********************** (******CONV2SND.DOC******) *********************** ********************** * *********************** ** CONV2SND.PAS: 2.02* * *********************** ********************** * *********************** *** February 3, 1993** * *********************** ***** Kenneth Udut**** * *********************** *** Revised 10/15/96** * *********************** ****** J.L. Hayes***** * *********************** ********************** * ********************** (******CONV2SND.DOC******)      CONV2SND.EXE, CONV2SND.PAS, CONV2SND.DOC, CONV2SND.HST HEREBY ARE RELEASED INTO THE PUBLIC DOMAIN.* UNLESS SPECIAL CIRCUMSTANCES WARRANT OTHERWISE, EVERY SOFTWARE PRODUCT CREATED BY (_KENNETH UDUT_) WILL:  1. BE RELEASED INTO THE PUBLIC DOMAIN, FREELY COPYABLE AND CHANGEABLE.  2. INCLUDE THE SOURCE CODE.  AS THE AUTHOR OF (_CONV2SND_), I WOULDN'T MIND CREDIT FOR MY WORK. * HOWEVER, IT IS *NOT* A NECESSITY, SO DON'T WORRY ABOUT IT! :)  [Note: My modifications are also public domain.* Ken wrote much of the  following
, but I have changed a lot of it to match this version. - J.L.  Hayes]  Now enough of this screaming!    1.* CONV2SND is a program with one thing in mind:* To open the worlds *** of digitized sounds created on other machines and by other methods *** to anyone using a Tandy 1000 SL, TL, RL, RLX, 2500 RLX, RSX and *** every other machine that uses:  *** SOUND.PDM  *** Ken wrote: "I wish that Tandy had created something a few years ago  *** that would have done this, as it might have kept the Tandy's with the  *** superb 8-bit DAC a little more in step with the SoundBlaster, the  *** Macintosh, the Amiga and the Atari ST, all of which are able to create  *** superb digitized sounds and have programs that allow you to share them  *** with one another.  *** "So, here it is."   2.* To start the program, type:  ******* CONV2SND <infile>  *** or:  ******* CONV2SND <infile> <outfile>  *** The process will then take place (almost) automatically.* Filenames can  *** include drive and path if you want.* If you use the f
irst form, the  *** output filename will be the same as the input filename, including drive  *** and path if present, but with an ".snd" extension attached.* If you use  *** the second form, the output filename will be what you specified.* If  *** the output filename does not have an extension, it defaults to ".snd". * *** There is no default for the input file, so you have to specify the  *** extension for it.* If you want the output file not to have an  *** extension, end its name with a period.  *** As of version 2.00, CONV2SND can read and understand .wav headers.* The  *** .wav will be detected by the header, not by the filename.* CONV2SND  *** will check that the .wav is an 8-bit mono file and display some advice  *** if it isn't (see below).* If the .wav is 8-bit mono, CONV2SND will read  *** the sampling rate from the header and check to see if it is within 5%  *** of one of the three rates supported by SOUND.PDM.* If so, the sampling  *** rate will be taken from the .wav header, and you won't be ask
ed about  *** that.* If not, you will be told to resample the file.* There is a  *** program available to do this, discussed below.* If the .wav header is  *** invalid, you will be given the option of continuing as if the file were  *** not a .wav.* I don't know how useful that will be.  *** You will be asked a couple of questions.  *** A) Asks about the sampling rate (speed of playback).* You won't be  ****** asked this question if CONV2SND is able to determine the sampling  ****** rate from the .wav header.  ****** These are the options (Ken's notes):  ****** 5500 is a slow sampling rate, often used for speech purposes or some *********** games.* [5500 seems to be rare nowadays. - JLH]  ****** 11000 is the the most common sampling rate.* It is #2 on the  ************ selection list and the one you will want to try first.* The  ************ only reason it is so popular is that it is of sufficient  ************ quality for music and such.* 5500 really isn't.* WAV files are  ************ usually sampled at 11
