This component decodes modules, or music files combining both samples and sequence data and effects, using libopenmpt. Its version number is usually derived from the associated libopenmpt version, with a plus sign and trailing number if I add any extra commits following a library update. It supports files in the following formats: .MPTM, .MOD, .S3M, .XM, .IT, .669, .AMF, .AMS, .C67, .DBM, .DIGI, .DMF, .DSM, .DTM, .FAR, .IMF, .ICE, .J2B, .M15, .MDL, .MED, .MMS, .MT2, .MTM, .NST, .OKT, .PLM, .PSM, .PT36, .PTM, .SFX, .SFX2, .ST26, .STK, .STM, .STP, .ULT, .WOW, and .GDM. It also supports Ian Luck's .MO3 compressed format with MP3, Ogg Vorbis, or ADPCM samples, and it also supports arbitrary archive formats with the extensions of .MDZ, .MDR, .S3Z, .S3R, .XMZ, .ITZ, or .MPTMZ, assuming foobar2000's Archive Support plugin is installed, or any other archive plugin.

https://git.lopez-snowhill.net/chris/foo_openmpt/-/commits/master
https://lib.openmpt.org/doc/changelog.html

Note: The included "sinc" resampler is equivalent to the OpenMPT software's "sinc + lowpass", aka sinc plus anti-aliasing filter. The option to disable this filter is not currently exposed in the libOpenMPT API, and even if it were, it would be an implementation detail only relevant to the "sinc" resampling filter size.

• AMS: An upper bound for uncompressed sample size is now established to avoid memory exhaustion from malformed files.
• DMF: Support early format beta versions (in particular versions 1-4).
• MED: Also use octave wrapping in 8-channel mode for MMD0/MMD1 modules.
• MED: If 8-channel mode is activated, ignore BPM mode.
• MED: Emulate tempo commands F01 and F02 quirk.
• MED: Tempo commands below 32 BPM were interpreted as tempo slides.
• IMF: Instrument sample mapping was off by one octave, notable in the guitar part of Astaris by Karsten Koch.