TXTH is a simple text file with 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:
(filename.ext).txth.(ext).txth.txth
If found and parsed correctly, vgmstream will play the file as described.
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 .vag.txth or .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: .txth.
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.
Also check the examples section for some quick recipes, of varying complexity.
The .txth may be rejected if incorrect commands are found. Errors are shown in the console log (see USAGE guide), better try starting with a simple case from examples then add more complex commands until it fully works.
Extension must be accepted/added to vgmstream (plugins like foobar2000 only load extensions from an accepted list in formats.c), or one could rename to any supported extension (like .vgmstream), or leave the file extensionless. Before renaming consider reporting the unknown extension so it can be added to the list (so similar games benefit, as long as the extension is a good fit). Some plugins allow playing unknown extensions too.
Note that TXTH has lower priority than most (not all) vgmstream formats, by design. This means your .txth may be ignored if vgmstream thinks it can play your file better. If vgmstream plays your file somewhat off, rather renaming to force a .txth, report the bug so that and similar cases can be fixed. TXTH isn't meant to be a replacement of vgmstream's parsers, but a way to play cases that aren't a good fit be added directly to vgmstream.
If you put debug = 1 on top of the TXTP, vgmstream will ouput to the plugin/CLI's console some info about values being read, useful while testing more complex cases.
The file is made of lines with key = value commands describing a header. Commands are all case sensitive and spaces are optional: key=value, key = value, and so on are all ok, while Key = VaLuE is not. 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 interleave, sample_rate, channels and num_samples, or use the "subfile" feature.
The following can be used in place of (value) for (key) = (value) commands.
(number): constant number in dec/hex, unsigned (no +10 or -10).- Examples:
44100, 40, 0x40 (decimal=64)
- Examples:
(offset): read a value at offset inside the file, format being@(number)[:LE|BE][$1|2|3|4]@(number): offset of the value (required)- if
base_offsetis defined this value is modified (see later)
- if
: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)- Example:
@0x10:BE$2meansget big endian 16b value at 0x10
(field): uses current value of some fields (interleave,channels,start_offset,data_size,num_samples,subsong_count,subfile_size,base_offset,name_valueX, etc)subsongis a special field for current subsong
(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 [Fatal Frame (PS2), namCollection: Tekken (PS2)]
# - HEVAG Vita/PS4 ADPCM
# * For some Vita/PS4 games
# * Interleave is multiple of 0x10 (default)
#
# - DSP|NGC_DSP Nintendo GameCube ADPCM
# * For many GC/Wii/3DS/Switch 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)
# - 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_SB PCM 8-bit with sign bit
# * Variation with modified encoding
# * For few rare games [Sonic CD (SCD)]
# - PCM24LE PCM 24-bit little endian
# * For few rare games [100% Orange Juice (PC)-sfx]
# * Interleave is multiple of 0x3 (default)
# - PCM24BE PCM 24-bit little endian
# * For few rare games [Deadly Premonition (PS3)]
# * Interleave is multiple of 0x3 (default)
# - PCM_FLOAT_LE PCM 32-bit float little endian
# * For few rare games [Ikinari Maou (Switch)]
# * Interleave is multiple of 0x4 (default)
#
# - ULAW mu-Law 8-bit PCM
# * For few rare games [Burnout (GC)]
# * Interleave is multiple of 0x1 (default)
# - ALAW A-Law 8-bit PCM
# * For few rare games [Illwinter Game Design games: Conquest of Elysium 3 (PC), Dominions 3/4 (PC)]
# * Interleave is multiple of 0x1 (default)
# - DPCM_KCEJ DPCM 8-bit (KCE Japan)
# * For rare games [Metal Gear Solid 2 (PS2)-cutscenes]
#
# - 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
# - XBOX Xbox IMA ADPCM (mono/stereo)
# * For many XBOX games, and some PC games
# * Special interleave is multiple of 0x24 (mono) or 0x48 (stereo)
# - MS_IMA Microsoft IMA ADPCM
# * For some PC games
# * frame_size (or interleave) varies, often multiple of 0x100 [required]
# - APPLE_IMA4 Apple Quicktime IMA ADPCM
# * For some Mac/iOS games
# - IMA_HV High Voltage's IMA ADPCM
# * For some High Voltage Software PC games [NBA Hangtime (PC), NHL Open Ice (PC)]
#
# - MSADPCM Microsoft ADPCM (mono/stereo)
# * For some PC games
