TXTH is a simple text file that uses text commands to simulate a header for files unsupported by vgmstream, mainly headerless audio.
When an unsupported file is loaded (for instance "bgm01.snd"), vgmstream tries to find a TXTH header in the same dir, in this order:
If found and parsed correctly (the .txth may be rejected if incorrect commands are found) vgmstream will try to play the file as described. Extension must be accepted/added to vgmstream (plugins like foobar2000 only load extensions from a whitelist in formats.c), or one could rename to any supported extension (like .vgmstream), or leave the file extensionless.
You can also use
.(sub).(ext).txth (if the file is
filename.sub.ext), to allow mixing slightly different files in the same folder. The
sub part doesn't need to be an extension, for example:
For an unsupported
bgm01.vag this would be a simple TXTH for it:
codec = PSX #data uses PS-ADPCM sample_rate = @0x10$2 #get sample rate at offset 0x10, 16 bit value channels = @0x14 #get number of channels at offset 14 interleave = 0x1000 #fixed value start_offset = 0x100 #data starts after exactly this value num_samples = data_size #find automatically number of samples in the file loop_flag = auto #find loop points in PS-ADPCM
A text file with the above commands must be saved as
.txth (preferably the former), notice it starts with a "." (dot). On some Windows versions files starting with a dot need to be created by appending a dot at the end when renaming:
While the main point is playing the file, many of TXTH's features are aimed towards keeping original data intact, for documentation and preservation purposes; try leaving data as untouched as possible and consider how the game plays the file, as there is a good chance some feature can mimic it.
The file is made of lines with
key = value commands describing a header. Commands are all case sensitive and spaces are optional:
key = value, and so on are all ok. Comments start with # and can be inlined.
The parser is fairly simple and may be buggy or unexpected in some cases. The order of keys is variable but some things won't work if others aren't defined (ex. bytes-to-samples may not work without channels or interleave) or need to be done in a certain order (due to technical reasons) as explained below.
To get a file playing you need to correctly set, at least:
codec and sometimes
num_samples, or use the "subfile" feature.
The following can be used in place of
(key) = (value) commands.
(number): constant number in dec/hex, unsigned (no +10 or -10).
44100, 40, 0x40 (decimal=64)
(offset): read a value at offset inside the file, format being
@(number): offset of the value (required)
base_offsetis defined this value is modified (see later)
:LE|BE: value is little/big endian (optional, defaults to LE)
$1|2|3|4: value has size of 8/16/24/32 bit (optional, defaults to 4)
get big endian 16b value at 0x10
(field): uses current value of some fields. Accepted strings:
interleave, interleave_last, channels, sample_rate, start_offset, data_size, num_samples, loop_start_sample, loop_end_sample, subsong_count, subsong_spacing, subfile_offset, subfile_size, base_offset, name_valueX
(other): other special values for certain keys, described per key
The above may be combined with math operations (+-*/&):
(key) = (number) (op) (offset) (op) (field) (...)
