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	diff --git a/util/sdl/sound/CHANGELOG b/util/sdl/sound/CHANGELOG
	deleted file mode 100644
	index 1db243b8..00000000
	--- a/util/sdl/sound/CHANGELOG
	+++ /dev/null
	@@ -1,376 +0,0 @@
	-/*
	- * CHANGELOG.
	- */
	-
	-04202008 - Upped version to 1.0.3 (brown paper bag release for soname bug).
	-04192008 - Apparently MICRO_VERSION in configure.in doesn't do what I think;
	- reset for binary compatibility (thanks, Hans!).
	-04182008 - Include <math.h> in shn.c.
	-04172008 - Look for Speex includes in new directory. Converted all text
	- encoding from ISO-8859-1 to UTF-8. Fixed "make dist" script for
	- dealing with Subversion instead of CVS. Added Speex to the README.
	- Upped version to 1.0.2.
	-04112008 - Check if Speex header has bogus data (CVE-2008-1686).
	-08062007 - Updated my email address.
	-07152007 - Minor correction in Timidity resampling code (Thanks, Sam!).
	-07062007 - Fixed uninitialized buffer in mpglib. (Thanks, Phil!).
	-10292006 - Fixed bogus memory dereference when SMPEG fails init (thanks, Chris!)
	-10272006 - FLAC 1.1.3 breaks their API _again_, so we try to do the right
	- thing at build time. (Thanks, Josh!).
	-05122006 - Patched to get mpglib compiling again (thanks, Sam!).
	-12172005 - Fixed gcc4 whining in playsound_simple.c.
	-12062005 - Trimmed a bunch of junk out of the build system, and now it works
	- on Mac OS X again.
	-10122005 - Check for libmodplug headers in two possible places (thanks, Tyler!)
	-10012005 - Added playsound_simple.c.
	-05302005 - Backport from devtree: Fixed automake nonsense.
	-11122004 - Backport from devtree: fix .voc decoder crash on file open.
	-05082004 - Fixed "bootstrap" to work with MacOSX.
	-05072004 - Backed out some commits, converted repository to Subversion, and
	- branched off to a 1.1.0 development tree. Changed MikMod URL...old
	- one is now a porn site. :(
	-10252003 - VOC decoder was broken. Now it isn't.
	-10142003 - Build system fix: acinclude.m4 had some word wrapping badness.
	-10122003 - Fixed "make dist" behaviour to not packaged generated docs, and
	- made sure other files are always packaged, regardless of config.
	- Upped version to 1.0.1.
	-10102003 - Changed some SDL_Error()s to __Sound_SetError() in new DLS code
	- to fix linking issues.
	-10052003 - Fixed memory corruption when freeing DLS instruments,
	- and bug when timidity is initialized multiple times (Thanks, Sam!).
	-09252003 - Sam Lantinga added support for DLS instruments to the MIDI decoder.
	-09132003 - Happy September. Added Speex (.spx) decoder.
	-08052003 - Fixed MIDI decoder on bigendian systems.
	-03102003 - Never actually created samplelist_mutex (Thanks, Glenn Maynard!).
	-01302003 - Patches to make SDL_sound more Visual C happy (Thanks, Eric!).
	-01122003 - Fix to smpeg.c's rewinding code (Thanks, Eric). Put Visual C 6
	- project files in CVS, without external binaries (Thanks, Eric).
	-12212002 - Fixed ogg.c to decode a full buffer at a time instead of one ogg
	- packet per call, and mikmod has a check during initialization to
	- prevent a clash with SDL_mixer (Thanks, Eric).
	-12092002 - Changed Sound_Init()'s call to SDL_Init() to SDL_InitSubSystem(),
	- to prevent unwanted use of the SDL parachute (thanks, Glenn).
	-10092002 - Fixed a "make dist" issue and upped version to 1.0.0! Woohoo!
	-09302002 - libFLAC broke their API (again!) for version 1.0.4. That was the
	- last straw. I ripped the version detection and obsolete FLAC
	- support out, so you need libFLAC 1.0.4 for that decoder now (and
	- they'll probably break the API again for 1.0.5. Argh).
	-09262002 - Happy September. Fixed SDLCALL issues in SDL_sound.h, so it should
	- work with Win32/WinCE builds again. I hope. Merged latest altcvt
	- from Frank into CVS.
	-08222002 - Borland project files in CVS, thanks to Dominique Louis. There are
	- project files for C++ Builder 6 (Windows), C++ Builder for Linux
	- (aka Kylix 3) and Borland's C++ Command line compiler.
	-08172002 - Timidity memory leak cleanup by Torbjörn.
	-07292002 - Valgrind cleanups; memory leak patches, etc.
	-07212002 - done_flag was not being reset between files in playsound, so the
	- first file would playback, and then any following tracks in a given
	- run would "finish" immediately. Fixed.
	-07132002 - More altcvt fixes from Frank Ranostaj.
	-07122002 - Changed inline keyword to compile universally.
	-07102002 - Fixed a bug in command line handling in playsound.c. Fixes from
	- Torbjörn and myself to get flac.c friendly between versions of
	- libFLAC. Mutex'd a potential race condition in decoders/modplug.c.
	- FIXME cleanups here and there.
	-07092002 - Fixed typo in documentation (SDL_sound.h).
	-07052002 - Cleaned up some stuff in playsound.c, removing some FIXMEs.
	- Commandline validation is improved, too. FIXME removal in
	- voc.c; should report i/o errors correctly now. Changed DECLSPEC
	- to SNDDECLSPEC to prevent SDL conflict, and added SDLCALL support.
	- Removed all instances of Sound_SetError()...now they are either
	- __Sound_SetError or BAIL*_MACRO.
	-07022002 - Added WinCE support pack to website, updated INSTALL with CE info.
	- More altcvt fixes from Frank Ranostaj.
	-07012002 - Fixed configure.in to work around bug in older autoconfs. Started
	- merging Tyler's WinCE (PocketPC) port. Added checks for assert.h
	- and signal.h to configure.in/config.h.in, and #if HAVE_*_H checks
	- where appropriate in the code. Moved #include <assert.h> (along
	- with the HAVE_ASSERT_H check) to SDL_sound_internal.h, and removed
	- unnecessary #includes from the individual source files. Added
	- "md_reverb = 1;" to MIKMOD_init(). Modplug got some WinCE-specific
	- setting tweaks, and some settings maintanance code. configure.in
	- checks if setbuf() is available.
	-06292002 - More altcvt fixes from Frank Ranostaj...mostly working now?
	-06252002 - More altcvt fixes from Frank Ranostaj.
	-06132002 - Patch from Torbjörn to fix stereo AIFF files.
	-06212002 - More altcvt fixes from Frank Ranostaj.
	-06132002 - Patch from Torbjörn to make the WAV decoder more tolerant.
	-06122002 - Committed some altcvt enhancements from Frank Ranostaj.
	-06112002 - Fixed some debug messages in smpeg.c and mpglib.c.
	-06072002 - Manpages! Finally installed Doxygen and scratched together a
	- Doxyfile. After some revision to physfs.h, we've got a rather
	- nice API reference.
	-06062002 - Added URLs for official and unofficial versions of ModPlug in
	- decoders/modplug.c. Cleaned up some FIXMEs.
	-05222002 - Torbjörn sent in some more fixes for altcvt: mono to stereo
	- conversion works, now.
	-05222002 - Torbjörn sent in some initial cleanups and fixes for altcvt, and
	- fixed a bug in playsound when not all three of --rate, --channels
	- and --format are specified.
	-05202002 - Some .cvsignores from Max and me. Added a seek implementations for
	- the SMPEG, ogg, aiff, wav-adpcm, voc, and au decoders. Added a seek
	- stub to quicktime.c. playsound now takes milliseconds in the seek
	- lists: --seek "00:00:400" or whatnot. Corrected playsound's usage
	- text. Other au.c cleanups for extra robustness. Added an
	- experimental audio converter that Frank Ranostaj sent to the SDL
	- mailing list about a month ago: enable it with --enable-altcvt at
	- configure time, but be warned that it doesn't work very well right
	- now.
	-04292002 - Darrell Walisser updated the Mac Classic and OS X project
	- files, fixed some portability issues, and added an
	- experimental decoder that uses Apple's QuickTime libraries
	- (see decoders/quicktime.c). I've included the Mac project files
	- in CVS, now. Removed all use of alloca() from playsound.
	-04242002 - Added --seek option and bugfixes to playsound.c. Torbjörn comes
	- through with seek support for the FLAC, MIDI, and ModPlug
	- decoders (and some stub code for MikMod), and a bugfix for sample
	- flag manipulation in the base library (and his own --seek code for
	- playsound, which unfortunately we're not using).
	-04232002 - Cleaned up the playsound command line handling. Most command line
	- options (--rate, --format, --predecode, etc) are specified per-file
	- and reset to their defaults after each sample is played back.
	- --loop now takes a numeric argument: --loop 2 will playback the
	- sample three times (one playback and two loops). Added Darrell
	- to the playsound credits.
	-04212002 - Initial work to add a Sound_Seek() API. Removed the NEEDSEEK
	- sample flag (replaced it with CANSEEK). Hack to change the internal
	- Sound_SetError() function to __Sound_SetError(). Added internal
	- function __Sound_convertMsToBytePos().
	-04082002 - Cleaned up the archive support in playsound a little bit, and
	- fixed a PhysicsFS bug in the process.
	-03252002 - Win32 patches and fixes from Tyler Montbriand: handled "inline"
	- keyword, fixed SNDDBG macros in mpglib, and renamed a conflicting
	- file (decoders/mpglib/common.c to decoders/mpglib/mpglib_common.c).
	-03172002 - Removed an unneeded #include in mpglib that broke build on BeOS.
	- mpglib seems to work find on BeOS. Reworked some of mpglib.c so we
	- can determine the audio format when accepting the data stream. Some
	- other minor cleanups here and there.
	-03162002 - Tied the PhysicsFS code into the build system (code disabled if
	- physfs not found or --disable-physfs passed to ./configure.)
	-03152002 - Added PhysicsFS support to playsound, so you can play sound files
	- that are in ZIP files without unzipping them. Needs to be merged
	- into build system (I was just testing my PhysFS->RWops glue code).
	-03142002 - Changed configure script's --enable-vorbis to --enable-ogg. Removed
	- global state variable from mpglib, so it should be reentrant now
	- (patches sent to mpglib's actual maintainer). playsound can now
	- read from stdin.
	-03102002 - Added a FIXME note to decoders/mpglib.c. playsound now reports
	- errors in the thread where they occured, which also fixes a double
	- report of errors during predecoding. Removed all calls to exit() in
	- mpglib. These calls now report errors correctly to SDL_sound, which
	- passes them on to the application (patch also sent to mpglib's
	- actual maintainer). Replaced all stderr chatter in mpglib with
	- Sound_SetError() calls.
	-03072002 - decoders/mpglib.c now disregards ID3 tags instead of passing them
	- on as valid MP3 data to mpglib. Added some (buggy) example code for
	- adjusting an audio stream's volume (via the new --volume command
	- line in playsound).
	-03032002 - Fixed mpglib's build configuration to include general build flags
	- so that things like --enable-debug work as expected.
	-02212002 - Changed SMPEG's URL to point to the icculus.org site. Added an
	- mpglib decoder (internal to SDL_sound; relies on no external libs)
	- and changes mp3.c to smpeg.c (and other associated things).
	-02112002 - Committed a patch from Torbjörn to fix incorrect memory accesses
	- in the Timidity code. Changed the magic number in the AU decoder
	- to be bigendian (seems appropriate). Updated README for
	- completeness, and TODO for accuracy. Darrell sent in updated
	- MacOS X Project Builder files (on the website).
	-02072002 - Committed a patch Torbjörn sent in awhile ago for preventing
	- confusion with Timidity++-specific stuff in the timidity.cfg file.
	- Tyler Montbriand sent in an updated Visual C package.
	- Updated SDL_sound.h's comments a little. Upped version to 0.1.5.
	-02052002 - Fixed a cleanup I broke last night. Added CWProject.sit to the
	- EXTRA_DIST section of Makefile.am, and updated the README with
	- MacOS (9/X) install instructions.
	-02042002 - Darrell Walisser submitted some cleanups and CodeWarrior project
	- files for MacOS 9. Sweet!
	-01232002 - Max fixed decoders/Makefile.am to work with seperate build
	- directories, and corrected some dates in this file.
	-01192002 - Torbjörn sent in patches implementing the rewind method for the
	- rest of the decoders except shn.c, for which I added a kludged
	- implementation. Added more info to the README. Hunted down the
	- reason why SMPEG can't decode before calling SDL_OpenAudio(), and
	- it can't be fixed without a change to SMPEG (not MY fault! :) ).
	- Made ModPlug take priority over MikMod when selecting a decoder.
	- Mutex-protected the internal samples list, and fixed some bugs in
	- the management of that list. Changed some stuff to use uniform
	- coding conventions.
	-01182002 - SDL_sound/playsound builds and runs on BeOS now. Fixed an assertion
	- bug I introduced yesterday.
	-01172002 - Implemented Sound_Rewind(), and added a --loop command line to
	- playsound for testing. Rewrote the audio callback to handle looping
	- with both predecoded and streamed samples. Most of the decoders
	- just have an assert(0) in their internal rewinding method at this
	- point. I implemented the WAV, VOC, AU, AIFF, and RAW ones, for now.
	- (...and skeleton.c, for what that's worth.) A few tweaks in the
	- core API implementation to fix unlikely but possible leaks.
	-01112002 - Mattias Engdegård sent in an .AU decoder. Nice! He also tweaked
	- playsound to try and wait until SDL has completed playing a given
	- sound before closing the audio device. Changed a macro in
	- decoders/shn.c to be more uniform with the other decoders.
	- SDL_sound error messages are now maintained on a per-thread basis,
	- and do not interfere with SDL_[GS]etError() anymore.
	-01112002 - Committed the rest of Torbjörn's MOD patches, to clean up file
	- extension handling.
	-01092002 - Torbjörn comes through with a ModPlug-based decoder, which should
	- work nicely for decoding multiple .MODs at once. Now we need to
	- figure out what to do with two decoders that can decode the same
	- file. For now, if you explicitly want either MikMod or ModPlug, you
	- should explicitly enable one decoder and disable the other on the
	- configure command line ("--enable-modplug --disable-mikmod", for
	- example), otherwise configure will try to sort out the best one for
	- your system. Choice is a wonderful thing. :)
	-01042002 - Forgot to bump playsound's version to match SDL_sound's. Fixed.
	- Added some notes to the top of COPYING about other libraries, etc.
	- A real MIDI decoder (using a hacked version of the hacked version
	- of Timidity from SDL_mixer) is now in place and working well,
	- thanks to Torbjörn.
	-01012002 - Happy New Year. Added some debug output to wav.c for future
	- codecs (GSM comes to mind). Fixed the SMPEG decoder's URL to point
	- to Loki's webpage.
	-12302001 - Upped version to 0.1.4.
	-12272001 - Added --audiobuf and --decodebuf options to playsound to make
	- tracking down a bug in the ADPCM decoder easier (plus, it could
	- help for benchmarking, etc later on...). Found a printf() bug in
	- playsound (extra comma in there...). ADPCM decoder appears to be
	- functional now. Tried to add ElectricFence support to
	- configure.in, and failed. All this libtool/autoconf stuff makes my
	- head hurt.
	-12262001 - Changed remaining references to the "LICENSE" file into "COPYING".
	- Work progresses on the ADPCM-compressed .wav decoder. Updates to
	- the documentation in SDL_sound.h. Hhmm...find_chunk() in wav.c was
	- badly broken. Fixed.
	-12162001 - FLAC decoder now checks for the magic number unless the file
	- extension is recognized. This was changed back because searching
	- for metadata, while probably more effective, is VERY expensive (and
	- useless) on non-FLAC streams.
	-12052001 - Put our names in a "--credits" option in playsound, and put the
	- standard GNU disclaimers in there too, for good measure. Renamed
	- LICENSE to COPYING to match GNU standards more closely (and to
	- end Max's torment. :) ) Tweaks to wav.c, and work on aiff.c to
	- make it easier to support multiple audio formats (for compression
	- handling later down the road).
	-11302001 - Torbjörn and I make Sound_DecodeAll() more robust: checks for
	- previous decoding failures and sets an appropriate error, handles
	- decoders that change their buffers on the fly (such as the FLAC
	- decoder), and deals with out-of-memory conditions more gracefully.
	-11252001 - (With thanks to Andreas Umbach for pointing it out) Fixed some
	- problems with Sound_DecodeAll(). For local testing of this bug,
	- added a --predecode command line to playsound. Minor fixes to
	- theoretical bugs in Sound_FreeSample(). playsound no longer
	- buffers stdout and stderr. Updated Sound_DecodeAll()'s comments in
	- SDL_sound.h ...
	-11192001 - FLAC decoder cleanups from Torbjörn.
	-11092001 - Torbjörn fixes playsound's audio callback after I broke it, again.
	- A bug in configure.in was preventing SMPEG from being used unless
	- --enable-debug was set; fixed. Changed this file to list latest
	- changes first. Torbjörn submitted a FLAC decoder that utilizes
	- libFLAC (http://flac.sf.net/). Cool.
	-11012001 - API COMPATIBILITY BREAKAGE: Decoders can now list multiple file
	- extensions each. Playsound has been updated to handle this.
	- Playsound now registers a SIGINT handler, so you can skip tracks
	- and/or abort the way that mpg123 does.
	-10232001 - Rewrote playsound.c's audio_callback() to no longer need the
	- overflow buffer hack, which streamlines it a little and trims the
	- memory requirements for playsound by about 16 kilobytes.
	-10172001 - Torbjörn catches a problem with the overflow buffer in playsound's
	- audio callback.
	-10152001 - Torbjörn sends in a default sample format for the MIDI decoder,
	- and the starts of the audio conversion funcitonality (ripped
	- from SDL). Officially released 0.1.3. Added LICENSE and
	- CHANGELOG to the distribution. (Again, from Torbjörn) added in
	- the start of a tweaked audio converter.
	-10122001 - Torbjörn Andersson submitted command line enhancements to
	- playsound, and I cleaned up the --help output.
	-10092001 - Patches to shn.c for Visual C compatibility. Visual C project files
	- available from the website. Changed Corona688 to Tyler Montbriand
	- in CREDITS. Upped version to 0.1.3.
	-10082001 - Restructured decoders/wav.c to allow for multiple formats, and
	- put the start of a handler for the ADPCM format in place.
	-10072001 - Changed the way decoders/mod.c handles samplerate so that it should
	- work universally. This isn't an ideal solution, but it's probably
	- the best we can do without rewriting mikmod. Made a change to ogg.c
	- for portability: changed an int64_t to ogg_int64_t.
	-10062001 - Made a change to SDL_sound.c for compiling on non-GNU toolchains.
	-10052001 - Removed #include "SDL_endian" from aiff.c.
	-10042001 - Changed some #if (defined SOUND_SUPPORTS_*) lines to
	- #ifdef SOUND_SUPPORTS_* in voc.c and shn.c, for consistency with
	- the other decoders.
	-10032001 - After hours of tracking down a bogus pointer, the SHN decoder works!
	- I can die happy. :) Max placated me with an --enable-debug option
	- so I could stop my whining. Other autoconf goodies (such as
	- reenabling -Werror for debug builds, etc). Torbjörn brings in a
	- MIDI decoder, which reads from a Timidity process through a pipe.
	- Changed playsound to open the audio device to match the properties
	- of each sound file, which results in less conversion (and therefore,
	- more chance of correct playback).
	-10022001 - Changed a comment in mod.c to not refer to "the mikmod
	- directory" anymore. Committed Torbjörn's patch for MP3 detection.
	- (better late than never). __Sound_strcasecmp() now handles NULL
	- strings gracefully, fixing the crash with "playsound bootstrap".
	- More work on the SHN decoder.
	-10012001 - Fixed a memory leak that Torbjörn found in the MOD decoder.
	-09252001 - More autoconf work. Gave Max Horn write access to the CVS
	- repository, so I don't drive him nuts tweaking this thing. :)
	- Fixed a const complaint and some other stuff needed for compilation
	- under Visual C++ 6.0 (no, it isn't ported yet). Put the SHN source
	- in CVS, even though it isn't ready (and doesn't even compile). Do
	- NOT enable it in your build!
	-09242001 - Thank goodness, Torbjörn came through with the MP3 fix. Apparently
	- SMPEG mixes each chunk of decoded data with whatever is already
	- in the buffer you give it. I hate that. I'm going to patch SMPEG
	- to let the programmer enable and disable that behaviour in a given
	- (SMPEG *), since it's just a CPU eater in this case. The _D(())
	- macro is now SNDDBG(()), since _D is taken on MacOS X's version of
	- gcc (which was bound to happen on some platform sooner than later
	- anyhow). Renamed test_sdlsound to playsound, and made it more
	- robust in general: fixed potential overflow in audio_callback,
	- made it chatter less, made it take multiple files and some other
	- command lines. Initial autoconf support, thanks to Max Horn.
	-09222001 - Torbjörn Andersson strikes again, with a collection of patches.
	- First, some cosmetic tweaks for decoders/aiff.c. Next, a MOD player
	- based on MikMod. This inspired me to add two more methods to
	- Sound_DecoderFunctions: init() and quit(). Third, a fix to
	- decoders/mp3.c so that SMPEG won't claim every stream it sees, MP3
	- or not. I removed the multiple-streams-per-rwops code, after
	- discussion on the mailing list. The init() and quit() methods
	- led to the possibility that certain decoders will flag themselves
	- as unavailable at runtime, and SDL_sound now handles this.
	- Added [LIB\|INC]PATH_[OGG\|MOD]. Bigendian fixes; now works on
	- PowerPC Linux. MikMod tweaks. Changed version to 0.1.2.
	-09202001 - Torbjörn Andersson submitted several patches: fixed a comment in
	- the .WAV decoder (whoops...screwed up my own search-and-replace.
	- Hah.), made an attempt at putting multiple sound streams behind
	- one RWops (gotta think on that one first), and, most importantly,
	- added an AIFF decoder, which is very cool.
	-09192001 - Added a skeleton decoder source file. Changed voc_read() to
	- voc_read_waveform(), so it wouldn't be confused with VOC_read().
	- Fixed a byte ordering bug in voc.c (reported as AUDIO_S16LSB, but
	- we were swapping byte order of data ourselves. Fixed). Added basic
	- .WAV support. Fixed Makefile so that -I. is always first;
	- otherwise, a previously installed header might get used for the
	- compiles, which is not good. SDL_sound.h now includes SDL_endian.h,
	- since SDL.h doesn't, for some reason. Moved version defines in
	- SDL_sound.h to top of file so I can find them. :)
	- Changed version to 0.1.1. Committed patch from Tsuyoshi Iguchi to
	- fix a segfault (I forgot to put a NULL terminator at the end of
	- the available_decoders array), fixing the only bug preventing the
	- test program from running on FreeBSD 4.3. Sweet. Added Ogg Vorbis
	- decoder. Rewrote the test program's SDL audio callback to be more
	- robust (Ogg exposed a nasty bug in it). Fixed a byte-ordering issue
	- in the VOC decoder.
	-09182001 - Implemented MP3 support through SMPEG (not working yet, though) and
	- wrote the Reference Counting RWops wrapper. Added other little
	- things like the _D(()) macro. Added VOC support, which went up with
	- surprisingly little struggle, which means it MUST be leaking
	- memory. :)
	-09172001 - Changed some overlooked "voice" to "sound". Implemented base API.
	- So...tired. Everything's different. :)
	- Also put in a RAW decoder and a simple test program.
	-09142001 - Changed name to SDL_sound, added Sound_DecodeAll() to spec.
	-09132001 - Initial spec proposed on SDL mailing list, under name "SDL_voice".
	-
	---ryan. (icculus@icculus.org)
	-
	-/* end of CHANGELOG ... */
	-
	diff --git a/util/sdl/sound/COPYING b/util/sdl/sound/COPYING
	deleted file mode 100644
	index 6228aac2..00000000
	--- a/util/sdl/sound/COPYING
	+++ /dev/null
	@@ -1,524 +0,0 @@
	-Please note that the included source from Timidity, the MIDI decoder, is also
	- licensed under the following terms (GNU LGPL), but can also be used
	- separately under the GNU GPL, or the Perl Artistic License. Those licensing
	- terms are not reprinted here, but can be found on the web easily.
	-
	-Other external libraries (such as Ogg Vorbis, SMPEG, etc) have their own
	- licenses which you should be aware of before including the related code
	- in your configuration. Most (if not all) are also under the LGPL, but are
	- external projects and we've got no control over them.
	-
	-If you want to use SDL_sound under a closed-source license, please contact
	- Ryan (icculus@icculus.org), and we can discuss an alternate license for
	- money to be distributed between the contributors to this work, but I'd
	- encourage you to abide by the LGPL, since the usual concern is whether you
	- can use this library without releasing your own source code (you can).
	-
	-
	--------------------
	-
	-
	- GNU LESSER GENERAL PUBLIC LICENSE
	- Version 2.1, February 1999
	-
	- Copyright (C) 1991, 1999 Free Software Foundation, Inc.
	- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
	- Everyone is permitted to copy and distribute verbatim copies
	- of this license document, but changing it is not allowed.
	-
	-[This is the first released version of the Lesser GPL. It also counts
	- as the successor of the GNU Library Public License, version 2, hence
	- the version number 2.1.]
	-
	- Preamble
	-
	- The licenses for most software are designed to take away your
	-freedom to share and change it. By contrast, the GNU General Public
	-Licenses are intended to guarantee your freedom to share and change
	-free software--to make sure the software is free for all its users.
	-
	- This license, the Lesser General Public License, applies to some
	-specially designated software packages--typically libraries--of the
	-Free Software Foundation and other authors who decide to use it. You
	-can use it too, but we suggest you first think carefully about whether
	-this license or the ordinary General Public License is the better
	-strategy to use in any particular case, based on the explanations below.
	-
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	diff --git a/util/sdl/sound/CREDITS b/util/sdl/sound/CREDITS
	deleted file mode 100644
	index bb9af496..00000000
	--- a/util/sdl/sound/CREDITS
	+++ /dev/null
	@@ -1,66 +0,0 @@
	- ----------------------
	- \| SDL_sound credits. \|
	- ----------------------
	-
	-Initial API interface and implementation,
	-RAW driver,
	-VOC driver,
	-SMPEG driver,
	-MPGLIB driver,
	-WAV driver,
	-OGG driver,
	-SHN driver,
	-Unix support,
	-BeOS support:
	- Ryan C. Gordon
	-
	-Bug fixes,
	-FreeBSD testing:
	- Tsuyoshi Iguchi
	-
	-Code cleanups,
	-SMPEG fixes,
	-AIFF driver,
	-MikMod driver,
	-MIDI driver,
	-ModPlug driver,
	-FLAC driver:
	- Torbjörn Andersson
	-
	-autoconf,
	-MacOS X support:
	- Max Horn
	-
	-win32 support,
	-PocketPC support,
	-other fixes:
	- Tyler Montbriand
	-
	-AU driver,
	- Mattias Engdegård
	-
	-MacOS Classic support,
	-quicktime decoder,
	-OS X fixes:
	- Darrell Walisser
	-
	-Alternate audio conversion code:
	- Frank Ranostaj
	-
	-Initial Borland C++ project files:
	- Dominique Louis
	-
	-Bugfixes and stuff:
	- Eric Wing
	-
	-FLAC 1.1.3 updates:
	- Josh Coalson
	-
	-SMPEG fixes:
	- Chris Nelson
	-
	-Other stuff:
	- Your name here! Patches go to icculus@icculus.org ...
	-
	-/* end of CREDITS ... */
	-
	diff --git a/util/sdl/sound/Doxyfile b/util/sdl/sound/Doxyfile
	deleted file mode 100644
	index 33cf6708..00000000
	--- a/util/sdl/sound/Doxyfile
	+++ /dev/null
	@@ -1,946 +0,0 @@
	-# Doxyfile 1.2.16
	-
	-# This file describes the settings to be used by the documentation system
	-# doxygen (www.doxygen.org) for a project
	-#
	-# All text after a hash (#) is considered a comment and will be ignored
	-# The format is:
	-# TAG = value [value, ...]
	-# For lists items can also be appended using:
	-# TAG += value [value, ...]
	-# Values that contain spaces should be placed between quotes (" ")
	-
	-#---------------------------------------------------------------------------
	-# General configuration options
	-#---------------------------------------------------------------------------
	-
	-# The PROJECT_NAME tag is a single word (or a sequence of words surrounded
	-# by quotes) that should identify the project.
	-
	-PROJECT_NAME = SDL_sound
	-
	-# The PROJECT_NUMBER tag can be used to enter a project or revision number.
	-# This could be handy for archiving the generated documentation or
	-# if some version control system is used.
	-
	-PROJECT_NUMBER = 1.0.1
	-
	-# The OUTPUT_DIRECTORY tag is used to specify the (relative or absolute)
	-# base path where the generated documentation will be put.
	-# If a relative path is entered, it will be relative to the location
	-# where doxygen was started. If left blank the current directory will be used.
	-
	-OUTPUT_DIRECTORY = docs
	-
	-# The OUTPUT_LANGUAGE tag is used to specify the language in which all
	-# documentation generated by doxygen is written. Doxygen will use this
	-# information to generate all constant output in the proper language.
	-# The default language is English, other supported languages are:
	-# Brazilian, Chinese, Chinese-Traditional, Croatian, Czech, Danish, Dutch,
	-# Finnish, French, German, Greek, Hungarian, Italian, Japanese, Korean,
	-# Norwegian, Polish, Portuguese, Romanian, Russian, Slovak, Slovene,
	-# Spanish, Swedish and Ukrainian.
	-
	-OUTPUT_LANGUAGE = English
	-
	-# If the EXTRACT_ALL tag is set to YES doxygen will assume all entities in
	-# documentation are documented, even if no documentation was available.
	-# Private class members and static file members will be hidden unless
	-# the EXTRACT_PRIVATE and EXTRACT_STATIC tags are set to YES
	-
	-EXTRACT_ALL = NO
	-
	-# If the EXTRACT_PRIVATE tag is set to YES all private members of a class
	-# will be included in the documentation.
	-
	-EXTRACT_PRIVATE = NO
	-
	-# If the EXTRACT_STATIC tag is set to YES all static members of a file
	-# will be included in the documentation.
	-
	-EXTRACT_STATIC = NO
	-
	-# If the EXTRACT_LOCAL_CLASSES tag is set to YES classes (and structs)
	-# defined locally in source files will be included in the documentation.
	-# If set to NO only classes defined in header files are included.
	-
	-EXTRACT_LOCAL_CLASSES = NO
	-
	-# If the HIDE_UNDOC_MEMBERS tag is set to YES, Doxygen will hide all
	-# undocumented members of documented classes, files or namespaces.
	-# If set to NO (the default) these members will be included in the
	-# various overviews, but no documentation section is generated.
	-# This option has no effect if EXTRACT_ALL is enabled.
	-
	-HIDE_UNDOC_MEMBERS = NO
	-
	-# If the HIDE_UNDOC_CLASSES tag is set to YES, Doxygen will hide all
	-# undocumented classes that are normally visible in the class hierarchy.
	-# If set to NO (the default) these class will be included in the various
	-# overviews. This option has no effect if EXTRACT_ALL is enabled.
	-
	-HIDE_UNDOC_CLASSES = NO
	-
	-# If the BRIEF_MEMBER_DESC tag is set to YES (the default) Doxygen will
	-# include brief member descriptions after the members that are listed in
	-# the file and class documentation (similar to JavaDoc).
	-# Set to NO to disable this.
	-
	-BRIEF_MEMBER_DESC = YES
	-
	-# If the REPEAT_BRIEF tag is set to YES (the default) Doxygen will prepend
	-# the brief description of a member or function before the detailed description.
	-# Note: if both HIDE_UNDOC_MEMBERS and BRIEF_MEMBER_DESC are set to NO, the
	-# brief descriptions will be completely suppressed.
	-
	-REPEAT_BRIEF = YES
	-
	-# If the ALWAYS_DETAILED_SEC and REPEAT_BRIEF tags are both set to YES then
	-# Doxygen will generate a detailed section even if there is only a brief
	-# description.
	-
	-ALWAYS_DETAILED_SEC = NO
	-
	-# If the INLINE_INHERITED_MEMB tag is set to YES, doxygen will show all inherited
	-# members of a class in the documentation of that class as if those members were
	-# ordinary class members. Constructors, destructors and assignment operators of
	-# the base classes will not be shown.
	-
	-INLINE_INHERITED_MEMB = NO
	-
	-# If the FULL_PATH_NAMES tag is set to YES then Doxygen will prepend the full
	-# path before files name in the file list and in the header files. If set
	-# to NO the shortest path that makes the file name unique will be used.
	-
	-FULL_PATH_NAMES = NO
	-
	-# If the FULL_PATH_NAMES tag is set to YES then the STRIP_FROM_PATH tag
	-# can be used to strip a user defined part of the path. Stripping is
	-# only done if one of the specified strings matches the left-hand part of
	-# the path. It is allowed to use relative paths in the argument list.
	-
	-STRIP_FROM_PATH =
	-
	-# The INTERNAL_DOCS tag determines if documentation
	-# that is typed after a \internal command is included. If the tag is set
	-# to NO (the default) then the documentation will be excluded.
	-# Set it to YES to include the internal documentation.
	-
	-INTERNAL_DOCS = NO
	-
	-# Setting the STRIP_CODE_COMMENTS tag to YES (the default) will instruct
	-# doxygen to hide any special comment blocks from generated source code
	-# fragments. Normal C and C++ comments will always remain visible.
	-
	-STRIP_CODE_COMMENTS = YES
	-
	-# If the CASE_SENSE_NAMES tag is set to NO then Doxygen will only generate
	-# file names in lower case letters. If set to YES upper case letters are also
	-# allowed. This is useful if you have classes or files whose names only differ
	-# in case and if your file system supports case sensitive file names. Windows
	-# users are adviced to set this option to NO.
	-
	-CASE_SENSE_NAMES = YES
	-
	-# If the SHORT_NAMES tag is set to YES, doxygen will generate much shorter
	-# (but less readable) file names. This can be useful is your file systems
	-# doesn't support long names like on DOS, Mac, or CD-ROM.
	-
	-SHORT_NAMES = NO
	-
	-# If the HIDE_SCOPE_NAMES tag is set to NO (the default) then Doxygen
	-# will show members with their full class and namespace scopes in the
	-# documentation. If set to YES the scope will be hidden.
	-
	-HIDE_SCOPE_NAMES = NO
	-
	-# If the VERBATIM_HEADERS tag is set to YES (the default) then Doxygen
	-# will generate a verbatim copy of the header file for each class for
	-# which an include is specified. Set to NO to disable this.
	-
	-VERBATIM_HEADERS = YES
	-
	-# If the SHOW_INCLUDE_FILES tag is set to YES (the default) then Doxygen
	-# will put list of the files that are included by a file in the documentation
	-# of that file.
	-
	-SHOW_INCLUDE_FILES = YES
	-
	-# If the JAVADOC_AUTOBRIEF tag is set to YES then Doxygen
	-# will interpret the first line (until the first dot) of a JavaDoc-style
	-# comment as the brief description. If set to NO, the JavaDoc
	-# comments will behave just like the Qt-style comments (thus requiring an
	-# explict @brief command for a brief description.
	-
	-JAVADOC_AUTOBRIEF = NO
	-
	-# If the DETAILS_AT_TOP tag is set to YES then Doxygen
	-# will output the detailed description near the top, like JavaDoc.
	-# If set to NO, the detailed description appears after the member
	-# documentation.
	-
	-DETAILS_AT_TOP = NO
	-
	-# If the INHERIT_DOCS tag is set to YES (the default) then an undocumented
	-# member inherits the documentation from any documented member that it
	-# reimplements.
	-
	-INHERIT_DOCS = YES
	-
	-# If the INLINE_INFO tag is set to YES (the default) then a tag [inline]
	-# is inserted in the documentation for inline members.
	-
	-INLINE_INFO = YES
	-
	-# If the SORT_MEMBER_DOCS tag is set to YES (the default) then doxygen
	-# will sort the (detailed) documentation of file and class members
	-# alphabetically by member name. If set to NO the members will appear in
	-# declaration order.
	-
	-SORT_MEMBER_DOCS = YES
	-
	-# If member grouping is used in the documentation and the DISTRIBUTE_GROUP_DOC
	-# tag is set to YES, then doxygen will reuse the documentation of the first
	-# member in the group (if any) for the other members of the group. By default
	-# all members of a group must be documented explicitly.
	-
	-DISTRIBUTE_GROUP_DOC = NO
	-
	-# The TAB_SIZE tag can be used to set the number of spaces in a tab.
	-# Doxygen uses this value to replace tabs by spaces in code fragments.
	-
	-TAB_SIZE = 4
	-
	-# The GENERATE_TODOLIST tag can be used to enable (YES) or
	-# disable (NO) the todo list. This list is created by putting \todo
	-# commands in the documentation.
	-
	-GENERATE_TODOLIST = YES
	-
	-# The GENERATE_TESTLIST tag can be used to enable (YES) or
	-# disable (NO) the test list. This list is created by putting \test
	-# commands in the documentation.
	-
	-GENERATE_TESTLIST = YES
	-
	-# The GENERATE_BUGLIST tag can be used to enable (YES) or
	-# disable (NO) the bug list. This list is created by putting \bug
	-# commands in the documentation.
	-
	-GENERATE_BUGLIST = YES
	-
	-# This tag can be used to specify a number of aliases that acts
	-# as commands in the documentation. An alias has the form "name=value".
	-# For example adding "sideeffect=\par Side Effects:\n" will allow you to
	-# put the command \sideeffect (or @sideeffect) in the documentation, which
	-# will result in a user defined paragraph with heading "Side Effects:".
	-# You can put \n's in the value part of an alias to insert newlines.
	-
	-ALIASES =
	-
	-# The ENABLED_SECTIONS tag can be used to enable conditional
	-# documentation sections, marked by \if sectionname ... \endif.
	-
	-ENABLED_SECTIONS =
	-
	-# The MAX_INITIALIZER_LINES tag determines the maximum number of lines
	-# the initial value of a variable or define consist of for it to appear in
	-# the documentation. If the initializer consists of more lines than specified
	-# here it will be hidden. Use a value of 0 to hide initializers completely.
	-# The appearance of the initializer of individual variables and defines in the
	-# documentation can be controlled using \showinitializer or \hideinitializer
	-# command in the documentation regardless of this setting.
	-
	-MAX_INITIALIZER_LINES = 30
	-
	-# Set the OPTIMIZE_OUTPUT_FOR_C tag to YES if your project consists of C sources
	-# only. Doxygen will then generate output that is more tailored for C.
	-# For instance some of the names that are used will be different. The list
	-# of all members will be omitted, etc.
	-
	-OPTIMIZE_OUTPUT_FOR_C = YES
	-
	-# Set the OPTIMIZE_OUTPUT_JAVA tag to YES if your project consists of Java sources
	-# only. Doxygen will then generate output that is more tailored for Java.
	-# For instance namespaces will be presented as packages, qualified scopes
	-# will look different, etc.
	-
	-OPTIMIZE_OUTPUT_JAVA = NO
	-
	-# Set the SHOW_USED_FILES tag to NO to disable the list of files generated
	-# at the bottom of the documentation of classes and structs. If set to YES the
	-# list will mention the files that were used to generate the documentation.
	-
	-SHOW_USED_FILES = YES
	-
	-#---------------------------------------------------------------------------
	-# configuration options related to warning and progress messages
	-#---------------------------------------------------------------------------
	-
	-# The QUIET tag can be used to turn on/off the messages that are generated
	-# by doxygen. Possible values are YES and NO. If left blank NO is used.
	-
	-QUIET = NO
	-
	-# The WARNINGS tag can be used to turn on/off the warning messages that are
	-# generated by doxygen. Possible values are YES and NO. If left blank
	-# NO is used.
	-
	-WARNINGS = YES
	-
	-# If WARN_IF_UNDOCUMENTED is set to YES, then doxygen will generate warnings
	-# for undocumented members. If EXTRACT_ALL is set to YES then this flag will
	-# automatically be disabled.
	-
	-WARN_IF_UNDOCUMENTED = YES
	-
	-# The WARN_FORMAT tag determines the format of the warning messages that
	-# doxygen can produce. The string should contain the $file, $line, and $text
	-# tags, which will be replaced by the file and line number from which the
	-# warning originated and the warning text.
	-
	-WARN_FORMAT = "$file:$line: $text"
	-
	-# The WARN_LOGFILE tag can be used to specify a file to which warning
	-# and error messages should be written. If left blank the output is written
	-# to stderr.
	-
	-WARN_LOGFILE =
	-
	-#---------------------------------------------------------------------------
	-# configuration options related to the input files
	-#---------------------------------------------------------------------------
	-
	-# The INPUT tag can be used to specify the files and/or directories that contain
	-# documented source files. You may enter file names like "myfile.cpp" or
	-# directories like "/usr/src/myproject". Separate the files or directories
	-# with spaces.
	-
	-INPUT = SDL_sound.h
	-
	-# If the value of the INPUT tag contains directories, you can use the
	-# FILE_PATTERNS tag to specify one or more wildcard pattern (like *.cpp
	-# and *.h) to filter out the source-files in the directories. If left
	-# blank the following patterns are tested:
	-# .c .cc .cxx .cpp .c++ .java .ii .ixx .ipp .i++ .inl .h .hh .hxx *.hpp
	-# .h++ .idl *.odl
	-
	-FILE_PATTERNS =
	-
	-# The RECURSIVE tag can be used to turn specify whether or not subdirectories
	-# should be searched for input files as well. Possible values are YES and NO.
	-# If left blank NO is used.
	-
	-RECURSIVE = NO
	-
	-# The EXCLUDE tag can be used to specify files and/or directories that should
	-# excluded from the INPUT source files. This way you can easily exclude a
	-# subdirectory from a directory tree whose root is specified with the INPUT tag.
	-
	-EXCLUDE =
	-
	-# The EXCLUDE_SYMLINKS tag can be used select whether or not files or directories
	-# that are symbolic links (a Unix filesystem feature) are excluded from the input.
	-
	-EXCLUDE_SYMLINKS = NO
	-
	-# If the value of the INPUT tag contains directories, you can use the
	-# EXCLUDE_PATTERNS tag to specify one or more wildcard patterns to exclude
	-# certain files from those directories.
	-
	-EXCLUDE_PATTERNS =
	-
	-# The EXAMPLE_PATH tag can be used to specify one or more files or
	-# directories that contain example code fragments that are included (see
	-# the \include command).
	-
	-EXAMPLE_PATH =
	-
	-# If the value of the EXAMPLE_PATH tag contains directories, you can use the
	-# EXAMPLE_PATTERNS tag to specify one or more wildcard pattern (like *.cpp
	-# and *.h) to filter out the source-files in the directories. If left
	-# blank all files are included.
	-
	-EXAMPLE_PATTERNS =
	-
	-# If the EXAMPLE_RECURSIVE tag is set to YES then subdirectories will be
	-# searched for input files to be used with the \include or \dontinclude
	-# commands irrespective of the value of the RECURSIVE tag.
	-# Possible values are YES and NO. If left blank NO is used.
	-
	-EXAMPLE_RECURSIVE = NO
	-
	-# The IMAGE_PATH tag can be used to specify one or more files or
	-# directories that contain image that are included in the documentation (see
	-# the \image command).
	-
	-IMAGE_PATH =
	-
	-# The INPUT_FILTER tag can be used to specify a program that doxygen should
	-# invoke to filter for each input file. Doxygen will invoke the filter program
	-# by executing (via popen()) the command <filter> <input-file>, where <filter>
	-# is the value of the INPUT_FILTER tag, and <input-file> is the name of an
	-# input file. Doxygen will then use the output that the filter program writes
	-# to standard output.
	-
	-INPUT_FILTER =
	-
	-# If the FILTER_SOURCE_FILES tag is set to YES, the input filter (if set using
	-# INPUT_FILTER) will be used to filter the input files when producing source
	-# files to browse.
	-
	-FILTER_SOURCE_FILES = NO
	-
	-#---------------------------------------------------------------------------
	-# configuration options related to source browsing
	-#---------------------------------------------------------------------------
	-
	-# If the SOURCE_BROWSER tag is set to YES then a list of source files will
	-# be generated. Documented entities will be cross-referenced with these sources.
	-
	-SOURCE_BROWSER = NO
	-
	-# Setting the INLINE_SOURCES tag to YES will include the body
	-# of functions and classes directly in the documentation.
	-
	-INLINE_SOURCES = NO
	-
	-# If the REFERENCED_BY_RELATION tag is set to YES (the default)
	-# then for each documented function all documented
	-# functions referencing it will be listed.
	-
	-REFERENCED_BY_RELATION = YES
	-
	-# If the REFERENCES_RELATION tag is set to YES (the default)
	-# then for each documented function all documented entities
	-# called/used by that function will be listed.
	-
	-REFERENCES_RELATION = YES
	-
	-#---------------------------------------------------------------------------
	-# configuration options related to the alphabetical class index
	-#---------------------------------------------------------------------------
	-
	-# If the ALPHABETICAL_INDEX tag is set to YES, an alphabetical index
	-# of all compounds will be generated. Enable this if the project
	-# contains a lot of classes, structs, unions or interfaces.
	-
	-ALPHABETICAL_INDEX = NO
	-
	-# If the alphabetical index is enabled (see ALPHABETICAL_INDEX) then
	-# the COLS_IN_ALPHA_INDEX tag can be used to specify the number of columns
	-# in which this list will be split (can be a number in the range [1..20])
	-
	-COLS_IN_ALPHA_INDEX = 5
	-
	-# In case all classes in a project start with a common prefix, all
	-# classes will be put under the same header in the alphabetical index.
	-# The IGNORE_PREFIX tag can be used to specify one or more prefixes that
	-# should be ignored while generating the index headers.
	-
	-IGNORE_PREFIX =
	-
	-#---------------------------------------------------------------------------
	-# configuration options related to the HTML output
	-#---------------------------------------------------------------------------
	-
	-# If the GENERATE_HTML tag is set to YES (the default) Doxygen will
	-# generate HTML output.
	-
	-GENERATE_HTML = YES
	-
	-# The HTML_OUTPUT tag is used to specify where the HTML docs will be put.
	-# If a relative path is entered the value of OUTPUT_DIRECTORY will be
	-# put in front of it. If left blank `html' will be used as the default path.
	-
	-HTML_OUTPUT = html
	-
	-# The HTML_FILE_EXTENSION tag can be used to specify the file extension for
	-# each generated HTML page (for example: .htm,.php,.asp). If it is left blank
	-# doxygen will generate files with .html extension.
	-
	-HTML_FILE_EXTENSION = .html
	-
	-# The HTML_HEADER tag can be used to specify a personal HTML header for
	-# each generated HTML page. If it is left blank doxygen will generate a
	-# standard header.
	-
	-HTML_HEADER =
	-
	-# The HTML_FOOTER tag can be used to specify a personal HTML footer for
	-# each generated HTML page. If it is left blank doxygen will generate a
	-# standard footer.
	-
	-HTML_FOOTER =
	-
	-# The HTML_STYLESHEET tag can be used to specify a user defined cascading
	-# style sheet that is used by each HTML page. It can be used to
	-# fine-tune the look of the HTML output. If the tag is left blank doxygen
	-# will generate a default style sheet
	-
	-HTML_STYLESHEET =
	-
	-# If the HTML_ALIGN_MEMBERS tag is set to YES, the members of classes,
	-# files or namespaces will be aligned in HTML using tables. If set to
	-# NO a bullet list will be used.
	-
	-HTML_ALIGN_MEMBERS = YES
	-
	-# If the GENERATE_HTMLHELP tag is set to YES, additional index files
	-# will be generated that can be used as input for tools like the
	-# Microsoft HTML help workshop to generate a compressed HTML help file (.chm)
	-# of the generated HTML documentation.
	-
	-GENERATE_HTMLHELP = NO
	-
	-# If the GENERATE_HTMLHELP tag is set to YES, the GENERATE_CHI flag
	-# controls if a separate .chi index file is generated (YES) or that
	-# it should be included in the master .chm file (NO).
	-
	-GENERATE_CHI = NO
	-
	-# If the GENERATE_HTMLHELP tag is set to YES, the BINARY_TOC flag
	-# controls whether a binary table of contents is generated (YES) or a
	-# normal table of contents (NO) in the .chm file.
	-
	-BINARY_TOC = NO
	-
	-# The TOC_EXPAND flag can be set to YES to add extra items for group members
	-# to the contents of the Html help documentation and to the tree view.
	-
	-TOC_EXPAND = NO
	-
	-# The DISABLE_INDEX tag can be used to turn on/off the condensed index at
	-# top of each HTML page. The value NO (the default) enables the index and
	-# the value YES disables it.
	-
	-DISABLE_INDEX = NO
	-
	-# This tag can be used to set the number of enum values (range [1..20])
	-# that doxygen will group on one line in the generated HTML documentation.
	-
	-ENUM_VALUES_PER_LINE = 4
	-
	-# If the GENERATE_TREEVIEW tag is set to YES, a side panel will be
	-# generated containing a tree-like index structure (just like the one that
	-# is generated for HTML Help). For this to work a browser that supports
	-# JavaScript and frames is required (for instance Mozilla, Netscape 4.0+,
	-# or Internet explorer 4.0+). Note that for large projects the tree generation
	-# can take a very long time. In such cases it is better to disable this feature.
	-# Windows users are probably better off using the HTML help feature.
	-
	-GENERATE_TREEVIEW = NO
	-
	-# If the treeview is enabled (see GENERATE_TREEVIEW) then this tag can be
	-# used to set the initial width (in pixels) of the frame in which the tree
	-# is shown.
	-
	-TREEVIEW_WIDTH = 250
	-
	-#---------------------------------------------------------------------------
	-# configuration options related to the LaTeX output
	-#---------------------------------------------------------------------------
	-
	-# If the GENERATE_LATEX tag is set to YES (the default) Doxygen will
	-# generate Latex output.
	-
	-GENERATE_LATEX = YES
	-
	-# The LATEX_OUTPUT tag is used to specify where the LaTeX docs will be put.
	-# If a relative path is entered the value of OUTPUT_DIRECTORY will be
	-# put in front of it. If left blank `latex' will be used as the default path.
	-
	-LATEX_OUTPUT = latex
	-
	-# The LATEX_CMD_NAME tag can be used to specify the LaTeX command name to be invoked. If left blank `latex' will be used as the default command name.
	-
	-LATEX_CMD_NAME = latex
	-
	-# The MAKEINDEX_CMD_NAME tag can be used to specify the command name to
	-# generate index for LaTeX. If left blank `makeindex' will be used as the
	-# default command name.
	-
	-MAKEINDEX_CMD_NAME = makeindex
	-
	-# If the COMPACT_LATEX tag is set to YES Doxygen generates more compact
	-# LaTeX documents. This may be useful for small projects and may help to
	-# save some trees in general.
	-
	-COMPACT_LATEX = NO
	-
	-# The PAPER_TYPE tag can be used to set the paper type that is used
	-# by the printer. Possible values are: a4, a4wide, letter, legal and
	-# executive. If left blank a4wide will be used.
	-
	-PAPER_TYPE = a4wide
	-
	-# The EXTRA_PACKAGES tag can be to specify one or more names of LaTeX
	-# packages that should be included in the LaTeX output.
	-
	-EXTRA_PACKAGES =
	-
	-# The LATEX_HEADER tag can be used to specify a personal LaTeX header for
	-# the generated latex document. The header should contain everything until
	-# the first chapter. If it is left blank doxygen will generate a
	-# standard header. Notice: only use this tag if you know what you are doing!
	-
	-LATEX_HEADER =
	-
	-# If the PDF_HYPERLINKS tag is set to YES, the LaTeX that is generated
	-# is prepared for conversion to pdf (using ps2pdf). The pdf file will
	-# contain links (just like the HTML output) instead of page references
	-# This makes the output suitable for online browsing using a pdf viewer.
	-
	-PDF_HYPERLINKS = NO
	-
	-# If the USE_PDFLATEX tag is set to YES, pdflatex will be used instead of
	-# plain latex in the generated Makefile. Set this option to YES to get a
	-# higher quality PDF documentation.
	-
	-USE_PDFLATEX = NO
	-
	-# If the LATEX_BATCHMODE tag is set to YES, doxygen will add the \\batchmode.
	-# command to the generated LaTeX files. This will instruct LaTeX to keep
	-# running if errors occur, instead of asking the user for help.
	-# This option is also used when generating formulas in HTML.
	-
	-LATEX_BATCHMODE = NO
	-
	-#---------------------------------------------------------------------------
	-# configuration options related to the RTF output
	-#---------------------------------------------------------------------------
	-
	-# If the GENERATE_RTF tag is set to YES Doxygen will generate RTF output
	-# The RTF output is optimised for Word 97 and may not look very pretty with
	-# other RTF readers or editors.
	-
	-GENERATE_RTF = NO
	-
	-# The RTF_OUTPUT tag is used to specify where the RTF docs will be put.
	-# If a relative path is entered the value of OUTPUT_DIRECTORY will be
	-# put in front of it. If left blank `rtf' will be used as the default path.
	-
	-RTF_OUTPUT = rtf
	-
	-# If the COMPACT_RTF tag is set to YES Doxygen generates more compact
	-# RTF documents. This may be useful for small projects and may help to
	-# save some trees in general.
	-
	-COMPACT_RTF = NO
	-
	-# If the RTF_HYPERLINKS tag is set to YES, the RTF that is generated
	-# will contain hyperlink fields. The RTF file will
	-# contain links (just like the HTML output) instead of page references.
	-# This makes the output suitable for online browsing using WORD or other
	-# programs which support those fields.
	-# Note: wordpad (write) and others do not support links.
	-
	-RTF_HYPERLINKS = NO
	-
	-# Load stylesheet definitions from file. Syntax is similar to doxygen's
	-# config file, i.e. a series of assigments. You only have to provide
	-# replacements, missing definitions are set to their default value.
	-
	-RTF_STYLESHEET_FILE =
	-
	-# Set optional variables used in the generation of an rtf document.
	-# Syntax is similar to doxygen's config file.
	-
	-RTF_EXTENSIONS_FILE =
	-
	-#---------------------------------------------------------------------------
	-# configuration options related to the man page output
	-#---------------------------------------------------------------------------
	-
	-# If the GENERATE_MAN tag is set to YES (the default) Doxygen will
	-# generate man pages
	-
	-GENERATE_MAN = YES
	-
	-# The MAN_OUTPUT tag is used to specify where the man pages will be put.
	-# If a relative path is entered the value of OUTPUT_DIRECTORY will be
	-# put in front of it. If left blank `man' will be used as the default path.
	-
	-MAN_OUTPUT = man
	-
	-# The MAN_EXTENSION tag determines the extension that is added to
	-# the generated man pages (default is the subroutine's section .3)
	-
	-MAN_EXTENSION = .3
	-
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	-# documented in the real man page(s). These additional files
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	-
	-# If the GENERATE_XML tag is set to YES Doxygen will
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	-GENERATE_XML = NO
	-
	-#---------------------------------------------------------------------------
	-# configuration options for the AutoGen Definitions output
	-#---------------------------------------------------------------------------
	-
	-# If the GENERATE_AUTOGEN_DEF tag is set to YES Doxygen will
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	-ENABLE_PREPROCESSING = YES
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	-MACRO_EXPANSION = YES
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	-# If the EXPAND_ONLY_PREDEF and MACRO_EXPANSION tags are both set to YES
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	-EXPAND_ONLY_PREDEF = YES
	-
	-# If the SEARCH_INCLUDES tag is set to YES (the default) the includes files
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	-SEARCH_INCLUDES = YES
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	-# The INCLUDE_PATH tag can be used to specify one or more directories that
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	-INCLUDE_PATH =
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	-# You can use the INCLUDE_FILE_PATTERNS tag to specify one or more wildcard
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	-INCLUDE_FILE_PATTERNS =
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	-# gcc). The argument of the tag is a list of macros of the form: name
	-# or name=definition (no spaces). If the definition and the = are
	-# omitted =1 is assumed.
	-
	-PREDEFINED = DOXYGEN_SHOULD_IGNORE_THIS=1 SDLCALL= SNDDECLSPEC=
	-
	-# If the MACRO_EXPANSION and EXPAND_PREDEF_ONLY tags are set to YES then
	-# this tag can be used to specify a list of macro names that should be expanded.
	-# The macro definition that is found in the sources will be used.
	-# Use the PREDEFINED tag if you want to use a different macro definition.
	-
	-EXPAND_AS_DEFINED =
	-
	-# If the SKIP_FUNCTION_MACROS tag is set to YES (the default) then
	-# doxygen's preprocessor will remove all function-like macros that are alone
	-# on a line and do not end with a semicolon. Such function macros are typically
	-# used for boiler-plate code, and will confuse the parser if not removed.
	-
	-SKIP_FUNCTION_MACROS = YES
	-
	-#---------------------------------------------------------------------------
	-# Configuration::addtions related to external references
	-#---------------------------------------------------------------------------
	-
	-# The TAGFILES tag can be used to specify one or more tagfiles.
	-
	-TAGFILES =
	-
	-# When a file name is specified after GENERATE_TAGFILE, doxygen will create
	-# a tag file that is based on the input files it reads.
	-
	-GENERATE_TAGFILE =
	-
	-# If the ALLEXTERNALS tag is set to YES all external classes will be listed
	-# in the class index. If set to NO only the inherited external classes
	-# will be listed.
	-
	-ALLEXTERNALS = NO
	-
	-# If the EXTERNAL_GROUPS tag is set to YES all external groups will be listed
	-# in the modules index. If set to NO, only the current project's groups will
	-# be listed.
	-
	-EXTERNAL_GROUPS = YES
	-
	-# The PERL_PATH should be the absolute path and name of the perl script
	-# interpreter (i.e. the result of `which perl').
	-
	-PERL_PATH = /usr/bin/perl
	-
	-#---------------------------------------------------------------------------
	-# Configuration options related to the dot tool
	-#---------------------------------------------------------------------------
	-
	-# If the CLASS_DIAGRAMS tag is set to YES (the default) Doxygen will
	-# generate a inheritance diagram (in Html, RTF and LaTeX) for classes with base or
	-# super classes. Setting the tag to NO turns the diagrams off. Note that this
	-# option is superceded by the HAVE_DOT option below. This is only a fallback. It is
	-# recommended to install and use dot, since it yields more powerful graphs.
	-
	-CLASS_DIAGRAMS = NO
	-
	-# If set to YES, the inheritance and collaboration graphs will hide
	-# inheritance and usage relations if the target is undocumented
	-# or is not a class.
	-
	-HIDE_UNDOC_RELATIONS = YES
	-
	-# If you set the HAVE_DOT tag to YES then doxygen will assume the dot tool is
	-# available from the path. This tool is part of Graphviz, a graph visualization
	-# toolkit from AT&T and Lucent Bell Labs. The other options in this section
	-# have no effect if this option is set to NO (the default)
	-
	-HAVE_DOT = NO
	-
	-# If the CLASS_GRAPH and HAVE_DOT tags are set to YES then doxygen
	-# will generate a graph for each documented class showing the direct and
	-# indirect inheritance relations. Setting this tag to YES will force the
	-# the CLASS_DIAGRAMS tag to NO.
	-
	-CLASS_GRAPH = NO
	-
	-# If the COLLABORATION_GRAPH and HAVE_DOT tags are set to YES then doxygen
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	-# indirect implementation dependencies (inheritance, containment, and
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	-COLLABORATION_GRAPH = NO
	-
	-# If set to YES, the inheritance and collaboration graphs will show the
	-# relations between templates and their instances.
	-
	-TEMPLATE_RELATIONS = NO
	-
	-# If the ENABLE_PREPROCESSING, SEARCH_INCLUDES, INCLUDE_GRAPH, and HAVE_DOT
	-# tags are set to YES then doxygen will generate a graph for each documented
	-# file showing the direct and indirect include dependencies of the file with
	-# other documented files.
	-
	-INCLUDE_GRAPH = NO
	-
	-# If the ENABLE_PREPROCESSING, SEARCH_INCLUDES, INCLUDED_BY_GRAPH, and
	-# HAVE_DOT tags are set to YES then doxygen will generate a graph for each
	-# documented header file showing the documented files that directly or
	-# indirectly include this file.
	-
	-INCLUDED_BY_GRAPH = YES
	-
	-# If the GRAPHICAL_HIERARCHY and HAVE_DOT tags are set to YES then doxygen
	-# will graphical hierarchy of all classes instead of a textual one.
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	-GRAPHICAL_HIERARCHY = YES
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	-# The DOT_IMAGE_FORMAT tag can be used to set the image format of the images
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	-# If left blank png will be used.
	-
	-DOT_IMAGE_FORMAT = png
	-
	-# The tag DOT_PATH can be used to specify the path where the dot tool can be
	-# found. If left blank, it is assumed the dot tool can be found on the path.
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	-DOT_PATH =
	-
	-# The DOTFILE_DIRS tag can be used to specify one or more directories that
	-# contain dot files that are included in the documentation (see the
	-# \dotfile command).
	-
	-DOTFILE_DIRS =
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	-# The MAX_DOT_GRAPH_WIDTH tag can be used to set the maximum allowed width
	-# (in pixels) of the graphs generated by dot. If a graph becomes larger than
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	-
	-MAX_DOT_GRAPH_WIDTH = 1024
	-
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	-# large images.
	-
	-MAX_DOT_GRAPH_HEIGHT = 1024
	-
	-# If the GENERATE_LEGEND tag is set to YES (the default) Doxygen will
	-# generate a legend page explaining the meaning of the various boxes and
	-# arrows in the dot generated graphs.
	-
	-GENERATE_LEGEND = YES
	-
	-# If the DOT_CLEANUP tag is set to YES (the default) Doxygen will
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	-# the various graphs.
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	-DOT_CLEANUP = YES
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	diff --git a/util/sdl/sound/INSTALL b/util/sdl/sound/INSTALL
	deleted file mode 100644
	index 8ac35599..00000000
	--- a/util/sdl/sound/INSTALL
	+++ /dev/null
	@@ -1,105 +0,0 @@
	-Building is pretty easy. Please read README, too, as it duplicates and
	-expands upon much of this information.
	-
	-
	-ALL PLATFORMS:
	-
	-Please understand your rights and mine: read the text file COPYING in the root
	-of the source tree. If you can't abide by it, delete this source tree now.
	-
	-The best documentation for the SDL_sound API is SDL_sound.h. It is VERY
	-heavily commented, and makes an excellent, in-depth reference to all the
	-functions. The official API reference is generated from this file with
	-a program called "Doxygen" (http://www.doxygen.org/)
	-
	-
	-Borland C++ Builder for Linux (Kylix 3):
	- Unzip the "borland.zip" file in the root of the source tree and use the
	- project files in the newly-created Borland/k3 directory. Makefiles for the
	- command line compiler are in Borland/freebcc ...
	-
	-
	-Unix:
	- (If you pulled the source from CVS), run ./bootstrap
	-
	- run ./configure --help, and see if there's any options you need. Rerun
	- configure with those options. If this is confusing to you, just run
	- ./configure with no options: the defaults are generally decent, and
	- configure is usually smart enough to figure out what's best..
	-
	- If configuration succeeded, run "make".
	-
	- Run "make install" as root to install the library for use on your system.
	-
	- This should work for most Unix-style systems, including Linux, *BSD, BeOS, and
	- MacOS X. Reports of success and failure are welcome.
	-
	-
	-MacOS 9 users:
	- Included with the source is CWProject.sit, which contains project files for
	- CodeWarrior 5.0 and later.
	-
	-
	-MacOS X command line tools:
	- You can use the "UNIX" instructions above if you like the command line tools.
	-
	-
	-MacOS X Project Builder:
	- If you prefer to use Project Builder, use the project files included with
	- this source: PBProjects.tar.gz...unpack it in the root of the SDL_sound
	- folder. This archive contains several external libraries you would have
	- to download/install manually if you used the command line tools (these
	- libraries are for extra decoders, and are NOT required for SDL_sound to
	- function...however, without them, the number of sound formats you can
	- decode is reduced.)
	-
	-
	-BeOS:
	- You can use the "UNIX" instructions above, too.
	-
	-
	-Win32 Visual C:
	- For Visual C, use:
	- http://icculus.org/SDL_sound/downloads/sdl_sound_visualc_srcs.zip
	- ...and unzip it somewhere. This zipfile has a complete copy of the
	- SDL_sound sources, Visual C project files, and several external libraries,
	- too. This zip is everything you should need, and you can scrap this copy of
	- the source.
	-
	-
	-Win32 Cygwin:
	- Cygwin users can try their luck with the Unix build instructions in this
	- tarball instead.
	-
	-
	-Win32 Borland C++ Builder 6:
	- Unzip the "borland.zip" file in the root of the source tree and use the
	- project files in the newly-created Borland/bcb6 directory. Makefiles for the
	- command line compiler are in Borland/freebcc ... these are unmaintained, and
	- you will need to go find the external libraries you want to use (those that
	- wish to maintain these project files should contact me).
	-
	-
	-If building is successful, there will be a shared library and a binary
	- called "playsound".
	-
	-
	-Windows CE (Microsoft PocketPC):
	- You'll need Microsoft's PocketPC development environment, and this zipfile:
	- http://icculus.org/SDL_sound/downloads/SDL_soundCE.zip
	-
	- Unzip that into the root of this source tree. The new "wce" directory has
	- project files, and the source to some of the external decoders is included.
	- Note that not all of the decoders are supported on PocketPC (but please, do
	- send us patches if you get them working!)
	-
	-
	-OTHER PLATFORMS:
	-
	-Send me patches, and instructions, and I'll list them here. Consider
	-joining the SDL_sound mailing list. Details are at:
	- http://icculus.org/SDL_sound/
	-
	---ryan. (icculus@icculus.org)
	-
	-
	diff --git a/util/sdl/sound/Makefile.am b/util/sdl/sound/Makefile.am
	deleted file mode 100644
	index 4927a521..00000000
	--- a/util/sdl/sound/Makefile.am
	+++ /dev/null
	@@ -1,53 +0,0 @@
	-lib_LTLIBRARIES = libSDL_sound.la
	-
	-SUBDIRS = decoders . playsound
	-
	-libSDL_soundincludedir = $(includedir)/SDL
	-libSDL_soundinclude_HEADERS = \
	- SDL_sound.h
	-
	-libSDL_sound_la_SOURCES = \
	- SDL_sound.c \
	- SDL_sound_internal.h \
	- alt_audio_convert.c \
	- alt_audio_convert.h \
	- audio_convert.c \
	- extra_rwops.c \
	- extra_rwops.h
	-
	-if USE_TIMIDITY
	-TIMIDITY_LIB = decoders/timidity/libtimidity.la
	-else
	-TIMIDITY_LIB =
	-endif
	-
	-if USE_MPGLIB
	-MPGLIB_LIB = decoders/mpglib/libmpglib.la
	-else
	-MPGLIB_LIB =
	-endif
	-
	-libSDL_sound_la_LDFLAGS = \
	- -release $(LT_RELEASE) \
	- -version-info $(LT_CURRENT):$(LT_REVISION):$(LT_AGE)
	-libSDL_sound_la_LIBADD = \
	- decoders/libdecoders.la \
	- $(TIMIDITY_LIB) $(MPGLIB_LIB)
	-
	-EXTRA_DIST = \
	- CREDITS \
	- COPYING \
	- CHANGELOG \
	- CWProject.sit \
	- PBProjects.tar.gz \
	- borland.zip \
	- Doxyfile \
	- VisualC
	-
	-dist-hook:
	- mkdir $(distdir)/docs
	- echo "Docs are generated with the program "Doxygen" (http://www.doxygen.org/)," >> $(distdir)/docs/README
	- echo " or can be read online at http://icculus.org/SDL_sound/docs/" >> $(distdir)/docs/README
	- echo >> $(distdir)/docs/README
	- rm -rf `find $(distdir) -type d -name ".svn"`
	-
	diff --git a/util/sdl/sound/Makefile.in b/util/sdl/sound/Makefile.in
	deleted file mode 100644
	index 35cce45a..00000000
	--- a/util/sdl/sound/Makefile.in
	+++ /dev/null
	@@ -1,808 +0,0 @@
	-# Makefile.in generated by automake 1.9.6 from Makefile.am.
	-# @configure_input@
	-
	-# Copyright (C) 1994, 1995, 1996, 1997, 1998, 1999, 2000, 2001, 2002,
	-# 2003, 2004, 2005 Free Software Foundation, Inc.
	-# This Makefile.in is free software; the Free Software Foundation
	-# gives unlimited permission to copy and/or distribute it,
	-# with or without modifications, as long as this notice is preserved.
	-
	-# This program is distributed in the hope that it will be useful,
	-# but WITHOUT ANY WARRANTY, to the extent permitted by law; without
	-# even the implied warranty of MERCHANTABILITY or FITNESS FOR A
	-# PARTICULAR PURPOSE.
	-
	-@SET_MAKE@
	-
	-
	-srcdir = @srcdir@
	-top_srcdir = @top_srcdir@
	-VPATH = @srcdir@
	-pkgdatadir = $(datadir)/@PACKAGE@
	-pkglibdir = $(libdir)/@PACKAGE@
	-pkgincludedir = $(includedir)/@PACKAGE@
	-top_builddir = .
	-am__cd = CDPATH="$${ZSH_VERSION+.}$(PATH_SEPARATOR)" && cd
	-INSTALL = @INSTALL@
	-install_sh_DATA = $(install_sh) -c -m 644
	-install_sh_PROGRAM = $(install_sh) -c
	-install_sh_SCRIPT = $(install_sh) -c
	-INSTALL_HEADER = $(INSTALL_DATA)
	-transform = $(program_transform_name)
	-NORMAL_INSTALL = :
	-PRE_INSTALL = :
	-POST_INSTALL = :
	-NORMAL_UNINSTALL = :
	-PRE_UNINSTALL = :
	-POST_UNINSTALL = :
	-build_triplet = @build@
	-host_triplet = @host@
	-target_triplet = @target@
	-DIST_COMMON = README $(am__configure_deps) \
	- $(libSDL_soundinclude_HEADERS) $(srcdir)/Makefile.am \
	- $(srcdir)/Makefile.in $(srcdir)/config.h.in \
	- $(top_srcdir)/configure COPYING INSTALL TODO compile \
	- config.guess config.sub depcomp install-sh ltmain.sh missing
	-subdir = .
	-ACLOCAL_M4 = $(top_srcdir)/aclocal.m4
	-am__aclocal_m4_deps = $(top_srcdir)/acinclude.m4 \
	- $(top_srcdir)/configure.in
	-am__configure_deps = $(am__aclocal_m4_deps) $(CONFIGURE_DEPENDENCIES) \
	- $(ACLOCAL_M4)
	-am__CONFIG_DISTCLEAN_FILES = config.status config.cache config.log \
	- configure.lineno configure.status.lineno
	-mkinstalldirs = $(install_sh) -d
	-CONFIG_HEADER = config.h
	-CONFIG_CLEAN_FILES =
	-am__vpath_adj_setup = srcdirstrip=`echo "$(srcdir)" \| sed 's\|.\|.\|g'`;
	-am__vpath_adj = case $$p in \
	- $(srcdir)/*) f=`echo "$$p" \| sed "s\|^$$srcdirstrip/\|\|"`;; \
	- *) f=$$p;; \
	- esac;
	-am__strip_dir = `echo $$p \| sed -e 's\|^.*/\|\|'`;
	-am__installdirs = "$(DESTDIR)$(libdir)" \
	- "$(DESTDIR)$(libSDL_soundincludedir)"
	-libLTLIBRARIES_INSTALL = $(INSTALL)
	-LTLIBRARIES = $(lib_LTLIBRARIES)
	-@USE_TIMIDITY_TRUE@am__DEPENDENCIES_1 = \
	-@USE_TIMIDITY_TRUE@ decoders/timidity/libtimidity.la
	-@USE_MPGLIB_TRUE@am__DEPENDENCIES_2 = decoders/mpglib/libmpglib.la
	-libSDL_sound_la_DEPENDENCIES = decoders/libdecoders.la \
	- $(am__DEPENDENCIES_1) $(am__DEPENDENCIES_2)
	-am_libSDL_sound_la_OBJECTS = SDL_sound.lo alt_audio_convert.lo \
	- audio_convert.lo extra_rwops.lo
	-libSDL_sound_la_OBJECTS = $(am_libSDL_sound_la_OBJECTS)
	-DEFAULT_INCLUDES = -I. -I$(srcdir) -I.