000 samples per second.  ****** 22000 is most often used with the Macintosh Sound files.* It is of a  ************ higher quality, but it does take up more disk space, which is  ************ why it is the 2nd most popular sampling rate to record at. * ************ [This rate is becoming more common. - JLH]  ****** If you screw up and select the wrong one, that's okay!* Just redo the ****** procedure and select a different rate next time!  ****** [It should be mentioned that 44000, or 44kHz, is an increasingly  ****** common rate.* This is the rate that audio CD's are sampled at.* The  ****** Tandy sound chip is perfectly capable of going that fast, but  ****** SOUND.PDM does not support that speed.* I have written a .wav player  ****** for the Tandy chip that will play .wav's at 44kHz - see Snd2wav.doc. * ****** Included here with CONV2SND is a utility called Ppwav that can cut  ****** the sampling rate of these .wav's down to 22kHz so that CONV2SND can  ****** convert them.* See below. - JLH]  *** B) Asks f
or the "name" of the sound.  ****** This isn't the file name.* This is the name that appears when you  ****** are in SOUND.PDM, right next to "Name: ", in the right hand box.* If  ****** you don't specify a name, none will be used.* This is a change from  ****** version 1.98, which required you to enter a name.* It is a good idea  ****** to specify a name anyway, in case the name of the file gets changed  ****** at some point in the future.  ****** You can read the name of the sound by typing DEBUG filename.snd and  ****** then "d".* You will see the name of the sound file in the right hand  ****** side of the screen.* (NOTE: "q" gets you out of DEBUG)  *** C) Asks whether you want to convert to the new DeskMate .snd file  ****** format.* The old format is native to the 1988 version of SOUND.PDM;  ****** the new format is native to the 1989 version.* Any version of  ****** SOUND.PDM can read the old format, so it is a good idea to convert  ****** to that one unless you have a reason not to.* In other words,
you  ****** probably want to say "no" to this question.  ****** Programs using Frank Durda's PSSJ Digital Sound Toolkit (not from  ****** Tandy) do not understand the old format, so you need to convert to  ****** the new format for them.  *** D) Asks for your name.* "No real reason yet, but I'm thinking of one!"  ****** writes Ken.* Version 1.98 required you to enter your name; this  ****** version will call you "CONV2SND user" if you don't enter a name.   *** A new file with your sound will be created.* This is the DeskMate SOUND  *** version of the other digitized file.   3.* You will need to do the following after the file is created:   *** * Go into SOUND.PDM, and load the new .SND file (F2, Open)  *** * Play the file and see if the conversion was done right.* If the file ***** sounds like Alvin the Chipmunk, the sampling rate you gave was too  ***** low; go back and run CONV2SND again with a higher rate.* If it sounds  ***** like Lurch on "The Addams Family," the rate is too high, and you need  ***** to
try a lower one.* (That is, unless the file *is* a recording of  ***** Lurch! :-)* If the file sounds like nothing at all, just static, it  ***** may not be an 8-bit mono PCM file.* You will have to use a program  ***** called Sox to convert it to one before CONV2SND can use it; see  ***** below.  ***** If the input file was a .wav, you should not have this problem.* If  ***** the .wav is of a type that can't be converted (most can), CONV2SND  ***** will tell you why and halt without creating an output file.  *** * If the sampling rate seems right, carefully remove any noise at the  ***** beginning of the file, using a very small increment (use "sample"). * ***** Beginning noise, if present, represents the header of the original  ***** sound file; it is not actual sound.* There are a few types of sound  ***** files that don't have a header ("raw" files made by Sox are one - see  ***** below), but most do.* The noise will be a short click or pop.  ***** Again, if the input file was a .wav, the .wav header wi
ll be deleted  ***** automatically, and this problem will not arise.  *** * Hit F2 and Save the file.  ***** Example: ROCKY.WAV is converted into ROCKY.SND ******** After listening to the file to check the conversion and getting  ******** rid of the little beginning noise, hit F2, and Save.* Exit  ******** SOUND.PDM if you don't want to do more with the file at that time.   Here are some problems you may have, and what to do about them:  *** * If the file sounds like nothing at all, just static, it may not be an  ***** 8-bit unsigned mono PCM file.* That means the sound data in the file  ***** is not the right type for SOUND.PDM, and it will have to be converted  ***** to 8-bit unsigned mono PCM before CONV2SND can use it.* There is a  ***** program called Sox that can tell you what kind of file it is and do  ***** the conversion.* As of this writing (8/15/1996), Sox is available by  ***** ftp from:  ******* oak.oakland.edu  ***** Get file:  ******* /pub/simtelnet/msdos/sound/sox7dos.zip  ***** and read the