# * frame_size (or interleave) varies, often 0x2c/0x8c/0x100/0x400 and max 0x800 [required]
# * frame_size + interleave forces mono mode
#
# - AICA Yamaha AICA ADPCM (mono/stereo)
# * For some Dreamcast games, and some arcade (Naomi) games
# * Special interleave is multiple of 0x1
# - YMZ Yamaha YMZ263B/YMZ280B ADPCM (mono/stereo)
# * Variation of AICA
# * For rare arcade games [VJ: Visual & Music Slap (AC)]
# - CP_YM Capcom's Yamaha ADPCM
# * For rare Saturn games [Marvel Super Heroes vs Street Fighter (SAT)]
#
# - 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 (MP2: around 240 or 480, MP3: around 1152)
#
# - ATRAC3 Sony ATRAC3
# * For some PS2 and PS3 games
# * frame_size (or interleave) can be 0x60/0x98/0xC0 * channels [required]
# * Should set skip_samples (around 1024+69 but varies)
# - ATRAC3PLUS Sony ATRAC3plus
# * For many PSP games and rare PS3 games
# * frame_size (or interleave) 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 (around 2048+184 but varies)
#
# - XMA1 Microsoft XMA1
# * For early X360 games
# - XMA2 Microsoft XMA2
# * For later X360 games
#
# - AC3 AC3/SPDIF
# * For few PS2 games [Burnout (PS2)]
# * Should set skip_samples (around 256 but varies)
# * bytes-to-samples needs interleave, but only works for PS2-style AC3 (use sync work 0x72F8/0x770B rather than 0x0b77)
# - AAC Advanced Audio Coding (raw outside .mp4)
# * For some 3DS games and many iOS games
# * Should set skip_samples (typically 1024 but varies, 2112 is also common)
# - FFMPEG Any headered FFmpeg format
# * For uncommon games
# * May set skip_samples
#
# - OKI16 OKI ADPCM with 16-bit output (not VOX/Dialogic 12-bit)
# * For rare PS2 games [Sweet Legacy (PS2), Hooligan (PS2)]
# - OKI4S OKI ADPCM with 16-bit output and adjusted tables
# * For later Konami arcade games [Gitadora (AC), Metal Gear Arcade (AC)]
# - 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
# - TGC Tiger Game.com 4-bit ADPCM
# * For Tiger Game.com games
# - 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)]
# - XA CD-XA ADPCM (ISO 2048 mode1 streams without subchannel data)
# * For rare Saturn and PS2 games [Phantasy Star Collection (SAT), Fantavision (PS2), EA SAT videos]
# - XA_EA Electronic Arts XA ADPCM variation
# * For rare Saturn games [EA SAT videos]
codec = (codec string)
Changes the behavior of some codecs:
# - XBOX|EAXA: 0=standard (mono or stereo interleave), 1=force mono interleave mode
# - NGC_DSP: 0=normal interleave, 1=byte interleave, 2=no interleave
# - XMA1|XMA2: 0=dual multichannel (2ch xN), 1=single multichannel (1ch xN)
# - 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, and while optional (defaults described in the "codec" section) you'll often need it to get proper sound.
Roughly speaking interleave is the separation between data of each channel (or the size of a data chunk). For example interleave = 0x02 means there are 2 bytes of data from channel 1, then 2 bytes from channel 2, then bytes for other channels if any (this is common for PCM16LE, where 2 bytes = 1 sample). While interleave = 0x1000 works the same but means there is a lot more data of one channel before next channel:
interleave = 0x02: | ch1 | ch2 | ch1 | ch2 | ...
interleave = 0x08: | ch1 ch1 ch1 ch1 | ch2 ch2 ch2 ch2 | ch1 ch1 ch1 ch1 | ch2 ch2 ch2 ch2 | ...
Incorrect interleave will usually sound like audio is "fragmented" or noisy (since channel data is misinterpreted), but it's usually easy enough to try a few common values until it sounds right. Interleave needs to be a multiple of some value (PCM16LE is multiple of 0x02 so can't use interleave 0x05).
Special values:
half_size: sets interleave as data_size / channels automatically
interleave = (value)|half_size
Depending on the codec itself interleave has certain implications:
- mono-interleaved codecs: uses default value or value in
interleaveas described above - mono/stereo codecs: no default, setting interleave usually forces mono-interleaved mode (if known cases exists)
- codecs with frame sizes: if
interleaveis set butframe_sizeisn't set, it'll use the former asframe_size(see below) - other codecs: ignored
Mono/stereo codecs are a bit particular in that they have two modes. For 1 channel files mono mode is always used. But for stereo files, different games may have either mono-interleave data or stereo data:
- stereo file uses mono data: set interleave (forces mono mode with interleaved chunks)
- stereo file uses stereo data: don't set interleave (forces stereo mode with linear chunks)
It's technically possible that a game could use stereo mode with interleave for multichannel, but isn't handled at the moment.