	-depcomp = $(SHELL) $(top_srcdir)/depcomp
	-am__depfiles_maybe = depfiles
	-COMPILE = $(CC) $(DEFS) $(DEFAULT_INCLUDES) $(INCLUDES) $(AM_CPPFLAGS) \
	- $(CPPFLAGS) $(AM_CFLAGS) $(CFLAGS)
	-LTCOMPILE = $(LIBTOOL) --tag=CC --mode=compile $(CC) $(DEFS) \
	- $(DEFAULT_INCLUDES) $(INCLUDES) $(AM_CPPFLAGS) $(CPPFLAGS) \
	- $(AM_CFLAGS) $(CFLAGS)
	-CCLD = $(CC)
	-LINK = $(LIBTOOL) --tag=CC --mode=link $(CCLD) $(AM_CFLAGS) $(CFLAGS) \
	- $(AM_LDFLAGS) $(LDFLAGS) -o $@
	-SOURCES = $(libSDL_sound_la_SOURCES)
	-DIST_SOURCES = $(libSDL_sound_la_SOURCES)
	-RECURSIVE_TARGETS = all-recursive check-recursive dvi-recursive \
	- html-recursive info-recursive install-data-recursive \
	- install-exec-recursive install-info-recursive \
	- install-recursive installcheck-recursive installdirs-recursive \
	- pdf-recursive ps-recursive uninstall-info-recursive \
	- uninstall-recursive
	-libSDL_soundincludeHEADERS_INSTALL = $(INSTALL_HEADER)
	-HEADERS = $(libSDL_soundinclude_HEADERS)
	-ETAGS = etags
	-CTAGS = ctags
	-DIST_SUBDIRS = $(SUBDIRS)
	-DISTFILES = $(DIST_COMMON) $(DIST_SOURCES) $(TEXINFOS) $(EXTRA_DIST)
	-distdir = $(PACKAGE)-$(VERSION)
	-top_distdir = $(distdir)
	-am__remove_distdir = \
	- { test ! -d $(distdir) \
	- \|\| { find $(distdir) -type d ! -perm -200 -exec chmod u+w {} ';' \
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	-DIST_ARCHIVES = $(distdir).tar.gz
	-GZIP_ENV = --best
	-distuninstallcheck_listfiles = find . -type f -print
	-distcleancheck_listfiles = find . -type f -print
	-ACLOCAL = @ACLOCAL@
	-AMDEP_FALSE = @AMDEP_FALSE@
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	-AMTAR = @AMTAR@
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	-AUTOHEADER = @AUTOHEADER@
	-AUTOMAKE = @AUTOMAKE@
	-AWK = @AWK@
	-BINARY_AGE = @BINARY_AGE@
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	-CCDEPMODE = @CCDEPMODE@
	-CFLAGS = @CFLAGS@
	-CPP = @CPP@
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	- && $(MAKE) $(AM_MAKEFLAGS) uninstall \
	- && $(MAKE) $(AM_MAKEFLAGS) distuninstallcheck_dir="$$dc_install_base" \
	- distuninstallcheck \
	- && chmod -R a-w "$$dc_install_base" \
	- && ({ \
	- (cd ../.. && umask 077 && mkdir "$$dc_destdir") \
	- && $(MAKE) $(AM_MAKEFLAGS) DESTDIR="$$dc_destdir" install \
	- && $(MAKE) $(AM_MAKEFLAGS) DESTDIR="$$dc_destdir" uninstall \
	- && $(MAKE) $(AM_MAKEFLAGS) DESTDIR="$$dc_destdir" \
	- distuninstallcheck_dir="$$dc_destdir" distuninstallcheck; \
	- } \|\| { rm -rf "$$dc_destdir"; exit 1; }) \
	- && rm -rf "$$dc_destdir" \
	- && $(MAKE) $(AM_MAKEFLAGS) dist \
	- && rm -rf $(DIST_ARCHIVES) \
	- && $(MAKE) $(AM_MAKEFLAGS) distcleancheck
	- $(am__remove_distdir)
	- @(echo "$(distdir) archives ready for distribution: "; \
	- list='$(DIST_ARCHIVES)'; for i in $$list; do echo $$i; done) \| \
	- sed -e '1{h;s/./=/g;p;x;}' -e '$${p;x;}'
	-distuninstallcheck:
	- @cd $(distuninstallcheck_dir) \
	- && test `$(distuninstallcheck_listfiles) \| wc -l` -le 1 \
	- \|\| { echo "ERROR: files left after uninstall:" ; \
	- if test -n "$(DESTDIR)"; then \
	- echo " (check DESTDIR support)"; \
	- fi ; \
	- $(distuninstallcheck_listfiles) ; \
	- exit 1; } >&2
	-distcleancheck: distclean
	- @if test '$(srcdir)' = . ; then \
	- echo "ERROR: distcleancheck can only run from a VPATH build" ; \
	- exit 1 ; \
	- fi
	- @test `$(distcleancheck_listfiles) \| wc -l` -eq 0 \
	- \|\| { echo "ERROR: files left in build directory after distclean:" ; \
	- $(distcleancheck_listfiles) ; \
	- exit 1; } >&2
	-check-am: all-am
	-check: check-recursive
	-all-am: Makefile $(LTLIBRARIES) $(HEADERS) config.h
	-installdirs: installdirs-recursive
	-installdirs-am:
	- for dir in "$(DESTDIR)$(libdir)" "$(DESTDIR)$(libSDL_soundincludedir)"; do \
	- test -z "$$dir" \|\| $(mkdir_p) "$$dir"; \
	- done
	-install: install-recursive
	-install-exec: install-exec-recursive
	-install-data: install-data-recursive
	-uninstall: uninstall-recursive
	-
	-install-am: all-am
	- @$(MAKE) $(AM_MAKEFLAGS) install-exec-am install-data-am
	-
	-installcheck: installcheck-recursive
	-install-strip:
	- $(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
	- install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
	- `test -z '$(STRIP)' \|\| \
	- echo "INSTALL_PROGRAM_ENV=STRIPPROG='$(STRIP)'"` install
	-mostlyclean-generic:
	-
	-clean-generic:
	-
	-distclean-generic:
	- -test -z "$(CONFIG_CLEAN_FILES)" \|\| rm -f $(CONFIG_CLEAN_FILES)
	-
	-maintainer-clean-generic:
	- @echo "This command is intended for maintainers to use"
	- @echo "it deletes files that may require special tools to rebuild."
	-clean: clean-recursive
	-
	-clean-am: clean-generic clean-libLTLIBRARIES clean-libtool \
	- mostlyclean-am
	-
	-distclean: distclean-recursive
	- -rm -f $(am__CONFIG_DISTCLEAN_FILES)
	- -rm -rf ./$(DEPDIR)
	- -rm -f Makefile
	-distclean-am: clean-am distclean-compile distclean-generic \
	- distclean-hdr distclean-libtool distclean-tags
	-
	-dvi: dvi-recursive
	-
	-dvi-am:
	-
	-html: html-recursive
	-
	-info: info-recursive
	-
	-info-am:
	-
	-install-data-am: install-libSDL_soundincludeHEADERS
	-
	-install-exec-am: install-libLTLIBRARIES
	-
	-install-info: install-info-recursive
	-
	-install-man:
	-
	-installcheck-am:
	-
	-maintainer-clean: maintainer-clean-recursive
	- -rm -f $(am__CONFIG_DISTCLEAN_FILES)
	- -rm -rf $(top_srcdir)/autom4te.cache
	- -rm -rf ./$(DEPDIR)
	- -rm -f Makefile
	-maintainer-clean-am: distclean-am maintainer-clean-generic
	-
	-mostlyclean: mostlyclean-recursive
	-
	-mostlyclean-am: mostlyclean-compile mostlyclean-generic \
	- mostlyclean-libtool
	-
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	-
	-ps-am:
	-
	-uninstall-am: uninstall-info-am uninstall-libLTLIBRARIES \
	- uninstall-libSDL_soundincludeHEADERS
	-
	-uninstall-info: uninstall-info-recursive
	-
	-.PHONY: $(RECURSIVE_TARGETS) CTAGS GTAGS all all-am am--refresh check \
	- check-am clean clean-generic clean-libLTLIBRARIES \
	- clean-libtool clean-recursive ctags ctags-recursive dist \
	- dist-all dist-bzip2 dist-gzip dist-hook dist-shar dist-tarZ \
	- dist-zip distcheck distclean distclean-compile \
	- distclean-generic distclean-hdr distclean-libtool \
	- distclean-recursive distclean-tags distcleancheck distdir \
	- distuninstallcheck dvi dvi-am html html-am info info-am \
	- install install-am install-data install-data-am install-exec \
	- install-exec-am install-info install-info-am \
	- install-libLTLIBRARIES install-libSDL_soundincludeHEADERS \
	- install-man install-strip installcheck installcheck-am \
	- installdirs installdirs-am maintainer-clean \
	- maintainer-clean-generic maintainer-clean-recursive \
	- mostlyclean mostlyclean-compile mostlyclean-generic \
	- mostlyclean-libtool mostlyclean-recursive pdf pdf-am ps ps-am \
	- tags tags-recursive uninstall uninstall-am uninstall-info-am \
	- uninstall-libLTLIBRARIES uninstall-libSDL_soundincludeHEADERS
	-
	-
	-dist-hook:
	- mkdir $(distdir)/docs
	- echo "Docs are generated with the program "Doxygen" (http://www.doxygen.org/)," >> $(distdir)/docs/README
	- echo " or can be read online at http://icculus.org/SDL_sound/docs/" >> $(distdir)/docs/README
	- echo >> $(distdir)/docs/README
	- rm -rf `find $(distdir) -type d -name ".svn"`
	-# Tell versions [3.59,3.63) of GNU make to not export all variables.
	-# Otherwise a system limit (for SysV at least) may be exceeded.
	-.NOEXPORT:
	diff --git a/util/sdl/sound/README b/util/sdl/sound/README
	deleted file mode 100644
	index 54bbd5c1..00000000
	--- a/util/sdl/sound/README
	+++ /dev/null
	@@ -1,58 +0,0 @@
	-SDL_sound. An abstract soundfile decoder.
	-
	-SDL_sound is a library that handles the decoding of several popular sound file
	- formats, such as .WAV and .MP3. It is meant to make the programmer's sound
	- playback tasks simpler. The programmer gives SDL_sound a filename, or feeds
	- it data directly from one of many sources, and then reads the decoded
	- waveform data back at her leisure. If resource constraints are a concern,
	- SDL_sound can process sound data in programmer-specified blocks. Alternately,
	- SDL_sound can decode a whole sound file and hand back a single pointer to the
	- whole waveform. SDL_sound can also handle sample rate, audio format, and
	- channel conversion on-the-fly and behind-the-scenes, if the programmer
	- desires.
	-
	-Please check the website for the most up-to-date information about SDL_sound:
	- http://icculus.org/SDL_sound/
	-
	-SDL_sound _REQUIRES_ Simple Directmedia Layer (SDL) to function, and cannot
	- be built without it. You can get SDL from http://www.libsdl.org/. SDL_sound
	- has only been tried with the SDL 1.2 series, but may work on older versions.
	- Reports of success or failure are welcome.
	-
	-Some optional external libraries that SDL_sound can use and where to find them:
	- SMPEG (used to decode MP3s): http://icculus.org/smpeg/
	- libvorbisfile (used to decode OGGs): http://www.xiph.org/ogg/vorbis/
	- libSpeex (used to decode SPXs): http://speex.org/
	- libFLAC (used to decode FLACs): http://flac.sourceforge.net/
	- libModPlug (used to decode MODs, etc): http://modplug-xmms.sourceforge.net/
	- libMikMod (used to decode MODs, etc, too): http://www.mikmod.org/
	-
	- Experimental QuickTime support for the Mac is included, but has not been
	- integrated with the build system, and probably doesn't work with
	- QuickTime for Windows.
	-
	-These external libraries are OPTIONAL. SDL_sound will build and function
	- without them, but various sound file formats are not supported unless these
	- libraries are available. Unless explicitly disabled during initial build
	- configuration, SDL_sound always supports these file formats internally:
	-
	- - Microsoft .WAV files (uncompressed and MS-ADPCM encoded).
	- - Creative Labs .VOC files
	- - Shorten (.SHN) files
	- - Audio Interchange format (AIFF) files
	- - Sun Audio (.AU) files
	- - MIDI files
	- - MP3 files (internal decoder, different than the one SMPEG uses)
	- - Raw waveform data
	-
	-Building/Installing:
	- Please read the INSTALL document.
	-
	-Reporting bugs/commenting:
	- There is a mailing list available. To subscribe, send a blank email to
	- sdlsound-subscribe@icculus.org. This is the best way to get in touch with
	- SDL_sound developers.
	-
	---ryan. (icculus@icculus.org)
	-
	-
	diff --git a/util/sdl/sound/SDL_sound.c b/util/sdl/sound/SDL_sound.c
	deleted file mode 100644
	index 8631c298..00000000
	--- a/util/sdl/sound/SDL_sound.c
	+++ /dev/null
	@@ -1,905 +0,0 @@
	-/*
	- * SDL_sound -- An abstract sound format decoding API.
	- * Copyright (C) 2001 Ryan C. Gordon.
	- *
	- * This library is free software; you can redistribute it and/or
	- * modify it under the terms of the GNU Lesser General Public
	- * License as published by the Free Software Foundation; either
	- * version 2.1 of the License, or (at your option) any later version.
	- *
	- * This library is distributed in the hope that it will be useful,
	- * but WITHOUT ANY WARRANTY; without even the implied warranty of
	- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	- * Lesser General Public License for more details.
	- *
	- * You should have received a copy of the GNU Lesser General Public
	- * License along with this library; if not, write to the Free Software
	- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
	- */
	-
	-/**
	- * This file implements the core API, which is relatively simple.
	- * The real meat of SDL_sound is in the decoders directory.
	- *
	- * Documentation is in SDL_sound.h ... It's verbose, honest. :)
	- *
	- * Please see the file COPYING in the source's root directory.
	- *
	- * This file written by Ryan C. Gordon. (icculus@icculus.org)
	- */
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#include <stdio.h>
	-#include <stdlib.h>
	-#include <string.h>
	-#include <ctype.h>
	-
	-#include "SDL.h"
	-#include "SDL_thread.h"
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-
	-/* The various decoder drivers... */
	-
	-#if (defined SOUND_SUPPORTS_SMPEG)
	-extern const Sound_DecoderFunctions __Sound_DecoderFunctions_SMPEG;
	-#endif
	-
	-#if (defined SOUND_SUPPORTS_MPGLIB)
	-extern const Sound_DecoderFunctions __Sound_DecoderFunctions_MPGLIB;
	-#endif
	-
	-#if (defined SOUND_SUPPORTS_MIKMOD)
	-extern const Sound_DecoderFunctions __Sound_DecoderFunctions_MIKMOD;
	-#endif
	-
	-#if (defined SOUND_SUPPORTS_MODPLUG)
	-extern const Sound_DecoderFunctions __Sound_DecoderFunctions_MODPLUG;
	-#endif
	-
	-#if (defined SOUND_SUPPORTS_WAV)
	-extern const Sound_DecoderFunctions __Sound_DecoderFunctions_WAV;
	-#endif
	-
	-#if (defined SOUND_SUPPORTS_AIFF)
	-extern const Sound_DecoderFunctions __Sound_DecoderFunctions_AIFF;
	-#endif
	-
	-#if (defined SOUND_SUPPORTS_AU)
	-extern const Sound_DecoderFunctions __Sound_DecoderFunctions_AU;
	-#endif
	-
	-#if (defined SOUND_SUPPORTS_OGG)
	-extern const Sound_DecoderFunctions __Sound_DecoderFunctions_OGG;
	-#endif
	-
	-#if (defined SOUND_SUPPORTS_VOC)
	-extern const Sound_DecoderFunctions __Sound_DecoderFunctions_VOC;
	-#endif
	-
	-#if (defined SOUND_SUPPORTS_RAW)
	-extern const Sound_DecoderFunctions __Sound_DecoderFunctions_RAW;
	-#endif
	-
	-#if (defined SOUND_SUPPORTS_SHN)
	-extern const Sound_DecoderFunctions __Sound_DecoderFunctions_SHN;
	-#endif
	-
	-#if (defined SOUND_SUPPORTS_MIDI)
	-extern const Sound_DecoderFunctions __Sound_DecoderFunctions_MIDI;
	-#endif
	-
	-#if (defined SOUND_SUPPORTS_FLAC)
	-extern const Sound_DecoderFunctions __Sound_DecoderFunctions_FLAC;
	-#endif
	-
	-#if (defined SOUND_SUPPORTS_QUICKTIME)
	-extern const Sound_DecoderFunctions __Sound_DecoderFunctions_QuickTime;
	-#endif
	-
	-#if (defined SOUND_SUPPORTS_SPEEX)
	-extern const Sound_DecoderFunctions __Sound_DecoderFunctions_SPEEX;
	-#endif
	-
	-typedef struct
	-{
	- int available;
	- const Sound_DecoderFunctions *funcs;
	-} decoder_element;
	-
	-static decoder_element decoders[] =
	-{
	-#if (defined SOUND_SUPPORTS_SMPEG)
	- { 0, &__Sound_DecoderFunctions_SMPEG },
	-#endif
	-
	-#if (defined SOUND_SUPPORTS_MPGLIB)
	- { 0, &__Sound_DecoderFunctions_MPGLIB },
	-#endif
	-
	-#if (defined SOUND_SUPPORTS_MODPLUG)
	- { 0, &__Sound_DecoderFunctions_MODPLUG },
	-#endif
	-
	-#if (defined SOUND_SUPPORTS_MIKMOD)
	- { 0, &__Sound_DecoderFunctions_MIKMOD },
	-#endif
	-
	-#if (defined SOUND_SUPPORTS_WAV)
	- { 0, &__Sound_DecoderFunctions_WAV },
	-#endif
	-
	-#if (defined SOUND_SUPPORTS_AIFF)
	- { 0, &__Sound_DecoderFunctions_AIFF },
	-#endif
	-
	-#if (defined SOUND_SUPPORTS_AU)
	- { 0, &__Sound_DecoderFunctions_AU },
	-#endif
	-
	-#if (defined SOUND_SUPPORTS_OGG)
	- { 0, &__Sound_DecoderFunctions_OGG },
	-#endif
	-
	-#if (defined SOUND_SUPPORTS_VOC)
	- { 0, &__Sound_DecoderFunctions_VOC },
	-#endif
	-
	-#if (defined SOUND_SUPPORTS_RAW)
	- { 0, &__Sound_DecoderFunctions_RAW },
	-#endif
	-
	-#if (defined SOUND_SUPPORTS_SHN)
	- { 0, &__Sound_DecoderFunctions_SHN },
	-#endif
	-
	-#if (defined SOUND_SUPPORTS_FLAC)
	- { 0, &__Sound_DecoderFunctions_FLAC },
	-#endif
	-
	-#if (defined SOUND_SUPPORTS_MIDI)
	- { 0, &__Sound_DecoderFunctions_MIDI },
	-#endif
	-
	-#if (defined SOUND_SUPPORTS_QUICKTIME)
	- { 0, &__Sound_DecoderFunctions_QuickTime },
	-#endif
	-
	-#if (defined SOUND_SUPPORTS_SPEEX)
	- { 0, &__Sound_DecoderFunctions_SPEEX },
	-#endif
	-
	- { 0, NULL }
	-};
	-
	-
	-
	-/* General SDL_sound state ... */
	-
	-typedef struct __SOUND_ERRMSGTYPE__
	-{
	- Uint32 tid;
	- int error_available;
	- char error_string[128];
	- struct __SOUND_ERRMSGTYPE__ *next;
	-} ErrMsg;
	-
	-static ErrMsg *error_msgs = NULL;
	-static SDL_mutex *errorlist_mutex = NULL;
	-
	-static Sound_Sample sample_list = NULL; / this is a linked list. */
	-static SDL_mutex *samplelist_mutex = NULL;
	-
	-static const Sound_DecoderInfo **available_decoders = NULL;
	-static int initialized = 0;
	-
	-
	-/* functions ... */
	-
	-void Sound_GetLinkedVersion(Sound_Version *ver)
	-{
	- if (ver != NULL)
	- {
	- ver->major = SOUND_VER_MAJOR;
	- ver->minor = SOUND_VER_MINOR;
	- ver->patch = SOUND_VER_PATCH;
	- } /* if */
	-} /* Sound_GetLinkedVersion */
	-
	-
	-int Sound_Init(void)
	-{
	- size_t i;
	- size_t pos = 0;
	- size_t total = sizeof (decoders) / sizeof (decoders[0]);
	- BAIL_IF_MACRO(initialized, ERR_IS_INITIALIZED, 0);
	-
	- sample_list = NULL;
	- error_msgs = NULL;
	-
	- available_decoders = (const Sound_DecoderInfo **)
	- malloc((total) * sizeof (Sound_DecoderInfo *));
	- BAIL_IF_MACRO(available_decoders == NULL, ERR_OUT_OF_MEMORY, 0);
	-
	- SDL_InitSubSystem(SDL_INIT_AUDIO);
	-
	- errorlist_mutex = SDL_CreateMutex();
	- samplelist_mutex = SDL_CreateMutex();
	-
	- for (i = 0; decoders[i].funcs != NULL; i++)
	- {
	- decoders[i].available = decoders[i].funcs->init();
	- if (decoders[i].available)
	- {
	- available_decoders[pos] = &(decoders[i].funcs->info);
	- pos++;
	- } /* if */
	- } /* for */
	-
	- available_decoders[pos] = NULL;
	-
	- initialized = 1;
	- return(1);
	-} /* Sound_Init */
	-
	-
	-int Sound_Quit(void)
	-{
	- ErrMsg *err;
	- ErrMsg *nexterr = NULL;
	- size_t i;
	-
	- BAIL_IF_MACRO(!initialized, ERR_NOT_INITIALIZED, 0);
	-
	- while (((volatile Sound_Sample *) sample_list) != NULL)
	- Sound_FreeSample(sample_list);
	-
	- initialized = 0;
	-
	- SDL_DestroyMutex(samplelist_mutex);
	- samplelist_mutex = NULL;
	- sample_list = NULL;
	-
	- for (i = 0; decoders[i].funcs != NULL; i++)
	- {
	- if (decoders[i].available)
	- {
	- decoders[i].funcs->quit();
	- decoders[i].available = 0;
	- } /* if */
	- } /* for */
	-
	- if (available_decoders != NULL)
	- free((void *) available_decoders);
	- available_decoders = NULL;
	-
	- /* clean up error state for each thread... */
	- SDL_LockMutex(errorlist_mutex);
	- for (err = error_msgs; err != NULL; err = nexterr)
	- {
	- nexterr = err->next;
	- free(err);
	- } /* for */
	- error_msgs = NULL;
	- SDL_UnlockMutex(errorlist_mutex);
	- SDL_DestroyMutex(errorlist_mutex);
	- errorlist_mutex = NULL;
	-
	- return(1);
	-} /* Sound_Quit */
	-
	-
	-const Sound_DecoderInfo **Sound_AvailableDecoders(void)
	-{
	- return(available_decoders); /* READ. ONLY. */
	-} /* Sound_AvailableDecoders */
	-
	-
	-static ErrMsg *findErrorForCurrentThread(void)
	-{
	- ErrMsg *i;
	- Uint32 tid;
	-
	- if (error_msgs != NULL)
	- {
	- tid = SDL_ThreadID();
	-
	- SDL_LockMutex(errorlist_mutex);
	- for (i = error_msgs; i != NULL; i = i->next)
	- {
	- if (i->tid == tid)
	- {
	- SDL_UnlockMutex(errorlist_mutex);
	- return(i);
	- } /* if */
	- } /* for */
	- SDL_UnlockMutex(errorlist_mutex);
	- } /* if */
	-
	- return(NULL); /* no error available. */
	-} /* findErrorForCurrentThread */
	-
	-
	-const char *Sound_GetError(void)
	-{
	- const char *retval = NULL;
	- ErrMsg *err;
	-
	- if (!initialized)
	- return(ERR_NOT_INITIALIZED);
	-
	- err = findErrorForCurrentThread();
	- if ((err != NULL) && (err->error_available))
	- {
	- retval = err->error_string;
	- err->error_available = 0;
	- } /* if */
	-
	- return(retval);
	-} /* Sound_GetError */
	-
	-
	-void Sound_ClearError(void)
	-{
	- ErrMsg *err;
	-
	- if (!initialized)
	- return;
	-
	- err = findErrorForCurrentThread();
	- if (err != NULL)
	- err->error_available = 0;
	-} /* Sound_ClearError */
	-
	-
	-/*
	- * This is declared in the internal header.
	- */
	-void __Sound_SetError(const char *str)
	-{
	- ErrMsg *err;
	-
	- if (str == NULL)
	- return;
	-
	- SNDDBG(("__Sound_SetError(\"%s\");%s\n", str,
	- (initialized) ? "" : " [NOT INITIALIZED!]"));
	-
	- if (!initialized)
	- return;
	-
	- err = findErrorForCurrentThread();
	- if (err == NULL)
	- {
	- err = (ErrMsg *) malloc(sizeof (ErrMsg));
	- if (err == NULL)
	- return; /* uhh...? */
	-
	- memset((void *) err, '\0', sizeof (ErrMsg));
	- err->tid = SDL_ThreadID();
	-
	- SDL_LockMutex(errorlist_mutex);
	- err->next = error_msgs;
	- error_msgs = err;
	- SDL_UnlockMutex(errorlist_mutex);
	- } /* if */
	-
	- err->error_available = 1;
	- strncpy(err->error_string, str, sizeof (err->error_string));
	- err->error_string[sizeof (err->error_string) - 1] = '\0';
	-} /* __Sound_SetError */
	-
	-
	-Uint32 __Sound_convertMsToBytePos(Sound_AudioInfo *info, Uint32 ms)
	-{
	- /* "frames" == "sample frames" */
	- float frames_per_ms = ((float) info->rate) / 1000.0f;
	- Uint32 frame_offset = (Uint32) (frames_per_ms * ((float) ms));
	- Uint32 frame_size = (Uint32) ((info->format & 0xFF) / 8) * info->channels;
	- return(frame_offset * frame_size);
	-} /* __Sound_convertMsToBytePos */
	-
	-
	-/*
	- * -ansi and -pedantic flags prevent use of strcasecmp() on Linux, and
	- * I honestly don't want to mess around with figuring out if a given
	- * platform has "strcasecmp", "stricmp", or
	- * "compare_two_damned_strings_case_insensitive", which I hear is in the
	- * next release of Carbon. :) This is exported so decoders may use it if
	- * they like.
	- */
	-int __Sound_strcasecmp(const char x, const char y)
	-{
	- int ux, uy;
	-
	- if (x == y) /* same pointer? Both NULL? */
	- return(0);
	-
	- if (x == NULL)
	- return(-1);
	-
	- if (y == NULL)
	- return(1);
	-
	- do
	- {
	- ux = toupper((int) *x);
	- uy = toupper((int) *y);
	- if (ux > uy)
	- return(1);
	- else if (ux < uy)
	- return(-1);
	- x++;
	- y++;
	- } while ((ux) && (uy));
	-
	- return(0);
	-} /* __Sound_strcasecmp */
	-
	-
	-/*
	- * Allocate a Sound_Sample, and fill in most of its fields. Those that need
	- * to be filled in later, by a decoder, will be initialized to zero.
	- */
	-static Sound_Sample alloc_sample(SDL_RWops rw, Sound_AudioInfo *desired,
	- Uint32 bufferSize)
	-{
	- Sound_Sample *retval = malloc(sizeof (Sound_Sample));
	- Sound_SampleInternal *internal = malloc(sizeof (Sound_SampleInternal));
	- if ((retval == NULL) \|\| (internal == NULL))
	- {
	- __Sound_SetError(ERR_OUT_OF_MEMORY);
	- if (retval)
	- free(retval);
	- if (internal)
	- free(internal);
	-
	- return(NULL);
	- } /* if */
	-
	- memset(retval, '\0', sizeof (Sound_Sample));
	- memset(internal, '\0', sizeof (Sound_SampleInternal));
	-
	- assert(bufferSize > 0);
	- retval->buffer = malloc(bufferSize); /* pure ugly. */
	- if (!retval->buffer)
	- {
	- __Sound_SetError(ERR_OUT_OF_MEMORY);
	- free(internal);
	- free(retval);
	- return(NULL);
	- } /* if */
	- memset(retval->buffer, '\0', bufferSize);
	- retval->buffer_size = bufferSize;
	-
	- if (desired != NULL)
	- memcpy(&retval->desired, desired, sizeof (Sound_AudioInfo));
	-
	- internal->rw = rw;
	- retval->opaque = internal;
	- return(retval);
	-} /* alloc_sample */
	-
	-
	-#if (defined DEBUG_CHATTER)
	-static __inline__ const char *fmt_to_str(Uint16 fmt)
	-{
	- switch(fmt)
	- {
	- case AUDIO_U8:
	- return("U8");
	- case AUDIO_S8:
	- return("S8");
	- case AUDIO_U16LSB:
	- return("U16LSB");
	- case AUDIO_S16LSB:
	- return("S16LSB");
	- case AUDIO_U16MSB:
	- return("U16MSB");
	- case AUDIO_S16MSB:
	- return("S16MSB");
	- } /* switch */
	-
	- return("Unknown");
	-} /* fmt_to_str */
	-#endif
	-
	-
	-/*
	- * The bulk of the Sound_NewSample() work is done here...
	- * Ask the specified decoder to handle the data in (rw), and if
	- * so, construct the Sound_Sample. Otherwise, try to wind (rw)'s stream
	- * back to where it was, and return false.
	- */
	-static int init_sample(const Sound_DecoderFunctions *funcs,
	- Sound_Sample sample, const char ext,
	- Sound_AudioInfo *_desired)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- Sound_AudioInfo desired;
	- int pos = SDL_RWtell(internal->rw);
	-
	- /* fill in the funcs for this decoder... */
	- sample->decoder = &funcs->info;
	- internal->funcs = funcs;
	- if (!funcs->open(sample, ext))
	- {
	- SDL_RWseek(internal->rw, pos, SEEK_SET); /* set for next try... */
	- return(0);
	- } /* if */
	-
	- /* success; we've got a decoder! */
	-
	- /* Now we need to set up the conversion buffer... */
	-
	- memcpy(&desired, (_desired != NULL) ? _desired : &sample->actual,
	- sizeof (Sound_AudioInfo));
	-
	- if (desired.format == 0)
	- desired.format = sample->actual.format;
	- if (desired.channels == 0)
	- desired.channels = sample->actual.channels;
	- if (desired.rate == 0)
	- desired.rate = sample->actual.rate;
	-
	- if (Sound_BuildAudioCVT(&internal->sdlcvt,
	- sample->actual.format,
	- sample->actual.channels,
	- sample->actual.rate,
	- desired.format,
	- desired.channels,
	- desired.rate,
	- sample->buffer_size) == -1)
	- {
	- __Sound_SetError(SDL_GetError());
	- funcs->close(sample);
	- SDL_RWseek(internal->rw, pos, SEEK_SET); /* set for next try... */
	- return(0);
	- } /* if */
	-
	- if (internal->sdlcvt.len_mult > 1)
	- {
	- void *rc = realloc(sample->buffer,
	- sample->buffer_size * internal->sdlcvt.len_mult);
	- if (rc == NULL)
	- {
	- funcs->close(sample);
	- SDL_RWseek(internal->rw, pos, SEEK_SET); /* set for next try... */
	- return(0);
	- } /* if */
	-
	- sample->buffer = rc;
	- } /* if */
	-
	- /* these pointers are all one and the same. */
	- memcpy(&sample->desired, &desired, sizeof (Sound_AudioInfo));
	- internal->sdlcvt.buf = internal->buffer = sample->buffer;
	- internal->buffer_size = sample->buffer_size / internal->sdlcvt.len_mult;
	- internal->sdlcvt.len = internal->buffer_size;
	-
	- /* Prepend our new Sound_Sample to the sample_list... */
	- SDL_LockMutex(samplelist_mutex);
	- internal->next = sample_list;
	- if (sample_list != NULL)
	- ((Sound_SampleInternal *) sample_list->opaque)->prev = sample;
	- sample_list = sample;
	- SDL_UnlockMutex(samplelist_mutex);
	-
	- SNDDBG(("New sample DESIRED format: %s format, %d rate, %d channels.\n",
	- fmt_to_str(sample->desired.format),
	- sample->desired.rate,
	- sample->desired.channels));
	-
	- SNDDBG(("New sample ACTUAL format: %s format, %d rate, %d channels.\n",
	- fmt_to_str(sample->actual.format),
	- sample->actual.rate,
	- sample->actual.channels));
	-
	- SNDDBG(("On-the-fly conversion: %s.\n",
	- internal->sdlcvt.needed ? "ENABLED" : "DISABLED"));
	-
	- return(1);
	-} /* init_sample */
	-
	-
	-Sound_Sample Sound_NewSample(SDL_RWops rw, const char *ext,
	- Sound_AudioInfo *desired, Uint32 bSize)
	-{
	- Sound_Sample *retval;
	- decoder_element *decoder;
	-
	- /* sanity checks. */
	- BAIL_IF_MACRO(!initialized, ERR_NOT_INITIALIZED, NULL);
	- BAIL_IF_MACRO(rw == NULL, ERR_INVALID_ARGUMENT, NULL);
	-
	- retval = alloc_sample(rw, desired, bSize);
	- if (!retval)
	- return(NULL); /* alloc_sample() sets error message... */
	-
	- if (ext != NULL)
	- {
	- for (decoder = &decoders[0]; decoder->funcs != NULL; decoder++)
	- {
	- if (decoder->available)
	- {
	- const char **decoderExt = decoder->funcs->info.extensions;
	- while (*decoderExt)
	- {
	- if (__Sound_strcasecmp(*decoderExt, ext) == 0)
	- {
	- if (init_sample(decoder->funcs, retval, ext, desired))
	- return(retval);
	- break; /* done with this decoder either way. */
	- } /* if */
	- decoderExt++;
	- } /* while */
	- } /* if */
	- } /* for */
	- } /* if */
	-
	- /* no direct extension match? Try everything we've got... */
	- for (decoder = &decoders[0]; decoder->funcs != NULL; decoder++)
	- {
	- if (decoder->available)
	- {
	- int should_try = 1;
	- const char **decoderExt = decoder->funcs->info.extensions;
	-
	- /* skip if we would have tried decoder above... */
	- while (*decoderExt)
	- {
	- if (__Sound_strcasecmp(*decoderExt, ext) == 0)
	- {
	- should_try = 0;
	- break;
	- } /* if */
	- decoderExt++;
	- } /* while */
	-
	- if (should_try)
	- {
	- if (init_sample(decoder->funcs, retval, ext, desired))
	- return(retval);
	- } /* if */
	- } /* if */
	- } /* for */
	-
	- /* nothing could handle the sound data... */
	- free(retval->opaque);
	- if (retval->buffer != NULL)
	- free(retval->buffer);
	- free(retval);
	- SDL_RWclose(rw);
	- __Sound_SetError(ERR_UNSUPPORTED_FORMAT);
	- return(NULL);
	-} /* Sound_NewSample */
	-
	-
	-Sound_Sample Sound_NewSampleFromFile(const char filename,
	- Sound_AudioInfo *desired,
	- Uint32 bufferSize)
	-{
	- const char *ext;
	- SDL_RWops *rw;
	-
	- BAIL_IF_MACRO(!initialized, ERR_NOT_INITIALIZED, NULL);
	- BAIL_IF_MACRO(filename == NULL, ERR_INVALID_ARGUMENT, NULL);
	-
	- ext = strrchr(filename, '.');
	- rw = SDL_RWFromFile(filename, "rb");
	- BAIL_IF_MACRO(rw == NULL, SDL_GetError(), NULL);
	-
	- if (ext != NULL)
	- ext++;
	-
	- return(Sound_NewSample(rw, ext, desired, bufferSize));
	-} /* Sound_NewSampleFromFile */
	-
	-
	-void Sound_FreeSample(Sound_Sample *sample)
	-{
	- Sound_SampleInternal *internal;
	-
	- if (!initialized)
	- {
	- __Sound_SetError(ERR_NOT_INITIALIZED);
	- return;
	- } /* if */
	-
	- if (sample == NULL)
	- {
	- __Sound_SetError(ERR_INVALID_ARGUMENT);
	- return;
	- } /* if */
	-
	- internal = (Sound_SampleInternal *) sample->opaque;
	-
	- SDL_LockMutex(samplelist_mutex);
	-
	- /* update the sample_list... */
	- if (internal->prev != NULL)
	- {
	- Sound_SampleInternal *prevInternal;
	- prevInternal = (Sound_SampleInternal *) internal->prev->opaque;
	- prevInternal->next = internal->next;
	- } /* if */
	- else
	- {
	- assert(sample_list == sample);
	- sample_list = internal->next;
	- } /* else */
	-
	- if (internal->next != NULL)
	- {
	- Sound_SampleInternal *nextInternal;
	- nextInternal = (Sound_SampleInternal *) internal->next->opaque;
	- nextInternal->prev = internal->prev;
	- } /* if */
	-
	- SDL_UnlockMutex(samplelist_mutex);
	-
	- /* nuke it... */
	- internal->funcs->close(sample);
	-
	- if (internal->rw != NULL) /* this condition is a "just in case" thing. */
	- SDL_RWclose(internal->rw);
	-
	- if ((internal->buffer != NULL) && (internal->buffer != sample->buffer))
	- free(internal->buffer);
	-
	- free(internal);
	-
	- if (sample->buffer != NULL)
	- free(sample->buffer);
	-
	- free(sample);
	-} /* Sound_FreeSample */
	-
	-
	-int Sound_SetBufferSize(Sound_Sample *sample, Uint32 newSize)
	-{
	- void *newBuf = NULL;
	- Sound_SampleInternal *internal = NULL;
	-
	- BAIL_IF_MACRO(!initialized, ERR_NOT_INITIALIZED, 0);
	- BAIL_IF_MACRO(sample == NULL, ERR_INVALID_ARGUMENT, 0);
	- internal = ((Sound_SampleInternal *) sample->opaque);
	- newBuf = realloc(sample->buffer, newSize * internal->sdlcvt.len_mult);
	- BAIL_IF_MACRO(newBuf == NULL, ERR_OUT_OF_MEMORY, 0);
	-
	- internal->sdlcvt.buf = internal->buffer = sample->buffer = newBuf;
	- sample->buffer_size = newSize;
	- internal->buffer_size = newSize / internal->sdlcvt.len_mult;
	- internal->sdlcvt.len = internal->buffer_size;
	-
	- return(1);
	-} /* Sound_SetBufferSize */
	-
	-
	-Uint32 Sound_Decode(Sound_Sample *sample)
	-{
	- Sound_SampleInternal *internal = NULL;
	- Uint32 retval = 0;
	-
	- /* a boatload of sanity checks... */
	- BAIL_IF_MACRO(!initialized, ERR_NOT_INITIALIZED, 0);
	- BAIL_IF_MACRO(sample == NULL, ERR_INVALID_ARGUMENT, 0);
	- BAIL_IF_MACRO(sample->flags & SOUND_SAMPLEFLAG_ERROR, ERR_PREV_ERROR, 0);
	- BAIL_IF_MACRO(sample->flags & SOUND_SAMPLEFLAG_EOF, ERR_PREV_EOF, 0);
	-
	- internal = (Sound_SampleInternal *) sample->opaque;
	-
	- assert(sample->buffer != NULL);
	- assert(sample->buffer_size > 0);
	- assert(internal->buffer != NULL);
	- assert(internal->buffer_size > 0);
	-
	- /* reset EAGAIN. Decoder can flip it back on if it needs to. */
	- sample->flags &= ~SOUND_SAMPLEFLAG_EAGAIN;
	- retval = internal->funcs->read(sample);
	-
	- if (retval > 0 && internal->sdlcvt.needed)
	- {
	- internal->sdlcvt.len = retval;
	- Sound_ConvertAudio(&internal->sdlcvt);
	- retval = internal->sdlcvt.len_cvt;
	- } /* if */
	-
	- return(retval);
	-} /* Sound_Decode */
	-
	-
	-Uint32 Sound_DecodeAll(Sound_Sample *sample)
	-{
	- Sound_SampleInternal *internal = NULL;
	- void *buf = NULL;
	- Uint32 newBufSize = 0;
	-
	- BAIL_IF_MACRO(!initialized, ERR_NOT_INITIALIZED, 0);
	- BAIL_IF_MACRO(sample->flags & SOUND_SAMPLEFLAG_EOF, ERR_PREV_EOF, 0);
	- BAIL_IF_MACRO(sample->flags & SOUND_SAMPLEFLAG_ERROR, ERR_PREV_ERROR, 0);
	-
	- internal = (Sound_SampleInternal *) sample->opaque;
	-
	- while ( ((sample->flags & SOUND_SAMPLEFLAG_EOF) == 0) &&
	- ((sample->flags & SOUND_SAMPLEFLAG_ERROR) == 0) )
	- {
	- Uint32 br = Sound_Decode(sample);
	- void *ptr = realloc(buf, newBufSize + br);
	- if (ptr == NULL)
	- {
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- __Sound_SetError(ERR_OUT_OF_MEMORY);
	- } /* if */
	- else
	- {
	- buf = ptr;
	- memcpy( ((char *) buf) + newBufSize, sample->buffer, br );
	- newBufSize += br;
	- } /* else */
	- } /* while */
	-
	- if (buf == NULL) /* ...in case first call to realloc() fails... */
	- return(sample->buffer_size);
	-
	- if (internal->buffer != sample->buffer)
	- free(internal->buffer);
	-
	- free(sample->buffer);
	-
	- internal->sdlcvt.buf = internal->buffer = sample->buffer = buf;
	- sample->buffer_size = newBufSize;
	- internal->buffer_size = newBufSize / internal->sdlcvt.len_mult;
	- internal->sdlcvt.len = internal->buffer_size;
	-
	- return(newBufSize);
	-} /* Sound_DecodeAll */
	-
	-
	-int Sound_Rewind(Sound_Sample *sample)
	-{
	- Sound_SampleInternal *internal;
	- BAIL_IF_MACRO(!initialized, ERR_NOT_INITIALIZED, 0);
	-
	- internal = (Sound_SampleInternal *) sample->opaque;
	- if (!internal->funcs->rewind(sample))
	- {
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- return(0);
	- } /* if */
	-
	- sample->flags &= ~SOUND_SAMPLEFLAG_EAGAIN;
	- sample->flags &= ~SOUND_SAMPLEFLAG_ERROR;
	- sample->flags &= ~SOUND_SAMPLEFLAG_EOF;
	-
	- return(1);
	-} /* Sound_Rewind */
	-
	-
	-int Sound_Seek(Sound_Sample *sample, Uint32 ms)
	-{
	- Sound_SampleInternal *internal;
	-
	- BAIL_IF_MACRO(!initialized, ERR_NOT_INITIALIZED, 0);
	- if (!(sample->flags & SOUND_SAMPLEFLAG_CANSEEK))
	- BAIL_MACRO(ERR_CANNOT_SEEK, 0);
	-
	- internal = (Sound_SampleInternal *) sample->opaque;
	- BAIL_IF_MACRO(!internal->funcs->seek(sample, ms), NULL, 0);
	-
	- sample->flags &= ~SOUND_SAMPLEFLAG_EAGAIN;
	- sample->flags &= ~SOUND_SAMPLEFLAG_ERROR;
	- sample->flags &= ~SOUND_SAMPLEFLAG_EOF;
	-
	- return(1);
	-} /* Sound_Rewind */
	-
	-
	-/* end of SDL_sound.c ... */
	-
	diff --git a/util/sdl/sound/SDL_sound.h b/util/sdl/sound/SDL_sound.h
	deleted file mode 100644
	index b0b8c978..00000000
	--- a/util/sdl/sound/SDL_sound.h
	+++ /dev/null
	@@ -1,674 +0,0 @@
	-/** \file SDL_sound.h */
	-
	-/*
	- * SDL_sound -- An abstract sound format decoding API.
	- * Copyright (C) 2001 Ryan C. Gordon.
	- *
	- * This library is free software; you can redistribute it and/or
	- * modify it under the terms of the GNU Lesser General Public
	- * License as published by the Free Software Foundation; either
	- * version 2.1 of the License, or (at your option) any later version.
	- *
	- * This library is distributed in the hope that it will be useful,
	- * but WITHOUT ANY WARRANTY; without even the implied warranty of
	- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	- * Lesser General Public License for more details.
	- *
	- * You should have received a copy of the GNU Lesser General Public
	- * License along with this library; if not, write to the Free Software
	- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
	- */
	-
	-/**
	- * \mainpage SDL_sound
	- *
	- * The latest version of SDL_sound can be found at:
	- * http://icculus.org/SDL_sound/
	- *
	- * The basic gist of SDL_sound is that you use an SDL_RWops to get sound data
	- * into this library, and SDL_sound will take that data, in one of several
	- * popular formats, and decode it into raw waveform data in the format of
	- * your choice. This gives you a nice abstraction for getting sound into your
	- * game or application; just feed it to SDL_sound, and it will handle
	- * decoding and converting, so you can just pass it to your SDL audio
	- * callback (or whatever). Since it gets data from an SDL_RWops, you can get
	- * the initial sound data from any number of sources: file, memory buffer,
	- * network connection, etc.
	- *
	- * As the name implies, this library depends on SDL: Simple Directmedia Layer,
	- * which is a powerful, free, and cross-platform multimedia library. It can
	- * be found at http://www.libsdl.org/
	- *
	- * Support is in place or planned for the following sound formats:
	- * - .WAV (Microsoft WAVfile RIFF data, internal.)
	- * - .VOC (Creative Labs' Voice format, internal.)
	- * - .MP3 (MPEG-1 Layer 3 support, via the SMPEG and mpglib libraries.)
	- * - .MID (MIDI music converted to Waveform data, internal.)
	- * - .MOD (MOD files, via MikMod and ModPlug.)
	- * - .OGG (Ogg files, via Ogg Vorbis libraries.)
	- * - .SPX (Speex files, via libspeex.)
	- * - .SHN (Shorten files, internal.)
	- * - .RAW (Raw sound data in any format, internal.)
	- * - .AU (Sun's Audio format, internal.)
	- * - .AIFF (Audio Interchange format, internal.)
	- * - .FLAC (Lossless audio compression, via libFLAC.)
	- *
	- * (...and more to come...)
	- *
	- * Please see the file COPYING in the source's root directory.
	- *
	- * \author Ryan C. Gordon (icculus@icculus.org)
	- * \author many others, please see CREDITS in the source's root directory.
	- */
	-
	-#ifndef _INCLUDE_SDL_SOUND_H_
	-#define _INCLUDE_SDL_SOUND_H_
	-
	-#include "SDL.h"
	-#include "SDL_endian.h"
	-
	-#ifdef __cplusplus
	-extern "C" {
	-#endif
	-
	-#ifndef DOXYGEN_SHOULD_IGNORE_THIS
	-
	-#ifndef SDLCALL /* may not be defined with older SDL releases. */
	-#define SDLCALL
	-#endif
	-
	-#ifdef SDL_SOUND_DLL_EXPORTS
	-# define SNDDECLSPEC __declspec(dllexport)
	-#else
	-# define SNDDECLSPEC
	-#endif
	-
	-#define SOUND_VER_MAJOR 1
	-#define SOUND_VER_MINOR 0
	-#define SOUND_VER_PATCH 3
	-#endif
	-
	-
	-/**
	- * \enum Sound_SampleFlags
	- * \brief Flags that are used in a Sound_Sample to show various states.
	- *
	- * To use:
	- * \code
	- * if (sample->flags & SOUND_SAMPLEFLAG_ERROR) { dosomething(); }
	- * \endcode
	- *
	- * \sa Sound_SampleNew
	- * \sa Sound_SampleNewFromFile
	- * \sa Sound_SampleDecode
	- * \sa Sound_SampleDecodeAll
	- * \sa Sound_SampleSeek
	- */
	-typedef enum
	-{
	- SOUND_SAMPLEFLAG_NONE = 0, /*< No special attributes. /
	-
	- /* these are set at sample creation time... */
	- SOUND_SAMPLEFLAG_CANSEEK = 1, /*< Sample can seek to arbitrary points. /
	-
	- /* these are set during decoding... */
	- SOUND_SAMPLEFLAG_EOF = 1 << 29, /*< End of input stream. /
	- SOUND_SAMPLEFLAG_ERROR = 1 << 30, /*< Unrecoverable error. /
	- SOUND_SAMPLEFLAG_EAGAIN = 1 << 31 /*< Function would block, or temp error. /
	-} Sound_SampleFlags;
	-
	-
	-/**
	- * \struct Sound_AudioInfo
	- * \brief Information about an existing sample's format.
	- *
	- * These are the basics of a decoded sample's data structure: data format
	- * (see AUDIO_U8 and friends in SDL_audio.h), number of channels, and sample
	- * rate. If you need more explanation than that, you should stop developing
	- * sound code right now.
	- *
	- * \sa Sound_SampleNew
	- * \sa Sound_SampleNewFromFile
	- */
	-typedef struct
	-{
	- Uint16 format; /*< Equivalent of SDL_AudioSpec.format. /
	- Uint8 channels; /*< Number of sound channels. 1 == mono, 2 == stereo. /
	- Uint32 rate; /*< Sample rate; frequency of sample points per second. /
	-} Sound_AudioInfo;
	-
	-
	-/**
	- * \struct Sound_DecoderInfo
	- * \brief Information about available soudn decoders.
	- *
	- * Each decoder sets up one of these structs, which can be retrieved via
	- * the Sound_AvailableDecoders() function. EVERY FIELD IN THIS IS READ-ONLY.
	- *
	- * The extensions field is a NULL-terminated list of ASCIZ strings. You
	- * should read it like this:
	- *
	- * \code
	- * const char **ext;
	- * for (ext = info->extensions; *ext != NULL; ext++) {
	- * printf(" File extension \"%s\"\n", *ext);
	- * }
	- * \endcode
	- *
	- * \sa Sound_AvailableDecoders
	- */
	-typedef struct
	-{
	- const char extensions; /< File extensions, list ends with NULL. */
	- const char description; /< Human readable description of decoder. /
	- const char author; /< "Name Of Author \<email@emailhost.dom\>" /
	- const char url; /< URL specific to this decoder. /
	-} Sound_DecoderInfo;
	-
	-
	-
	-/**
	- * \struct Sound_Sample
	- * \brief Represents sound data in the process of being decoded.
	- *
	- * The Sound_Sample structure is the heart of SDL_sound. This holds
	- * information about a source of sound data as it is being decoded.
	- * EVERY FIELD IN THIS IS READ-ONLY. Please use the API functions to
	- * change them.
	- */
	-typedef struct
	-{
	- void opaque; /< Internal use only. Don't touch. /
	- const Sound_DecoderInfo decoder; /< Decoder used for this sample. /
	- Sound_AudioInfo desired; /*< Desired audio format for conversion. /
	- Sound_AudioInfo actual; /*< Actual audio format of sample. /
	- void buffer; /< Decoded sound data lands in here. /
	- Uint32 buffer_size; /*< Current size of (buffer), in bytes (Uint8). /
	- Sound_SampleFlags flags; /*< Flags relating to this sample. /
	-} Sound_Sample;
	-
	-
	-/**
	- * \struct Sound_Version
	- * \brief Information the version of SDL_sound in use.
	- *
	- * Represents the library's version as three levels: major revision
	- * (increments with massive changes, additions, and enhancements),
	- * minor revision (increments with backwards-compatible changes to the
	- * major revision), and patchlevel (increments with fixes to the minor
	- * revision).
	- *
	- * \sa SOUND_VERSION
	- * \sa Sound_GetLinkedVersion
	- */
	-typedef struct
	-{
	- int major; /*< major revision /
	- int minor; /*< minor revision /
	- int patch; /*< patchlevel /
	-} Sound_Version;
	-
	-
	-/* functions and macros... */
	-
	-/**
	- * \def SOUND_VERSION(x)
	- * \brief Macro to determine SDL_sound version program was compiled against.
	- *
	- * This macro fills in a Sound_Version structure with the version of the
	- * library you compiled against. This is determined by what header the
	- * compiler uses. Note that if you dynamically linked the library, you might
	- * have a slightly newer or older version at runtime. That version can be
	- * determined with Sound_GetLinkedVersion(), which, unlike SOUND_VERSION,
	- * is not a macro.
	- *
	- * \param x A pointer to a Sound_Version struct to initialize.
	- *
	- * \sa Sound_Version
	- * \sa Sound_GetLinkedVersion
	- */
	-#define SOUND_VERSION(x) \
	-{ \
	- (x)->major = SOUND_VER_MAJOR; \
	- (x)->minor = SOUND_VER_MINOR; \
	- (x)->patch = SOUND_VER_PATCH; \
	-}
	-
	-
	-/**
	- * \fn void Sound_GetLinkedVersion(Sound_Version *ver)
	- * \brief Get the version of SDL_sound that is linked against your program.
	- *
	- * If you are using a shared library (DLL) version of SDL_sound, then it is
	- * possible that it will be different than the version you compiled against.
	- *
	- * This is a real function; the macro SOUND_VERSION tells you what version
	- * of SDL_sound you compiled against:
	- *
	- * \code
	- * Sound_Version compiled;
	- * Sound_Version linked;
	- *
	- * SOUND_VERSION(&compiled);
	- * Sound_GetLinkedVersion(&linked);
	- * printf("We compiled against SDL_sound version %d.%d.%d ...\n",
	- * compiled.major, compiled.minor, compiled.patch);
	- * printf("But we linked against SDL_sound version %d.%d.%d.\n",
	- * linked.major, linked.minor, linked.patch);
	- * \endcode
	- *
	- * This function may be called safely at any time, even before Sound_Init().
	- *
	- * \param ver Sound_Version structure to fill with shared library's version.
	- *
	- * \sa Sound_Version
	- * \sa SOUND_VERSION
	- */
	-SNDDECLSPEC void SDLCALL Sound_GetLinkedVersion(Sound_Version *ver);
	-
	-
	-/**
	- * \fn Sound_Init(void)
	- * \brief Initialize SDL_sound.
	- *
	- * This must be called before any other SDL_sound function (except perhaps
	- * Sound_GetLinkedVersion()). You should call SDL_Init() before calling this.
	- * Sound_Init() will attempt to call SDL_Init(SDL_INIT_AUDIO), just in case.
	- * This is a safe behaviour, but it may not configure SDL to your liking by
	- * itself.
	- *
	- * \return nonzero on success, zero on error. Specifics of the
	- * error can be gleaned from Sound_GetError().
	- *
	- * \sa Sound_Quit
	- */
	-SNDDECLSPEC int SDLCALL Sound_Init(void);
	-
	-
	-/**
	- * \fn Sound_Quit(void)
	- * \brief Shutdown SDL_sound.
	- *
	- * This closes any SDL_RWops that were being used as sound sources, and frees
	- * any resources in use by SDL_sound.
	- *
	- * All Sound_Sample pointers you had prior to this call are INVALIDATED.
	- *
	- * Once successfully deinitialized, Sound_Init() can be called again to
	- * restart the subsystem. All default API states are restored at this
	- * point.
	- *
	- * You should call this BEFORE SDL_Quit(). This will NOT call SDL_Quit()
	- * for you!
	- *
	- * \return nonzero on success, zero on error. Specifics of the error
	- * can be gleaned from Sound_GetError(). If failure, state of
	- * SDL_sound is undefined, and probably badly screwed up.
	- *
	- * \sa Sound_Init
	- */
	-SNDDECLSPEC int SDLCALL Sound_Quit(void);
	-
	-
	-/**
	- * \fn const Sound_DecoderInfo **Sound_AvailableDecoders(void)
	- * \brief Get a list of sound formats supported by this version of SDL_sound.
	- *
	- * This is for informational purposes only. Note that the extension listed is
	- * merely convention: if we list "MP3", you can open an MPEG-1 Layer 3 audio
	- * file with an extension of "XYZ", if you like. The file extensions are
	- * informational, and only required as a hint to choosing the correct
	- * decoder, since the sound data may not be coming from a file at all, thanks
	- * to the abstraction that an SDL_RWops provides.
	- *
	- * The returned value is an array of pointers to Sound_DecoderInfo structures,
	- * with a NULL entry to signify the end of the list:
	- *
	- * \code
	- * Sound_DecoderInfo **i;
	- *
	- * for (i = Sound_AvailableDecoders(); *i != NULL; i++)
	- * {
	- * printf("Supported sound format: [%s], which is [%s].\n",
	- * i->extension, i->description);
	- * // ...and other fields...
	- * }
	- * \endcode
	- *
	- * The return values are pointers to static internal memory, and should
	- * be considered READ ONLY, and never freed.
	- *
	- * \return READ ONLY Null-terminated array of READ ONLY structures.
	- *
	- * \sa Sound_DecoderInfo
	- */
	-SNDDECLSPEC const Sound_DecoderInfo ** SDLCALL Sound_AvailableDecoders(void);
	-
	-
	-/**
	- * \fn const char *Sound_GetError(void)
	- * \brief Get the last SDL_sound error message as a null-terminated string.
	- *
	- * This will be NULL if there's been no error since the last call to this
	- * function. The pointer returned by this call points to an internal buffer,
	- * and should not be deallocated. Each thread has a unique error state
	- * associated with it, but each time a new error message is set, it will
	- * overwrite the previous one associated with that thread. It is safe to call
	- * this function at anytime, even before Sound_Init().
	- *
	- * \return READ ONLY string of last error message.
	- *
	- * \sa Sound_ClearError
	- */
	-SNDDECLSPEC const char * SDLCALL Sound_GetError(void);
	-
	-
	-/**
	- * \fn void Sound_ClearError(void)
	- * \brief Clear the current error message.
	- *
	- * The next call to Sound_GetError() after Sound_ClearError() will return NULL.
	- *
	- * \sa Sound_GetError
	- */
	-SNDDECLSPEC void SDLCALL Sound_ClearError(void);
	-
	-
	-/**
	- * \fn Sound_Sample Sound_NewSample(SDL_RWops rw, const char ext, Sound_AudioInfo desired, Uint32 bufferSize)
	- * \brief Start decoding a new sound sample.
	- *
	- * The data is read via an SDL_RWops structure (see SDL_rwops.h in the SDL
	- * include directory), so it may be coming from memory, disk, network stream,
	- * etc. The (ext) parameter is merely a hint to determining the correct
	- * decoder; if you specify, for example, "mp3" for an extension, and one of
	- * the decoders lists that as a handled extension, then that decoder is given
	- * first shot at trying to claim the data for decoding. If none of the
	- * extensions match (or the extension is NULL), then every decoder examines
	- * the data to determine if it can handle it, until one accepts it. In such a
	- * case your SDL_RWops will need to be capable of rewinding to the start of
	- * the stream.
	- *
	- * If no decoders can handle the data, a NULL value is returned, and a human
	- * readable error message can be fetched from Sound_GetError().
	- *
	- * Optionally, a desired audio format can be specified. If the incoming data
	- * is in a different format, SDL_sound will convert it to the desired format
	- * on the fly. Note that this can be an expensive operation, so it may be
	- * wise to convert data before you need to play it back, if possible, or
	- * make sure your data is initially in the format that you need it in.
	- * If you don't want to convert the data, you can specify NULL for a desired
	- * format. The incoming format of the data, preconversion, can be found
	- * in the Sound_Sample structure.
	- *
	- * Note that the raw sound data "decoder" needs you to specify both the
	- * extension "RAW" and a "desired" format, or it will refuse to handle
	- * the data. This is to prevent it from catching all formats unsupported
	- * by the other decoders.
	- *
	- * Finally, specify an initial buffer size; this is the number of bytes that
	- * will be allocated to store each read from the sound buffer. The more you
	- * can safely allocate, the more decoding can be done in one block, but the
	- * more resources you have to use up, and the longer each decoding call will
	- * take. Note that different data formats require more or less space to
	- * store. This buffer can be resized via Sound_SetBufferSize() ...
	- *
	- * The buffer size specified must be a multiple of the size of a single
	- * sample point. So, if you want 16-bit, stereo samples, then your sample
	- * point size is (2 channels * 16 bits), or 32 bits per sample, which is four
	- * bytes. In such a case, you could specify 128 or 132 bytes for a buffer,
	- * but not 129, 130, or 131 (although in reality, you'll want to specify a
	- * MUCH larger buffer).
	- *
	- * When you are done with this Sound_Sample pointer, you can dispose of it
	- * via Sound_FreeSample().
	- *
	- * You do not have to keep a reference to (rw) around. If this function
	- * suceeds, it stores (rw) internally (and disposes of it during the call
	- * to Sound_FreeSample()). If this function fails, it will dispose of the
	- * SDL_RWops for you.
	- *
	- * \param rw SDL_RWops with sound data.
	- * \param ext File extension normally associated with a data format.
	- * Can usually be NULL.
	- * \param desired Format to convert sound data into. Can usually be NULL,
	- * if you don't need conversion.
	- * \param bufferSize Size, in bytes, to allocate for the decoding buffer.
	- * \return Sound_Sample pointer, which is used as a handle to several other
	- * SDL_sound APIs. NULL on error. If error, use
	- * Sound_GetError() to see what went wrong.
	- *
	- * \sa Sound_NewSampleFromFile
	- * \sa Sound_SetBufferSize
	- * \sa Sound_Decode
	- * \sa Sound_DecodeAll
	- * \sa Sound_Seek
	- * \sa Sound_Rewind
	- * \sa Sound_FreeSample
	- */
	-SNDDECLSPEC Sound_Sample * SDLCALL Sound_NewSample(SDL_RWops *rw,
	- const char *ext,
	- Sound_AudioInfo *desired,
	- Uint32 bufferSize);
	-
	-/**
	- * \fn Sound_Sample Sound_NewSampleFromFile(const char filename, Sound_AudioInfo *desired, Uint32 bufferSize)
	- * \brief Start decoding a new sound sample from a file on disk.
	- *
	- * This is identical to Sound_NewSample(), but it creates an SDL_RWops for you
	- * from the file located in (filename). Note that (filename) is specified in
	- * platform-dependent notation. ("C:\\music\\mysong.mp3" on windows, and
	- * "/home/icculus/music/mysong.mp3" or whatever on Unix, etc.)
	- * Sound_NewSample()'s "ext" parameter is gleaned from the contents of
	- * (filename).
	- *
	- * \param filename file containing sound data.
	- * \param desired Format to convert sound data into. Can usually be NULL,
	- * if you don't need conversion.
	- * \param bufferSize size, in bytes, of initial read buffer.
	- * \return Sound_Sample pointer, which is used as a handle to several other
	- * SDL_sound APIs. NULL on error. If error, use
	- * Sound_GetError() to see what went wrong.
	- *
	- * \sa Sound_NewSample
	- * \sa Sound_SetBufferSize
	- * \sa Sound_Decode
	- * \sa Sound_DecodeAll
	- * \sa Sound_Seek
	- * \sa Sound_Rewind
	- * \sa Sound_FreeSample
	- */
	-SNDDECLSPEC Sound_Sample * SDLCALL Sound_NewSampleFromFile(const char *fname,
	- Sound_AudioInfo *desired,
	- Uint32 bufferSize);
	-
	-/**
	- * \fn void Sound_FreeSample(Sound_Sample *sample)
	- * \brief Dispose of a Sound_Sample.
	- *
	- * This will also close/dispose of the SDL_RWops that was used at creation
	- * time, so there's no need to keep a reference to that around.
	- * The Sound_Sample pointer is invalid after this call, and will almost
	- * certainly result in a crash if you attempt to keep using it.
	- *
	- * \param sample The Sound_Sample to delete.
	- *
	- * \sa Sound_NewSample
	- * \sa Sound_NewSampleFromFile
	- */
	-SNDDECLSPEC void SDLCALL Sound_FreeSample(Sound_Sample *sample);
	-
	-
	-/**
	- * \fn int Sound_SetBufferSize(Sound_Sample *sample, Uint32 new_size)
	- * \brief Change the current buffer size for a sample.
	- *
	- * If the buffer size could be changed, then the sample->buffer and
	- * sample->buffer_size fields will reflect that. If they could not be
	- * changed, then your original sample state is preserved. If the buffer is
	- * shrinking, the data at the end of buffer is truncated. If the buffer is
	- * growing, the contents of the new space at the end is undefined until you
	- * decode more into it or initialize it yourself.
	- *
	- * The buffer size specified must be a multiple of the size of a single
	- * sample point. So, if you want 16-bit, stereo samples, then your sample
	- * point size is (2 channels * 16 bits), or 32 bits per sample, which is four
	- * bytes. In such a case, you could specify 128 or 132 bytes for a buffer,
	- * but not 129, 130, or 131 (although in reality, you'll want to specify a
	- * MUCH larger buffer).
	- *
	- * \param sample The Sound_Sample whose buffer to modify.
	- * \param new_size The desired size, in bytes, of the new buffer.
	- * \return non-zero if buffer size changed, zero on failure.
	- *
	- * \sa Sound_Decode
	- * \sa Sound_DecodeAll
	- */
	-SNDDECLSPEC int SDLCALL Sound_SetBufferSize(Sound_Sample *sample,
	- Uint32 new_size);
	-
	-
	-/**
	- * \fn Uint32 Sound_Decode(Sound_Sample *sample)
	- * \brief Decode more of the sound data in a Sound_Sample.
	- *
	- * It will decode at most sample->buffer_size bytes into sample->buffer in the
	- * desired format, and return the number of decoded bytes.
	- * If sample->buffer_size bytes could not be decoded, then please refer to
	- * sample->flags to determine if this was an end-of-stream or error condition.
	- *
	- * \param sample Do more decoding to this Sound_Sample.
	- * \return number of bytes decoded into sample->buffer. If it is less than
	- * sample->buffer_size, then you should check sample->flags to see
	- * what the current state of the sample is (EOF, error, read again).
	- *
	- * \sa Sound_DecodeAll
	- * \sa Sound_SetBufferSize
	- * \sa Sound_Seek
	- * \sa Sound_Rewind
	- */
	-SNDDECLSPEC Uint32 SDLCALL Sound_Decode(Sound_Sample *sample);
	-
	-
	-/**
	- * \fn Uint32 Sound_DecodeAll(Sound_Sample *sample)
	- * \brief Decode the remainder of the sound data in a Sound_Sample.
	- *
	- * This will dynamically allocate memory for the ENTIRE remaining sample.
	- * sample->buffer_size and sample->buffer will be updated to reflect the
	- * new buffer. Please refer to sample->flags to determine if the decoding
	- * finished due to an End-of-stream or error condition.
	- *
	- * Be aware that sound data can take a large amount of memory, and that
	- * this function may block for quite awhile while processing. Also note
	- * that a streaming source (for example, from a SDL_RWops that is getting
	- * fed from an Internet radio feed that doesn't end) may fill all available
	- * memory before giving up...be sure to use this on finite sound sources
	- * only!
	- *
	- * When decoding the sample in its entirety, the work is done one buffer at a
	- * time. That is, sound is decoded in sample->buffer_size blocks, and
	- * appended to a continually-growing buffer until the decoding completes.
	- * That means that this function will need enough RAM to hold approximately
	- * sample->buffer_size bytes plus the complete decoded sample at most. The
	- * larger your buffer size, the less overhead this function needs, but beware
	- * the possibility of paging to disk. Best to make this user-configurable if
	- * the sample isn't specific and small.
	- *
	- * \param sample Do all decoding for this Sound_Sample.
	- * \return number of bytes decoded into sample->buffer. You should check
	- * sample->flags to see what the current state of the sample is
	- * (EOF, error, read again).
	- *
	- * \sa Sound_Decode
	- * \sa Sound_SetBufferSize
	- */
	-SNDDECLSPEC Uint32 SDLCALL Sound_DecodeAll(Sound_Sample *sample);
	-
	-
	-/**
	- * \fn int Sound_Rewind(Sound_Sample *sample)
	- * \brief Rewind a sample to the start.
	- *
	- * Restart a sample at the start of its waveform data, as if newly
	- * created with Sound_NewSample(). If successful, the next call to
	- * Sound_Decode[All]() will give audio data from the earliest point
	- * in the stream.
	- *
	- * Beware that this function will fail if the SDL_RWops that feeds the
	- * decoder can not be rewound via it's seek method, but this can
	- * theoretically be avoided by wrapping it in some sort of buffering
	- * SDL_RWops.
	- *
	- * This function should ONLY fail if the RWops is not seekable, or
	- * SDL_sound is not initialized. Both can be controlled by the application,
	- * and thus, it is up to the developer's paranoia to dictate whether this
	- * function's return value need be checked at all.
	- *
	- * If this function fails, the state of the sample is undefined, but it
	- * is still safe to call Sound_FreeSample() to dispose of it.
	- *
	- * On success, ERROR, EOF, and EAGAIN are cleared from sample->flags. The
	- * ERROR flag is set on error.
	- *
	- * \param sample The Sound_Sample to rewind.
	- * \return nonzero on success, zero on error. Specifics of the
	- * error can be gleaned from Sound_GetError().
	- *
	- * \sa Sound_Seek
	- */
	-SNDDECLSPEC int SDLCALL Sound_Rewind(Sound_Sample *sample);
	-
	-
	-/**
	- * \fn int Sound_Seek(Sound_Sample *sample, Uint32 ms)
	- * \brief Seek to a different point in a sample.
	- *
	- * Reposition a sample's stream. If successful, the next call to
	- * Sound_Decode[All]() will give audio data from the offset you
	- * specified.
	- *
	- * The offset is specified in milliseconds from the start of the
	- * sample.
	- *
	- * Beware that this function can fail for several reasons. If the
	- * SDL_RWops that feeds the decoder can not seek, this call will almost
	- * certainly fail, but this can theoretically be avoided by wrapping it
	- * in some sort of buffering SDL_RWops. Some decoders can never seek,
	- * others can only seek with certain files. The decoders will set a flag
	- * in the sample at creation time to help you determine this.
	- *
	- * You should check sample->flags & SOUND_SAMPLEFLAG_CANSEEK
	- * before attempting. Sound_Seek() reports failure immediately if this
	- * flag isn't set. This function can still fail for other reasons if the
	- * flag is set.
	- *
	- * This function can be emulated in the application with Sound_Rewind()
	- * and predecoding a specific amount of the sample, but this can be
	- * extremely inefficient. Sound_Seek() accelerates the seek on a
	- * with decoder-specific code.
	- *
	- * If this function fails, the sample should continue to function as if
	- * this call was never made. If there was an unrecoverable error,
	- * sample->flags & SOUND_SAMPLEFLAG_ERROR will be set, which you regular
	- * decoding loop can pick up.
	- *
	- * On success, ERROR, EOF, and EAGAIN are cleared from sample->flags.
	- *
	- * \param sample The Sound_Sample to seek.
	- * \param ms The new position, in milliseconds from start of sample.
	- * \return nonzero on success, zero on error. Specifics of the
	- * error can be gleaned from Sound_GetError().
	- *
	- * \sa Sound_Rewind
	- */
	-SNDDECLSPEC int SDLCALL Sound_Seek(Sound_Sample *sample, Uint32 ms);
	-
	-#ifdef __cplusplus
	-}
	-#endif
	-
	-#endif /* !defined _INCLUDE_SDL_SOUND_H_ */
	-
	-/* end of SDL_sound.h ... */
	-
	diff --git a/util/sdl/sound/SDL_sound_internal.h b/util/sdl/sound/SDL_sound_internal.h
	deleted file mode 100644
	index d467fc8d..00000000
	--- a/util/sdl/sound/SDL_sound_internal.h
	+++ /dev/null
	@@ -1,326 +0,0 @@
	-/*
	- * SDL_sound -- An abstract sound format decoding API.
	- * Copyright (C) 2001 Ryan C. Gordon.
	- *
	- * This library is free software; you can redistribute it and/or
	- * modify it under the terms of the GNU Lesser General Public
	- * License as published by the Free Software Foundation; either
	- * version 2.1 of the License, or (at your option) any later version.
	- *
	- * This library is distributed in the hope that it will be useful,
	- * but WITHOUT ANY WARRANTY; without even the implied warranty of
	- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	- * Lesser General Public License for more details.
	- *
	- * You should have received a copy of the GNU Lesser General Public
	- * License along with this library; if not, write to the Free Software
	- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
	- */
	-
	-/*
	- * Internal function/structure declaration. Do NOT include in your
	- * application.
	- *
	- * Please see the file COPYING in the source's root directory.
	- *
	- * This file written by Ryan C. Gordon. (icculus@icculus.org)
	- */
	-
	-#ifndef _INCLUDE_SDL_SOUND_INTERNAL_H_
	-#define _INCLUDE_SDL_SOUND_INTERNAL_H_
	-
	-#ifndef __SDL_SOUND_INTERNAL__
	-#error Do not include this header from your applications.
	-#endif
	-
	-#include "SDL.h"
	-
	-/* SDL 1.2.4 defines this, but better safe than sorry. */
	-#if (!defined(__inline__))
	-# define __inline__
	-#endif
	-
	-#if (defined DEBUG_CHATTER)
	-#define SNDDBG(x) printf x
	-#else
	-#define SNDDBG(x)
	-#endif
	-
	-#if HAVE_ASSERT_H
	-# include <assert.h>
	-#endif
	-
	-#ifdef _WIN32_WCE
	- extern char strrchr(const char s, int c);
	-# ifdef NDEBUG
	-# define assert(x)
	-# else
	-# define assert(x) if(!x) { fprintf(stderr,"Assertion failed in %s, line %s.\n",__FILE__,__LINE__); fclose(stderr); fclose(stdout); exit(1); }
	-# endif
	-#endif
	-
	-
	-#if (!defined assert) /* if all else fails. */
	-# define assert(x)
	-#endif
	-
	-
	-typedef struct __SOUND_DECODERFUNCTIONS__
	-{
	- /* This is a block of info about your decoder. See SDL_sound.h. */
	- const Sound_DecoderInfo info;
	-
	- /*
	- * This is called during the Sound_Init() function. Use this to
	- * set up any global state that your decoder needs, such as
	- * initializing an external library, etc.
	- *
	- * Return non-zero if initialization is successful, zero if there's
	- * a fatal error. If this method fails, then this decoder is
	- * flagged as unavailable until SDL_sound() is shut down and
	- * reinitialized, in which case this method will be tried again.
	- *
	- * Note that the decoders quit() method won't be called if this
	- * method fails, so if you can't intialize, you'll have to clean
	- * up the half-initialized state in this method.
	- */
	- int (*init)(void);
	-
	- /*
	- * This is called during the Sound_Quit() function. Use this to
	- * clean up any global state that your decoder has used during its
	- * lifespan.
	- */
	- void (*quit)(void);
	-
	- /*
	- * Returns non-zero if (sample) has a valid fileformat that this
	- * driver can handle. Zero if this driver can NOT handle the data.
	- *
	- * Extension, which may be NULL, is just a hint as to the form of
	- * data that is being passed in. Most decoders should determine if
	- * they can handle the data by the data itself, but others, like
	- * the raw data handler, need this hint to know if they should
	- * accept the data in the first place.
	- *
	- * (sample)'s (opaque) field should be cast to a Sound_SampleInternal
	- * pointer:
	- *
	- * Sound_SampleInternal *internal;
	- * internal = (Sound_SampleInternal *) sample->opaque;
	- *
	- * Certain fields of sample will be filled in for the decoder before
	- * this call, and others should be filled in by the decoder. Some
	- * fields are offlimits, and should NOT be modified. The list:
	- *
	- * in Sound_SampleInternal section:
	- * Sound_Sample *next; (offlimits)
	- * Sound_Sample *prev; (offlimits)
	- * SDL_RWops *rw; (can use, but do NOT close it)
	- * const Sound_DecoderFunctions *funcs; (that's this structure)
	- * Sound_AudioCVT sdlcvt; (offlimits)
	- * void *buffer; (offlimits until read() method)
	- * Uint32 buffer_size; (offlimits until read() method)
	- * void *decoder_private; (read and write access)
	- *
	- * in rest of Sound_Sample:
	- * void *opaque; (this was internal section, above)
	- * const Sound_DecoderInfo *decoder; (read only)
	- * Sound_AudioInfo desired; (read only, usually not needed here)
	- * Sound_AudioInfo actual; (please fill this in)
	- * void *buffer; (offlimits)
	- * Uint32 buffer_size; (offlimits)
	- * Sound_SampleFlags flags; (set appropriately)
	- */
	- int (open)(Sound_Sample sample, const char *ext);
	-
	- /*
	- * Clean up. SDL_sound is done with this sample, so the decoder should
	- * clean up any resources it allocated. Anything that wasn't
	- * explicitly allocated by the decoder should be LEFT ALONE, since
	- * the higher-level SDL_sound layer will clean up its own mess.
	- */
	- void (close)(Sound_Sample sample);
	-
	- /*
	- * Get more data from (sample). The decoder should get a pointer to
	- * the internal structure...
	- *
	- * Sound_SampleInternal *internal;
	- * internal = (Sound_SampleInternal *) sample->opaque;
	- *
	- * ...and then start decoding. Fill in up to internal->buffer_size
	- * bytes of decoded sound in the space pointed to by
	- * internal->buffer. The encoded data is read in from internal->rw.
	- * Data should be decoded in the format specified during the
	- * decoder's open() method in the sample->actual field. The
	- * conversion to the desired format is done at a higher level.
	- *
	- * The return value is the number of bytes decoded into
	- * internal->buffer, which can be no more than internal->buffer_size,
	- * but can be less. If it is less, you should set a state flag:
	- *
	- * If there's just no more data (end of file, etc), then do:
	- * sample->flags \|= SOUND_SAMPLEFLAG_EOF;
	- *
	- * If there's an unrecoverable error, then do:
	- * __Sound_SetError(ERR_EXPLAIN_WHAT_WENT_WRONG);
	- * sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- *
	- * If there's more data, but you'd have to block for considerable
	- * amounts of time to get at it, or there's a recoverable error,
	- * then do:
	- * __Sound_SetError(ERR_EXPLAIN_WHAT_WENT_WRONG);
	- * sample->flags \|= SOUND_SAMPLEFLAG_EAGAIN;
	- *
	- * SDL_sound will not call your read() method for any samples with
	- * SOUND_SAMPLEFLAG_EOF or SOUND_SAMPLEFLAG_ERROR set. The
	- * SOUND_SAMPLEFLAG_EAGAIN flag is reset before each call to this
	- * method.
	- */
	- Uint32 (read)(Sound_Sample sample);
	-
	- /*
	- * Reset the decoding to the beginning of the stream. Nonzero on
	- * success, zero on failure.
