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Scream Tracker 3.

20 File Formats And Mixing Info


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This document finally containts the OFFICIAL information on s3m format


and much more. There might be some errors here, so if something seems
weird, don't just blindly believe it. Think first if it could be just
a typo or something.

-----------------------------------------------------------------------------
What is the S3M file format?
What is the samplefile format?
What is the adlib instrument format?

The first table describes the S3M header. All other blocks are
pointer to by pointers, so in theory they could be anywhere in
the file. However, the practical standard order is:
- header
- instruments in order
- patterns in order
- samples in order

Next the instrument header is described. It is stored to S3M


for each instrument and also saved to the start of all samples
saved from ST3. Same header is also used by Advance Digiplayer.

The third part is the description of the packed pattern format.

S3M Module header


0 1 2 3 4 5 6 7 8 9 A B C D E F
���������������������������������������������������������������Ŀ
0000: � Song name, max 28 chars (end with NUL (0)) �
���������������������������������������������������������������Ĵ
0010: � �1Ah�Typ� x � x �
���������������������������������������������������������������Ĵ
0020: �OrdNum �InsNum �PatNum � Flags � Cwt/v � Ffi �'S'�'C'�'R'�'M'�
���������������������������������������������������������������Ĵ
0030: �g.v�i.s�i.t�m.v�u.c�d.p� x � x � x � x � x � x � x � x �Special�
���������������������������������������������������������������Ĵ
0040: �Channel settings for 32 channels, 255=unused,+128=disabled �
���������������������������������������������������������������Ĵ
0050: � �
���������������������������������������������������������������Ĵ
0060: �Orders; length=OrdNum (should be even) �
���������������������������������������������������������������Ĵ
xxx1: �Parapointers to instruments; length=InsNum*2 �
���������������������������������������������������������������Ĵ
xxx2: �Parapointers to patterns; length=PatNum*2 �
���������������������������������������������������������������Ĵ
xxx3: �Channel default pan positions �
���������������������������������������������������������������Ĵ
xxx1=70h+orders
xxx2=70h+orders+instruments*2
xxx3=70h+orders+instruments*2+patterns*2

Parapointers to file offset Y is (Y-Offset of file header)/16.


You could think of parapointers as segments relative to the
start of the S3M file.

Type = File type: 16=ST3 module


Ordnum = Number of orders in file (should be even!)
Insnum = Number of instruments in file
Patnum = Number of patterns in file
Cwt/v = Created with tracker / version: &0xfff=version, >>12=tracker
ST3.00:0x1300 (NOTE: volumeslides on EVERY frame)
ST3.01:0x1301
ST3.03:0x1303
ST3.20:0x1320
Ffi = File format information
1=[VERY OLD] signed samples
2=unsigned samples
Flags = [ These are old flags for Ffv1. Not supported in ST3.01
| +1:st2vibrato
| +2:st2tempo
| +4:amigaslides
| +32:enable filter/sfx with sb
]
+8: 0vol optimizations
Automatically turn off looping notes whose volume
is zero for >2 note rows.
+16: amiga limits
Disallow any notes that go beond the amiga hardware
limits (like amiga does). This means that sliding
up stops at B#5 etc. Also affects some minor amiga
compatibility issues.
+64: st3.00 volumeslides
Normally volumeslide is NOT performed on first
frame of each row (this is according to amiga
playing). If this is set, volumeslide is performed
ALSO on the first row. This is set by default
if the Cwt/v files is 0x1300
+128: special custom data in file (see below)
Special = pointer to special custom data (not used by ST3.01)
ExtHead = pointer to extended header data area (generally after)
the main header)
g.v = global volume (see next section)
m.v = master volume (see next section) 7 lower bits
bit 8: stereo(1) / mono(0)
i.s = initial speed (command A)
i.t = initial tempo (command T)
u.c = ultra click removal. ST3 uses u.c gus channels to
guarantee, that u.c/2 channels run without any clicks.
If more channels are used, some clicks might appear.
The number displayed in ST3 order page is u.c/2
d.p = 252 when default channel pan positions are present
in the end of the header (xxx3). If !=252 ST3 doesn't
try to load channel pan settings.

Channel settings (byte per channel):


bit 8: channel enabled
bit 0-7: channel type
0..7 : Left Sample Channel 1-8
8..15 : Right Sample Channel 1-8
16..31 : Adlib channels (9 melody + 5 drums)
Channel pan settings (byte per channel):
bit 6-7: reserved
bit 5: 1=default pan position specified, 0=use defaults:
for mono 7, for stereo 3 or C.
bit 0-3: default pan position

Global volume directly divides the volume numbers used. So


if the module has a note with volume 48 and master volume
is 32, the note will be played with volume 24. This affects
both Gravis & SoundBlasters.

Master volume only affects the SoundBlaster. It controls


the amount of sample multiplication (see mixing section
of this doc). The bigger the value the bigger the output
volume (and thus quality) will be. However if the value
is too big, the mixer may have to clip the output to
fit the 8 bit output stream. The default value works
pretty well. Note that in stereo, the mastermul is
internally multiplied by 11/8 inside the player since
there is generally more room in the output stream.