Sets codec used to encode the data. Some codecs need interleave or other config as explained below, but often will use default values. Accepted codec strings:
# - PSX PlayStation ADPCM # * For many PS1/PS2/PS3 games # * Interleave is multiple of 0x10 (default), often +0x1000 # - PSX_bf PlayStation ADPCM with bad flags # * Variation with garbage data, for rare PS2 games # - XBOX Xbox IMA ADPCM (mono/stereo) # * For many XBOX games, and some PC games # * Special interleave is multiple of 0x24 (mono) or 0x48 (stereo) # - DSP|NGC_DSP Nintendo GameCube ADPCM # * For many GC/Wii/3DS games # * Interleave is multiple of 0x08 (default), often +0x1000 # * Must set decoding coefficients (coef_offset/spacing/etc) # * Should set ADPCM state (hist_offset/spacing/etc) # - DTK|NGC_DTK Nintendo ADP/DTK ADPCM # * For rare GC games # - PCM16LE PCM 16-bit little endian # * For many games (usually on PC) # * Interleave is multiple of 0x2 (default) # - PCM16BE PCM 16-bit big endian # * Variation for certain consoles (GC/Wii/PS3/X360/etc) # - PCM8 PCM 8-bit signed # * For some games (usually on PC) # * Interleave is multiple of 0x1 (default) # - PCM8_U PCM 8-bit unsigned # * Variation with modified encoding # - PCM8_U_int PCM 8-bit unsigned (interleave block) # * Variation with modified encoding # - IMA IMA ADPCM (mono/stereo) # * For some PC games, and rarely consoles # * Special interleave is multiple of 0x1, often +0x80 # - DVI_IMA IMA ADPCM (DVI order) # * Variation with modified encoding # - AICA Yamaha AICA ADPCM (mono/stereo) # * For some Dreamcast games, and some arcade (Naomi) games # * Special interleave is multiple of 0x1 # - APPLE_IMA4 Apple Quicktime IMA ADPCM # * For some Mac/iOS games # - MS_IMA Microsoft IMA ADPCM # * For some PC games # * Interleave (frame size) varies, often multiple of 0x100 [required] # - MSADPCM Microsoft ADPCM (mono/stereo) # * For some PC games # * Interleave (frame size) varies, often multiple of 0x100 [required] # - SDX2 Squareroot-delta-exact 8-bit DPCM # * For many 3DO games # - MPEG MPEG Audio Layer file (MP1/2/3) # * For some games (usually PC/PS3) # * May set skip_samples # - ATRAC3 Sony ATRAC3 # * For some PS2 and PS3 games # * Interleave (frame size) can be 0x60/0x98/0xC0 * channels [required] # * Should set skip_samples (more than 1024+69 but varies) # - ATRAC3PLUS Sony ATRAC3plus # * For many PSP games and rare PS3 games # * Interleave (frame size) can be: [required] # Mono: 0x0118|0178|0230|02E8 # Stereo: 0x0118|0178|0230|02E8|03A8|0460|05D0|0748|0800 # 6/8 channels: multiple of one of the above # * Should set skip_samples (more than 2048+184 but varies) # - XMA1 Microsoft XMA1 # * For early X360 games # - XMA2 Microsoft XMA2 # * For later X360 games # - FFMPEG Any headered FFmpeg format # * For uncommon games # * May set skip_samples # - AC3 AC3/SPDIF # * For few PS2 games # * Should set skip_samples (around 256 but varies) # - PCFX PC-FX ADPCM # * For many PC-FX games # * Interleave is multiple of 0x1, often +0x8000 # * Sample rate may be ~31468/~15734/~10489/~7867 # - PCM4 PCM 4-bit signed # * For early consoles # - PCM4_U PCM 4-bit unsigned # * Variation with modified encoding # - OKI16 OKI ADPCM with 16-bit output (not VOX/Dialogic 12-bit) # * For rare PS2 games (Sweet Legacy, Hooligan) # - OKI4S OKI ADPCM with 16-bit output and adjusted tables # * For later Konami rhythm games # - AAC Advanced Audio Coding (raw without .mp4) # * For some 3DS games and many iOS games # * Should set skip_samples (around 1024 but varies) # - TGC Tiger Game.com 4-bit ADPCM # * For Tiger Game.com # - ASF Argonaut ASF ADPCM # * For rare Argonaut games [Croc (SAT)] # - EAXA Electronic Arts EA-XA ADPCM # * For rare EA games [Harry Potter and the Chamber of Secrets (PC)] codec = (codec string)
Changes the behavior of some codecs:
# - NGC_DSP: 0=normal interleave, 1=byte interleave, 2=no interleave # - XMA1|XMA2: 0=dual multichannel (2ch xN), 1=single multichannel (1ch xN) # - XBOX|EAXA: 0=standard (mono or stereo interleave), 1=force mono interleave mode # - PCFX: 0=standard, 1='buggy encoder' mode, 2/3=same as 0/1 but with double volume # - PCM4|PCM4_U: 0=low nibble first, 1=high nibble first # - others: ignored codec_mode = (variation)
Use inline math instead of this.
Changes next read to:
(key) = (value) */+- value_(op). Set to 0 when done using, as it affects ANY value. Priority is as listed.
value_mul|value_* = (value) value_div|value_/ = (value) value_add|value_+ = (value) value_sub|value_- = (value)
This value changes how data is read depending on the codec:
half_size: sets interleave as data_size / channels automatically
interleave = (value)|half_size
In some files with interleaved data the last block (
interleave * channels) of data is smaller than normal, so
interleave is smaller for that block. Setting this fixes decoding glitches at the end.