Codecs with custom frame sizes (MSADPCM, MS-IMA, ATRAC3/plus) need frame_size. "frame size" is the amount of data the decoder needs as a single unit. Conceptually it's similar to interleave, so to simplify usage you may set interleave instead of frame_size.
It's possible though rare (seen in some MSADPCM files) that a game needs a frame_size then sets another interleave value, for example has frame_size 0x100 with mono interleave 0x400. This means it reads multiple smaller 0x100 for one channel up to 0x400, then next channel, etc.
frame_size = (value)
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).
Special values:
auto: calculate based on channels, interleave and data_size/start_offset
interleave_last = (value)|auto
Similar to the above, in rare cases the file starts with a different interleave (bigger or smaller), then uses another value.
For example, file has start_offset at 0x100, first interleave_first of 0x800 then interleave of 0x400.
In trickier cases, file at 0x100 has 0x10 garbage (before each channel data), then data up to 0x800, then interleave of 0x800. So interleave sizes are consistent, but first block has less data. Here we need to set interleave_first_skip = 0x10 so block sizes can be properly calculated and garbage skipped. Notice that if file was 4ch this means total garbage of 0x40 ((0x10 garbage + 0x7F0 data) * 4).
Be aware that certain features like autodetecting PS-ADPCM loop points may not handle interleave_first at the moment.
interleave_first = (value)
interleave_first_skip = (value)
Validates that id_value (normally set as constant value) matches value read at id_check. 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_check is found. id_offset can be used as an alt for id_check
id_value = (value)
id_check = (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 framesauto-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.
Accepted values:
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- Value is internally converted from blocks to bytes first:
bytes = (value * interleave*channels)
- Value is internally converted from blocks to bytes first:
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_start and loop_end instead as aliases of loop_start_sample and loop_end_sample (no difference).
To activate loops you need to define both loop_start and loop_end (rather than say, only defining start and defaulting end to num_samples). This is to make the .txth more descriptive, and avoid ambiguity in cases where start value is 0. See loop settings below to fine tune when to loop.
Special values:
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 not bigger than num_samples.
Special values:
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 loop_behavior before loop_flag:
default: values 0 or 0xFFFF/0xFFFFFFFF (-1) disable looping, but not 0xFF (loop endlessly)negative: values 0xFF/0xFFFF/0xFFFFFFFF (-1) enable loopingpositive: values 0xFF/0xFFFF/0xFFFFFFFF (-1) disable looping (0 also enables it)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.
Supported codecs: ATRAC3/ATRAC3PLUS/XMA/FFMPEG/AC3/AAC
skip_samples = (value)
DSP needs a "coefs" list to decode correctly. These are 8*2 16-bit values per channel, starting from coef_offset.
They typically look like positive-negative values one after other (0x0nnn 0xFnnn 0x...). Usually each channel uses its own list, so we may need to set coef_spacing (separation per channel), often 0x20 (16 values * 2 bytes). Channel N coefs are read at offset coef_offset + coef_spacing * ch.
Those 16-bit coefs can be little or big endian (BE in GC/Wii, LE in 3DS/Switch). Set coef_endianness directly or in an offset value where 0=LE, >0=BE. This also allows adding a _split suffix, that means coefs are divided into 8 positive then 8 negatives (instead of the usual 1 positive, 1 negative up to 16), as found in a few Capcom games.
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_spacing and coef_endianness though.
coef_offset is adjusted by base_offset and subsong_spacing. If offset points to some absolute offset that doesn't depend on subsong, set first offset_absolute = 1.
coef_offset = (value)
coef_spacing = (value)
coef_endianness = BE|LE|BE_split|LE_split\(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 hist_offset.
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).
Allowed values:
- (filename): open any file, subdirs also work (dir/filename)
- *.(extension): opens with same name as the "base" file (the one you open, not the .txth) plus another extension
- null: unloads file and goes back to defaults (body/header = base file).
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.
Instead of 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)
A experimental field is subsong_sum = (value), that sums all subsong values up to current subsong. Mainly meant when offsets are the sum of subsong sizes: if you have a table of sizes at 0x10 for 3 subsongs, each of size 0x1000, then subsong_sum = @0x10 for first subsong sums 0x0000, 0x1000 for second, 0x2000 for third (can be used later as start_offset = subsong_sum).