	- *
	- * The purpose of this method is to allow for higher efficiency than
	- * an application could get by just recreating the sample externally;
	- * not only do they not have to reopen the RWops, reallocate buffers,
	- * and potentially pass the data through several rejecting decoders,
	- * but certain decoders will not have to recreate their existing
	- * state (search for metadata, etc) since they already know they
	- * have a valid audio stream with a given set of characteristics.
	- *
	- * The decoder is responsible for calling seek() on the associated
	- * SDL_RWops. A failing call to seek() should be the ONLY reason that
	- * this method should ever fail!
	- */
	- int (rewind)(Sound_Sample sample);
	-
	- /*
	- * Reposition the decoding to an arbitrary point. Nonzero on
	- * success, zero on failure.
	- *
	- * The purpose of this method is to allow for higher efficiency than
	- * an application could get by just rewinding the sample and
	- * decoding to a given point.
	- *
	- * The decoder is responsible for calling seek() on the associated
	- * SDL_RWops.
	- *
	- * If there is an error, try to recover so that the next read will
	- * continue as if nothing happened.
	- */
	- int (seek)(Sound_Sample sample, Uint32 ms);
	-} Sound_DecoderFunctions;
	-
	-
	-/* A structure to hold a set of audio conversion filters and buffers */
	-#if (defined SOUND_USE_ALTCVT)
	-#include "alt_audio_convert.h"
	-#else
	-typedef struct Sound_AudioCVT
	-{
	- int needed; /* Set to 1 if conversion possible */
	- Uint16 src_format; /* Source audio format */
	- Uint16 dst_format; /* Target audio format */
	- double rate_incr; /* Rate conversion increment */
	- Uint8 buf; / Buffer to hold entire audio data */
	- int len; /* Length of original audio buffer */
	- int len_cvt; /* Length of converted audio buffer */
	- int len_mult; /* buffer must be lenlen_mult big /
	- double len_ratio; /* Given len, final size is lenlen_ratio /
	- void (filters[20])(struct Sound_AudioCVT cvt, Uint16 *format);
	- int filter_index; /* Current audio conversion function */
	-} Sound_AudioCVT;
	-#endif
	-
	-extern SNDDECLSPEC int Sound_BuildAudioCVT(Sound_AudioCVT *cvt,
	- Uint16 src_format, Uint8 src_channels, Uint32 src_rate,
	- Uint16 dst_format, Uint8 dst_channels, Uint32 dst_rate,
	- Uint32 dst_size);
	-
	-extern SNDDECLSPEC int Sound_ConvertAudio(Sound_AudioCVT *cvt);
	-
	-
	-
	-typedef struct __SOUND_SAMPLEINTERNAL__
	-{
	- Sound_Sample *next;
	- Sound_Sample *prev;
	- SDL_RWops *rw;
	- const Sound_DecoderFunctions *funcs;
	- Sound_AudioCVT sdlcvt;
	- void *buffer;
	- Uint32 buffer_size;
	- void *decoder_private;
	-} Sound_SampleInternal;
	-
	-
	-/* error messages... */
	-#define ERR_IS_INITIALIZED "Already initialized"
	-#define ERR_NOT_INITIALIZED "Not initialized"
	-#define ERR_INVALID_ARGUMENT "Invalid argument"
	-#define ERR_OUT_OF_MEMORY "Out of memory"
	-#define ERR_NOT_SUPPORTED "Operation not supported"
	-#define ERR_UNSUPPORTED_FORMAT "Sound format unsupported"
	-#define ERR_NOT_A_HANDLE "Not a file handle"
	-#define ERR_NO_SUCH_FILE "No such file"
	-#define ERR_PAST_EOF "Past end of file"
	-#define ERR_IO_ERROR "I/O error"
	-#define ERR_COMPRESSION "(De)compression error"
	-#define ERR_PREV_ERROR "Previous decoding already caused an error"
	-#define ERR_PREV_EOF "Previous decoding already triggered EOF"
	-#define ERR_CANNOT_SEEK "Sample is not seekable"
	-
	-/*
	- * Call this to set the message returned by Sound_GetError().
	- * Please only use the ERR_* constants above, or add new constants to the
	- * above group, but I want these all in one place.
	- *
	- * Calling this with a NULL argument is a safe no-op.
	- */
	-void __Sound_SetError(const char *err);
	-
	-/*
	- * Call this to convert milliseconds to an actual byte position, based on
	- * audio data characteristics.
	- */
	-Uint32 __Sound_convertMsToBytePos(Sound_AudioInfo *info, Uint32 ms);
	-
	-/*
	- * Use this if you need a cross-platform stricmp().
	- */
	-int __Sound_strcasecmp(const char x, const char y);
	-
	-
	-/* These get used all over for lessening code clutter. */
	-#define BAIL_MACRO(e, r) { __Sound_SetError(e); return r; }
	-#define BAIL_IF_MACRO(c, e, r) if (c) { __Sound_SetError(e); return r; }
	-
	-
	-
	-
	-/--------------------------------------------------------------------------/
	-/--------------------------------------------------------------------------/
	-/------------ ----------------/
	-/------------ You MUST implement the following functions ----------------/
	-/------------ if porting to a new platform. ----------------/
	-/------------ (see platform/unix.c for an example) ----------------/
	-/------------ ----------------/
	-/--------------------------------------------------------------------------/
	-/--------------------------------------------------------------------------/
	-
	-
	-/* (None, right now.) */
	-
	-
	-#ifdef __cplusplus
	-extern "C" {
	-#endif
	-
	-#endif /* defined _INCLUDE_SDL_SOUND_INTERNAL_H_ */
	-
	-/* end of SDL_sound_internal.h ... */
	-
	diff --git a/util/sdl/sound/TODO b/util/sdl/sound/TODO
	deleted file mode 100644
	index 99a13c01..00000000
	--- a/util/sdl/sound/TODO
	+++ /dev/null
	@@ -1,29 +0,0 @@
	-More immediate:
	-- Fix the crappy rewind implementation in shn.c's SHN_rewind().
	-- Finish implementing seek() method in decoders, see below.
	-- Add a sdlsound-config script?
	-- Make sure we can build shared libs on Cygwin, BeOS, Mac OS X...
	-
	-Decoders still needing seek() method:
	- (If decoder can't seek, clean up the stub and report an error.)
	-- mikmod.c
	-- shn.c
	-- mpglib.c
	-- quicktime.c
	-
	-General stuff TODO:
	-- Hack on the experimental audio conversion routines.
	-- Handle compression and other chunks in WAV files.
	-- Handle compression and other chunks in AIFF-C files.
	-
	-Quicktime stuff that'd be cool, but isn't crucial:
	-- Integrate decoders/quicktime.c with build system (for OS X)?
	-- Make decoders/quicktime.c more robust.
	-- Make decoders/quicktime.c work on win32?
	-- There's no seek() method.
	-
	-Ongoing:
	-- look for "FIXME"s in the code.
	-
	-/* end of TODO ... */
	-
	diff --git a/util/sdl/sound/acinclude.m4 b/util/sdl/sound/acinclude.m4
	deleted file mode 100644
	index 99f8af36..00000000
	--- a/util/sdl/sound/acinclude.m4
	+++ /dev/null
	@@ -1,173 +0,0 @@
	-dnl AM_PATH_SDL([MINIMUM-VERSION, [ACTION-IF-FOUND [, ACTION-IF-NOT-FOUND]]])
	-dnl Test for SDL, and define SDL_CFLAGS and SDL_LIBS
	-dnl
	-AC_DEFUN([AM_PATH_SDL],
	-[dnl
	-dnl Get the cflags and libraries from the sdl-config script
	-dnl
	-AC_ARG_WITH(sdl-prefix,[ --with-sdl-prefix=PFX Prefix where SDL is installed (optional)],
	- sdl_prefix="$withval", sdl_prefix="")
	-AC_ARG_WITH(sdl-exec-prefix,[ --with-sdl-exec-prefix=PFX Exec prefix where SDL is installed (optional)],
	- sdl_exec_prefix="$withval", sdl_exec_prefix="")
	-AC_ARG_ENABLE(sdltest, [ --disable-sdltest Do not try to compile and run a test SDL program],
	- , enable_sdltest=yes)
	-
	- if test x$sdl_exec_prefix != x ; then
	- sdl_args="$sdl_args --exec-prefix=$sdl_exec_prefix"
	- if test x${SDL_CONFIG+set} != xset ; then
	- SDL_CONFIG=$sdl_exec_prefix/bin/sdl-config
	- fi
	- fi
	- if test x$sdl_prefix != x ; then
	- sdl_args="$sdl_args --prefix=$sdl_prefix"
	- if test x${SDL_CONFIG+set} != xset ; then
	- SDL_CONFIG=$sdl_prefix/bin/sdl-config
	- fi
	- fi
	-
	- AC_REQUIRE([AC_CANONICAL_TARGET])
	- PATH="$prefix/bin:$prefix/usr/bin:$PATH"
	- AC_PATH_PROG(SDL_CONFIG, sdl-config, no, [$PATH])
	- min_sdl_version=ifelse([$1], ,0.11.0,$1)
	- AC_MSG_CHECKING(for SDL - version >= $min_sdl_version)
	- no_sdl=""
	- if test "$SDL_CONFIG" = "no" ; then
	- no_sdl=yes
	- else
	- SDL_CFLAGS=`$SDL_CONFIG $sdlconf_args --cflags`
	- SDL_LIBS=`$SDL_CONFIG $sdlconf_args --libs`
	-
	- sdl_major_version=`$SDL_CONFIG $sdl_args --version \| \
	- sed 's/$[[0-9]]$.$[[0-9]]$.$[[0-9]]*$/\1/'`
	- sdl_minor_version=`$SDL_CONFIG $sdl_args --version \| \
	- sed 's/$[[0-9]]$.$[[0-9]]$.$[[0-9]]*$/\2/'`
	- sdl_micro_version=`$SDL_CONFIG $sdl_config_args --version \| \
	- sed 's/$[[0-9]]$.$[[0-9]]$.$[[0-9]]*$/\3/'`
	- if test "x$enable_sdltest" = "xyes" ; then
	- ac_save_CFLAGS="$CFLAGS"
	- ac_save_CXXFLAGS="$CXXFLAGS"
	- ac_save_LIBS="$LIBS"
	- CFLAGS="$CFLAGS $SDL_CFLAGS"
	- CXXFLAGS="$CXXFLAGS $SDL_CFLAGS"
	- LIBS="$LIBS $SDL_LIBS"
	-dnl
	-dnl Now check if the installed SDL is sufficiently new. (Also sanity
	-dnl checks the results of sdl-config to some extent
	-dnl
	- rm -f conf.sdltest
	- AC_TRY_RUN([
	-#include <stdio.h>
	-#include <stdlib.h>
	-#include <string.h>
	-#include "SDL.h"
	-
	-char*
	-my_strdup (char *str)
	-{
	- char *new_str;
	-
	- if (str)
	- {
	- new_str = (char )malloc ((strlen (str) + 1) sizeof(char));
	- strcpy (new_str, str);
	- }
	- else
	- new_str = NULL;
	-
	- return new_str;
	-}
	-
	-int main (int argc, char *argv[])
	-{
	- int major, minor, micro;
	- char *tmp_version;
	-
	- /* This hangs on some systems (?)
	- system ("touch conf.sdltest");
	- */
	- { FILE *fp = fopen("conf.sdltest", "a"); if ( fp ) fclose(fp); }
	-
	- /* HP/UX 9 (%@#!) writes to sscanf strings */
	- tmp_version = my_strdup("$min_sdl_version");
	- if (sscanf(tmp_version, "%d.%d.%d", &major, &minor, &micro) != 3) {
	- printf("%s, bad version string\n", "$min_sdl_version");
	- exit(1);
	- }
	-
	- if (($sdl_major_version > major) \|\|
	- (($sdl_major_version == major) && ($sdl_minor_version > minor)) \|\|
	- (($sdl_major_version == major) && ($sdl_minor_version == minor) && ($sdl_micro_version >= micro)))
	- {
	- return 0;
	- }
	- else
	- {
	- printf("\n*** 'sdl-config --version' returned %d.%d.%d, but the minimum version\n", $sdl_major_version, $sdl_minor_version, $sdl_micro_version);
	- printf("*** of SDL required is %d.%d.%d. If sdl-config is correct, then it is\n", major, minor, micro);
	- printf("*** best to upgrade to the required version.\n");
	- printf("*** If sdl-config was wrong, set the environment variable SDL_CONFIG\n");
	- printf("*** to point to the correct copy of sdl-config, and remove the file\n");
	- printf("*** config.cache before re-running configure\n");
	- return 1;
	- }
	-}
	-
	-],, no_sdl=yes,[echo $ac_n "cross compiling; assumed OK... $ac_c"])
	- CFLAGS="$ac_save_CFLAGS"
	- LIBS="$ac_save_LIBS"
	- fi
	- fi
	- if test "x$no_sdl" = x ; then
	- AC_MSG_RESULT(yes)
	- ifelse([$2], , :, [$2])
	- else
	- AC_MSG_RESULT(no)
	- if test "$SDL_CONFIG" = "no" ; then
	- echo "*** The sdl-config script installed by SDL could not be found"
	- echo "*** If SDL was installed in PREFIX, make sure PREFIX/bin is in"
	- echo "*** your path, or set the SDL_CONFIG environment variable to the"
	- echo "*** full path to sdl-config."
	- else
	- if test -f conf.sdltest ; then
	- :
	- else
	- echo "*** Could not run SDL test program, checking why..."
	- CFLAGS="$CFLAGS $SDL_CFLAGS"
	- CXXFLAGS="$CXXFLAGS $SDL_CFLAGS"
	- LIBS="$LIBS $SDL_LIBS"
	- AC_TRY_LINK([
	-#include <stdio.h>
	-#include "SDL.h"
	-
	-int main(int argc, char *argv[])
	-{ return 0; }
	-#undef main
	-#define main K_and_R_C_main
	-], [ return 0; ],
	- [ echo "*** The test program compiled, but did not run. This usually means"
	- echo "*** that the run-time linker is not finding SDL or finding the wrong"
	- echo "*** version of SDL. If it is not finding SDL, you'll need to set your"
	- echo "*** LD_LIBRARY_PATH environment variable, or edit /etc/ld.so.conf to point"
	- echo "*** to the installed location Also, make sure you have run ldconfig if that"
	- echo "*** is required on your system"
	- echo "***"
	- echo "*** If you have an old version installed, it is best to remove it, although"
	- echo "*** you may also be able to get things to work by modifying LD_LIBRARY_PATH"],
	- [ echo "*** The test program failed to compile or link. See the file config.log for the"
	- echo "*** exact error that occured. This usually means SDL was incorrectly installed"
	- echo "*** or that you have moved SDL since it was installed. In the latter case, you"
	- echo "*** may want to edit the sdl-config script: $SDL_CONFIG" ])
	- CFLAGS="$ac_save_CFLAGS"
	- CXXFLAGS="$ac_save_CXXFLAGS"
	- LIBS="$ac_save_LIBS"
	- fi
	- fi
	- SDL_CFLAGS=""
	- SDL_CXXFLAGS=""
	- SDL_LIBS=""
	- ifelse([$3], , :, [$3])
	- fi
	- AC_SUBST(SDL_CFLAGS)
	- AC_SUBST(SDL_LIBS)
	- rm -f conf.sdltest
	-])
	diff --git a/util/sdl/sound/alt_audio_convert.c b/util/sdl/sound/alt_audio_convert.c
	deleted file mode 100644
	index de9b8fd2..00000000
	--- a/util/sdl/sound/alt_audio_convert.c
	+++ /dev/null
	@@ -1,1057 +0,0 @@
	-/*
	- * Extended Audio Converter for SDL (Simple DirectMedia Layer)
	- * Copyright (C) 2002 Frank Ranostaj
	- * Institute of Applied Physik
	- * Johann Wolfgang Goethe-Universität
	- * Frankfurt am Main, Germany
	- *
	- * This library is free software; you can redistribute it and/or
	- * modify it under the terms of the GNU Library General Public
	- * License as published by the Free Software Foundation; either
	- * version 2 of the License, or (at your option) any later version.
	- *
	- * This library is distributed in the hope that it will be useful,
	- * but WITHOUT ANY WARRANTY; without even the implied warranty of
	- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	- * Library General Public License for more details.
	- *
	- * You should have received a copy of the GNU Library General Public
	- * License along with this library; if not, write to the Free
	- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
	- *
	- * Frank Ranostaj
	- * ranostaj@stud.uni-frankfurt.de
	- *
	- * (This code blatantly abducted for SDL_sound. Thanks, Frank! --ryan.)
	- */
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#if SOUND_USE_ALTCVT
	-
	-#include "alt_audio_convert.h"
	-#include <math.h>
	-
	-/* just to make sure this is defined... */
	-
	-#ifndef min
	-#define min(x, y) ( ((x) < (y)) ? (x) : (y) )
	-#endif
	-
	-#ifndef max
	-#define max(x, y) ( ((x) > (y)) ? (x) : (y) )
	-#endif
	-
	-#ifndef abs
	-#define abs(x) ( ((x) > (0)) ? (x) : -(x) )
	-#endif
	-
	-
	-/* some macros for "parsing" format */
	-
	-#define IS_8BIT(x) ((x).format & 0x0008)
	-#define IS_16BIT(x) ((x).format & 0x0010)
	-#define IS_FLOAT(x) ((x).format & 0x0020)
	-#define IS_SIGNED(x) ((x).format & 0x8000)
	-#define IS_SYSENDIAN(x) ((~AUDIO_U16SYS ^ (x).format) & 0x1000)
	-#define SDL_MSB_POSITION_IN_SHORT ((0x1000 & AUDIO_U16SYS)>>12)
	-
	-
	-/-------------------------------------------------------------------------/
	-/* the purpose of the RateConverterBuffer is to provide a continous storage
	- for head and tail of the (sample)-buffer. This allows a simple and
	- perfomant implemantation of the sample rate converters. Depending of the
	- operation mode, two layouts for the RateConverterBuffer.inbuffer are
	- possible:
	-
	- in the Loop Mode:
	- ... T-4 T-3 T-2 T-1 H+0 H+1 H+2 H+3 H+4 ...
	- \|
	- linp, finp
	-
	- in the Single Mode (non Loop):
	- ... T-4 T-3 T-2 T-1 0 0 0 ... 0 0 0 H+0 H+1 H+2 H+3 H+4 ...
	- \| \|
	- linp finp
	-
	- The RateConverterBuffer allows an accurate attack and decay of the
	- filters in the rate Converters.
	-
	- The pointer finp are actually shifted against the depicted position so
	- that on the first invocation of the rate converter the input of the
	- filter is nearly complete in the zero region, only one input value is
	- used. After the calculation of the first output value, the pointer are
	- incremented or decremented depending on down or up conversion and the
	- first two input value are taken into account. This procedure repeats
	- until the filter has processed all zeroes. The distance of the pointer
	- movement is stored in flength, always positive.
	-
	- Further a pointer cinp to the sample buffer itself is stored. The pointer
	- to the sample buffer is shifted too, so that on the first use of this
	- pointer the filter is complete in the sample buffer. The pointer moves
	- over the sample buffer until it reaches the other end. The distance of
	- the movement is stored in clength.
	-
	- Finally the decay of the filter is done by linp and llength like finp,
	- flength, but in reverse order.
	-
	- buffer denotes the start or the end of the output buffer, depending
	- on direction of the rate conversion.
	-
	- All pointer and length referring the buffer as Sint16. All length
	- are refering to the input buffer */
	-
	-typedef struct
	-{
	- Sint16 inbuffer[24*_fsize];
	- Sint16 finp, cinp, *linp;
	- int flength, clength, llength;
	- Sint16 *buffer;
	- VarFilter *filter;
	-} RateConverterBuffer;
	-
	-typedef struct
	-{
	- Sint16 carry;
	- Sint16 pos;
	-} RateAux;
	-
	-
	-/* Mono (1 channel ) */
	-#define Suffix(x) x##1
	-#include "filter_templates.h"
	-#undef Suffix
	-
	-/* Stereo (2 channel ) */
	-#define Suffix(x) x##2
	-#include "filter_templates.h"
	-#undef Suffix
	-
	-
	-/-------------------------------------------------------------------------/
	-int Sound_estimateBufferSize( Sound_AudioCVT *Data, int size )
	-{
	- size *= Data->len_mult;
	- size += Data->len_add;
	- return ( size + 3 ) & -4; /* force Size in multipels of 4 Byte */
	-}
	-
	-/-------------------------------------------------------------------------/
	-int Sound_AltConvertAudio( Sound_AudioCVT *Data,
	- Uint8* buffer, int length, int mode )
	-{
	- AdapterC Temp;
	- int i;
	-
	- /* Make sure there's a converter */
	- if( Data == NULL ) {
	- SDL_SetError("No converter given");
	- return(-1);
	- }
	-
	- /* Make sure there's data to convert */
	- if( buffer == NULL ) {
	- SDL_SetError("No buffer allocated for conversion");
	- return(-1);
	- }
	-
	- if( length < 0 ) {
	- SDL_SetError("Lenght < 0");
	- return(-1);
	- }
	-
	- /* Set up the conversion and go! */
	- Temp.buffer = buffer;
	- Temp.mode = mode;
	- Temp.filter = &Data->filter;
	-
	- for( i = 0; Data->adapter[i] != NULL; i++ )
	- length = (*Data->adapter[i])( Temp, length);
	-
	- return length;
	-}
	-
	-int Sound_ConvertAudio( Sound_AudioCVT *Data )
	-{
	- int length;
	- /* !!! FIXME: Try the looping stuff under certain circumstances? --ryan. */
	- length = Sound_AltConvertAudio( Data, Data->buf, Data->len, 0 );
	- Data->len_cvt = length;
	- return length;
	-}
	-
	-/-------------------------------------------------------------------------/
	-static int expand8BitTo16BitSys( AdapterC Data, int length )
	-{
	- int i;
	- Uint8* inp = Data.buffer - 1;
	- Uint16* buffer = (Uint16*)Data.buffer - 1;
	- for( i = length + 1; --i; )
	- buffer[i] = inp[i]<<8;
	- return 2*length;
	-}
	-
	-static int expand8BitTo16BitWrong( AdapterC Data, int length )
	-{
	- int i;
	- Uint8* inp = Data.buffer - 1;
	- Uint16* buffer = (Uint16*)Data.buffer - 1;
	- for( i = length + 1; --i; )
	- buffer[i] = inp[i];
	- return 2*length;
	-}
	-
	-/-------------------------------------------------------------------------/
	-static int expand16BitToFloat( AdapterC Data, int length )
	-{
	- int i;
	- Sint16* inp = (Sint16*)Data.buffer - 1;
	- float* buffer = (float*)Data.buffer - 1;
	- for( i = length>>1 + 1; --i; )
	- buffer[i] = inp[i]*(1./32767);
	- return 2*length;
	-}
	-
	-/-------------------------------------------------------------------------/
	-static int swapBytes( AdapterC Data, int length )
	-{
	- /*
	- * !!! FIXME !!!
	- *
	- *
	- * Use the faster SDL-Macros to swap
	- * - Frank
	- */
	-
	- int i;
	- Uint16 a,b;
	- Uint16* buffer = (Uint16*) Data.buffer - 1;
	- for( i = length>>1 + 1; --i; )
	- {
	- a = b = buffer[i];
	- buffer[i] = ( a << 8 ) \| ( b >> 8 );
	- }
	- return length;
	-}
	-
	-/-------------------------------------------------------------------------/
	-static int cutFloatTo16Bit( AdapterC Data, int length )
	-{
	- int i;
	- float* inp = (float*) Data.buffer;
	- Sint16* buffer = (Sint16*) Data.buffer;
	- length>>=2;
	- for( i = 0; i < length; i++ )
	- {
	- if( inp[i] > 1. )
	- buffer[i] = 32767;
	- else if( inp[i] < -1. )
	- buffer[i] = -32768;
	- else
	- buffer[i] = 32767 * inp[i];
	- }
	- return 2*length;
	-}
	-
	-/-------------------------------------------------------------------------/
	-static int cut16BitTo8Bit( AdapterC Data, int length, int off )
	-{
	- int i;
	- Uint8* inp = Data.buffer + off;
	- Uint8* buffer = Data.buffer;
	- length >>= 1;
	- for( i = 0; i < length; i++ )
	- buffer[i] = inp[2*i];
	- return length;
	-}
	-
	-static int cut16BitSysTo8Bit( AdapterC Data, int length )
	-{
	- return cut16BitTo8Bit( Data, length, SDL_MSB_POSITION_IN_SHORT );
	-}
	-
	-static int cut16BitWrongTo8Bit( AdapterC Data, int length )
	-{
	- return cut16BitTo8Bit( Data, length, 1-SDL_MSB_POSITION_IN_SHORT );
	-}
	-
	-/-------------------------------------------------------------------------/
	-/* poor mans mmx :-) */
	-static int changeSigned( AdapterC Data, int length, Uint32 XOR )
	-{
	- int i;
	- Uint32* buffer = (Uint32*) Data.buffer - 1;
	- for( i = ( length + 7 ) >> 2; --i; )
	- buffer[i] ^= XOR;
	- return length;
	-}
	-
	-static int changeSigned16BitSys( AdapterC Data, int length )
	-{
	- return changeSigned( Data, length, 0x80008000 );
	-}
	-
	-static int changeSigned16BitWrong( AdapterC Data, int length )
	-{
	- return changeSigned( Data, length, 0x00800080 );
	-}
	-
	-static int changeSigned8Bit( AdapterC Data, int length )
	-{
	- return changeSigned( Data, length, 0x80808080 );
	-}
	-
	-/-------------------------------------------------------------------------/
	-static int convertStereoToMonoS16Bit( AdapterC Data, int length )
	-{
	- int i;
	- Sint16* buffer = (Sint16*) Data.buffer;
	- Sint16* src = (Sint16*) Data.buffer;
	- length >>= 2;
	- for( i = 0; i < length; i++, src+=2 )
	- buffer[i] = ((int) src[0] + src[1] ) >> 1;
	- return 2*length;
	-}
	-
	-static int convertStereoToMonoU16Bit( AdapterC Data, int length )
	-{
	- int i;
	- Uint16* buffer = (Uint16*) Data.buffer;
	- Uint16* src = (Uint16*) Data.buffer;
	- length >>= 2;
	- for( i = 0; i < length; i++, src+=2 )
	- buffer[i] = ((int) src[0] + src[1] ) >> 1;
	- return 2*length;
	-}
	-
	-static int convertStereoToMonoS8Bit( AdapterC Data, int length )
	-{
	- int i;
	- Sint8* buffer = (Sint8*) Data.buffer;
	- Sint8* src = (Sint8*) Data.buffer;
	- length >>= 1;
	- for( i = 0; i < length; i++, src+=2 )
	- buffer[i] = ((int) src[0] + src[1] ) >> 1;
	- return length;
	-}
	-
	-static int convertStereoToMonoU8Bit( AdapterC Data, int length )
	-{
	- int i;
	- Uint8* buffer = (Uint8*) Data.buffer;
	- Uint8* src = (Uint8*) Data.buffer;
	- length >>= 1;
	- for( i = 0; i < length; i++, src+=2 )
	- buffer[i] = ((int) src[0] + src[1] ) >> 1;
	- return length;
	-}
	-
	-/-------------------------------------------------------------------------/
	-static int convertMonoToStereo16Bit( AdapterC Data, int length )
	-{
	- int i;
	- Uint16* buffer;
	- Uint16* dst;
	-
	- length >>=1;
	- buffer = (Uint16*)Data.buffer - 1;
	- dst = (Uint16)Data.buffer + 2length - 2;
	- for( i = length + 1; --i; dst-=2 )
	- dst[0] = dst[1] = buffer[i];
	- return 4*length;
	-}
	-
	-static int convertMonoToStereo8Bit( AdapterC Data, int length )
	-{
	- int i;
	- Uint8* buffer = Data.buffer - 1;
	- Uint8* dst = Data.buffer + 2*length - 2;
	- for( i = length + 1; --i; dst-=2 )
	- dst[0] = dst[1] = buffer[i];
	- return 2*length;
	-}
	-
	-/-------------------------------------------------------------------------/
	-static int minus5dB( AdapterC Data, int length )
	-{
	- int i;
	- Sint16* buffer = (Sint16*) Data.buffer;
	- for(i = length>>1 + 1; --i; )
	- buffer[i] = (38084 * (int)buffer[i]) >> 16;
	- return length;
	-}
	-
	-/-------------------------------------------------------------------------/
	-const Fraction Half = {1, 2};
	-const Fraction Double = {2, 1};
	-const Fraction One = {1, 1};
	-
	-
	-static void initStraigthBuffer( RateConverterBuffer *rcb,
	- int length, Fraction r )
	-{
	- int i, size, minsize;
	- size = 8 * _fsize;
	- minsize = min( size, length );
	-
	- for( i = 0; i < minsize; i++ )
	- {
	- rcb->inbuffer[i] = rcb->buffer[length-size+i];
	- rcb->inbuffer[i+size] = 0;
	- rcb->inbuffer[i+2*size] = rcb->buffer[i];
	- }
	- for( ; i < size; i++ )
	- {
	- rcb->inbuffer[i] = 0;
	- rcb->inbuffer[i+size] = 0;
	- rcb->inbuffer[i+2*size] = 0;
	- }
	-
	- length = max( length, size );
	- rcb->flength = rcb->llength = size;
	- rcb->clength = length - size;
	-
	- if( r.numerator < r.denominator )
	- {
	- rcb->finp = rcb->inbuffer + 5*size/2;
	- rcb->cinp = rcb->buffer + length - size/2;
	- rcb->linp = rcb->inbuffer + 3*size/2;
	- rcb->buffer += ( 1 + r.denominator * ( length + size )
	- / r.numerator ) & -2;
	- }
	- else
	- {
	- rcb->finp = rcb->inbuffer + size/2;
	- rcb->cinp = rcb->buffer + size/2;
	- rcb->linp = rcb->inbuffer + 3*size/2;
	- }
	-}
	-
	-static void initLoopBuffer( RateConverterBuffer *rcb,
	- int length, Fraction r )
	-{
	- /* !!!FIXME: modulo length, take scale into account,
	- check against the Straight part -frank */
	- int i, size;
	- size = 8 * _fsize;
	- for( i = 0; i < size; i++ )
	- {
	- rcb->inbuffer[i] = rcb->buffer[length-size+i];
	- rcb->inbuffer[i+size] = rcb->buffer[i];
	- }
	- rcb->finp = rcb->linp = rcb->inbuffer + size;
	- if( size < 0 )
	- rcb->buffer += r.numerator * ( length + 2 * size )
	- / r.denominator;
	-}
	-
	-static void initRateConverterBuffer( RateConverterBuffer *rcb,
	- AdapterC* Data, int length, Fraction ratio )
	-{
	- length >>= 1;
	- rcb->buffer = (Sint16*)( Data->buffer );
	- rcb->filter = Data->filter;
	-
	- if( Data->mode & SDL_SOUND_Loop )
	- initLoopBuffer( rcb, length, ratio );
	- else
	- initStraigthBuffer( rcb, length, ratio );
	-
	- fprintf( stderr, " finp: %8x length: %8x\n", rcb->finp, rcb->flength );
	- fprintf( stderr, " cinp: %8x length: %8x\n", rcb->cinp, rcb->clength );
	- fprintf( stderr, " linp: %8x length: %8x\n", rcb->linp, rcb->llength );
	-}
	-
	-static void nextRateConverterBuffer( RateConverterBuffer *rcb )
	-{
	- rcb->buffer++;
	- rcb->finp++;
	- rcb->cinp++;
	- rcb->linp++;
	-}
	-
	-typedef Sint16* (RateConverter)( Sint16, Sint16*, int,
	- VarFilter, RateAux );
	-
	-static Sint16* doRateConversion( RateConverterBuffer* rcb, RateConverter rc )
	-{
	- RateAux aux = {0,0};
	- Sint16 *outp = rcb->buffer;
	- VarFilter* filter = rcb->filter;
	-
	- outp = (*rc)( outp, rcb->finp, rcb->flength, filter, &aux );
	- fprintf( stderr, " outp: %8x aux.carry: %8x\n", outp, aux.carry );
	- outp = (*rc)( outp, rcb->cinp, rcb->clength, filter, &aux );
	- fprintf( stderr, " outp: %8x aux.carry: %8x\n", outp, aux.carry );
	- outp = (*rc)( outp, rcb->linp, rcb->llength, filter, &aux );
	- fprintf( stderr, " outp: %8x aux.carry: %8x\n", outp, aux.carry );
	- return outp;
	-}
	-
	-
	-/-------------------------------------------------------------------------/
	-static void clearSint16Buffer( Sint8* buffer, Sint16*r )
	-{
	- while( r >= (Sint16)buffer ) r-- = 0;
	-}
	-
	-/-------------------------------------------------------------------------/
	-static int doubleRateMono( AdapterC Data, int length )
	-{
	- Sint16* r;
	- RateConverterBuffer rcb;
	- initRateConverterBuffer( &rcb, &Data, length, Half );
	- r = 1 + doRateConversion( &rcb, doubleRate1 );
	- clearSint16Buffer( Data.buffer, r );
	- return 2 * ( rcb.buffer - (Sint16*)Data.buffer + 2 );
	-}
	-
	-static int doubleRateStereo( AdapterC Data, int length )
	-{
	- Sint16* r;
	- RateConverterBuffer rcb;
	- fprintf( stderr, "\n Buffer: %8x length: %8x\n", Data.buffer, length );
	- initRateConverterBuffer( &rcb, &Data, length, Half );
	- doRateConversion( &rcb, doubleRate2 );
	- nextRateConverterBuffer( &rcb );
	- r = 2 + doRateConversion( &rcb, doubleRate2 );
	- clearSint16Buffer( Data.buffer, r );
	- return 2 * ( rcb.buffer - (Sint16*)Data.buffer + 3 );
	-}
	-
	-/-------------------------------------------------------------------------/
	-static int halfRateMono( AdapterC Data, int length )
	-{
	- Sint16* r;
	- RateConverterBuffer rcb;
	- initRateConverterBuffer( &rcb, &Data, length, Double );
	- r = doRateConversion( &rcb, halfRate1 );
	- return 2 * ( r - (Sint16*)Data.buffer );
	-}
	-
	-static int halfRateStereo( AdapterC Data, int length )
	-{
	- Sint16* r;
	- RateConverterBuffer rcb;
	- initRateConverterBuffer( &rcb, &Data, length, Double );
	- doRateConversion( &rcb, halfRate2 );
	- nextRateConverterBuffer( &rcb );
	- r = doRateConversion( &rcb, halfRate2 );
	- return 2 * ( r - (Sint16*)Data.buffer );
	-}
	-
	-/-------------------------------------------------------------------------/
	-static int increaseRateMono( AdapterC Data, int length )
	-{
	- Sint16* r;
	- RateConverterBuffer rcb;
	- initRateConverterBuffer( &rcb, &Data, length, Data.filter->ratio );
	- r = doRateConversion( &rcb, increaseRate1 );
	- clearSint16Buffer( Data.buffer, r );
	- return 2 * ( rcb.buffer - (Sint16*)Data.buffer + 1 );
	-}
	-
	-static int increaseRateStereo( AdapterC Data, int length )
	-{
	- Sint16* r;
	- RateConverterBuffer rcb;
	- fprintf( stderr, "\n Buffer: %8x length: %8x\n", Data.buffer, length );
	- initRateConverterBuffer( &rcb, &Data, length, Data.filter->ratio );
	- doRateConversion( &rcb, increaseRate2 );
	- nextRateConverterBuffer( &rcb );
	- r = doRateConversion( &rcb, increaseRate2 );
	- clearSint16Buffer( Data.buffer, r );
	- return 2 * ( rcb.buffer - (Sint16*)Data.buffer + 1 );
	-}
	-
	-/-------------------------------------------------------------------------/
	-static int decreaseRateMono( AdapterC Data, int length )
	-{
	- Sint16* r;
	- RateConverterBuffer rcb;
	- initRateConverterBuffer( &rcb, &Data, length, Data.filter->ratio );
	- r = doRateConversion( &rcb, decreaseRate1 );
	- return 2 * ( r - (Sint16*)Data.buffer );
	-}
	-
	-static int decreaseRateStereo( AdapterC Data, int length )
	-{
	- Sint16* r;
	- RateConverterBuffer rcb;
	- initRateConverterBuffer( &rcb, &Data, length, Data.filter->ratio );
	- doRateConversion( &rcb, decreaseRate2 );
	- nextRateConverterBuffer( &rcb );
	- r = doRateConversion( &rcb, decreaseRate2 );
	- return 2 * ( r - (Sint16*)Data.buffer );
	-}
	-
	-/-------------------------------------------------------------------------/
	-/* gives a maximal error of 3% and typical less than 0.2% */
	-static Fraction findFraction( float Value )
	-{
	- const Sint8 frac[95]={
	- 2, -1, /* /1 */
	- 1, 3, -1, /* /2 */
	- 2, 4, 5, -1, /* /3 */
	- 3, 5, 7, -1, /* /4 */
	- 3, 4, 6, 7, 8, 9, -1, /* /5 */
	- 5, 7, 11, -1, /* /6 */
	- 4, 5, 6, 8, 9, 10, 11, 12, 13, -1, /* /7 */
	- 5, 7, 9, 11, 13, 15, -1, /* /8 */
	- 5, 7, 8, 10, 11, 13, 14, 16, -1, /* /9 */
	- 7, 9, 11, 13, -1, /* /10 */
	- 6, 7, 8, 9, 10, 12, 13, 14, 15, 16, -1, /* /11 */
	- 7, 11, 13, -1, /* /12 */
	- 7, 8, 9, 10, 11, 12, 14, 15, 16, -1, /* /13 */
	- 9, 11, 13, 15, -1, /* /14 */
	- 8, 11, 13, 14, 16, -1, /* /15 */
	- 9, 11, 13, 15 }; /* /16 */
	-
	-
	- Fraction Result = {0,0};
	- int i,num,den=1;
	-
	- float RelErr, BestErr = 0;
	- if( Value < 31/64. \|\| Value > 64/31. ) return Result;
	-
	- for( i = 0; i < SDL_TABLESIZE(frac); i++ )
	- {
	- num = frac[i];
	- if( num < 0 ) den++;
	- RelErr = Value * num / den;
	- RelErr = min( RelErr, 1/RelErr );
	- if( RelErr > BestErr )
	- {
	- BestErr = RelErr;
	- Result.denominator = den;
	- Result.numerator = num;
	- }
	- }
	- return Result;
	-}
	-
	-/-------------------------------------------------------------------------/
	-static float sinc( float x )
	-{
	- if( x > -1e-24 && x < 1e-24 ) return 1.;
	- else return sin(x)/x;
	-}
	-
	-static float calculateVarFilter( Sint16* dst,
	- float Ratio, float phase, float scale )
	-{
	- const Uint16 KaiserWindow7[]= {
	- 22930, 16292, 14648, 14288, 14470, 14945, 15608, 16404,
	- 17304, 18289, 19347, 20467, 21644, 22872, 24145, 25460,
	- 26812, 28198, 29612, 31052, 32513, 33991, 35482, 36983,
	- 38487, 39993, 41494, 42986, 44466, 45928, 47368, 48782,
	- 50165, 51513, 52821, 54086, 55302, 56466, 57575, 58624,
	- 59610, 60529, 61379, 62156, 62858, 63483, 64027, 64490,
	- 64870, 65165, 65375, 65498, 65535, 65484, 65347, 65124,
	- 64815, 64422, 63946, 63389, 62753, 62039, 61251, 60391 };
	- int i;
	- float w;
	- const float fg = -.018 + .5 * Ratio;
	- const float omega = 2 * M_PI * fg;
	- fprintf( stderr, " phase: %6g \n", phase );
	- phase += 63;
	- for( i = 0; i < 64; i++)
	- {
	- w = scale * ( KaiserWindow7[i] * ( i + 1 ));
	- dst[i] = w * sinc( omega * (i-phase) );
	- dst[127-i] = w * sinc( omega * (127-i-phase) );
	- }
	- fprintf( stderr, " center: %6d %6d \n", dst[63], dst[64] );
	- return fg;
	-}
	-
	-static Fraction setupVarFilter( Sound_AudioCVT *Data, float Ratio )
	-{
	- int pos,n,d, incr, phase = 0;
	- float Scale, rd, fg;
	- Fraction IRatio;
	- VarFilter* filter = &Data->filter;
	-
	- IRatio = findFraction( Ratio );
	-// Scale = Ratio < 1. ? 0.0364733 : 0.0211952;
	- Scale = 0.0084778;
	- Ratio = min( Ratio, 0.97 );
	-
	- filter->ratio = IRatio;
	- n = IRatio.numerator;
	- d = IRatio.denominator;
	- rd = 1. / d;
	-
	- fprintf( stderr, "Filter:\n" );
	-
	- for( pos = 0; pos < d; pos++ )
	- {
	- fg = calculateVarFilter( filter->c[pos], Ratio, phase*rd, Scale );
	- phase += n;
	- filter->incr[pos] = phase / d;
	- phase %= d;
	- }
	- fprintf( stderr, " fg: %6g\n\n", fg );
	-/* !!!FIXME: get rid of the inversion -Frank*/
	- IRatio.numerator = d;
	- IRatio.denominator = n;
	- return IRatio;
	-}
	-/-------------------------------------------------------------------------/
	-static void initAudioCVT( Sound_AudioCVT *Data )
	-{
	- Data->len_ratio = 1.;
	- Data->len_mult = 1;
	- Data->add = 0;
	- Data->len_add = 0;
	- Data->filter_index = 0;
	-}
	-
	-static void adjustSize( Sound_AudioCVT *Data, int add, Fraction f )
	-{
	- double ratio = f.numerator / (double) f.denominator;
	- Data->len_ratio *= ratio;
	- Data->len_mult = max( Data->len_mult, ceil(Data->len_ratio) );
	- Data->add = ratio * (Data->add + add);
	- Data->len_add = max( Data->len_add, ceil(Data->add) );
	-}
	-
	-static Adapter* addAdapter( Sound_AudioCVT *Data, Adapter a )
	-{
	- Data->adapter[Data->filter_index] = a;
	- return &Data->adapter[Data->filter_index++];
	-}
	-
	-static void addHAdapter( Sound_AudioCVT *Data, Adapter a )
	-{
	- adjustSize( Data, 0, Half );
	- addAdapter( Data, a );
	-}
	-
	-static void addDAdapter( Sound_AudioCVT *Data, Adapter a )
	-{
	- adjustSize( Data, 0, Double );
	- addAdapter( Data, a );
	-}
	-
	-
	-/-------------------------------------------------------------------------/
	-const Adapter doubleRate[2] = { doubleRateMono, doubleRateStereo };
	-const Adapter halfRate[2] = { halfRateMono, halfRateStereo };
	-const Adapter increaseRate[2] = { increaseRateMono, increaseRateStereo };
	-const Adapter decreaseRate[2] = { decreaseRateMono, decreaseRateStereo };
	-
	-static int createRateConverter( Sound_AudioCVT *Data,
	- int SrcRate, int DestRate, int channel )
	-{
	- const int c = channel - 1;
	- const int size = 16 * channel * _fsize;
	- Adapter* AdapterPos;
	- float Ratio = DestRate;
	- Fraction f;
	-
	- if( SrcRate < 1 \|\| SrcRate > 1<<18 \|\|
	- DestRate < 1 \|\| DestRate > 1<<18 ) return -1;
	- Ratio /= SrcRate;
	-
	- AdapterPos = addAdapter( Data, minus5dB );
	-
	- while( Ratio > 64./31.)
	- {
	- Ratio /= 2.;
	- addAdapter( Data, doubleRate[c] );
	- adjustSize( Data, size, Double );
	- }
	-
	- while( Ratio < 31./64. )
	- {
	- Ratio *= 2;
	- addAdapter( Data, halfRate[c] );
	- adjustSize( Data, size, Half );
	- }
	-
	- if( Ratio > 1. )
	- {
	- *AdapterPos = increaseRate[c];
	- f = setupVarFilter( Data, Ratio );
	- adjustSize( Data, size, f );
	- }
	- else
	- {
	- f = setupVarFilter( Data, Ratio );
	- addAdapter( Data, decreaseRate[c]);
	- adjustSize( Data, size, f );
	- }
	-
	- return 0;
	-}
	-
	-/-------------------------------------------------------------------------/
	-static void createFormatConverter16Bit(Sound_AudioCVT *Data,
	- SDL_AudioSpec src, SDL_AudioSpec dst )
	-{
	- if( src.channels == 2 && dst.channels == 1 )
	- {
	- if( !IS_SYSENDIAN(src) )
	- addAdapter( Data, swapBytes );
	-
	- if( IS_SIGNED(src) )
	- addHAdapter( Data, convertStereoToMonoS16Bit );
	- else
	- addHAdapter( Data, convertStereoToMonoU16Bit );
	-
	- if( !IS_SYSENDIAN(dst) )
	- addAdapter( Data, swapBytes );
	- }
	- else if( IS_SYSENDIAN(src) != IS_SYSENDIAN(dst) )
	- addAdapter( Data, swapBytes );
	-
	- if( IS_SIGNED(src) != IS_SIGNED(dst) )
	- {
	- if( IS_SYSENDIAN(dst) )
	- addAdapter( Data, changeSigned16BitSys );
	- else
	- addAdapter( Data, changeSigned16BitWrong );
	- }
	-
	- if( src.channels == 1 && dst.channels == 2 )
	- addDAdapter( Data, convertMonoToStereo16Bit );
	-}
	-
	-/-------------------------------------------------------------------------/
	-static void createFormatConverter8Bit(Sound_AudioCVT *Data,
	- SDL_AudioSpec src, SDL_AudioSpec dst )
	-{
	- if( IS_16BIT(src) )
	- {
	- if( IS_SYSENDIAN(src) )
	- addHAdapter( Data, cut16BitSysTo8Bit );
	- else
	- addHAdapter( Data, cut16BitWrongTo8Bit );
	- }
	-
	- if( src.channels == 2 && dst.channels == 1 )
	- {
	- if( IS_SIGNED(src) )
	- addHAdapter( Data, convertStereoToMonoS8Bit );
	- else
	- addHAdapter( Data, convertStereoToMonoU8Bit );
	- }
	-
	- if( IS_SIGNED(src) != IS_SIGNED(dst) )
	- addDAdapter( Data, changeSigned8Bit );
	-
	- if( src.channels == 1 && dst.channels == 2 )
	- addDAdapter( Data, convertMonoToStereo8Bit );
	-
	- if( !IS_8BIT(dst) )
	- {
	- if( IS_SYSENDIAN(dst) )
	- addDAdapter( Data, expand8BitTo16BitSys );
	- else
	- addDAdapter( Data, expand8BitTo16BitWrong );
	- }
	-}
	-
	-/-------------------------------------------------------------------------/
	-static void createFormatConverter(Sound_AudioCVT *Data,
	- SDL_AudioSpec src, SDL_AudioSpec dst )
	-{
	- if( IS_FLOAT(src) )
	- addHAdapter( Data, cutFloatTo16Bit );
	-
	- if( IS_8BIT(src) \|\| IS_8BIT(dst) )
	- createFormatConverter8Bit( Data, src, dst);
	- else
	- createFormatConverter16Bit( Data, src, dst);
	-
	- if( IS_FLOAT(dst) )
	- addDAdapter( Data, expand16BitToFloat );
	-}
	-
	-/-------------------------------------------------------------------------/
	-int Sound_AltBuildAudioCVT( Sound_AudioCVT *Data,
	- SDL_AudioSpec src, SDL_AudioSpec dst )
	-{
	- SDL_AudioSpec im;
	-
	- if( Data == NULL ) return -1;
	-
	- initAudioCVT( Data );
	- Data->filter.ratio.denominator = 0;
	- Data->filter.mask = dst.size - 1;
	-
	- /* Check channels */
	- if( src.channels < 1 \|\| src.channels > 2 \|\|
	- dst.channels < 1 \|\| dst.channels > 2 ) goto error_exit;
	-
	- if( src.freq != dst.freq )
	- {
	- /* Convert to intermidiate format: signed 16Bit System-Endian */
	- im.format = AUDIO_S16SYS;
	- im.channels = min( src.channels, dst.channels );
	- createFormatConverter( Data, src, im );
	-
	- /* Do rate conversion */
	- if( createRateConverter( Data, src.freq, dst.freq, im.channels ) )
	- goto error_exit;
	-
	- src = im;
	- }
	-
	- /* Convert to final format */
	- createFormatConverter( Data, src, dst );
	-
	- /* Finalize adapter list */
	- addAdapter( Data, NULL );
	-/* !!! FIXME: Is it okay to assign NULL to a function pointer?
	- Borland says no. -frank */
	- return 0;
	-
	-error_exit:
	-/* !!! FIXME: Is it okay to assign NULL to a function pointer?
	- Borland says no. -frank */
	- Data->adapter[0] = NULL;
	- return -1;
	-}
	-
	-/-------------------------------------------------------------------------/
	-static char *fmt_to_str(Uint16 fmt)
	-{
	- switch (fmt)
	- {
	- case AUDIO_U8: return " U8";
	- case AUDIO_S8: return " S8";
	- case AUDIO_U16MSB: return "U16MSB";
	- case AUDIO_S16MSB: return "S16MSB";
	- case AUDIO_U16LSB: return "U16LSB";
	- case AUDIO_S16LSB: return "S16LSB";
	- }
	- return "??????";
	-}
	-
	-#define AdapterDesc(x) { x, #x }
	-
	-static void show_AudioCVT( Sound_AudioCVT *Data )
	-{
	- int i,j;
	- const struct{ int (adapter) ( AdapterC, int); Sint8 name; }
	- AdapterDescription[] = {
	- AdapterDesc(expand8BitTo16BitSys),
	- AdapterDesc(expand8BitTo16BitWrong),
	- AdapterDesc(expand16BitToFloat),
	- AdapterDesc(swapBytes),
	- AdapterDesc(cut16BitSysTo8Bit),
	- AdapterDesc(cut16BitWrongTo8Bit),
	- AdapterDesc(cutFloatTo16Bit),
	- AdapterDesc(changeSigned16BitSys),
	- AdapterDesc(changeSigned16BitWrong),
	- AdapterDesc(changeSigned8Bit),
	- AdapterDesc(convertStereoToMonoS16Bit),
	- AdapterDesc(convertStereoToMonoU16Bit),
	- AdapterDesc(convertStereoToMonoS8Bit),
	- AdapterDesc(convertStereoToMonoU8Bit),
	- AdapterDesc(convertMonoToStereo16Bit),
	- AdapterDesc(convertMonoToStereo8Bit),
	- AdapterDesc(minus5dB),
	- AdapterDesc(doubleRateMono),
	- AdapterDesc(doubleRateStereo),
	- AdapterDesc(halfRateMono),
	- AdapterDesc(halfRateStereo),
	- AdapterDesc(increaseRateMono),
	- AdapterDesc(increaseRateStereo),
	- AdapterDesc(decreaseRateMono),
	- AdapterDesc(decreaseRateStereo),
	- { NULL, "----------NULL-----------\n" }
	- };
	-
	- fprintf( stderr, "Sound_AudioCVT:\n" );
	- fprintf( stderr, " needed: %8d\n", Data->needed );
	- fprintf( stderr, " add: %8g\n", Data->add );
	- fprintf( stderr, " len_add: %8d\n", Data->len_add );
	- fprintf( stderr, " len_ratio: %8g\n", Data->len_ratio );
	- fprintf( stderr, " len_mult: %8d\n", Data->len_mult );
	- fprintf( stderr, " filter->mask: %#7x\n", Data->filter.mask );
	- fprintf( stderr, "\n" );
	-
	- fprintf( stderr, "Adapter List: \n" );
	- for( i = 0; i < 32; i++ )
	- {
	- for( j = 0; j < SDL_TABLESIZE(AdapterDescription); j++ )
	- {
	- if( Data->adapter[i] == AdapterDescription[j].adapter )
	- {
	- fprintf( stderr, " %s \n", AdapterDescription[j].name );
	- if( Data->adapter[i] == NULL ) goto sucess_exit;
	- goto cont;
	- }
	- }
	- fprintf( stderr, " Error: unknown adapter\n" );
	-
	- cont:
	- }
	- fprintf( stderr, " Error: NULL adapter missing\n" );
	- sucess_exit:
	- if( Data->filter.ratio.denominator )
	- {
	- fprintf( stderr, "Variable Rate Converter:\n"
	- " numerator: %3d\n"
	- " denominator: %3d\n",
	- Data->filter.ratio.denominator,
	- Data->filter.ratio.numerator );
	-
	- fprintf( stderr, " increment sequence:\n"
	- " " );
	- for( i = 0; i < Data->filter.ratio.denominator; i++ )
	- fprintf( stderr, "%1d ", Data->filter.incr[i] );
	-
	- fprintf( stderr, "\n" );
	- }
	- else
	- {
	- fprintf( stderr, "No Variable Rate Converter\n" );
	- }
	-}
	-
	-
	-int Sound_BuildAudioCVT(Sound_AudioCVT *Data,
	- Uint16 src_format, Uint8 src_channels, Uint32 src_rate,
	- Uint16 dst_format, Uint8 dst_channels, Uint32 dst_rate, Uint32 bufsize)
	-{
	- SDL_AudioSpec src, dst;
	- int ret;
	-
	- fprintf (stderr,
	- "Sound_BuildAudioCVT():\n"
	- "-----------------------------\n"
	- "format: %s -> %s\n"
	- "channels: %6d -> %6d\n"
	- "rate: %6d -> %6d\n"
	- "size: don't care -> %#7x\n\n",
	- fmt_to_str (src_format), fmt_to_str (dst_format),
	- src_channels, dst_channels,
	- src_rate, dst_rate );
	-
	- src.format = src_format;
	- src.channels = src_channels;
	- src.freq = src_rate;
	-
	- dst.format = dst_format;
	- dst.channels = dst_channels;
	- dst.freq = dst_rate;
	-
	- ret = Sound_AltBuildAudioCVT( Data, src, dst );
	- Data->needed = 1;
	-
	- show_AudioCVT( Data );
	- fprintf (stderr, "\n"
	- "return value: %d \n\n\n", ret );
	- return ret;
	-}
	-
	-#endif /* SOUND_USE_ALTCVT */
	-
	-/* end of alt_audio_convert.c ... */
	-
	diff --git a/util/sdl/sound/alt_audio_convert.h b/util/sdl/sound/alt_audio_convert.h
	deleted file mode 100644
	index 8dd4670e..00000000
	--- a/util/sdl/sound/alt_audio_convert.h
	+++ /dev/null
	@@ -1,89 +0,0 @@
	-/*
	- * Extended Audio Converter for SDL (Simple DirectMedia Layer)
	- * Copyright (C) 2002 Frank Ranostaj
	- * Institute of Applied Physik
	- * Johann Wolfgang Goethe-Universität
	- * Frankfurt am Main, Germany
	- *
	- * This library is free software; you can redistribute it and/or
	- * modify it under the terms of the GNU Library General Public
	- * License as published by the Free Software Foundation; either
	- * version 2 of the License, or (at your option) any later version.
	- *
	- * This library is distributed in the hope that it will be useful,
	- * but WITHOUT ANY WARRANTY; without even the implied warranty of
	- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	- * Library General Public License for more details.
	- *
	- * You should have received a copy of the GNU Library General Public
	- * License along with this library; if not, write to the Free
	- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
	- *
	- * Frank Ranostaj
	- * ranostaj@stud.uni-frankfurt.de
	- *
	- * (This code blatantly abducted for SDL_sound. Thanks, Frank! --ryan.)
	- */
	-
	-#ifndef _INCLUDE_AUDIO_CONVERT_H_
	-#define _INCLUDE_AUDIO_CONVERT_H_
	-
	-#include "SDL_audio.h"
	-#define Sound_AI_Loop 0x2
	-#define _fsize 32
	-
	-typedef struct{
	- Sint16 numerator;
	- Sint16 denominator;
	-} Fraction;
	-
	-typedef struct{
	- Sint16 c[16][4*_fsize];
	- Uint8 incr[16];
	- Fraction ratio;
	- int mask;
	-} VarFilter;
	-
	-typedef struct{
	- Uint8* buffer;
	- int mode;
	- VarFilter *filter;
	-} AdapterC;
	-
	-typedef int (*Adapter) ( AdapterC Data, int length );
	-
	-typedef struct{
	- VarFilter filter;
	- int filter_index;
	- Adapter adapter[32];
	-/* buffer must be lenlen_mult(+len_add) big /
	- int len_mult;
	- int len_add;
	- double add;
	-
	-/* the following elements are provided for compatibility: */
	-/* the size of the output is approx lenlen_ratio /
	- double len_ratio;
	- Uint8* buf; /* input/output buffer */
	- int needed; /* 0 if nothing to be done, 1 otherwise */
	- int len; /* Length of the input */
	- int len_cvt; /* Length of converted audio buffer */
	-} Sound_AudioCVT;
	-
	-#define SDL_SOUND_Loop 0x10
	-
	-#ifndef SNDDECLSPEC
	-#define SNDDECLSPEC DECLSPEC
	-#endif
	-
	-extern SNDDECLSPEC int Sound_AltConvertAudio( Sound_AudioCVT *Data,
	- Uint8* buffer, int length, int mode );
	-
	-extern SNDDECLSPEC int Sound_AltBuildAudioCVT( Sound_AudioCVT *Data,
	- SDL_AudioSpec src, SDL_AudioSpec dst );
	-
	-extern SNDDECLSPEC int Sound_estimateBufferSize( Sound_AudioCVT *Data,
	- int length );
	-
	-#endif /* _INCLUDE_AUDIO_CONVERT_H_ */
	-
	diff --git a/util/sdl/sound/audio_convert.c b/util/sdl/sound/audio_convert.c
	deleted file mode 100644
	index 05564dc4..00000000
	--- a/util/sdl/sound/audio_convert.c
	+++ /dev/null
	@@ -1,739 +0,0 @@
	-/*
	- SDL - Simple DirectMedia Layer
	- Copyright (C) 1997, 1998, 1999, 2000, 2001 Sam Lantinga
	-
	- This library is free software; you can redistribute it and/or
	- modify it under the terms of the GNU Library General Public
	- License as published by the Free Software Foundation; either
	- version 2 of the License, or (at your option) any later version.
	-
	- This library is distributed in the hope that it will be useful,
	- but WITHOUT ANY WARRANTY; without even the implied warranty of
	- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	- Library General Public License for more details.
	-
	- You should have received a copy of the GNU Library General Public
	- License along with this library; if not, write to the Free
	- Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
	-
	- Sam Lantinga
	- slouken@devolution.com
	-*/
	-
	-/*
	- * This file was derived from SDL's SDL_audiocvt.c and is an attempt to
	- * address the shortcomings of it.
	- *
	- * Perhaps we can adapt some good filters from SoX?
	- */
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#if !SOUND_USE_ALTCVT
	-
	-#include "SDL.h"
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-/* Functions for audio drivers to perform runtime conversion of audio format */
	-
	-
	-/*
	- * Toggle endianness. This filter is, of course, only applied to 16-bit
	- * audio data.
	- */
	-
	-static void Sound_ConvertEndian(Sound_AudioCVT cvt, Uint16 format)
	-{
	- int i;
	- Uint8 *data, tmp;
	-
	- /* SNDDBG(("Converting audio endianness\n")); */
	-
	- data = cvt->buf;
	-
	- for (i = cvt->len_cvt / 2; i; --i)
	- {
	- tmp = data[0];
	- data[0] = data[1];
	- data[1] = tmp;
	- data += 2;
	- } /* for */
	-
	- format = (format ^ 0x1000);
	-} /* Sound_ConvertEndian */
	-
	-
	-/*
	- * Toggle signed/unsigned. Apparently this is done by toggling the most
	- * significant bit of each sample.
	- */
	-
	-static void Sound_ConvertSign(Sound_AudioCVT cvt, Uint16 format)
	-{
	- int i;
	- Uint8 *data;
	-
	- /* SNDDBG(("Converting audio signedness\n")); */
	-
	- data = cvt->buf;
	-
	- /* 16-bit sound? */
	- if ((*format & 0xFF) == 16)
	- {
	- /* Little-endian? */
	- if ((*format & 0x1000) != 0x1000)
	- ++data;
	-
	- for (i = cvt->len_cvt / 2; i; --i)
	- {
	- *data ^= 0x80;
	- data += 2;
	- } /* for */
	- } /* if */
	- else
	- {
	- for (i = cvt->len_cvt; i; --i)
	- *data++ ^= 0x80;
	- } /* else */
	-
	- format = (format ^ 0x8000);
	-} /* Sound_ConvertSign */
	-
	-
	-/*
	- * Convert 16-bit to 8-bit. This is done by taking the most significant byte
	- * of each 16-bit sample.
	- */
	-
	-static void Sound_Convert8(Sound_AudioCVT cvt, Uint16 format)
	-{
	- int i;
	- Uint8 src, dst;
	-
	- /* SNDDBG(("Converting to 8-bit\n")); */
	-
	- src = cvt->buf;
	- dst = cvt->buf;
	-
	- /* Little-endian? */
	- if ((*format & 0x1000) != 0x1000)
	- ++src;
	-
	- for (i = cvt->len_cvt / 2; i; --i)
	- {
	- dst = src;
	- src += 2;
	- dst += 1;
	- } /* for */
	-
	- format = ((format & ~0x9010) \| AUDIO_U8);
	- cvt->len_cvt /= 2;
	-} /* Sound_Convert8 */
	-
	-
	-/* Convert 8-bit to 16-bit - LSB */
	-
	-static void Sound_Convert16LSB(Sound_AudioCVT cvt, Uint16 format)
	-{
	- int i;
	- Uint8 src, dst;
	-
	- /* SNDDBG(("Converting to 16-bit LSB\n")); */
	-
	- src = cvt->buf + cvt->len_cvt;
	- dst = cvt->buf + cvt->len_cvt * 2;
	-
	- for (i = cvt->len_cvt; i; --i)
	- {
	- src -= 1;
	- dst -= 2;
	- dst[1] = *src;
	- dst[0] = 0;
	- } /* for */
	-
	- format = ((format & ~0x0008) \| AUDIO_U16LSB);
	- cvt->len_cvt *= 2;
	-} /* Sound_Convert16LSB */
	-
	-
	-/* Convert 8-bit to 16-bit - MSB */
	-
	-static void Sound_Convert16MSB(Sound_AudioCVT cvt, Uint16 format)
	-{
	- int i;
	- Uint8 src, dst;
	-
	- /* SNDDBG(("Converting to 16-bit MSB\n")); */
	-
	- src = cvt->buf + cvt->len_cvt;
	- dst = cvt->buf + cvt->len_cvt * 2;
	-
	- for (i = cvt->len_cvt; i; --i)
	- {
	- src -= 1;
	- dst -= 2;
	- dst[0] = *src;
	- dst[1] = 0;
	- } /* for */
	-
	- format = ((format & ~0x0008) \| AUDIO_U16MSB);
	- cvt->len_cvt *= 2;
	-} /* Sound_Convert16MSB */
	-
	-
	-/* Duplicate a mono channel to both stereo channels */
	-
	-static void Sound_ConvertStereo(Sound_AudioCVT cvt, Uint16 format)
	-{
	- int i;
	-
	- /* SNDDBG(("Converting to stereo\n")); */
	-
	- /* 16-bit sound? */
	- if ((*format & 0xFF) == 16)
	- {
	- Uint16 src, dst;
	-
	- src = (Uint16 *) (cvt->buf + cvt->len_cvt);
	- dst = (Uint16 ) (cvt->buf + cvt->len_cvt 2);
	-
	- for (i = cvt->len_cvt/2; i; --i)
	- {
	- dst -= 2;
	- src -= 1;
	- dst[0] = src[0];
	- dst[1] = src[0];
	- } /* for */
	- } /* if */
	- else
	- {
	- Uint8 src, dst;
	-
	- src = cvt->buf + cvt->len_cvt;
	- dst = cvt->buf + cvt->len_cvt * 2;
	-
	- for (i = cvt->len_cvt; i; --i)
	- {
	- dst -= 2;
	- src -= 1;
	- dst[0] = src[0];
	- dst[1] = src[0];
	- } /* for */
	- } /* else */
	-
	- cvt->len_cvt *= 2;
	-} /* Sound_ConvertStereo */
	-
	-
	-/* Effectively mix right and left channels into a single channel */
	-
	-static void Sound_ConvertMono(Sound_AudioCVT cvt, Uint16 format)
	-{
	- int i;
	- Sint32 sample;
	- Uint8 u_src, u_dst;
	- Sint8 s_src, s_dst;
	-
	- /* SNDDBG(("Converting to mono\n")); */
	-
	- switch (*format)
	- {
	- case AUDIO_U8:
	- u_src = cvt->buf;
	- u_dst = cvt->buf;
	-
	- for (i = cvt->len_cvt / 2; i; --i)
	- {
	- sample = u_src[0] + u_src[1];
	- *u_dst = (sample > 255) ? 255 : sample;
	- u_src += 2;
	- u_dst += 1;
	- } /* for */
	- break;
	-
	- case AUDIO_S8:
	- s_src = (Sint8 *) cvt->buf;
	- s_dst = (Sint8 *) cvt->buf;
	-
	- for (i = cvt->len_cvt / 2; i; --i)
	- {
	- sample = s_src[0] + s_src[1];
	- if (sample > 127)
	- *s_dst = 127;
	- else if (sample < -128)
	- *s_dst = -128;
	- else
	- *s_dst = sample;
	-
	- s_src += 2;
	- s_dst += 1;
	- } /* for */
	- break;
	-
	- case AUDIO_U16MSB:
	- u_src = cvt->buf;
	- u_dst = cvt->buf;
	-
	- for (i = cvt->len_cvt / 4; i; --i)
	- {
	- sample = (Uint16) ((u_src[0] << 8) \| u_src[1])
	- + (Uint16) ((u_src[2] << 8) \| u_src[3]);
	- if (sample > 65535)
	- {
	- u_dst[0] = 0xFF;
	- u_dst[1] = 0xFF;
	- } /* if */
	- else
	- {
	- u_dst[1] = (sample & 0xFF);
	- sample >>= 8;
	- u_dst[0] = (sample & 0xFF);
	- } /* else */
	- u_src += 4;
	- u_dst += 2;
	- } /* for */
	- break;
	-
	- case AUDIO_U16LSB:
	- u_src = cvt->buf;
	- u_dst = cvt->buf;
	-
	- for (i = cvt->len_cvt / 4; i; --i)
	- {
	- sample = (Uint16) ((u_src[1] << 8) \| u_src[0])
	- + (Uint16) ((u_src[3] << 8) \| u_src[2]);
	- if (sample > 65535)
	- {
	- u_dst[0] = 0xFF;
	- u_dst[1] = 0xFF;
	- } /* if */
	- else
	- {
	- u_dst[0] = (sample & 0xFF);
	- sample >>= 8;
	- u_dst[1] = (sample & 0xFF);
	- } /* else */
	- u_src += 4;
	- u_dst += 2;
	- } /* for */
	- break;
	-
	- case AUDIO_S16MSB:
	- u_src = cvt->buf;
	- u_dst = cvt->buf;
	-
	- for (i = cvt->len_cvt / 4; i; --i)
	- {
	- sample = (Sint16) ((u_src[0] << 8) \| u_src[1])
	- + (Sint16) ((u_src[2] << 8) \| u_src[3]);
	- if (sample > 32767)
	- {
	- u_dst[0] = 0x7F;
	- u_dst[1] = 0xFF;
	- } /* if */
	- else if (sample < -32768)
	- {
	- u_dst[0] = 0x80;
	- u_dst[1] = 0x00;
	- } /* else if */
	- else
	- {
	- u_dst[1] = (sample & 0xFF);
	- sample >>= 8;
	- u_dst[0] = (sample & 0xFF);
	- } /* else */
	- u_src += 4;
	- u_dst += 2;
	- } /* for */
	- break;
	-
	- case AUDIO_S16LSB:
	- u_src = cvt->buf;
	- u_dst = cvt->buf;
	-
	- for (i = cvt->len_cvt / 4; i; --i)
	- {
	- sample = (Sint16) ((u_src[1] << 8) \| u_src[0])
	- + (Sint16) ((u_src[3] << 8) \| u_src[2]);
	- if (sample > 32767)
	- {
	- u_dst[1] = 0x7F;
	- u_dst[0] = 0xFF;
	- } /* if */
	- else if (sample < -32768)
	- {
	- u_dst[1] = 0x80;
	- u_dst[0] = 0x00;
	- } /* else if */
	- else
	- {
	- u_dst[0] = (sample & 0xFF);
	- sample >>= 8;
	- u_dst[1] = (sample & 0xFF);
	- } /* else */
	- u_src += 4;
	- u_dst += 2;
	- } /* for */
	- break;
	- } /* switch */
	-
	- cvt->len_cvt /= 2;
	-} /* Sound_ConvertMono */
	-
	-
	-/* Convert rate up by multiple of 2 */
	-
	-static void Sound_RateMUL2(Sound_AudioCVT cvt, Uint16 format)
	-{
	- int i;
	- Uint8 src, dst;
	-
	- /* SNDDBG(("Converting audio rate * 2\n")); */
	-
	- src = cvt->buf + cvt->len_cvt;
	- dst = cvt->buf + cvt->len_cvt*2;
	-
	- /* 8- or 16-bit sound? */
	- switch (*format & 0xFF)
	- {
	- case 8:
	- for (i = cvt->len_cvt; i; --i)
	- {
	- src -= 1;
	- dst -= 2;
	- dst[0] = src[0];
	- dst[1] = src[0];
	- } /* for */
	- break;
	-
	- case 16:
	- for (i = cvt->len_cvt / 2; i; --i)
	- {
	- src -= 2;
	- dst -= 4;
	- dst[0] = src[0];
	- dst[1] = src[1];
	- dst[2] = src[0];
	- dst[3] = src[1];
	- } /* for */
	- break;
	- } /* switch */
	-
	- cvt->len_cvt *= 2;
	-} /* Sound_RateMUL2 */
	-
	-
	-/* Convert rate down by multiple of 2 */
	-
	-static void Sound_RateDIV2(Sound_AudioCVT cvt, Uint16 format)
	-{
	- int i;
	- Uint8 src, dst;
	-
	- /* SNDDBG(("Converting audio rate / 2\n")); */
	-
	- src = cvt->buf;
	- dst = cvt->buf;
	-
	- /* 8- or 16-bit sound? */
	- switch (*format & 0xFF)
	- {
	- case 8:
	- for (i = cvt->len_cvt / 2; i; --i)
	- {
	- dst[0] = src[0];
	- src += 2;
	- dst += 1;
	- } /* for */
	- break;
	-
	- case 16:
	- for (i = cvt->len_cvt / 4; i; --i)
	- {
	- dst[0] = src[0];
	- dst[1] = src[1];
	- src += 4;
	- dst += 2;
	- }
	- break;
	- } /* switch */
	-
	- cvt->len_cvt /= 2;
	-} /* Sound_RateDIV2 */
	-
	-
	-/* Very slow rate conversion routine */
	-
	-static void Sound_RateSLOW(Sound_AudioCVT cvt, Uint16 format)
	-{
	- double ipos;
	- int i, clen;
	- Uint8 *output8;
	- Uint16 *output16;
	-
	- /* SNDDBG(("Converting audio rate * %4.4f\n", 1.0/cvt->rate_incr)); */
	-
	- clen = (int) ((double) cvt->len_cvt / cvt->rate_incr);
	-
	- if (cvt->rate_incr > 1.0)
	- {
	- /* 8- or 16-bit sound? */
	- switch (*format & 0xFF)
	- {
	- case 8:
	- output8 = cvt->buf;
	-
	- ipos = 0.0;
	- for (i = clen; i; --i)
	- {
	- *output8 = cvt->buf[(int) ipos];
	- ipos += cvt->rate_incr;
	- output8 += 1;
	- } /* for */
	- break;
	-
	- case 16:
	- output16 = (Uint16 *) cvt->buf;
	-
	- clen &= ~1;
	- ipos = 0.0;
	- for (i = clen / 2; i; --i)
	- {
	- output16 = ((Uint16 ) cvt->buf)[(int) ipos];
	- ipos += cvt->rate_incr;
	- output16 += 1;
	- } /* for */
	- break;
	- } /* switch */
	- } /* if */
	- else
	- {
	- /* 8- or 16-bit sound */
	- switch (*format & 0xFF)
	- {
	- case 8:
	- output8 = cvt->buf + clen;
	-
	- ipos = (double) cvt->len_cvt;
	- for (i = clen; i; --i)
	- {
	- ipos -= cvt->rate_incr;
	- output8 -= 1;
	- *output8 = cvt->buf[(int) ipos];
	- } /* for */
	- break;
	-
	- case 16:
	- clen &= ~1;
	- output16 = (Uint16 *) (cvt->buf + clen);
	- ipos = (double) cvt->len_cvt / 2;
	- for (i = clen / 2; i; --i)
	- {
	- ipos -= cvt->rate_incr;
	- output16 -= 1;
	- output16 = ((Uint16 ) cvt->buf)[(int) ipos];
	- } /* for */
	- break;
	- } /* switch */
	- } /* else */
	-
	- cvt->len_cvt = clen;
	-} /* Sound_RateSLOW */
	-
	-
	-int Sound_ConvertAudio(Sound_AudioCVT *cvt)
	-{
	- Uint16 format;
	-
	- /* Make sure there's data to convert */
	- if (cvt->buf == NULL)
	- {
	- __Sound_SetError("No buffer allocated for conversion");
	- return(-1);
	- } /* if */
	-
	- /* Return okay if no conversion is necessary */
	- cvt->len_cvt = cvt->len;
	- if (cvt->filters[0] == NULL)
	- return(0);
	-
	- /* Set up the conversion and go! */
	- format = cvt->src_format;
	- for (cvt->filter_index = 0; cvt->filters[cvt->filter_index];
	- cvt->filter_index++)
	- {
	- cvt->filters[cvt->filter_index](cvt, &format);
	- }
	- return(0);
	-} /* Sound_ConvertAudio */
	-
	-
	-/*
	- * Creates a set of audio filters to convert from one format to another.
	- * Returns -1 if the format conversion is not supported, or 1 if the
	- * audio filter is set up.
	- */
	-
	-int Sound_BuildAudioCVT(Sound_AudioCVT *cvt,
	- Uint16 src_format, Uint8 src_channels, Uint32 src_rate,
	- Uint16 dst_format, Uint8 dst_channels, Uint32 dst_rate,
	- Uint32 dst_size)
	-{
	- /* Start off with no conversion necessary */
	- cvt->needed = 0;
	- cvt->filter_index = 0;
	- cvt->filters[0] = NULL;
	- cvt->len_mult = 1;
	- cvt->len_ratio = 1.0;
	-
	- /* First filter: Endian conversion from src to dst */
	- if ((src_format & 0x1000) != (dst_format & 0x1000) &&
	- ((src_format & 0xff) != 8))
	- {
	- SNDDBG(("Adding filter: Sound_ConvertEndian\n"));
	- cvt->filters[cvt->filter_index++] = Sound_ConvertEndian;
	- } /* if */
	-
	- /* Second filter: Sign conversion -- signed/unsigned */
	- if ((src_format & 0x8000) != (dst_format & 0x8000))
	- {
	- SNDDBG(("Adding filter: Sound_ConvertSign\n"));
	- cvt->filters[cvt->filter_index++] = Sound_ConvertSign;
	- } /* if */
	-
	- /* Next filter: Convert 16 bit <--> 8 bit PCM. */
	- if ((src_format & 0xFF) != (dst_format & 0xFF))
	- {
	- switch (dst_format & 0x10FF)
	- {
	- case AUDIO_U8:
	- SNDDBG(("Adding filter: Sound_Convert8\n"));
	- cvt->filters[cvt->filter_index++] = Sound_Convert8;
	- cvt->len_ratio /= 2;
	- break;
	-
	- case AUDIO_U16LSB:
	- SNDDBG(("Adding filter: Sound_Convert16LSB\n"));
	- cvt->filters[cvt->filter_index++] = Sound_Convert16LSB;
	- cvt->len_mult *= 2;
	- cvt->len_ratio *= 2;
	- break;
	-
	- case AUDIO_U16MSB:
	- SNDDBG(("Adding filter: Sound_Convert16MSB\n"));
	- cvt->filters[cvt->filter_index++] = Sound_Convert16MSB;
	- cvt->len_mult *= 2;
	- cvt->len_ratio *= 2;
	- break;
	- } /* switch */
	- } /* if */
	-
	- /* Next filter: Mono/Stereo conversion */
	- if (src_channels != dst_channels)
	- {
	- while ((src_channels * 2) <= dst_channels)
	- {
	- SNDDBG(("Adding filter: Sound_ConvertStereo\n"));
	- cvt->filters[cvt->filter_index++] = Sound_ConvertStereo;
	- cvt->len_mult *= 2;
	- src_channels *= 2;
	- cvt->len_ratio *= 2;
	- } /* while */
	-
	- /* This assumes that 4 channel audio is in the format:
	- * Left {front/back} + Right {front/back}
	- * so converting to L/R stereo works properly.