Note that this doesn't affect files with padding data in the last block (as the
interleave itself is constant).
auto: calculate based on channels, interleave and data_size/start_offset
interleave_last = (value)|auto
id_value (normally set as constant value) matches value read at
id_offset. The file will be rejected and won't play if values don't match.
Can be redefined several times, it's checked whenever a new id_offset is found.
id_value = (value) id_offset = (value)
channels = (value)
sample_rate = (value)
Where encoded data actually starts, after the header part. Defaults to 0.
start_offset = (value)
Special variable that can be used in sample values. Defaults to
(file_size - start_offset), re-calculated when
start_offset is set. With multiple subsongs,
block_size or padding are set this it's recalculated as well.
If data_size is manually set it stays constant and won't be auto changed.
data_size = (value)
Some files have extra padding at the end that is meant to be ignored. This adjusts the padding in
data_size, manually or auto-calculated.
Special values (for PS-ADPCM only):
auto: discards null frames
auto-empty: discards null and 'empty' frames (for games with weird padding)
padding_size = (value)|auto|auto-empty
Modifies the meaning of sample fields when set before them.
samples: exact sample (default)
bytes: automatically converts bytes/offset to samples (applies after */+-& modifiers)
blocks: same as bytes, but value is given in blocks/frames
bytes = (value * interleave*channels)
Some codecs can't convert bytes-to-samples at the moment:
FFMPEG. For XMA1/2, bytes does special parsing, with loop values being bit offsets within data (as XMA has a peculiar way to loop).
sample_type = samples|bytes|blocks
Those tell vgmstream how long the song is. Define loop points for the track to repeat at those points (if plugin is configured to loop).
You can use
loop_end instead as aliases of
loop_end_sample (no difference).
data_size: automatically converts bytes-to-samples (a few codecs don't allow this)
num_samples = (value)|data_size loop_start_sample = (value) loop_end_sample = (value)|data_size
Force loop on or off, as loop start/end may be defined but not used. If not set, by default it loops when loop_end_sample is defined and less than num_samples.
auto: tries to autodetect loop points for PS-ADPCM data using data loop flags.
Sometimes games give loop flags different meaning, so behavior can be tweaked by defining
default: values 0 or 0xFFFF/0xFFFFFFFF (-1) disable looping, but not 0xFF (loop endlessly)
negative: values 0xFF/0xFFFF/0xFFFFFFFF (-1) enable looping
positive: values 0xFF/0xFFFF/0xFFFFFFFF (-1) disable looping
inverted: values not 0 disable looping
loop_behavior = default|negative|positive|inverted loop_flag = (value)|auto
For XMA1/2 + sample_type=bytes it means loop subregion, if read after loop values.
For other codecs its added to loop start/end, if read before loop values (a format may rarely have rough loop offset/bytes, then a loop adjust in samples).
loop_adjust = (value)
Beginning samples to skip, a.k.a. priming samples or encoder delay, that some codecs use to "warm up" the decoder. This is needed for proper gapless support.
skip_samples = (value)
DSP needs a "coefs" list to decode correctly. These are 8*2 16-bit values per channel, starting from
Usually each channel uses its own list, so we may need to set separation per channel, usually 0x20 (16 values * 2 bytes). So channel N coefs are read at
coef_offset + coef_spacing * N
Those 16-bit coefs can be little or big endian (usually BE), set
coef_endianness directly or in an offset value where
While the coef table is almost always included per-file, some games have their coef table in the executable or precalculated somehow. You can set inline coefs instead of coef_offset. Format is a long string of bytes (optionally space-separated) like
coef_table = 0x1E02DE01 3C0C0EFA .... You still need to set
coef_offset = (value) coef_spacing = (value) coef_endianness = BE|LE|(value) coef_table = (string)
Some ADPCM codecs need to set up their initial or "history" state, normally one or two 16-bit PCM samples per channel, starting from
Usually each channel uses its own state, so we may need to set separation per channel.
State values can be little or big endian (usually BE for DSP), set
hist_endianness directly or in an offset value where ´0=LE, >0=BE´.
