 for 51.2Kc
 *          =5 for 102.4Kc, =6 for 204.8Kc
 *  WD 2 - pack
 *      0 for 12 bit
 *      1 for 16 bit (18 bit)
 *      2 for 9 bit floating point incremental
 *      3 for 36-bit floating point
 *      N>9 for N bit bytes in ILDB format
 *      has # samples per word in LH.
 *  WD 3 - # channels
 *      1 for MONO
 *      2 for STEREO
 *      4 for QUAD
 *  WD 4 - Maximum amplitude (if known)
 *      is a floating point number
 *      is zero if not known
 *      is maximum magnitude (abs value) of signal
 *  WDs 5-77 Reserved for future expansion
 *  WDs 100-177 Text description of file (in ASCIZ format)
 */

-----------------------------------------------------------------------
Tandy Deskmate .snd Format Notes
--------------------------------

From: Jeffrey L. Hayes <[e-mail]>

Tandy .snd files are created by Sound.pdm, a program that came with the
proprietary DeskMate environment.  They are used by Music.pdm to create
music modules (.sng files).  DeskMate Sound and Music require the Tandy
sound chip.  There is a program to convert RIFF WAVE and other 8-bit PCM
formats to .snd, Conv2snd, by Kenneth Udut.  Conv2snd v.2.00 comes with
Snd2wav, which converts .snd to RIFF WAVE.

There are two types of DeskMate .snd files, sound files and instrument
files.  Both contain 8-bit unsigned PCM samples.

Sound files are simpler.  These are garden-variety sample files with a
fixed-length header giving the name of the sound, the recording frequency,
and the length of the sound.  Sound files may be recorded at 5500Hz, 11kHz
or 22kHz.

Instrument files contain samples as well as frequency and looping
information used by Music.pdm to represent an instrument.  Instrument files
provide for attack, sustain, and decay with several samples having
different implied frequencies and being used by Music.pdm to represent the
instrument in different pitch ranges.  Up to 16 different notes (with 16
different samples) can be contained in one instrument file.  Instrument
files are always recorded at 11kHz.  Both sound files and instrument files
may be compressed in one of two ways, "music" compression or "speech"
compression, or they may be uncompressed.  I don't know the compression
algorithms, but simple file comparison reveals that "music" and "speech"
compression are almost identical.

The DeskMate .snd file header consists of 16 bytes of fixed header
information followed by one or more 28-byte note records.  The sample
information, which may be compressed, follows the header.

DeskMate .snd File Format - Fixed Header
----------------------------------------

  offset    size      what
  ------    ----      ----

  0         byte      1Ah (.snd ID byte)

  1         byte      Compression code:  0 = no compression; 1 = music
                      compression; 2 = sound compression.

  2         byte      Number of notes in the instrument file.  1 if sound
                      file.

  3         byte      Instrument number.  0 if sound file; 0FFh if instrument
                      file with no number set.  Valid instrument numbers in
                      an instrument file are 1 to 32.  Use this field to
                      distinguish a sound file from an instrument file.

  4         10 bytes  Sound or instrument name.  Filled on the right with
                      nulls if less than 10 characters.

  0Eh       word      Sampling rate in samples per second.  Note that although
                      a sampling rate other than 5500, 11000 and 22000 can be
                      entered here, Sound.pdm will not actually play at other
                      rates.

  10h       variable  Note records begin, 28 bytes each.  Number of records
                      given in byte 2 above.


DeskMate .snd File Format - Note Record
---------------------------------------

  0         byte      Pitch of the note:  1 = A1 in American Standard Pitch;
                      2 = A#1; etc.  A1 is lowest note allowed; highest note
                      allowed is B6 (3Fh).  Sound files have 0FFh here; so do
                      instrument files with no note set.
                          Note that Sound.pdm does not designate notes in the
                      standard manner to the user.  Although A1 and B6 in
                      Sound.pdm are the same as A1 and B6 in standard pitch,
                      Sound.pdm starts octaves at A rather than at C (as is
                      standard).  Thus, middle C, C4 in standard pitch, is C3
                      in Sound.pdm.

  1         byte      Sound files, and instrument files with no pitch set,
                      have 0 here.  If the pitch is set, this byte is 0FFh.

  2         2 bytes   Range of the note, first byte is lower limit, second
                      is higher limit.  Byte encoding as for offset 0 (i.e.,
                      01h to 3Fh).  Sound files have FF FF here; so do
                      instrument files with no range set.

  4         dword     Offset in the file where samples for this note begin
                      (zero-relative), after compression if that was done.

  8         dword     If compressed, the length of the compressed data in the
                      file for this note.  Uncompressed files have 0 here.

  0Ch       4 bytes   Unknown.  Set to zero.

  10h       dword     Number of samples in the note, after decompression if
                      necessary.

  14h       dword     Number of sample at start of sustain region for the
                      note, relative to the first (zeroth) sample of the note.
                      For sound files, or if sustain is not set, this field is
                      0.

  18h       dword     Number of sample at end of sustain region for the note,
                      relative to the first (zeroth) sample of the note.  For
                      sound files, or if sustain is not set, this field is 0.


New Tandy .Snd File Format
--------------------------

This is the new .snd file format used on the 2500-series.  From information
provided by John Ball ([e-mail]).

Like the old format, the new format header consists of a fixed part
followed by one or more sample descriptors.  The fixed part is 114 bytes;
the sample descriptors are 46 bytes each.  Samples are still 8-bit unsigned
PCM.

Fixed header:
    offset       size        what
      0          10 bytes    ASCIIZ name of sound.
      0Ah        34 bytes    unknown
      2Ch        2 bytes     New .snd ID:  1Ah 80h.
      2Eh        word        Number of samples in file.
      30h        word        Sound (instrument) number.
      32h        16 bytes    unknown
      42h        word        Compression code (0 = no compression, 1 =
                             music compression, 2 = speech compression).
      44h        20 bytes    unknown
      58h        word        Sampling rate in Hz.
      5Ah        24 bytes    unknown
      72h        variable    Sample descriptors begin.

Sample descriptors (number given by word at 2Eh above):
    offset       size        what
      0          dword       Link to next sample descriptor (offset in file
                             of next sample descriptor record).  0 if last.
      4          2 bytes     unknown
      6          byte        Pitch of note (01h-3Fh), 01 = A1 in American
                             Standard Pitch; 0FFh if not set.
      7          byte        unknown (compare old .Snd format; value is 00
                             or FF, but seemingly unrelated to pitch setting)
      8          2 bytes     Range of note.  First byte is lower limit,
                             second is higher limit.  Values as for byte
                             at offset 6 above; FF FFh if not set.
      0Ah        dword       Offset in file of start of sound data for
                             this sample.
      0Eh        dword       Length of sample sound data in bytes.
      12h        dword       Uncompressed length of sound data (number of
                             samples).
      16h        24 bytes    unknown
------------------------------------------------------------------------