documentation that comes with Sox.* You will need to  ***** tell Sox to make its output file a RIFF WAVE file, size "byte," "1  ***** channel" and select a sampling rate of 5500, 11000, or 22000.* Use  ***** the "verbose" option with the "stat" effect to find out what the  ***** sampling rate of the original file is; if it's close to one of those  ***** three, leave it alone.* CONV2SND will accept sampling rates within 5%  ***** of 5500, 11000, or 22000.* It is best not to resample if you don't  ***** have to since it introduces noise.* Warning:* Sox is a little  ***** confusing to use at first.  ***** As an alternative to Sox, this distribution includes a utility called  ***** Ppwav that operates only on .wav files.* Ppwav can convert wav's with  ***** more than one channel to mono and larger sample sizes to 8-bit.* It  ***** can also cut the sampling rate of a file in half.* Sox does all that  ***** Ppwav does and more, but Ppwav is much faster and easier to use.* If  ***** you have a .wav that can't be co
nverted because of one of the  ***** problems that Ppwav can solve, CONV2SND will refuse to convert the  ***** file and tell you to use Ppwav.* See the separate documentation file  ***** on Ppwav.* Note that there are rare types of .wav's that neither Sox  ***** nor Ppwav can convert.  *** * If you've tried all 3 sampling rates, 5500, 11000, and 22000, and  ***** none of them seems to be right, you will have to resample the sound  ***** to change the sampling rate to one of those three.* SOUND.PDM can  ***** only use one of those 3 rates.* Sox (see above) can be used to do  ***** this.* Changing the sampling rate will introduce noise, unfortun-  ***** ately; change to a sampling rate that is an integer multiple of the  ***** original to minimize the noise.  ***** Other sampling rates that I have seen are 8000, 15000, and 44000. * ***** Sun .au files are usually sampled at 8000 (note that Sun .au is not  ***** PCM but mu-law - Sox can convert these to .wav).* If the input file  ***** is a .wav, CONV2SND will
refuse to convert the file if the header  ***** indicates that the sampling rate is not one of the three that  ***** SOUND.PDM supports.  *** * If CONV2SND tells you the disk where the output file goes is full,  ***** you will need to specify another drive for the output file, one with  ***** more space on it.* If you have a single-floppy system, you can  ***** specify drive B:, and DOS will prompt you to swap disks when needed. * ***** It will be awkward to do it this way, but it will work.  *** * If SOUND.PDM tells you the .snd file won't fit in memory, there is  ***** unfortunately not much you can do.* If your system can take  ***** additional DOS memory, you can install additional memory if you  ***** haven't already.* This is a very cheap upgrade and highly  ***** recommended; check your owner's manual to find out how.* I'm  ***** referring to systems with less than 640k of DOS memory; I doubt that  ***** SOUND.PDM will use expanded or extended.   That's it!* Ken's phone (908) 241-6246 is open for your
use up 'till about  11pm or so.* He is Kenneth Udut, 170 East Clay Avenue, Roselle Park, NJ USA  07204-2050.    Note:* Ken hasn't worked  with this program in a while  and doesn't seem to be on the  Internet any more. * -JL Hayes, Oct 15, 1996  I'm still here, Jeff. :P -K Udut, March 23, 2015 *  I am responsible for versions 2.00 and above,  including much of the above text.* My phone is (207) 866-7903.* I'm  [email protected] on the Internet.* Snailmail: Jeffrey L. Hayes, 130 Forest  Ave., Lot 1, Orono, Maine 04473.  "Thanks for reading and hope you NOT ONLY enjoy using it, but also play  around with the source code and work with it! *  "I want to see a world where source code is freely distributable, where  there isn't all that "data-hiding" and all that.* I would hate to see a  society where the 'elite programmers' only knew the secrets of 'real'  programming, while the peons were kept down low...  "I prefer openness and honesty, personally :)  "Talk to you soon!  "--Ken  "February 3, 1993"   FIDONET: Kenn