Sets the name of the stream, most useful when used with subsongs. TXTH will read a string at name_offset, with name_size characters.
name_size defaults to 0, which reads until null-terminator or a non-ascii character is found.
name_offset is adjusted by base_offset and subsong_spacing. If offset points to some absolute offset that doesn't depend on subsong, set first offset_absolute = 1.
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 channels or 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. Or maybe 0x100 of useless header + 0x10000 of valid data. Chunk settings allow vgmstream to play valid chunks while ignoring the rest (read 0x10000 data, skip rest).
File is first "dechunked" before being played, so other settings work over this final file (start_offset would be a point within the internal dechunked" file). Use combinations of chunk settings to make vgmstream "see" only actual codec data.
Main settings:
chunk_count: total number of interleaved chunks (ex. 3=3 interleaved songs)chunk_start: absolute offset where chunks start (normally 0x00)chunk_size: amount of data in a single chunk (ex. 0x10000)
Optional settings (set before main):
chunk_number: first chunk to start (ex. 1=0x00000, 2=0x10000, 3=0x20000...)- If you set
subsong_countandchunk_countfirst,chunk_numberwill be auto-set per subsong (subsong 1 starts from chunk number 1, subsong 2 from chunk 2, etc)
- If you set
chunk_header_size: header to skip before chunk data (part of chunk_size)- If size is 0x1000 and header_size 0x100, data_size is implicitly set to 0xF00
chunk_data_size: actual data size (part of chunk_size, rest is header/padding)- If size is 0x1000 and data_size 0x800 last 0x200 is ignored padding.
Dynamic settings (set before main, requires chunk_header_size):
chunk_value: ignores chunks that don't match this value at chunk offset 0x00 (32-bit, inchunk_endianness). Can be used to ignore video chunks in movie files, for example.chunk_size_offset: reads chunk size at this offset, in header (32-bit inchunk_endianness). For chunks of dynamic sizes (no need to setchunk_sizeas will be ignored). Includes header size.chunk_data_size_offset: same, for data sizes (not including headers). Note that you can use header+data+size configs to handle padding between blocks.chunk_endianness: sets endianness of the above two settings.
For technical reasons, "dechunking" activates when setting all main settings, so set optional config first. Note that chunk_size is static (read once from a fixed offset) while chunk_size_offset is dynamic (read on every chunk).
# optional - fixed
chunk_number = (value)
chunk_header_size = (value)
chunk_data_size = (value)
# optional - dynamic
chunk_value = (value)
chunk_size_offset = (value)
chunk_data_size_offset = (value)
chunk_endianness = LE|BE
# main
chunk_count = (value)
chunk_start = (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)
If you set name_table = * it'll default to .names.txt (most common).
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 (bgm05: 5*0x010). You can also set a codec string too (bgm01: PCM16LE; codec = name_value).
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.
This function also works with subsongs, with this syntax:
# for subsong 1 in filename
(filename)#1: (value 1)
# for any subsong 2
#2: (value 2)
...
Then subsong N would automatically use its own name_value.
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 name_table, like base_offset = name_value, channels = @0x04.
It also allows parsing formats that set offsets to another offset, by "chaining" base_offset. With base_offset = @0x10 (pointing to 0x40) then 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 0x20
base_offset = (value)
TXTH can't do conditions (if) but sometimes you have have variations of the same format in the same dir. You can set multiple .txth files to try until one works. Use id_value/id_check to reject wrong .txth (otherwise the first one will be selected).
multi_txth = (filename), (filename), ...
For example:
multi_txth = .2ch.txth, .4ch.txth
.2ch.txth
id_value = 2
id_check = @0x00 # 2ch only
... #some settings for stereo
.4ch.txth
id_value = 4
id_check = @0x00 # 4ch only
... #different settings for 4ch
As an interesting side-effect, you can use this to force load .txth in other paths. For example it can be useful if you have files in subdirs and want to point to a base .txth in root.
multi_txth = ../.main.txth
Most commands are evaluated and calculated immediately, every time they are found. This is by design, as it can be used to adjust and trick for certain calculations.
It does make TXTHs a bit harder to follow, as they are order dependent, 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 data_size in num_samples and loop_end_samples they must be alone to trigger.
data_size = @0x100
num_samples = data_size * 2 # doesn't transform 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:
sample_type = bytes
num_samples = @0x10
sample_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 num_samples and loop_end_sample and will always convert as bytes-to-samples, though.