Normally audio starts with silence or hist samples are set to zero and can be ignored, but it does affect a bit resulting output.
Currently used by DSP.
hist_offset = (value) hist_spacing = (value) hist_endianness = BE|LE|(value)
Changes internal header/body representation to external files.
TXTH commands are done on a "header", and decoding on "body". When loading an unsupported file it becomes the "base" file that loads the .txth, and is both header and body.
You can alter those, mainly for files that split header and body in separate files (load base file and txth sets header on another file). It's also possible to load the .txth directly with a set body, as a sort of "reverse TXTH" (useful with bigfiles, as you could have one .txth per song).
header_file = (filename)|*.(extension)|null body_file = (filename)|*.(extension)|null
Sets the number of subsongs in the file, adjusting reads per subsong N:
value = @(offset) + subsong_spacing*N. Number/constants values aren't adjusted though.
subsong_spacing you can use
subsong_offset (older alias).
Mainly for bigfiles with consecutive headers per subsong, set subsong_offset to 0 when done as it affects any reads. The current subsong number is handled externally by plugins or TXTP.
subsong_count = (value) subsong_spacing = (value)
Sets the name of the stream, most useful when used with subsongs. TXTH will read a string at
name_size defaults to 0, which reads until null-terminator or a non-ascii character is found.
name_offset can be a (number) value, but being an offset it's also adjusted by
subsong_spacing. If you need to point to some absolute offset (for example a subsong pointings to name in another table) that doesn't depend on subsong (must not be changed by
name_offset = (value) name_size = (value) name_offset_absolute = (value)
Tells TXTH to parse a full file (ex. an Ogg) at
subfile_offset, with size of
subfile_size (defaults to
file size - subfile_offset if not set). This is useful for files that are just container of other files, so you don't have to remove the extra data (since it could contain useful stuff like loop info).
Internal subfile extension can be changed to
subfile_extension if needed, as vgmstream won't accept unknown extensions (for example if your file uses .vgmstream or .pogg you may need to set subfile_extension = ogg).
Setting any of those three will trigger this mode (it's ok to set offset 0). Once triggered most fields are ignored, but not all, explained later. This will also set some values like
sample_rate if not set for calculations/convenience.
subfile_offset = (value) subfile_size = (value) subfile_extension = (string)
Some files interleave data chunks, for example 3 stereo songs pasted together, alternating 0x10000 bytes of data each. These settings allow vgmstream to play one of the chunks while ignoring the rest (read 0x10000 data, skip 0x10000*2).
File is first "dechunked" then played with using other settings (
start_offset would point within the internal dechunked" file). It can be used to remove garbage data that affects decoding, too.
You need to set:
chunk_count: total number of interleaved chunks (ex. 3=3 interleaved songs)
chunk_number: first chunk to start (ex. 1=0x00000, 2=0x10000, 3=0x20000...)
chunk_numberwill be auto-set per subsong (subsong 1 starts from chunk number 1, subsong 2 from chunk 2, etc)
chunk_start: absolute offset where chunks start (normally 0x00)
chunk_size: amount of data in a single chunk (ex. 0x10000)
For fine-tuning you can optionally set (before
chunk_size, for reasons):
chunk_header_size: header to skip before chunk data (part of chunk_size)
chunk_data_size: actual data size (part of chunk_size, rest is header/padding)
So, if you set size to 0x1000, header_size 0x100, data_size is implicitly 0xF00, or if size is 0x1000 and data_size 0x800 last 0x200 is ignored padding. Use combinations of the above to make vgmstream "see" only actual codec data.
chunk_count = (value) chunk_number = (value) chunk_start = (value) chunk_header_size = (value) chunk_data_size = (value) chunk_size = (value)
Some games have headers for all files pasted together separate from the actual data, but this order may be hard-coded or even alphabetically ordered by filename. In those cases you can set a "name table" that assigns constant values (one or many) to filenames. This table is loaded from an external text file (for clarity) and can be set to any name, for example
name_table = .names.txt
name_table = (filename)
Inside the table you define lines mapping a filename to a bunch of values, in this format:
# base definition (filename1): (value) ... # may put multiple comma-separated values, spaces are ok (filenameN) : (value1), (...) , (valueN) # inline comments too # put no name before the : to set default values : (value1), (...), (valueN)
Then I'll find your current file name, and you can then reference its numbers from the list as a
name_value field, like
base_offset = name_value,
start_offset = 0x1000 + name_value1,
interleave = name_value5, etc.