Order list lists the order in which to play the patterns. 255=--
is the end of tune mark and 254=++ is just a marker that is
skipped.

Digiplayer/ST3 samplefileformat
0 1 2 3 4 5 6 7 8 9 A B C D E F
���������������������������������������������������������������Ŀ
0000: �[T]� Dos filename (12345678.ABC) � MemSeg �
���������������������������������������������������������������Ĵ
0010: �Length �HI:leng�LoopBeg�HI:LBeg�LoopEnd�HI:Lend�Vol� x �[P]�[F]�
���������������������������������������������������������������Ĵ
0020: �C2Spd �HI:C2sp� x � x � x � x �Int:Gp �Int:512�Int:lastused �
���������������������������������������������������������������Ĵ
0030: � Sample name, 28 characters max... (incl. NUL) �
���������������������������������������������������������������Ĵ
0040: � ...sample name... �'S'�'C'�'R'�'S'�
���������������������������������������������������������������Ĵ
xxxx: sampledata

Length / LoopBegin / LoopEnd are all 32 bit parameters although


ST3 only support file sizes up to 64,000 bytes. Files bigger
than that are clipped to 64,000 bytes when loaded to ST3. NOTE
that LoopEnd points to one byte AFTER the end of the sample,
so LoopEnd=100 means that byte 99.9999 (fixed) is the last one
played.
C2Spd = Herz for middle C. ST3 only uses lower 16 bits.
Vol = Default volume 0..64
Memseg = Pointer to sampledata
Inside a sample or S3M, MemSeg tells the parapointer to
the actual sampledata. In files all 24 bits are used.
In memory the value points to the actual sample segment
or Fxxx if sample is in EMS under handle xxx. In memory
the first memseg byte is overwritten with 0 to create
the dos filename terminator nul.
Int:Gp = Internal: Address of sample in gravis memory /32
(only used while module in memory)
Int:512= Internal: flags for soundblaster loop expansion
(only used while module in memory)
Int:las= Internal: last used position (only works with sb)
(only used while module in memory)
[T]ype 1=Sample, 2=adlib melody, 3+=adlib drum (see below for
adlib structure)
[F]lags, +1=loop on
+2=stereo (after Length bytes for LEFT channel,
another Length bytes for RIGHT channel)
+4=16-bit sample (intel LO-HI byteorder)
(+2/+4 not supported by ST3.01)
[P]ack 0=unpacked, 1=DP30ADPCM packing (not used by ST3.01)

adlib instrument format


0 1 2 3 4 5 6 7 8 9 A B C D E F
���������������������������������������������������������������Ŀ
0000: �[T]� Dos filename (12345678.123) �00h�00h�00h�
���������������������������������������������������������������Ĵ
0010: �D00�D01�D02�D03�D04�D05�D06�D07�D08�D09�D0A�D0B�Vol�Dsk� x � x �
���������������������������������������������������������������Ĵ
0020: �C2Spd �HI:C2sp� x � x � x � x � x � x � x � x � x � x � x � x �
���������������������������������������������������������������Ĵ
0030: � Sample name, 28 characters max... (incl. NUL) �
���������������������������������������������������������������Ĵ
0040: � ...sample name... �'S'�'C'�'R'�'I'�
���������������������������������������������������������������Ĵ

[T]ype= 2:amel 3:abd 4:asnare 5:atom 6:acym 7:ahihat


C2Spd = 'Herz' for middle C. ST3 only uses lower 16 bits.
Actually this is a modifier since there is no
clear frequency for adlib instruments. It scales
the note freq sent to adlib.
D00..D0B contains the adlib instrument specs packed like this:
modulator: carrier:
D00=[freq.muliplier]+[?scale env.]*16+[?sustain]*32+ =D01
[?pitch vib]*64+[?vol.vib]*128
D02=[63-volume]+[levelscale&1]*128+[l.s.&2]*64 =D03
D04=[attack]*16+[decay] =D05
D06=[15-sustain]*16+[release] =D07
D08=[wave select] =D09
D0A=[modulation feedback]*2+[?additive synthesis]
D0B=unused

packed pattern format


0 1 2 3 4 5 6 7 8 9 A B C D E F
���������������������������������������������������������������Ŀ
0000: �Length � packed data, see below...
���������������������������������������������������������������Ĵ

Length = length of packed pattern

Unpacked pattern is always 32 channels by 64 rows. Below


is the unpacked format st uses for reference:
Unpacked Internal memoryformat for patterns (not used in files):
NOTE: each channel takes 320 bytes, rows for each channel are
sequential, so one unpacked pattern takes 10K.
byte 0 - Note; hi=oct, lo=note, 255=empty note,
254=key off (used with adlib, with samples stops smp)
byte 1 - Instrument ;0=..
byte 2 - Volume ;255=..
byte 3 - Special command ;255=..
byte 4 - Command info ;

Packed data consits of following entries:


BYTE:what 0=end of row
&31=channel
&32=follows; BYTE:note, BYTE:instrument
&64=follows; BYTE:volume
&128=follows; BYTE:command, BYTE:info

So to unpack, first read one byte. If it's zero, this row is


done (64 rows in entire pattern). If nonzero, the channel
this entry belongs to is in BYTE AND 31. Then if bit 32
is set, read NOTE and INSTRUMENT (2 bytes). Then if bit
64 is set read VOLUME (1 byte). Then if bit 128 is set
read COMMAND and INFO (2 bytes).