	- */
	- while (((src_channels % 2) == 0) &&
	- ((src_channels / 2) >= dst_channels))
	- {
	- SNDDBG(("Adding filter: Sound_ConvertMono\n"));
	- cvt->filters[cvt->filter_index++] = Sound_ConvertMono;
	- src_channels /= 2;
	- cvt->len_ratio /= 2;
	- } /* while */
	-
	- if ( src_channels != dst_channels ) {
	- /* Uh oh.. */;
	- } /* if */
	- } /* if */
	-
	- /* Do rate conversion */
	- cvt->rate_incr = 0.0;
	- if ((src_rate / 100) != (dst_rate / 100))
	- {
	- Uint32 hi_rate, lo_rate;
	- int len_mult;
	- double len_ratio;
	- void (rate_cvt)(Sound_AudioCVT cvt, Uint16 *format);
	-
	- if (src_rate > dst_rate)
	- {
	- hi_rate = src_rate;
	- lo_rate = dst_rate;
	- SNDDBG(("Adding filter: Sound_RateDIV2\n"));
	- rate_cvt = Sound_RateDIV2;
	- len_mult = 1;
	- len_ratio = 0.5;
	- } /* if */
	- else
	- {
	- hi_rate = dst_rate;
	- lo_rate = src_rate;
	- SNDDBG(("Adding filter: Sound_RateMUL2\n"));
	- rate_cvt = Sound_RateMUL2;
	- len_mult = 2;
	- len_ratio = 2.0;
	- } /* else */
	-
	- /* If hi_rate = lo_rate2^x then conversion is easy /
	- while (((lo_rate * 2) / 100) <= (hi_rate / 100))
	- {
	- cvt->filters[cvt->filter_index++] = rate_cvt;
	- cvt->len_mult *= len_mult;
	- lo_rate *= 2;
	- cvt->len_ratio *= len_ratio;
	- } /* while */
	-
	- /* We may need a slow conversion here to finish up */
	- if ((lo_rate / 100) != (hi_rate / 100))
	- {
	- if (src_rate < dst_rate)
	- {
	- cvt->rate_incr = (double) lo_rate / hi_rate;
	- cvt->len_mult *= 2;
	- cvt->len_ratio /= cvt->rate_incr;
	- } /* if */
	- else
	- {
	- cvt->rate_incr = (double) hi_rate / lo_rate;
	- cvt->len_ratio *= cvt->rate_incr;
	- } /* else */
	- SNDDBG(("Adding filter: Sound_RateSLOW\n"));
	- cvt->filters[cvt->filter_index++] = Sound_RateSLOW;
	- } /* if */
	- } /* if */
	-
	- /* Set up the filter information */
	- if (cvt->filter_index != 0)
	- {
	- cvt->needed = 1;
	- cvt->src_format = src_format;
	- cvt->dst_format = dst_format;
	- cvt->len = 0;
	- cvt->buf = NULL;
	- cvt->filters[cvt->filter_index] = NULL;
	- } /* if */
	-
	- return(cvt->needed);
	-} /* Sound_BuildAudioCVT */
	-
	-#endif /* !SOUND_USE_ALTCVT */
	-
	-/* end of audio_convert.c ... */
	-
	diff --git a/util/sdl/sound/compile b/util/sdl/sound/compile
	deleted file mode 100755
	index 1b1d2321..00000000
	--- a/util/sdl/sound/compile
	+++ /dev/null
	@@ -1,142 +0,0 @@
	-#! /bin/sh
	-# Wrapper for compilers which do not understand `-c -o'.
	-
	-scriptversion=2005-05-14.22
	-
	-# Copyright (C) 1999, 2000, 2003, 2004, 2005 Free Software Foundation, Inc.
	-# Written by Tom Tromey <tromey@cygnus.com>.
	-#
	-# This program is free software; you can redistribute it and/or modify
	-# it under the terms of the GNU General Public License as published by
	-# the Free Software Foundation; either version 2, or (at your option)
	-# any later version.
	-#
	-# This program is distributed in the hope that it will be useful,
	-# but WITHOUT ANY WARRANTY; without even the implied warranty of
	-# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
	-# GNU General Public License for more details.
	-#
	-# You should have received a copy of the GNU General Public License
	-# along with this program; if not, write to the Free Software
	-# Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
	-
	-# As a special exception to the GNU General Public License, if you
	-# distribute this file as part of a program that contains a
	-# configuration script generated by Autoconf, you may include it under
	-# the same distribution terms that you use for the rest of that program.
	-
	-# This file is maintained in Automake, please report
	-# bugs to <bug-automake@gnu.org> or send patches to
	-# <automake-patches@gnu.org>.
	-
	-case $1 in
	- '')
	- echo "$0: No command. Try \`$0 --help' for more information." 1>&2
	- exit 1;
	- ;;
	- -h \| --h*)
	- cat <<\EOF
	-Usage: compile [--help] [--version] PROGRAM [ARGS]
	-
	-Wrapper for compilers which do not understand `-c -o'.
	-Remove `-o dest.o' from ARGS, run PROGRAM with the remaining
	-arguments, and rename the output as expected.
	-
	-If you are trying to build a whole package this is not the
	-right script to run: please start by reading the file `INSTALL'.
	-
	-Report bugs to <bug-automake@gnu.org>.
	-EOF
	- exit $?
	- ;;
	- -v \| --v*)
	- echo "compile $scriptversion"
	- exit $?
	- ;;
	-esac
	-
	-ofile=
	-cfile=
	-eat=
	-
	-for arg
	-do
	- if test -n "$eat"; then
	- eat=
	- else
	- case $1 in
	- -o)
	- # configure might choose to run compile as `compile cc -o foo foo.c'.
	- # So we strip `-o arg' only if arg is an object.
	- eat=1
	- case $2 in
	- .o \| .obj)
	- ofile=$2
	- ;;
	- *)
	- set x "$@" -o "$2"
	- shift
	- ;;
	- esac
	- ;;
	- *.c)
	- cfile=$1
	- set x "$@" "$1"
	- shift
	- ;;
	- *)
	- set x "$@" "$1"
	- shift
	- ;;
	- esac
	- fi
	- shift
	-done
	-
	-if test -z "$ofile" \|\| test -z "$cfile"; then
	- # If no `-o' option was seen then we might have been invoked from a
	- # pattern rule where we don't need one. That is ok -- this is a
	- # normal compilation that the losing compiler can handle. If no
	- # `.c' file was seen then we are probably linking. That is also
	- # ok.
	- exec "$@"
	-fi
	-
	-# Name of file we expect compiler to create.
	-cofile=`echo "$cfile" \| sed -e 's\|^.*/\|\|' -e 's/\.c$/.o/'`
	-
	-# Create the lock directory.
	-# Note: use `[/.-]' here to ensure that we don't use the same name
	-# that we are using for the .o file. Also, base the name on the expected
	-# object file name, since that is what matters with a parallel build.
	-lockdir=`echo "$cofile" \| sed -e 's\|[/.-]\|_\|g'`.d
	-while true; do
	- if mkdir "$lockdir" >/dev/null 2>&1; then
	- break
	- fi
	- sleep 1
	-done
	-# FIXME: race condition here if user kills between mkdir and trap.
	-trap "rmdir '$lockdir'; exit 1" 1 2 15
	-
	-# Run the compile.
	-"$@"
	-ret=$?
	-
	-if test -f "$cofile"; then
	- mv "$cofile" "$ofile"
	-elif test -f "${cofile}bj"; then
	- mv "${cofile}bj" "$ofile"
	-fi
	-
	-rmdir "$lockdir"
	-exit $ret
	-
	-# Local Variables:
	-# mode: shell-script
	-# sh-indentation: 2
	-# eval: (add-hook 'write-file-hooks 'time-stamp)
	-# time-stamp-start: "scriptversion="
	-# time-stamp-format: "%:y-%02m-%02d.%02H"
	-# time-stamp-end: "$"
	-# End:
	diff --git a/util/sdl/sound/config.h.in b/util/sdl/sound/config.h.in
	deleted file mode 100644
	index d07258da..00000000
	--- a/util/sdl/sound/config.h.in
	+++ /dev/null
	@@ -1,133 +0,0 @@
	-/* config.h.in. Generated from configure.in by autoheader. */
	-
	-/* Define for debug builds. */
	-#undef DEBUG
	-
	-/* Define for debug build chattering. */
	-#undef DEBUG_CHATTER
	-
	-/* Define to 1 if you have the <assert.h> header file. */
	-#undef HAVE_ASSERT_H
	-
	-/* Define to 1 if you have the <dlfcn.h> header file. */
	-#undef HAVE_DLFCN_H
	-
	-/* Define to 1 if you have the <inttypes.h> header file. */
	-#undef HAVE_INTTYPES_H
	-
	-/* Define to 1 if you have the <memory.h> header file. */
	-#undef HAVE_MEMORY_H
	-
	-/* Define to 1 if you have the `memset' function. */
	-#undef HAVE_MEMSET
	-
	-/* Define to 1 if you have the `setbuf' function. */
	-#undef HAVE_SETBUF
	-
	-/* Define to 1 if you have the <signal.h> header file. */
	-#undef HAVE_SIGNAL_H
	-
	-/* Define to 1 if you have the <stdint.h> header file. */
	-#undef HAVE_STDINT_H
	-
	-/* Define to 1 if you have the <stdlib.h> header file. */
	-#undef HAVE_STDLIB_H
	-
	-/* Define to 1 if you have the <strings.h> header file. */
	-#undef HAVE_STRINGS_H
	-
	-/* Define to 1 if you have the <string.h> header file. */
	-#undef HAVE_STRING_H
	-
	-/* Define to 1 if you have the `strrchr' function. */
	-#undef HAVE_STRRCHR
	-
	-/* Define to 1 if you have the <sys/stat.h> header file. */
	-#undef HAVE_SYS_STAT_H
	-
	-/* Define to 1 if you have the <sys/types.h> header file. */
	-#undef HAVE_SYS_TYPES_H
	-
	-/* Define to 1 if you have the <unistd.h> header file. */
	-#undef HAVE_UNISTD_H
	-
	-/* Define to disable debugging. */
	-#undef NDEBUG
	-
	-/* Name of package */
	-#undef PACKAGE
	-
	-/* Define to the address where bug reports for this package should be sent. */
	-#undef PACKAGE_BUGREPORT
	-
	-/* Define to the full name of this package. */
	-#undef PACKAGE_NAME
	-
	-/* Define to the full name and version of this package. */
	-#undef PACKAGE_STRING
	-
	-/* Define to the one symbol short name of this package. */
	-#undef PACKAGE_TARNAME
	-
	-/* Define to the version of this package. */
	-#undef PACKAGE_VERSION
	-
	-/* Define if modplug header is in own directory. */
	-#undef SOUND_MODPLUG_IN_OWN_PATH
	-
	-/* Define if AIFF support is desired. */
	-#undef SOUND_SUPPORTS_AIFF
	-
	-/* Define if AU support is desired. */
	-#undef SOUND_SUPPORTS_AU
	-
	-/* Define if FLAC support is desired. */
	-#undef SOUND_SUPPORTS_FLAC
	-
	-/* Define if MIDI support is desired. */
	-#undef SOUND_SUPPORTS_MIDI
	-
	-/* Define if MIKMOD support is desired. */
	-#undef SOUND_SUPPORTS_MIKMOD
	-
	-/* Define if MODPLUG support is desired. */
	-#undef SOUND_SUPPORTS_MODPLUG
	-
	-/* Define if MPGLIB support is desired. */
	-#undef SOUND_SUPPORTS_MPGLIB
	-
	-/* Define if OGG support is desired. */
	-#undef SOUND_SUPPORTS_OGG
	-
	-/* Define if RAW support is desired. */
	-#undef SOUND_SUPPORTS_RAW
	-
	-/* Define if SHN support is desired. */
	-#undef SOUND_SUPPORTS_SHN
	-
	-/* Define if SMPEG support is desired. */
	-#undef SOUND_SUPPORTS_SMPEG
	-
	-/* Define if SPEEX support is desired. */
	-#undef SOUND_SUPPORTS_SPEEX
	-
	-/* Define if VOC support is desired. */
	-#undef SOUND_SUPPORTS_VOC
	-
	-/* Define if WAV support is desired. */
	-#undef SOUND_SUPPORTS_WAV
	-
	-/* Define to use alternate audio converter. */
	-#undef SOUND_USE_ALTCVT
	-
	-/* Define to 1 if you have the ANSI C header files. */
	-#undef STDC_HEADERS
	-
	-/* Version number of package */
	-#undef VERSION
	-
	-/* Define to empty if `const' does not conform to ANSI C. */
	-#undef const
	-
	-/* Define to `unsigned int' if <sys/types.h> does not define. */
	-#undef size_t
	diff --git a/util/sdl/sound/decoders/Makefile.am b/util/sdl/sound/decoders/Makefile.am
	deleted file mode 100644
	index e6d4f9b4..00000000
	--- a/util/sdl/sound/decoders/Makefile.am
	+++ /dev/null
	@@ -1,22 +0,0 @@
	-noinst_LTLIBRARIES = libdecoders.la
	-
	-SUBDIRS = timidity mpglib
	-
	-INCLUDES = -I$(top_srcdir) -I$(top_srcdir)/decoders/timidity
	-
	-libdecoders_la_SOURCES = \
	- aiff.c \
	- au.c \
	- mikmod.c \
	- modplug.c \
	- mpglib.c \
	- smpeg.c \
	- ogg.c \
	- raw.c \
	- shn.c \
	- voc.c \
	- midi.c \
	- flac.c \
	- speex.c \
	- quicktime.c \
	- wav.c
	diff --git a/util/sdl/sound/decoders/Makefile.in b/util/sdl/sound/decoders/Makefile.in
	deleted file mode 100644
	index b1860112..00000000
	--- a/util/sdl/sound/decoders/Makefile.in
	+++ /dev/null
	@@ -1,593 +0,0 @@
	-# Makefile.in generated by automake 1.9.6 from Makefile.am.
	-# @configure_input@
	-
	-# Copyright (C) 1994, 1995, 1996, 1997, 1998, 1999, 2000, 2001, 2002,
	-# 2003, 2004, 2005 Free Software Foundation, Inc.
	-# This Makefile.in is free software; the Free Software Foundation
	-# gives unlimited permission to copy and/or distribute it,
	-# with or without modifications, as long as this notice is preserved.
	-
	-# This program is distributed in the hope that it will be useful,
	-# but WITHOUT ANY WARRANTY, to the extent permitted by law; without
	-# even the implied warranty of MERCHANTABILITY or FITNESS FOR A
	-# PARTICULAR PURPOSE.
	-
	-@SET_MAKE@
	-
	-srcdir = @srcdir@
	-top_srcdir = @top_srcdir@
	-VPATH = @srcdir@
	-pkgdatadir = $(datadir)/@PACKAGE@
	-pkglibdir = $(libdir)/@PACKAGE@
	-pkgincludedir = $(includedir)/@PACKAGE@
	-top_builddir = ..
	-am__cd = CDPATH="$${ZSH_VERSION+.}$(PATH_SEPARATOR)" && cd
	-INSTALL = @INSTALL@
	-install_sh_DATA = $(install_sh) -c -m 644
	-install_sh_PROGRAM = $(install_sh) -c
	-install_sh_SCRIPT = $(install_sh) -c
	-INSTALL_HEADER = $(INSTALL_DATA)
	-transform = $(program_transform_name)
	-NORMAL_INSTALL = :
	-PRE_INSTALL = :
	-POST_INSTALL = :
	-NORMAL_UNINSTALL = :
	-PRE_UNINSTALL = :
	-POST_UNINSTALL = :
	-build_triplet = @build@
	-host_triplet = @host@
	-target_triplet = @target@
	-subdir = decoders
	-DIST_COMMON = $(srcdir)/Makefile.am $(srcdir)/Makefile.in
	-ACLOCAL_M4 = $(top_srcdir)/aclocal.m4
	-am__aclocal_m4_deps = $(top_srcdir)/acinclude.m4 \
	- $(top_srcdir)/configure.in
	-am__configure_deps = $(am__aclocal_m4_deps) $(CONFIGURE_DEPENDENCIES) \
	- $(ACLOCAL_M4)
	-mkinstalldirs = $(install_sh) -d
	-CONFIG_HEADER = $(top_builddir)/config.h
	-CONFIG_CLEAN_FILES =
	-LTLIBRARIES = $(noinst_LTLIBRARIES)
	-libdecoders_la_LIBADD =
	-am_libdecoders_la_OBJECTS = aiff.lo au.lo mikmod.lo modplug.lo \
	- mpglib.lo smpeg.lo ogg.lo raw.lo shn.lo voc.lo midi.lo flac.lo \
	- speex.lo quicktime.lo wav.lo
	-libdecoders_la_OBJECTS = $(am_libdecoders_la_OBJECTS)
	-DEFAULT_INCLUDES = -I. -I$(srcdir) -I$(top_builddir)
	-depcomp = $(SHELL) $(top_srcdir)/depcomp
	-am__depfiles_maybe = depfiles
	-COMPILE = $(CC) $(DEFS) $(DEFAULT_INCLUDES) $(INCLUDES) $(AM_CPPFLAGS) \
	- $(CPPFLAGS) $(AM_CFLAGS) $(CFLAGS)
	-LTCOMPILE = $(LIBTOOL) --tag=CC --mode=compile $(CC) $(DEFS) \
	- $(DEFAULT_INCLUDES) $(INCLUDES) $(AM_CPPFLAGS) $(CPPFLAGS) \
	- $(AM_CFLAGS) $(CFLAGS)
	-CCLD = $(CC)
	-LINK = $(LIBTOOL) --tag=CC --mode=link $(CCLD) $(AM_CFLAGS) $(CFLAGS) \
	- $(AM_LDFLAGS) $(LDFLAGS) -o $@
	-SOURCES = $(libdecoders_la_SOURCES)
	-DIST_SOURCES = $(libdecoders_la_SOURCES)
	-RECURSIVE_TARGETS = all-recursive check-recursive dvi-recursive \
	- html-recursive info-recursive install-data-recursive \
	- install-exec-recursive install-info-recursive \
	- install-recursive installcheck-recursive installdirs-recursive \
	- pdf-recursive ps-recursive uninstall-info-recursive \
	- uninstall-recursive
	-ETAGS = etags
	-CTAGS = ctags
	-DIST_SUBDIRS = $(SUBDIRS)
	-DISTFILES = $(DIST_COMMON) $(DIST_SOURCES) $(TEXINFOS) $(EXTRA_DIST)
	-ACLOCAL = @ACLOCAL@
	-AMDEP_FALSE = @AMDEP_FALSE@
	-AMDEP_TRUE = @AMDEP_TRUE@
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	- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	- * Lesser General Public License for more details.
	- *
	- * You should have received a copy of the GNU Lesser General Public
	- * License along with this library; if not, write to the Free Software
	- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
	- */
	-
	-/*
	- * AIFF decoder for SDL_sound
	- *
	- * [Insert something profound about the AIFF file format here.]
	- *
	- * This code was ripped from a decoder I had written for SDL_mixer, which was
	- * based on SDL_mixer's old AIFF music loader. (This loader was unfortunately
	- * completely broken, but it was still useful because all the pieces were
	- * still there, so to speak.)
	- *
	- * When rewriting it for SDL_sound, I changed its structure to be more like
	- * the WAV loader Ryan wrote. Had they not both been part of the same project
	- * it would have been embarrassing how similar they are.
	- *
	- * It is not the most feature-complete AIFF loader the world has ever seen.
	- * For instance, it only makes a token attempt at implementing the AIFF-C
	- * standard; basically the parts of it that I can easily understand and test.
	- * It's a start, though.
	- *
	- * Please see the file COPYING in the source's root directory.
	- *
	- * This file was written by Torbjörn Andersson. (d91tan@Update.UU.SE)
	- */
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#ifdef SOUND_SUPPORTS_AIFF
	-
	-#include <stdio.h>
	-#include <stdlib.h>
	-#include <string.h>
	-
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-static Uint32 SANE_to_Uint32 (Uint8 *sanebuf);
	-
	-
	-static int AIFF_init(void);
	-static void AIFF_quit(void);
	-static int AIFF_open(Sound_Sample sample, const char ext);
	-static void AIFF_close(Sound_Sample *sample);
	-static Uint32 AIFF_read(Sound_Sample *sample);
	-static int AIFF_rewind(Sound_Sample *sample);
	-static int AIFF_seek(Sound_Sample *sample, Uint32 ms);
	-
	-static const char *extensions_aiff[] = { "AIFF", NULL };
	-const Sound_DecoderFunctions __Sound_DecoderFunctions_AIFF =
	-{
	- {
	- extensions_aiff,
	- "Audio Interchange File Format",
	- "Torbjörn Andersson <d91tan@Update.UU.SE>",
	- "http://www.icculus.org/SDL_sound/"
	- },
	-
	- AIFF_init, /* init() method */
	- AIFF_quit, /* quit() method */
	- AIFF_open, /* open() method */
	- AIFF_close, /* close() method */
	- AIFF_read, /* read() method */
	- AIFF_rewind, /* rewind() method */
	- AIFF_seek /* seek() method */
	-};
	-
	-
	-/*****************************************************************************
	- * aiff_t is what we store in our internal->decoder_private field... *
	- *****************************************************************************/
	-typedef struct S_AIFF_FMT_T
	-{
	- Uint32 type;
	-
	- Uint32 total_bytes;
	- Uint32 data_starting_offset;
	-
	- void (free)(struct S_AIFF_FMT_T fmt);
	- Uint32 (read_sample)(Sound_Sample sample);
	- int (rewind_sample)(Sound_Sample sample);
	- int (seek_sample)(Sound_Sample sample, Uint32 ms);
	-
	-
	-#if 0
	-/*
	- this is ripped from wav.c as ann example of format-specific data.
	- please replace with something more appropriate when the need arises.
	-*/
	- union
	- {
	- struct
	- {
	- Uint16 cbSize;
	- Uint16 wSamplesPerBlock;
	- Uint16 wNumCoef;
	- ADPCMCOEFSET *aCoeff;
	- } adpcm;
	-
	- /* put other format-specific data here... */
	- } fmt;
	-#endif
	-} fmt_t;
	-
	-
	-typedef struct
	-{
	- fmt_t fmt;
	- Sint32 bytesLeft;
	-} aiff_t;
	-
	-
	-
	- /* Chunk management code... */
	-
	-#define formID 0x4D524F46 /* "FORM", in ascii. */
	-#define aiffID 0x46464941 /* "AIFF", in ascii. */
	-#define aifcID 0x43464941 /* "AIFC", in ascii. */
	-#define ssndID 0x444E5353 /* "SSND", in ascii. */
	-
	-
	-/*****************************************************************************
	- * The COMM chunk... *
	- *****************************************************************************/
	-
	-#define commID 0x4D4D4F43 /* "COMM", in ascii. */
	-
	-/* format/compression types... */
	-#define noneID 0x454E4F4E /* "NONE", in ascii. */
	-
	-typedef struct
	-{
	- Uint32 ckID;
	- Uint32 ckDataSize;
	- Uint16 numChannels;
	- Uint32 numSampleFrames;
	- Uint16 sampleSize;
	- Uint32 sampleRate;
	- /*
	- * We don't handle AIFF-C compressed audio yet, but for those
	- * interested the allowed compression types are supposed to be
	- *
	- * compressionType compressionName meaning
	- * ---------------------------------------------------------------
	- * 'NONE' "not compressed" uncompressed, that is,
	- * straight digitized samples
	- * 'ACE2' "ACE 2-to-1" 2-to-1 IIGS ACE (Audio
	- * Compression / Expansion)
	- * 'ACE8' "ACE 8-to-3" 8-to-3 IIGS ACE (Audio
	- * Compression / Expansion)
	- * 'MAC3' "MACE 3-to-1" 3-to-1 Macintosh Audio
	- * Compression / Expansion
	- * 'MAC6' "MACE 6-to-1" 6-to-1 Macintosh Audio
	- * Compression / Expansion
	- *
	- * A pstring is a "Pascal-style string", that is, "one byte followed
	- * by test bytes followed when needed by one pad byte. The total
	- * number of bytes in a pstring must be even. The pad byte is
	- * included when the number of text bytes is even, so the total of
	- * text bytes + one count byte + one pad byte will be even. This pad
	- * byte is not reflected in the count."
	- *
	- * As for how these compression algorithms work, your guess is as
	- * good as mine.
	- */
	- Uint32 compressionType;
	-#if 0
	- pstring compressionName;
	-#endif
	-} comm_t;
	-
	-
	-/*
	- * Read in a comm_t from disk. This makes this process safe regardless of
	- * the processor's byte order or how the comm_t structure is packed.
	- */
	-
	-static int read_comm_chunk(SDL_RWops rw, comm_t comm)
	-{
	- Uint8 sampleRate[10];
	-
	- /* skip reading the chunk ID, since it was already read at this point... */
	- comm->ckID = commID;
	-
	- if (SDL_RWread(rw, &comm->ckDataSize, sizeof (comm->ckDataSize), 1) != 1)
	- return(0);
	- comm->ckDataSize = SDL_SwapBE32(comm->ckDataSize);
	-
	- if (SDL_RWread(rw, &comm->numChannels, sizeof (comm->numChannels), 1) != 1)
	- return(0);
	- comm->numChannels = SDL_SwapBE16(comm->numChannels);
	-
	- if (SDL_RWread(rw, &comm->numSampleFrames,
	- sizeof (comm->numSampleFrames), 1) != 1)
	- return(0);
	- comm->numSampleFrames = SDL_SwapBE32(comm->numSampleFrames);
	-
	- if (SDL_RWread(rw, &comm->sampleSize, sizeof (comm->sampleSize), 1) != 1)
	- return(0);
	- comm->sampleSize = SDL_SwapBE16(comm->sampleSize);
	-
	- if (SDL_RWread(rw, sampleRate, sizeof (sampleRate), 1) != 1)
	- return(0);
	- comm->sampleRate = SANE_to_Uint32(sampleRate);
	-
	- if (comm->ckDataSize > sizeof(comm->numChannels)
	- + sizeof(comm->numSampleFrames)
	- + sizeof(comm->sampleSize)
	- + sizeof(sampleRate))
	- {
	- if (SDL_RWread(rw, &comm->compressionType,
	- sizeof (comm->compressionType), 1) != 1)
	- return(0);
	- comm->compressionType = SDL_SwapBE32(comm->compressionType);
	- } /* if */
	- else
	- {
	- comm->compressionType = noneID;
	- } /* else */
	-
	- return(1);
	-} /* read_comm_chunk */
	-
	-
	-
	-/*****************************************************************************
	- * The SSND chunk... *
	- *****************************************************************************/
	-
	-typedef struct
	-{
	- Uint32 ckID;
	- Uint32 ckDataSize;
	- Uint32 offset;
	- Uint32 blockSize;
	- /*
	- * Then, comm->numSampleFrames sample frames. (It's better to get the
	- * length from numSampleFrames than from ckDataSize.)
	- */
	-} ssnd_t;
	-
	-
	-static int read_ssnd_chunk(SDL_RWops rw, ssnd_t ssnd)
	-{
	- /* skip reading the chunk ID, since it was already read at this point... */
	- ssnd->ckID = ssndID;
	-
	- if (SDL_RWread(rw, &ssnd->ckDataSize, sizeof (ssnd->ckDataSize), 1) != 1)
	- return(0);
	- ssnd->ckDataSize = SDL_SwapBE32(ssnd->ckDataSize);
	-
	- if (SDL_RWread(rw, &ssnd->offset, sizeof (ssnd->offset), 1) != 1)
	- return(0);
	- ssnd->offset = SDL_SwapBE32(ssnd->offset);
	-
	- if (SDL_RWread(rw, &ssnd->blockSize, sizeof (ssnd->blockSize), 1) != 1)
	- return(0);
	- ssnd->blockSize = SDL_SwapBE32(ssnd->blockSize);
	-
	- /* Leave the SDL_RWops position indicator at the start of the samples */
	- if (SDL_RWseek(rw, (int) ssnd->offset, SEEK_CUR) == -1)
	- return(0);
	-
	- return(1);
	-} /* read_ssnd_chunk */
	-
	-
	-
	-/*****************************************************************************
	- * Normal, uncompressed aiff handler... *
	- *****************************************************************************/
	-
	-static Uint32 read_sample_fmt_normal(Sound_Sample *sample)
	-{
	- Uint32 retval;
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- aiff_t a = (aiff_t ) internal->decoder_private;
	- Uint32 max = (internal->buffer_size < (Uint32) a->bytesLeft) ?
	- internal->buffer_size : (Uint32) a->bytesLeft;
	-
	- assert(max > 0);
	-
	- /*
	- * We don't actually do any decoding, so we read the AIFF data
	- * directly into the internal buffer...
	- */
	- retval = SDL_RWread(internal->rw, internal->buffer, 1, max);
	-
	- a->bytesLeft -= retval;
	-
	- /* Make sure the read went smoothly... */
	- if ((retval == 0) \|\| (a->bytesLeft == 0))
	- sample->flags \|= SOUND_SAMPLEFLAG_EOF;
	-
	- else if (retval == -1)
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	-
	- /* (next call this EAGAIN may turn into an EOF or error.) */
	- else if (retval < internal->buffer_size)
	- sample->flags \|= SOUND_SAMPLEFLAG_EAGAIN;
	-
	- return(retval);
	-} /* read_sample_fmt_normal */
	-
	-
	-static int rewind_sample_fmt_normal(Sound_Sample *sample)
	-{
	- /* no-op. */
	- return(1);
	-} /* rewind_sample_fmt_normal */
	-
	-
	-static int seek_sample_fmt_normal(Sound_Sample *sample, Uint32 ms)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- aiff_t a = (aiff_t ) internal->decoder_private;
	- fmt_t *fmt = &a->fmt;
	- int offset = __Sound_convertMsToBytePos(&sample->actual, ms);
	- int pos = (int) (fmt->data_starting_offset + offset);
	- int rc = SDL_RWseek(internal->rw, pos, SEEK_SET);
	- BAIL_IF_MACRO(rc != pos, ERR_IO_ERROR, 0);
	- a->bytesLeft = fmt->total_bytes - offset;
	- return(1); /* success. */
	-} /* seek_sample_fmt_normal */
	-
	-
	-static void free_fmt_normal(fmt_t *fmt)
	-{
	- /* it's a no-op. */
	-} /* free_fmt_normal */
	-
	-
	-static int read_fmt_normal(SDL_RWops rw, fmt_t fmt)
	-{
	- /* (don't need to read more from the RWops...) */
	- fmt->free = free_fmt_normal;
	- fmt->read_sample = read_sample_fmt_normal;
	- fmt->rewind_sample = rewind_sample_fmt_normal;
	- fmt->seek_sample = seek_sample_fmt_normal;
	- return(1);
	-} /* read_fmt_normal */
	-
	-
	-
	-
	-/*****************************************************************************
	- * Everything else... *
	- *****************************************************************************/
	-
	-static int AIFF_init(void)
	-{
	- return(1); /* always succeeds. */
	-} /* AIFF_init */
	-
	-
	-static void AIFF_quit(void)
	-{
	- /* it's a no-op. */
	-} /* AIFF_quit */
	-
	-
	-/*
	- * Sample rate is encoded as an "80 bit IEEE Standard 754 floating point
	- * number (Standard Apple Numeric Environment [SANE] data type Extended)".
	- * Whose bright idea was that?
	- *
	- * This function was adapted from libsndfile, and while I do know a little
	- * bit about the IEEE floating point standard I don't pretend to fully
	- * understand this.
	- */
	-static Uint32 SANE_to_Uint32 (Uint8 *sanebuf)
	-{
	- /* Is the frequency outside of what we can represent with Uint32? */
	- if ( (sanebuf[0] & 0x80)
	- \|\| (sanebuf[0] <= 0x3F)
	- \|\| (sanebuf[0] > 0x40)
	- \|\| (sanebuf[0] == 0x40 && sanebuf[1] > 0x1C) )
	- return 0;
	-
	- return ((sanebuf[2] << 23) \| (sanebuf[3] << 15) \| (sanebuf[4] << 7)
	- \| (sanebuf[5] >> 1)) >> (29 - sanebuf[1]);
	-} /* SANE_to_Uint32 */
	-
	-
	-static int find_chunk(SDL_RWops *rw, Uint32 id)
	-{
	- Sint32 siz = 0;
	- Uint32 _id = 0;
	-
	- while (1)
	- {
	- BAIL_IF_MACRO(SDL_RWread(rw, &_id, sizeof (_id), 1) != 1, NULL, 0);
	- if (SDL_SwapLE32(_id) == id)
	- return(1);
	-
	- BAIL_IF_MACRO(SDL_RWread(rw, &siz, sizeof (siz), 1) != 1, NULL, 0);
	- siz = SDL_SwapBE32(siz);
	- assert(siz > 0);
	- BAIL_IF_MACRO(SDL_RWseek(rw, siz, SEEK_CUR) == -1, NULL, 0);
	- } /* while */
	-
	- return(0); /* shouldn't hit this, but just in case... */
	-} /* find_chunk */
	-
	-
	-static int read_fmt(SDL_RWops rw, comm_t c, fmt_t *fmt)
	-{
	- fmt->type = c->compressionType;
	-
	- /* if it's in this switch statement, we support the format. */
	- switch (fmt->type)
	- {
	- case noneID:
	- SNDDBG(("AIFF: Appears to be uncompressed audio.\n"));
	- return(read_fmt_normal(rw, fmt));
	-
	- /* add other types here. */
	-
	- default:
	- SNDDBG(("AIFF: Format %lu is unknown.\n",
	- (unsigned int) fmt->type));
	- BAIL_MACRO("AIFF: Unsupported format", 0);
	- } /* switch */
	-
	- assert(0); /* shouldn't hit this point. */
	- return(0);
	-} /* read_fmt */
	-
	-
	-static int AIFF_open(Sound_Sample sample, const char ext)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- SDL_RWops *rw = internal->rw;
	- Uint32 chunk_id;
	- int bytes_per_sample;
	- long pos;
	- comm_t c;
	- ssnd_t s;
	- aiff_t *a;
	-
	- BAIL_IF_MACRO(SDL_ReadLE32(rw) != formID, "AIFF: Not a FORM file.", 0);
	- SDL_ReadBE32(rw); /* throw the length away; we don't need it. */
	-
	- chunk_id = SDL_ReadLE32(rw);
	- BAIL_IF_MACRO(chunk_id != aiffID && chunk_id != aifcID,
	- "AIFF: Not an AIFF or AIFC file.", 0);
	-
	- /* Chunks may appear in any order, so we establish base camp here. */
	- pos = SDL_RWtell(rw);
	-
	- BAIL_IF_MACRO(!find_chunk(rw, commID), "AIFF: No common chunk.", 0);
	- BAIL_IF_MACRO(!read_comm_chunk(rw, &c),
	- "AIFF: Can't read common chunk.", 0);
	-
	- sample->actual.channels = (Uint8) c.numChannels;
	- sample->actual.rate = c.sampleRate;
	-
	- if (c.sampleSize <= 8)
	- {
	- sample->actual.format = AUDIO_S8;
	- bytes_per_sample = c.numChannels;
	- } /* if */
	- else if (c.sampleSize <= 16)
	- {
	- sample->actual.format = AUDIO_S16MSB;
	- bytes_per_sample = 2 * c.numChannels;
	- } /* if */
	- else
	- {
	- BAIL_MACRO("AIFF: Unsupported sample size.", 0);
	- } /* else */
	-
	- BAIL_IF_MACRO(c.sampleRate == 0, "AIFF: Unsupported sample rate.", 0);
	-
	- a = (aiff_t *) malloc(sizeof(aiff_t));
	- BAIL_IF_MACRO(a == NULL, ERR_OUT_OF_MEMORY, 0);
	-
	- if (!read_fmt(rw, &c, &(a->fmt)))
	- {
	- free(a);
	- return(0);
	- } /* if */
	-
	- SDL_RWseek(rw, pos, SEEK_SET); /* if the seek fails, let it go... */
	-
	- if (!find_chunk(rw, ssndID))
	- {
	- free(a);
	- BAIL_MACRO("AIFF: No sound data chunk.", 0);
	- } /* if */
	-
	- if (!read_ssnd_chunk(rw, &s))
	- {
	- free(a);
	- BAIL_MACRO("AIFF: Can't read sound data chunk.", 0);
	- } /* if */
	-
	- a->fmt.total_bytes = a->bytesLeft = bytes_per_sample * c.numSampleFrames;
	- a->fmt.data_starting_offset = SDL_RWtell(rw);
	- internal->decoder_private = (void *) a;
	-
	- sample->flags = SOUND_SAMPLEFLAG_CANSEEK;
	-
	- SNDDBG(("AIFF: Accepting data stream.\n"));
	- return(1); /* we'll handle this data. */
	-} /* AIFF_open */
	-
	-
	-static void AIFF_close(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- aiff_t a = (aiff_t ) internal->decoder_private;
	- a->fmt.free(&(a->fmt));
	- free(a);
	-} /* AIFF_close */
	-
	-
	-static Uint32 AIFF_read(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- aiff_t a = (aiff_t ) internal->decoder_private;
	- return(a->fmt.read_sample(sample));
	-} /* AIFF_read */
	-
	-
	-static int AIFF_rewind(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- aiff_t a = (aiff_t ) internal->decoder_private;
	- fmt_t *fmt = &a->fmt;
	- int rc = SDL_RWseek(internal->rw, fmt->data_starting_offset, SEEK_SET);
	- BAIL_IF_MACRO(rc != fmt->data_starting_offset, ERR_IO_ERROR, 0);
	- a->bytesLeft = fmt->total_bytes;
	- return(fmt->rewind_sample(sample));
	-} /* AIFF_rewind */
	-
	-
	-static int AIFF_seek(Sound_Sample *sample, Uint32 ms)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- aiff_t a = (aiff_t ) internal->decoder_private;
	- return(a->fmt.seek_sample(sample, ms));
	-} /* AIFF_seek */
	-
	-#endif /* SOUND_SUPPORTS_AIFF */
	-
	-/* end of aiff.c ... */
	-
	diff --git a/util/sdl/sound/decoders/au.c b/util/sdl/sound/decoders/au.c
	deleted file mode 100644
	index ab0ff9f7..00000000
	--- a/util/sdl/sound/decoders/au.c
	+++ /dev/null
	@@ -1,376 +0,0 @@
	-/*
	- * SDL_sound -- An abstract sound format decoding API.
	- * Copyright (C) 2001 Ryan C. Gordon.
	- *
	- * This library is free software; you can redistribute it and/or
	- * modify it under the terms of the GNU Lesser General Public
	- * License as published by the Free Software Foundation; either
	- * version 2.1 of the License, or (at your option) any later version.
	- *
	- * This library is distributed in the hope that it will be useful,
	- * but WITHOUT ANY WARRANTY; without even the implied warranty of
	- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	- * Lesser General Public License for more details.
	- *
	- * You should have received a copy of the GNU Lesser General Public
	- * License along with this library; if not, write to the Free Software
	- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
	- */
	-
	-/*
	- * Sun/NeXT .au decoder for SDL_sound.
	- * Formats supported: 8 and 16 bit linear PCM, 8 bit µ-law.
	- * Files without valid header are assumed to be 8 bit µ-law, 8kHz, mono.
	- *
	- * Please see the file COPYING in the source's root directory.
	- *
	- * This file written by Mattias Engdegård. (f91-men@nada.kth.se)
	- */
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#ifdef SOUND_SUPPORTS_AU
	-
	-#include <stdio.h>
	-#include <stdlib.h>
	-#include <string.h>
	-
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-static int AU_init(void);
	-static void AU_quit(void);
	-static int AU_open(Sound_Sample sample, const char ext);
	-static void AU_close(Sound_Sample *sample);
	-static Uint32 AU_read(Sound_Sample *sample);
	-static int AU_rewind(Sound_Sample *sample);
	-static int AU_seek(Sound_Sample *sample, Uint32 ms);
	-
	-/*
	- * Sometimes the extension ".snd" is used for these files (mostly on the NeXT),
	- * and the magic number comes from this. However it may clash with other
	- * formats and is somewhat of an anachronism, so only .au is used here.
	- */
	-static const char *extensions_au[] = { "AU", NULL };
	-const Sound_DecoderFunctions __Sound_DecoderFunctions_AU =
	-{
	- {
	- extensions_au,
	- "Sun/NeXT audio file format",
	- "Mattias Engdegård <f91-men@nada.kth.se>",
	- "http://www.icculus.org/SDL_sound/"
	- },
	-
	- AU_init, /* init() method */
	- AU_quit, /* quit() method */
	- AU_open, /* open() method */
	- AU_close, /* close() method */
	- AU_read, /* read() method */
	- AU_rewind, /* rewind() method */
	- AU_seek /* seek() method */
	-};
	-
	-/* no init/deinit needed */
	-static int AU_init(void)
	-{
	- return(1);
	-} /* AU_init */
	-
	-static void AU_quit(void)
	-{
	- /* no-op. */
	-} /* AU_quit */
	-
	-struct au_file_hdr
	-{
	- Uint32 magic;
	- Uint32 hdr_size;
	- Uint32 data_size;
	- Uint32 encoding;
	- Uint32 sample_rate;
	- Uint32 channels;
	-};
	-
	-#define HDR_SIZE 24
	-
	-enum
	-{
	- AU_ENC_ULAW_8 = 1, /* 8-bit ISDN µ-law */
	- AU_ENC_LINEAR_8 = 2, /* 8-bit linear PCM */
	- AU_ENC_LINEAR_16 = 3, /* 16-bit linear PCM */
	-
	- /* the rest are unsupported (I have never seen them in the wild) */
	- AU_ENC_LINEAR_24 = 4, /* 24-bit linear PCM */
	- AU_ENC_LINEAR_32 = 5, /* 32-bit linear PCM */
	- AU_ENC_FLOAT = 6, /* 32-bit IEEE floating point */
	- AU_ENC_DOUBLE = 7, /* 64-bit IEEE floating point */
	- /* more Sun formats, not supported either */
	- AU_ENC_ADPCM_G721 = 23,
	- AU_ENC_ADPCM_G722 = 24,
	- AU_ENC_ADPCM_G723_3 = 25,
	- AU_ENC_ADPCM_G723_5 = 26,
	- AU_ENC_ALAW_8 = 27
	-};
	-
	-struct audec
	-{
	- Uint32 total;
	- Uint32 remaining;
	- Uint32 start_offset;
	- int encoding;
	-};
	-
	-
	-/*
	- * Read in the AU header from disk. This makes this process safe
	- * regardless of the processor's byte order or how the au_file_hdr
	- * structure is packed.
	- */
	-static int read_au_header(SDL_RWops rw, struct au_file_hdr hdr)
	-{
	- if (SDL_RWread(rw, &hdr->magic, sizeof (hdr->magic), 1) != 1)
	- return(0);
	- hdr->magic = SDL_SwapBE32(hdr->magic);
	-
	- if (SDL_RWread(rw, &hdr->hdr_size, sizeof (hdr->hdr_size), 1) != 1)
	- return(0);
	- hdr->hdr_size = SDL_SwapBE32(hdr->hdr_size);
	-
	- if (SDL_RWread(rw, &hdr->data_size, sizeof (hdr->data_size), 1) != 1)
	- return(0);
	- hdr->data_size = SDL_SwapBE32(hdr->data_size);
	-
	- if (SDL_RWread(rw, &hdr->encoding, sizeof (hdr->encoding), 1) != 1)
	- return(0);
	- hdr->encoding = SDL_SwapBE32(hdr->encoding);
	-
	- if (SDL_RWread(rw, &hdr->sample_rate, sizeof (hdr->sample_rate), 1) != 1)
	- return(0);
	- hdr->sample_rate = SDL_SwapBE32(hdr->sample_rate);
	-
	- if (SDL_RWread(rw, &hdr->channels, sizeof (hdr->channels), 1) != 1)
	- return(0);
	- hdr->channels = SDL_SwapBE32(hdr->channels);
	-
	- return(1);
	-} /* read_au_header */
	-
	-
	-#define AU_MAGIC 0x2E736E64 /* ".snd", in ASCII (bigendian number) */
	-
	-static int AU_open(Sound_Sample sample, const char ext)
	-{
	- Sound_SampleInternal *internal = sample->opaque;
	- SDL_RWops *rw = internal->rw;
	- int skip, hsize, i;
	- struct au_file_hdr hdr;
	- struct audec *dec;
	- char c;
	-
	- /* read_au_header() will do byte order swapping. */
	- BAIL_IF_MACRO(!read_au_header(rw, &hdr), "AU: bad header", 0);
	-
	- dec = malloc(sizeof *dec);
	- BAIL_IF_MACRO(dec == NULL, ERR_OUT_OF_MEMORY, 0);
	- internal->decoder_private = dec;
	-
	- if (hdr.magic == AU_MAGIC)
	- {
	- /* valid magic */
	- dec->encoding = hdr.encoding;
	- switch(dec->encoding)
	- {
	- case AU_ENC_ULAW_8:
	- /* Convert 8-bit µ-law to 16-bit linear on the fly. This is
	- slightly wasteful if the audio driver must convert them
	- back, but µ-law only devices are rare (mostly _old_ Suns) */
	- sample->actual.format = AUDIO_S16SYS;
	- break;
	-
	- case AU_ENC_LINEAR_8:
	- sample->actual.format = AUDIO_S8;
	- break;
	-
	- case AU_ENC_LINEAR_16:
	- sample->actual.format = AUDIO_S16MSB;
	- break;
	-
	- default:
	- free(dec);
	- BAIL_MACRO("AU: Unsupported .au encoding", 0);
	- } /* switch */
	-
	- sample->actual.rate = hdr.sample_rate;
	- sample->actual.channels = hdr.channels;
	- dec->remaining = hdr.data_size;
	- hsize = hdr.hdr_size;
	-
	- /* skip remaining part of header (input may be unseekable) */
	- for (i = HDR_SIZE; i < hsize; i++)
	- {
	- if (SDL_RWread(rw, &c, 1, 1) != 1)
	- {
	- free(dec);
	- BAIL_MACRO(ERR_IO_ERROR, 0);
	- } /* if */
	- } /* for */
	- } /* if */
	-
	- else if (__Sound_strcasecmp(ext, "au") == 0)
	- {
	- /*
	- * A number of files in the wild have the .au extension but no valid
	- * header; these are traditionally assumed to be 8kHz µ-law. Handle
	- * them here only if the extension is recognized.
	- */
	-
	- SNDDBG(("AU: Invalid header, assuming raw 8kHz µ-law.\n"));
	- /* if seeking fails, we lose 24 samples. big deal */
	- SDL_RWseek(rw, -HDR_SIZE, SEEK_CUR);
	- dec->encoding = AU_ENC_ULAW_8;
	- dec->remaining = (Uint32)-1; /* no limit */
	- sample->actual.format = AUDIO_S16SYS;
	- sample->actual.rate = 8000;
	- sample->actual.channels = 1;
	- } /* else if */
	-
	- else
	- {
	- free(dec);
	- BAIL_MACRO("AU: Not an .AU stream.", 0);
	- } /* else */
	-
	- sample->flags = SOUND_SAMPLEFLAG_CANSEEK;
	- dec->total = dec->remaining;
	- dec->start_offset = SDL_RWtell(rw);
	-
	- SNDDBG(("AU: Accepting data stream.\n"));
	- return(1);
	-} /* AU_open */
	-
	-
	-static void AU_close(Sound_Sample *sample)
	-{
	- Sound_SampleInternal *internal = sample->opaque;
	- free(internal->decoder_private);
	-} /* AU_close */
	-
	-
	-/* table to convert from µ-law encoding to signed 16-bit samples,
	- generated by a throwaway perl script */
	-static Sint16 ulaw_to_linear[256] = {
	- -32124,-31100,-30076,-29052,-28028,-27004,-25980,-24956,
	- -23932,-22908,-21884,-20860,-19836,-18812,-17788,-16764,
	- -15996,-15484,-14972,-14460,-13948,-13436,-12924,-12412,
	- -11900,-11388,-10876,-10364, -9852, -9340, -8828, -8316,
	- -7932, -7676, -7420, -7164, -6908, -6652, -6396, -6140,
	- -5884, -5628, -5372, -5116, -4860, -4604, -4348, -4092,
	- -3900, -3772, -3644, -3516, -3388, -3260, -3132, -3004,
	- -2876, -2748, -2620, -2492, -2364, -2236, -2108, -1980,
	- -1884, -1820, -1756, -1692, -1628, -1564, -1500, -1436,
	- -1372, -1308, -1244, -1180, -1116, -1052, -988, -924,
	- -876, -844, -812, -780, -748, -716, -684, -652,
	- -620, -588, -556, -524, -492, -460, -428, -396,
	- -372, -356, -340, -324, -308, -292, -276, -260,
	- -244, -228, -212, -196, -180, -164, -148, -132,
	- -120, -112, -104, -96, -88, -80, -72, -64,
	- -56, -48, -40, -32, -24, -16, -8, 0,
	- 32124, 31100, 30076, 29052, 28028, 27004, 25980, 24956,
	- 23932, 22908, 21884, 20860, 19836, 18812, 17788, 16764,
	- 15996, 15484, 14972, 14460, 13948, 13436, 12924, 12412,
	- 11900, 11388, 10876, 10364, 9852, 9340, 8828, 8316,
	- 7932, 7676, 7420, 7164, 6908, 6652, 6396, 6140,
	- 5884, 5628, 5372, 5116, 4860, 4604, 4348, 4092,
	- 3900, 3772, 3644, 3516, 3388, 3260, 3132, 3004,
	- 2876, 2748, 2620, 2492, 2364, 2236, 2108, 1980,
	- 1884, 1820, 1756, 1692, 1628, 1564, 1500, 1436,
	- 1372, 1308, 1244, 1180, 1116, 1052, 988, 924,
	- 876, 844, 812, 780, 748, 716, 684, 652,
	- 620, 588, 556, 524, 492, 460, 428, 396,
	- 372, 356, 340, 324, 308, 292, 276, 260,
	- 244, 228, 212, 196, 180, 164, 148, 132,
	- 120, 112, 104, 96, 88, 80, 72, 64,
	- 56, 48, 40, 32, 24, 16, 8, 0
	-};
	-
	-
	-static Uint32 AU_read(Sound_Sample *sample)
	-{
	- int ret;
	- Sound_SampleInternal *internal = sample->opaque;
	- struct audec *dec = internal->decoder_private;
	- int maxlen;
	- Uint8 *buf;
	-
	- maxlen = internal->buffer_size;
	- buf = internal->buffer;
	- if (dec->encoding == AU_ENC_ULAW_8)
	- {
	- /* We read µ-law samples into the second half of the buffer, so
	- we can expand them to 16-bit samples afterwards */
	- maxlen >>= 1;
	- buf += maxlen;
	- } /* if */
	-
	- if (maxlen > dec->remaining)
	- maxlen = dec->remaining;
	- ret = SDL_RWread(internal->rw, buf, 1, maxlen);
	- if (ret == 0)
	- sample->flags \|= SOUND_SAMPLEFLAG_EOF;
	- else if (ret == -1)
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- else
	- {
	- dec->remaining -= ret;
	- if (ret < maxlen)
	- sample->flags \|= SOUND_SAMPLEFLAG_EAGAIN;
	-
	- if (dec->encoding == AU_ENC_ULAW_8)
	- {
	- int i;
	- Sint16 *dst = internal->buffer;
	- for (i = 0; i < ret; i++)
	- dst[i] = ulaw_to_linear[buf[i]];
	- ret <<= 1; /* return twice as much as read */
	- } /* if */
	- } /* else */
	-
	- return(ret);
	-} /* AU_read */
	-
	-
	-static int AU_rewind(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- struct audec dec = (struct audec ) internal->decoder_private;
	- int rc = SDL_RWseek(internal->rw, dec->start_offset, SEEK_SET);
	- BAIL_IF_MACRO(rc != dec->start_offset, ERR_IO_ERROR, 0);
	- dec->remaining = dec->total;
	- return(1);
	-} /* AU_rewind */
	-
	-
	-static int AU_seek(Sound_Sample *sample, Uint32 ms)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- struct audec dec = (struct audec ) internal->decoder_private;
	- int offset = __Sound_convertMsToBytePos(&sample->actual, ms);
	- int rc;
	- int pos;
	-
	- if (dec->encoding == AU_ENC_ULAW_8)
	- offset >>= 1; /* halve the byte offset for compression. */
	-
	- pos = (int) (dec->start_offset + offset);
	- rc = SDL_RWseek(internal->rw, pos, SEEK_SET);
	- BAIL_IF_MACRO(rc != pos, ERR_IO_ERROR, 0);
	- dec->remaining = dec->total - offset;
	- return(1);
	-} /* AU_seek */
	-
	-#endif /* SOUND_SUPPORTS_AU */
	-
	diff --git a/util/sdl/sound/decoders/flac.c b/util/sdl/sound/decoders/flac.c
	deleted file mode 100644
	index 54aebc0c..00000000
	--- a/util/sdl/sound/decoders/flac.c
	+++ /dev/null
	@@ -1,566 +0,0 @@
	-/*
	- * SDL_sound -- An abstract sound format decoding API.
	- * Copyright (C) 2001 Ryan C. Gordon.
	- *
	- * This library is free software; you can redistribute it and/or
	- * modify it under the terms of the GNU Lesser General Public
	- * License as published by the Free Software Foundation; either
	- * version 2.1 of the License, or (at your option) any later version.
	- *
	- * This library is distributed in the hope that it will be useful,
	- * but WITHOUT ANY WARRANTY; without even the implied warranty of
	- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	- * Lesser General Public License for more details.
	- *
	- * You should have received a copy of the GNU Lesser General Public
	- * License along with this library; if not, write to the Free Software
	- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
	- */
	-
	-/*
	- * FLAC decoder for SDL_sound.
	- *
	- * This driver handles FLAC audio, that is to say the Free Lossless Audio
	- * Codec. It depends on libFLAC for decoding, which can be grabbed from:
	- * http://flac.sourceforge.net
	- *
	- * Please see the file COPYING in the source's root directory.
	- *
	- * This file written by Torbjörn Andersson. (d91tan@Update.UU.SE)
	- */
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#ifdef SOUND_SUPPORTS_FLAC
	-
	-#include <stdio.h>
	-#include <stdlib.h>
	-#include <string.h>
	-
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-#include <FLAC/export.h>
	-
	-/* FLAC 1.1.3 has FLAC_API_VERSION_CURRENT == 8 */
	-#if !defined(FLAC_API_VERSION_CURRENT) \|\| FLAC_API_VERSION_CURRENT < 8
	-#define LEGACY_FLAC
	-#else
	-#undef LEGACY_FLAC
	-#endif
	-
	-#ifdef LEGACY_FLAC
	-#include <FLAC/seekable_stream_decoder.h>
	-
	-#define D_END_OF_STREAM FLAC__SEEKABLE_STREAM_DECODER_END_OF_STREAM
	-
	-#define d_new() FLAC__seekable_stream_decoder_new()
	-#define d_init(x) FLAC__seekable_stream_decoder_init(x)
	-#define d_process_metadata(x) FLAC__seekable_stream_decoder_process_until_end_of_metadata(x)
	-#define d_process_one_frame(x) FLAC__seekable_stream_decoder_process_single(x)
	-#define d_get_state(x) FLAC__seekable_stream_decoder_get_state(x)
	-#define d_finish(x) FLAC__seekable_stream_decoder_finish(x)
	-#define d_delete(x) FLAC__seekable_stream_decoder_delete(x)
	-#define d_set_read_callback(x, y) FLAC__seekable_stream_decoder_set_read_callback(x, y)
	-#define d_set_write_callback(x, y) FLAC__seekable_stream_decoder_set_write_callback(x, y)
	-#define d_set_metadata_callback(x, y) FLAC__seekable_stream_decoder_set_metadata_callback(x, y)
	-#define d_set_error_callback(x, y) FLAC__seekable_stream_decoder_set_error_callback(x, y)
	-#define d_set_client_data(x, y) FLAC__seekable_stream_decoder_set_client_data(x, y)
	-
	-typedef FLAC__SeekableStreamDecoder decoder_t;
	-typedef FLAC__SeekableStreamDecoderReadStatus d_read_status_t;
	-
	-#define D_SEEK_STATUS_OK FLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_OK
	-#define D_SEEK_STATUS_ERROR FLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_ERROR
	-#define D_TELL_STATUS_OK FLAC__SEEKABLE_STREAM_DECODER_TELL_STATUS_OK
	-#define D_TELL_STATUS_ERROR FLAC__SEEKABLE_STREAM_DECODER_TELL_STATUS_ERROR
	-#define D_LENGTH_STATUS_OK FLAC__SEEKABLE_STREAM_DECODER_LENGTH_STATUS_OK
	-#define D_LENGTH_STATUS_ERROR FLAC__SEEKABLE_STREAM_DECODER_LENGTH_STATUS_ERROR
	-
	-#define d_set_seek_callback(x, y) FLAC__seekable_stream_decoder_set_seek_callback(x, y)
	-#define d_set_tell_callback(x, y) FLAC__seekable_stream_decoder_set_tell_callback(x, y)
	-#define d_set_length_callback(x, y) FLAC__seekable_stream_decoder_set_length_callback(x, y)
	-#define d_set_eof_callback(x, y) FLAC__seekable_stream_decoder_set_eof_callback(x, y)
	-#define d_seek_absolute(x, y) FLAC__seekable_stream_decoder_seek_absolute(x, y)
	-
	-typedef FLAC__SeekableStreamDecoderSeekStatus d_seek_status_t;
	-typedef FLAC__SeekableStreamDecoderTellStatus d_tell_status_t;
	-typedef FLAC__SeekableStreamDecoderLengthStatus d_length_status_t;
	-#else
	-#include <FLAC/stream_decoder.h>
	-
	-#define D_END_OF_STREAM FLAC__STREAM_DECODER_END_OF_STREAM
	-
	-#define d_new() FLAC__stream_decoder_new()
	-#define d_process_metadata(x) FLAC__stream_decoder_process_until_end_of_metadata(x)
	-#define d_process_one_frame(x) FLAC__stream_decoder_process_single(x)
	-#define d_get_state(x) FLAC__stream_decoder_get_state(x)
	-#define d_finish(x) FLAC__stream_decoder_finish(x)
	-#define d_delete(x) FLAC__stream_decoder_delete(x)
	-
	-typedef FLAC__StreamDecoder decoder_t;
	-typedef FLAC__StreamDecoderReadStatus d_read_status_t;
	-
	-#define D_SEEK_STATUS_OK FLAC__STREAM_DECODER_SEEK_STATUS_OK
	-#define D_SEEK_STATUS_ERROR FLAC__STREAM_DECODER_SEEK_STATUS_ERROR
	-#define D_TELL_STATUS_OK FLAC__STREAM_DECODER_TELL_STATUS_OK
	-#define D_TELL_STATUS_ERROR FLAC__STREAM_DECODER_TELL_STATUS_ERROR
	-#define D_LENGTH_STATUS_OK FLAC__STREAM_DECODER_LENGTH_STATUS_OK
	-#define D_LENGTH_STATUS_ERROR FLAC__STREAM_DECODER_LENGTH_STATUS_ERROR
	-
	-#define d_seek_absolute(x, y) FLAC__stream_decoder_seek_absolute(x, y)
	-
	-typedef FLAC__StreamDecoderSeekStatus d_seek_status_t;
	-typedef FLAC__StreamDecoderTellStatus d_tell_status_t;
	-typedef FLAC__StreamDecoderLengthStatus d_length_status_t;
	-#endif
	-
	-#define D_WRITE_CONTINUE FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE
	-#define D_READ_END_OF_STREAM FLAC__STREAM_DECODER_READ_STATUS_END_OF_STREAM
	-#define D_READ_ABORT FLAC__STREAM_DECODER_READ_STATUS_ABORT
	-#define D_READ_CONTINUE FLAC__STREAM_DECODER_READ_STATUS_CONTINUE
	-
	-#define d_error_status_string FLAC__StreamDecoderErrorStatusString
	-
	-typedef FLAC__StreamDecoderErrorStatus d_error_status_t;
	-typedef FLAC__StreamMetadata d_metadata_t;
	-typedef FLAC__StreamDecoderWriteStatus d_write_status_t;
	-
	-
	-static int FLAC_init(void);
	-static void FLAC_quit(void);
	-static int FLAC_open(Sound_Sample sample, const char ext);
	-static void FLAC_close(Sound_Sample *sample);
	-static Uint32 FLAC_read(Sound_Sample *sample);
	-static int FLAC_rewind(Sound_Sample *sample);
	-static int FLAC_seek(Sound_Sample *sample, Uint32 ms);
	-
	-static const char *extensions_flac[] = { "FLAC", "FLA", NULL };
	-
	-const Sound_DecoderFunctions __Sound_DecoderFunctions_FLAC =
	-{
	- {
	- extensions_flac,
	- "Free Lossless Audio Codec",
	- "Torbjörn Andersson <d91tan@Update.UU.SE>",
	- "http://flac.sourceforge.net/"
	- },
	-
	- FLAC_init, /* init() method */
	- FLAC_quit, /* quit() method */
	- FLAC_open, /* open() method */
	- FLAC_close, /* close() method */
	- FLAC_read, /* read() method */
	- FLAC_rewind, /* rewind() method */
	- FLAC_seek /* seek() method */
	-};
	-
	- /* This is what we store in our internal->decoder_private field. */
	-typedef struct
	-{
	- decoder_t *decoder;
	- SDL_RWops *rw;
	- Sound_Sample *sample;
	- Uint32 frame_size;
	- Uint8 is_flac;
	- Uint32 stream_length;
	-} flac_t;
	-
	-
	-static void free_flac(flac_t *f)
	-{
	- d_finish(f->decoder);
	- d_delete(f->decoder);
	- free(f);
	-} /* free_flac */
	-
	-
	-#ifdef LEGACY_FLAC
	-static d_read_status_t read_callback(
	- const decoder_t *decoder, FLAC__byte buffer[],
	- unsigned int bytes, void client_data)
	-#else
	-static d_read_status_t read_callback(
	- const decoder_t *decoder, FLAC__byte buffer[],
	- size_t bytes, void client_data)
	-#endif
	-{
	- flac_t f = (flac_t ) client_data;
	- Uint32 retval;
	-
	- retval = SDL_RWread(f->rw, (Uint8 ) buffer, 1, bytes);
	-
	- if (retval == 0)
	- {
	- *bytes = 0;
	- f->sample->flags \|= SOUND_SAMPLEFLAG_EOF;
	- return(D_READ_END_OF_STREAM);
	- } /* if */
	-
	- if (retval == -1)
	- {
	- *bytes = 0;
	- f->sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- return(D_READ_ABORT);
	- } /* if */
	-
	- if (retval < *bytes)
	- {
	- *bytes = retval;
	- f->sample->flags \|= SOUND_SAMPLEFLAG_EAGAIN;
	- } /* if */
	-
	- return(D_READ_CONTINUE);
	-} /* read_callback */
	-
	-
	-static d_write_status_t write_callback(
	- const decoder_t decoder, const FLAC__Frame frame,
	- const FLAC__int32 * const buffer[],
	- void *client_data)
	-{
	- flac_t f = (flac_t ) client_data;
	- Uint32 i, j;
	- Uint32 sample;
	- Uint8 *dst;
	-
	- f->frame_size = frame->header.channels * frame->header.blocksize
	- * frame->header.bits_per_sample / 8;
	-
	- if (f->frame_size > f->sample->buffer_size)
	- Sound_SetBufferSize(f->sample, f->frame_size);
	-
	- dst = f->sample->buffer;
	-
	- /* If the sample is neither exactly 8-bit nor 16-bit, it will have to
	- * be converted. Unfortunately the buffer is read-only, so we either
	- * have to check for each sample, or make a copy of the buffer. I'm
	- * not sure which way is best, so I've arbitrarily picked the former.
	- */
	- if (f->sample->actual.format == AUDIO_S8)
	- {
	- for (i = 0; i < frame->header.blocksize; i++)
	- for (j = 0; j < frame->header.channels; j++)
	- {
	- sample = buffer[j][i];
	- if (frame->header.bits_per_sample < 8)
	- sample <<= (8 - frame->header.bits_per_sample);
	- *dst++ = sample & 0x00ff;
	- } /* for */
	- } /* if */
	- else
	- {
	- for (i = 0; i < frame->header.blocksize; i++)
	- for (j = 0; j < frame->header.channels; j++)
	- {
	- sample = buffer[j][i];
	- if (frame->header.bits_per_sample < 16)
	- sample <<= (16 - frame->header.bits_per_sample);
	- else if (frame->header.bits_per_sample > 16)
	- sample >>= (frame->header.bits_per_sample - 16);
	- *dst++ = (sample & 0xff00) >> 8;
	- *dst++ = sample & 0x00ff;
	- } /* for */
	- } /* else */
	-
	- return(D_WRITE_CONTINUE);
	-} /* write_callback */
	-
	-
	-static void metadata_callback(
	- const decoder_t *decoder,
	- const d_metadata_t *metadata,
	- void *client_data)
	-{
	- flac_t f = (flac_t ) client_data;
	-
	- SNDDBG(("FLAC: Metadata callback.\n"));
	-
	- /* There are several kinds of metadata, but STREAMINFO is the only
	- * one that always has to be there.
	- */
	- if (metadata->type == FLAC__METADATA_TYPE_STREAMINFO)
	- {
	- SNDDBG(("FLAC: Metadata is streaminfo.\n"));
	-
	- f->is_flac = 1;
	- f->sample->actual.channels = metadata->data.stream_info.channels;
	- f->sample->actual.rate = metadata->data.stream_info.sample_rate;
	-
	- if (metadata->data.stream_info.bits_per_sample > 8)
	- f->sample->actual.format = AUDIO_S16MSB;
	- else
	- f->sample->actual.format = AUDIO_S8;
	- } /* if */
	-} /* metadata_callback */
	-
	-
	-static void error_callback(
	- const decoder_t *decoder,
	- d_error_status_t status,
	- void *client_data)
	-{
	- flac_t f = (flac_t ) client_data;
	-
	- __Sound_SetError(d_error_status_string[status]);
	- f->sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	-} /* error_callback */
	-
	-
	-static d_seek_status_t seek_callback(
	- const decoder_t *decoder,
	- FLAC__uint64 absolute_byte_offset,
	- void *client_data)
	-{
	- flac_t f = (flac_t ) client_data;
	-
	- if (SDL_RWseek(f->rw, absolute_byte_offset, SEEK_SET) >= 0)
	- {
	- return(D_SEEK_STATUS_OK);
	- } /* if */
	-
	- return(D_SEEK_STATUS_ERROR);
	-} /* seek_callback*/
	-
	-
	-static d_tell_status_t tell_callback(
	- const decoder_t *decoder,
	- FLAC__uint64 *absolute_byte_offset,
	- void *client_data)
	-{
	- flac_t f = (flac_t ) client_data;
	- int pos;
	-
	- pos = SDL_RWtell(f->rw);
	-
	- if (pos < 0)
	- {
	- return(D_TELL_STATUS_ERROR);
	- } /* if */
	-
	- *absolute_byte_offset = pos;
	- return(D_TELL_STATUS_OK);
	-} /* tell_callback */
	-
	-
	-static d_length_status_t length_callback(
	- const decoder_t *decoder,
	- FLAC__uint64 *stream_length,
	- void *client_data)
	-{
	- flac_t f = (flac_t ) client_data;
	-
	- if (f->sample->flags & SOUND_SAMPLEFLAG_CANSEEK)
	- {
	- *stream_length = f->stream_length;
	- return(D_LENGTH_STATUS_OK);
	- } /* if */
	-
	- return(D_LENGTH_STATUS_ERROR);
	-} /* length_callback */
	-
	-
	-static FLAC__bool eof_callback(
	- const decoder_t *decoder,
	- void *client_data)
	-{
	- flac_t f = (flac_t ) client_data;
	- int pos;
	-
	- /* Maybe we could check for SOUND_SAMPLEFLAG_EOF here instead? */
	- pos = SDL_RWtell(f->rw);
	-
	- if (pos >= 0 && pos >= f->stream_length)
	- {
	- return(true);
	- } /* if */
	-
	- return(false);
	-} /* eof_callback */
	-
	-
	-static int FLAC_init(void)
	-{
	- return(1); /* always succeeds. */
	-} /* FLAC_init */
	-
	-
	-static void FLAC_quit(void)
	-{
	- /* it's a no-op. */
	-} /* FLAC_quit */
	-
	-
	-#define FLAC_MAGIC 0x43614C66 /* "fLaC" in ASCII. */
	-
	-static int FLAC_open(Sound_Sample sample, const char ext)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- SDL_RWops *rw = internal->rw;
	- decoder_t *decoder;
	- flac_t *f;
	- int i;
	- int has_extension = 0;
	- Uint32 pos;
	-
	- /*
	- * If the extension is "flac", we'll believe that this is really meant
	- * to be a FLAC stream, and will try to grok it from existing metadata.
	- * metadata searching can be a very expensive operation, however, so
	- * unless the user swears that it is a FLAC stream through the extension,
	- * we decide what to do based on the existance of a 32-bit magic number.
	- */
	- for (i = 0; extensions_flac[i] != NULL; i++)
	- {
	- if (__Sound_strcasecmp(ext, extensions_flac[i]) == 0)
	- {
	- has_extension = 1;
	- break;
	- } /* if */
	- } /* for */
	-
	- if (!has_extension)
	- {
	- int rc;
	- Uint32 flac_magic = SDL_ReadLE32(rw);
	- BAIL_IF_MACRO(flac_magic != FLAC_MAGIC, "FLAC: Not a FLAC stream.", 0);
	-
	- /* move back over magic number for metadata scan... */
	- rc = SDL_RWseek(internal->rw, -sizeof (flac_magic), SEEK_CUR);
	- BAIL_IF_MACRO(rc < 0, ERR_IO_ERROR, 0);
	- } /* if */
	-
	- f = (flac_t *) malloc(sizeof (flac_t));
	- BAIL_IF_MACRO(f == NULL, ERR_OUT_OF_MEMORY, 0);
	-
	- decoder = d_new();
	- if (decoder == NULL)
	- {
	- free(f);
	- BAIL_MACRO(ERR_OUT_OF_MEMORY, 0);
	- } /* if */
	-
	-#ifdef LEGACY_FLAC
	- d_set_read_callback(decoder, read_callback);
	- d_set_write_callback(decoder, write_callback);
	- d_set_metadata_callback(decoder, metadata_callback);
	- d_set_error_callback(decoder, error_callback);
	- d_set_seek_callback(decoder, seek_callback);
	- d_set_tell_callback(decoder, tell_callback);
	- d_set_length_callback(decoder, length_callback);
	- d_set_eof_callback(decoder, eof_callback);
	-
	- d_set_client_data(decoder, f);
	-#endif
	-
	- f->rw = internal->rw;
	- f->sample = sample;
	- f->decoder = decoder;
	- f->sample->actual.format = 0;
	- f->is_flac = 0 /* !!! FIXME: should be "has_extension", not "0". */;
	-
	- internal->decoder_private = f;
	- /* really should check the init return value here: */
	-#ifdef LEGACY_FLAC
	- d_init(decoder);
	-#else
	- FLAC__stream_decoder_init_stream(decoder, read_callback, seek_callback,
	- tell_callback, length_callback,
	- eof_callback, write_callback,
	- metadata_callback, error_callback, f);
	-#endif
	-
	- sample->flags = SOUND_SAMPLEFLAG_NONE;
	-
	- pos = SDL_RWtell(f->rw);
	- if (SDL_RWseek(f->rw, 0, SEEK_END) > 0)
	- {
	- f->stream_length = SDL_RWtell(f->rw);
	- if (SDL_RWseek(f->rw, pos, SEEK_SET) == -1)
	- {
	- free_flac(f);
	- BAIL_MACRO(ERR_IO_ERROR, 0);
	- } /* if */
	- sample->flags = SOUND_SAMPLEFLAG_CANSEEK;
	- } /* if */
	-
	- /*
	- * If we are not sure this is a FLAC stream, check for the STREAMINFO
	- * metadata block. If not, we'd have to peek at the first audio frame
	- * and get the sound format from there, but that is not yet
	- * implemented.
	- */
	- if (!f->is_flac)
	- {
	- d_process_metadata(decoder);
	-
	- /* Still not FLAC? Give up. */
	- if (!f->is_flac)
	- {
	- free_flac(f);
	- BAIL_MACRO("FLAC: No metadata found. Not a FLAC stream?", 0);
	- } /* if */
	- } /* if */
	-
	- SNDDBG(("FLAC: Accepting data stream.\n"));
	- return(1);
	-} /* FLAC_open */
	-
	-
	-static void FLAC_close(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- flac_t f = (flac_t ) internal->decoder_private;
	-
	- free_flac(f);
	-} /* FLAC_close */
	-
	-
	-static Uint32 FLAC_read(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- flac_t f = (flac_t ) internal->decoder_private;
	- Uint32 len;
	-
	- if (!d_process_one_frame(f->decoder))
	- {
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- BAIL_MACRO("FLAC: Couldn't decode frame.", 0);
	- } /* if */
	-
	- if (d_get_state(f->decoder) == D_END_OF_STREAM)
	- {
	- sample->flags \|= SOUND_SAMPLEFLAG_EOF;
	- return(0);
	- } /* if */
	-
	- /* An error may have been signalled through the error callback. */
	- if (sample->flags & SOUND_SAMPLEFLAG_ERROR)
	- return(0);
	-
	- return(f->frame_size);
	-} /* FLAC_read */
	-
	-
	-static int FLAC_rewind(Sound_Sample *sample)
	-{
	- return FLAC_seek(sample, 0);
	-} /* FLAC_rewind */
	-
	-
	-static int FLAC_seek(Sound_Sample *sample, Uint32 ms)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- flac_t f = (flac_t ) internal->decoder_private;
	-
	- d_seek_absolute(f->decoder, (ms * sample->actual.rate) / 1000);
	- return(1);
	-} /* FLAC_seek */
	-
	-#endif /* SOUND_SUPPORTS_FLAC */
	-
	-/* end of flac.c ... */
	diff --git a/util/sdl/sound/decoders/midi.c b/util/sdl/sound/decoders/midi.c
	deleted file mode 100644
	index b283c5c6..00000000
	--- a/util/sdl/sound/decoders/midi.c
	+++ /dev/null
	@@ -1,175 +0,0 @@
	-/*
	- * SDL_sound -- An abstract sound format decoding API.