(filename) can be with or without extension (like
bgm01.vag or just
bgm01), and if the file's name isn't found it'll use default values, and if those aren't defined you'll get 0 instead. Being "values" they can use math or offsets too (
You can use wildcards to match multiple names too (it stops on first name that matches), and UTF-8 names should work, case insensitive even.
bgm_??_4: 4 # 4ch: files like bgm_00_4, bgm_01_4, etc bgm*_M: 1 # 1ch: some files end with _M for mono bgm*: 2 # 2ch: all other files, notice order matters
While you can put anything in the values, this feature is meant to be used to store some number that points to the actual data inside a real multi-header, that could be set with
header_file. If you feel the need to store many constant values per file, there is good chance it can be done in some better, simpler way.
You can set a default offset that affects next
@(offset) reads making them
@(offset + base_offset), for cleaner parsing.
This is particularly interesting when combined with offsets to some long value. For example instead of
channels = @0x714 you could set
base_offset = 0x710, channels = @0x04. Or values from the
base_offset = name_value, channels = @0x04.
It also allows parsing formats that set offsets to another offset, by "chaining"
base_offset = @0x10 (pointing to
base_offset = @0x20, it reads value at
0x60. Set to 0 when you want to disable/reset the chain:
base_offset = @0x10 then
base_offset = 0 then
base_offset = @0x20 reads value at
base_offset = (value)
Most commands are evaluated and calculated immediatedly, every time they are found. This is by design, as it can be used to adjust and trick for certain calculations.
It makes TXTHs a bit harder to follow, as they are order dependant, but otherwise it's hard to accomplish some things or others become ambiguous.
For example, normally you are given a data_size in bytes, that can be used to calculate num_samples for all channels.
channels = 2 sample_type = bytes num_samples = @0x10 #calculated from data_size
But sometimes this size is for a single channel only (even though the file may be stereo). You can set temporally change the channel number to force a correct calculation.
channels = 1 #not the actual number of channels sample_type = bytes num_samples = @0x10 #calculated from channel_size channels = 2 #change once calculations are done
You can also use:
channels = 2 sample_type = bytes num_samples = @0x10 * channels # resulting bytes is transformed to samples
Do note when using special values/strings like
loop_end_samples they must be alone to trigger.
data_size = @0x100 num_samples = data_size * 2 # doesn't tranform bytes-to-samples (do it before? after?)
data_size = @0x100 * 2 num_samples = data_size # ok
Also beware of order:
start_offset = 0x200 # recalculated data_size num_samples = data_size # transforms bytes-to-samples data_size = @0x100 # useless as num_samples is already transformed
Some commands alter the function of all next commands and can be redefined as needed:
samples_type = bytes num_samples = @0x10 samples_type = sample loop_end_sample = @0x14
When setting external files all commands are done on the "header" file, but with some creativity you can read in multiple files.
body_file = bgm01.bdy header_file = bgm01.hdr channels = @0x10 #base info in bgm01.hdr header_file = bgm01.bdy coef_offset = 0x00 #DSP coefs in bgm01.bdy
Note that DSP coefs are special in that aren't read immediately, and will use last header_file set.
Values may need to be reset (to 0 or other sensible value) when done. Subsong example:
subsong_count = 5 subsong_spacing = 0x20 # there are 5 subsong headers, 0x20 each channel_count = @0x10 # reads channels at 0x10+0x20*subsong # 1st subsong: 0x10+0x20*0: 0x10 # 2nd subsong: 0x10+0x20*1: 0x30 # 2nd subsong: 0x10+0x20*2: 0x50 # ... start_offset = @0x14 # reads offset within data at 0x14+0x20*subsong subsong_spacing = 0 # reset value sample_rate = 0x04 # sample rate is the same for all subsongs # Nth subsong ch: 0x04+0x00*N: 0x08
Sometimes header values are in "sectors" or similar concepts (typical in DVD games), and need to be adjusted to a real value using some complex math:
sample_type = bytes start_offset = @0x10 * 0x800 # 0x15 * DVD sector size, for example
You can use
+-*/& operators, and also certain fields' values:
num_samples = @0x10 * channels # byte-to-samples of channel_size
data_size is a special value for
loop_end_sample and will always convert as bytes-to-samples, though.