Priority is left-to-right only, due to technical reasons it doesn't handle proper math priority. 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 omitted 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:
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
...
codec = PCM8
sample_rate = 32000
channels = 1
start_offset = 0
num_samples = data_size
codec = DVI_IMA
interleave = 0x80
start_offset = 0x00
channels = 2
sample_rate = 44100
num_samples = data_size
codec = PSX
interleave = 0x10
sample_rate = 22050
channels = 1
padding_size = auto-empty
num_samples = data_size
codec = PSX
interleave = 0x2000
channels = 2
sample_rate = 48000
num_samples = data_size
interleave_last = auto
codec = PSX
channels = @0x8 + 1
sample_rate = 48000
interleave = 0x1000
interleave_first = 0x2000
interleave_first_skip = 0x10
padding_size = auto-empty
num_samples = data_size
#@0x00 interleave?
#@0x04 number of 0x800 sectors
# cutscene/voices use mono interleave for stereo tracks, set both frame_size and interleave
codec = MSADPCM
frame_size = 0x400
interleave = 0x800
sample_rate = @0x06:BE$2
channels = @0x08$1
# 0a: codec (11=msapdcm)
start_offset = 0x10
num_samples = data_size
# first check that value at 0x00 is really 0x00000000 (rarely needed though)
id_value = 0x00000000
id_check = @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 = XBOX
codec_mode = 1 #interleaved XBOX
interleave = 0xD800
channels = 12
sample_rate = 44100
start_offset = 0x00
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 = 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 = (((@0x0F$1 & 0x7F) * 256) + @0x0E$1) * interleave / channels / 0x10 * 28
loop_flag = @0x0F$1
padding_size = auto-empty
loop_end_sample = data_size
num_samples = data_size
.names.txt
# 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
.names.txt
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
.names.txt
*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 = CP_YM
sample_rate = 24000
channels = 2
#loops are in MM.BIN, table at 0x80700 + id*4 - 0x06018B00
#-- for files with video, remove if demuxed
chunk_count = 1
chunk_start = 0x00
chunk_data_size = 0x8000
chunk_size = 0x25800
#--
codec = PCM8_SB
channels = 1
sample_rate = 32768 #32500?
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
codec = XMA2
#00: id (0x819A584D)
#04: null
#08: fmt header offset
#0c: fmt size
start_offset = @0x10:BE
data_size = @0x14:BE
base_offset = @0x08:BE
channels = @0x02:BE$2
sample_rate = @0x04:BE
num_samples = @0x24:BE
loop_end = @0x24:BE #0x2c?
loop_start = @0x28:BE
loop_flag = @0x2c:BE
# once extracted from bigfiles there are 2 types of files with hardcoded settings,
# so we need 2 .txth
multi_txth = .type2.txth, .type4.txth
.type2.txth
id_value = 2
id_check = @0x1f80
codec = PSX
channels = 2
sample_rate = @0x1F44
interleave = 0x10000
start_offset = 0x00
chunk_count = 1
chunk_start = 0x1f84
chunk_data_size = 0x20000
chunk_size = 0x21000
padding_size = auto
num_samples = data_size
.type4.txth
id_value = 4
id_check = @0x1F80
name_table = .names.txt
subsong_count = 2
base_offset = name_value1
codec = PSX
channels = 2
sample_rate = @0x04
interleave = name_value2
start_offset = 0x00
chunk_count = 1
chunk_start = 0x1f84
chunk_header_size = name_value3
chunk_data_size = name_value4
chunk_size = 0x21000
padding_size = auto
num_samples = data_size
# base_offset = 0x1F40
# 00: stream size without padding
# 04: stream 1 sample rate
# 08: stream size without padding (same)
# 0c: stream 2 sample rate (same)
# repeat for stream 3 and 4 if any
# 1/2 are mini streams (interleave 0x800, chunk size 0x1000, padding 0x20000)
# 1/2 are standard streams (interleave 0x10000, padding 0x1000, chunk size 0x20000)
# (mini streams are muffled versions of the standard ones)
.names.txt
# base_offset, interleave, chunk_header_size, chunk_data_size
#1: 0x1F40, 0x800, 0x00, 0x1000
#2: 0x1F50, 0x10000, 0x1000, 0x20000
# "dechunks" videos with dynamic chunks realtime and plays audio only
codec = IMA
channels = 2
sample_rate = @0x20
# each chunk is 0x00: ID + 0x04 data size (not including header)
chunk_number = 1
chunk_header_size = 0x08
chunk_value = 0x00414D46 #"FMA\0" LE
chunk_data_size_offset = 0x04
chunk_count = 1
chunk_start = 0x28 #first chunk after header
num_samples = data_size