eth Udut @ 1:107/900 INTERNET: [email protected] America Online: K Udut PC-Link: K Udut CompuServe: INTERNET> [email protected] BITNET: kudut%[email protected]  Home Address: Kenneth Udut 170 East Clay Avenue Roselle Park, New Jersey 07204-2050  Phone Numbers: (908) 241-6246 (908) 241-8528* --* FAX number  Can be found on: The Juice Bar BBS, located in Roselle Park, New Jersey, USA (908) 298-0764.* I'm user #11, and a VERY VERY active poster!  I also run a list on BITNET.  [email protected] - discussion on the rights of kids and teens.   "Leave me a message somewhere or give me a phone call!* I'll set a pot of  tea out for you and we'll chat :)" - Ken   Also check out the Tandy sound package!* Play mono or stereo 8- or 16-bit  .wav's of arbitrary size at up to 65535 Hz, record .wav's of arbitrary  length at up to 62kHz, convert raw 8-bit sound files to .wav, convert 16-  bit or stereo .wav's to mono 8-bit, convert .snd instrument files to Amiga  .mod samples (including looped samples!
), use your Tandy DAC with programs  that support a generic DAC, and play your Tandy like a piano with the 3-  voice chip (chords too!).* All source code included, and the best, most  complete programming information on the Tandy sound that you will find  anywhere!* All free!* How do I get this marvellous package, you ask?* Just  ftp it from:  * ftp.agate.net:/users/01935/tandy1000/sound/tspak17.zip  - JLH     ********** -----------[technical stuff]------------     Many (but not all) types of sound files are basically identical to DeskMate  .snd.* That is, the actual sound data is encoded in exactly the same way. * The difference lies in the "header."  The header of a file is a small series of bytes, usually 44 in the case of  DeskMate .snd files.* The purpose of a header?  A header tells the program how big the file is, as most programs that read  digitized sound files need to be specially told how many bits of sound they  need to load up.* It also might tell you the "name" or description of the  sound file
* The sampling rate (how fast the sound was recorded and should  be played back) is also usually included.  Christopher Taveres was the first to try to decode the .snd header. * Chris's effort appeared as part of his SOUNDOFF.C program in _PCM_, the  Tandy magazine.* Here's what he came up with:  ======================================================================  /* Sound Off! /* Written by Christopher Taveres  /* Copyright (c) January 1992 /* Falsoft, Inc. /* PCM /** Definintion for the header block for Deskmate .snd files **/  struct dmheader {*********** /* Structure of the header block***** */ ****** int marker;********** /* Marker bytes - should be 00 1a**** */ ****** char note_count;***** /* Number of notes in instrument file */ ****** char inst_num;******* /* Instrument number***************** */ ****** char inst_name[10];** /* Instrument name******************* */ ****** int sample_rate;***** /* Sampling rate********************* */ ****** char filler[16];***** /* I don't know what this does****
*** */ ****** unsigned long sample_size;* /* Number of samples in file*** */ ****** char filler2[8];***** /* More unknown space**************** */  ======================================================================  Ken noticed that the marker bytes could be 1A 01 instead of 1A 00.* The  number of samples confused him, however.* After fiddling around with  SOUND.PDM for a while, I filled in most of the blanks as follows:   TANDY DESKMATE .SND FORMAT NOTES (old .snd file format) -------------------------------------------------------  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.  Tandy used two incompatible .snd file formats.* The first format, described  here, was used by the 1988 and 1989 versions of DeskMate Sound and Music -  the version bundled with the 1000SL and 1000TL.* The second format,  described below, was used
by the 1990 versions of these programs - the  version bundled with the 2500-series.* The 1990 version of Sound.pdm can  read but not write the old file format and can be used to convert to the  new format; the 1988 version of Sound.pdm does not understand the new  format.  There are two types of DeskMate .snd files, "sound" files and "instrument"  files.* Both contain 8-bit unsigned PCM samples.  Sound files 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 (I'm not sure whether the decay  section is ever actually played) with several samples having different  implied frequencies and being used by Music.pdm to represent the instrument  in different pitch ranges.* Up to 16 differ