Priority is left-to-right. Do add brackets though, they are accounted for and if they are implemented in the future your .txth will break with impunity.
# normal priority data_size = @0x10 * 0x800 + 0x800 # also works data_size = (@0x10 + 1) * 0x800 # same as above but don't do this # (may become @0x10 + (1 * 0x800) in the future data_size = @0x10 + 1 * 0x800 # doesn't work at the moment, so reorder as (1 * 0x800) + @0x10 data_size = @0x10 + (1 * 0x800) # fails, wrong bracket count data_size = (@0x10 + 1 * 0x800 # fails, wrong bracket count data_size = )@0x10 + 1 * 0x800
If a TXTH needs too many calculations it may be better to implement directly in vgmstream though, consider reporting.
Remnant of simpler math (priority is fixed to */+-), shouldn't be needed anymore.
value_multiply = 0x800 start_offset = @0x10 value_multiply = 0
value_add = 1 channels = @0x08 value_add = 0 value_multiply = channels sample_type = bytes num_samples = @0x10 value_multiply = 0
value_add = 0x10 value_mul = 0x800 start_offset = @0x10
Sometimes a file is just a wrapper for another common format. In those cases you can tell TXTH to just play the internal format:
subfile_offset = 0x20 # tell TXTH to parse a full file (ex. .ogg) at this offset subfile_size = @0x10 # defaults to (file size - subfile_offset) if not set subfile_extension = ogg # may be ommited if subfile extension is the same # many fields are ignored codec = PCM16LE interleave = 0x1000 channels = 2 # a few fields are applied sample_rate = @0x08 num_samples = @0x10 loop_start_sample = @0x14 loop_end_sample = @0x18
Most fields can't be changed after parsing since doesn't make much sense technically, as the parsed subfile should supply them. You can set them to use bytes-to-samples conversions, though.
# parses subfile at start with some num_samples subfile_offset = 0x20 # force recalculation of num_samples codec = PSX start_offset = 0x40 num_samples = data_size
Chunks affect some values (padding size, data size, etc) and are a bit sensitive to order at the moment, due to technical complexities:
# Street Fighter EX3 (PS2) # base config is defined normally codec = PSX sample_rate = 44100 channels = 2 interleave = 0x8000 # set subsong number instead of chunk_number for subsongs subsong_count = 26 #chunk_number = 1 chunk_start = 0 chunk_size = 0x10000 chunk_count = 26 # after setting chunks (sizes vary when 'dechunking') start_offset = 0x00 padding_size = auto-empty num_samples = data_size
Subfiles and chunks can coexist:
# Gitaroo Man (PSP) # 3 interleaved RIFF files subsong_count = 3 chunk_start = 0 chunk_size = 0x2800 chunk_count = 3 # the 3 de-interleaved chunks are treated and parsed as a subsong subfile_offset = 0 subfile_size = @0x04 + 0x08 #RIFF size subfile_extension = at3
It can be used to make blocks with padding playable:
# Mortal Kombat: Deception (PS2) codec = PSX interleave = 0x3F40 sample_rate = 32000 channels = 2 chunk_number = 1 chunk_count = 1 chunk_start = 0x00 chunk_data_size = interleave * channels chunk_size = 0x8000 num_samples = data_size
Some formats read an offset to another part of the file, then another offset, then other, etc.