For information on commands / how st3 plays them, see the


manual.

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What is the Stmik300old format?
What is the STIMPORT file format?
What is the SIMPLEXFILE format?

The old stmik 300 (never published because it has no


usable interface) want's that the samples in the module
are pre-extended for soundblaster. This means that
samples have an extra 512 bytes of loop after their
loopend (or silence if no loop). The save option for
old stmik does this extension. Otherwise the fileformat
is the same as for a normal S3M.

STIMPORT file format is supposed to be an easy way of


imputting weird data to st3. That is, it should be
easy to convert something to STIMPORT format. The format
goes like this:

0 1 2 3 4 5 6 7 8 9 A B C D E F
���������������������������������������������������������������Ŀ
0000: �'S'�'T'�'I'�'M'�'P'�'O'�'R'�'T'�i.s� x � x � x � x � x � x � x �
���������������������������������������������������������������Ĵ
0010: �Notedata...
���������������������������������������������������������������Ĵ
xxxx: �Instruments...
���������������������������������������������������������������Ĵ

There are no patterns or orders, just a continuos note stream


of following structures:
0 1 2 3 4 5
�����������������������������Ŀ
�Chan�Note�Inst�Volu�Cmnd�Info�
�������������������������������
Chan is the channel number for the note +128. Value 0 stands
for next row. Value 255 stands for end of note stream.
Note/Inst/Cmnd/Info are like in the unpacked memory pattern
(see previous section)

After the notedata there can be instruments (optional)


Every instrument consits of the following block:
0 1 2 3 4
����������������������������������������������������...��������...
�Inst�Flag�LoopBeg �LoopEnd �Length � Sampledata... � Name (opt)
����������������������������������������������������...��������...
Inst = number of instrument to load (0=end of instruments)
Flag = +1=loop enabled +128=name after sampledata (null terminated)
LoopBeg/LoopEnd/Length = guess
Sampledata = raw sampledata for Length bytes.

SIMPLEXFORMAT is designed for easy export of notedata.


The contents of a S3Y file is:

32 first instruments, 128 bytes each.


The 80 first bytes correspond to the ST3 Samplefile header
(described in this file). The rest correspond to nothing.
Raw notedata for the entire song - 9 channels stored.
Every row consists of 9 notes, each consisting of 5 bytes:
���������������������
�not�ins�vol�cmd�inf�
���������������������
byte 0 - Note; hi=oct, lo=note, 255=empty note,
254=key off (used with adlib, with samples stops smp)
byte 1 - Instrument ;0=..
byte 2 - Volume ;255=..
byte 3 - Special command ;255=..
byte 4 - Command info ;

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What is C2SPD?
How to calculate the note frequencies like ST3?
How does ST3 mix depending on master volume?

Finetuning (C2SPD) is actually the frequency in herz for


the note C4. Why is it C2SPD? Well, originally in ST2
the middle note was C2 and the name stuck. Later in ST3
the middle note was raised to C4 for more octaves... So
actually C2SPD should be called C4SPD...

Table for note frequencies used by ST3:

note: C C# D D# E F F# G G# A A# B
period: 1712,1616,1524,1440,1356,1280,1208,1140,1076,1016,0960,0907
middle octave is 4.

8363 * 16 * ( period(NOTE) >> octave(NOTE) )


note_st3period = --------------------------------------------
middle_c_finetunevalue(INSTRUMENT)

note_amigaperiod = note_st3period / 4

note_herz=14317056 / note_st3period

Note that ST3 uses period values that are 4 times larger than the
amiga to allow for extra fine slides (which are 4 times finer
than normal fine slides).

How ST3 mixes:

1) volumetable is created in the following way:

> volumetable[volume][sampledata]=volume*(sampledata-128)/64;

NOTE: sampledata in memory is unsigned in ST3, so the -128 in the


formula converts it so that the volumetable output is signed.

2) postprocessing table is created with this pseudocode:

> z=mastervol&127;
> if(z<0x10) z=0x10;
> c=2048*16/z;
> a=(2048-c)/2;
> b=a+c;
> { 0 , if x < a
> posttable[x+1024] = { (x-a)*256/(b-a) , if a <= x < b
> { 255 , if x > b

3) mixing the samples

output=1024
for i=0 to number of channels
output+=volumetable[volume*globalvolume/64][sampledata];
next
realoutput=posttable[output]

This is how the mixing is done in theory. In practice it's a bit


different for speed reasons, but the result is the same.

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That's it. If there are any more questions, that's too bad :-) If you have
problems with the S3M format, try to contact somone who already supports
it (there is quite a lot of support for the S3M already, so that shouldn't
be too hard...) Good luck for reading / writing Scream Tracker files.