ent notes (with 16 different  samples) can be contained in one instrument file.* Instrument files are  always recorded at 11kHz.  Frank Durda notes that if the sound was recorded on a Tandy 1000SL or  1000TL, the actual sampling rate will be slightly different from the rate  specified in the file.* The adjustment required does not appear to be a  simple function, though.* The old format does not describe the recording  hardware anyway.  Both sound files and instrument files may be compressed in one of two ways,  "music" compression or "speech" compression, or they may be uncompressed. * Music compression uses a combination of run length encoding and non-  adaptive differential PCM.* Speech compression adds a silence suppression  step before the run length encoding.* Both compression types are lossy,  speech compression more so.* The Tandy Digital Sound Toolkit, now in the  public domain in binary form, can be used for decompression.* Frank Durda's  PSSJ Digital Sound Toolkit also contains decompression code.
* It is  available in both source and binary forms but is restricted to  noncommercial use.* Sound.pdm can also be used to decompress an .snd file.  There are some programs for playing .snd files at my ftp/WWW site:  *** ftp://ftp.agate.net/users/01935/tandy1000/sound/ *** http://www.agate.net/~tvdog/sound.html  Basic stuff for beginners :-):  ** A sound can be represented mathematically as a waveform that is a  ** function of time; for example, y = sin(t) would represent a pure tone. * ** The sound wave is continuous, however, and computers deal with discrete  ** data.* Hence, to represent a sound in a computer, the waveform is  ** "sampled" at regular intervals and its level recorded.* The more  ** frequently the wave is sampled, and the finer the measurement of its  ** level, the closer the approximation to the original wave, and the better  ** the quality of the recorded sound.* The chip that samples the sound wave  ** is called an "analog-to-digital converter," or ADC.* When the recorded  ** sound is to
be played back, another chip takes the samples and changes  ** them back into an approximation of the sound wave.* That chip is called  ** a "digital-to-analog converter," or DAC.* In the Tandy, the same chip,  ** called "PSSJ," performs both functions.  ** There are different ways of representing the level of the sound wave  ** when sampling.* The simplest way is to record the sound level as an  ** integer, usually in 8 or 16 bits.* This is called "pulse code  ** modulation," or PCM.* In unsigned PCM, zero is the lowest possible level  ** of the sound wave, and the samples range from 0 to 255 (8 bits) or from  ** 0 to 65535 (16 bits).* In signed PCM, zero is the middle level of the  ** wave, and the samples range from -128 to 127 (8 bits) or -32768 to 32767  ** (16 bits).* The Tandy chip uses 8-bit unsigned PCM samples.* It is also  ** possible to record the level of the sound wave as a real number, or  ** rather an approximate real number.* A-law and mu-law encoding are  ** examples of that.* Once the sam
ples have been recorded, they can be  ** compressed by various methods.* "Run length encoding," "differential  ** pulse code modulation," and "adaptive differential pulse code  ** modulation" are a few compression methods.* Some compression methods are  ** "lossy," meaning that when you decompress the compressed file, you don't  ** get back *quite* the original sound.* It is frequently possible to get  ** more compression if you can tolerate some loss in the sound quality.  ** Many of the numbers below are given in hexadecimal, or base-16 notation. * ** Hexadecimal is a common way to express numbers when programming at the  ** machine level.* In base-16, letters A through F are used to represent  ** "digits" 10 through 15.* Also: "word" is a 16-bit unsigned integer, and  ** "dword" is a 32-bit unsigned integer.  ** "American Standard Pitch" refers to the sound frequencies assigned to  ** the various musical notes by the American Standards Association.* Sound  ** frequencies are based on A above middle C bein