You can simulate this chaining multiple
base_offset = @0x10 #sets current at 0x1000 channels = @0x04 #reads at 0x1004 (base_offset + 0x04) base_offset = base_offset + @0x10 #sets current at 0x1000 + 0x200 = 0x1200 sample_rate = @0x04 #reads at 0x1204 ...
id_value = 0x00000000 #check that value at 0x00 is really 0x00000000 id_offset = @0x00:BE codec = PCM16LE channels = 2 sample_rate = 32000 start_offset = 0x04 num_samples = data_size loop_start_sample = 0 loop_end_sample = data_size
codec = PSX interleave = 0x2000 channels = 2 sample_rate = 48000 num_samples = data_size interleave_last = auto
codec = XBOX codec_mode = 1 #interleaved XBOX interleave = 0xD800 channels = 12 sample_rate = 44100 start_offset = 0x00 num_samples = data_size
codec = DVI_IMA interleave = 0x80 start_offset = 0x00 channels = 2 sample_rate = 44100 num_samples = data_size
codec = PCM8 sample_rate = 32000 channels = 1 start_offset = 0 num_samples = data_size
codec = NGC_DSP interleave = 0x10000 channels = 2 start_offset = 0x00 num_samples = @0x00:BE sample_rate = @0x08:BE loop_flag = @0x0C:BE$2 sample_type = bytes loop_start_sample = @0x10:BE loop_end_sample = @0x14:BE coef_offset = 0x1c coef_spacing = 0x10000 coef_endianness = BE
codec = PSX interleave = 0x10 sample_rate = 24000 channels = 1 padding_size = auto-empty num_samples = data_size
codec = PSX interleave = 0x1000 # this .txth is meant to be loaded directly header_file = data/SLPM_660.69 body_file = data/BGM.BIN channels = 2 # subsong headers at 0x1A5A40, entry size 0x14, total 58 * 0x14 = 0x488 subsong_count = 58 subsong_spacing = 0x14 base_offset = 0x1A5A40 sample_rate = @0x00 start_offset = @0x04 * 0x800 #in sectors sample_type = bytes num_samples = @0x08 * channels #in 1ch sizes loop_start_sample = @0x0c * channels loop_end_sample = @0x10 * channels data_size = @0x08 * channels #for bitrate
# parse MP3 inside the .snd subfile_extension = mp3 subfile_offset = 0x14 #subfile_size = @0x10 # manually set looping codec = MPEG start_offset = 0x14 num_samples = data_size loop_start_sample = 0 loop_end_sample = data_size
header_file = TSNDDRVC.IRX name_table = .names.txt base_offset = 0xAC3c + name_value codec = PSX interleave = @0x10 sample_rate = @0x0A$2 * 48000 / 4096 #pitch value channels = @0x0D$1 loop_start_sample = @0x0E$1 * interleave / 2 / 0x10 * 28 loop_flag = @0x0F$1 padding_size = auto-empty loop_end_sample = data_size num_samples = data_size
# offset-to-header within TSNDDRVC.IRX at around 0xAC3C + position * 0x18 BGM001.XAG: 0x00 BGM002.XAG: 0x18 BGM000.XAG: 0x30 BGM003.XAG: 0x48 BGM008.XAG: 0xA8 BGM010.XAG: 0xD8 BGM011.XAG: 0xF0 BGM012.XAG: 0x108 PAD.XAG : 0x150 JIN002.XAG: 0x168 JIN003.XAG: 0x180
header_file = GM1.IDX body_file = GM1.STZ subsong_count = 394 #last doesn't have size though subsong_spacing = 0x04 subfile_offset = (@0x00 & 0xFFFFF) * 0x800 subfile_extension = seb subfile_size = ((@0x04 - @0x00) & 0xFFFFF) * 0x800
header_file = bgm_S01.srt name_table = .names.txt base_offset = @0x0c:BE base_offset = base_offset + @0x08:BE + name_value base_offset = base_offset + @0x00:BE - name_value codec = NGC_DSP channels = 2 interleave = half_size sample_rate = @0x08:BE loop_flag = @0x04:BE sample_type = bytes loop_start_sample = @0x10:BE loop_end_sample = @0x14:BE num_samples = @0x18:BE coef_offset = 0x20 coef_spacing = 0x40 coef_endianness = BE
st_s01_00a.ssd: 0*0x04 st_s01_00b.ssd: 1*0x04 st_s01_00c.ssd: 2*0x04 st_s01_01a.ssd: 3*0x04 st_s01_01b.ssd: 4*0x04 st_s01_02a.ssd: 5*0x04 st_s01_02b.ssd: 6*0x04 st_s01_02c.ssd: 7*0x04