	- * Copyright (C) 2001 Ryan C. Gordon.
	- *
	- * This library is free software; you can redistribute it and/or
	- * modify it under the terms of the GNU Lesser General Public
	- * License as published by the Free Software Foundation; either
	- * version 2.1 of the License, or (at your option) any later version.
	- *
	- * This library is distributed in the hope that it will be useful,
	- * but WITHOUT ANY WARRANTY; without even the implied warranty of
	- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	- * Lesser General Public License for more details.
	- *
	- * You should have received a copy of the GNU Lesser General Public
	- * License along with this library; if not, write to the Free Software
	- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
	- */
	-
	-/*
	- * MIDI decoder for SDL_sound.
	- *
	- * This driver handles MIDI data through a stripped-down version of TiMidity.
	- * See the documentation in the timidity subdirectory.
	- *
	- * Please see the file COPYING in the source's root directory.
	- *
	- * This file written by Torbjörn Andersson. (d91tan@Update.UU.SE)
	- */
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#ifdef SOUND_SUPPORTS_MIDI
	-
	-#include <stdio.h>
	-#include <stdlib.h>
	-#include <string.h>
	-
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-#include "timidity.h"
	-
	-
	-static int MIDI_init(void);
	-static void MIDI_quit(void);
	-static int MIDI_open(Sound_Sample sample, const char ext);
	-static void MIDI_close(Sound_Sample *sample);
	-static Uint32 MIDI_read(Sound_Sample *sample);
	-static int MIDI_rewind(Sound_Sample *sample);
	-static int MIDI_seek(Sound_Sample *sample, Uint32 ms);
	-
	-static const char *extensions_midi[] = { "MIDI", "MID", NULL };
	-const Sound_DecoderFunctions __Sound_DecoderFunctions_MIDI =
	-{
	- {
	- extensions_midi,
	- "MIDI decoder, using a subset of TiMidity",
	- "Torbjörn Andersson <d91tan@Update.UU.SE>",
	- "http://www.icculus.org/SDL_sound/"
	- },
	-
	- MIDI_init, /* init() method */
	- MIDI_quit, /* quit() method */
	- MIDI_open, /* open() method */
	- MIDI_close, /* close() method */
	- MIDI_read, /* read() method */
	- MIDI_rewind, /* rewind() method */
	- MIDI_seek /* seek() method */
	-};
	-
	-
	-static int MIDI_init(void)
	-{
	- BAIL_IF_MACRO(Timidity_Init() < 0, "MIDI: Could not initialise", 0);
	- return(1);
	-} /* MIDI_init */
	-
	-
	-static void MIDI_quit(void)
	-{
	- Timidity_Exit();
	-} /* MIDI_quit */
	-
	-
	-static int MIDI_open(Sound_Sample sample, const char ext)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- SDL_RWops *rw = internal->rw;
	- SDL_AudioSpec spec;
	- MidiSong *song;
	-
	- spec.channels = 2;
	- spec.format = AUDIO_S16SYS;
	- spec.freq = 44100;
	- spec.samples = 4096;
	-
	- song = Timidity_LoadSong(rw, &spec);
	- BAIL_IF_MACRO(song == NULL, "MIDI: Not a MIDI file.", 0);
	- Timidity_SetVolume(song, 100);
	- Timidity_Start(song);
	-
	- SNDDBG(("MIDI: Accepting data stream.\n"));
	-
	- internal->decoder_private = (void *) song;
	-
	- sample->actual.channels = 2;
	- sample->actual.rate = 44100;
	- sample->actual.format = AUDIO_S16SYS;
	-
	- sample->flags = SOUND_SAMPLEFLAG_CANSEEK;
	- return(1); /* we'll handle this data. */
	-} /* MIDI_open */
	-
	-
	-static void MIDI_close(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- MidiSong song = (MidiSong ) internal->decoder_private;
	-
	- Timidity_FreeSong(song);
	-} /* MIDI_close */
	-
	-
	-static Uint32 MIDI_read(Sound_Sample *sample)
	-{
	- Uint32 retval;
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- MidiSong song = (MidiSong ) internal->decoder_private;
	-
	- retval = Timidity_PlaySome(song, internal->buffer, internal->buffer_size);
	-
	- /* Make sure the read went smoothly... */
	- if (retval == 0)
	- sample->flags \|= SOUND_SAMPLEFLAG_EOF;
	-
	- else if (retval == -1)
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	-
	- /* (next call this EAGAIN may turn into an EOF or error.) */
	- else if (retval < internal->buffer_size)
	- sample->flags \|= SOUND_SAMPLEFLAG_EAGAIN;
	-
	- return(retval);
	-} /* MIDI_read */
	-
	-
	-static int MIDI_rewind(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- MidiSong song = (MidiSong ) internal->decoder_private;
	-
	- Timidity_Start(song);
	- return(1);
	-} /* MIDI_rewind */
	-
	-
	-static int MIDI_seek(Sound_Sample *sample, Uint32 ms)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- MidiSong song = (MidiSong ) internal->decoder_private;
	-
	- Timidity_Seek(song, ms);
	- return(1);
	-} /* MIDI_seek */
	-
	-#endif /* SOUND_SUPPORTS_MIDI */
	-
	-
	-/* end of midi.c ... */
	-
	diff --git a/util/sdl/sound/decoders/mikmod.c b/util/sdl/sound/decoders/mikmod.c
	deleted file mode 100644
	index ebfed455..00000000
	--- a/util/sdl/sound/decoders/mikmod.c
	+++ /dev/null
	@@ -1,344 +0,0 @@
	-/*
	- * SDL_sound -- An abstract sound format decoding API.
	- * Copyright (C) 2001 Ryan C. Gordon.
	- *
	- * This library is free software; you can redistribute it and/or
	- * modify it under the terms of the GNU Lesser General Public
	- * License as published by the Free Software Foundation; either
	- * version 2.1 of the License, or (at your option) any later version.
	- *
	- * This library is distributed in the hope that it will be useful,
	- * but WITHOUT ANY WARRANTY; without even the implied warranty of
	- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	- * Lesser General Public License for more details.
	- *
	- * You should have received a copy of the GNU Lesser General Public
	- * License along with this library; if not, write to the Free Software
	- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
	- */
	-
	-/*
	- * Module player for SDL_sound. This driver handles anything MikMod does, and
	- * is based on SDL_mixer.
	- *
	- * Please see the file COPYING in the source's root directory.
	- *
	- * This file written by Torbjörn Andersson (d91tan@Update.UU.SE)
	- */
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#ifdef SOUND_SUPPORTS_MIKMOD
	-
	-#include <stdio.h>
	-#include <stdlib.h>
	-#include <string.h>
	-
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-#include "mikmod.h"
	-
	-
	-static int MIKMOD_init(void);
	-static void MIKMOD_quit(void);
	-static int MIKMOD_open(Sound_Sample sample, const char ext);
	-static void MIKMOD_close(Sound_Sample *sample);
	-static Uint32 MIKMOD_read(Sound_Sample *sample);
	-static int MIKMOD_rewind(Sound_Sample *sample);
	-static int MIKMOD_seek(Sound_Sample *sample, Uint32 ms);
	-
	-static const char *extensions_mikmod[] =
	-{
	- "669", /* Composer 669 */
	- "AMF", /* DMP Advanced Module Format */
	- "DSM", /* DSIK internal format */
	- "FAR", /* Farandole module */
	- "GDM", /* General DigiMusic module */
	- "IMF", /* Imago Orpheus module */
	- "IT", /* Impulse tracker */
	- "M15", /* 15 instrument MOD / Ultimate Sound Tracker (old M15 format) */
	- "MED", /* Amiga MED module */
	- "MOD", /* Generic MOD (Protracker, StarTracker, FastTracker, etc) */
	- "MTM", /* MTM module */
	- "OKT", /* Oktalyzer module */
	- "S3M", /* Screamtracker module */
	- "STM", /* Screamtracker 2 module */
	- "STX", /* STMIK 0.2 module */
	- "ULT", /* Ultratracker module */
	- "UNI", /* UNIMOD - libmikmod's and APlayer's internal module format */
	- "XM", /* Fasttracker module */
	- NULL
	-};
	-
	-const Sound_DecoderFunctions __Sound_DecoderFunctions_MIKMOD =
	-{
	- {
	- extensions_mikmod,
	- "Play modules through MikMod",
	- "Torbjörn Andersson <d91tan@Update.UU.SE>",
	- "http://mikmod.raphnet.net/"
	- },
	-
	- MIKMOD_init, /* init() method */
	- MIKMOD_quit, /* quit() method */
	- MIKMOD_open, /* open() method */
	- MIKMOD_close, /* close() method */
	- MIKMOD_read, /* read() method */
	- MIKMOD_rewind, /* rewind() method */
	- MIKMOD_seek /* seek() method */
	-};
	-
	-
	-/* Make MikMod read from a RWops... */
	-
	-typedef struct MRWOPSREADER {
	- MREADER core;
	- Sound_Sample *sample;
	- int end;
	-} MRWOPSREADER;
	-
	-static BOOL _mm_RWopsReader_eof(MREADER *reader)
	-{
	- MRWOPSREADER rwops_reader = (MRWOPSREADER ) reader;
	- Sound_Sample *sample = rwops_reader->sample;
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- int pos = SDL_RWtell(internal->rw);
	-
	- if (rwops_reader->end == pos)
	- return(1);
	-
	- return(0);
	-} /* _mm_RWopsReader_eof */
	-
	-
	-static BOOL _mm_RWopsReader_read(MREADER reader, void ptr, size_t size)
	-{
	- MRWOPSREADER rwops_reader = (MRWOPSREADER ) reader;
	- Sound_Sample *sample = rwops_reader->sample;
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- return(SDL_RWread(internal->rw, ptr, size, 1));
	-} /* _mm_RWopsReader_Read */
	-
	-
	-static int _mm_RWopsReader_get(MREADER *reader)
	-{
	- char buf;
	- MRWOPSREADER rwops_reader = (MRWOPSREADER ) reader;
	- Sound_Sample *sample = rwops_reader->sample;
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	-
	- if (SDL_RWread(internal->rw, &buf, 1, 1) != 1)
	- return(EOF);
	-
	- return((int) buf);
	-} /* _mm_RWopsReader_get */
	-
	-
	-static BOOL _mm_RWopsReader_seek(MREADER *reader, long offset, int whence)
	-{
	- MRWOPSREADER rwops_reader = (MRWOPSREADER ) reader;
	- Sound_Sample *sample = rwops_reader->sample;
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	-
	- return(SDL_RWseek(internal->rw, offset, whence));
	-} /* _mm_RWopsReader_seek */
	-
	-
	-static long _mm_RWopsReader_tell(MREADER *reader)
	-{
	- MRWOPSREADER rwops_reader = (MRWOPSREADER ) reader;
	- Sound_Sample *sample = rwops_reader->sample;
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	-
	- return(SDL_RWtell(internal->rw));
	-} /* _mm_RWopsReader_tell */
	-
	-
	-static MREADER _mm_new_rwops_reader(Sound_Sample sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	-
	- MRWOPSREADER reader = (MRWOPSREADER ) malloc(sizeof (MRWOPSREADER));
	- if (reader != NULL)
	- {
	- int failed_seek = 1;
	- int here;
	- reader->core.Eof = _mm_RWopsReader_eof;
	- reader->core.Read = _mm_RWopsReader_read;
	- reader->core.Get = _mm_RWopsReader_get;
	- reader->core.Seek = _mm_RWopsReader_seek;
	- reader->core.Tell = _mm_RWopsReader_tell;
	- reader->sample = sample;
	-
	- /* RWops does not explicitly support an eof check, so we shall find
	- the end manually - this requires seek support for the RWop */
	- here = SDL_RWtell(internal->rw);
	- if (here != -1)
	- {
	- reader->end = SDL_RWseek(internal->rw, 0, SEEK_END);
	- if (reader->end != -1)
	- {
	- /* Move back */
	- if (SDL_RWseek(internal->rw, here, SEEK_SET) != -1)
	- failed_seek = 0;
	- } /* if */
	- } /* if */
	-
	- if (failed_seek)
	- {
	- free(reader);
	- reader = NULL;
	- } /* if */
	- } /* if */
	-
	- return((MREADER *) reader);
	-} /* _mm_new_rwops_reader */
	-
	-
	-static void _mm_delete_rwops_reader(MREADER *reader)
	-{
	- /* SDL_sound will delete the RWops and sample at a higher level... */
	- if (reader != NULL)
	- free(reader);
	-} /* _mm_delete_rwops_reader */
	-
	-
	-
	-static int MIKMOD_init(void)
	-{
	- MikMod_RegisterDriver(&drv_nos);
	-
	- /* Quick and dirty hack to prevent an infinite loop problem
	- * found when using SDL_mixer and SDL_sound together and
	- * both have MikMod compiled in. So, check to see if
	- * MikMod has already been registered first before calling
	- * RegisterAllLoaders()
	- */
	- if(MikMod_InfoLoader() == NULL)
	- {
	- MikMod_RegisterAllLoaders();
	- }
	- /*
	- * Both DMODE_SOFT_MUSIC and DMODE_16BITS should be set by default,
	- * so this is just for clarity. I haven't experimented with any of
	- * the other flags. There are a few which are said to give better
	- * sound quality.
	- */
	- md_mode \|= (DMODE_SOFT_MUSIC \| DMODE_16BITS);
	- md_mixfreq = 0;
	- md_reverb = 1;
	-
	- BAIL_IF_MACRO(MikMod_Init(""), MikMod_strerror(MikMod_errno), 0);
	-
	- return(1); /* success. */
	-} /* MIKMOD_init */
	-
	-
	-static void MIKMOD_quit(void)
	-{
	- MikMod_Exit();
	- md_mixfreq = 0;
	-} /* MIKMOD_quit */
	-
	-
	-static int MIKMOD_open(Sound_Sample sample, const char ext)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- MREADER *reader;
	- MODULE *module;
	-
	- reader = _mm_new_rwops_reader(sample);
	- BAIL_IF_MACRO(reader == NULL, ERR_OUT_OF_MEMORY, 0);
	- module = Player_LoadGeneric(reader, 64, 0);
	- _mm_delete_rwops_reader(reader);
	- BAIL_IF_MACRO(module == NULL, "MIKMOD: Not a module file.", 0);
	-
	- module->extspd = 1;
	- module->panflag = 1;
	- module->wrap = 0;
	- module->loop = 0;
	-
	- if (md_mixfreq == 0)
	- md_mixfreq = (!sample->desired.rate) ? 44100 : sample->desired.rate;
	-
	- sample->actual.channels = 2;
	- sample->actual.rate = md_mixfreq;
	- sample->actual.format = AUDIO_S16SYS;
	- internal->decoder_private = (void *) module;
	-
	- Player_Start(module);
	- Player_SetPosition(0);
	-
	- sample->flags = SOUND_SAMPLEFLAG_NONE;
	-
	- SNDDBG(("MIKMOD: Name: %s\n", module->songname));
	- SNDDBG(("MIKMOD: Type: %s\n", module->modtype));
	- SNDDBG(("MIKMOD: Accepting data stream\n"));
	-
	- return(1); /* we'll handle this data. */
	-} /* MIKMOD_open */
	-
	-
	-static void MIKMOD_close(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- MODULE module = (MODULE ) internal->decoder_private;
	-
	- Player_Free(module);
	-} /* MIKMOD_close */
	-
	-
	-static Uint32 MIKMOD_read(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- MODULE module = (MODULE ) internal->decoder_private;
	-
	- /* Switch to the current module, stopping any previous one. */
	- Player_Start(module);
	- if (!Player_Active())
	- {
	- sample->flags \|= SOUND_SAMPLEFLAG_EOF;
	- return(0);
	- } /* if */
	- return((Uint32) VC_WriteBytes(internal->buffer, internal->buffer_size));
	-} /* MIKMOD_read */
	-
	-
	-static int MIKMOD_rewind(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- MODULE module = (MODULE ) internal->decoder_private;
	-
	- Player_Start(module);
	- Player_SetPosition(0);
	- return(1);
	-} /* MIKMOD_rewind */
	-
	-
	-static int MIKMOD_seek(Sound_Sample *sample, Uint32 ms)
	-{
	-#if 0
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- MODULE module = (MODULE ) internal->decoder_private;
	-
	- /*
	- * Heaven may know what the argument to Player_SetPosition() is.
	- * I, however, haven't the faintest idea.
	- */
	- Player_Start(module);
	- Player_SetPosition(ms);
	- return(1);
	-#else
	- BAIL_MACRO("MIKMOD: Seeking not implemented", 0);
	-#endif
	-} /* MIKMOD_seek */
	-
	-#endif /* SOUND_SUPPORTS_MIKMOD */
	-
	-
	-/* end of mikmod.c ... */
	diff --git a/util/sdl/sound/decoders/modplug.c b/util/sdl/sound/decoders/modplug.c
	deleted file mode 100644
	index b2b3233d..00000000
	--- a/util/sdl/sound/decoders/modplug.c
	+++ /dev/null
	@@ -1,340 +0,0 @@
	-/*
	- * SDL_sound -- An abstract sound format decoding API.
	- * Copyright (C) 2001 Ryan C. Gordon.
	- *
	- * This library is free software; you can redistribute it and/or
	- * modify it under the terms of the GNU Lesser General Public
	- * License as published by the Free Software Foundation; either
	- * version 2.1 of the License, or (at your option) any later version.
	- *
	- * This library is distributed in the hope that it will be useful,
	- * but WITHOUT ANY WARRANTY; without even the implied warranty of
	- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	- * Lesser General Public License for more details.
	- *
	- * You should have received a copy of the GNU Lesser General Public
	- * License along with this library; if not, write to the Free Software
	- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
	- */
	-
	-/*
	- * Module player for SDL_sound. This driver handles anything that ModPlug does.
	- *
	- * ModPlug can be found at http://sourceforge.net/projects/modplug-xmms
	- *
	- * An unofficial version of modplug with all C++ dependencies removed is also
	- * available: http://freecraft.net/snapshots/
	- * (Look for something like "libmodplug-johns-*.tar.gz")
	- *
	- * Please see the file COPYING in the source's root directory.
	- *
	- * This file written by Torbjörn Andersson (d91tan@Update.UU.SE)
	- */
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#ifdef SOUND_SUPPORTS_MODPLUG
	-
	-#include <stdio.h>
	-#include <stdlib.h>
	-#include <string.h>
	-
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-#if SOUND_MODPLUG_IN_OWN_PATH
	-#include "libmodplug/modplug.h"
	-#else
	-#include "modplug.h"
	-#endif
	-
	-static int MODPLUG_init(void);
	-static void MODPLUG_quit(void);
	-static int MODPLUG_open(Sound_Sample sample, const char ext);
	-static void MODPLUG_close(Sound_Sample *sample);
	-static Uint32 MODPLUG_read(Sound_Sample *sample);
	-static int MODPLUG_rewind(Sound_Sample *sample);
	-static int MODPLUG_seek(Sound_Sample *sample, Uint32 ms);
	-
	-static const char *extensions_modplug[] =
	-{
	- /* The XMMS plugin is apparently able to load compressed modules as
	- * well, but libmodplug does not handle this.
	- */
	- "669", /* Composer 669 / UNIS 669 module */
	- "AMF", /* ASYLUM Music Format / Advanced Music Format(DSM) */
	- "AMS", /* AMS module */
	- "DBM", /* DigiBooster Pro Module */
	- "DMF", /* DMF DELUSION DIGITAL MUSIC FILEFORMAT (X-Tracker) */
	- "DSM", /* DSIK Internal Format module */
	- "FAR", /* Farandole module */
	- "IT", /* Impulse Tracker IT file */
	- "MDL", /* DigiTracker module */
	-#if 0
	- "J2B", /* Not implemented? What is it anyway? */
	-#endif
	- "MED", /* OctaMed MED file */
	- "MOD", /* ProTracker / NoiseTracker MOD/NST file */
	- "MT2", /* MadTracker 2.0 */
	- "MTM", /* MTM file */
	- "OKT", /* Oktalyzer module */
	- "PTM", /* PTM PolyTracker module */
	- "PSM", /* PSM module */
	- "S3M", /* ScreamTracker file */
	- "STM", /* ST 2.xx */
	- "ULT",
	- "UMX",
	- "XM", /* FastTracker II */
	- NULL
	-};
	-
	-const Sound_DecoderFunctions __Sound_DecoderFunctions_MODPLUG =
	-{
	- {
	- extensions_modplug,
	- "Play modules through ModPlug",
	- "Torbjörn Andersson <d91tan@Update.UU.SE>",
	- "http://modplug-xmms.sourceforge.net/"
	- },
	-
	- MODPLUG_init, /* init() method */
	- MODPLUG_quit, /* quit() method */
	- MODPLUG_open, /* open() method */
	- MODPLUG_close, /* close() method */
	- MODPLUG_read, /* read() method */
	- MODPLUG_rewind, /* rewind() method */
	- MODPLUG_seek /* seek() method */
	-};
	-
	-
	-static ModPlug_Settings settings;
	-static Sound_AudioInfo current_audioinfo;
	-static unsigned int total_mods_decoding = 0;
	-static SDL_mutex *modplug_mutex = NULL;
	-
	-static int MODPLUG_init(void)
	-{
	- assert(modplug_mutex == NULL);
	-
	- /*
	- * The settings will require some experimenting. I've borrowed some
	- * of them from the XMMS ModPlug plugin.
	- */
	- settings.mFlags = MODPLUG_ENABLE_OVERSAMPLING;
	-
	-#ifndef _WIN32_WCE
	- settings.mFlags \|= MODPLUG_ENABLE_NOISE_REDUCTION \|
	- MODPLUG_ENABLE_REVERB \|
	- MODPLUG_ENABLE_MEGABASS \|
	- MODPLUG_ENABLE_SURROUND;
	-
	- settings.mReverbDepth = 30;
	- settings.mReverbDelay = 100;
	- settings.mBassAmount = 40;
	- settings.mBassRange = 30;
	- settings.mSurroundDepth = 20;
	- settings.mSurroundDelay = 20;
	-#endif
	-
	- settings.mChannels = 2;
	- settings.mBits = 16;
	- settings.mFrequency = 44100;
	- settings.mResamplingMode = MODPLUG_RESAMPLE_FIR;
	- settings.mLoopCount = 0;
	-
	- current_audioinfo.channels = 2;
	- current_audioinfo.rate = 44100;
	- current_audioinfo.format = AUDIO_S16SYS;
	- total_mods_decoding = 0;
	-
	- modplug_mutex = SDL_CreateMutex();
	-
	- ModPlug_SetSettings(&settings);
	- return(1); /* success. */
	-} /* MODPLUG_init */
	-
	-
	-static void MODPLUG_quit(void)
	-{
	- assert(total_mods_decoding == 0);
	-
	- if (modplug_mutex != NULL)
	- {
	- SDL_DestroyMutex(modplug_mutex);
	- modplug_mutex = NULL;
	- } /* if */
	-} /* MODPLUG_quit */
	-
	-
	-/*
	- * Most MOD files I've seen have tended to be a few hundred KB, even if some
	- * of them were much smaller than that.
	- */
	-#define CHUNK_SIZE 65536
	-
	-static int MODPLUG_open(Sound_Sample sample, const char ext)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- ModPlugFile *module;
	- Uint8 *data;
	- size_t size;
	- Uint32 retval;
	- int has_extension = 0;
	- int i;
	-
	- /*
	- * Apparently ModPlug's loaders are too forgiving. They gladly accept
	- * streams that they shouldn't. For now, rely on file extension instead.
	- */
	- for (i = 0; extensions_modplug[i] != NULL; i++)
	- {
	- if (__Sound_strcasecmp(ext, extensions_modplug[i]) == 0)
	- {
	- has_extension = 1;
	- break;
	- } /* if */
	- } /* for */
	-
	- if (!has_extension)
	- {
	- SNDDBG(("MODPLUG: Unrecognized file type: %s\n", ext));
	- BAIL_MACRO("MODPLUG: Not a module file.", 0);
	- } /* if */
	-
	- /*
	- * ModPlug needs the entire stream in one big chunk. I don't like it,
	- * but I don't think there's any way around it.
	- */
	- data = (Uint8 *) malloc(CHUNK_SIZE);
	- BAIL_IF_MACRO(data == NULL, ERR_OUT_OF_MEMORY, 0);
	- size = 0;
	-
	- do
	- {
	- retval = SDL_RWread(internal->rw, &data[size], 1, CHUNK_SIZE);
	- size += retval;
	- if (retval == CHUNK_SIZE)
	- {
	- data = (Uint8 *) realloc(data, size + CHUNK_SIZE);
	- BAIL_IF_MACRO(data == NULL, ERR_OUT_OF_MEMORY, 0);
	- } /* if */
	- } while (retval > 0);
	-
	- /*
	- * It's only safe to change these settings when there're
	- * no other mods being decoded...
	- */
	- if (modplug_mutex != NULL)
	- SDL_LockMutex(modplug_mutex);
	-
	- if (total_mods_decoding > 0)
	- {
	- /* other mods decoding: use the same settings they are. */
	- memcpy(&sample->actual, &current_audioinfo, sizeof (Sound_AudioInfo));
	- } /* if */
	- else
	- {
	- /* no other mods decoding: define the new ModPlug output settings. */
	- memcpy(&sample->actual, &sample->desired, sizeof (Sound_AudioInfo));
	- if (sample->actual.rate == 0)
	- sample->actual.rate = 44100;
	- if (sample->actual.channels == 0)
	- sample->actual.channels = 2;
	- if (sample->actual.format == 0)
	- sample->actual.format = AUDIO_S16SYS;
	-
	- memcpy(&current_audioinfo, &sample->actual, sizeof (Sound_AudioInfo));
	- settings.mChannels=sample->actual.channels;
	- settings.mFrequency=sample->actual.rate;
	- settings.mBits = sample->actual.format & 0xFF;
	- ModPlug_SetSettings(&settings);
	- } /* else */
	-
	- /*
	- * The buffer may be a bit too large, but that doesn't matter. I think
	- * it's safe to free it as soon as ModPlug_Load() is finished anyway.
	- */
	- module = ModPlug_Load((void *) data, size);
	- free(data);
	- if (module == NULL)
	- {
	- if (modplug_mutex != NULL)
	- SDL_UnlockMutex(modplug_mutex);
	-
	- BAIL_MACRO("MODPLUG: Not a module file.", 0);
	- } /* if */
	-
	- total_mods_decoding++;
	-
	- if (modplug_mutex != NULL)
	- SDL_UnlockMutex(modplug_mutex);
	-
	- SNDDBG(("MODPLUG: [%d ms] %s\n",
	- ModPlug_GetLength(module), ModPlug_GetName(module)));
	-
	- internal->decoder_private = (void *) module;
	- sample->flags = SOUND_SAMPLEFLAG_CANSEEK;
	-
	- SNDDBG(("MODPLUG: Accepting data stream\n"));
	- return(1); /* we'll handle this data. */
	-} /* MODPLUG_open */
	-
	-
	-static void MODPLUG_close(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- ModPlugFile module = (ModPlugFile ) internal->decoder_private;
	-
	- if (modplug_mutex != NULL)
	- SDL_LockMutex(modplug_mutex);
	-
	- total_mods_decoding--;
	-
	- if (modplug_mutex != NULL)
	- SDL_UnlockMutex(modplug_mutex);
	-
	- ModPlug_Unload(module);
	-} /* MODPLUG_close */
	-
	-
	-static Uint32 MODPLUG_read(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- ModPlugFile module = (ModPlugFile ) internal->decoder_private;
	- int retval;
	-
	- retval = ModPlug_Read(module, internal->buffer, internal->buffer_size);
	- if (retval == 0)
	- sample->flags \|= SOUND_SAMPLEFLAG_EOF;
	- return(retval);
	-} /* MODPLUG_read */
	-
	-
	-static int MODPLUG_rewind(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- ModPlugFile module = (ModPlugFile ) internal->decoder_private;
	-
	- ModPlug_Seek(module, 0);
	- return(1);
	-} /* MODPLUG_rewind */
	-
	-
	-static int MODPLUG_seek(Sound_Sample *sample, Uint32 ms)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- ModPlugFile module = (ModPlugFile ) internal->decoder_private;
	-
	- /* Assume that this will work. */
	- ModPlug_Seek(module, ms);
	- return(1);
	-} /* MODPLUG_seek */
	-
	-#endif /* SOUND_SUPPORTS_MODPLUG */
	-
	-
	-/* end of modplug.c ... */
	diff --git a/util/sdl/sound/decoders/mpglib.c b/util/sdl/sound/decoders/mpglib.c
	deleted file mode 100644
	index d7ee4686..00000000
	--- a/util/sdl/sound/decoders/mpglib.c
	+++ /dev/null
	@@ -1,298 +0,0 @@
	-/*
	- * SDL_sound -- An abstract sound format decoding API.
	- * Copyright (C) 2001 Ryan C. Gordon.
	- *
	- * This library is free software; you can redistribute it and/or
	- * modify it under the terms of the GNU Lesser General Public
	- * License as published by the Free Software Foundation; either
	- * version 2.1 of the License, or (at your option) any later version.
	- *
	- * This library is distributed in the hope that it will be useful,
	- * but WITHOUT ANY WARRANTY; without even the implied warranty of
	- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	- * Lesser General Public License for more details.
	- *
	- * You should have received a copy of the GNU Lesser General Public
	- * License along with this library; if not, write to the Free Software
	- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
	- */
	-
	-/*
	- * MPGLIB decoder for SDL_sound. This is a very lightweight MP3 decoder,
	- * which is included with the SDL_sound source, so that it doesn't rely on
	- * unnecessary external libraries.
	- *
	- * The SMPEG decoder plays back more forms of MPEGs, and may behave better or
	- * worse under various conditions. mpglib is (apparently) more efficient than
	- * SMPEG, and, again, doesn't need an external library. You should test both
	- * decoders and use what you find works best for you.
	- *
	- * mpglib is an LGPL'd portion of mpg123, which can be found in its original
	- * form at: http://www.mpg123.de/
	- *
	- * Please see the file COPYING in the source's root directory. The included
	- * source code for mpglib falls under the LGPL, which is the same license as
	- * SDL_sound (so you can consider it a single work).
	- *
	- * This file written by Ryan C. Gordon. (icculus@icculus.org)
	- */
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#ifdef SOUND_SUPPORTS_MPGLIB
	-
	-#include <stdio.h>
	-#include <stdlib.h>
	-#include <string.h>
	-#include "mpglib/mpg123_sdlsound.h"
	-#include "mpglib/mpglib_sdlsound.h"
	-
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-static int MPGLIB_init(void);
	-static void MPGLIB_quit(void);
	-static int MPGLIB_open(Sound_Sample sample, const char ext);
	-static void MPGLIB_close(Sound_Sample *sample);
	-static Uint32 MPGLIB_read(Sound_Sample *sample);
	-static int MPGLIB_rewind(Sound_Sample *sample);
	-static int MPGLIB_seek(Sound_Sample *sample, Uint32 ms);
	-
	-static const char *extensions_mpglib[] = { "MP3", NULL };
	-const Sound_DecoderFunctions __Sound_DecoderFunctions_MPGLIB =
	-{
	- {
	- extensions_mpglib,
	- "MP3 decoding via internal mpglib",
	- "Ryan C. Gordon <icculus@icculus.org>",
	- "http://www.icculus.org/SDL_sound/"
	- },
	-
	- MPGLIB_init, /* init() method */
	- MPGLIB_quit, /* quit() method */
	- MPGLIB_open, /* open() method */
	- MPGLIB_close, /* close() method */
	- MPGLIB_read, /* read() method */
	- MPGLIB_rewind, /* rewind() method */
	- MPGLIB_seek /* seek() method */
	-};
	-
	-
	-/* this is what we store in our internal->decoder_private field... */
	-typedef struct
	-{
	- struct mpstr mp;
	- Uint8 inbuf[16384];
	- Uint8 outbuf[8192];
	- int outleft;
	- int outpos;
	-} mpglib_t;
	-
	-
	-
	-static int MPGLIB_init(void)
	-{
	- return(1); /* always succeeds. */
	-} /* MPGLIB_init */
	-
	-
	-static void MPGLIB_quit(void)
	-{
	- /* it's a no-op. */
	-} /* MPGLIB_quit */
	-
	-
	-static int MPGLIB_open(Sound_Sample sample, const char ext)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- mpglib_t *mpg = NULL;
	- int rc;
	-
	- /*
	- * If I understand things correctly, MP3 files don't really have any
	- * magic header we can check for. The MP3 player is expected to just
	- * pick the first thing that looks like a valid frame and start
	- * playing from there.
	- *
	- * So here's what we do: If the caller insists that this is really
	- * MP3 we'll take his word for it. Otherwise, use the same test as
	- * SDL_mixer does and check if the stream starts with something that
	- * looks like a frame.
	- *
	- * A frame begins with 11 bits of frame sync (all bits must be set),
	- * followed by a two-bit MPEG Audio version ID:
	- *
	- * 00 - MPEG Version 2.5 (later extension of MPEG 2)
	- * 01 - reserved
	- * 10 - MPEG Version 2 (ISO/IEC 13818-3)
	- * 11 - MPEG Version 1 (ISO/IEC 11172-3)
	- *
	- * Apparently we don't handle MPEG Version 2.5.
	- */
	- if (__Sound_strcasecmp(ext, "MP3") != 0)
	- {
	- Uint8 mp3_magic[2];
	-
	- if (SDL_RWread(internal->rw, mp3_magic, sizeof (mp3_magic), 1) != 1)
	- BAIL_MACRO("MPGLIB: Could not read MP3 magic.", 0);
	-
	- if (mp3_magic[0] != 0xFF \|\| (mp3_magic[1] & 0xF0) != 0xF0)
	- BAIL_MACRO("MPGLIB: Not an MP3 stream.", 0);
	-
	- /* If the seek fails, we'll probably miss a frame, but oh well. */
	- SDL_RWseek(internal->rw, -sizeof (mp3_magic), SEEK_CUR);
	- } /* if */
	-
	- mpg = (mpglib_t *) malloc(sizeof (mpglib_t));
	- BAIL_IF_MACRO(mpg == NULL, ERR_OUT_OF_MEMORY, 0);
	- memset(mpg, '\0', sizeof (mpglib_t));
	- InitMP3(&mpg->mp);
	-
	- rc = SDL_RWread(internal->rw, mpg->inbuf, 1, sizeof (mpg->inbuf));
	- if (rc <= 0)
	- {
	- free(mpg);
	- BAIL_MACRO("MPGLIB: Failed to read any data at all", 0);
	- } /* if */
	-
	- if (decodeMP3(&mpg->mp, mpg->inbuf, rc,
	- mpg->outbuf, sizeof (mpg->outbuf),
	- &mpg->outleft) == MP3_ERR)
	- {
	- free(mpg);
	- BAIL_MACRO("MPGLIB: Not an MP3 stream?", 0);
	- } /* if */
	-
	- SNDDBG(("MPGLIB: Accepting data stream.\n"));
	-
	- internal->decoder_private = mpg;
	- sample->actual.rate = mpglib_freqs[mpg->mp.fr.sampling_frequency];
	- sample->actual.channels = mpg->mp.fr.stereo;
	- sample->actual.format = AUDIO_S16SYS;
	- sample->flags = SOUND_SAMPLEFLAG_NONE;
	-
	- return(1); /* we'll handle this data. */
	-} /* MPGLIB_open */
	-
	-
	-static void MPGLIB_close(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- mpglib_t mpg = ((mpglib_t ) internal->decoder_private);
	- ExitMP3(&mpg->mp);
	- free(mpg);
	-} /* MPGLIB_close */
	-
	-
	-static Uint32 MPGLIB_read(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- mpglib_t mpg = ((mpglib_t ) internal->decoder_private);
	- Uint32 bw = 0;
	- int rc;
	-
	- while (bw < internal->buffer_size)
	- {
	- if (mpg->outleft > 0)
	- {
	- size_t cpysize = internal->buffer_size - bw;
	- if (cpysize > mpg->outleft)
	- cpysize = mpg->outleft;
	- memcpy(((Uint8 *) internal->buffer) + bw,
	- mpg->outbuf + mpg->outpos, cpysize);
	- bw += cpysize;
	- mpg->outpos += cpysize;
	- mpg->outleft -= cpysize;
	- continue;
	- } /* if */
	-
	- /* need to decode more from the MP3 stream... */
	- mpg->outpos = 0;
	- rc = decodeMP3(&mpg->mp, NULL, 0, mpg->outbuf,
	- sizeof (mpg->outbuf), &mpg->outleft);
	- if (rc == MP3_ERR)
	- {
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- return(bw);
	- } /* if */
	-
	- else if (rc == MP3_NEED_MORE)
	- {
	- rc = SDL_RWread(internal->rw, mpg->inbuf, 1, sizeof (mpg->inbuf));
	- if (rc == -1)
	- {
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- return(bw);
	- } /* if */
	-
	- else if (rc == 0)
	- {
	- sample->flags \|= SOUND_SAMPLEFLAG_EOF;
	- return(bw);
	- } /* else if */
	-
	- /* make sure there isn't an ID3 tag. */
	- /*
	- * !!! FIXME: This can fail under the following circumstances:
	- * First, if there's the sequence "TAG" 128 bytes from the end
	- * of a read that isn't the EOF. This is unlikely.
	- * Second, if the TAG sequence is split between two reads (ie,
	- * the last byte of a read is 'T', and the next read is the
	- * final 127 bytes of the stream, being the rest of the ID3 tag).
	- * While this is more likely, it's still not very likely at all.
	- * Still, something SHOULD be done about this.
	- * ID3v2 tags are more complex, too, not to mention LYRICS tags,
	- * etc, which aren't handled, either. Hey, this IS meant to be
	- * a lightweight decoder. Use SMPEG if you need an all-purpose
	- * decoder. mpglib really assumes you control all your assets.
	- */
	- if (rc >= 128)
	- {
	- Uint8 *ptr = &mpg->inbuf[rc - 128];
	- if ((ptr[0] == 'T') && (ptr[1] == 'A') && (ptr[2] == 'G'))
	- rc -= 128; /* disregard it. */
	- } /* if */
	-
	- rc = decodeMP3(&mpg->mp, mpg->inbuf, rc,
	- mpg->outbuf, sizeof (mpg->outbuf),
	- &mpg->outleft);
	- if (rc == MP3_ERR)
	- {
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- return(bw);
	- } /* if */
	- } /* else if */
	- } /* while */
	-
	- return(bw);
	-} /* MPGLIB_read */
	-
	-
	-static int MPGLIB_rewind(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- mpglib_t mpg = ((mpglib_t ) internal->decoder_private);
	- BAIL_IF_MACRO(SDL_RWseek(internal->rw, 0, SEEK_SET) != 0, ERR_IO_ERROR, 0);
	-
	- /* this is just resetting some fields in a structure; it's very fast. */
	- ExitMP3(&mpg->mp);
	- InitMP3(&mpg->mp);
	- mpg->outpos = mpg->outleft = 0;
	- return(1);
	-} /* MPGLIB_rewind */
	-
	-
	-static int MPGLIB_seek(Sound_Sample *sample, Uint32 ms)
	-{
	- BAIL_MACRO("MPGLIB: Seeking not implemented", 0);
	-} /* MPGLIB_seek */
	-
	-#endif /* SOUND_SUPPORTS_MPGLIB */
	-
	-
	-/* end of mpglib.c ... */
	-
	diff --git a/util/sdl/sound/decoders/mpglib/CHANGES b/util/sdl/sound/decoders/mpglib/CHANGES
	deleted file mode 100644
	index b9a8c50c..00000000
	--- a/util/sdl/sound/decoders/mpglib/CHANGES
	+++ /dev/null
	@@ -1,4 +0,0 @@
	-14/Oct/1999:
	- - VBR fix
	- - Layer2 and Layer1 added
	-
	diff --git a/util/sdl/sound/decoders/mpglib/Makefile.am b/util/sdl/sound/decoders/mpglib/Makefile.am
	deleted file mode 100644
	index 8f9a8d8e..00000000
	--- a/util/sdl/sound/decoders/mpglib/Makefile.am
	+++ /dev/null
	@@ -1,23 +0,0 @@
	-if USE_MPGLIB
	-noinst_LTLIBRARIES = libmpglib.la
	-endif
	-
	-INCLUDES = -I$(top_srcdir)
	-libmpglib_la_CFLAGS = -DLAYER1 -DLAYER2 -DLAYER3
	-
	-libmpglib_la_SOURCES = \
	- mpglib_common.c \
	- huffman.h \
	- layer1.c \
	- tabinit.c \
	- dct64_i386.c \
	- interface.c \
	- layer2.c \
	- mpg123_sdlsound.h \
	- decode_i386.c \
	- l2tables.h \
	- layer3.c \
	- mpglib_sdlsound.h
	-
	-EXTRA_DIST = CHANGES README README-sdlsound TODO main.c
	-
	diff --git a/util/sdl/sound/decoders/mpglib/Makefile.in b/util/sdl/sound/decoders/mpglib/Makefile.in
	deleted file mode 100644
	index dd346f88..00000000
	--- a/util/sdl/sound/decoders/mpglib/Makefile.in
	+++ /dev/null
	@@ -1,532 +0,0 @@
	-# Makefile.in generated by automake 1.9.6 from Makefile.am.
	-# @configure_input@
	-
	-# Copyright (C) 1994, 1995, 1996, 1997, 1998, 1999, 2000, 2001, 2002,
	-# 2003, 2004, 2005 Free Software Foundation, Inc.
	-# This Makefile.in is free software; the Free Software Foundation
	-# gives unlimited permission to copy and/or distribute it,
	-# with or without modifications, as long as this notice is preserved.
	-
	-# This program is distributed in the hope that it will be useful,
	-# but WITHOUT ANY WARRANTY, to the extent permitted by law; without
	-# even the implied warranty of MERCHANTABILITY or FITNESS FOR A
	-# PARTICULAR PURPOSE.
	-
	-@SET_MAKE@
	-
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	-top_srcdir = @top_srcdir@
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	-subdir = decoders/mpglib
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	diff --git a/util/sdl/sound/decoders/mpglib/README b/util/sdl/sound/decoders/mpglib/README
	deleted file mode 100644
	index 2465ffaa..00000000
	--- a/util/sdl/sound/decoders/mpglib/README
	+++ /dev/null
	@@ -1,39 +0,0 @@
	-MP3 library
	------------
	-Version 0.2a
	-
	-This decoder is a 'light' version (thrown out all unnecessay parts)
	-from the mpg123 package. I made this for a company.
	-
	-Currently only Layer3 is enabled to save some space. Layer1,2 isn't
	-tested at all. The interface will not change significantly.
	-A backport to the mpg123 package is planed.
	-
	-compiled and tested only on Solaris 2.6
	-main.c contains a simple demo application for library.
	-
	-COPYING: you may use this source under LGPL terms!
	- (Yes, I switched to LGPL for the _mpglib_ part!)
	-
	-PLEASE NOTE: This software may contain patented algorithms (at least
	- patented in some countries). It may be not allowed to sell/use products
	- based on this source code in these countries. Check this out first!
	-
	-COPYRIGHT of MP3 music:
	- Please note, that the duplicating of copyrighted music without explicit
	- permission violates the rights of the owner.
	-
	-SENDING PATCHES:
	- The current version is under LGPL. Please consider this when sending patches or
	- changes. I also want to have the freedom to sell the code to companies that
	- cannot or do not want to use the code under LGPL. So, if you send me
	- significant patches, I need your explicit permission to do this. Of course,
	- there will always be the LGPLed open source version of the 100% same code.
	- In the case you cannot accept this: the code is free, it's your freedom
	- to distribute your changes again under LGPL.
	-
	-FEEDBACK:
	- I'm interessted to here from you, when you use this package as part
	- of another project.
	-
	-
	diff --git a/util/sdl/sound/decoders/mpglib/README-sdlsound b/util/sdl/sound/decoders/mpglib/README-sdlsound
	deleted file mode 100644
	index e24073b4..00000000
	--- a/util/sdl/sound/decoders/mpglib/README-sdlsound
	+++ /dev/null
	@@ -1,7 +0,0 @@
	-This package, according to the README, is under the LGPL, which means it uses
	-the same license as SDL_sound.
	-
	-mpglib is part of mpg123, which can be found at http://www.mpg123.de/ ...
	-
	---ryan.
	-
	diff --git a/util/sdl/sound/decoders/mpglib/TODO b/util/sdl/sound/decoders/mpglib/TODO
	deleted file mode 100644
	index e69de29b..00000000
	diff --git a/util/sdl/sound/decoders/mpglib/dct64_i386.c b/util/sdl/sound/decoders/mpglib/dct64_i386.c
	deleted file mode 100644
	index 67c1fa5e..00000000
	--- a/util/sdl/sound/decoders/mpglib/dct64_i386.c
	+++ /dev/null
	@@ -1,315 +0,0 @@
	-
	-/*
	- * Discrete Cosine Tansform (DCT) for subband synthesis
	- * optimized for machines with no auto-increment.
	- * The performance is highly compiler dependend. Maybe
	- * the dct64.c version for 'normal' processor may be faster
	- * even for Intel processors.
	- */
	-
	-#include "mpg123_sdlsound.h"
	-
	-static void dct64_1(real out0,real out1,real b1,real b2,real *samples)
	-{
	-
	- {
	- register real *costab = pnts[0];
	-
	- b1[0x00] = samples[0x00] + samples[0x1F];
	- b1[0x1F] = (samples[0x00] - samples[0x1F]) * costab[0x0];
	-
	- b1[0x01] = samples[0x01] + samples[0x1E];
	- b1[0x1E] = (samples[0x01] - samples[0x1E]) * costab[0x1];
	-
	- b1[0x02] = samples[0x02] + samples[0x1D];
	- b1[0x1D] = (samples[0x02] - samples[0x1D]) * costab[0x2];
	-
	- b1[0x03] = samples[0x03] + samples[0x1C];
	- b1[0x1C] = (samples[0x03] - samples[0x1C]) * costab[0x3];
	-
	- b1[0x04] = samples[0x04] + samples[0x1B];
	- b1[0x1B] = (samples[0x04] - samples[0x1B]) * costab[0x4];
	-
	- b1[0x05] = samples[0x05] + samples[0x1A];
	- b1[0x1A] = (samples[0x05] - samples[0x1A]) * costab[0x5];
	-
	- b1[0x06] = samples[0x06] + samples[0x19];
	- b1[0x19] = (samples[0x06] - samples[0x19]) * costab[0x6];
	-
	- b1[0x07] = samples[0x07] + samples[0x18];
	- b1[0x18] = (samples[0x07] - samples[0x18]) * costab[0x7];
	-
	- b1[0x08] = samples[0x08] + samples[0x17];
	- b1[0x17] = (samples[0x08] - samples[0x17]) * costab[0x8];
	-
	- b1[0x09] = samples[0x09] + samples[0x16];
	- b1[0x16] = (samples[0x09] - samples[0x16]) * costab[0x9];
	-
	- b1[0x0A] = samples[0x0A] + samples[0x15];
	- b1[0x15] = (samples[0x0A] - samples[0x15]) * costab[0xA];
	-
	- b1[0x0B] = samples[0x0B] + samples[0x14];
	- b1[0x14] = (samples[0x0B] - samples[0x14]) * costab[0xB];
	-
	- b1[0x0C] = samples[0x0C] + samples[0x13];
	- b1[0x13] = (samples[0x0C] - samples[0x13]) * costab[0xC];
	-
	- b1[0x0D] = samples[0x0D] + samples[0x12];
	- b1[0x12] = (samples[0x0D] - samples[0x12]) * costab[0xD];
	-
	- b1[0x0E] = samples[0x0E] + samples[0x11];
	- b1[0x11] = (samples[0x0E] - samples[0x11]) * costab[0xE];
	-
	- b1[0x0F] = samples[0x0F] + samples[0x10];
	- b1[0x10] = (samples[0x0F] - samples[0x10]) * costab[0xF];
	- }
	-
	-
	- {
	- register real *costab = pnts[1];
	-
	- b2[0x00] = b1[0x00] + b1[0x0F];
	- b2[0x0F] = (b1[0x00] - b1[0x0F]) * costab[0];
	- b2[0x01] = b1[0x01] + b1[0x0E];
	- b2[0x0E] = (b1[0x01] - b1[0x0E]) * costab[1];
	- b2[0x02] = b1[0x02] + b1[0x0D];
	- b2[0x0D] = (b1[0x02] - b1[0x0D]) * costab[2];
	- b2[0x03] = b1[0x03] + b1[0x0C];
	- b2[0x0C] = (b1[0x03] - b1[0x0C]) * costab[3];
	- b2[0x04] = b1[0x04] + b1[0x0B];
	- b2[0x0B] = (b1[0x04] - b1[0x0B]) * costab[4];
	- b2[0x05] = b1[0x05] + b1[0x0A];
	- b2[0x0A] = (b1[0x05] - b1[0x0A]) * costab[5];
	- b2[0x06] = b1[0x06] + b1[0x09];
	- b2[0x09] = (b1[0x06] - b1[0x09]) * costab[6];
	- b2[0x07] = b1[0x07] + b1[0x08];
	- b2[0x08] = (b1[0x07] - b1[0x08]) * costab[7];
	-
	- b2[0x10] = b1[0x10] + b1[0x1F];
	- b2[0x1F] = (b1[0x1F] - b1[0x10]) * costab[0];
	- b2[0x11] = b1[0x11] + b1[0x1E];
	- b2[0x1E] = (b1[0x1E] - b1[0x11]) * costab[1];
	- b2[0x12] = b1[0x12] + b1[0x1D];
	- b2[0x1D] = (b1[0x1D] - b1[0x12]) * costab[2];
	- b2[0x13] = b1[0x13] + b1[0x1C];
	- b2[0x1C] = (b1[0x1C] - b1[0x13]) * costab[3];
	- b2[0x14] = b1[0x14] + b1[0x1B];
	- b2[0x1B] = (b1[0x1B] - b1[0x14]) * costab[4];
	- b2[0x15] = b1[0x15] + b1[0x1A];
	- b2[0x1A] = (b1[0x1A] - b1[0x15]) * costab[5];
	- b2[0x16] = b1[0x16] + b1[0x19];
	- b2[0x19] = (b1[0x19] - b1[0x16]) * costab[6];
	- b2[0x17] = b1[0x17] + b1[0x18];
	- b2[0x18] = (b1[0x18] - b1[0x17]) * costab[7];
	- }
	-
	- {
	- register real *costab = pnts[2];
	-
	- b1[0x00] = b2[0x00] + b2[0x07];
	- b1[0x07] = (b2[0x00] - b2[0x07]) * costab[0];
	- b1[0x01] = b2[0x01] + b2[0x06];
	- b1[0x06] = (b2[0x01] - b2[0x06]) * costab[1];
	- b1[0x02] = b2[0x02] + b2[0x05];
	- b1[0x05] = (b2[0x02] - b2[0x05]) * costab[2];
	- b1[0x03] = b2[0x03] + b2[0x04];
	- b1[0x04] = (b2[0x03] - b2[0x04]) * costab[3];
	-
	- b1[0x08] = b2[0x08] + b2[0x0F];
	- b1[0x0F] = (b2[0x0F] - b2[0x08]) * costab[0];
	- b1[0x09] = b2[0x09] + b2[0x0E];
	- b1[0x0E] = (b2[0x0E] - b2[0x09]) * costab[1];
	- b1[0x0A] = b2[0x0A] + b2[0x0D];
	- b1[0x0D] = (b2[0x0D] - b2[0x0A]) * costab[2];
	- b1[0x0B] = b2[0x0B] + b2[0x0C];
	- b1[0x0C] = (b2[0x0C] - b2[0x0B]) * costab[3];
	-
	- b1[0x10] = b2[0x10] + b2[0x17];
	- b1[0x17] = (b2[0x10] - b2[0x17]) * costab[0];
	- b1[0x11] = b2[0x11] + b2[0x16];
	- b1[0x16] = (b2[0x11] - b2[0x16]) * costab[1];
	- b1[0x12] = b2[0x12] + b2[0x15];
	- b1[0x15] = (b2[0x12] - b2[0x15]) * costab[2];
	- b1[0x13] = b2[0x13] + b2[0x14];
	- b1[0x14] = (b2[0x13] - b2[0x14]) * costab[3];
	-
	- b1[0x18] = b2[0x18] + b2[0x1F];
	- b1[0x1F] = (b2[0x1F] - b2[0x18]) * costab[0];
	- b1[0x19] = b2[0x19] + b2[0x1E];
	- b1[0x1E] = (b2[0x1E] - b2[0x19]) * costab[1];
	- b1[0x1A] = b2[0x1A] + b2[0x1D];
	- b1[0x1D] = (b2[0x1D] - b2[0x1A]) * costab[2];
	- b1[0x1B] = b2[0x1B] + b2[0x1C];
	- b1[0x1C] = (b2[0x1C] - b2[0x1B]) * costab[3];
	- }
	-
	- {
	- register real const cos0 = pnts[3][0];
	- register real const cos1 = pnts[3][1];
	-
	- b2[0x00] = b1[0x00] + b1[0x03];
	- b2[0x03] = (b1[0x00] - b1[0x03]) * cos0;
	- b2[0x01] = b1[0x01] + b1[0x02];
	- b2[0x02] = (b1[0x01] - b1[0x02]) * cos1;
	-
	- b2[0x04] = b1[0x04] + b1[0x07];
	- b2[0x07] = (b1[0x07] - b1[0x04]) * cos0;
	- b2[0x05] = b1[0x05] + b1[0x06];
	- b2[0x06] = (b1[0x06] - b1[0x05]) * cos1;
	-
	- b2[0x08] = b1[0x08] + b1[0x0B];
	- b2[0x0B] = (b1[0x08] - b1[0x0B]) * cos0;
	- b2[0x09] = b1[0x09] + b1[0x0A];
	- b2[0x0A] = (b1[0x09] - b1[0x0A]) * cos1;
	-
	- b2[0x0C] = b1[0x0C] + b1[0x0F];
	- b2[0x0F] = (b1[0x0F] - b1[0x0C]) * cos0;
	- b2[0x0D] = b1[0x0D] + b1[0x0E];
	- b2[0x0E] = (b1[0x0E] - b1[0x0D]) * cos1;
	-
	- b2[0x10] = b1[0x10] + b1[0x13];
	- b2[0x13] = (b1[0x10] - b1[0x13]) * cos0;
	- b2[0x11] = b1[0x11] + b1[0x12];
	- b2[0x12] = (b1[0x11] - b1[0x12]) * cos1;
	-
	- b2[0x14] = b1[0x14] + b1[0x17];
	- b2[0x17] = (b1[0x17] - b1[0x14]) * cos0;
	- b2[0x15] = b1[0x15] + b1[0x16];
	- b2[0x16] = (b1[0x16] - b1[0x15]) * cos1;
	-
	- b2[0x18] = b1[0x18] + b1[0x1B];
	- b2[0x1B] = (b1[0x18] - b1[0x1B]) * cos0;
	- b2[0x19] = b1[0x19] + b1[0x1A];
	- b2[0x1A] = (b1[0x19] - b1[0x1A]) * cos1;
	-
	- b2[0x1C] = b1[0x1C] + b1[0x1F];
	- b2[0x1F] = (b1[0x1F] - b1[0x1C]) * cos0;
	- b2[0x1D] = b1[0x1D] + b1[0x1E];
	- b2[0x1E] = (b1[0x1E] - b1[0x1D]) * cos1;
	- }
	-
	- {
	- register real const cos0 = pnts[4][0];
	-
	- b1[0x00] = b2[0x00] + b2[0x01];
	- b1[0x01] = (b2[0x00] - b2[0x01]) * cos0;
	- b1[0x02] = b2[0x02] + b2[0x03];
	- b1[0x03] = (b2[0x03] - b2[0x02]) * cos0;
	- b1[0x02] += b1[0x03];
	-
	- b1[0x04] = b2[0x04] + b2[0x05];
	- b1[0x05] = (b2[0x04] - b2[0x05]) * cos0;
	- b1[0x06] = b2[0x06] + b2[0x07];
	- b1[0x07] = (b2[0x07] - b2[0x06]) * cos0;
	- b1[0x06] += b1[0x07];
	- b1[0x04] += b1[0x06];
	- b1[0x06] += b1[0x05];
	- b1[0x05] += b1[0x07];
	-
	- b1[0x08] = b2[0x08] + b2[0x09];
	- b1[0x09] = (b2[0x08] - b2[0x09]) * cos0;
	- b1[0x0A] = b2[0x0A] + b2[0x0B];
	- b1[0x0B] = (b2[0x0B] - b2[0x0A]) * cos0;
	- b1[0x0A] += b1[0x0B];
	-
	- b1[0x0C] = b2[0x0C] + b2[0x0D];
	- b1[0x0D] = (b2[0x0C] - b2[0x0D]) * cos0;
	- b1[0x0E] = b2[0x0E] + b2[0x0F];
	- b1[0x0F] = (b2[0x0F] - b2[0x0E]) * cos0;
	- b1[0x0E] += b1[0x0F];
	- b1[0x0C] += b1[0x0E];
	- b1[0x0E] += b1[0x0D];
	- b1[0x0D] += b1[0x0F];
	-
	- b1[0x10] = b2[0x10] + b2[0x11];
	- b1[0x11] = (b2[0x10] - b2[0x11]) * cos0;
	- b1[0x12] = b2[0x12] + b2[0x13];
	- b1[0x13] = (b2[0x13] - b2[0x12]) * cos0;
	- b1[0x12] += b1[0x13];
	-
	- b1[0x14] = b2[0x14] + b2[0x15];
	- b1[0x15] = (b2[0x14] - b2[0x15]) * cos0;
	- b1[0x16] = b2[0x16] + b2[0x17];
	- b1[0x17] = (b2[0x17] - b2[0x16]) * cos0;
	- b1[0x16] += b1[0x17];
	- b1[0x14] += b1[0x16];
	- b1[0x16] += b1[0x15];
	- b1[0x15] += b1[0x17];
	-
	- b1[0x18] = b2[0x18] + b2[0x19];
	- b1[0x19] = (b2[0x18] - b2[0x19]) * cos0;
	- b1[0x1A] = b2[0x1A] + b2[0x1B];
	- b1[0x1B] = (b2[0x1B] - b2[0x1A]) * cos0;
	- b1[0x1A] += b1[0x1B];
	-
	- b1[0x1C] = b2[0x1C] + b2[0x1D];
	- b1[0x1D] = (b2[0x1C] - b2[0x1D]) * cos0;
	- b1[0x1E] = b2[0x1E] + b2[0x1F];
	- b1[0x1F] = (b2[0x1F] - b2[0x1E]) * cos0;
	- b1[0x1E] += b1[0x1F];
	- b1[0x1C] += b1[0x1E];
	- b1[0x1E] += b1[0x1D];
	- b1[0x1D] += b1[0x1F];
	- }
	-
	- out0[0x10*16] = b1[0x00];
	- out0[0x10*12] = b1[0x04];
	- out0[0x10* 8] = b1[0x02];
	- out0[0x10* 4] = b1[0x06];
	- out0[0x10* 0] = b1[0x01];
	- out1[0x10* 0] = b1[0x01];
	- out1[0x10* 4] = b1[0x05];
	- out1[0x10* 8] = b1[0x03];
	- out1[0x10*12] = b1[0x07];
	-
	- b1[0x08] += b1[0x0C];
	- out0[0x10*14] = b1[0x08];
	- b1[0x0C] += b1[0x0a];
	- out0[0x10*10] = b1[0x0C];
	- b1[0x0A] += b1[0x0E];
	- out0[0x10* 6] = b1[0x0A];
	- b1[0x0E] += b1[0x09];
	- out0[0x10* 2] = b1[0x0E];
	- b1[0x09] += b1[0x0D];
	- out1[0x10* 2] = b1[0x09];
	- b1[0x0D] += b1[0x0B];
	- out1[0x10* 6] = b1[0x0D];
	- b1[0x0B] += b1[0x0F];
	- out1[0x10*10] = b1[0x0B];
	- out1[0x10*14] = b1[0x0F];
	-
	- b1[0x18] += b1[0x1C];
	- out0[0x10*15] = b1[0x10] + b1[0x18];
	- out0[0x10*13] = b1[0x18] + b1[0x14];
	- b1[0x1C] += b1[0x1a];
	- out0[0x10*11] = b1[0x14] + b1[0x1C];
	- out0[0x10* 9] = b1[0x1C] + b1[0x12];
	- b1[0x1A] += b1[0x1E];
	- out0[0x10* 7] = b1[0x12] + b1[0x1A];
	- out0[0x10* 5] = b1[0x1A] + b1[0x16];
	- b1[0x1E] += b1[0x19];
	- out0[0x10* 3] = b1[0x16] + b1[0x1E];
	- out0[0x10* 1] = b1[0x1E] + b1[0x11];
	- b1[0x19] += b1[0x1D];
	- out1[0x10* 1] = b1[0x11] + b1[0x19];
	- out1[0x10* 3] = b1[0x19] + b1[0x15];
	- b1[0x1D] += b1[0x1B];
	- out1[0x10* 5] = b1[0x15] + b1[0x1D];
	- out1[0x10* 7] = b1[0x1D] + b1[0x13];
	- b1[0x1B] += b1[0x1F];
	- out1[0x10* 9] = b1[0x13] + b1[0x1B];
	- out1[0x10*11] = b1[0x1B] + b1[0x17];
	- out1[0x10*13] = b1[0x17] + b1[0x1F];
	- out1[0x10*15] = b1[0x1F];
	-}
	-
	-/*
	- * the call via dct64 is a trick to force GCC to use
	- * (new) registers for the b1,b2 pointer to the bufs[xx] field
	- */
	-void dct64(real a,real b,real *c)
	-{
	- real bufs[0x40];
	- dct64_1(a,b,bufs,bufs+0x20,c);
	-}
	-
	diff --git a/util/sdl/sound/decoders/mpglib/decode_i386.c b/util/sdl/sound/decoders/mpglib/decode_i386.c
	deleted file mode 100644
	index 0afd5259..00000000
	--- a/util/sdl/sound/decoders/mpglib/decode_i386.c
	+++ /dev/null
	@@ -1,153 +0,0 @@
	-/*
	- * Mpeg Layer-1,2,3 audio decoder
	- * ------------------------------
	- * copyright (c) 1995,1996,1997 by Michael Hipp, All rights reserved.
	- * See also 'README'
	- *
	- * slighlty optimized for machines without autoincrement/decrement.
	- * The performance is highly compiler dependend. Maybe
	- * the decode.c version for 'normal' processor may be faster
	- * even for Intel processors.