g 440 cycles per second. * ** This A is called A4.* An octave below that is A3, an octave above, A5. * ** The frequency doubles with each octave increase in pitch.* Sound.pdm  ** does not display standard notes to you, though.* Standard octaves begin  ** at C, so you would have A#2, B2, C3.* Sound.pdm starts its octaves at A.  ** The sampling rate has nothing to do with the pitch frequency, except  ** that the rate must be at least twice as high as the highest frequency  ** you want to record.  The old 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 (<control>-Z, prevents DOS TYPE from trying to ********************* put the file on the screen).  * 1******** byte***** Compression code:* 0 = no compressio
n; 1 = music ********************* compression; 2 = speech compression.  * 2******** byte***** Number of notes in the instrument file.* 1 if sound ********************* file.* There can be a maximum of 16 notes in an ********************* instrument file.* Note:* when an instrument file is ********************* created, it automatically has one unset note placed ********************* in it.  * 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.* The name is null-terminated, ********************* so the name may be at most 9 characters long.  * 0Eh****** word***** Sampling rate in samples per second.* Note that although ********************* a sampling rate other than 5500, 11000 and 22000 can be ********************* en
tered 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****** dword**** 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 --------------------------  These files are created by the 1990 version of DeskMate Sound.pdm, bundled  with the Tandy 2500-series computers.  This information was provided by John Ball ([email protected]), author of  Playsnd, and by the PSSJ Digital Sound Toolkit of Frank Durda IV.  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, possibly compressed.  Fixed header: *** offset****** size******* what *** ------****** ----******* ---- ***** 0********* 44 bytes*** Name or description o
f the sound.* According  **************************** to John Ball, this is null-terminated ASCII. * **************************** According to Frank Durda, this is ASCII  **************************** terminated with carriage return, line feed,  **************************** and <control>-Z (1Ah). ***** 2Ch******* byte******* <Control>-Z (1Ah).* As with the old format,  **************************** this prevents the file from being typed to the  **************************** screen with the DOS TYPE command. ***** 2Dh******* byte******* Format code:* 80h. ***** 2Eh******* word******* Number of sounds in the file, 1 for sound  **************************** files.* Instrument files may have more than 1  **************************** sound (note). ***** 30h******* byte******* Instrument number of an instrument file. * **************************** Sound files have 0 here. ***** 31h******* 17 bytes*** Reserved, set to 0. ***** 42h******* word******* Compression code (0 = no compression, 1 = ***************************
* music compression, 2 = speech compression). ***** 44h******* 20 bytes*** Compression information; refer to the PSSJ  **************************** Digital Sound Toolkit.* This information does  **************************** not seem to be needed to decompress if  **************************** compression types 1 or 2 were used. ***** 58h******* word******* Sampling rate in Hz. ***** 5Ah******* word******* Reserved, set to 0. ***** 5Ch******* word******* Hardware bias value, refers to the recording  **************************** hardware used.* Refer to the PSSJ Digital  **************************** Sound Toolkit.* If writing the file, you can  **************************** set this to 0. ***** 5Eh******* 20 bytes*** Reserved, set to 0. ***** 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********* word******* Sound options, refer to the PSSJ Digital Sound  **************************** Toolkit.* 1 = player should close the file  **************************** after playback.* If writing the file, you can  **************************** set this to 0. ***** 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).* Only used by Sound.pdm, according  **************************** to Mr. Durda. ***** 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, after  **************************** compression if that was done. ***** 12h******* dword****** Uncompressed length of sound data (number of **************************** samples). ***** 16h******* dword****** Start of sustain region. ***** 1Ah******* dword****** End of sustain region. ***** 1Eh******* 16 bytes*** Reserved, set to 0.  //// end of conv2snd.doc ////

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