#alt from above with untouched folders header_file = Sound/BGM/bgm_S01.srt body_file = snd/stream/st_s01_00a.ssd name_table = .names.txt base_offset = @0x0c:BE base_offset = base_offset + @0x08:BE + name_value base_offset = base_offset + @0x00:BE - name_value codec = NGC_DSP channels = 2 interleave = half_size sample_rate = @0x08:BE loop_flag = @0x04:BE sample_type = bytes loop_start_sample = @0x10:BE loop_end_sample = @0x14:BE num_samples = @0x18:BE coef_offset = 0x20 coef_spacing = 0x40 coef_endianness = BE
*snd/stream/st_s01_00a.ssd: 0*0x04 *snd/stream/st_s01_00b.ssd: 1*0x04 *snd/stream/st_s01_00c.ssd: 2*0x04 *snd/stream/st_s01_01a.ssd: 3*0x04 *snd/stream/st_s01_01b.ssd: 4*0x04 *snd/stream/st_s01_02a.ssd: 5*0x04 *snd/stream/st_s01_02b.ssd: 6*0x04 *snd/stream/st_s01_02c.ssd: 7*0x04 # uses wildcards for full paths from plugins
codec = ASF sample_rate = 22050 channels = 2 num_samples = data_size
codec = PCM16LE header_file = ALL_AUDIO.sfx body_file = ALL_AUDIO.sfx #header format # 0..0x100: garbage/info? # 0x100 table1 offset (points to audio configs w/ floats, etc) # 0x104 table1 count # 0x108 table2 offset (points to stream offsets for audio configs?) # 0x10c table2 count # 0x110 table3 offset (points to headers) # 0x114 table3 count # 0x118 table3 offset (points to stream offsets) # 0x11c table3 count # read stream header using table3 subsong_count = @0x114 base_offset = @0x110 subsong_spacing = 0xc8 name_offset = 0x00 #0xc0: file number base_offset = @0xc4 #absolute jump subsong_spacing = 0 #stop offsetting for next vals channels = @0xC0 sample_rate = @0xC4 data_size = @0xC8 #without header num_samples = data_size # read stream offset using table4 base_offset = 0 #reset current jump base_offset = @0x118 subsong_spacing = 0xc8 start_offset = @0xc4 + 0xc0
codec = PCM16LE header_file = EnglishStreamHeader.stm body_file = EnglishStreamData.stm #header format # 0..0x100: garbage/info? # 0x100 table1 offset (points to headers) # 0x104 table1 count # 0x108 table2 offset (points to stream offsets) # 0x10c table2 count # read stream header using table1 subsong_count = @0x104 base_offset = @0x100 subsong_spacing = 0xc8 name_offset = 0x00 #0xc0: file number base_offset = @0xc4 #absolute jump subsong_spacing = 0 #stop offsetting for next vals channels = @0xC0 sample_rate = @0xC4 data_size = @0xC8 #without header num_samples = data_size # read stream offset using table1 base_offset = 0 #reset current jump base_offset = @0x108 subsong_spacing = 0xc8 start_offset = @0xc4 + 0xc0
codec = PSX channels = 1 sample_type = bytes header_file = MUSICPS2.WAD body_file = MUSICPS2.WAD subsong_count = 0xC subsong_spacing = 0x30 sample_rate = 32000 base_offset = 0x70 start_offset = @0x14 + 0x380 num_samples = @0x18 data_size = num_samples loop_flag = auto #@0x10 is an absolute offset to another table, that shouldn't be affected by subsong_spacing name_offset_absolute = @0x10 + 0x270
#00: MWAV #04: flags? #08: subsongs #0c: data size #10: null #14: sizes offset #18: offsets table #1c: offset to tables? #20: header offset subsong_count = @0x08 # size table subsong_spacing = 0 base_offset = 0 base_offset = @0x14 subsong_spacing = 0x04 data_size = @0x00 # offset table subsong_spacing = 0 base_offset = 0 base_offset = @0x18 subsong_spacing = 0x04 start_offset = @0x00 # header (standard "fmt") subsong_spacing = 0 base_offset = 0 base_offset = @0x20 channels = @0x02$2 sample_rate = @0x04 codec = XBOX num_samples = data_size #todo: there are dummy entries