	- */
	-
	-#include <stdlib.h>
	-#include <math.h>
	-#include <string.h>
	-
	-#include "mpg123_sdlsound.h"
	-#include "mpglib_sdlsound.h"
	-
	- /* old WRITE_SAMPLE */
	-#define WRITE_SAMPLE(samples,sum,clip) \
	- if( (sum) > 32767.0) { *(samples) = 0x7fff; (clip)++; } \
	- else if( (sum) < -32768.0) { *(samples) = -0x8000; (clip)++; } \
	- else { *(samples) = sum; }
	-
	-int synth_1to1_mono(real bandPtr,unsigned char samples,
	- int pnt, struct mpstr mp)
	-{
	- short samples_tmp[64];
	- short *tmp1 = samples_tmp;
	- int i,ret;
	- int pnt1 = 0;
	-
	- ret = synth_1to1(bandPtr,0,(unsigned char *) samples_tmp,&pnt1,mp);
	- samples += *pnt;
	-
	- for(i=0;i<32;i++) {
	- ( (short ) samples) = *tmp1;
	- samples += 2;
	- tmp1 += 2;
	- }
	- *pnt += 64;
	-
	- return ret;
	-}
	-
	-
	-int synth_1to1(real bandPtr,int channel,unsigned char out,
	- int pnt, struct mpstr mp)
	-{
	- static const int step = 2;
	- int bo;
	- short samples = (short ) (out + *pnt);
	-
	- real b0,(buf)[0x110];
	- int clip = 0;
	- int bo1;
	-
	- bo = mp->synth_bo;
	-
	- if(!channel) {
	- bo--;
	- bo &= 0xf;
	- buf = mp->synth_buffs[0];
	- }
	- else {
	- samples++;
	- buf = mp->synth_buffs[1];
	- }
	-
	- if(bo & 0x1) {
	- b0 = buf[0];
	- bo1 = bo;
	- dct64(buf[1]+((bo+1)&0xf),buf[0]+bo,bandPtr);
	- }
	- else {
	- b0 = buf[1];
	- bo1 = bo+1;
	- dct64(buf[0]+bo,buf[1]+bo+1,bandPtr);
	- }
	-
	- mp->synth_bo = bo;
	-
	- {
	- register int j;
	- real *window = decwin + 16 - bo1;
	-
	- for (j=16;j;j--,b0+=0x10,window+=0x20,samples+=step)
	- {
	- real sum;
	- sum = window[0x0] * b0[0x0];
	- sum -= window[0x1] * b0[0x1];
	- sum += window[0x2] * b0[0x2];
	- sum -= window[0x3] * b0[0x3];
	- sum += window[0x4] * b0[0x4];
	- sum -= window[0x5] * b0[0x5];
	- sum += window[0x6] * b0[0x6];
	- sum -= window[0x7] * b0[0x7];
	- sum += window[0x8] * b0[0x8];
	- sum -= window[0x9] * b0[0x9];
	- sum += window[0xA] * b0[0xA];
	- sum -= window[0xB] * b0[0xB];
	- sum += window[0xC] * b0[0xC];
	- sum -= window[0xD] * b0[0xD];
	- sum += window[0xE] * b0[0xE];
	- sum -= window[0xF] * b0[0xF];
	-
	- WRITE_SAMPLE(samples,sum,clip);
	- }
	-
	- {
	- real sum;
	- sum = window[0x0] * b0[0x0];
	- sum += window[0x2] * b0[0x2];
	- sum += window[0x4] * b0[0x4];
	- sum += window[0x6] * b0[0x6];
	- sum += window[0x8] * b0[0x8];
	- sum += window[0xA] * b0[0xA];
	- sum += window[0xC] * b0[0xC];
	- sum += window[0xE] * b0[0xE];
	- WRITE_SAMPLE(samples,sum,clip);
	- b0-=0x10,window-=0x20,samples+=step;
	- }
	- window += bo1<<1;
	-
	- for (j=15;j;j--,b0-=0x10,window-=0x20,samples+=step)
	- {
	- real sum;
	- sum = -window[-0x1] * b0[0x0];
	- sum -= window[-0x2] * b0[0x1];
	- sum -= window[-0x3] * b0[0x2];
	- sum -= window[-0x4] * b0[0x3];
	- sum -= window[-0x5] * b0[0x4];
	- sum -= window[-0x6] * b0[0x5];
	- sum -= window[-0x7] * b0[0x6];
	- sum -= window[-0x8] * b0[0x7];
	- sum -= window[-0x9] * b0[0x8];
	- sum -= window[-0xA] * b0[0x9];
	- sum -= window[-0xB] * b0[0xA];
	- sum -= window[-0xC] * b0[0xB];
	- sum -= window[-0xD] * b0[0xC];
	- sum -= window[-0xE] * b0[0xD];
	- sum -= window[-0xF] * b0[0xE];
	- sum -= window[-0x0] * b0[0xF];
	-
	- WRITE_SAMPLE(samples,sum,clip);
	- }
	- }
	- *pnt += 128;
	-
	- return clip;
	-}
	-
	diff --git a/util/sdl/sound/decoders/mpglib/huffman.h b/util/sdl/sound/decoders/mpglib/huffman.h
	deleted file mode 100644
	index 7fec0d58..00000000
	--- a/util/sdl/sound/decoders/mpglib/huffman.h
	+++ /dev/null
	@@ -1,332 +0,0 @@
	-/*
	- * huffman tables ... recalcualted to work with my optimzed
	- * decoder scheme (MH)
	- *
	- * probably we could save a few bytes of memory, because the
	- * smaller tables are often the part of a bigger table
	- */
	-
	-struct newhuff
	-{
	- unsigned int linbits;
	- short *table;
	-};
	-
	-static short tab0[] =
	-{
	- 0
	-};
	-
	-static short tab1[] =
	-{
	- -5, -3, -1, 17, 1, 16, 0
	-};
	-
	-static short tab2[] =
	-{
	- -15, -11, -9, -5, -3, -1, 34, 2, 18, -1, 33, 32, 17, -1, 1,
	- 16, 0
	-};
	-
	-static short tab3[] =
	-{
	- -13, -11, -9, -5, -3, -1, 34, 2, 18, -1, 33, 32, 16, 17, -1,
	- 1, 0
	-};
	-
	-static short tab5[] =
	-{
	- -29, -25, -23, -15, -7, -5, -3, -1, 51, 35, 50, 49, -3, -1, 19,
	- 3, -1, 48, 34, -3, -1, 18, 33, -1, 2, 32, 17, -1, 1, 16,
	- 0
	-};
	-
	-static short tab6[] =
	-{
	- -25, -19, -13, -9, -5, -3, -1, 51, 3, 35, -1, 50, 48, -1, 19,
	- 49, -3, -1, 34, 2, 18, -3, -1, 33, 32, 1, -1, 17, -1, 16,
	- 0
	-};
	-
	-static short tab7[] =
	-{
	- -69, -65, -57, -39, -29, -17, -11, -7, -3, -1, 85, 69, -1, 84, 83,
	- -1, 53, 68, -3, -1, 37, 82, 21, -5, -1, 81, -1, 5, 52, -1,
	- 80, -1, 67, 51, -5, -3, -1, 36, 66, 20, -1, 65, 64, -11, -7,
	- -3, -1, 4, 35, -1, 50, 3, -1, 19, 49, -3, -1, 48, 34, 18,
	- -5, -1, 33, -1, 2, 32, 17, -1, 1, 16, 0
	-};
	-
	-static short tab8[] =
	-{
	- -65, -63, -59, -45, -31, -19, -13, -7, -5, -3, -1, 85, 84, 69, 83,
	- -3, -1, 53, 68, 37, -3, -1, 82, 5, 21, -5, -1, 81, -1, 52,
	- 67, -3, -1, 80, 51, 36, -5, -3, -1, 66, 20, 65, -3, -1, 4,
	- 64, -1, 35, 50, -9, -7, -3, -1, 19, 49, -1, 3, 48, 34, -1,
	- 2, 32, -1, 18, 33, 17, -3, -1, 1, 16, 0
	-};
	-
	-static short tab9[] =
	-{
	- -63, -53, -41, -29, -19, -11, -5, -3, -1, 85, 69, 53, -1, 83, -1,
	- 84, 5, -3, -1, 68, 37, -1, 82, 21, -3, -1, 81, 52, -1, 67,
	- -1, 80, 4, -7, -3, -1, 36, 66, -1, 51, 64, -1, 20, 65, -5,
	- -3, -1, 35, 50, 19, -1, 49, -1, 3, 48, -5, -3, -1, 34, 2,
	- 18, -1, 33, 32, -3, -1, 17, 1, -1, 16, 0
	-};
	-
	-static short tab10[] =
	-{
	--125,-121,-111, -83, -55, -35, -21, -13, -7, -3, -1, 119, 103, -1, 118,
	- 87, -3, -1, 117, 102, 71, -3, -1, 116, 86, -1, 101, 55, -9, -3,
	- -1, 115, 70, -3, -1, 85, 84, 99, -1, 39, 114, -11, -5, -3, -1,
	- 100, 7, 112, -1, 98, -1, 69, 53, -5, -1, 6, -1, 83, 68, 23,
	- -17, -5, -1, 113, -1, 54, 38, -5, -3, -1, 37, 82, 21, -1, 81,
	- -1, 52, 67, -3, -1, 22, 97, -1, 96, -1, 5, 80, -19, -11, -7,
	- -3, -1, 36, 66, -1, 51, 4, -1, 20, 65, -3, -1, 64, 35, -1,
	- 50, 3, -3, -1, 19, 49, -1, 48, 34, -7, -3, -1, 18, 33, -1,
	- 2, 32, 17, -1, 1, 16, 0
	-};
	-
	-static short tab11[] =
	-{
	--121,-113, -89, -59, -43, -27, -17, -7, -3, -1, 119, 103, -1, 118, 117,
	- -3, -1, 102, 71, -1, 116, -1, 87, 85, -5, -3, -1, 86, 101, 55,
	- -1, 115, 70, -9, -7, -3, -1, 69, 84, -1, 53, 83, 39, -1, 114,
	- -1, 100, 7, -5, -1, 113, -1, 23, 112, -3, -1, 54, 99, -1, 96,
	- -1, 68, 37, -13, -7, -5, -3, -1, 82, 5, 21, 98, -3, -1, 38,
	- 6, 22, -5, -1, 97, -1, 81, 52, -5, -1, 80, -1, 67, 51, -1,
	- 36, 66, -15, -11, -7, -3, -1, 20, 65, -1, 4, 64, -1, 35, 50,
	- -1, 19, 49, -5, -3, -1, 3, 48, 34, 33, -5, -1, 18, -1, 2,
	- 32, 17, -3, -1, 1, 16, 0
	-};
	-
	-static short tab12[] =
	-{
	--115, -99, -73, -45, -27, -17, -9, -5, -3, -1, 119, 103, 118, -1, 87,
	- 117, -3, -1, 102, 71, -1, 116, 101, -3, -1, 86, 55, -3, -1, 115,
	- 85, 39, -7, -3, -1, 114, 70, -1, 100, 23, -5, -1, 113, -1, 7,
	- 112, -1, 54, 99, -13, -9, -3, -1, 69, 84, -1, 68, -1, 6, 5,
	- -1, 38, 98, -5, -1, 97, -1, 22, 96, -3, -1, 53, 83, -1, 37,
	- 82, -17, -7, -3, -1, 21, 81, -1, 52, 67, -5, -3, -1, 80, 4,
	- 36, -1, 66, 20, -3, -1, 51, 65, -1, 35, 50, -11, -7, -5, -3,
	- -1, 64, 3, 48, 19, -1, 49, 34, -1, 18, 33, -7, -5, -3, -1,
	- 2, 32, 0, 17, -1, 1, 16
	-};
	-
	-static short tab13[] =
	-{
	--509,-503,-475,-405,-333,-265,-205,-153,-115, -83, -53, -35, -21, -13, -9,
	- -7, -5, -3, -1, 254, 252, 253, 237, 255, -1, 239, 223, -3, -1, 238,
	- 207, -1, 222, 191, -9, -3, -1, 251, 206, -1, 220, -1, 175, 233, -1,
	- 236, 221, -9, -5, -3, -1, 250, 205, 190, -1, 235, 159, -3, -1, 249,
	- 234, -1, 189, 219, -17, -9, -3, -1, 143, 248, -1, 204, -1, 174, 158,
	- -5, -1, 142, -1, 127, 126, 247, -5, -1, 218, -1, 173, 188, -3, -1,
	- 203, 246, 111, -15, -7, -3, -1, 232, 95, -1, 157, 217, -3, -1, 245,
	- 231, -1, 172, 187, -9, -3, -1, 79, 244, -3, -1, 202, 230, 243, -1,
	- 63, -1, 141, 216, -21, -9, -3, -1, 47, 242, -3, -1, 110, 156, 15,
	- -5, -3, -1, 201, 94, 171, -3, -1, 125, 215, 78, -11, -5, -3, -1,
	- 200, 214, 62, -1, 185, -1, 155, 170, -1, 31, 241, -23, -13, -5, -1,
	- 240, -1, 186, 229, -3, -1, 228, 140, -1, 109, 227, -5, -1, 226, -1,
	- 46, 14, -1, 30, 225, -15, -7, -3, -1, 224, 93, -1, 213, 124, -3,
	- -1, 199, 77, -1, 139, 184, -7, -3, -1, 212, 154, -1, 169, 108, -1,
	- 198, 61, -37, -21, -9, -5, -3, -1, 211, 123, 45, -1, 210, 29, -5,
	- -1, 183, -1, 92, 197, -3, -1, 153, 122, 195, -7, -5, -3, -1, 167,
	- 151, 75, 209, -3, -1, 13, 208, -1, 138, 168, -11, -7, -3, -1, 76,
	- 196, -1, 107, 182, -1, 60, 44, -3, -1, 194, 91, -3, -1, 181, 137,
	- 28, -43, -23, -11, -5, -1, 193, -1, 152, 12, -1, 192, -1, 180, 106,
	- -5, -3, -1, 166, 121, 59, -1, 179, -1, 136, 90, -11, -5, -1, 43,
	- -1, 165, 105, -1, 164, -1, 120, 135, -5, -1, 148, -1, 119, 118, 178,
	- -11, -3, -1, 27, 177, -3, -1, 11, 176, -1, 150, 74, -7, -3, -1,
	- 58, 163, -1, 89, 149, -1, 42, 162, -47, -23, -9, -3, -1, 26, 161,
	- -3, -1, 10, 104, 160, -5, -3, -1, 134, 73, 147, -3, -1, 57, 88,
	- -1, 133, 103, -9, -3, -1, 41, 146, -3, -1, 87, 117, 56, -5, -1,
	- 131, -1, 102, 71, -3, -1, 116, 86, -1, 101, 115, -11, -3, -1, 25,
	- 145, -3, -1, 9, 144, -1, 72, 132, -7, -5, -1, 114, -1, 70, 100,
	- 40, -1, 130, 24, -41, -27, -11, -5, -3, -1, 55, 39, 23, -1, 113,
	- -1, 85, 7, -7, -3, -1, 112, 54, -1, 99, 69, -3, -1, 84, 38,
	- -1, 98, 53, -5, -1, 129, -1, 8, 128, -3, -1, 22, 97, -1, 6,
	- 96, -13, -9, -5, -3, -1, 83, 68, 37, -1, 82, 5, -1, 21, 81,
	- -7, -3, -1, 52, 67, -1, 80, 36, -3, -1, 66, 51, 20, -19, -11,
	- -5, -1, 65, -1, 4, 64, -3, -1, 35, 50, 19, -3, -1, 49, 3,
	- -1, 48, 34, -3, -1, 18, 33, -1, 2, 32, -3, -1, 17, 1, 16,
	- 0
	-};
	-
	-static short tab15[] =
	-{
	--495,-445,-355,-263,-183,-115, -77, -43, -27, -13, -7, -3, -1, 255, 239,
	- -1, 254, 223, -1, 238, -1, 253, 207, -7, -3, -1, 252, 222, -1, 237,
	- 191, -1, 251, -1, 206, 236, -7, -3, -1, 221, 175, -1, 250, 190, -3,
	- -1, 235, 205, -1, 220, 159, -15, -7, -3, -1, 249, 234, -1, 189, 219,
	- -3, -1, 143, 248, -1, 204, 158, -7, -3, -1, 233, 127, -1, 247, 173,
	- -3, -1, 218, 188, -1, 111, -1, 174, 15, -19, -11, -3, -1, 203, 246,
	- -3, -1, 142, 232, -1, 95, 157, -3, -1, 245, 126, -1, 231, 172, -9,
	- -3, -1, 202, 187, -3, -1, 217, 141, 79, -3, -1, 244, 63, -1, 243,
	- 216, -33, -17, -9, -3, -1, 230, 47, -1, 242, -1, 110, 240, -3, -1,
	- 31, 241, -1, 156, 201, -7, -3, -1, 94, 171, -1, 186, 229, -3, -1,
	- 125, 215, -1, 78, 228, -15, -7, -3, -1, 140, 200, -1, 62, 109, -3,
	- -1, 214, 227, -1, 155, 185, -7, -3, -1, 46, 170, -1, 226, 30, -5,
	- -1, 225, -1, 14, 224, -1, 93, 213, -45, -25, -13, -7, -3, -1, 124,
	- 199, -1, 77, 139, -1, 212, -1, 184, 154, -7, -3, -1, 169, 108, -1,
	- 198, 61, -1, 211, 210, -9, -5, -3, -1, 45, 13, 29, -1, 123, 183,
	- -5, -1, 209, -1, 92, 208, -1, 197, 138, -17, -7, -3, -1, 168, 76,
	- -1, 196, 107, -5, -1, 182, -1, 153, 12, -1, 60, 195, -9, -3, -1,
	- 122, 167, -1, 166, -1, 192, 11, -1, 194, -1, 44, 91, -55, -29, -15,
	- -7, -3, -1, 181, 28, -1, 137, 152, -3, -1, 193, 75, -1, 180, 106,
	- -5, -3, -1, 59, 121, 179, -3, -1, 151, 136, -1, 43, 90, -11, -5,
	- -1, 178, -1, 165, 27, -1, 177, -1, 176, 105, -7, -3, -1, 150, 74,
	- -1, 164, 120, -3, -1, 135, 58, 163, -17, -7, -3, -1, 89, 149, -1,
	- 42, 162, -3, -1, 26, 161, -3, -1, 10, 160, 104, -7, -3, -1, 134,
	- 73, -1, 148, 57, -5, -1, 147, -1, 119, 9, -1, 88, 133, -53, -29,
	- -13, -7, -3, -1, 41, 103, -1, 118, 146, -1, 145, -1, 25, 144, -7,
	- -3, -1, 72, 132, -1, 87, 117, -3, -1, 56, 131, -1, 102, 71, -7,
	- -3, -1, 40, 130, -1, 24, 129, -7, -3, -1, 116, 8, -1, 128, 86,
	- -3, -1, 101, 55, -1, 115, 70, -17, -7, -3, -1, 39, 114, -1, 100,
	- 23, -3, -1, 85, 113, -3, -1, 7, 112, 54, -7, -3, -1, 99, 69,
	- -1, 84, 38, -3, -1, 98, 22, -3, -1, 6, 96, 53, -33, -19, -9,
	- -5, -1, 97, -1, 83, 68, -1, 37, 82, -3, -1, 21, 81, -3, -1,
	- 5, 80, 52, -7, -3, -1, 67, 36, -1, 66, 51, -1, 65, -1, 20,
	- 4, -9, -3, -1, 35, 50, -3, -1, 64, 3, 19, -3, -1, 49, 48,
	- 34, -9, -7, -3, -1, 18, 33, -1, 2, 32, 17, -3, -1, 1, 16,
	- 0
	-};
	-
	-static short tab16[] =
	-{
	--509,-503,-461,-323,-103, -37, -27, -15, -7, -3, -1, 239, 254, -1, 223,
	- 253, -3, -1, 207, 252, -1, 191, 251, -5, -1, 175, -1, 250, 159, -3,
	- -1, 249, 248, 143, -7, -3, -1, 127, 247, -1, 111, 246, 255, -9, -5,
	- -3, -1, 95, 245, 79, -1, 244, 243, -53, -1, 240, -1, 63, -29, -19,
	- -13, -7, -5, -1, 206, -1, 236, 221, 222, -1, 233, -1, 234, 217, -1,
	- 238, -1, 237, 235, -3, -1, 190, 205, -3, -1, 220, 219, 174, -11, -5,
	- -1, 204, -1, 173, 218, -3, -1, 126, 172, 202, -5, -3, -1, 201, 125,
	- 94, 189, 242, -93, -5, -3, -1, 47, 15, 31, -1, 241, -49, -25, -13,
	- -5, -1, 158, -1, 188, 203, -3, -1, 142, 232, -1, 157, 231, -7, -3,
	- -1, 187, 141, -1, 216, 110, -1, 230, 156, -13, -7, -3, -1, 171, 186,
	- -1, 229, 215, -1, 78, -1, 228, 140, -3, -1, 200, 62, -1, 109, -1,
	- 214, 155, -19, -11, -5, -3, -1, 185, 170, 225, -1, 212, -1, 184, 169,
	- -5, -1, 123, -1, 183, 208, 227, -7, -3, -1, 14, 224, -1, 93, 213,
	- -3, -1, 124, 199, -1, 77, 139, -75, -45, -27, -13, -7, -3, -1, 154,
	- 108, -1, 198, 61, -3, -1, 92, 197, 13, -7, -3, -1, 138, 168, -1,
	- 153, 76, -3, -1, 182, 122, 60, -11, -5, -3, -1, 91, 137, 28, -1,
	- 192, -1, 152, 121, -1, 226, -1, 46, 30, -15, -7, -3, -1, 211, 45,
	- -1, 210, 209, -5, -1, 59, -1, 151, 136, 29, -7, -3, -1, 196, 107,
	- -1, 195, 167, -1, 44, -1, 194, 181, -23, -13, -7, -3, -1, 193, 12,
	- -1, 75, 180, -3, -1, 106, 166, 179, -5, -3, -1, 90, 165, 43, -1,
	- 178, 27, -13, -5, -1, 177, -1, 11, 176, -3, -1, 105, 150, -1, 74,
	- 164, -5, -3, -1, 120, 135, 163, -3, -1, 58, 89, 42, -97, -57, -33,
	- -19, -11, -5, -3, -1, 149, 104, 161, -3, -1, 134, 119, 148, -5, -3,
	- -1, 73, 87, 103, 162, -5, -1, 26, -1, 10, 160, -3, -1, 57, 147,
	- -1, 88, 133, -9, -3, -1, 41, 146, -3, -1, 118, 9, 25, -5, -1,
	- 145, -1, 144, 72, -3, -1, 132, 117, -1, 56, 131, -21, -11, -5, -3,
	- -1, 102, 40, 130, -3, -1, 71, 116, 24, -3, -1, 129, 128, -3, -1,
	- 8, 86, 55, -9, -5, -1, 115, -1, 101, 70, -1, 39, 114, -5, -3,
	- -1, 100, 85, 7, 23, -23, -13, -5, -1, 113, -1, 112, 54, -3, -1,
	- 99, 69, -1, 84, 38, -3, -1, 98, 22, -1, 97, -1, 6, 96, -9,
	- -5, -1, 83, -1, 53, 68, -1, 37, 82, -1, 81, -1, 21, 5, -33,
	- -23, -13, -7, -3, -1, 52, 67, -1, 80, 36, -3, -1, 66, 51, 20,
	- -5, -1, 65, -1, 4, 64, -1, 35, 50, -3, -1, 19, 49, -3, -1,
	- 3, 48, 34, -3, -1, 18, 33, -1, 2, 32, -3, -1, 17, 1, 16,
	- 0
	-};
	-
	-static short tab24[] =
	-{
	--451,-117, -43, -25, -15, -7, -3, -1, 239, 254, -1, 223, 253, -3, -1,
	- 207, 252, -1, 191, 251, -5, -1, 250, -1, 175, 159, -1, 249, 248, -9,
	- -5, -3, -1, 143, 127, 247, -1, 111, 246, -3, -1, 95, 245, -1, 79,
	- 244, -71, -7, -3, -1, 63, 243, -1, 47, 242, -5, -1, 241, -1, 31,
	- 240, -25, -9, -1, 15, -3, -1, 238, 222, -1, 237, 206, -7, -3, -1,
	- 236, 221, -1, 190, 235, -3, -1, 205, 220, -1, 174, 234, -15, -7, -3,
	- -1, 189, 219, -1, 204, 158, -3, -1, 233, 173, -1, 218, 188, -7, -3,
	- -1, 203, 142, -1, 232, 157, -3, -1, 217, 126, -1, 231, 172, 255,-235,
	--143, -77, -45, -25, -15, -7, -3, -1, 202, 187, -1, 141, 216, -5, -3,
	- -1, 14, 224, 13, 230, -5, -3, -1, 110, 156, 201, -1, 94, 186, -9,
	- -5, -1, 229, -1, 171, 125, -1, 215, 228, -3, -1, 140, 200, -3, -1,
	- 78, 46, 62, -15, -7, -3, -1, 109, 214, -1, 227, 155, -3, -1, 185,
	- 170, -1, 226, 30, -7, -3, -1, 225, 93, -1, 213, 124, -3, -1, 199,
	- 77, -1, 139, 184, -31, -15, -7, -3, -1, 212, 154, -1, 169, 108, -3,
	- -1, 198, 61, -1, 211, 45, -7, -3, -1, 210, 29, -1, 123, 183, -3,
	- -1, 209, 92, -1, 197, 138, -17, -7, -3, -1, 168, 153, -1, 76, 196,
	- -3, -1, 107, 182, -3, -1, 208, 12, 60, -7, -3, -1, 195, 122, -1,
	- 167, 44, -3, -1, 194, 91, -1, 181, 28, -57, -35, -19, -7, -3, -1,
	- 137, 152, -1, 193, 75, -5, -3, -1, 192, 11, 59, -3, -1, 176, 10,
	- 26, -5, -1, 180, -1, 106, 166, -3, -1, 121, 151, -3, -1, 160, 9,
	- 144, -9, -3, -1, 179, 136, -3, -1, 43, 90, 178, -7, -3, -1, 165,
	- 27, -1, 177, 105, -1, 150, 164, -17, -9, -5, -3, -1, 74, 120, 135,
	- -1, 58, 163, -3, -1, 89, 149, -1, 42, 162, -7, -3, -1, 161, 104,
	- -1, 134, 119, -3, -1, 73, 148, -1, 57, 147, -63, -31, -15, -7, -3,
	- -1, 88, 133, -1, 41, 103, -3, -1, 118, 146, -1, 25, 145, -7, -3,
	- -1, 72, 132, -1, 87, 117, -3, -1, 56, 131, -1, 102, 40, -17, -7,
	- -3, -1, 130, 24, -1, 71, 116, -5, -1, 129, -1, 8, 128, -1, 86,
	- 101, -7, -5, -1, 23, -1, 7, 112, 115, -3, -1, 55, 39, 114, -15,
	- -7, -3, -1, 70, 100, -1, 85, 113, -3, -1, 54, 99, -1, 69, 84,
	- -7, -3, -1, 38, 98, -1, 22, 97, -5, -3, -1, 6, 96, 53, -1,
	- 83, 68, -51, -37, -23, -15, -9, -3, -1, 37, 82, -1, 21, -1, 5,
	- 80, -1, 81, -1, 52, 67, -3, -1, 36, 66, -1, 51, 20, -9, -5,
	- -1, 65, -1, 4, 64, -1, 35, 50, -1, 19, 49, -7, -5, -3, -1,
	- 3, 48, 34, 18, -1, 33, -1, 2, 32, -3, -1, 17, 1, -1, 16,
	- 0
	-};
	-
	-static short tab_c0[] =
	-{
	- -29, -21, -13, -7, -3, -1, 11, 15, -1, 13, 14, -3, -1, 7, 5,
	- 9, -3, -1, 6, 3, -1, 10, 12, -3, -1, 2, 1, -1, 4, 8,
	- 0
	-};
	-
	-static short tab_c1[] =
	-{
	- -15, -7, -3, -1, 15, 14, -1, 13, 12, -3, -1, 11, 10, -1, 9,
	- 8, -7, -3, -1, 7, 6, -1, 5, 4, -3, -1, 3, 2, -1, 1,
	- 0
	-};
	-
	-
	-
	-static struct newhuff ht[] =
	-{
	- { /* 0 */ 0 , tab0 } ,
	- { /* 2 */ 0 , tab1 } ,
	- { /* 3 */ 0 , tab2 } ,
	- { /* 3 */ 0 , tab3 } ,
	- { /* 0 */ 0 , tab0 } ,
	- { /* 4 */ 0 , tab5 } ,
	- { /* 4 */ 0 , tab6 } ,
	- { /* 6 */ 0 , tab7 } ,
	- { /* 6 */ 0 , tab8 } ,
	- { /* 6 */ 0 , tab9 } ,
	- { /* 8 */ 0 , tab10 } ,
	- { /* 8 */ 0 , tab11 } ,
	- { /* 8 */ 0 , tab12 } ,
	- { /* 16 */ 0 , tab13 } ,
	- { /* 0 */ 0 , tab0 } ,
	- { /* 16 */ 0 , tab15 } ,
	-
	- { /* 16 */ 1 , tab16 } ,
	- { /* 16 */ 2 , tab16 } ,
	- { /* 16 */ 3 , tab16 } ,
	- { /* 16 */ 4 , tab16 } ,
	- { /* 16 */ 6 , tab16 } ,
	- { /* 16 */ 8 , tab16 } ,
	- { /* 16 */ 10, tab16 } ,
	- { /* 16 */ 13, tab16 } ,
	- { /* 16 */ 4 , tab24 } ,
	- { /* 16 */ 5 , tab24 } ,
	- { /* 16 */ 6 , tab24 } ,
	- { /* 16 */ 7 , tab24 } ,
	- { /* 16 */ 8 , tab24 } ,
	- { /* 16 */ 9 , tab24 } ,
	- { /* 16 */ 11, tab24 } ,
	- { /* 16 */ 13, tab24 }
	-};
	-
	-static struct newhuff htc[] =
	-{
	- { /* 1 , 1 , */ 0 , tab_c0 } ,
	- { /* 1 , 1 , */ 0 , tab_c1 }
	-};
	-
	-
	diff --git a/util/sdl/sound/decoders/mpglib/interface.c b/util/sdl/sound/decoders/mpglib/interface.c
	deleted file mode 100644
	index db9a3a5d..00000000
	--- a/util/sdl/sound/decoders/mpglib/interface.c
	+++ /dev/null
	@@ -1,243 +0,0 @@
	-
	-#include <stdlib.h>
	-#include <stdio.h>
	-
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-#include "mpg123_sdlsound.h"
	-#include "mpglib_sdlsound.h"
	-
	-
	-BOOL InitMP3(struct mpstr *mp)
	-{
	- static int init = 0;
	-
	- memset(mp,0,sizeof(struct mpstr));
	-
	- mp->framesize = 0;
	- mp->fsizeold = -1;
	- mp->bsize = 0;
	- mp->head = mp->tail = NULL;
	- mp->fr.single = -1;
	- mp->bsnum = 0;
	- mp->synth_bo = 1;
	-
	- if(!init) {
	- init = 1;
	- make_decode_tables(32767);
	- init_layer2();
	- init_layer3(SBLIMIT);
	- }
	-
	- return !0;
	-}
	-
	-void ExitMP3(struct mpstr *mp)
	-{
	- struct buf b,bn;
	-
	- b = mp->tail;
	- while(b) {
	- free(b->pnt);
	- bn = b->next;
	- free(b);
	- b = bn;
	- }
	-}
	-
	-static struct buf addbuf(struct mpstr mp,char *buf,int size)
	-{
	- struct buf *nbuf;
	-
	- nbuf = malloc( sizeof(struct buf) );
	- BAIL_IF_MACRO(!nbuf, ERR_OUT_OF_MEMORY, NULL);
	-
	- nbuf->pnt = malloc(size);
	- if(!nbuf->pnt) {
	- free(nbuf);
	- BAIL_MACRO(ERR_OUT_OF_MEMORY, NULL);
	- }
	- nbuf->size = size;
	- memcpy(nbuf->pnt,buf,size);
	- nbuf->next = NULL;
	- nbuf->prev = mp->head;
	- nbuf->pos = 0;
	-
	- if(!mp->tail) {
	- mp->tail = nbuf;
	- }
	- else {
	- mp->head->next = nbuf;
	- }
	-
	- mp->head = nbuf;
	- mp->bsize += size;
	-
	- return nbuf;
	-}
	-
	-static void remove_buf(struct mpstr *mp)
	-{
	- struct buf *buf = mp->tail;
	-
	- mp->tail = buf->next;
	- if(mp->tail)
	- mp->tail->prev = NULL;
	- else {
	- mp->tail = mp->head = NULL;
	- }
	-
	- free(buf->pnt);
	- free(buf);
	-
	-}
	-
	-static int read_buf_byte(struct mpstr mp, unsigned long retval)
	-{
	- int pos;
	-
	- pos = mp->tail->pos;
	- while(pos >= mp->tail->size) {
	- remove_buf(mp);
	- pos = mp->tail->pos;
	- if(!mp->tail) {
	- BAIL_MACRO("MPGLIB: Short read in read_buf_byte()!", 0);
	- }
	- }
	-
	- if (retval != NULL)
	- *retval = mp->tail->pnt[pos];
	-
	- mp->bsize--;
	- mp->tail->pos++;
	-
	- return 1;
	-}
	-
	-static int read_head(struct mpstr *mp)
	-{
	- unsigned long val;
	- unsigned long head;
	-
	- if (!read_buf_byte(mp, &val))
	- return 0;
	-
	- head = val << 8;
	-
	- if (!read_buf_byte(mp, &val))
	- return 0;
	-
	- head \|= val;
	- head <<= 8;
	-
	- if (!read_buf_byte(mp, &val))
	- return 0;
	-
	- head \|= val;
	- head <<= 8;
	-
	- if (!read_buf_byte(mp, &val))
	- return 0;
	-
	- head \|= val;
	- mp->header = head;
	- return 1;
	-}
	-
	-int decodeMP3(struct mpstr mp,char in,int isize,char *out,
	- int osize,int *done)
	-{
	- int len;
	-
	- BAIL_IF_MACRO(osize < 4608, "MPGLIB: Output buffer too small", MP3_ERR);
	-
	- if(in) {
	- if(addbuf(mp,in,isize) == NULL) {
	- return MP3_ERR;
	- }
	- }
	-
	- /* First decode header */
	- if(mp->framesize == 0) {
	- if(mp->bsize < 4) {
	- return MP3_NEED_MORE;
	- }
	-
	- if (!read_head(mp))
	- return MP3_ERR;
	-
	- if (!decode_header(&mp->fr,mp->header))
	- return MP3_ERR;
	-
	- mp->framesize = mp->fr.framesize;
	- }
	-
	- if(mp->fr.framesize > mp->bsize)
	- return MP3_NEED_MORE;
	-
	- wordpointer = mp->bsspace[mp->bsnum] + 512;
	- mp->bsnum = (mp->bsnum + 1) & 0x1;
	- bitindex = 0;
	-
	- len = 0;
	- while(len < mp->framesize) {
	- int nlen;
	- int blen = mp->tail->size - mp->tail->pos;
	- if( (mp->framesize - len) <= blen) {
	- nlen = mp->framesize-len;
	- }
	- else {
	- nlen = blen;
	- }
	- memcpy(wordpointer+len,mp->tail->pnt+mp->tail->pos,nlen);
	- len += nlen;
	- mp->tail->pos += nlen;
	- mp->bsize -= nlen;
	- if(mp->tail->pos == mp->tail->size) {
	- remove_buf(mp);
	- }
	- }
	-
	- *done = 0;
	- if(mp->fr.error_protection)
	- getbits(16);
	- switch(mp->fr.lay) {
	- case 1:
	- do_layer1(&mp->fr,(unsigned char *) out,done,mp);
	- break;
	- case 2:
	- do_layer2(&mp->fr,(unsigned char *) out,done,mp);
	- break;
	- case 3:
	- do_layer3(&mp->fr,(unsigned char *) out,done,mp);
	- break;
	- }
	-
	- mp->fsizeold = mp->framesize;
	- mp->framesize = 0;
	-
	- return MP3_OK;
	-}
	-
	-int set_pointer(long backstep, struct mpstr *mp)
	-{
	- unsigned char *bsbufold;
	- if(mp->fsizeold < 0 && backstep > 0) {
	- char err[128];
	- snprintf(err, sizeof (err), "MPGLIB: Can't step back! %ld!", backstep);
	- BAIL_MACRO(err, MP3_ERR);
	- }
	- bsbufold = mp->bsspace[mp->bsnum] + 512;
	- wordpointer -= backstep;
	- if (backstep)
	- memcpy(wordpointer,bsbufold+mp->fsizeold-backstep,backstep);
	- bitindex = 0;
	- return MP3_OK;
	-}
	-
	-
	-
	-
	diff --git a/util/sdl/sound/decoders/mpglib/l2tables.h b/util/sdl/sound/decoders/mpglib/l2tables.h
	deleted file mode 100644
	index 06d21353..00000000
	--- a/util/sdl/sound/decoders/mpglib/l2tables.h
	+++ /dev/null
	@@ -1,160 +0,0 @@
	-/*
	- * Layer 2 Alloc tables ..
	- * most other tables are calculated on program start (which is (of course)
	- * not ISO-conform) ..
	- * Layer-3 huffman table is in huffman.h
	- */
	-
	-struct al_table
	-{
	- short bits;
	- short d;
	-};
	-
	-struct al_table alloc_0[] = {
	- {4,0},{5,3},{3,-3},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},{9,-255},{10,-511},
	- {11,-1023},{12,-2047},{13,-4095},{14,-8191},{15,-16383},{16,-32767},
	- {4,0},{5,3},{3,-3},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},{9,-255},{10,-511},
	- {11,-1023},{12,-2047},{13,-4095},{14,-8191},{15,-16383},{16,-32767},
	- {4,0},{5,3},{3,-3},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},{9,-255},{10,-511},
	- {11,-1023},{12,-2047},{13,-4095},{14,-8191},{15,-16383},{16,-32767},
	- {4,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},
	- {9,-255},{10,-511},{11,-1023},{12,-2047},{13,-4095},{16,-32767},
	- {4,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},
	- {9,-255},{10,-511},{11,-1023},{12,-2047},{13,-4095},{16,-32767},
	- {4,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},
	- {9,-255},{10,-511},{11,-1023},{12,-2047},{13,-4095},{16,-32767},
	- {4,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},
	- {9,-255},{10,-511},{11,-1023},{12,-2047},{13,-4095},{16,-32767},
	- {4,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},
	- {9,-255},{10,-511},{11,-1023},{12,-2047},{13,-4095},{16,-32767},
	- {4,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},
	- {9,-255},{10,-511},{11,-1023},{12,-2047},{13,-4095},{16,-32767},
	- {4,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},
	- {9,-255},{10,-511},{11,-1023},{12,-2047},{13,-4095},{16,-32767},
	- {4,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},
	- {9,-255},{10,-511},{11,-1023},{12,-2047},{13,-4095},{16,-32767},
	- {3,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{16,-32767},
	- {3,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{16,-32767},
	- {3,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{16,-32767},
	- {3,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{16,-32767},
	- {3,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{16,-32767},
	- {3,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{16,-32767},
	- {3,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{16,-32767},
	- {3,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{16,-32767},
	- {3,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{16,-32767},
	- {3,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{16,-32767},
	- {3,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{16,-32767},
	- {3,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{16,-32767},
	- {2,0},{5,3},{7,5},{16,-32767},
	- {2,0},{5,3},{7,5},{16,-32767},
	- {2,0},{5,3},{7,5},{16,-32767},
	- {2,0},{5,3},{7,5},{16,-32767} };
	-
	-struct al_table alloc_1[] = {
	- {4,0},{5,3},{3,-3},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},{9,-255},{10,-511},
	- {11,-1023},{12,-2047},{13,-4095},{14,-8191},{15,-16383},{16,-32767},
	- {4,0},{5,3},{3,-3},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},{9,-255},{10,-511},
	- {11,-1023},{12,-2047},{13,-4095},{14,-8191},{15,-16383},{16,-32767},
	- {4,0},{5,3},{3,-3},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},{9,-255},{10,-511},
	- {11,-1023},{12,-2047},{13,-4095},{14,-8191},{15,-16383},{16,-32767},
	- {4,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},
	- {9,-255},{10,-511},{11,-1023},{12,-2047},{13,-4095},{16,-32767},
	- {4,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},
	- {9,-255},{10,-511},{11,-1023},{12,-2047},{13,-4095},{16,-32767},
	- {4,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},
	- {9,-255},{10,-511},{11,-1023},{12,-2047},{13,-4095},{16,-32767},
	- {4,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},
	- {9,-255},{10,-511},{11,-1023},{12,-2047},{13,-4095},{16,-32767},
	- {4,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},
	- {9,-255},{10,-511},{11,-1023},{12,-2047},{13,-4095},{16,-32767},
	- {4,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},
	- {9,-255},{10,-511},{11,-1023},{12,-2047},{13,-4095},{16,-32767},
	- {4,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},
	- {9,-255},{10,-511},{11,-1023},{12,-2047},{13,-4095},{16,-32767},
	- {4,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},
	- {9,-255},{10,-511},{11,-1023},{12,-2047},{13,-4095},{16,-32767},
	- {3,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{16,-32767},
	- {3,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{16,-32767},
	- {3,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{16,-32767},
	- {3,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{16,-32767},
	- {3,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{16,-32767},
	- {3,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{16,-32767},
	- {3,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{16,-32767},
	- {3,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{16,-32767},
	- {3,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{16,-32767},
	- {3,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{16,-32767},
	- {3,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{16,-32767},
	- {3,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{16,-32767},
	- {2,0},{5,3},{7,5},{16,-32767},
	- {2,0},{5,3},{7,5},{16,-32767},
	- {2,0},{5,3},{7,5},{16,-32767},
	- {2,0},{5,3},{7,5},{16,-32767},
	- {2,0},{5,3},{7,5},{16,-32767},
	- {2,0},{5,3},{7,5},{16,-32767},
	- {2,0},{5,3},{7,5},{16,-32767} };
	-
	-struct al_table alloc_2[] = {
	- {4,0},{5,3},{7,5},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},{9,-255},
	- {10,-511},{11,-1023},{12,-2047},{13,-4095},{14,-8191},{15,-16383},
	- {4,0},{5,3},{7,5},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},{9,-255},
	- {10,-511},{11,-1023},{12,-2047},{13,-4095},{14,-8191},{15,-16383},
	- {3,0},{5,3},{7,5},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},
	- {3,0},{5,3},{7,5},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},
	- {3,0},{5,3},{7,5},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},
	- {3,0},{5,3},{7,5},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},
	- {3,0},{5,3},{7,5},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},
	- {3,0},{5,3},{7,5},{10,9},{4,-7},{5,-15},{6,-31},{7,-63} };
	-
	-struct al_table alloc_3[] = {
	- {4,0},{5,3},{7,5},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},{9,-255},
	- {10,-511},{11,-1023},{12,-2047},{13,-4095},{14,-8191},{15,-16383},
	- {4,0},{5,3},{7,5},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},{9,-255},
	- {10,-511},{11,-1023},{12,-2047},{13,-4095},{14,-8191},{15,-16383},
	- {3,0},{5,3},{7,5},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},
	- {3,0},{5,3},{7,5},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},
	- {3,0},{5,3},{7,5},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},
	- {3,0},{5,3},{7,5},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},
	- {3,0},{5,3},{7,5},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},
	- {3,0},{5,3},{7,5},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},
	- {3,0},{5,3},{7,5},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},
	- {3,0},{5,3},{7,5},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},
	- {3,0},{5,3},{7,5},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},
	- {3,0},{5,3},{7,5},{10,9},{4,-7},{5,-15},{6,-31},{7,-63} };
	-
	-struct al_table alloc_4[] = {
	- {4,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},
	- {9,-255},{10,-511},{11,-1023},{12,-2047},{13,-4095},{14,-8191},
	- {4,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},
	- {9,-255},{10,-511},{11,-1023},{12,-2047},{13,-4095},{14,-8191},
	- {4,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},
	- {9,-255},{10,-511},{11,-1023},{12,-2047},{13,-4095},{14,-8191},
	- {4,0},{5,3},{7,5},{3,-3},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},{8,-127},
	- {9,-255},{10,-511},{11,-1023},{12,-2047},{13,-4095},{14,-8191},
	- {3,0},{5,3},{7,5},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},
	- {3,0},{5,3},{7,5},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},
	- {3,0},{5,3},{7,5},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},
	- {3,0},{5,3},{7,5},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},
	- {3,0},{5,3},{7,5},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},
	- {3,0},{5,3},{7,5},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},
	- {3,0},{5,3},{7,5},{10,9},{4,-7},{5,-15},{6,-31},{7,-63},
	- {2,0},{5,3},{7,5},{10,9},
	- {2,0},{5,3},{7,5},{10,9},
	- {2,0},{5,3},{7,5},{10,9},
	- {2,0},{5,3},{7,5},{10,9},
	- {2,0},{5,3},{7,5},{10,9},
	- {2,0},{5,3},{7,5},{10,9},
	- {2,0},{5,3},{7,5},{10,9},
	- {2,0},{5,3},{7,5},{10,9},
	- {2,0},{5,3},{7,5},{10,9},
	- {2,0},{5,3},{7,5},{10,9},
	- {2,0},{5,3},{7,5},{10,9},
	- {2,0},{5,3},{7,5},{10,9},
	- {2,0},{5,3},{7,5},{10,9},
	- {2,0},{5,3},{7,5},{10,9},
	- {2,0},{5,3},{7,5},{10,9},
	- {2,0},{5,3},{7,5},{10,9},
	- {2,0},{5,3},{7,5},{10,9},
	- {2,0},{5,3},{7,5},{10,9},
	- {2,0},{5,3},{7,5},{10,9} };
	-
	diff --git a/util/sdl/sound/decoders/mpglib/layer1.c b/util/sdl/sound/decoders/mpglib/layer1.c
	deleted file mode 100644
	index 3df430a8..00000000
	--- a/util/sdl/sound/decoders/mpglib/layer1.c
	+++ /dev/null
	@@ -1,148 +0,0 @@
	-/*
	- * Mpeg Layer-1 audio decoder
	- * --------------------------
	- * copyright (c) 1995 by Michael Hipp, All rights reserved. See also 'README'
	- * near unoptimzed ...
	- *
	- * may have a few bugs after last optimization ...
	- *
	- */
	-
	-#include "mpg123_sdlsound.h"
	-
	-void I_step_one(unsigned int balloc[], unsigned int scale_index[2][SBLIMIT],struct frame *fr)
	-{
	- unsigned int *ba=balloc;
	- unsigned int sca = (unsigned int ) scale_index;
	-
	- if(fr->stereo) {
	- int i;
	- int jsbound = fr->jsbound;
	- for (i=0;i<jsbound;i++) {
	- *ba++ = getbits(4);
	- *ba++ = getbits(4);
	- }
	- for (i=jsbound;i<SBLIMIT;i++)
	- *ba++ = getbits(4);
	-
	- ba = balloc;
	-
	- for (i=0;i<jsbound;i++) {
	- if ((*ba++))
	- *sca++ = getbits(6);
	- if ((*ba++))
	- *sca++ = getbits(6);
	- }
	- for (i=jsbound;i<SBLIMIT;i++)
	- if ((*ba++)) {
	- *sca++ = getbits(6);
	- *sca++ = getbits(6);
	- }
	- }
	- else {
	- int i;
	- for (i=0;i<SBLIMIT;i++)
	- *ba++ = getbits(4);
	- ba = balloc;
	- for (i=0;i<SBLIMIT;i++)
	- if ((*ba++))
	- *sca++ = getbits(6);
	- }
	-}
	-
	-void I_step_two(real fraction[2][SBLIMIT],unsigned int balloc[2*SBLIMIT],
	- unsigned int scale_index[2][SBLIMIT],struct frame *fr)
	-{
	- int i,n;
	- int smpb[2SBLIMIT]; / values: 0-65535 */
	- int *sample;
	- register unsigned int *ba;
	- register unsigned int sca = (unsigned int ) scale_index;
	-
	- if(fr->stereo) {
	- int jsbound = fr->jsbound;
	- register real *f0 = fraction[0];
	- register real *f1 = fraction[1];
	- ba = balloc;
	- for (sample=smpb,i=0;i<jsbound;i++) {
	- if ((n = *ba++))
	- *sample++ = getbits(n+1);
	- if ((n = *ba++))
	- *sample++ = getbits(n+1);
	- }
	- for (i=jsbound;i<SBLIMIT;i++)
	- if ((n = *ba++))
	- *sample++ = getbits(n+1);
	-
	- ba = balloc;
	- for (sample=smpb,i=0;i<jsbound;i++) {
	- if((n=*ba++))
	- f0++ = (real) ( ((-1)<<n) + (sample++) + 1) * muls[n+1][*sca++];
	- else
	- *f0++ = 0.0;
	- if((n=*ba++))
	- f1++ = (real) ( ((-1)<<n) + (sample++) + 1) * muls[n+1][*sca++];
	- else
	- *f1++ = 0.0;
	- }
	- for (i=jsbound;i<SBLIMIT;i++) {
	- if ((n=*ba++)) {
	- real samp = ( ((-1)<<n) + (*sample++) + 1);
	- f0++ = samp muls[n+1][*sca++];
	- f1++ = samp muls[n+1][*sca++];
	- }
	- else
	- f0++ = f1++ = 0.0;
	- }
	- }
	- else {
	- register real *f0 = fraction[0];
	- ba = balloc;
	- for (sample=smpb,i=0;i<SBLIMIT;i++)
	- if ((n = *ba++))
	- *sample++ = getbits(n+1);
	- ba = balloc;
	- for (sample=smpb,i=0;i<SBLIMIT;i++) {
	- if((n=*ba++))
	- f0++ = (real) ( ((-1)<<n) + (sample++) + 1) * muls[n+1][*sca++];
	- else
	- *f0++ = 0.0;
	- }
	- }
	-}
	-
	-int do_layer1(struct frame fr,unsigned char pcm_sample,
	- int pcm_point,struct mpstr mp)
	-{
	- int clip=0;
	- int i,stereo = fr->stereo;
	- unsigned int balloc[2*SBLIMIT];
	- unsigned int scale_index[2][SBLIMIT];
	- real fraction[2][SBLIMIT];
	- int single = fr->single;
	-
	- fr->jsbound = (fr->mode == MPG_MD_JOINT_STEREO) ? (fr->mode_ext<<2)+4 : 32;
	-
	- if(stereo == 1 \|\| single == 3)
	- single = 0;
	-
	- I_step_one(balloc,scale_index,fr);
	-
	- for (i=0;i<SCALE_BLOCK;i++)
	- {
	- I_step_two(fraction,balloc,scale_index,fr);
	-
	- if(single >= 0) {
	- clip += synth_1to1_mono( (real*)fraction[single],pcm_sample,pcm_point,mp);
	- }
	- else {
	- int p1 = *pcm_point;
	- clip += synth_1to1( (real*)fraction[0],0,pcm_sample,&p1,mp);
	- clip += synth_1to1( (real*)fraction[1],1,pcm_sample,pcm_point,mp);
	- }
	- }
	-
	- return clip;
	-}
	-
	-
	diff --git a/util/sdl/sound/decoders/mpglib/layer2.c b/util/sdl/sound/decoders/mpglib/layer2.c
	deleted file mode 100644
	index b5ef1e9b..00000000
	--- a/util/sdl/sound/decoders/mpglib/layer2.c
	+++ /dev/null
	@@ -1,289 +0,0 @@
	-/*
	- * Mpeg Layer-2 audio decoder
	- * --------------------------
	- * copyright (c) 1995 by Michael Hipp, All rights reserved. See also 'README'
	- *
	- */
	-
	-#include "mpg123_sdlsound.h"
	-#include "l2tables.h"
	-
	-static int grp_3tab[32 * 3] = { 0, }; /* used: 27 */
	-static int grp_5tab[128 * 3] = { 0, }; /* used: 125 */
	-static int grp_9tab[1024 * 3] = { 0, }; /* used: 729 */
	-
	-real muls[27][64]; /* also used by layer 1 */
	-
	-void init_layer2(void)
	-{
	- static double mulmul[27] = {
	- 0.0 , -2.0/3.0 , 2.0/3.0 ,
	- 2.0/7.0 , 2.0/15.0 , 2.0/31.0, 2.0/63.0 , 2.0/127.0 , 2.0/255.0 ,
	- 2.0/511.0 , 2.0/1023.0 , 2.0/2047.0 , 2.0/4095.0 , 2.0/8191.0 ,
	- 2.0/16383.0 , 2.0/32767.0 , 2.0/65535.0 ,
	- -4.0/5.0 , -2.0/5.0 , 2.0/5.0, 4.0/5.0 ,
	- -8.0/9.0 , -4.0/9.0 , -2.0/9.0 , 2.0/9.0 , 4.0/9.0 , 8.0/9.0 };
	- static int base[3][9] = {
	- { 1 , 0, 2 , } ,
	- { 17, 18, 0 , 19, 20 , } ,
	- { 21, 1, 22, 23, 0, 24, 25, 2, 26 } };
	- int i,j,k,l,len;
	- real *table;
	- static int tablen[3] = { 3 , 5 , 9 };
	- static int itable,tables[3] = { grp_3tab , grp_5tab , grp_9tab };
	-
	- for(i=0;i<3;i++)
	- {
	- itable = tables[i];
	- len = tablen[i];
	- for(j=0;j<len;j++)
	- for(k=0;k<len;k++)
	- for(l=0;l<len;l++)
	- {
	- *itable++ = base[i][l];
	- *itable++ = base[i][k];
	- *itable++ = base[i][j];
	- }
	- }
	-
	- for(k=0;k<27;k++)
	- {
	- double m=mulmul[k];
	- table = muls[k];
	- for(j=3,i=0;i<63;i++,j--)
	- table++ = m pow(2.0,(double) j / 3.0);
	- *table++ = 0.0;
	- }
	-}
	-
	-
	-void II_step_one(unsigned int bit_alloc,int scale,struct frame *fr)
	-{
	- int stereo = fr->stereo-1;
	- int sblimit = fr->II_sblimit;
	- int jsbound = fr->jsbound;
	- int sblimit2 = fr->II_sblimit<<stereo;
	- struct al_table *alloc1 = fr->alloc;
	- int i;
	- static unsigned int scfsi_buf[64];
	- unsigned int scfsi,bita;
	- int sc,step;
	-
	- bita = bit_alloc;
	- if(stereo)
	- {
	- for (i=jsbound;i;i--,alloc1+=(1<<step))
	- {
	- *bita++ = (char) getbits(step=alloc1->bits);
	- *bita++ = (char) getbits(step);
	- }
	- for (i=sblimit-jsbound;i;i--,alloc1+=(1<<step))
	- {
	- bita[0] = (char) getbits(step=alloc1->bits);
	- bita[1] = bita[0];
	- bita+=2;
	- }
	- bita = bit_alloc;
	- scfsi=scfsi_buf;
	- for (i=sblimit2;i;i--)
	- if (*bita++)
	- *scfsi++ = (char) getbits_fast(2);
	- }
	- else /* mono */
	- {
	- for (i=sblimit;i;i--,alloc1+=(1<<step))
	- *bita++ = (char) getbits(step=alloc1->bits);
	- bita = bit_alloc;
	- scfsi=scfsi_buf;
	- for (i=sblimit;i;i--)
	- if (*bita++)
	- *scfsi++ = (char) getbits_fast(2);
	- }
	-
	- bita = bit_alloc;
	- scfsi=scfsi_buf;
	- for (i=sblimit2;i;i--)
	- if (*bita++)
	- switch (*scfsi++)
	- {
	- case 0:
	- *scale++ = getbits_fast(6);
	- *scale++ = getbits_fast(6);
	- *scale++ = getbits_fast(6);
	- break;
	- case 1 :
	- *scale++ = sc = getbits_fast(6);
	- *scale++ = sc;
	- *scale++ = getbits_fast(6);
	- break;
	- case 2:
	- *scale++ = sc = getbits_fast(6);
	- *scale++ = sc;
	- *scale++ = sc;
	- break;
	- default: /* case 3 */
	- *scale++ = getbits_fast(6);
	- *scale++ = sc = getbits_fast(6);
	- *scale++ = sc;
	- break;
	- }
	-
	-}
	-
	-void II_step_two(unsigned int bit_alloc,real fraction[2][4][SBLIMIT],int scale,struct frame *fr,int x1)
	-{
	- int i,j,k,ba;
	- int stereo = fr->stereo;
	- int sblimit = fr->II_sblimit;
	- int jsbound = fr->jsbound;
	- struct al_table alloc2,alloc1 = fr->alloc;
	- unsigned int *bita=bit_alloc;
	- int d1,step;
	-
	- for (i=0;i<jsbound;i++,alloc1+=(1<<step))
	- {
	- step = alloc1->bits;
	- for (j=0;j<stereo;j++)
	- {
	- if ( (ba=*bita++) )
	- {
	- k=(alloc2 = alloc1+ba)->bits;
	- if( (d1=alloc2->d) < 0)
	- {
	- real cm=muls[k][scale[x1]];
	- fraction[j][0][i] = ((real) ((int)getbits(k) + d1)) * cm;
	- fraction[j][1][i] = ((real) ((int)getbits(k) + d1)) * cm;
	- fraction[j][2][i] = ((real) ((int)getbits(k) + d1)) * cm;
	- }
	- else
	- {
	- static int *table[] = { 0,0,0,grp_3tab,0,grp_5tab,0,0,0,grp_9tab };
	- unsigned int idx,*tab,m=scale[x1];
	- idx = (unsigned int) getbits(k);
	- tab = (unsigned int *) (table[d1] + idx + idx + idx);
	- fraction[j][0][i] = muls[*tab++][m];
	- fraction[j][1][i] = muls[*tab++][m];
	- fraction[j][2][i] = muls[*tab][m];
	- }
	- scale+=3;
	- }
	- else
	- fraction[j][0][i] = fraction[j][1][i] = fraction[j][2][i] = 0.0;
	- }
	- }
	-
	- for (i=jsbound;i<sblimit;i++,alloc1+=(1<<step))
	- {
	- step = alloc1->bits;
	- bita++; /* channel 1 and channel 2 bitalloc are the same */
	- if ( (ba=*bita++) )
	- {
	- k=(alloc2 = alloc1+ba)->bits;
	- if( (d1=alloc2->d) < 0)
	- {
	- real cm;
	- cm=muls[k][scale[x1+3]];
	- fraction[1][0][i] = (fraction[0][0][i] = (real) ((int)getbits(k) + d1) ) * cm;
	- fraction[1][1][i] = (fraction[0][1][i] = (real) ((int)getbits(k) + d1) ) * cm;
	- fraction[1][2][i] = (fraction[0][2][i] = (real) ((int)getbits(k) + d1) ) * cm;
	- cm=muls[k][scale[x1]];
	- fraction[0][0][i] = cm; fraction[0][1][i] = cm; fraction[0][2][i] *= cm;
	- }
	- else
	- {
	- static int *table[] = { 0,0,0,grp_3tab,0,grp_5tab,0,0,0,grp_9tab };
	- unsigned int idx,*tab,m1,m2;
	- m1 = scale[x1]; m2 = scale[x1+3];
	- idx = (unsigned int) getbits(k);
	- tab = (unsigned int *) (table[d1] + idx + idx + idx);
	- fraction[0][0][i] = muls[tab][m1]; fraction[1][0][i] = muls[tab++][m2];
	- fraction[0][1][i] = muls[tab][m1]; fraction[1][1][i] = muls[tab++][m2];
	- fraction[0][2][i] = muls[tab][m1]; fraction[1][2][i] = muls[tab][m2];
	- }
	- scale+=6;
	- }
	- else {
	- fraction[0][0][i] = fraction[0][1][i] = fraction[0][2][i] =
	- fraction[1][0][i] = fraction[1][1][i] = fraction[1][2][i] = 0.0;
	- }
	-/*
	- should we use individual scalefac for channel 2 or
	- is the current way the right one , where we just copy channel 1 to
	- channel 2 ??
	- The current 'strange' thing is, that we throw away the scalefac
	- values for the second channel ...!!
	--> changed .. now we use the scalefac values of channel one !!
	-*/
	- }
	-
	- for(i=sblimit;i<SBLIMIT;i++)
	- for (j=0;j<stereo;j++)
	- fraction[j][0][i] = fraction[j][1][i] = fraction[j][2][i] = 0.0;
	-
	-}
	-
	-static void II_select_table(struct frame *fr)
	-{
	- static int translate[3][2][16] =
	- { { { 0,2,2,2,2,2,2,0,0,0,1,1,1,1,1,0 } ,
	- { 0,2,2,0,0,0,1,1,1,1,1,1,1,1,1,0 } } ,
	- { { 0,2,2,2,2,2,2,0,0,0,0,0,0,0,0,0 } ,
	- { 0,2,2,0,0,0,0,0,0,0,0,0,0,0,0,0 } } ,
	- { { 0,3,3,3,3,3,3,0,0,0,1,1,1,1,1,0 } ,
	- { 0,3,3,0,0,0,1,1,1,1,1,1,1,1,1,0 } } };
	-
	- int table,sblim;
	- static struct al_table *tables[5] =
	- { alloc_0, alloc_1, alloc_2, alloc_3 , alloc_4 };
	- static int sblims[5] = { 27 , 30 , 8, 12 , 30 };
	-
	- if(fr->lsf)
	- table = 4;
	- else
	- table = translate[fr->sampling_frequency][2-fr->stereo][fr->bitrate_index];
	- sblim = sblims[table];
	-
	- fr->alloc = tables[table];
	- fr->II_sblimit = sblim;
	-}
	-
	-int do_layer2(struct frame fr,unsigned char pcm_sample,
	- int pcm_point,struct mpstr mp)
	-{
	- int clip=0;
	- int i,j;
	- int stereo = fr->stereo;
	- real fraction[2][4][SBLIMIT]; /* pick_table clears unused subbands */
	- unsigned int bit_alloc[64];
	- int scale[192];
	- int single = fr->single;
	-
	- II_select_table(fr);
	- fr->jsbound = (fr->mode == MPG_MD_JOINT_STEREO) ?
	- (fr->mode_ext<<2)+4 : fr->II_sblimit;
	-
	- if(stereo == 1 \|\| single == 3)
	- single = 0;
	-
	- II_step_one(bit_alloc, scale, fr);
	-
	- for (i=0;i<SCALE_BLOCK;i++)
	- {
	- II_step_two(bit_alloc,fraction,scale,fr,i>>2);
	- for (j=0;j<3;j++) {
	- if(single >= 0) {
	- clip += synth_1to1_mono(fraction[0][j],pcm_sample,pcm_point,mp);
	- }
	- else {
	- int p1 = *pcm_point;
	- clip += synth_1to1(fraction[0][j],0,pcm_sample,&p1,mp);
	- clip += synth_1to1(fraction[1][j],1,pcm_sample,pcm_point,mp);
	- }
	-
	- }
	- }
	-
	- return clip;
	-}
	-
	-
	diff --git a/util/sdl/sound/decoders/mpglib/layer3.c b/util/sdl/sound/decoders/mpglib/layer3.c
	deleted file mode 100644
	index 87abb19e..00000000
	--- a/util/sdl/sound/decoders/mpglib/layer3.c
	+++ /dev/null
	@@ -1,2020 +0,0 @@
	-/*
	- * Mpeg Layer-3 audio decoder
	- * --------------------------
	- * copyright (c) 1995,1996,1997 by Michael Hipp.
	- * All rights reserved. See also 'README'
	- */
	-
	-#include <stdio.h>
	-#include <stdlib.h>
	-
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-#include "mpg123_sdlsound.h"
	-#include "mpglib_sdlsound.h"
	-#include "huffman.h"
	-
	-#define MPEG1
	-
	-static real ispow[8207];
	-static real aa_ca[8],aa_cs[8];
	-static real COS1[12][6];
	-static real win[4][36];
	-static real win1[4][36];
	-static real gainpow2[256+118+4];
	-static real COS9[9];
	-static real COS6_1,COS6_2;
	-static real tfcos36[9];
	-static real tfcos12[3];
	-
	-struct bandInfoStruct {
	- short longIdx[23];
	- short longDiff[22];
	- short shortIdx[14];
	- short shortDiff[13];
	-};
	-
	-int longLimit[9][23];
	-int shortLimit[9][14];
	-
	-struct bandInfoStruct bandInfo[9] = {
	-
	-/* MPEG 1.0 */
	- { {0,4,8,12,16,20,24,30,36,44,52,62,74, 90,110,134,162,196,238,288,342,418,576},
	- {4,4,4,4,4,4,6,6,8, 8,10,12,16,20,24,28,34,42,50,54, 76,158},
	- {0,43,83,123,163,223,303,403,523,663, 843,1063,1363,192*3},
	- {4,4,4,4,6,8,10,12,14,18,22,30,56} } ,
	-
	- { {0,4,8,12,16,20,24,30,36,42,50,60,72, 88,106,128,156,190,230,276,330,384,576},
	- {4,4,4,4,4,4,6,6,6, 8,10,12,16,18,22,28,34,40,46,54, 54,192},
	- {0,43,83,123,163,223,283,383,503,643, 803,1003,1263,192*3},
	- {4,4,4,4,6,6,10,12,14,16,20,26,66} } ,
	-
	- { {0,4,8,12,16,20,24,30,36,44,54,66,82,102,126,156,194,240,296,364,448,550,576} ,
	- {4,4,4,4,4,4,6,6,8,10,12,16,20,24,30,38,46,56,68,84,102, 26} ,
	- {0,43,83,123,163,223,303,423,583,783,1043,1383,1803,192*3} ,
	- {4,4,4,4,6,8,12,16,20,26,34,42,12} } ,
	-
	-/* MPEG 2.0 */
	- { {0,6,12,18,24,30,36,44,54,66,80,96,116,140,168,200,238,284,336,396,464,522,576},
	- {6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54 } ,
	- {0,43,83,123,183,243,323,423,563,743,1003,1323,1743,192*3} ,
	- {4,4,4,6,6,8,10,14,18,26,32,42,18 } } ,
	-
	- { {0,6,12,18,24,30,36,44,54,66,80,96,114,136,162,194,232,278,330,394,464,540,576},
	- {6,6,6,6,6,6,8,10,12,14,16,18,22,26,32,38,46,52,64,70,76,36 } ,
	- {0,43,83,123,183,263,363,483,623,803,1043,1363,1803,192*3} ,
	- {4,4,4,6,8,10,12,14,18,24,32,44,12 } } ,
	-
	- { {0,6,12,18,24,30,36,44,54,66,80,96,116,140,168,200,238,284,336,396,464,522,576},
	- {6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54 },
	- {0,43,83,123,183,263,363,483,623,803,1043,1343,1743,192*3},
	- {4,4,4,6,8,10,12,14,18,24,30,40,18 } } ,
	-/* MPEG 2.5 */
	- { {0,6,12,18,24,30,36,44,54,66,80,96,116,140,168,200,238,284,336,396,464,522,576} ,
	- {6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54},
	- {0,12,24,36,54,78,108,144,186,240,312,402,522,576},
	- {4,4,4,6,8,10,12,14,18,24,30,40,18} },
	- { {0,6,12,18,24,30,36,44,54,66,80,96,116,140,168,200,238,284,336,396,464,522,576} ,
	- {6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54},
	- {0,12,24,36,54,78,108,144,186,240,312,402,522,576},
	- {4,4,4,6,8,10,12,14,18,24,30,40,18} },
	- { {0,12,24,36,48,60,72,88,108,132,160,192,232,280,336,400,476,566,568,570,572,574,576},
	- {12,12,12,12,12,12,16,20,24,28,32,40,48,56,64,76,90,2,2,2,2,2},
	- {0, 24, 48, 72,108,156,216,288,372,480,486,492,498,576},
	- {8,8,8,12,16,20,24,28,36,2,2,2,26} } ,
	-};
	-
	-static int mapbuf0[9][152];
	-static int mapbuf1[9][156];
	-static int mapbuf2[9][44];
	-static int *map[9][3];
	-static int *mapend[9][3];
	-
	-static unsigned int n_slen2[512]; /* MPEG 2.0 slen for 'normal' mode */
	-static unsigned int i_slen2[256]; /* MPEG 2.0 slen for intensity stereo */
	-
	-static real tan1_1[16],tan2_1[16],tan1_2[16],tan2_2[16];
	-static real pow1_1[2][16],pow2_1[2][16],pow1_2[2][16],pow2_2[2][16];
	-
	-/*
	- * init tables for layer-3
	- */
	-void init_layer3(int down_sample_sblimit)
	-{
	- int i,j,k,l;
	-
	- for(i=-256;i<118+4;i++)
	- gainpow2[i+256] = pow((double)2.0,-0.25 * (double) (i+210) );
	-
	- for(i=0;i<8207;i++)
	- ispow[i] = pow((double)i,(double)4.0/3.0);
	-
	- for (i=0;i<8;i++)
	- {
	- static double Ci[8]={-0.6,-0.535,-0.33,-0.185,-0.095,-0.041,-0.0142,-0.0037};
	- double sq=sqrt(1.0+Ci[i]*Ci[i]);
	- aa_cs[i] = 1.0/sq;
	- aa_ca[i] = Ci[i]/sq;
	- }
	-
	- for(i=0;i<18;i++)
	- {
	- win[0][i] = win[1][i] = 0.5 * sin( M_PI / 72.0 * (double) (2(i+0) +1) ) / cos ( M_PI (double) (2*(i+0) +19) / 72.0 );
	- win[0][i+18] = win[3][i+18] = 0.5 * sin( M_PI / 72.0 * (double) (2(i+18)+1) ) / cos ( M_PI (double) (2*(i+18)+19) / 72.0 );
	- }
	- for(i=0;i<6;i++)
	- {
	- win[1][i+18] = 0.5 / cos ( M_PI * (double) (2*(i+18)+19) / 72.0 );
	- win[3][i+12] = 0.5 / cos ( M_PI * (double) (2*(i+12)+19) / 72.0 );
	- win[1][i+24] = 0.5 * sin( M_PI / 24.0 * (double) (2i+13) ) / cos ( M_PI (double) (2*(i+24)+19) / 72.0 );
	- win[1][i+30] = win[3][i] = 0.0;
	- win[3][i+6 ] = 0.5 * sin( M_PI / 24.0 * (double) (2i+1) ) / cos ( M_PI (double) (2*(i+6 )+19) / 72.0 );
	- }
	-
	- for(i=0;i<9;i++)
	- COS9[i] = cos( M_PI / 18.0 * (double) i);
	-
	- for(i=0;i<9;i++)
	- tfcos36[i] = 0.5 / cos ( M_PI * (double) (i*2+1) / 36.0 );
	- for(i=0;i<3;i++)
	- tfcos12[i] = 0.5 / cos ( M_PI * (double) (i*2+1) / 12.0 );
	-
	- COS6_1 = cos( M_PI / 6.0 * (double) 1);
	- COS6_2 = cos( M_PI / 6.0 * (double) 2);
	-
	- for(i=0;i<12;i++)
	- {
	- win[2][i] = 0.5 * sin( M_PI / 24.0 * (double) (2i+1) ) / cos ( M_PI (double) (2*i+7) / 24.0 );
	- for(j=0;j<6;j++)
	- COS1[i][j] = cos( M_PI / 24.0 * (double) ((2i+7)(2*j+1)) );
	- }
	-
	- for(j=0;j<4;j++) {
	- static int len[4] = { 36,36,12,36 };
	- for(i=0;i<len[j];i+=2)
	- win1[j][i] = + win[j][i];
	- for(i=1;i<len[j];i+=2)
	- win1[j][i] = - win[j][i];
	- }
	-
	- for(i=0;i<16;i++)
	- {
	- double t = tan( (double) i * M_PI / 12.0 );
	- tan1_1[i] = t / (1.0+t);
	- tan2_1[i] = 1.0 / (1.0 + t);
	- tan1_2[i] = M_SQRT2 * t / (1.0+t);
	- tan2_2[i] = M_SQRT2 / (1.0 + t);
	-
	- for(j=0;j<2;j++) {
	- double base = pow(2.0,-0.25*(j+1.0));
	- double p1=1.0,p2=1.0;
	- if(i > 0) {
	- if( i & 1 )
	- p1 = pow(base,(i+1.0)*0.5);
	- else
	- p2 = pow(base,i*0.5);
	- }
	- pow1_1[j][i] = p1;
	- pow2_1[j][i] = p2;
	- pow1_2[j][i] = M_SQRT2 * p1;
	- pow2_2[j][i] = M_SQRT2 * p2;
	- }
	- }
	-
	- for(j=0;j<9;j++)
	- {
	- struct bandInfoStruct *bi = &bandInfo[j];
	- int *mp;
	- int cb,lwin;
	- short *bdf;
	-
	- mp = map[j][0] = mapbuf0[j];
	- bdf = bi->longDiff;
	- for(i=0,cb = 0; cb < 8 ; cb++,i+=*bdf++) {
	- mp++ = (bdf) >> 1;
	- *mp++ = i;
	- *mp++ = 3;
	- *mp++ = cb;
	- }
	- bdf = bi->shortDiff+3;
	- for(cb=3;cb<13;cb++) {
	- int l = (*bdf++) >> 1;
	- for(lwin=0;lwin<3;lwin++) {
	- *mp++ = l;
	- *mp++ = i + lwin;
	- *mp++ = lwin;
	- *mp++ = cb;
	- }
	- i += 6*l;
	- }
	- mapend[j][0] = mp;
	-
	- mp = map[j][1] = mapbuf1[j];
	- bdf = bi->shortDiff+0;
	- for(i=0,cb=0;cb<13;cb++) {
	- int l = (*bdf++) >> 1;
	- for(lwin=0;lwin<3;lwin++) {
	- *mp++ = l;
	- *mp++ = i + lwin;
	- *mp++ = lwin;
	- *mp++ = cb;
	- }
	- i += 6*l;
	- }
	- mapend[j][1] = mp;
	-
	- mp = map[j][2] = mapbuf2[j];
	- bdf = bi->longDiff;
	- for(cb = 0; cb < 22 ; cb++) {
	- mp++ = (bdf++) >> 1;
	- *mp++ = cb;
	- }
	- mapend[j][2] = mp;
	-
	- }
	-
	- for(j=0;j<9;j++) {
	- for(i=0;i<23;i++) {
	- longLimit[j][i] = (bandInfo[j].longIdx[i] - 1 + 8) / 18 + 1;
	- if(longLimit[j][i] > (down_sample_sblimit) )
	- longLimit[j][i] = down_sample_sblimit;
	- }
	- for(i=0;i<14;i++) {
	- shortLimit[j][i] = (bandInfo[j].shortIdx[i] - 1) / 18 + 1;
	- if(shortLimit[j][i] > (down_sample_sblimit) )
	- shortLimit[j][i] = down_sample_sblimit;
	- }
	- }
	-
	- for(i=0;i<5;i++) {
	- for(j=0;j<6;j++) {
	- for(k=0;k<6;k++) {
	- int n = k + j * 6 + i * 36;
	- i_slen2[n] = i\|(j<<3)\|(k<<6)\|(3<<12);
	- }
	- }
	- }
	- for(i=0;i<4;i++) {
	- for(j=0;j<4;j++) {
	- for(k=0;k<4;k++) {
	- int n = k + j * 4 + i * 16;
	- i_slen2[n+180] = i\|(j<<3)\|(k<<6)\|(4<<12);
	- }
	- }
	- }
	- for(i=0;i<4;i++) {
	- for(j=0;j<3;j++) {
	- int n = j + i * 3;
	- i_slen2[n+244] = i\|(j<<3) \| (5<<12);
	- n_slen2[n+500] = i\|(j<<3) \| (2<<12) \| (1<<15);
	- }
	- }
	-
	- for(i=0;i<5;i++) {
	- for(j=0;j<5;j++) {
	- for(k=0;k<4;k++) {
	- for(l=0;l<4;l++) {
	- int n = l + k * 4 + j * 16 + i * 80;
	- n_slen2[n] = i\|(j<<3)\|(k<<6)\|(l<<9)\|(0<<12);
	- }
	- }
	- }
	- }
	- for(i=0;i<5;i++) {
	- for(j=0;j<5;j++) {
	- for(k=0;k<4;k++) {
	- int n = k + j * 4 + i * 20;
	- n_slen2[n+400] = i\|(j<<3)\|(k<<6)\|(1<<12);
	- }
	- }
	- }
	-}
	-
	-/*
	- * read additional side information
	- */
	-#ifdef MPEG1
	-static int III_get_side_info_1(struct III_sideinfo *si,int stereo,
	- int ms_stereo,long sfreq,int single)
	-{
	- int ch, gr;
	- int powdiff = (single == 3) ? 4 : 0;
	-
	- si->main_data_begin = getbits(9);
	- if (stereo == 1)
	- si->private_bits = getbits_fast(5);
	- else
	- si->private_bits = getbits_fast(3);
	-
	- for (ch=0; ch<stereo; ch++) {
	- si->ch[ch].gr[0].scfsi = -1;
	- si->ch[ch].gr[1].scfsi = getbits_fast(4);
	- }
	-
	- for (gr=0; gr<2; gr++)
	- {
	- for (ch=0; ch<stereo; ch++)
	- {
	- register struct gr_info_s *gr_info = &(si->ch[ch].gr[gr]);
	-
	- gr_info->part2_3_length = getbits(12);
	- gr_info->big_values = getbits_fast(9);
	- if(gr_info->big_values > 288) {
	- SNDDBG(("MPGLIB: big_values too large!\n"));
	- gr_info->big_values = 288;
	- }
	- gr_info->pow2gain = gainpow2+256 - getbits_fast(8) + powdiff;
	- if(ms_stereo)
	- gr_info->pow2gain += 2;
	- gr_info->scalefac_compress = getbits_fast(4);
	-/* window-switching flag == 1 for block_Type != 0 .. and block-type == 0 -> win-sw-flag = 0 */
	- if(get1bit())
	- {
	- int i;
	- gr_info->block_type = getbits_fast(2);
	- gr_info->mixed_block_flag = get1bit();
	- gr_info->table_select[0] = getbits_fast(5);
	- gr_info->table_select[1] = getbits_fast(5);
	- /*
	- * table_select[2] not needed, because there is no region2,
	- * but to satisfy some verifications tools we set it either.
	- */
	- gr_info->table_select[2] = 0;
	- for(i=0;i<3;i++)
	- gr_info->full_gain[i] = gr_info->pow2gain + (getbits_fast(3)<<3);
	-
	- if(gr_info->block_type == 0) {
	- BAIL_MACRO("MPGLIB: Blocktype == 0 and window-switching == 1 not allowed.", 0);
	- }
	- /* region_count/start parameters are implicit in this case. */
	- gr_info->region1start = 36>>1;
	- gr_info->region2start = 576>>1;
	- }
	- else
	- {
	- int i,r0c,r1c;
	- for (i=0; i<3; i++)
	- gr_info->table_select[i] = getbits_fast(5);
	- r0c = getbits_fast(4);
	- r1c = getbits_fast(3);
	- gr_info->region1start = bandInfo[sfreq].longIdx[r0c+1] >> 1 ;
	- gr_info->region2start = bandInfo[sfreq].longIdx[r0c+1+r1c+1] >> 1;
	- gr_info->block_type = 0;
	- gr_info->mixed_block_flag = 0;
	- }
	- gr_info->preflag = get1bit();
	- gr_info->scalefac_scale = get1bit();
	- gr_info->count1table_select = get1bit();
	- }
	- }
	- return !0;
	-}
	-#endif
	-
	-/*
	- * Side Info for MPEG 2.0 / LSF
	- */
	-static int III_get_side_info_2(struct III_sideinfo *si,int stereo,
	- int ms_stereo,long sfreq,int single)
	-{
	- int ch;
	- int powdiff = (single == 3) ? 4 : 0;
	-
	- si->main_data_begin = getbits(8);
	- if (stereo == 1)
	- si->private_bits = get1bit();
	- else
	- si->private_bits = getbits_fast(2);
	-
	- for (ch=0; ch<stereo; ch++)
	- {
	- register struct gr_info_s *gr_info = &(si->ch[ch].gr[0]);
	-
	- gr_info->part2_3_length = getbits(12);
	- gr_info->big_values = getbits_fast(9);
	- if(gr_info->big_values > 288) {
	- SNDDBG(("MPGLIB: big_values too large!\n"));
	- gr_info->big_values = 288;
	- }
	- gr_info->pow2gain = gainpow2+256 - getbits_fast(8) + powdiff;
	- if(ms_stereo)
	- gr_info->pow2gain += 2;
	- gr_info->scalefac_compress = getbits(9);
	-/* window-switching flag == 1 for block_Type != 0 .. and block-type == 0 -> win-sw-flag = 0 */
	- if(get1bit())
	- {
	- int i;
	- gr_info->block_type = getbits_fast(2);
	- gr_info->mixed_block_flag = get1bit();
	- gr_info->table_select[0] = getbits_fast(5);
	- gr_info->table_select[1] = getbits_fast(5);
	- /*
	- * table_select[2] not needed, because there is no region2,
	- * but to satisfy some verifications tools we set it either.
	- */
	- gr_info->table_select[2] = 0;
	- for(i=0;i<3;i++)
	- gr_info->full_gain[i] = gr_info->pow2gain + (getbits_fast(3)<<3);
	-
	- if(gr_info->block_type == 0) {
	- BAIL_MACRO("MPGLIB: Blocktype == 0 and window-switching == 1 not allowed.", 0);
	- }
	- /* region_count/start parameters are implicit in this case. */
	-/* check this again! */
	- if(gr_info->block_type == 2)
	- gr_info->region1start = 36>>1;
	- else if(sfreq == 8)
	-/* check this for 2.5 and sfreq=8 */
	- gr_info->region1start = 108>>1;
	- else
	- gr_info->region1start = 54>>1;
	- gr_info->region2start = 576>>1;
	- }
	- else
	- {
	- int i,r0c,r1c;
	- for (i=0; i<3; i++)
	- gr_info->table_select[i] = getbits_fast(5);
	- r0c = getbits_fast(4);
	- r1c = getbits_fast(3);
	- gr_info->region1start = bandInfo[sfreq].longIdx[r0c+1] >> 1 ;
	- gr_info->region2start = bandInfo[sfreq].longIdx[r0c+1+r1c+1] >> 1;
	- gr_info->block_type = 0;
	- gr_info->mixed_block_flag = 0;
	- }
	- gr_info->scalefac_scale = get1bit();
	- gr_info->count1table_select = get1bit();
	- }
	- return !0;
	-}
	-
	-/*
	- * read scalefactors
	- */
	-#ifdef MPEG1
	-static int III_get_scale_factors_1(int scf,struct gr_info_s gr_info)
	-{
	- static const unsigned char slen[2][16] = {
	- {0, 0, 0, 0, 3, 1, 1, 1, 2, 2, 2, 3, 3, 3, 4, 4},
	- {0, 1, 2, 3, 0, 1, 2, 3, 1, 2, 3, 1, 2, 3, 2, 3}
	- };
	- int numbits;
	- int num0 = slen[0][gr_info->scalefac_compress];
	- int num1 = slen[1][gr_info->scalefac_compress];
	-
	- if (gr_info->block_type == 2) {
	- int i=18;
	- numbits = (num0 + num1) * 18;
	-
	- if (gr_info->mixed_block_flag) {
	- for (i=8;i;i--)
	- *scf++ = getbits_fast(num0);
	- i = 9;
	- numbits -= num0; /* num0 * 17 + num1 * 18 */
	- }
	-
	- for (;i;i--)
	- *scf++ = getbits_fast(num0);
	- for (i = 18; i; i--)
	- *scf++ = getbits_fast(num1);
	- scf++ = 0; scf++ = 0; scf++ = 0; / short[13][0..2] = 0 */
	- }
	- else {
	- int i;
	- int scfsi = gr_info->scfsi;
	-
	- if(scfsi < 0) { /* scfsi < 0 => granule == 0 */
	- for(i=11;i;i--)
	- *scf++ = getbits_fast(num0);
	- for(i=10;i;i--)
	- *scf++ = getbits_fast(num1);
	- numbits = (num0 + num1) * 10 + num0;
	- *scf++ = 0;
	- }
	- else {
	- numbits = 0;
	- if(!(scfsi & 0x8)) {
	- for (i=0;i<6;i++)
	- *scf++ = getbits_fast(num0);
	- numbits += num0 * 6;
	- }
	- else {
	- scf += 6;
	- }
	-
	- if(!(scfsi & 0x4)) {
	- for (i=0;i<5;i++)
	- *scf++ = getbits_fast(num0);
	- numbits += num0 * 5;
	- }
	- else {
	- scf += 5;
	- }
	-
	- if(!(scfsi & 0x2)) {
	- for(i=0;i<5;i++)
	- *scf++ = getbits_fast(num1);
	- numbits += num1 * 5;
	- }
	- else {
	- scf += 5;
	- }
	-
	- if(!(scfsi & 0x1)) {
	- for (i=0;i<5;i++)
	- *scf++ = getbits_fast(num1);
	- numbits += num1 * 5;
	- }
	- else {
	- scf += 5;
	- }
	- scf++ = 0; / no l[21] in original sources */
	- }
	- }
	- return numbits;
	-}
	-#endif
	-
	-
	-static int III_get_scale_factors_2(int scf,struct gr_info_s gr_info,int i_stereo)
	-{
	- unsigned char *pnt;
	- int i,j;
	- unsigned int slen;
	- int n = 0;
	- int numbits = 0;
	-
	- static unsigned char stab[3][6][4] = {
	- { { 6, 5, 5,5 } , { 6, 5, 7,3 } , { 11,10,0,0} ,
	- { 7, 7, 7,0 } , { 6, 6, 6,3 } , { 8, 8,5,0} } ,
	- { { 9, 9, 9,9 } , { 9, 9,12,6 } , { 18,18,0,0} ,
	- {12,12,12,0 } , {12, 9, 9,6 } , { 15,12,9,0} } ,
	- { { 6, 9, 9,9 } , { 6, 9,12,6 } , { 15,18,0,0} ,
	- { 6,15,12,0 } , { 6,12, 9,6 } , { 6,18,9,0} } };
	-
	- if(i_stereo) /* i_stereo AND second channel -> do_layer3() checks this */
	- slen = i_slen2[gr_info->scalefac_compress>>1];
	- else
	- slen = n_slen2[gr_info->scalefac_compress];
	-
	- gr_info->preflag = (slen>>15) & 0x1;
	-
	- n = 0;
	- if( gr_info->block_type == 2 ) {
	- n++;
	- if(gr_info->mixed_block_flag)
	- n++;
	- }
	-
	- pnt = stab[n][(slen>>12)&0x7];
	-
	- for(i=0;i<4;i++) {
	- int num = slen & 0x7;
	- slen >>= 3;
	- if(num) {
	- for(j=0;j<(int)(pnt[i]);j++)
	- *scf++ = getbits_fast(num);
	- numbits += pnt[i] * num;
	- }
	- else {
	- for(j=0;j<(int)(pnt[i]);j++)
	- *scf++ = 0;
	- }
	- }
	-
	- n = (n << 1) + 1;
	- for(i=0;i<n;i++)
	- *scf++ = 0;
	-
	- return numbits;
	-}
	-
	-static int pretab1[22] = {0,0,0,0,0,0,0,0,0,0,0,1,1,1,1,2,2,3,3,3,2,0};
	-static int pretab2[22] = {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0};
	-
	-/*
	- * don't forget to apply the same changes to III_dequantize_sample_ms() !!!
	- */
	-static int III_dequantize_sample(real xr[SBLIMIT][SSLIMIT],int *scf,
	- struct gr_info_s *gr_info,int sfreq,int part2bits)
	-{
	- int shift = 1 + gr_info->scalefac_scale;
	- real xrpnt = (real ) xr;
	- int l[3],l3;
	- int part2remain = gr_info->part2_3_length - part2bits;
	- int *me;
	-
	- {
	- int bv = gr_info->big_values;
	- int region1 = gr_info->region1start;
	- int region2 = gr_info->region2start;
	-
	- l3 = ((576>>1)-bv)>>1;
	-/*
	- * we may lose the 'odd' bit here !!
	- * check this later again
	- */
	- if(bv <= region1) {
	- l[0] = bv; l[1] = 0; l[2] = 0;
	- }
	- else {
	- l[0] = region1;
	- if(bv <= region2) {
	- l[1] = bv - l[0]; l[2] = 0;
	- }
	- else {
	- l[1] = region2 - l[0]; l[2] = bv - region2;
	- }
	- }
	- }
	-
	- if(gr_info->block_type == 2) {
	- /*
	- * decoding with short or mixed mode BandIndex table
	- */
	- int i,max[4];
	- int step=0,lwin=0,cb=0;
	- register real v = 0.0;
	- register int *m,mc;
	-
	- if(gr_info->mixed_block_flag) {
	- max[3] = -1;
	- max[0] = max[1] = max[2] = 2;
	- m = map[sfreq][0];
	- me = mapend[sfreq][0];
	- }
	- else {
	- max[0] = max[1] = max[2] = max[3] = -1;
	- /* max[3] not really needed in this case */
	- m = map[sfreq][1];
	- me = mapend[sfreq][1];
	- }
	-
	- mc = 0;
	- for(i=0;i<2;i++) {
	- int lp = l[i];
	- struct newhuff *h = ht+gr_info->table_select[i];
	- for(;lp;lp--,mc--) {
	- register int x,y;
	- if( (!mc) ) {
	- mc = *m++;
	- xrpnt = ((real ) xr) + (m++);
	- lwin = *m++;
	- cb = *m++;
	- if(lwin == 3) {
	- v = gr_info->pow2gain[(*scf++) << shift];
	- step = 1;
	- }
	- else {
	- v = gr_info->full_gain[lwin][(*scf++) << shift];
	- step = 3;
	- }
	- }
	- {
	- register short *val = h->table;
	- while((y=*val++)<0) {
	- if (get1bit())
	- val -= y;
	- part2remain--;
	- }
	- x = y >> 4;
	- y &= 0xf;
	- }
	- if(x == 15) {
	- max[lwin] = cb;
	- part2remain -= h->linbits+1;
	- x += getbits(h->linbits);
	- if(get1bit())
	- xrpnt = -ispow[x] v;
	- else
	- xrpnt = ispow[x] v;
	- }
	- else if(x) {
	- max[lwin] = cb;
	- if(get1bit())
	- xrpnt = -ispow[x] v;
	- else
	- xrpnt = ispow[x] v;
	- part2remain--;
	- }
	- else
	- *xrpnt = 0.0;
	- xrpnt += step;
	- if(y == 15) {
	- max[lwin] = cb;
	- part2remain -= h->linbits+1;
	- y += getbits(h->linbits);
	- if(get1bit())
	- xrpnt = -ispow[y] v;
	- else
	- xrpnt = ispow[y] v;
	- }
	- else if(y) {
	- max[lwin] = cb;
	- if(get1bit())
	- xrpnt = -ispow[y] v;
	- else
	- xrpnt = ispow[y] v;
	- part2remain--;
	- }
	- else
	- *xrpnt = 0.0;
	- xrpnt += step;
	- }
	- }
	- for(;l3 && (part2remain > 0);l3--) {
	- struct newhuff *h = htc+gr_info->count1table_select;
	- register short *val = h->table,a;
	-
	- while((a=*val++)<0) {
	- part2remain--;
	- if(part2remain < 0) {
	- part2remain++;
	- a = 0;
	- break;
	- }
	- if (get1bit())
	- val -= a;
	- }
	-
	- for(i=0;i<4;i++) {
	- if(!(i & 1)) {
	- if(!mc) {
	- mc = *m++;
	- xrpnt = ((real ) xr) + (m++);
	- lwin = *m++;
	- cb = *m++;
	- if(lwin == 3) {
	- v = gr_info->pow2gain[(*scf++) << shift];
	- step = 1;
	- }
	- else {
	- v = gr_info->full_gain[lwin][(*scf++) << shift];
	- step = 3;
	- }
	- }
	- mc--;
	- }
	- if( (a & (0x8>>i)) ) {
	- max[lwin] = cb;
	- part2remain--;
	- if(part2remain < 0) {
	- part2remain++;
	- break;
	- }
	- if(get1bit())
	- *xrpnt = -v;
	- else
	- *xrpnt = v;
	- }
	- else
	- *xrpnt = 0.0;
	- xrpnt += step;
	- }
	- }
	-
	- while( m < me ) {
	- if(!mc) {
	- mc = *m++;
	- xrpnt = ((real ) xr) + m++;
	- if( (*m++) == 3)
	- step = 1;
	- else
	- step = 3;
	- m++; /* cb */
	- }
	- mc--;
	- *xrpnt = 0.0;
	- xrpnt += step;
	- *xrpnt = 0.0;
	- xrpnt += step;
	-/* we could add a little opt. here:
	- * if we finished a band for window 3 or a long band
	- * further bands could copied in a simple loop without a
	- * special 'map' decoding
	- */
	- }
	-
	- gr_info->maxband[0] = max[0]+1;
	- gr_info->maxband[1] = max[1]+1;
	- gr_info->maxband[2] = max[2]+1;
	- gr_info->maxbandl = max[3]+1;
	-
	- {
	- int rmax = max[0] > max[1] ? max[0] : max[1];
	- rmax = (rmax > max[2] ? rmax : max[2]) + 1;
	- gr_info->maxb = rmax ? shortLimit[sfreq][rmax] : longLimit[sfreq][max[3]+1];
	- }
	-
	- }
	- else {
	- /*
	- * decoding with 'long' BandIndex table (block_type != 2)
	- */
	- int *pretab = gr_info->preflag ? pretab1 : pretab2;
	- int i,max = -1;
	- int cb = 0;
	- register int *m = map[sfreq][2];
	- register real v = 0.0;
	- register int mc = 0;
	-#if 0
	- me = mapend[sfreq][2];
	-#endif
	-
	- /*
	- * long hash table values
	- */
	- for(i=0;i<3;i++) {
	- int lp = l[i];
	- struct newhuff *h = ht+gr_info->table_select[i];
	-
	- for(;lp;lp--,mc--) {
	- int x,y;
	-
	- if(!mc) {
	- mc = *m++;
	- v = gr_info->pow2gain[((scf++) + (pretab++)) << shift];
	- cb = *m++;
	- }
	- {
	- register short *val = h->table;
	- while((y=*val++)<0) {
	- if (get1bit())
	- val -= y;
	- part2remain--;
	- }
	- x = y >> 4;
	- y &= 0xf;
	- }
	- if (x == 15) {
	- max = cb;
	- part2remain -= h->linbits+1;
	- x += getbits(h->linbits);
	- if(get1bit())
	- xrpnt++ = -ispow[x] v;
	- else
	- xrpnt++ = ispow[x] v;
	- }
	- else if(x) {
	- max = cb;
	- if(get1bit())
	- xrpnt++ = -ispow[x] v;
	- else
	- xrpnt++ = ispow[x] v;
	- part2remain--;
	- }
	- else
	- *xrpnt++ = 0.0;
	-
	- if (y == 15) {
	- max = cb;
	- part2remain -= h->linbits+1;
	- y += getbits(h->linbits);
	- if(get1bit())
	- xrpnt++ = -ispow[y] v;
	- else
	- xrpnt++ = ispow[y] v;
	- }
	- else if(y) {
	- max = cb;
	- if(get1bit())
	- xrpnt++ = -ispow[y] v;
	- else
	- xrpnt++ = ispow[y] v;
	- part2remain--;
	- }
	- else
	- *xrpnt++ = 0.0;
	- }
	- }
	-
	- /*
	- * short (count1table) values
	- */
	- for(;l3 && (part2remain > 0);l3--) {
	- struct newhuff *h = htc+gr_info->count1table_select;
	- register short *val = h->table,a;
	-
	- while((a=*val++)<0) {
	- part2remain--;
	- if(part2remain < 0) {
	- part2remain++;
	- a = 0;
	- break;
	- }
	- if (get1bit())
	- val -= a;
	- }
	-
	- for(i=0;i<4;i++) {
	- if(!(i & 1)) {
	- if(!mc) {
	- mc = *m++;
	- cb = *m++;
	- v = gr_info->pow2gain[((scf++) + (pretab++)) << shift];
	- }
	- mc--;
	- }
	- if ( (a & (0x8>>i)) ) {
	- max = cb;
	- part2remain--;
	- if(part2remain < 0) {
	- part2remain++;
	- break;
	- }
	- if(get1bit())
	- *xrpnt++ = -v;
	- else
	- *xrpnt++ = v;
	- }
	- else
	- *xrpnt++ = 0.0;
	- }
	- }
	-
	- /*
	- * zero part
	- */
	- for(i=(&xr[SBLIMIT][0]-xrpnt)>>1;i;i--) {
	- *xrpnt++ = 0.0;
	- *xrpnt++ = 0.0;
	- }
	-
	- gr_info->maxbandl = max+1;
	- gr_info->maxb = longLimit[sfreq][gr_info->maxbandl];
	- }
	-
	- while( part2remain > 16 ) {
	- getbits(16); /* Dismiss stuffing Bits */
	- part2remain -= 16;
	- }
	- if(part2remain > 0)
	- getbits(part2remain);
	- else if(part2remain < 0) {
	- char err[128];
	- snprintf(err, sizeof (err),
	- "MPGLIB: Can't rewind stream by %d bits!",
	- -part2remain);
	- BAIL_MACRO(err, 1); /* -> error */
	- }
	- return 0;
	-}
	-
	-#if 0
	-static int III_dequantize_sample_ms(real xr[2][SBLIMIT][SSLIMIT],int *scf,
	- struct gr_info_s *gr_info,int sfreq,int part2bits)
	-{
	- int shift = 1 + gr_info->scalefac_scale;
	- real xrpnt = (real ) xr[1];
	- real xr0pnt = (real ) xr[0];
	- int l[3],l3;
	- int part2remain = gr_info->part2_3_length - part2bits;
	- int *me;
	-
	- {
	- int bv = gr_info->big_values;
	- int region1 = gr_info->region1start;
	- int region2 = gr_info->region2start;
	-
	- l3 = ((576>>1)-bv)>>1;
	-/*
	- * we may lose the 'odd' bit here !!
	- * check this later gain
	- */
	- if(bv <= region1) {
	- l[0] = bv; l[1] = 0; l[2] = 0;
	- }
	- else {
	- l[0] = region1;
	- if(bv <= region2) {
	- l[1] = bv - l[0]; l[2] = 0;
	- }
	- else {
	- l[1] = region2 - l[0]; l[2] = bv - region2;
	- }
	- }
	- }
	-
	- if(gr_info->block_type == 2) {
	- int i,max[4];
	- int step=0,lwin=0,cb=0;
	- register real v = 0.0;
	- register int *m,mc = 0;
	-
	- if(gr_info->mixed_block_flag) {
	- max[3] = -1;
	- max[0] = max[1] = max[2] = 2;
	- m = map[sfreq][0];
	- me = mapend[sfreq][0];
	- }
	- else {
	- max[0] = max[1] = max[2] = max[3] = -1;
	- /* max[3] not really needed in this case */
	- m = map[sfreq][1];
	- me = mapend[sfreq][1];
	- }
	-
	- for(i=0;i<2;i++) {
	- int lp = l[i];
	- struct newhuff *h = ht+gr_info->table_select[i];
	- for(;lp;lp--,mc--) {
	- int x,y;
	-
	- if(!mc) {
	- mc = *m++;
	- xrpnt = ((real ) xr[1]) + m;
	- xr0pnt = ((real ) xr[0]) + m++;
	- lwin = *m++;
	- cb = *m++;
	- if(lwin == 3) {
	- v = gr_info->pow2gain[(*scf++) << shift];
	- step = 1;
	- }
	- else {
	- v = gr_info->full_gain[lwin][(*scf++) << shift];
	- step = 3;
	- }
	- }
	- {
	- register short *val = h->table;
	- while((y=*val++)<0) {
	- if (get1bit())
	- val -= y;
	- part2remain--;
	- }
	- x = y >> 4;
	- y &= 0xf;
	- }
	- if(x == 15) {
	- max[lwin] = cb;
	- part2remain -= h->linbits+1;
	- x += getbits(h->linbits);
	- if(get1bit()) {
	- real a = ispow[x] * v;
	- xrpnt = xr0pnt + a;
	- *xr0pnt -= a;
	- }
	- else {
	- real a = ispow[x] * v;
	- xrpnt = xr0pnt - a;
	- *xr0pnt += a;
	- }
	- }
	- else if(x) {
	- max[lwin] = cb;
	- if(get1bit()) {
	- real a = ispow[x] * v;
	- xrpnt = xr0pnt + a;
	- *xr0pnt -= a;
	- }
	- else {
	- real a = ispow[x] * v;
	- xrpnt = xr0pnt - a;
	- *xr0pnt += a;
	- }
	- part2remain--;
	- }
	- else
	- xrpnt = xr0pnt;
	- xrpnt += step;
	- xr0pnt += step;
	-
	- if(y == 15) {
	- max[lwin] = cb;
	- part2remain -= h->linbits+1;
	- y += getbits(h->linbits);
	- if(get1bit()) {
	- real a = ispow[y] * v;
	- xrpnt = xr0pnt + a;
	- *xr0pnt -= a;
	- }
	- else {
	- real a = ispow[y] * v;
	- xrpnt = xr0pnt - a;
	- *xr0pnt += a;
	- }
	- }
	- else if(y) {
	- max[lwin] = cb;
	- if(get1bit()) {
	- real a = ispow[y] * v;
	- xrpnt = xr0pnt + a;
	- *xr0pnt -= a;
	- }
	- else {
	- real a = ispow[y] * v;
	- xrpnt = xr0pnt - a;
	- *xr0pnt += a;
	- }
	- part2remain--;
	- }
	- else
	- xrpnt = xr0pnt;
	- xrpnt += step;
	- xr0pnt += step;
	- }
	- }
	-
	- for(;l3 && (part2remain > 0);l3--) {
	- struct newhuff *h = htc+gr_info->count1table_select;
	- register short *val = h->table,a;
	-
	- while((a=*val++)<0) {
	- part2remain--;
	- if(part2remain < 0) {
	- part2remain++;
	- a = 0;
	- break;
	- }
	- if (get1bit())
	- val -= a;
	- }
	-
	- for(i=0;i<4;i++) {
	- if(!(i & 1)) {
	- if(!mc) {
	- mc = *m++;
	- xrpnt = ((real ) xr[1]) + m;
	- xr0pnt = ((real ) xr[0]) + m++;
	- lwin = *m++;
	- cb = *m++;
	- if(lwin == 3) {
	- v = gr_info->pow2gain[(*scf++) << shift];
	- step = 1;
	- }
	- else {
	- v = gr_info->full_gain[lwin][(*scf++) << shift];
	- step = 3;
	- }
	- }
	- mc--;
	- }
	- if( (a & (0x8>>i)) ) {
	- max[lwin] = cb;
	- part2remain--;
	- if(part2remain < 0) {
	- part2remain++;
	- break;
	- }
	- if(get1bit()) {
	- xrpnt = xr0pnt + v;
	- *xr0pnt -= v;
	- }
	- else {
	- xrpnt = xr0pnt - v;
	- *xr0pnt += v;
	- }
	- }
	- else
	- xrpnt = xr0pnt;
	- xrpnt += step;
	- xr0pnt += step;
	- }
	- }
	-
	- while( m < me ) {
	- if(!mc) {
	- mc = *m++;
	- xrpnt = ((real ) xr[1]) + m;
	- xr0pnt = ((real ) xr[0]) + m++;
	- if(*m++ == 3)
	- step = 1;
	- else
	- step = 3;
	- m++; /* cb */
	- }
	- mc--;
	- xrpnt = xr0pnt;
	- xrpnt += step;
	- xr0pnt += step;
	- xrpnt = xr0pnt;
	- xrpnt += step;
	- xr0pnt += step;
	-/* we could add a little opt. here:
	- * if we finished a band for window 3 or a long band
	- * further bands could copied in a simple loop without a
	- * special 'map' decoding
	- */
	- }
	-
	- gr_info->maxband[0] = max[0]+1;
	- gr_info->maxband[1] = max[1]+1;
	- gr_info->maxband[2] = max[2]+1;
	- gr_info->maxbandl = max[3]+1;
	-
	- {
	- int rmax = max[0] > max[1] ? max[0] : max[1];
	- rmax = (rmax > max[2] ? rmax : max[2]) + 1;
	- gr_info->maxb = rmax ? shortLimit[sfreq][rmax] : longLimit[sfreq][max[3]+1];
	- }
	- }
	- else {
	- int *pretab = gr_info->preflag ? pretab1 : pretab2;
	- int i,max = -1;
	- int cb = 0;
	- register int mc=0,*m = map[sfreq][2];
	- register real v = 0.0;
	-#if 0
	- me = mapend[sfreq][2];
	-#endif
	-
	- for(i=0;i<3;i++) {
	- int lp = l[i];
	- struct newhuff *h = ht+gr_info->table_select[i];
	-
	- for(;lp;lp--,mc--) {
	- int x,y;
	- if(!mc) {
	- mc = *m++;
	- cb = *m++;
	- v = gr_info->pow2gain[((scf++) + (pretab++)) << shift];
	- }
	- {
	- register short *val = h->table;
	- while((y=*val++)<0) {
	- if (get1bit())
	- val -= y;
	- part2remain--;
	- }
	- x = y >> 4;
	- y &= 0xf;
	- }
	- if (x == 15) {
	- max = cb;
	- part2remain -= h->linbits+1;
	- x += getbits(h->linbits);
	- if(get1bit()) {
	- real a = ispow[x] * v;
	- xrpnt++ = xr0pnt + a;
	- *xr0pnt++ -= a;
	- }
	- else {
	- real a = ispow[x] * v;
	- xrpnt++ = xr0pnt - a;
	- *xr0pnt++ += a;
	- }
	- }
	- else if(x) {
	- max = cb;
	- if(get1bit()) {
	- real a = ispow[x] * v;
	- xrpnt++ = xr0pnt + a;
	- *xr0pnt++ -= a;
	- }
	- else {
	- real a = ispow[x] * v;
	- xrpnt++ = xr0pnt - a;
	- *xr0pnt++ += a;
	- }
	- part2remain--;
	- }
	- else
	- xrpnt++ = xr0pnt++;
	-
	- if (y == 15) {
	- max = cb;
	- part2remain -= h->linbits+1;
	- y += getbits(h->linbits);
	- if(get1bit()) {
	- real a = ispow[y] * v;
	- xrpnt++ = xr0pnt + a;
	- *xr0pnt++ -= a;
	- }
	- else {
	- real a = ispow[y] * v;
	- xrpnt++ = xr0pnt - a;
	- *xr0pnt++ += a;
	- }
	- }
	- else if(y) {
	- max = cb;
	- if(get1bit()) {
	- real a = ispow[y] * v;
	- xrpnt++ = xr0pnt + a;
	- *xr0pnt++ -= a;
	- }
	- else {
	- real a = ispow[y] * v;
	- xrpnt++ = xr0pnt - a;
	- *xr0pnt++ += a;
	- }
	- part2remain--;
	- }
	- else
	- xrpnt++ = xr0pnt++;
	- }
	- }
	-
	- for(;l3 && (part2remain > 0);l3--) {
	- struct newhuff *h = htc+gr_info->count1table_select;
	- register short *val = h->table,a;
	-
	- while((a=*val++)<0) {
	- part2remain--;
	- if(part2remain < 0) {
	- part2remain++;
	- a = 0;
	- break;
	- }
	- if (get1bit())
	- val -= a;
	- }
	-
	- for(i=0;i<4;i++) {
	- if(!(i & 1)) {
	- if(!mc) {
	- mc = *m++;
	- cb = *m++;
	- v = gr_info->pow2gain[((scf++) + (pretab++)) << shift];
	- }
	- mc--;
	- }
	- if ( (a & (0x8>>i)) ) {
	- max = cb;
	- part2remain--;
	- if(part2remain <= 0) {
	- part2remain++;
	- break;
	- }
	- if(get1bit()) {
	- xrpnt++ = xr0pnt + v;
	- *xr0pnt++ -= v;
	- }
	- else {
	- xrpnt++ = xr0pnt - v;
	- *xr0pnt++ += v;
	- }
	- }
	- else
	- xrpnt++ = xr0pnt++;
	- }
	- }
	- for(i=(&xr[1][SBLIMIT][0]-xrpnt)>>1;i;i--) {
	- xrpnt++ = xr0pnt++;
	- xrpnt++ = xr0pnt++;
	- }
	-
	- gr_info->maxbandl = max+1;
	- gr_info->maxb = longLimit[sfreq][gr_info->maxbandl];
	- }
	-
	- while ( part2remain > 16 ) {
	- getbits(16); /* Dismiss stuffing Bits */
	- part2remain -= 16;
	- }
	- if(part2remain > 0 )
	- getbits(part2remain);
	- else if(part2remain < 0) {
	- char err[128];
	- snprintf(err, sizeof (err),
	- "MPGLIB: Can't rewind stream by %d bits!",
	- -part2remain);
	- BAIL_MACRO(err, 1); /* -> error */
	- }
	- return 0;
	-}
	-#endif
	-
	-/*
	- * III_stereo: calculate real channel values for Joint-I-Stereo-mode
	- */
	-static void III_i_stereo(real xr_buf[2][SBLIMIT][SSLIMIT],int *scalefac,
	- struct gr_info_s *gr_info,int sfreq,int ms_stereo,int lsf)
	-{
	- real (xr)[SBLIMITSSLIMIT] = (real ()[SBLIMITSSLIMIT] ) xr_buf;
	- struct bandInfoStruct *bi = &bandInfo[sfreq];
	- real tab1,tab2;
	-
	- if(lsf) {
	- int p = gr_info->scalefac_compress & 0x1;
	- if(ms_stereo) {
	- tab1 = pow1_2[p]; tab2 = pow2_2[p];
	- }
	- else {
	- tab1 = pow1_1[p]; tab2 = pow2_1[p];
	- }
	- }
	- else {
	- if(ms_stereo) {
	- tab1 = tan1_2; tab2 = tan2_2;
	- }
	- else {
	- tab1 = tan1_1; tab2 = tan2_1;
	- }
	- }
	-
	- if (gr_info->block_type == 2)
	- {
	- int lwin,do_l = 0;
	- if( gr_info->mixed_block_flag )
	- do_l = 1;
	-
	- for (lwin=0;lwin<3;lwin++) /* process each window */
	- {
	- /* get first band with zero values */
	- int is_p,sb,idx,sfb = gr_info->maxband[lwin]; /* sfb is minimal 3 for mixed mode */
	- if(sfb > 3)
	- do_l = 0;
	-
	- for(;sfb<12;sfb++)
	- {
	- is_p = scalefac[sfb3+lwin-gr_info->mixed_block_flag]; / scale: 0-15 */
	- if(is_p != 7) {
	- real t1,t2;
	- sb = bi->shortDiff[sfb];
	- idx = bi->shortIdx[sfb] + lwin;
	- t1 = tab1[is_p]; t2 = tab2[is_p];
	- for (; sb > 0; sb--,idx+=3)
	- {
	- real v = xr[0][idx];
	- xr[0][idx] = v * t1;
	- xr[1][idx] = v * t2;
	- }
	- }
	- }
	-
	-#if 1
	-/* in the original: copy 10 to 11 , here: copy 11 to 12
	-maybe still wrong??? (copy 12 to 13?) */
	- is_p = scalefac[113+lwin-gr_info->mixed_block_flag]; / scale: 0-15 */
	- sb = bi->shortDiff[12];
	- idx = bi->shortIdx[12] + lwin;
	-#else
	- is_p = scalefac[103+lwin-gr_info->mixed_block_flag]; / scale: 0-15 */
	- sb = bi->shortDiff[11];
	- idx = bi->shortIdx[11] + lwin;
	-#endif
	- if(is_p != 7)
	- {
	- real t1,t2;
	- t1 = tab1[is_p]; t2 = tab2[is_p];
	- for ( ; sb > 0; sb--,idx+=3 )
	- {
	- real v = xr[0][idx];
	- xr[0][idx] = v * t1;
	- xr[1][idx] = v * t2;
	- }
	- }
	- } /* end for(lwin; .. ; . ) */
	-
	- if (do_l)
	- {
	-/* also check l-part, if ALL bands in the three windows are 'empty'
	- * and mode = mixed_mode
	- */
	- int sfb = gr_info->maxbandl;
	- int idx = bi->longIdx[sfb];
	-
	- for ( ; sfb<8; sfb++ )
	- {
	- int sb = bi->longDiff[sfb];
	- int is_p = scalefac[sfb]; /* scale: 0-15 */
	- if(is_p != 7) {
	- real t1,t2;
	- t1 = tab1[is_p]; t2 = tab2[is_p];
	- for ( ; sb > 0; sb--,idx++)
	- {
	- real v = xr[0][idx];
	- xr[0][idx] = v * t1;
	- xr[1][idx] = v * t2;
	- }
	- }
	- else
	- idx += sb;
	- }
	- }
	- }
	- else /* ((gr_info->block_type != 2)) */
	- {
	- int sfb = gr_info->maxbandl;
	- int is_p,idx = bi->longIdx[sfb];
	- for ( ; sfb<21; sfb++)
	- {
	- int sb = bi->longDiff[sfb];
	- is_p = scalefac[sfb]; /* scale: 0-15 */
	- if(is_p != 7) {
	- real t1,t2;
	- t1 = tab1[is_p]; t2 = tab2[is_p];
	- for ( ; sb > 0; sb--,idx++)
	- {
	- real v = xr[0][idx];
	- xr[0][idx] = v * t1;
	- xr[1][idx] = v * t2;
	- }
	- }
	- else
	- idx += sb;
	- }
	-
	- is_p = scalefac[20]; /* copy l-band 20 to l-band 21 */
	- if(is_p != 7)
	- {
	- int sb;
	- real t1 = tab1[is_p],t2 = tab2[is_p];
	-
	- for ( sb = bi->longDiff[21]; sb > 0; sb--,idx++ )
	- {
	- real v = xr[0][idx];
	- xr[0][idx] = v * t1;
	- xr[1][idx] = v * t2;
	- }
	- }
	- } /* ... */
	-}
	-
	-static void III_antialias(real xr[SBLIMIT][SSLIMIT],struct gr_info_s *gr_info)
	-{
	- int sblim;
	-
	- if(gr_info->block_type == 2)
	- {
	- if(!gr_info->mixed_block_flag)
	- return;
	- sblim = 1;
	- }
	- else {
	- sblim = gr_info->maxb-1;
	- }
	-
	- /* 31 alias-reduction operations between each pair of sub-bands */
	- /* with 8 butterflies between each pair */
	-
	- {
	- int sb;
	- real xr1=(real ) xr[1];
	-
	- for(sb=sblim;sb;sb--,xr1+=10)
	- {
	- int ss;
	- real cs=aa_cs,ca=aa_ca;
	- real *xr2 = xr1;
	-
	- for(ss=7;ss>=0;ss--)
	- { /* upper and lower butterfly inputs */
	- register real bu = --xr2,bd = xr1;
	- xr2 = (bu (cs) ) - (bd (*ca) );
	- xr1++ = (bd (cs++) ) + (bu (*ca++) );
	- }
	- }
	- }
	-}
	-
	-/*
	- DCT insipired by Jeff Tsay's DCT from the maplay package
	- this is an optimized version with manual unroll.
	-
	- References:
	- [1] S. Winograd: "On Computing the Discrete Fourier Transform",
	- Mathematics of Computation, Volume 32, Number 141, January 1978,
	- Pages 175-199
	-*/
	-
	-static void dct36(real inbuf,real o1,real o2,real wintab,real *tsbuf)
	-{
	- {
	- register real *in = inbuf;
	-
	- in[17]+=in[16]; in[16]+=in[15]; in[15]+=in[14];
	- in[14]+=in[13]; in[13]+=in[12]; in[12]+=in[11];
	- in[11]+=in[10]; in[10]+=in[9]; in[9] +=in[8];
	- in[8] +=in[7]; in[7] +=in[6]; in[6] +=in[5];
	- in[5] +=in[4]; in[4] +=in[3]; in[3] +=in[2];
	- in[2] +=in[1]; in[1] +=in[0];
	-
	- in[17]+=in[15]; in[15]+=in[13]; in[13]+=in[11]; in[11]+=in[9];
	- in[9] +=in[7]; in[7] +=in[5]; in[5] +=in[3]; in[3] +=in[1];
	-
	-
	- {
	-
	-#define MACRO0(v) { \
	- real tmp; \
	- out2[9+(v)] = (tmp = sum0 + sum1) * w[27+(v)]; \
	- out2[8-(v)] = tmp * w[26-(v)]; } \
	- sum0 -= sum1; \
	- ts[SBLIMIT(8-(v))] = out1[8-(v)] + sum0 w[8-(v)]; \
	- ts[SBLIMIT(9+(v))] = out1[9+(v)] + sum0 w[9+(v)];
	-#define MACRO1(v) { \
	- real sum0,sum1; \
	- sum0 = tmp1a + tmp2a; \
	- sum1 = (tmp1b + tmp2b) * tfcos36[(v)]; \
	- MACRO0(v); }
	-#define MACRO2(v) { \
	- real sum0,sum1; \
	- sum0 = tmp2a - tmp1a; \
	- sum1 = (tmp2b - tmp1b) * tfcos36[(v)]; \
	- MACRO0(v); }
	-
	- register const real *c = COS9;
	- register real *out2 = o2;
	- register real *w = wintab;
	- register real *out1 = o1;
	- register real *ts = tsbuf;
	-
	- real ta33,ta66,tb33,tb66;
	-
	- ta33 = in[23+0] c[3];
	- ta66 = in[26+0] c[6];
	- tb33 = in[23+1] c[3];
	- tb66 = in[26+1] c[6];
	-
	- {
	- real tmp1a,tmp2a,tmp1b,tmp2b;
	- tmp1a = in[21+0] c[1] + ta33 + in[25+0] c[5] + in[27+0] c[7];
	- tmp1b = in[21+1] c[1] + tb33 + in[25+1] c[5] + in[27+1] c[7];
	- tmp2a = in[20+0] + in[22+0] * c[2] + in[24+0] c[4] + ta66 + in[28+0] c[8];
	- tmp2b = in[20+1] + in[22+1] * c[2] + in[24+1] c[4] + tb66 + in[28+1] c[8];
	-
	- MACRO1(0);
	- MACRO2(8);
	- }
	-
	- {
	- real tmp1a,tmp2a,tmp1b,tmp2b;
	- tmp1a = ( in[21+0] - in[25+0] - in[27+0] ) c[3];
	- tmp1b = ( in[21+1] - in[25+1] - in[27+1] ) c[3];
	- tmp2a = ( in[22+0] - in[24+0] - in[28+0] ) c[6] - in[26+0] + in[20+0];
	- tmp2b = ( in[22+1] - in[24+1] - in[28+1] ) c[6] - in[26+1] + in[20+1];
	-
	- MACRO1(1);
	- MACRO2(7);
	- }
	-
	- {
	- real tmp1a,tmp2a,tmp1b,tmp2b;
	- tmp1a = in[21+0] c[5] - ta33 - in[25+0] c[7] + in[27+0] c[1];
	- tmp1b = in[21+1] c[5] - tb33 - in[25+1] c[7] + in[27+1] c[1];
	- tmp2a = in[20+0] - in[22+0] * c[8] - in[24+0] c[2] + ta66 + in[28+0] c[4];
	- tmp2b = in[20+1] - in[22+1] * c[8] - in[24+1] c[2] + tb66 + in[28+1] c[4];
	-
	- MACRO1(2);
	- MACRO2(6);
	- }
	-
	- {
	- real tmp1a,tmp2a,tmp1b,tmp2b;
	- tmp1a = in[21+0] c[7] - ta33 + in[25+0] c[1] - in[27+0] c[5];
	- tmp1b = in[21+1] c[7] - tb33 + in[25+1] c[1] - in[27+1] c[5];
	- tmp2a = in[20+0] - in[22+0] * c[4] + in[24+0] c[8] + ta66 - in[28+0] c[2];
	- tmp2b = in[20+1] - in[22+1] * c[4] + in[24+1] c[8] + tb66 - in[28+1] c[2];
	-
	- MACRO1(3);
	- MACRO2(5);
	- }
	-
	- {
	- real sum0,sum1;
	- sum0 = in[20+0] - in[22+0] + in[24+0] - in[26+0] + in[2*8+0];
	- sum1 = (in[20+1] - in[22+1] + in[24+1] - in[26+1] + in[28+1] ) tfcos36[4];
	- MACRO0(4);
	- }
	- }
	-
	- }
	-}
	-
	-/*
	- * new DCT12
	- */
	-static void dct12(real in,real rawout1,real rawout2,register real wi,register real *ts)
	-{
	-#define DCT12_PART1 \
	- in5 = in[5*3]; \
	- in5 += (in4 = in[4*3]); \
	- in4 += (in3 = in[3*3]); \
	- in3 += (in2 = in[2*3]); \
	- in2 += (in1 = in[1*3]); \
	- in1 += (in0 = in[0*3]); \
	- \
	- in5 += in3; in3 += in1; \
	- \
	- in2 *= COS6_1; \
	- in3 *= COS6_1; \
	-
	-#define DCT12_PART2 \
	- in0 += in4 * COS6_2; \
	- \
	- in4 = in0 + in2; \
	- in0 -= in2; \
	- \
	- in1 += in5 * COS6_2; \
	- \
	- in5 = (in1 + in3) * tfcos12[0]; \
	- in1 = (in1 - in3) * tfcos12[2]; \
	- \
	- in3 = in4 + in5; \
	- in4 -= in5; \
	- \
	- in2 = in0 + in1; \
	- in0 -= in1;
	-
	-
	- {
	- real in0,in1,in2,in3,in4,in5;
	- register real *out1 = rawout1;
	- ts[SBLIMIT0] = out1[0]; ts[SBLIMIT1] = out1[1]; ts[SBLIMIT*2] = out1[2];
	- ts[SBLIMIT3] = out1[3]; ts[SBLIMIT4] = out1[4]; ts[SBLIMIT*5] = out1[5];
	-
	- DCT12_PART1
	-
	- {
	- real tmp0,tmp1 = (in0 - in4);
	- {
	- real tmp2 = (in1 - in5) * tfcos12[1];
	- tmp0 = tmp1 + tmp2;
	- tmp1 -= tmp2;
	- }
	- ts[(17-1)SBLIMIT] = out1[17-1] + tmp0 wi[11-1];
	- ts[(12+1)SBLIMIT] = out1[12+1] + tmp0 wi[6+1];
	- ts[(6 +1)SBLIMIT] = out1[6 +1] + tmp1 wi[1];
	- ts[(11-1)SBLIMIT] = out1[11-1] + tmp1 wi[5-1];
	- }
	-
	- DCT12_PART2
	-
	- ts[(17-0)SBLIMIT] = out1[17-0] + in2 wi[11-0];
	- ts[(12+0)SBLIMIT] = out1[12+0] + in2 wi[6+0];
	- ts[(12+2)SBLIMIT] = out1[12+2] + in3 wi[6+2];
	- ts[(17-2)SBLIMIT] = out1[17-2] + in3 wi[11-2];
	-
	- ts[(6+0)SBLIMIT] = out1[6+0] + in0 wi[0];
	- ts[(11-0)SBLIMIT] = out1[11-0] + in0 wi[5-0];
	- ts[(6+2)SBLIMIT] = out1[6+2] + in4 wi[2];
	- ts[(11-2)SBLIMIT] = out1[11-2] + in4 wi[5-2];
	- }
	-
	- in++;
	-
	- {
	- real in0,in1,in2,in3,in4,in5;
	- register real *out2 = rawout2;
	-
	- DCT12_PART1
	-
	- {
	- real tmp0,tmp1 = (in0 - in4);
	- {
	- real tmp2 = (in1 - in5) * tfcos12[1];
	- tmp0 = tmp1 + tmp2;
	- tmp1 -= tmp2;
	- }
	- out2[5-1] = tmp0 * wi[11-1];
	- out2[0+1] = tmp0 * wi[6+1];
	- ts[(12+1)SBLIMIT] += tmp1 wi[1];
	- ts[(17-1)SBLIMIT] += tmp1 wi[5-1];
	- }
	-
	- DCT12_PART2
	-
	- out2[5-0] = in2 * wi[11-0];
	- out2[0+0] = in2 * wi[6+0];
	- out2[0+2] = in3 * wi[6+2];
	- out2[5-2] = in3 * wi[11-2];
	-
	- ts[(12+0)SBLIMIT] += in0 wi[0];
	- ts[(17-0)SBLIMIT] += in0 wi[5-0];
	- ts[(12+2)SBLIMIT] += in4 wi[2];
	- ts[(17-2)SBLIMIT] += in4 wi[5-2];
	- }
	-
	- in++;
	-
	- {
	- real in0,in1,in2,in3,in4,in5;
	- register real *out2 = rawout2;
	- out2[12]=out2[13]=out2[14]=out2[15]=out2[16]=out2[17]=0.0;
	-
	- DCT12_PART1
	-
	- {
	- real tmp0,tmp1 = (in0 - in4);
	- {
	- real tmp2 = (in1 - in5) * tfcos12[1];
	- tmp0 = tmp1 + tmp2;
	- tmp1 -= tmp2;
	- }
	- out2[11-1] = tmp0 * wi[11-1];
	- out2[6 +1] = tmp0 * wi[6+1];
	- out2[0+1] += tmp1 * wi[1];
	- out2[5-1] += tmp1 * wi[5-1];
	- }
	-
	- DCT12_PART2
	-
	- out2[11-0] = in2 * wi[11-0];
	- out2[6 +0] = in2 * wi[6+0];
	- out2[6 +2] = in3 * wi[6+2];
	- out2[11-2] = in3 * wi[11-2];
	-
	- out2[0+0] += in0 * wi[0];
	- out2[5-0] += in0 * wi[5-0];
	- out2[0+2] += in4 * wi[2];
	- out2[5-2] += in4 * wi[5-2];
	- }
	-}
	-
	-/*
	- * III_hybrid
	- */
	-static void III_hybrid(real fsIn[SBLIMIT][SSLIMIT],real tsOut[SSLIMIT][SBLIMIT],
	- int ch,struct gr_info_s gr_info,struct mpstr mp)
	-{
	- real tspnt = (real ) tsOut;
	- real (block)[2][SBLIMITSSLIMIT] = mp->hybrid_block;
	- int *blc = mp->hybrid_blc;
	- real rawout1,rawout2;
	- int bt;
	- int sb = 0;
	-
	- {
	- int b = blc[ch];
	- rawout1=block[b][ch];
	- b=-b+1;
	- rawout2=block[b][ch];
	- blc[ch] = b;
	- }
	-
	-
	- if(gr_info->mixed_block_flag) {
	- sb = 2;
	- dct36(fsIn[0],rawout1,rawout2,win[0],tspnt);
	- dct36(fsIn[1],rawout1+18,rawout2+18,win1[0],tspnt+1);
	- rawout1 += 36; rawout2 += 36; tspnt += 2;
	- }
	-
	- bt = gr_info->block_type;
	- if(bt == 2) {
	- for (; sb<gr_info->maxb; sb+=2,tspnt+=2,rawout1+=36,rawout2+=36) {
	- dct12(fsIn[sb],rawout1,rawout2,win[2],tspnt);
	- dct12(fsIn[sb+1],rawout1+18,rawout2+18,win1[2],tspnt+1);
	- }
	- }
	- else {
	- for (; sb<gr_info->maxb; sb+=2,tspnt+=2,rawout1+=36,rawout2+=36) {
	- dct36(fsIn[sb],rawout1,rawout2,win[bt],tspnt);
	- dct36(fsIn[sb+1],rawout1+18,rawout2+18,win1[bt],tspnt+1);
	- }
	- }
	-
	- for(;sb<SBLIMIT;sb++,tspnt++) {
	- int i;
	- for(i=0;i<SSLIMIT;i++) {
	- tspnt[iSBLIMIT] = rawout1++;
	- *rawout2++ = 0.0;
	- }
	- }
	-}
	-
	-/*
	- * main layer3 handler
	- */
	-int do_layer3(struct frame fr,unsigned char pcm_sample,
	- int pcm_point,struct mpstr mp)
	-{
	- int gr, ch, ss,clip=0;
	- int scalefacs[2][39]; /* max 39 for short[13][3] mode, mixed: 38, long: 22 */
	- struct III_sideinfo sideinfo;
	- int stereo = fr->stereo;
	- int single = fr->single;
	- int ms_stereo,i_stereo;
	- int sfreq = fr->sampling_frequency;
	- int stereo1,granules;
	-
	- if(stereo == 1) { /* stream is mono */
	- stereo1 = 1;
	- single = 0;
	- }
	- else if(single >= 0) /* stream is stereo, but force to mono */
	- stereo1 = 1;
	- else
	- stereo1 = 2;
	-
	- if(fr->mode == MPG_MD_JOINT_STEREO) {
	- ms_stereo = fr->mode_ext & 0x2;
	- i_stereo = fr->mode_ext & 0x1;
	- }
	- else
	- ms_stereo = i_stereo = 0;
	-
	- if(fr->lsf) {
	- granules = 1;
	- if(!III_get_side_info_2(&sideinfo,stereo,ms_stereo,sfreq,single))
	- return -1;
	- }
	- else {
	- granules = 2;
	-#ifdef MPEG1
	- if(!III_get_side_info_1(&sideinfo,stereo,ms_stereo,sfreq,single))
	- return -1;
	-#else
	- __Sound_SetError("MPGLIB: Not supported!");
	-#endif
	- }
	-
	- if(set_pointer(sideinfo.main_data_begin,mp) == MP3_ERR)
	- return -1;
	-
	- for (gr=0;gr<granules;gr++)
	- {
	- real hybridIn[2][SBLIMIT][SSLIMIT];
	- real hybridOut[2][SSLIMIT][SBLIMIT];
	- memset(hybridIn, '\0', sizeof (hybridIn));
	-
	- {
	- struct gr_info_s *gr_info = &(sideinfo.ch[0].gr[gr]);
	- long part2bits;
	- if(fr->lsf)
	- part2bits = III_get_scale_factors_2(scalefacs[0],gr_info,0);
	- else {
	-#ifdef MPEG1
	- part2bits = III_get_scale_factors_1(scalefacs[0],gr_info);
	-#else
	- __Sound_SetError("MPGLIB: Not supported!");
	-#endif
	- }
	- if(III_dequantize_sample(hybridIn[0], scalefacs[0],gr_info,sfreq,part2bits))
	- return clip;
	- }
	- if(stereo == 2) {
	- struct gr_info_s *gr_info = &(sideinfo.ch[1].gr[gr]);
	- long part2bits;
	- if(fr->lsf)
	- part2bits = III_get_scale_factors_2(scalefacs[1],gr_info,i_stereo);
	- else {
	-#ifdef MPEG1
	- part2bits = III_get_scale_factors_1(scalefacs[1],gr_info);
	-#else
	- __Sound_SetError("MPGLIB: Not supported!");
	-#endif
	- }
	-
	- if(III_dequantize_sample(hybridIn[1],scalefacs[1],gr_info,sfreq,part2bits))
	- return clip;
	-
	- if(ms_stereo) {
	- int i;
	- for(i=0;i<SBLIMIT*SSLIMIT;i++) {
	- real tmp0,tmp1;
	- tmp0 = ((real *) hybridIn[0])[i];
	- tmp1 = ((real *) hybridIn[1])[i];
	- ((real *) hybridIn[0])[i] = tmp0 + tmp1;
	- ((real *) hybridIn[1])[i] = tmp0 - tmp1;
	- }
	- }
	-
	- if(i_stereo)
	- III_i_stereo(hybridIn,scalefacs[1],gr_info,sfreq,ms_stereo,fr->lsf);
	-
	- if(ms_stereo \|\| i_stereo \|\| (single == 3) ) {
	- if(gr_info->maxb > sideinfo.ch[0].gr[gr].maxb)
	- sideinfo.ch[0].gr[gr].maxb = gr_info->maxb;
	- else
	- gr_info->maxb = sideinfo.ch[0].gr[gr].maxb;
	- }
	-
	- switch(single) {
	- case 3:
	- {
	- register int i;
	- register real in0 = (real ) hybridIn[0],in1 = (real ) hybridIn[1];
	- for(i=0;i<SSLIMIT*gr_info->maxb;i++,in0++)
	- in0 = (in0 + in1++); / 0.5 done by pow-scale /
	- }
	- break;
	- case 1:
	- {
	- register int i;
	- register real in0 = (real ) hybridIn[0],in1 = (real ) hybridIn[1];
	- for(i=0;i<SSLIMIT*gr_info->maxb;i++)
	- in0++ = in1++;
	- }
	- break;
	- }
	- }
	-
	- for(ch=0;ch<stereo1;ch++) {
	- struct gr_info_s *gr_info = &(sideinfo.ch[ch].gr[gr]);
	- III_antialias(hybridIn[ch],gr_info);
	- III_hybrid(hybridIn[ch], hybridOut[ch], ch,gr_info,mp);
	- }
	-
	- for(ss=0;ss<SSLIMIT;ss++) {
	- if(single >= 0) {
	- clip += synth_1to1_mono(hybridOut[0][ss],pcm_sample,pcm_point,mp);
	- }
	- else {
	- int p1 = *pcm_point;
	- clip += synth_1to1(hybridOut[0][ss],0,pcm_sample,&p1,mp);
	- clip += synth_1to1(hybridOut[1][ss],1,pcm_sample,pcm_point,mp);
	- }
	- }
	- }
	-
	- return clip;
	-}
	-
	-
	diff --git a/util/sdl/sound/decoders/mpglib/main.c b/util/sdl/sound/decoders/mpglib/main.c
	deleted file mode 100644
	index 5063958d..00000000
	--- a/util/sdl/sound/decoders/mpglib/main.c
	+++ /dev/null
	@@ -1,33 +0,0 @@
	-
	-#include "mpg123_sdlsound.h"
	-#include "mpglib_sdlsound.h"
	-
	-#error This is for example usage. Do not compile for SDL_sound.
	-
	-char buf[16384];
	-struct mpstr mp;
	-
	-int main(int argc,char **argv)
	-{
	- int size;
	- char out[8192];
	- int len,ret;
	-
	-
	- InitMP3(&mp);
	-
	- while(1) {
	- len = read(0,buf,16384);
	- if(len <= 0)
	- break;
	- ret = decodeMP3(&mp,buf,len,out,8192,&size);
	- while(ret == MP3_OK) {
	- write(1,out,size);
	- ret = decodeMP3(&mp,NULL,0,out,8192,&size);
	- }
	- }
	-
	- return 0;
	-
	-}
	-
	diff --git a/util/sdl/sound/decoders/mpglib/mpg123_sdlsound.h b/util/sdl/sound/decoders/mpglib/mpg123_sdlsound.h
	deleted file mode 100644
	index d8fc9bd3..00000000
	--- a/util/sdl/sound/decoders/mpglib/mpg123_sdlsound.h
	+++ /dev/null
	@@ -1,199 +0,0 @@
	-#include <stdio.h>
	-#include <string.h>
	-
	-#if !defined(WIN32) && !defined(macintosh) && !defined(_WIN32_WCE)
	-#include <unistd.h>
	-#endif
	-
	-#include <math.h>
	-
	-#if defined(_WIN32)
	-# undef WIN32
	-# define WIN32
	-#endif
	-
	-#if defined(WIN32) \|\| defined(macintosh) \|\| defined(_WIN32_WCE)
	-
	-# define M_PI 3.14159265358979323846
	-# define M_SQRT2 1.41421356237309504880
	-# define REAL_IS_FLOAT
	-# define NEW_DCT9
	-
	-# define random rand
	-# define srandom srand
	-
	-#endif
	-
	-#ifdef REAL_IS_FLOAT
	-# define real float
	-#elif defined(REAL_IS_LONG_DOUBLE)
	-# define real long double
	-#else
	-# define real double
	-#endif
	-
	-#ifdef __GNUC__
	-#define INLINE inline
	-#elif ((defined _MSC_VER) \|\| (defined __inline__))
	-#define INLINE __inline__
	-#else
	-#define INLINE
	-#endif
	-
	-/* AUDIOBUFSIZE = n64 with n=1,2,3 ... /
	-#define AUDIOBUFSIZE 16384
	-
	-#ifndef FALSE
	-#define FALSE 0
	-#endif
	-#ifndef FALSE
	-#define TRUE 1
	-#endif
	-
	-#define SBLIMIT 32
	-#define SSLIMIT 18
	-
	-#define SCALE_BLOCK 12
	-
	-
	-#define MPG_MD_STEREO 0
	-#define MPG_MD_JOINT_STEREO 1
	-#define MPG_MD_DUAL_CHANNEL 2
	-#define MPG_MD_MONO 3
	-
	-#define MAXFRAMESIZE 1792
	-
	-
	-/* Pre Shift fo 16 to 8 bit converter table */
	-#define AUSHIFT (3)
	-
	-struct frame {
	- int stereo;
	- int jsbound;
	- int single;
	- int lsf;
	- int mpeg25;
	- int header_change;
	- int lay;
	- int error_protection;
	- int bitrate_index;
	- int sampling_frequency;
	- int padding;
	- int extension;
	- int mode;
	- int mode_ext;
	- int copyright;
	- int original;
	- int emphasis;
	- int framesize; /* computed framesize */
	-
	- /* layer2 stuff */
	- int II_sblimit;
	- void *alloc;
	-};
	-
	-struct parameter {
	- int quiet; /* shut up! */
	- int tryresync; /* resync stream after error */
	- int verbose; /* verbose level */
	- int checkrange;
	-};
	-
	-struct mpstr; /* forward declaration. */
	-
	-extern unsigned int get1bit(void);
	-extern unsigned int getbits(int);
	-extern unsigned int getbits_fast(int);
	-extern int set_pointer(long,struct mpstr *);
	-
	-extern unsigned char *wordpointer;
	-extern int bitindex;
	-
	-extern void make_decode_tables(long scaleval);
	-extern int do_layer3(struct frame fr,unsigned char ,int ,struct mpstr );
	-extern int do_layer2(struct frame fr,unsigned char ,int ,struct mpstr );
	-extern int do_layer1(struct frame fr,unsigned char ,int ,struct mpstr );
	-extern int decode_header(struct frame *fr,unsigned long newhead);
	-
	-
	-
	-struct gr_info_s {
	- int scfsi;
	- unsigned part2_3_length;
	- unsigned big_values;
	- unsigned scalefac_compress;
	- unsigned block_type;
	- unsigned mixed_block_flag;
	- unsigned table_select[3];
	- unsigned subblock_gain[3];
	- unsigned maxband[3];
	- unsigned maxbandl;
	- unsigned maxb;
	- unsigned region1start;
	- unsigned region2start;
	- unsigned preflag;
	- unsigned scalefac_scale;
	- unsigned count1table_select;
	- real *full_gain[3];
	- real *pow2gain;
	-};
	-
	-struct III_sideinfo
	-{
	- unsigned main_data_begin;
	- unsigned private_bits;
	- struct {
	- struct gr_info_s gr[2];
	- } ch[2];
	-};
	-
	-
	-extern int synth_1to1 (real ,int,unsigned char ,int ,struct mpstr );
	-extern int synth_1to1_8bit (real ,int,unsigned char ,int *);
	-extern int synth_1to1_mono (real ,unsigned char ,int ,struct mpstr );
	-extern int synth_1to1_mono2stereo (real ,unsigned char ,int *);
	-extern int synth_1to1_8bit_mono (real ,unsigned char ,int *);
	-extern int synth_1to1_8bit_mono2stereo (real ,unsigned char ,int *);
	-
	-extern int synth_2to1 (real ,int,unsigned char ,int *);
	-extern int synth_2to1_8bit (real ,int,unsigned char ,int *);
	-extern int synth_2to1_mono (real ,unsigned char ,int *);
	-extern int synth_2to1_mono2stereo (real ,unsigned char ,int *);
	-extern int synth_2to1_8bit_mono (real ,unsigned char ,int *);
	-extern int synth_2to1_8bit_mono2stereo (real ,unsigned char ,int *);
	-
	-extern int synth_4to1 (real ,int,unsigned char ,int *);
	-extern int synth_4to1_8bit (real ,int,unsigned char ,int *);
	-extern int synth_4to1_mono (real ,unsigned char ,int *);
	-extern int synth_4to1_mono2stereo (real ,unsigned char ,int *);
	-extern int synth_4to1_8bit_mono (real ,unsigned char ,int *);
	-extern int synth_4to1_8bit_mono2stereo (real ,unsigned char ,int *);
	-
	-extern int synth_ntom (real ,int,unsigned char ,int *);
	-extern int synth_ntom_8bit (real ,int,unsigned char ,int *);
	-extern int synth_ntom_mono (real ,unsigned char ,int *);
	-extern int synth_ntom_mono2stereo (real ,unsigned char ,int *);
	-extern int synth_ntom_8bit_mono (real ,unsigned char ,int *);
	-extern int synth_ntom_8bit_mono2stereo (real ,unsigned char ,int *);
	-
	-extern void rewindNbits(int bits);
	-extern int hsstell(void);
	-extern int get_songlen(struct frame *fr,int no);
	-
	-extern void init_layer3(int);
	-extern void init_layer2(void);
	-extern void make_decode_tables(long scale);
	-extern void make_conv16to8_table(int);
	-extern void dct64(real ,real ,real *);
	-
	-extern void synth_ntom_set_step(long,long);
	-
	-extern unsigned char *conv16to8;
	-extern long mpglib_freqs[9];
	-extern real muls[27][64];
	-extern real decwin[512+32];
	-extern real *pnts[5];
	-
	-extern struct parameter param;
	-
	-
	diff --git a/util/sdl/sound/decoders/mpglib/mpglib_common.c b/util/sdl/sound/decoders/mpglib/mpglib_common.c
	deleted file mode 100644
	index cc14662d..00000000
	--- a/util/sdl/sound/decoders/mpglib/mpglib_common.c
	+++ /dev/null
	@@ -1,243 +0,0 @@
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#include <ctype.h>
	-#include <stdlib.h>
	-
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-#include "mpg123_sdlsound.h"
	-
	-struct parameter param = { 1 , 1 , 0 , 0 };
	-
	-int tabsel_123[2][3][16] = {
	- { {0,32,64,96,128,160,192,224,256,288,320,352,384,416,448,},
	- {0,32,48,56, 64, 80, 96,112,128,160,192,224,256,320,384,},
	- {0,32,40,48, 56, 64, 80, 96,112,128,160,192,224,256,320,} },
	-
	- { {0,32,48,56,64,80,96,112,128,144,160,176,192,224,256,},
	- {0,8,16,24,32,40,48,56,64,80,96,112,128,144,160,},
	- {0,8,16,24,32,40,48,56,64,80,96,112,128,144,160,} }
	-};
	-
	-long mpglib_freqs[9] = { 44100, 48000, 32000,
	- 22050, 24000, 16000 ,
	- 11025 , 12000 , 8000 };
	-
	-int bitindex;
	-unsigned char *wordpointer;
	-unsigned char *pcm_sample;
	-int pcm_point = 0;
	-
	-
	-#define HDRCMPMASK 0xfffffd00
	-
	-#if 0
	-int head_check(unsigned long head)
	-{
	- if( (head & 0xffe00000) != 0xffe00000)
	- return FALSE;
	- if(!((head>>17)&3))
	- return FALSE;
	- if( ((head>>12)&0xf) == 0xf)
	- return FALSE;
	- if( ((head>>10)&0x3) == 0x3 )
	- return FALSE;
	- return TRUE;
	-}
	-#endif
	-
	-
	-/*
	- * the code a header and write the information
	- * into the frame structure
	- */
	-int decode_header(struct frame *fr,unsigned long newhead)
	-{
	- if( newhead & (1<<20) ) {
	- fr->lsf = (newhead & (1<<19)) ? 0x0 : 0x1;
	- fr->mpeg25 = 0;
	- }
	- else {
	- fr->lsf = 1;
	- fr->mpeg25 = 1;
	- }
	-
	- fr->lay = 4-((newhead>>17)&3);
	- if( ((newhead>>10)&0x3) == 0x3) {
	- BAIL_MACRO("MPGLIB: Corrupted header", 0);
	- }
	- if(fr->mpeg25) {
	- fr->sampling_frequency = 6 + ((newhead>>10)&0x3);
	- }
	- else
	- fr->sampling_frequency = ((newhead>>10)&0x3) + (fr->lsf*3);
	- fr->error_protection = ((newhead>>16)&0x1)^0x1;
	-
	- if(fr->mpeg25) /* allow Bitrate change for 2.5 ... */
	- fr->bitrate_index = ((newhead>>12)&0xf);
	-
	- fr->bitrate_index = ((newhead>>12)&0xf);
	- fr->padding = ((newhead>>9)&0x1);
	- fr->extension = ((newhead>>8)&0x1);
	- fr->mode = ((newhead>>6)&0x3);
	- fr->mode_ext = ((newhead>>4)&0x3);
	- fr->copyright = ((newhead>>3)&0x1);
	- fr->original = ((newhead>>2)&0x1);
	- fr->emphasis = newhead & 0x3;
	-
	- fr->stereo = (fr->mode == MPG_MD_MONO) ? 1 : 2;
	-
	- if(!fr->bitrate_index)
	- {
	- BAIL_MACRO("MPGLIB: Free format not supported.", 0);
	- }
	-
	- switch(fr->lay)
	- {
	- case 1:
	-#ifdef LAYER1
	-#if 0
	- fr->jsbound = (fr->mode == MPG_MD_JOINT_STEREO) ?
	- (fr->mode_ext<<2)+4 : 32;
	-#endif
	- fr->framesize = (long) tabsel_123[fr->lsf][0][fr->bitrate_index] * 12000;
	- fr->framesize /= mpglib_freqs[fr->sampling_frequency];
	- fr->framesize = ((fr->framesize+fr->padding)<<2)-4;
	-#else
	- __Sound_SetError("MPGLIB: Not supported!");
	-#endif
	- break;
	- case 2:
	-#ifdef LAYER2
	-#if 0
	- fr->jsbound = (fr->mode == MPG_MD_JOINT_STEREO) ?
	- (fr->mode_ext<<2)+4 : fr->II_sblimit;
	-#endif
	- fr->framesize = (long) tabsel_123[fr->lsf][1][fr->bitrate_index] * 144000;
	- fr->framesize /= mpglib_freqs[fr->sampling_frequency];
	- fr->framesize += fr->padding - 4;
	-#else
	- __Sound_SetError("MPGLIB: Not supported!");
	-#endif
	- break;
	- case 3:
	-#if 0
	- fr->do_layer = do_layer3;
	- if(fr->lsf)
	- ssize = (fr->stereo == 1) ? 9 : 17;
	- else
	- ssize = (fr->stereo == 1) ? 17 : 32;
	-#endif
	-
	-#if 0
	- if(fr->error_protection)
	- ssize += 2;
	-#endif
	- fr->framesize = (long) tabsel_123[fr->lsf][2][fr->bitrate_index] * 144000;
	- fr->framesize /= mpglib_freqs[fr->sampling_frequency]<<(fr->lsf);
	- fr->framesize = fr->framesize + fr->padding - 4;
	- break;
	- default:
	- BAIL_MACRO("MPGLIB: Unknown layer type.", 0);
	- }
	- return 1;
	-}
	-
	-#if 0
	-void print_header(struct frame *fr)
	-{
	- static char *modes[4] = { "Stereo", "Joint-Stereo", "Dual-Channel", "Single-Channel" };
	- static char *layers[4] = { "Unknown" , "I", "II", "III" };
	-
	- fprintf(stderr,"MPEG %s, Layer: %s, Freq: %ld, mode: %s, modext: %d, BPF : %d\n",
	- fr->mpeg25 ? "2.5" : (fr->lsf ? "2.0" : "1.0"),
	- layers[fr->lay],mpglib_freqs[fr->sampling_frequency],
	- modes[fr->mode],fr->mode_ext,fr->framesize+4);
	- fprintf(stderr,"Channels: %d, copyright: %s, original: %s, CRC: %s, emphasis: %d.\n",
	- fr->stereo,fr->copyright?"Yes":"No",
	- fr->original?"Yes":"No",fr->error_protection?"Yes":"No",
	- fr->emphasis);
	- fprintf(stderr,"Bitrate: %d Kbits/s, Extension value: %d\n",
	- tabsel_123[fr->lsf][fr->lay-1][fr->bitrate_index],fr->extension);
	-}
	-
	-void print_header_compact(struct frame *fr)
	-{
	- static char *modes[4] = { "stereo", "joint-stereo", "dual-channel", "mono" };
	- static char *layers[4] = { "Unknown" , "I", "II", "III" };
	-
	- fprintf(stderr,"MPEG %s layer %s, %d kbit/s, %ld Hz %s\n",
	- fr->mpeg25 ? "2.5" : (fr->lsf ? "2.0" : "1.0"),
	- layers[fr->lay],
	- tabsel_123[fr->lsf][fr->lay-1][fr->bitrate_index],
	- mpglib_freqs[fr->sampling_frequency], modes[fr->mode]);
	-}
	-
	-#endif
	-
	-unsigned int getbits(int number_of_bits)
	-{
	- unsigned long rval;
	-
	- if(!number_of_bits)
	- return 0;
	-
	- {
	- rval = wordpointer[0];
	- rval <<= 8;
	- rval \|= wordpointer[1];
	- rval <<= 8;
	- rval \|= wordpointer[2];
	- rval <<= bitindex;
	- rval &= 0xffffff;
	-
	- bitindex += number_of_bits;
	-
	- rval >>= (24-number_of_bits);
	-
	- wordpointer += (bitindex>>3);
	- bitindex &= 7;
	- }
	- return rval;
	-}
	-
	-unsigned int getbits_fast(int number_of_bits)
	-{
	- unsigned long rval;
	-
	- {
	- rval = wordpointer[0];
	- rval <<= 8;
	- rval \|= wordpointer[1];
	- rval <<= bitindex;
	- rval &= 0xffff;
	- bitindex += number_of_bits;
	-
	- rval >>= (16-number_of_bits);
	-
	- wordpointer += (bitindex>>3);
	- bitindex &= 7;
	- }
	- return rval;
	-}
	-
	-unsigned int get1bit(void)
	-{
	- unsigned char rval;
	- rval = *wordpointer << bitindex;
	-
	- bitindex++;
	- wordpointer += (bitindex>>3);
	- bitindex &= 7;
	-
	- return rval>>7;
	-}
	-
	-
	-
	diff --git a/util/sdl/sound/decoders/mpglib/mpglib_sdlsound.h b/util/sdl/sound/decoders/mpglib/mpglib_sdlsound.h
	deleted file mode 100644
	index 58c7041c..00000000
	--- a/util/sdl/sound/decoders/mpglib/mpglib_sdlsound.h
	+++ /dev/null
	@@ -1,63 +0,0 @@
	-
	-#ifndef _INCLUDE_MPGLIB_SDLSOUND_H_
	-#define _INCLUDE_MPGLIB_SDLSOUND_H_
	-
	-#ifdef _MSC_VER
	- #define snprintf _snprintf
	-#endif
	-
	-struct buf {
	- unsigned char *pnt;
	- long size;
	- long pos;
	- struct buf *next;
	- struct buf *prev;
	-};
	-
	-struct framebuf {
	- struct buf *buf;
	- long pos;
	- struct frame *next;
	- struct frame *prev;
	-};
	-
	-struct mpstr {
	- struct buf head,tail;
	- int bsize;
	- int framesize;
	- int fsizeold;
	- struct frame fr;
	- unsigned char bsspace[2][MAXFRAMESIZE+512]; /* MAXFRAMESIZE */
	- real hybrid_block[2][2][SBLIMIT*SSLIMIT];
	- int hybrid_blc[2];
	- unsigned long header;
	- int bsnum;
	- real synth_buffs[2][2][0x110];
	- int synth_bo;
	-};
	-
	-#ifndef BOOL
	-#define BOOL int
	-#endif
	-
	-#define MP3_ERR -1
	-#define MP3_OK 0
	-#define MP3_NEED_MORE 1
	-
	-
	-#ifdef __cplusplus
	-extern "C" {
	-#endif
	-
	-BOOL InitMP3(struct mpstr *mp);
	-int decodeMP3(struct mpstr mp,char inmemory,int inmemsize,
	- char outmemory,int outmemsize,int done);
	-void ExitMP3(struct mpstr *mp);
	-
	-#ifdef __cplusplus
	-}
	-#endif
	-
	-#endif
	-
	-
	diff --git a/util/sdl/sound/decoders/mpglib/tabinit.c b/util/sdl/sound/decoders/mpglib/tabinit.c
	deleted file mode 100644
	index dbd3965c..00000000
	--- a/util/sdl/sound/decoders/mpglib/tabinit.c
	+++ /dev/null
	@@ -1,80 +0,0 @@
	-
	-#include <stdlib.h>
	-
	-#include "mpg123_sdlsound.h"
	-
	-real decwin[512+32];
	-static real cos64[16],cos32[8],cos16[4],cos8[2],cos4[1];
	-real *pnts[] = { cos64,cos32,cos16,cos8,cos4 };
	-
	-#if 0
	-static unsigned char *conv16to8_buf = NULL;
	-unsigned char *conv16to8;
	-#endif
	-
	-static long intwinbase[] = {
	- 0, -1, -1, -1, -1, -1, -1, -2, -2, -2,
	- -2, -3, -3, -4, -4, -5, -5, -6, -7, -7,
	- -8, -9, -10, -11, -13, -14, -16, -17, -19, -21,
	- -24, -26, -29, -31, -35, -38, -41, -45, -49, -53,
	- -58, -63, -68, -73, -79, -85, -91, -97, -104, -111,
	- -117, -125, -132, -139, -147, -154, -161, -169, -176, -183,
	- -190, -196, -202, -208, -213, -218, -222, -225, -227, -228,
	- -228, -227, -224, -221, -215, -208, -200, -189, -177, -163,
	- -146, -127, -106, -83, -57, -29, 2, 36, 72, 111,
	- 153, 197, 244, 294, 347, 401, 459, 519, 581, 645,
	- 711, 779, 848, 919, 991, 1064, 1137, 1210, 1283, 1356,
	- 1428, 1498, 1567, 1634, 1698, 1759, 1817, 1870, 1919, 1962,
	- 2001, 2032, 2057, 2075, 2085, 2087, 2080, 2063, 2037, 2000,
	- 1952, 1893, 1822, 1739, 1644, 1535, 1414, 1280, 1131, 970,
	- 794, 605, 402, 185, -45, -288, -545, -814, -1095, -1388,
	- -1692, -2006, -2330, -2663, -3004, -3351, -3705, -4063, -4425, -4788,
	- -5153, -5517, -5879, -6237, -6589, -6935, -7271, -7597, -7910, -8209,
	- -8491, -8755, -8998, -9219, -9416, -9585, -9727, -9838, -9916, -9959,
	- -9966, -9935, -9863, -9750, -9592, -9389, -9139, -8840, -8492, -8092,
	- -7640, -7134, -6574, -5959, -5288, -4561, -3776, -2935, -2037, -1082,
	- -70, 998, 2122, 3300, 4533, 5818, 7154, 8540, 9975, 11455,
	- 12980, 14548, 16155, 17799, 19478, 21189, 22929, 24694, 26482, 28289,
	- 30112, 31947, 33791, 35640, 37489, 39336, 41176, 43006, 44821, 46617,
	- 48390, 50137, 51853, 53534, 55178, 56778, 58333, 59838, 61289, 62684,
	- 64019, 65290, 66494, 67629, 68692, 69679, 70590, 71420, 72169, 72835,
	- 73415, 73908, 74313, 74630, 74856, 74992, 75038 };
	-
	-void make_decode_tables(long scaleval)
	-{
	- int i,j,k,kr,divv;
	- real table,costab;
	-
	-
	- for(i=0;i<5;i++)
	- {
	- kr=0x10>>i; divv=0x40>>i;
	- costab = pnts[i];
	- for(k=0;k<kr;k++)
	- costab[k] = 1.0 / (2.0 * cos(M_PI * ((double) k * 2.0 + 1.0) / (double) divv));
	- }
	-
	- table = decwin;
	- scaleval = -scaleval;
	- for(i=0,j=0;i<256;i++,j++,table+=32)
	- {
	- if(table < decwin+512+16)
	- table[16] = table[0] = (double) intwinbase[j] / 65536.0 * (double) scaleval;
	- if(i % 32 == 31)
	- table -= 1023;
	- if(i % 64 == 63)
	- scaleval = - scaleval;
	- }
	-
	- for( /* i=256 */ ;i<512;i++,j--,table+=32)
	- {
	- if(table < decwin+512+16)
	- table[16] = table[0] = (double) intwinbase[j] / 65536.0 * (double) scaleval;
	- if(i % 32 == 31)
	- table -= 1023;
	- if(i % 64 == 63)
	- scaleval = - scaleval;
	- }
	-}
	-
	-
	diff --git a/util/sdl/sound/decoders/ogg.c b/util/sdl/sound/decoders/ogg.c
	deleted file mode 100644
	index da81c387..00000000
	--- a/util/sdl/sound/decoders/ogg.c
	+++ /dev/null
	@@ -1,375 +0,0 @@
	-/*
	- * SDL_sound -- An abstract sound format decoding API.
	- * Copyright (C) 2001 Ryan C. Gordon.
	- *
	- * This library is free software; you can redistribute it and/or
	- * modify it under the terms of the GNU Lesser General Public
	- * License as published by the Free Software Foundation; either
	- * version 2.1 of the License, or (at your option) any later version.
	- *
	- * This library is distributed in the hope that it will be useful,
	- * but WITHOUT ANY WARRANTY; without even the implied warranty of
	- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	- * Lesser General Public License for more details.
	- *
	- * You should have received a copy of the GNU Lesser General Public
	- * License along with this library; if not, write to the Free Software
	- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
	- */
	-
	-/*
	- * Ogg Vorbis decoder for SDL_sound.
	- *
	- * This driver handles .OGG audio files, and depends on libvorbisfile to
	- * do the actual decoding work. libvorbisfile is part of libvorbis, which
	- * is part of the Ogg Vorbis project.
	- *
	- * Ogg Vorbis: http://www.xiph.org/ogg/vorbis/
	- * vorbisfile documentation: http://www.xiph.org/ogg/vorbis/doc/vorbisfile/
	- *
	- * Please see the file COPYING in the source's root directory.
	- *
	- * This file written by Ryan C. Gordon. (icculus@icculus.org)
	- */
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#ifdef SOUND_SUPPORTS_OGG
	-
	-#include <stdio.h>
	-#include <stdlib.h>
	-#include <string.h>
	-#include <math.h>
	-
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-#include <vorbis/codec.h>
	-#include <vorbis/vorbisfile.h>
	-
	-
	-static int OGG_init(void);
	-static void OGG_quit(void);
	-static int OGG_open(Sound_Sample sample, const char ext);
	-static void OGG_close(Sound_Sample *sample);
	-static Uint32 OGG_read(Sound_Sample *sample);
	-static int OGG_rewind(Sound_Sample *sample);
	-static int OGG_seek(Sound_Sample *sample, Uint32 ms);
	-
	-static const char *extensions_ogg[] = { "OGG", NULL };
	-const Sound_DecoderFunctions __Sound_DecoderFunctions_OGG =
	-{
	- {
	- extensions_ogg,
	- "Ogg Vorbis audio through VorbisFile",
	- "Ryan C. Gordon <icculus@icculus.org>",
	- "http://www.icculus.org/SDL_sound/"
	- },
	-
	- OGG_init, /* init() method */
	- OGG_quit, /* quit() method */
	- OGG_open, /* open() method */
	- OGG_close, /* close() method */
	- OGG_read, /* read() method */
	- OGG_rewind, /* rewind() method */
	- OGG_seek /* seek() method */
	-};
	-
	-
	-static int OGG_init(void)
	-{
	- return(1); /* always succeeds. */
	-} /* OGG_init */
	-
	-
	-static void OGG_quit(void)
	-{
	- /* it's a no-op. */
	-} /* OGG_quit */
	-
	-
	-
	- /*
	- * These are callbacks from vorbisfile that let them read data from
	- * a RWops...
	- */
	-
	-static size_t RWops_ogg_read(void ptr, size_t size, size_t nmemb, void datasource)
	-{
	- return((size_t) SDL_RWread((SDL_RWops *) datasource, ptr, size, nmemb));
	-} /* RWops_ogg_read */
	-
	-static int RWops_ogg_seek(void *datasource, ogg_int64_t offset, int whence)
	-{
	- return(SDL_RWseek((SDL_RWops *) datasource, offset, whence));
	-} /* RWops_ogg_seek */
	-
	-static int RWops_ogg_close(void *datasource)
	-{
	- /* do nothing; SDL_sound will delete the RWops at a higher level. */
	- return(0); /* this is success in fclose(), so I guess that's okay. */
	-} /* RWops_ogg_close */
	-
	-static long RWops_ogg_tell(void *datasource)
	-{
	- return((long) SDL_RWtell((SDL_RWops *) datasource));
	-} /* RWops_ogg_tell */
	-
	-static const ov_callbacks RWops_ogg_callbacks =
	-{
	- RWops_ogg_read,
	- RWops_ogg_seek,
	- RWops_ogg_close,
	- RWops_ogg_tell
	-};
	-
	-
	- /* Return a human readable version of an VorbisFile error code... */
	-#if (defined DEBUG_CHATTER)
	-static const char *ogg_error(int errnum)
	-{
	- switch(errnum)
	- {
	- case OV_EREAD:
	- return("i/o error");
	- case OV_ENOTVORBIS:
	- return("not a vorbis file");
	- case OV_EVERSION:
	- return("Vorbis version mismatch");
	- case OV_EBADHEADER:
	- return("invalid Vorbis bitstream header");
	- case OV_EFAULT:
	- return("internal logic fault in Vorbis library");
	- } /* switch */
	-
	- return("unknown error");
	-} /* ogg_error */
	-#endif
	-
	-static __inline__ void output_ogg_comments(OggVorbis_File *vf)
	-{
	-#if (defined DEBUG_CHATTER)
	- int i;
	- vorbis_comment *vc = ov_comment(vf, -1);
	-
	- if (vc == NULL)
	- return;
	-
	- SNDDBG(("OGG: vendor == [%s].\n", vc->vendor));
	- for (i = 0; i < vc->comments; i++)
	- {
	- SNDDBG(("OGG: user comment [%s].\n", vc->user_comments[i]));
	- } /* for */
	-#endif
	-} /* output_ogg_comments */
	-
	-
	-static int OGG_open(Sound_Sample sample, const char ext)
	-{
	- int rc;
	- OggVorbis_File *vf;
	- vorbis_info *info;
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	-
	- vf = (OggVorbis_File *) malloc(sizeof (OggVorbis_File));
	- BAIL_IF_MACRO(vf == NULL, ERR_OUT_OF_MEMORY, 0);
	-
	- rc = ov_open_callbacks(internal->rw, vf, NULL, 0, RWops_ogg_callbacks);
	- if (rc != 0)
	- {
	- SNDDBG(("OGG: can't grok data. reason: [%s].\n", ogg_error(rc)));
	- free(vf);
	- BAIL_MACRO("OGG: Not valid Ogg Vorbis data.", 0);
	- } /* if */
	-
	- info = ov_info(vf, -1);
	- if (info == NULL)
	- {
	- ov_clear(vf);
	- free(vf);
	- BAIL_MACRO("OGG: failed to retrieve bitstream info", 0);
	- } /* if */
	-
	- SNDDBG(("OGG: Accepting data stream.\n"));
	-
	- output_ogg_comments(vf);
	- SNDDBG(("OGG: bitstream version == (%d).\n", info->version));
	- SNDDBG(("OGG: bitstream channels == (%d).\n", info->channels));
	- SNDDBG(("OGG: bitstream sampling rate == (%ld).\n", info->rate));
	- SNDDBG(("OGG: seekable == {%s}.\n", ov_seekable(vf) ? "TRUE" : "FALSE"));
	- SNDDBG(("OGG: number of logical bitstreams == (%ld).\n", ov_streams(vf)));
	- SNDDBG(("OGG: serial number == (%ld).\n", ov_serialnumber(vf, -1)));
	- SNDDBG(("OGG: total seconds of sample == (%f).\n", ov_time_total(vf, -1)));
	-
	- internal->decoder_private = vf;
	- sample->flags = SOUND_SAMPLEFLAG_CANSEEK;
	- sample->actual.rate = (Uint32) info->rate;
	- sample->actual.channels = (Uint8) info->channels;
	-
	- /*
	- * Since we might have more than one logical bitstream in the OGG file,
	- * and these bitstreams may be in different formats, we might be
	- * converting two or three times: once in vorbisfile, once again in
	- * SDL_sound, and perhaps a third time to get it to the sound device's
	- * format. That's wickedly inefficient.
	- *
	- * To combat this a little, if the user specified a desired format, we
	- * claim that to be the "actual" format of the collection of logical
	- * bitstreams. This means that VorbisFile will do a conversion as
	- * necessary, and SDL_sound will not. If the user didn't specify a
	- * desired format, then we pretend the "actual" format is something that
	- * OGG files are apparently commonly encoded in.
	- */
	- sample->actual.format = (sample->desired.format == 0) ?
	- AUDIO_S16LSB : sample->desired.format;
	- return(1);
	-} /* OGG_open */
	-
	-
	-static void OGG_close(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- OggVorbis_File vf = (OggVorbis_File ) internal->decoder_private;
	- ov_clear(vf);
	- free(vf);
	-} /* OGG_close */
	-
	-/* Note: According to the Vorbis documentation:
	- * "ov_read() will decode at most one vorbis packet per invocation,
	- * so the value returned will generally be less than length."
	- * Due to this, for buffer sizes like 16384, SDL_Sound was always getting
	- * an underfilled buffer and always setting the EAGAIN flag.
	- * Since the SDL_Sound API implies that the entire buffer
	- * should be filled unless EOF, additional code has been added
	- * to this function to call ov_read() until the buffer is filled.
	- * However, there may still be some corner cases where the buffer
	- * cannot be entirely filled. So be aware.
	- */
	-static Uint32 OGG_read(Sound_Sample *sample)
	-{
	- int rc;
	- int bitstream;
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- OggVorbis_File vf = (OggVorbis_File ) internal->decoder_private;
	-
	- rc = ov_read(vf, internal->buffer, internal->buffer_size,
	- ((sample->actual.format & 0x1000) ? 1 : 0), /* bigendian? */
	- ((sample->actual.format & 0xFF) / 8), /* bytes per sample point */
	- ((sample->actual.format & 0x8000) ? 1 : 0), /* signed data? */
	- &bitstream);
	-
	- /* Make sure the read went smoothly... */
	- if (rc == 0)
	- sample->flags \|= SOUND_SAMPLEFLAG_EOF;
	-
	- else if (rc < 0)
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	-
	- /* If the buffer isn't filled, keep trying to fill it
	- * until no more data can be grabbed */
	- else if ((Uint32) rc < internal->buffer_size)
	- {
	- /* Creating a pointer to the buffer that denotes where to start
	- * writing new data. */
	- Uint8* buffer_start_point = NULL;
	- int total_bytes_read = rc;
	- int bytes_remaining = internal->buffer_size - rc;
	-
	- /* Keep grabbing data until something prevents
	- * us from getting more. (Could be EOF,
	- * packets are too large to fit in remaining
	- * space, or an error.)
	- */
	- while( (rc > 0) && (bytes_remaining > 0) )
	- {
	- /* Set buffer pointer to end of last write */
	- /* All the messiness is to get rid of the warning for
	- * dereferencing a void*
	- */
	- buffer_start_point = &(((Uint8*)internal->buffer)[total_bytes_read]);
	- rc = ov_read(vf, buffer_start_point, bytes_remaining,
	- ((sample->actual.format & 0x1000) ? 1 : 0), /* bigendian? */
	- ((sample->actual.format & 0xFF) / 8), /* bytes per sample point */
	- ((sample->actual.format & 0x8000) ? 1 : 0), /* signed data? */
	- &bitstream);
	- /* Make sure rc > 0 because we don't accidently want
	- * to change the counters if there was an error
	- */
	- if(rc > 0)
	- {
	- total_bytes_read += rc;
	- bytes_remaining = bytes_remaining - rc;
	- }
	- }
	- /* I think the minimum read size is 2, though I'm
	- * not sure about this. (I've hit cases where I
	- * couldn't read less than 4.) What I don't want to do is
	- * accidently claim we hit EOF when the reason rc == 0
	- * is because the requested amount of data was smaller
	- * than the minimum packet size.
	- * For now, I will be conservative
	- * and not set the EOF flag, and let the next call to
	- * this function figure it out.
	- * I think the ERROR flag is safe to set because
	- * it looks like OGG simply returns 0 if the
	- * read size is too small.
	- * And in most cases for sensible buffer sizes,
	- * this fix will fill the buffer,
	- * so we can set the EAGAIN flag without worrying
	- * that it will always be set.
	- */
	- if(rc < 0)
	- {
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- }
	- else if(rc == 0)
	- {
	- /* Do nothing for now until there is a better solution */
	- /* sample->flags \|= SOUND_SAMPLEFLAG_EOF; */
	- }
	-
	- /* Test for a buffer underrun. It should occur less frequently
	- * now, but it still may happen and not necessarily mean
	- * anything useful. */
	- if ((Uint32) total_bytes_read < internal->buffer_size)
	- {
	- sample->flags \|= SOUND_SAMPLEFLAG_EAGAIN;
	- }
	- /* change rc to the total bytes read so function
	- * can return the correct value.
	- */
	- rc = total_bytes_read;
	- }
	-
	- return((Uint32) rc);
	-} /* OGG_read */
	-
	-
	-static int OGG_rewind(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- OggVorbis_File vf = (OggVorbis_File ) internal->decoder_private;
	-
	- BAIL_IF_MACRO(ov_raw_seek(vf, 0) < 0, ERR_IO_ERROR, 0);
	- return(1);
	-} /* OGG_rewind */
	-
	-
	-static int OGG_seek(Sound_Sample *sample, Uint32 ms)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- OggVorbis_File vf = (OggVorbis_File ) internal->decoder_private;
	- double timepos = (((double) ms) / 1000.0);
	- BAIL_IF_MACRO(ov_time_seek(vf, timepos) < 0, ERR_IO_ERROR, 0);
	- return(1);
	-} /* OGG_seek */
	-
	-#endif /* SOUND_SUPPORTS_OGG */
	-
	-
	-/* end of ogg.c ... */
	-
	diff --git a/util/sdl/sound/decoders/quicktime.c b/util/sdl/sound/decoders/quicktime.c
	deleted file mode 100644
	index 355ca6e7..00000000
	--- a/util/sdl/sound/decoders/quicktime.c
	+++ /dev/null
	@@ -1,591 +0,0 @@
	-/*
	- * SDL_sound -- An abstract sound format decoding API.
	- * Copyright (C) 2001 Ryan C. Gordon.
	- *
	- * This library is free software; you can redistribute it and/or
	- * modify it under the terms of the GNU Lesser General Public
	- * License as published by the Free Software Foundation; either
	- * version 2.1 of the License, or (at your option) any later version.
	- *
	- * This library is distributed in the hope that it will be useful,
	- * but WITHOUT ANY WARRANTY; without even the implied warranty of
	- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	- * Lesser General Public License for more details.
	- *
	- * You should have received a copy of the GNU Lesser General Public
	- * License along with this library; if not, write to the Free Software
	- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
	- */
	-
	-/*
	- * QuickTime decoder for sound formats that QuickTime supports.
	- * April 28, 2002
	- *
	- * This driver handles .mov files with a sound track. In
	- * theory, it could handle any format that QuickTime supports.
	- * In practice, it may only handle a select few of these formats.
	- *
	- * It seems able to play back AIFF and other standard Mac formats.
	- * Rewinding is not supported yet.
	- *
	- * The routine QT_create_data_ref() needs to be
	- * tweaked to support different media types.
	- * This code was originally written to get MP3 support,
	- * as it turns out, this isn't possible using this method.
	- *
	- * The only way to get streaming MP3 support through QuickTime,
	- * and hence support for SDL_RWops, is to write
	- * a DataHandler component, which suddenly gets much more difficult :-(
	- *
	- * This file was written by Darrell Walisser (walisser@mac.com)
	- * Portions have been borrowed from the "MP3Player" sample code,
	- * courtesy of Apple.
	- */
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#ifdef SOUND_SUPPORTS_QUICKTIME
	-#ifdef macintosh
	-typedef long int32_t;
	-# define OPAQUE_UPP_TYPES 0
	-# include <QuickTime.h>
	-#else
	-# include <QuickTime/QuickTime.h>
	-# include <Carbon/Carbon.h>
	-#endif
	-
	-#include <stdio.h>
	-#include <stdlib.h>
	-#include <stdint.h>
	-#include <string.h>
	-
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-static int QT_init(void);
	-static void QT_quit(void);
	-static int QT_open(Sound_Sample sample, const char ext);
	-static void QT_close(Sound_Sample *sample);
	-static Uint32 QT_read(Sound_Sample *sample);
	-static int QT_rewind(Sound_Sample *sample);
	-static int QT_seek(Sound_Sample *sample, Uint32 ms);
	-
	-#define QT_MAX_INPUT_BUFFER (321024) / Maximum size of internal buffer (internal->buffer_size) */
	-
	-static const char *extensions_quicktime[] = { "mov", NULL };
	-const Sound_DecoderFunctions __Sound_DecoderFunctions_QuickTime =
	- {
	- {
	- extensions_quicktime,
	- "QuickTime format",
	- "Darrell Walisser <dwaliss1@purdue.edu>",
	- "http://www.icculus.org/SDL_sound/"
	- },
	-
	- QT_init, /* init() method */
	- QT_quit, /* quit() method */
	- QT_open, /* open() method */
	- QT_close, /* close() method */
	- QT_read, /* read() method */
	- QT_rewind, /* rewind() method */
	- QT_seek /* seek() method */
	- };
	-
	-typedef struct {
	-
	- ExtendedSoundComponentData compData;
	- Handle hSource; /* source media buffer */
	- Media sourceMedia; /* sound media identifier */
	- TimeValue getMediaAtThisTime;
	- TimeValue sourceDuration;
	- Boolean isThereMoreSource;
	- UInt32 maxBufferSize;
	-
	-} SCFillBufferData, *SCFillBufferDataPtr;
	-
	-typedef struct {
	-
	- Movie movie;
	- Track track;
	- Media media;
	- AudioFormatAtomPtr atom;
	- SoundComponentData source_format;
	- SoundComponentData dest_format;
	- SoundConverter converter;
	- SCFillBufferData buffer_data;
	- SoundConverterFillBufferDataUPP fill_buffer_proc;
	-
	-} qt_t;
	-
	-
	-
	-
	-/*
	- * This procedure creates a description of the raw data
	- * read from SDL_RWops so that QuickTime can identify
	- * the codec it needs to use to decompress it.
	- */
	-static Handle QT_create_data_ref (const char *file_extension) {
	-
	- Handle tmp_handle, data_ref;
	- StringPtr file_name = "\p"; /* empty since we don't know the file name! */
	- OSType file_type;
	- StringPtr mime_type;
	- long atoms[3];
	-
	-/*
	- if (__Sound_strcasecmp (file_extension, "mp3")==0) {
	- file_type = 'MPEG';
	- mime_type = "\pvideo/mpeg";
	- }
	- else {
	-
	- return NULL;
	- }
	-*/
	-
	- if (__Sound_strcasecmp (file_extension, "mov") == 0) {
	-
	- file_type = 'MooV';
	- mime_type = "\pvideo/quicktime";
	- }
	- else {
	-
	- return NULL;
	- }
	-
	- tmp_handle = NewHandle(0);
	- assert (tmp_handle != NULL);
	- assert (noErr == PtrToHand (&tmp_handle, &data_ref, sizeof(Handle)));
	- assert (noErr == PtrAndHand (file_name, data_ref, file_name[0]+1));
	-
	- atoms[0] = EndianU32_NtoB (sizeof(long) * 3);
	- atoms[1] = EndianU32_NtoB (kDataRefExtensionMacOSFileType);
	- atoms[2] = EndianU32_NtoB (file_type);
	-
	- assert (noErr == PtrAndHand (atoms, data_ref, sizeof(long)*3));
	-
	- atoms[0] = EndianU32_NtoB (sizeof(long)*2 + mime_type[0]+1);
	- atoms[1] = EndianU32_NtoB (kDataRefExtensionMIMEType);
	-
	- assert (noErr == PtrAndHand (atoms, data_ref, sizeof(long)*2));
	- assert (noErr == PtrAndHand (mime_type, data_ref, mime_type[0]+1));
	-
	- return data_ref;
	-}
	-
	-/*
	- * This procedure is a hook for QuickTime to grab data from the
	- * SDL_RWOps data structure when it needs it
	- */
	-static pascal OSErr QT_get_movie_data_proc (long offset, long size,
	- void data, void user_data)
	-{
	- SDL_RWops* rw = (SDL_RWops*)user_data;
	- OSErr error;
	-
	- if (offset == SDL_RWseek (rw, offset, SEEK_SET)) {
	-
	- if (size == SDL_RWread (rw, data, 1, size)) {
	- error = noErr;
	- }
	- else {
	- error = notEnoughDataErr;
	- }
	- }
	- else {
	- error = fileOffsetTooBigErr;
	- }
	-
	- return (error);
	-}
	-
	-/* * ----------------------------
	- * SoundConverterFillBufferDataProc
	- *
	- * the callback routine that provides the source data for conversion -
	- * it provides data by setting outData to a pointer to a properly
	- * filled out ExtendedSoundComponentData structure
	- */
	-static pascal Boolean QT_sound_converter_fill_buffer_data_proc (SoundComponentDataPtr outData, void inRefCon)
	-{
	- SCFillBufferDataPtr pFillData = (SCFillBufferDataPtr)inRefCon;
	-
	- OSErr err = noErr;
	-
	- /* if after getting the last chunk of data the total time is over
	- * the duration, we're done
	- */
	- if (pFillData->getMediaAtThisTime >= pFillData->sourceDuration) {
	- pFillData->isThereMoreSource = false;
	- pFillData->compData.desc.buffer = NULL;
	- pFillData->compData.desc.sampleCount = 0;
	- pFillData->compData.bufferSize = 0;
	- }
	-
	- if (pFillData->isThereMoreSource) {
	-
	- long sourceBytesReturned;
	- long numberOfSamples;
	- TimeValue sourceReturnedTime, durationPerSample;
	-
	- HUnlock(pFillData->hSource);
	-
	- err = GetMediaSample
	- (pFillData->sourceMedia,/* specifies the media for this operation */
	- pFillData->hSource, /* function returns the sample data into this handle */
	- pFillData->maxBufferSize, /* maximum number of bytes of sample data to be returned */
	- &sourceBytesReturned, /* the number of bytes of sample data returned */
	- pFillData->getMediaAtThisTime,/* starting time of the sample to
	- be retrieved (must be in
	- Media's TimeScale) */
	- &sourceReturnedTime,/* indicates the actual time of the returned sample data */
	- &durationPerSample, /* duration of each sample in the media */
	- NULL, /* sample description corresponding to the returned sample data (NULL to ignore) */
	- NULL, /* index value to the sample description that corresponds
	- to the returned sample data (NULL to ignore) */
	- 0, /* maximum number of samples to be returned (0 to use a
	- value that is appropriate for the media) */
	- &numberOfSamples, /* number of samples it actually returned */
	- NULL); /* flags that describe the sample (NULL to ignore) */
	-
	- HLock(pFillData->hSource);
	-
	- if ((noErr != err) \|\| (sourceBytesReturned == 0)) {
	- pFillData->isThereMoreSource = false;
	- pFillData->compData.desc.buffer = NULL;
	- pFillData->compData.desc.sampleCount = 0;
	-
	- if ((err != noErr) && (sourceBytesReturned > 0))
	- DebugStr("\pGetMediaSample - Failed in FillBufferDataProc");
	- }
	-
	- pFillData->getMediaAtThisTime = sourceReturnedTime + (durationPerSample * numberOfSamples);
	- pFillData->compData.bufferSize = sourceBytesReturned;
	- }
	-
	- /* set outData to a properly filled out ExtendedSoundComponentData struct */
	- *outData = (SoundComponentDataPtr)&pFillData->compData;
	-
	- return (pFillData->isThereMoreSource);
	-}
	-
	-
	-static int QT_init_internal () {
	-
	- OSErr error;
	-
	- error = EnterMovies(); /* initialize the movie toolbox */
	-
	- return (error == noErr);
	-}
	-
	-static void QT_quit_internal () {
	-
	- ExitMovies();
	-}
	-
	-static qt_t* QT_open_internal (Sound_Sample sample, const char extension)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	-
	- qt_t *instance;
	- OSErr error;
	- Movie movie;
	- Track sound_track;
	- Media sound_track_media;
	- AudioFormatAtomPtr source_sound_decomp_atom;
	-
	- SoundDescriptionV1Handle source_sound_description;
	- Handle source_sound_description_extension;
	- Size source_sound_description_extension_size;
	- Handle data_ref;
	-
	- data_ref = QT_create_data_ref (extension);
	-
	- /* create a movie that will read data using SDL_RWops */
	- error = NewMovieFromUserProc
	- (&movie,
	- 0,
	- NULL,
	- NewGetMovieUPP(QT_get_movie_data_proc),
	- (void*) internal->rw,
	- data_ref,
	- 'hndl');
	-
	- if (error != noErr) {
	-
	- return NULL;
	- }
	-
	- /* get the first sound track of the movie; other tracks will be ignored */
	- sound_track = GetMovieIndTrackType (movie, 1, SoundMediaType, movieTrackMediaType);
	- if (sound_track == NULL) {
	-
	- /* movie needs a sound track! */
	-
	- return NULL;
	- }
	-
	- /* get and return the sound track media */
	- sound_track_media = GetTrackMedia (sound_track);
	- if (sound_track_media == NULL) {
	-
	- return NULL;
	- }
	-
	- /* create a description of the source sound so we can convert it later */
	- source_sound_description = (SoundDescriptionV1Handle)NewHandle(0);
	- assert (source_sound_description != NULL); /* out of memory */
	-
	- GetMediaSampleDescription (sound_track_media, 1,
	- (SampleDescriptionHandle)source_sound_description);
	- error = GetMoviesError();
	- if (error != noErr) {
	-
	- return NULL;
	- }
	-
	- source_sound_description_extension = NewHandle(0);
	- assert (source_sound_description_extension != NULL); /* out of memory */
	-
	- error = GetSoundDescriptionExtension ((SoundDescriptionHandle) source_sound_description,
	- &source_sound_description_extension,
	- siDecompressionParams);
	-
	- if (error == noErr) {
	-
	- /* copy extension to atom format description if we have an extension */
	-
	- source_sound_description_extension_size =
	- GetHandleSize (source_sound_description_extension);
	- HLock (source_sound_description_extension);
	-
	- source_sound_decomp_atom = (AudioFormatAtom*)
	- NewPtr (source_sound_description_extension_size);
	- assert (source_sound_decomp_atom != NULL); /* out of memory */
	-
	- BlockMoveData (*source_sound_description_extension,
	- source_sound_decomp_atom,
	- source_sound_description_extension_size);
	-
	- HUnlock (source_sound_description_extension);
	- }
	-
	- else {
	-
	- source_sound_decomp_atom = NULL;
	- }
	-
	- instance = (qt_t) malloc (sizeof(instance));
	- assert (instance != NULL); /* out of memory */
	-
	- instance->movie = movie;
	- instance->track = sound_track;
	- instance->media = sound_track_media;
	- instance->atom = source_sound_decomp_atom;
	-
	- instance->source_format.flags = 0;
	- instance->source_format.format = (*source_sound_description)->desc.dataFormat;
	- instance->source_format.numChannels = (*source_sound_description)->desc.numChannels;
	- instance->source_format.sampleSize = (*source_sound_description)->desc.sampleSize;
	- instance->source_format.sampleRate = (*source_sound_description)->desc.sampleRate;
	- instance->source_format.sampleCount = 0;
	- instance->source_format.buffer = NULL;
	- instance->source_format.reserved = 0;
	-
	- instance->dest_format.flags = 0;
	- instance->dest_format.format = kSoundNotCompressed;
	- instance->dest_format.numChannels = (*source_sound_description)->desc.numChannels;
	- instance->dest_format.sampleSize = (*source_sound_description)->desc.sampleSize;
	- instance->dest_format.sampleRate = (*source_sound_description)->desc.sampleRate;
	- instance->dest_format.sampleCount = 0;
	- instance->dest_format.buffer = NULL;
	- instance->dest_format.reserved = 0;
	-
	- sample->actual.channels = (*source_sound_description)->desc.numChannels;
	- sample->actual.rate = (*source_sound_description)->desc.sampleRate >> 16;
	-
	- if ((*source_sound_description)->desc.sampleSize == 16) {
	-
	- sample->actual.format = AUDIO_S16SYS;
	- }
	- else if ((*source_sound_description)->desc.sampleSize == 8) {
	-
	- sample->actual.format = AUDIO_U8;
	- }
	- else {
	-
	- /* 24-bit or others... (which SDL can't handle) */
	- return NULL;
	- }
	-
	- DisposeHandle (source_sound_description_extension);
	- DisposeHandle ((Handle)source_sound_description);
	-
	- /* This next code sets up the SoundConverter component */
	- error = SoundConverterOpen (&instance->source_format, &instance->dest_format,
	- &instance->converter);
	-
	- if (error != noErr) {
	-
	- return NULL;
	- }
	-
	- error = SoundConverterSetInfo (instance->converter, siDecompressionParams,
	- instance->atom);
	- if (error == siUnknownInfoType) {
	-
	- /* ignore */
	- }
	- else if (error != noErr) {
	-
	- /* reall error */
	- return NULL;
	- }
	-
	- error = SoundConverterBeginConversion (instance->converter);
	- if (error != noErr) {
	-
	- return NULL;
	- }
	-
	- instance->buffer_data.sourceMedia = instance->media;
	- instance->buffer_data.getMediaAtThisTime = 0;
	- instance->buffer_data.sourceDuration = GetMediaDuration(instance->media);
	- instance->buffer_data.isThereMoreSource = true;
	- instance->buffer_data.maxBufferSize = QT_MAX_INPUT_BUFFER;
	- /* allocate source media buffer */
	- instance->buffer_data.hSource = NewHandle((long)instance->buffer_data.maxBufferSize);
	- assert (instance->buffer_data.hSource != NULL); /* out of memory */
	-
	- instance->buffer_data.compData.desc = instance->source_format;
	- instance->buffer_data.compData.desc.buffer = (Byte )instance->buffer_data.hSource;
	- instance->buffer_data.compData.desc.flags = kExtendedSoundData;
	- instance->buffer_data.compData.recordSize = sizeof(ExtendedSoundComponentData);
	- instance->buffer_data.compData.extendedFlags =
	- kExtendedSoundSampleCountNotValid \| kExtendedSoundBufferSizeValid;
	- instance->buffer_data.compData.bufferSize = 0;
	-
	- instance->fill_buffer_proc =
	- NewSoundConverterFillBufferDataUPP (QT_sound_converter_fill_buffer_data_proc);
	-
	- return (instance);
	-
	-} /* QT_open_internal */
	-
	-static void QT_close_internal (qt_t *instance)
	-{
	-
	-} /* QT_close_internal */
	-
	-static Uint32 QT_read_internal(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- qt_t instance = (qt_t) internal->decoder_private;
	- long output_bytes, output_frames, output_flags;
	- OSErr error;
	-
	- error = SoundConverterFillBuffer
	- (instance->converter, /* a sound converter */
	- instance->fill_buffer_proc, /* the callback UPP */
	- &instance->buffer_data, /* refCon passed to FillDataProc */
	- internal->buffer, /* the decompressed data 'play' buffer */
	- internal->buffer_size, /* size of the 'play' buffer */
	- &output_bytes, /* number of output bytes */
	- &output_frames, /* number of output frames */
	- &output_flags); /* fillbuffer retured advisor flags */
	-
	- if (output_flags & kSoundConverterHasLeftOverData) {
	-
	- sample->flags \|= SOUND_SAMPLEFLAG_EAGAIN;
	- }
	- else {
	-
	- sample->flags \|= SOUND_SAMPLEFLAG_EOF;
	- }
	-
	- if (error != noErr) {
	-
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- }
	-
	- return (output_bytes);
	-
	-} /* QT_read_internal */
	-
	-static int QT_rewind_internal (Sound_Sample *sample)
	-{
	-
	- return 0;
	-
	-} /* QT_rewind_internal */
	-
	-
	-
	-static int QT_init(void)
	-{
	- return (QT_init_internal());
	-
	-} /* QT_init */
	-
	-static void QT_quit(void)
	-{
	- QT_quit_internal();
	-
	-} /* QT_quit */
	-
	-static int QT_open(Sound_Sample sample, const char ext)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- qt_t *instance;
	-
	- instance = QT_open_internal(sample, ext);
	- internal->decoder_private = (void*)instance;
	-
	- return(instance != NULL);
	-
	-} /* QT_open */
	-
	-
	-static void QT_close(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- qt_t instance = (qt_t ) internal->decoder_private;
	-
	- QT_close_internal (instance);
	-
	- free (instance);
	-
	-} /* QT_close */
	-
	-
	-static Uint32 QT_read(Sound_Sample *sample)
	-{
	- return(QT_read_internal(sample));
	-
	-} /* QT_read */
	-
	-
	-static int QT_rewind(Sound_Sample *sample)
	-{
	-
	- return(QT_rewind_internal(sample));
	-
	-} /* QT_rewind */
	-
	-
	-static int QT_seek(Sound_Sample *sample, Uint32 ms)
	-{
	- BAIL_MACRO("QUICKTIME: Seeking not implemented", 0);
	-} /* QT_seek */
	-
	-
	-#endif /* SOUND_SUPPORTS_QUICKTIME */
	-
	-/* end of quicktime.c ... */
	-
	diff --git a/util/sdl/sound/decoders/raw.c b/util/sdl/sound/decoders/raw.c
	deleted file mode 100644
	index be3c810f..00000000
	--- a/util/sdl/sound/decoders/raw.c
	+++ /dev/null
	@@ -1,184 +0,0 @@
	-/*
	- * SDL_sound -- An abstract sound format decoding API.
	- * Copyright (C) 2001 Ryan C. Gordon.
	- *
	- * This library is free software; you can redistribute it and/or
	- * modify it under the terms of the GNU Lesser General Public
	- * License as published by the Free Software Foundation; either
	- * version 2.1 of the License, or (at your option) any later version.
	- *
	- * This library is distributed in the hope that it will be useful,
	- * but WITHOUT ANY WARRANTY; without even the implied warranty of
	- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	- * Lesser General Public License for more details.
	- *
	- * You should have received a copy of the GNU Lesser General Public
	- * License along with this library; if not, write to the Free Software
	- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
	- */
	-
	-/*
	- * RAW decoder for SDL_sound. This is as simple as it gets.
	- *
	- * This driver handles raw audio data. You must, regardless of where the
	- * data is actually coming from, specify the string "RAW" in the extension
	- * parameter of Sound_NewSample() (or, alternately, open a file with the
	- * extension ".raw" in Sound_NewSampleFromFile()). The string is checked
	- * case-insensitive. We need this check, because raw data, being raw, has
	- * no headers or magic number we can use to determine if we should handle a
	- * given file, so we needed some way to have this "decoder" discriminate.
	- *
	- * When calling Sound_NewSample*(), you must also specify a "desired"
	- * audio format. The "actual" format will always match what you specify, so
	- * there will be no conversion overhead, but these routines need to know how
	- * to treat the bits, since it's all random garbage otherwise.
	- *
	- * Please see the file COPYING in the source's root directory.
	- *
	- * This file written by Ryan C. Gordon. (icculus@icculus.org)
	- */
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#ifdef SOUND_SUPPORTS_RAW
	-
	-#include <stdio.h>
	-#include <stdlib.h>
	-#include <string.h>
	-
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-static int RAW_init(void);
	-static void RAW_quit(void);
	-static int RAW_open(Sound_Sample sample, const char ext);
	-static void RAW_close(Sound_Sample *sample);
	-static Uint32 RAW_read(Sound_Sample *sample);
	-static int RAW_rewind(Sound_Sample *sample);
	-static int RAW_seek(Sound_Sample *sample, Uint32 ms);
	-
	-static const char *extensions_raw[] = { "RAW", NULL };
	-const Sound_DecoderFunctions __Sound_DecoderFunctions_RAW =
	-{
	- {
	- extensions_raw,
	- "Raw audio",
	- "Ryan C. Gordon <icculus@icculus.org>",
	- "http://www.icculus.org/SDL_sound/"
	- },
	-
	- RAW_init, /* init() method */
	- RAW_quit, /* quit() method */
	- RAW_open, /* open() method */
	- RAW_close, /* close() method */
	- RAW_read, /* read() method */
	- RAW_rewind, /* rewind() method */
	- RAW_seek /* seek() method */
	-};
	-
	-
	-static int RAW_init(void)
	-{
	- return(1); /* always succeeds. */
	-} /* RAW_init */
	-
	-
	-static void RAW_quit(void)
	-{
	- /* it's a no-op. */
	-} /* RAW_quit */
	-
	-
	-static int RAW_open(Sound_Sample sample, const char ext)
	-{
	- /*
	- * We check this explicitly, since we have no other way to
	- * determine whether we should handle this data or not.
	- */
	- if (__Sound_strcasecmp(ext, "RAW") != 0)
	- BAIL_MACRO("RAW: extension isn't explicitly \"RAW\".", 0);
	-
	- /*
	- * You must also specify a desired format, so we know how to
	- * treat the bits that are otherwise binary garbage.
	- */
	- if ( (sample->desired.channels < 1) \|\|
	- (sample->desired.channels > 2) \|\|
	- (sample->desired.rate == 0) \|\|
	- (sample->desired.format == 0) )
	- {
	- BAIL_MACRO("RAW: invalid desired format.", 0);
	- } /* if */
	-
	- SNDDBG(("RAW: Accepting data stream.\n"));
	-
	- /*
	- * We never convert raw samples; what you ask for is what you get.
	- */
	- memcpy(&sample->actual, &sample->desired, sizeof (Sound_AudioInfo));
	- sample->flags = SOUND_SAMPLEFLAG_CANSEEK;
	-
	- return(1); /* we'll handle this data. */
	-} /* RAW_open */
	-
	-
	-static void RAW_close(Sound_Sample *sample)
	-{
	- /* we don't allocate anything that we need to free. That's easy, eh? */
	-} /* RAW_close */
	-
	-
	-static Uint32 RAW_read(Sound_Sample *sample)
	-{
	- Uint32 retval;
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	-
	- /*
	- * We don't actually do any decoding, so we read the raw data
	- * directly into the internal buffer...
	- */
	- retval = SDL_RWread(internal->rw, internal->buffer,
	- 1, internal->buffer_size);
	-
	- /* Make sure the read went smoothly... */
	- if (retval == 0)
	- sample->flags \|= SOUND_SAMPLEFLAG_EOF;
	-
	- else if (retval == -1)
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	-
	- /* (next call this EAGAIN may turn into an EOF or error.) */
	- else if (retval < internal->buffer_size)
	- sample->flags \|= SOUND_SAMPLEFLAG_EAGAIN;
	-
	- return(retval);
	-} /* RAW_read */
	-
	-
	-static int RAW_rewind(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- BAIL_IF_MACRO(SDL_RWseek(internal->rw, 0, SEEK_SET) != 0, ERR_IO_ERROR, 0);
	- return(1);
	-} /* RAW_rewind */
	-
	-
	-static int RAW_seek(Sound_Sample *sample, Uint32 ms)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- int pos = (int) __Sound_convertMsToBytePos(&sample->actual, ms);
	- int err = (SDL_RWseek(internal->rw, pos, SEEK_SET) != pos);
	- BAIL_IF_MACRO(err, ERR_IO_ERROR, 0);
	- return(1);
	-} /* RAW_seek */
	-
	-
	-#endif /* SOUND_SUPPORTS_RAW */
	-
	-
	-/* end of raw.c ... */
	-
	diff --git a/util/sdl/sound/decoders/shn.c b/util/sdl/sound/decoders/shn.c
	deleted file mode 100644
	index 62a316ff..00000000
	--- a/util/sdl/sound/decoders/shn.c
	+++ /dev/null
	@@ -1,1341 +0,0 @@
	-/*
	- * SDL_sound -- An abstract sound format decoding API.
	- * Copyright (C) 2001 Ryan C. Gordon.
	- *
	- * This library is free software; you can redistribute it and/or
	- * modify it under the terms of the GNU Lesser General Public
	- * License as published by the Free Software Foundation; either
	- * version 2.1 of the License, or (at your option) any later version.
	- *
	- * This library is distributed in the hope that it will be useful,
	- * but WITHOUT ANY WARRANTY; without even the implied warranty of
	- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	- * Lesser General Public License for more details.
	- *
	- * You should have received a copy of the GNU Lesser General Public
	- * License along with this library; if not, write to the Free Software
	- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
	- */
	-
	-/*
	- * Shorten decoder for SDL_sound.
	- *
	- * This driver handles Shorten-compressed waveforms. Despite the fact that
	- * SHNs tend to be much bigger than MP3s, they are still the de facto
	- * standard in online music trading communities. If an MP3 crunches the
	- * waveform to 10-20 percent of its original size, SHNs only go to about
	- * 50-60%. Why do the Phish fans of the world use this format then? Rabid
	- * music traders appreciate the sound quality; SHNs, unlike MP3s, do not
	- * throw away any part of the waveform. Yes, there are people that notice
	- * this, and further more, they demand it...and if they can't get a good
	- * transfer of those larger files over the 'net, they haven't underestimated
	- * the bandwidth of CDs travelling the world through the postal system.
	- *
	- * Shorten homepage: http://www.softsound.com/Shorten.html
	- *
	- * The Shorten format was gleaned from the shorten codebase, by Tony
	- * Robinson and SoftSound Limited.
	- *
	- * Please see the file COPYING in the source's root directory.
	- *
	- * This file written by Ryan C. Gordon. (icculus@icculus.org)
	- */
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#ifdef SOUND_SUPPORTS_SHN
	-
	-#include <stdio.h>
	-#include <stdlib.h>
	-#include <string.h>
	-#include <math.h>
	-
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-static int SHN_init(void);
	-static void SHN_quit(void);
	-static int SHN_open(Sound_Sample sample, const char ext);
	-static void SHN_close(Sound_Sample *sample);
	-static Uint32 SHN_read(Sound_Sample *sample);
	-static int SHN_rewind(Sound_Sample *sample);
	-static int SHN_seek(Sound_Sample *sample, Uint32 ms);
	-
	-static const char *extensions_shn[] = { "SHN", NULL };
	-const Sound_DecoderFunctions __Sound_DecoderFunctions_SHN =
	-{
	- {
	- extensions_shn,
	- "Shorten-compressed audio data",
	- "Ryan C. Gordon <icculus@icculus.org>",
	- "http://www.icculus.org/SDL_sound/"
	- },
	-
	- SHN_init, /* init() method */
	- SHN_quit, /* quit() method */
	- SHN_open, /* open() method */
	- SHN_close, /* close() method */
	- SHN_read, /* read() method */
	- SHN_rewind, /* rewind() method */
	- SHN_seek /* seek() method */
	-};
	-
	-
	-#define SHN_BUFSIZ 512
	-
	-typedef struct
	-{
	- Sint32 version;
	- Sint32 datatype;
	- Sint32 nchan;
	- Sint32 blocksize;
	- Sint32 maxnlpc;
	- Sint32 nmean;
	- Sint32 nwrap;
	- Sint32 **buffer;
	- Sint32 **offset;
	- Sint32 *qlpc;
	- Sint32 lpcqoffset;
	- Sint32 bitshift;
	- int nbitget;
	- int nbyteget;
	- Uint8 *getbuf;
	- Uint8 *getbufp;
	- Uint32 gbuffer;
	- Uint8 *backBuffer;
	- Uint32 backBufferSize;
	- Uint32 backBufLeft;
	- Uint32 start_pos;
	-} shn_t;
	-
	-
	-static const Uint32 mask_table[] =
	-{
	- 0x00000000, 0x00000001, 0x00000003, 0x00000007, 0x0000000F, 0x0000001F,
	- 0x0000003F, 0x0000007F, 0x000000FF, 0x000001FF, 0x000003FF, 0x000007FF,
	- 0x00000FFF, 0x00001FFF, 0x00003FFF, 0x00007FFF, 0x0000FFFF, 0x0001FFFF,
	- 0x0003FFFF, 0x0007FFFF, 0x000FFFFF, 0x001FFFFF, 0x003FFFFF, 0x007FFFFF,
	- 0x00FFFFFF, 0x01FFFFFF, 0x03FFFFFF, 0x07FFFFFF, 0x0FFFFFFF, 0x1FFFFFFF,
	- 0x3FFFFFFF, 0x7FFFFFFF, 0xFFFFFFFF
	-};
	-
	-
	-static const Uint8 ulaw_outward[13][256] = {
	-{127,0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15,16,17,18,19,20,21,22,23,24,25,26,27,28,29,30,31,32,33,34,35,36,37,38,39,40,41,42,43,44,45,46,47,48,49,50,51,52,53,54,55,56,57,58,59,60,61,62,63,64,65,66,67,68,69,70,71,72,73,74,75,76,77,78,79,80,81,82,83,84,85,86,87,88,89,90,91,92,93,94,95,96,97,98,99,100,101,102,103,104,105,106,107,108,109,110,111,112,113,114,115,116,117,118,119,120,121,122,123,124,125,126,255,254,253,252,251,250,249,248,247,246,245,244,243,242,241,240,239,238,237,236,235,234,233,232,231,230,229,228,227,226,225,224,223,222,221,220,219,218,217,216,215,214,213,212,211,210,209,208,207,206,205,204,203,202,201,200,199,198,197,196,195,194,193,192,191,190,189,188,187,186,185,184,183,182,181,180,179,178,177,176,175,174,173,172,171,170,169,168,167,166,165,164,163,162,161,160,159,158,157,156,155,154,153,152,151,150,149,148,147,146,145,144,143,142,141,140,139,138,137,136,135,134,133,132,131,130,129,128},
	-{112,114,116,118,120,122,124,126,127,0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15,16,17,18,19,20,21,22,23,24,25,26,27,28,29,30,31,32,33,34,35,36,37,38,39,40,41,42,43,44,45,46,47,48,49,50,51,52,53,54,55,56,57,58,59,60,61,62,63,64,65,66,67,68,69,70,71,72,73,74,75,76,77,78,79,80,81,82,83,84,85,86,87,88,89,90,91,92,93,94,95,96,97,98,99,100,101,102,103,104,105,106,107,108,109,110,111,113,115,117,119,121,123,125,255,253,251,249,247,245,243,241,239,238,237,236,235,234,233,232,231,230,229,228,227,226,225,224,223,222,221,220,219,218,217,216,215,214,213,212,211,210,209,208,207,206,205,204,203,202,201,200,199,198,197,196,195,194,193,192,191,190,189,188,187,186,185,184,183,182,181,180,179,178,177,176,175,174,173,172,171,170,169,168,167,166,165,164,163,162,161,160,159,158,157,156,155,154,153,152,151,150,149,148,147,146,145,144,143,142,141,140,139,138,137,136,135,134,133,132,131,130,129,128,254,252,250,248,246,244,242,240},
	-{96,98,100,102,104,106,108,110,112,113,114,116,117,118,120,121,122,124,125,126,127,0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15,16,17,18,19,20,21,22,23,24,25,26,27,28,29,30,31,32,33,34,35,36,37,38,39,40,41,42,43,44,45,46,47,48,49,50,51,52,53,54,55,56,57,58,59,60,61,62,63,64,65,66,67,68,69,70,71,72,73,74,75,76,77,78,79,80,81,82,83,84,85,86,87,88,89,90,91,92,93,94,95,97,99,101,103,105,107,109,111,115,119,123,255,251,247,243,239,237,235,233,231,229,227,225,223,222,221,220,219,218,217,216,215,214,213,212,211,210,209,208,207,206,205,204,203,202,201,200,199,198,197,196,195,194,193,192,191,190,189,188,187,186,185,184,183,182,181,180,179,178,177,176,175,174,173,172,171,170,169,168,167,166,165,164,163,162,161,160,159,158,157,156,155,154,153,152,151,150,149,148,147,146,145,144,143,142,141,140,139,138,137,136,135,134,133,132,131,130,129,128,254,253,252,250,249,248,246,245,244,242,241,240,238,236,234,232,230,228,226,224},
	-{80,82,84,86,88,90,92,94,96,97,98,100,101,102,104,105,106,108,109,110,112,113,114,115,116,117,118,120,121,122,123,124,125,126,127,0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15,16,17,18,19,20,21,22,23,24,25,26,27,28,29,30,31,32,33,34,35,36,37,38,39,40,41,42,43,44,45,46,47,48,49,50,51,52,53,54,55,56,57,58,59,60,61,62,63,64,65,66,67,68,69,70,71,72,73,74,75,76,77,78,79,81,83,85,87,89,91,93,95,99,103,107,111,119,255,247,239,235,231,227,223,221,219,217,215,213,211,209,207,206,205,204,203,202,201,200,199,198,197,196,195,194,193,192,191,190,189,188,187,186,185,184,183,182,181,180,179,178,177,176,175,174,173,172,171,170,169,168,167,166,165,164,163,162,161,160,159,158,157,156,155,154,153,152,151,150,149,148,147,146,145,144,143,142,141,140,139,138,137,136,135,134,133,132,131,130,129,128,254,253,252,251,250,249,248,246,245,244,243,242,241,240,238,237,236,234,233,232,230,229,228,226,225,224,222,220,218,216,214,212,210,208},
	-{64,66,68,70,72,74,76,78,80,81,82,84,85,86,88,89,90,92,93,94,96,97,98,99,100,101,102,104,105,106,107,108,109,110,112,113,114,115,116,117,118,119,120,121,122,123,124,125,126,127,0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15,16,17,18,19,20,21,22,23,24,25,26,27,28,29,30,31,32,33,34,35,36,37,38,39,40,41,42,43,44,45,46,47,48,49,50,51,52,53,54,55,56,57,58,59,60,61,62,63,65,67,69,71,73,75,77,79,83,87,91,95,103,111,255,239,231,223,219,215,211,207,205,203,201,199,197,195,193,191,190,189,188,187,186,185,184,183,182,181,180,179,178,177,176,175,174,173,172,171,170,169,168,167,166,165,164,163,162,161,160,159,158,157,156,155,154,153,152,151,150,149,148,147,146,145,144,143,142,141,140,139,138,137,136,135,134,133,132,131,130,129,128,254,253,252,251,250,249,248,247,246,245,244,243,242,241,240,238,237,236,235,234,233,232,230,229,228,227,226,225,224,222,221,220,218,217,216,214,213,212,210,209,208,206,204,202,200,198,196,194,192},
	-{49,51,53,55,57,59,61,63,64,66,67,68,70,71,72,74,75,76,78,79,80,81,82,84,85,86,87,88,89,90,92,93,94,95,96,97,98,99,100,101,102,104,105,106,107,108,109,110,111,112,113,114,115,116,117,118,119,120,121,122,123,124,125,126,127,0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15,16,17,18,19,20,21,22,23,24,25,26,27,28,29,30,31,32,33,34,35,36,37,38,39,40,41,42,43,44,45,46,47,48,50,52,54,56,58,60,62,65,69,73,77,83,91,103,255,231,219,211,205,201,197,193,190,188,186,184,182,180,178,176,175,174,173,172,171,170,169,168,167,166,165,164,163,162,161,160,159,158,157,156,155,154,153,152,151,150,149,148,147,146,145,144,143,142,141,140,139,138,137,136,135,134,133,132,131,130,129,128,254,253,252,251,250,249,248,247,246,245,244,243,242,241,240,239,238,237,236,235,234,233,232,230,229,228,227,226,225,224,223,222,221,220,218,217,216,215,214,213,212,210,209,208,207,206,204,203,202,200,199,198,196,195,194,192,191,189,187,185,183,181,179,177},
	-{32,34,36,38,40,42,44,46,48,49,51,52,53,55,56,57,59,60,61,63,64,65,66,67,68,70,71,72,73,74,75,76,78,79,80,81,82,83,84,85,86,87,88,89,90,92,93,94,95,96,97,98,99,100,101,102,103,104,105,106,107,108,109,110,111,112,113,114,115,116,117,118,119,120,121,122,123,124,125,126,127,0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15,16,17,18,19,20,21,22,23,24,25,26,27,28,29,30,31,33,35,37,39,41,43,45,47,50,54,58,62,69,77,91,255,219,205,197,190,186,182,178,175,173,171,169,167,165,163,161,159,158,157,156,155,154,153,152,151,150,149,148,147,146,145,144,143,142,141,140,139,138,137,136,135,134,133,132,131,130,129,128,254,253,252,251,250,249,248,247,246,245,244,243,242,241,240,239,238,237,236,235,234,233,232,231,230,229,228,227,226,225,224,223,222,221,220,218,217,216,215,214,213,212,211,210,209,208,207,206,204,203,202,201,200,199,198,196,195,194,193,192,191,189,188,187,185,184,183,181,180,179,177,176,174,172,170,168,166,164,162,160},
	-{16,18,20,22,24,26,28,30,32,33,34,36,37,38,40,41,42,44,45,46,48,49,50,51,52,53,55,56,57,58,59,60,61,63,64,65,66,67,68,69,70,71,72,73,74,75,76,78,79,80,81,82,83,84,85,86,87,88,89,90,91,92,93,94,95,96,97,98,99,100,101,102,103,104,105,106,107,108,109,110,111,112,113,114,115,116,117,118,119,120,121,122,123,124,125,126,127,0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15,17,19,21,23,25,27,29,31,35,39,43,47,54,62,77,255,205,190,182,175,171,167,163,159,157,155,153,151,149,147,145,143,142,141,140,139,138,137,136,135,134,133,132,131,130,129,128,254,253,252,251,250,249,248,247,246,245,244,243,242,241,240,239,238,237,236,235,234,233,232,231,230,229,228,227,226,225,224,223,222,221,220,219,218,217,216,215,214,213,212,211,210,209,208,207,206,204,203,202,201,200,199,198,197,196,195,194,193,192,191,189,188,187,186,185,184,183,181,180,179,178,177,176,174,173,172,170,169,168,166,165,164,162,161,160,158,156,154,152,150,148,146,144},
	-{2,4,6,8,10,12,14,16,17,18,20,21,22,24,25,26,28,29,30,32,33,34,35,36,37,38,40,41,42,43,44,45,46,48,49,50,51,52,53,54,55,56,57,58,59,60,61,63,64,65,66,67,68,69,70,71,72,73,74,75,76,77,78,79,80,81,82,83,84,85,86,87,88,89,90,91,92,93,94,95,96,97,98,99,100,101,102,103,104,105,106,107,108,109,110,111,112,113,114,115,116,117,118,119,120,121,122,123,124,125,126,127,0,1,3,5,7,9,11,13,15,19,23,27,31,39,47,62,255,190,175,167,159,155,151,147,143,141,139,137,135,133,131,129,254,253,252,251,250,249,248,247,246,245,244,243,242,241,240,239,238,237,236,235,234,233,232,231,230,229,228,227,226,225,224,223,222,221,220,219,218,217,216,215,214,213,212,211,210,209,208,207,206,205,204,203,202,201,200,199,198,197,196,195,194,193,192,191,189,188,187,186,185,184,183,182,181,180,179,178,177,176,174,173,172,171,170,169,168,166,165,164,163,162,161,160,158,157,156,154,153,152,150,149,148,146,145,144,142,140,138,136,134,132,130,128},
	-{1,2,4,5,6,8,9,10,12,13,14,16,17,18,19,20,21,22,24,25,26,27,28,29,30,32,33,34,35,36,37,38,39,40,41,42,43,44,45,46,48,49,50,51,52,53,54,55,56,57,58,59,60,61,62,63,64,65,66,67,68,69,70,71,72,73,74,75,76,77,78,79,80,81,82,83,84,85,86,87,88,89,90,91,92,93,94,95,96,97,98,99,100,101,102,103,104,105,106,107,108,109,110,111,112,113,114,115,116,117,118,119,120,121,122,123,124,125,126,127,0,3,7,11,15,23,31,47,255,175,159,151,143,139,135,131,254,253,252,251,250,249,248,247,246,245,244,243,242,241,240,239,238,237,236,235,234,233,232,231,230,229,228,227,226,225,224,223,222,221,220,219,218,217,216,215,214,213,212,211,210,209,208,207,206,205,204,203,202,201,200,199,198,197,196,195,194,193,192,191,190,189,188,187,186,185,184,183,182,181,180,179,178,177,176,174,173,172,171,170,169,168,167,166,165,164,163,162,161,160,158,157,156,155,154,153,152,150,149,148,147,146,145,144,142,141,140,138,137,136,134,133,132,130,129,128},
	-{1,2,3,4,5,6,8,9,10,11,12,13,14,16,17,18,19,20,21,22,23,24,25,26,27,28,29,30,32,33,34,35,36,37,38,39,40,41,42,43,44,45,46,47,48,49,50,51,52,53,54,55,56,57,58,59,60,61,62,63,64,65,66,67,68,69,70,71,72,73,74,75,76,77,78,79,80,81,82,83,84,85,86,87,88,89,90,91,92,93,94,95,96,97,98,99,100,101,102,103,104,105,106,107,108,109,110,111,112,113,114,115,116,117,118,119,120,121,122,123,124,125,126,127,0,7,15,31,255,159,143,135,254,253,252,251,250,249,248,247,246,245,244,243,242,241,240,239,238,237,236,235,234,233,232,231,230,229,228,227,226,225,224,223,222,221,220,219,218,217,216,215,214,213,212,211,210,209,208,207,206,205,204,203,202,201,200,199,198,197,196,195,194,193,192,191,190,189,188,187,186,185,184,183,182,181,180,179,178,177,176,175,174,173,172,171,170,169,168,167,166,165,164,163,162,161,160,158,157,156,155,154,153,152,151,150,149,148,147,146,145,144,142,141,140,139,138,137,136,134,133,132,131,130,129,128},
	-{1,2,3,4,5,6,7,8,9,10,11,12,13,14,16,17,18,19,20,21,22,23,24,25,26,27,28,29,30,31,32,33,34,35,36,37,38,39,40,41,42,43,44,45,46,47,48,49,50,51,52,53,54,55,56,57,58,59,60,61,62,63,64,65,66,67,68,69,70,71,72,73,74,75,76,77,78,79,80,81,82,83,84,85,86,87,88,89,90,91,92,93,94,95,96,97,98,99,100,101,102,103,104,105,106,107,108,109,110,111,112,113,114,115,116,117,118,119,120,121,122,123,124,125,126,127,0,15,255,143,254,253,252,251,250,249,248,247,246,245,244,243,242,241,240,239,238,237,236,235,234,233,232,231,230,229,228,227,226,225,224,223,222,221,220,219,218,217,216,215,214,213,212,211,210,209,208,207,206,205,204,203,202,201,200,199,198,197,196,195,194,193,192,191,190,189,188,187,186,185,184,183,182,181,180,179,178,177,176,175,174,173,172,171,170,169,168,167,166,165,164,163,162,161,160,159,158,157,156,155,154,153,152,151,150,149,148,147,146,145,144,142,141,140,139,138,137,136,135,134,133,132,131,130,129,128},
	-{1,2,3,4,5,6,7,8,9,10,11,12,13,14,15,16,17,18,19,20,21,22,23,24,25,26,27,28,29,30,31,32,33,34,35,36,37,38,39,40,41,42,43,44,45,46,47,48,49,50,51,52,53,54,55,56,57,58,59,60,61,62,63,64,65,66,67,68,69,70,71,72,73,74,75,76,77,78,79,80,81,82,83,84,85,86,87,88,89,90,91,92,93,94,95,96,97,98,99,100,101,102,103,104,105,106,107,108,109,110,111,112,113,114,115,116,117,118,119,120,121,122,123,124,125,126,127,0,255,254,253,252,251,250,249,248,247,246,245,244,243,242,241,240,239,238,237,236,235,234,233,232,231,230,229,228,227,226,225,224,223,222,221,220,219,218,217,216,215,214,213,212,211,210,209,208,207,206,205,204,203,202,201,200,199,198,197,196,195,194,193,192,191,190,189,188,187,186,185,184,183,182,181,180,179,178,177,176,175,174,173,172,171,170,169,168,167,166,165,164,163,162,161,160,159,158,157,156,155,154,153,152,151,150,149,148,147,146,145,144,143,142,141,140,139,138,137,136,135,134,133,132,131,130,129,128}
	-};
	-
	-
	-#ifndef MIN_MACRO
	-#define MIN_MACRO(a,b) (((a)<(b))?(a):(b))
	-#endif
	-
	-#ifndef MAX_MACRO
	-#define MAX_MACRO(a,b) (((a)>(b))?(a):(b))
	-#endif
	-
	-#define POSITIVE_ULAW_ZERO 0xff
	-#define NEGATIVE_ULAW_ZERO 0x7f
	-
	-#define CAPMAXSCHAR(x) ((x > 127) ? 127 : x)
	-#define CAPMAXUCHAR(x) ((x > 255) ? 255 : x)
	-#define CAPMAXSHORT(x) ((x > 32767) ? 32767 : x)
	-#define CAPMAXUSHORT(x) ((x > 65535) ? 65535 : x)
	-
	-#define UNDEFINED_UINT -1
	-#define DEFAULT_BLOCK_SIZE 256
	-#define DEFAULT_V0NMEAN 0
	-#define DEFAULT_V2NMEAN 4
	-#define DEFAULT_MAXNLPC 0
	-#define DEFAULT_NCHAN 1
	-#define DEFAULT_NSKIP 0
	-#define DEFAULT_NDISCARD 0
	-#define NBITPERLONG 32
	-#define DEFAULT_MINSNR 256
	-#define DEFAULT_QUANTERROR 0
	-#define MINBITRATE 2.5
	-
	-#define MEAN_VERSION0 0
	-#define MEAN_VERSION2 4
	-
	-#define SHN_FN_DIFF0 0
	-#define SHN_FN_DIFF1 1
	-#define SHN_FN_DIFF2 2
	-#define SHN_FN_DIFF3 3
	-#define SHN_FN_QUIT 4
	-#define SHN_FN_BLOCKSIZE 5
	-#define SHN_FN_BITSHIFT 6
	-#define SHN_FN_QLPC 7
	-#define SHN_FN_ZERO 8
	-#define SHN_FN_VERBATIM 9
	-
	-#define SHN_TYPE_AU1 0
	-#define SHN_TYPE_S8 1
	-#define SHN_TYPE_U8 2
	-#define SHN_TYPE_S16HL 3
	-#define SHN_TYPE_U16HL 4
	-#define SHN_TYPE_S16LH 5
	-#define SHN_TYPE_U16LH 6
	-#define SHN_TYPE_ULAW 7
	-#define SHN_TYPE_AU2 8
	-#define SHN_TYPE_AU3 9
	-#define SHN_TYPE_ALAW 10
	-#define SHN_TYPE_RIFF_WAVE 11
	-#define SHN_TYPE_EOF 12
	-#define SHN_TYPE_GENERIC_ULAW 128
	-#define SHN_TYPE_GENERIC_ALAW 129
	-
	-#define SHN_FNSIZE 2
	-#define SHN_CHANNELSIZE 0
	-#define SHN_TYPESIZE 4
	-#define SHN_ULONGSIZE 2
	-#define SHN_NSKIPSIZE 1
	-#define SHN_LPCQSIZE 2
	-#define SHN_LPCQUANT 5
	-#define SHN_XBYTESIZE 7
	-#define SHN_VERBATIM_CKSIZE_SIZE 5
	-#define SHN_VERBATIM_BYTE_SIZE 8
	-#define SHN_ENERGYSIZE 3
	-#define SHN_BITSHIFTSIZE 2
	-
	-#define SHN_LPCQOFFSET_VER2 (1 << SHN_LPCQUANT)
	-
	-
	-#define SHN_MAGIC 0x676B6A61 /* looks like "ajkg" as chars. */
	-
	-#ifndef M_LN2
	-#define M_LN2 0.69314718055994530942
	-#endif
	-
	-#ifndef M_PI
	-#define M_PI 3.14159265358979323846
	-#endif
	-
	-
	-static int word_get(shn_t shn, SDL_RWops rw, Uint32 *word)
	-{
	- if (shn->nbyteget < 4)
	- {
	- shn->nbyteget += SDL_RWread(rw, shn->getbuf, 1, SHN_BUFSIZ);
	- BAIL_IF_MACRO(shn->nbyteget < 4, NULL, 0);
	- shn->getbufp = shn->getbuf;
	- } /* if */
	-
	- if (word != NULL)
	- {
	- *word = (((Sint32) shn->getbufp[0]) << 24) \|
	- (((Sint32) shn->getbufp[1]) << 16) \|
	- (((Sint32) shn->getbufp[2]) << 8) \|
	- (((Sint32) shn->getbufp[3]) );
	- } /* if */
	-
	- shn->getbufp += 4;
	- shn->nbyteget -= 4;
	-
	- return(1);
	-} /* word_get */
	-
	-
	-static int uvar_get(int nbin, shn_t shn, SDL_RWops rw, Sint32 *word)
	-{
	- Sint32 result;
	-
	- if (shn->nbitget == 0)
	- {
	- BAIL_IF_MACRO(!word_get(shn, rw, &shn->gbuffer), NULL, 0);
	- shn->nbitget = 32;
	- } /* if */
	-
	- for (result = 0; !(shn->gbuffer & (1L << --shn->nbitget)); result++)
	- {
	- if (shn->nbitget == 0)
	- {
	- BAIL_IF_MACRO(!word_get(shn, rw, &shn->gbuffer), NULL, 0);
	- shn->nbitget = 32;
	- } /* if */
	- } /* for */
	-
	- while (nbin != 0)
	- {
	- if (shn->nbitget >= nbin)
	- {
	- result = ( (result << nbin) \|
	- ((shn->gbuffer >> (shn->nbitget - nbin)) &
	- mask_table[nbin]) );
	- shn->nbitget -= nbin;
	- break;
	- } /* if */
	- else
	- {
	- result = (result << shn->nbitget) \|
	- (shn->gbuffer & mask_table[shn->nbitget]);
	- BAIL_IF_MACRO(!word_get(shn, rw, &shn->gbuffer), NULL, 0);
	- nbin -= shn->nbitget;
	- shn->nbitget = 32;
	- } /* else */
	- } /* while */
	-
	- if (word != NULL)
	- *word = result;
	-
	- return(1);
	-} /* uvar_get */
	-
	-
	-static int var_get(int nbin, shn_t shn, SDL_RWops rw, Sint32 *word)
	-{
	- BAIL_IF_MACRO(!uvar_get(nbin + 1, shn, rw, word), NULL, 0);
	-
	- if ((*word) & 1)
	- word = (Sint32) ~((word) >> 1);
	- else
	- word = (Sint32) ((word) >> 1);
	-
	- return(1);
	-} /* var_get */
	-
	-
	-static int ulong_get(shn_t shn, SDL_RWops rw, Sint32 *word)
	-{
	- Sint32 nbit;
	- Sint32 retval;
	- BAIL_IF_MACRO(!uvar_get(SHN_ULONGSIZE, shn, rw, &nbit), NULL, 0);
	- BAIL_IF_MACRO(!uvar_get(nbit, shn, rw, &retval), NULL, 0);
	-
	- if (word != NULL)
	- *word = retval;
	-
	- return(1);
	-} /* ulong_get */
	-
	-
	-static __inline__ int uint_get(int nbit, shn_t shn, SDL_RWops rw, Sint32 *w)
	-{
	- return((shn->version == 0) ?
	- uvar_get(nbit, shn, rw, w) :
	- ulong_get(shn, rw, w));
	-} /* uint_get */
	-
	-
	-static int SHN_init(void)
	-{
	- return(1); /* initialization always successful. */
	-} /* SHN_init */
	-
	-
	-static void SHN_quit(void)
	-{
	- /* it's a no-op. */
	-} /* SHN_quit */
	-
	-
	-/*
	- * Look through the whole file for a SHN magic number. This is costly, so
	- * it should only be done if the user SWEARS they have a Shorten stream...
	- */
	-static __inline__ int extended_shn_magic_search(Sound_Sample *sample)
	-{
	- SDL_RWops rw = ((Sound_SampleInternal ) sample->opaque)->rw;
	- Uint32 word = 0;
	- Uint8 ch;
	-
	- while (1)
	- {
	- BAIL_IF_MACRO(SDL_RWread(rw, &ch, sizeof (ch), 1) != 1, NULL, -1);
	- word = ((word << 8) & 0xFFFFFF00) \| ch;
	- if (SDL_SwapBE32(word) == SHN_MAGIC)
	- {
	- BAIL_IF_MACRO(SDL_RWread(rw, &ch, sizeof (ch), 1) != 1, NULL, -1);
	- return((int) ch);
	- } /* if */
	- } /* while */
	-
	- return((int) ch);
	-} /* extended_shn_magic_search */
	-
	-
	-/* look for the magic number in the RWops and see what kind of file this is. */
	-static __inline__ int determine_shn_version(Sound_Sample *sample,
	- const char *ext)
	-{
	- SDL_RWops rw = ((Sound_SampleInternal ) sample->opaque)->rw;
	- Uint32 magic;
	- Uint8 ch;
	-
	- /*
	- * Apparently the magic number can start at any byte offset in the file,
	- * and we should just discard prior data, but I'm going to restrict it
	- * to offset zero for now, so we don't chug down every file that might
	- * happen to pass through here. If the extension is explicitly "SHN", we
	- * check the whole stream, though.
	- */
	-
	- if (__Sound_strcasecmp(ext, "shn") == 0)
	- return(extended_shn_magic_search(sample));
	-
	- BAIL_IF_MACRO(SDL_RWread(rw, &magic, sizeof (magic), 1) != 1, NULL, -1);
	- BAIL_IF_MACRO(SDL_SwapLE32(magic) != SHN_MAGIC, "SHN: Not a SHN file", -1);
	- BAIL_IF_MACRO(SDL_RWread(rw, &ch, sizeof (ch), 1) != 1, NULL, -1);
	- BAIL_IF_MACRO(ch > 3, "SHN: Unsupported file version", -1);
	-
	- return((int) ch);
	-} /* determine_shn_version */
	-
	-
	-static void init_shn_offset(Sint32 **offset, int nchan, int nblock, int ftype)
	-{
	- Sint32 mean = 0;
	- int chan;
	-
	- switch (ftype)
	- {
	- case SHN_TYPE_AU1:
	- case SHN_TYPE_S8:
	- case SHN_TYPE_S16HL:
	- case SHN_TYPE_S16LH:
	- case SHN_TYPE_ULAW:
	- case SHN_TYPE_AU2:
	- case SHN_TYPE_AU3:
	- case SHN_TYPE_ALAW:
	- mean = 0;
	- break;
	- case SHN_TYPE_U8:
	- mean = 0x80;
	- break;
	- case SHN_TYPE_U16HL:
	- case SHN_TYPE_U16LH:
	- mean = 0x8000;
	- break;
	- default:
	- __Sound_SetError("SHN: unknown file type");
	- return;
	- } /* switch */
	-
	- for(chan = 0; chan < nchan; chan++)
	- {
	- int i;
	- for(i = 0; i < nblock; i++)
	- offset[chan][i] = mean;
	- } /* for */
	-} /* init_shn_offset */
	-
	-
	-static __inline__ Uint16 cvt_shnftype_to_sdlfmt(Sint16 shntype)
	-{
	- switch (shntype)
	- {
	- case SHN_TYPE_S8:
	- return(AUDIO_S8);
	-
	- case SHN_TYPE_ALAW:
	- case SHN_TYPE_ULAW:
	- case SHN_TYPE_AU1:
	- case SHN_TYPE_AU2:
	- case SHN_TYPE_AU3:
	- case SHN_TYPE_U8:
	- return(AUDIO_U8);
	-
	- case SHN_TYPE_S16HL:
	- return(AUDIO_S16MSB);
	-
	- case SHN_TYPE_S16LH:
	- return(AUDIO_S16LSB);
	-
	- case SHN_TYPE_U16HL:
	- return(AUDIO_U16MSB);
	-
	- case SHN_TYPE_U16LH:
	- return(AUDIO_U16LSB);
	- } /* switch */
	-
	- return(0);
	-} /* cvt_shnftype_to_sdlfmt */
	-
	-
	-static __inline__ int skip_bits(shn_t shn, SDL_RWops rw)
	-{
	- int i;
	- Sint32 skip;
	- Sint32 trash;
	-
	- BAIL_IF_MACRO(!uint_get(SHN_NSKIPSIZE, shn, rw, &skip), NULL, 0);
	- for(i = 0; i < skip; i++)
	- {
	- BAIL_IF_MACRO(!uint_get(SHN_XBYTESIZE, shn, rw, &trash), NULL, 0);
	- } /* for */
	-
	- return(1);
	-} /* skip_bits */
	-
	-
	-static Sint32 **shn_long2d(Uint32 n0, Uint32 n1)
	-{
	- Sint32 **array0;
	- Uint32 size = (n0 * sizeof (Sint32 )) + (n0 n1 * sizeof (Sint32));
	-
	- array0 = (Sint32 **) malloc(size);
	- if (array0 != NULL)
	- {
	- int i;
	- Sint32 array1 = (Sint32 ) (array0 + n0);
	- for(i = 0; i < n0; i++)
	- array0[i] = array1 + (i * n1);
	- } /* if */
	-
	- return(array0);
	-} /* shn_long2d */
	-
	-#define riffID 0x46464952 /* "RIFF", in ascii. */
	-#define waveID 0x45564157 /* "WAVE", in ascii. */
	-#define fmtID 0x20746D66 /* "fmt ", in ascii. */
	-#define dataID 0x61746164 /* "data", in ascii. */
	-
	-static int verb_ReadLE32(shn_t shn, SDL_RWops rw, Uint32 *word)
	-{
	- int i;
	- Uint8 chars[4];
	- Sint32 byte;
	-
	- for (i = 0; i < 4; i++)
	- {
	- if (!uvar_get(SHN_VERBATIM_BYTE_SIZE, shn, rw, &byte))
	- return(0);
	- chars[i] = (Uint8) byte;
	- } /* for */
	-
	- memcpy(word, chars, sizeof (*word));
	- word = SDL_SwapLE32(word);
	-
	- return(1);
	-} /* verb_ReadLE32 */
	-
	-
	-static int verb_ReadLE16(shn_t shn, SDL_RWops rw, Uint16 *word)
	-{
	- int i;
	- Uint8 chars[2];
	- Sint32 byte;
	-
	- for (i = 0; i < 2; i++)
	- {
	- if (!uvar_get(SHN_VERBATIM_BYTE_SIZE, shn, rw, &byte))
	- return(0);
	- chars[i] = (Uint8) byte;
	- } /* for */
	-
	- memcpy(word, chars, sizeof (*word));
	- word = SDL_SwapLE16(word);
	-
	- return(1);
	-} /* verb_ReadLE16 */
	-
	-
	-static __inline__ int parse_riff_header(shn_t shn, Sound_Sample sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- SDL_RWops *rw = internal->rw;
	- Uint16 u16;
	- Uint32 u32;
	- Sint32 cklen;
	-
	- BAIL_IF_MACRO(!uvar_get(SHN_VERBATIM_CKSIZE_SIZE, shn, rw, &cklen), NULL, 0);
	-
	- BAIL_IF_MACRO(!verb_ReadLE32(shn, rw, &u32), NULL, 0); /* RIFF header */
	- BAIL_IF_MACRO(u32 != riffID, "SHN: No RIFF header.", 0);
	- BAIL_IF_MACRO(!verb_ReadLE32(shn, rw, &u32), NULL, 0); /* length */
	-
	- BAIL_IF_MACRO(!verb_ReadLE32(shn, rw, &u32), NULL, 0); /* WAVE header */
	- BAIL_IF_MACRO(u32 != waveID, "SHN: No WAVE header.", 0);
	-
	- BAIL_IF_MACRO(!verb_ReadLE32(shn, rw, &u32), NULL, 0); /* 'fmt ' header */
	- BAIL_IF_MACRO(u32 != fmtID, "SHN: No 'fmt ' header.", 0);
	-
	- BAIL_IF_MACRO(!verb_ReadLE32(shn, rw, &u32), NULL, 0); /* chunksize */
	- BAIL_IF_MACRO(!verb_ReadLE16(shn, rw, &u16), NULL, 0); /* format */
	- BAIL_IF_MACRO(!verb_ReadLE16(shn, rw, &u16), NULL, 0); /* channels */
	- sample->actual.channels = u16;
	- BAIL_IF_MACRO(!verb_ReadLE32(shn, rw, &u32), NULL, 0); /* sample rate */
	- sample->actual.rate = u32;
	-
	- BAIL_IF_MACRO(!verb_ReadLE32(shn, rw, &u32), NULL, 0); /* bytespersec */
	- BAIL_IF_MACRO(!verb_ReadLE16(shn, rw, &u16), NULL, 0); /* blockalign */
	- BAIL_IF_MACRO(!verb_ReadLE16(shn, rw, &u16), NULL, 0); /* bitspersample */
	-
	- BAIL_IF_MACRO(!verb_ReadLE32(shn, rw, &u32), NULL, 0); /* 'data' header */
	- BAIL_IF_MACRO(u32 != dataID, "SHN: No 'data' header.", 0);
	- BAIL_IF_MACRO(!verb_ReadLE32(shn, rw, &u32), NULL, 0); /* chunksize */
	-
	- return(1);
	-} /* parse_riff_header */
	-
	-
	-static int SHN_open(Sound_Sample sample, const char ext)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- SDL_RWops *rw = internal->rw;
	- shn_t _shn;
	- shn_t shn = &_shn; / malloc and copy later. */
	- Sint32 cmd;
	- Sint32 chan;
	-
	- memset(shn, '\0', sizeof (shn_t));
	- shn->getbufp = shn->getbuf = (Uint8 *) malloc(SHN_BUFSIZ);
	- shn->datatype = SHN_TYPE_EOF;
	- shn->nchan = DEFAULT_NCHAN;
	- shn->blocksize = DEFAULT_BLOCK_SIZE;
	- shn->maxnlpc = DEFAULT_MAXNLPC;
	- shn->nmean = UNDEFINED_UINT;
	- shn->version = determine_shn_version(sample, ext);
	-
	- if (shn->version == -1) goto shn_open_puke;
	- if (!uint_get(SHN_TYPESIZE, shn, rw, &shn->datatype)) goto shn_open_puke;
	- if (!uint_get(SHN_CHANNELSIZE, shn, rw, &shn->nchan)) goto shn_open_puke;
	-
	- sample->actual.format = cvt_shnftype_to_sdlfmt(shn->datatype);
	- if (sample->actual.format == 0)
	- {
	- SDL_SetError(ERR_UNSUPPORTED_FORMAT);
	- goto shn_open_puke;
	- } /* if */
	-
	- if (shn->version > 0)
	- {
	- int rc = uint_get((int) (log((double) DEFAULT_BLOCK_SIZE) / M_LN2),
	- shn, rw, &shn->blocksize);
	- if (!rc) goto shn_open_puke;;
	- if (!uint_get(SHN_LPCQSIZE, shn, rw, &shn->maxnlpc)) goto shn_open_puke;
	- if (!uint_get(0, shn, rw, &shn->nmean)) goto shn_open_puke;
	- if (!skip_bits(shn, rw)) goto shn_open_puke;
	- } /* else */
	-
	- shn->nwrap = (shn->maxnlpc > 3) ? shn->maxnlpc : 3;
	-
	- /* grab some space for the input buffer */
	- shn->buffer = shn_long2d((Uint32) shn->nchan, shn->blocksize + shn->nwrap);
	- shn->offset = shn_long2d((Uint32) shn->nchan, MAX_MACRO(1, shn->nmean));
	-
	- for (chan = 0; chan < shn->nchan; chan++)
	- {
	- int i;
	- for(i = 0; i < shn->nwrap; i++)
	- shn->buffer[chan][i] = 0;
	- shn->buffer[chan] += shn->nwrap;
	- } /* for */
	-
	- if (shn->maxnlpc > 0)
	- {
	- shn->qlpc = (int ) malloc((Uint32) (shn->maxnlpc sizeof (Sint32)));
	- if (shn->qlpc == NULL)
	- {
	- __Sound_SetError(ERR_OUT_OF_MEMORY);
	- goto shn_open_puke;
	- } /* if */
	- } /* if */
	-
	- if (shn->version > 1)
	- shn->lpcqoffset = SHN_LPCQOFFSET_VER2;
	-
	- init_shn_offset(shn->offset, shn->nchan,
	- MAX_MACRO(1, shn->nmean), shn->datatype);
	-
	- if ( (!uvar_get(SHN_FNSIZE, shn, rw, &cmd)) \|\|
	- (cmd != SHN_FN_VERBATIM) \|\|
	- (!parse_riff_header(shn, sample)) )
	- {
	- if (cmd != SHN_FN_VERBATIM) /* the other conditions set error state */
	- __Sound_SetError("SHN: Expected VERBATIM function");
	-
	- goto shn_open_puke;
	- return(0);
	- } /* if */
	-
	- shn->start_pos = SDL_RWtell(rw);
	-
	- shn = (shn_t *) malloc(sizeof (shn_t));
	- if (shn == NULL)
	- {
	- __Sound_SetError(ERR_OUT_OF_MEMORY);
	- goto shn_open_puke;
	- } /* if */
	-
	- memcpy(shn, &_shn, sizeof (shn_t));
	- internal->decoder_private = shn;
	-
	- SNDDBG(("SHN: Accepting data stream.\n"));
	- sample->flags = SOUND_SAMPLEFLAG_NONE;
	- return(1); /* we'll handle this data. */
	-
	-shn_open_puke:
	- if (_shn.getbuf)
	- free(_shn.getbuf);
	- if (_shn.buffer != NULL)
	- free(_shn.buffer);
	- if (_shn.offset != NULL)
	- free(_shn.offset);
	- if (_shn.qlpc != NULL)
	- free(_shn.qlpc);
	-
	- return(0);
	-} /* SHN_open */
	-
	-
	-static void fix_bitshift(Sint32 *buffer, int nitem, int bitshift, int ftype)
	-{
	- int i;
	-
	- if (ftype == SHN_TYPE_AU1)
	- {
	- for (i = 0; i < nitem; i++)
	- buffer[i] = ulaw_outward[bitshift][buffer[i] + 128];
	- } /* if */
	- else if (ftype == SHN_TYPE_AU2)
	- {
	- for(i = 0; i < nitem; i++)
	- {
	- if (buffer[i] >= 0)
	- buffer[i] = ulaw_outward[bitshift][buffer[i] + 128];
	- else if (buffer[i] == -1)
	- buffer[i] = NEGATIVE_ULAW_ZERO;
	- else
	- buffer[i] = ulaw_outward[bitshift][buffer[i] + 129];
	- } /* for */
	- } /* else if */
	- else
	- {
	- if (bitshift != 0)
	- {
	- for(i = 0; i < nitem; i++)
	- buffer[i] <<= bitshift;
	- } /* if */
	- } /* else */
	-} /* fix_bitshift */
	-
	-
	-static void SHN_close(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- shn_t shn = (shn_t ) internal->decoder_private;
	-
	- if (shn->qlpc != NULL)
	- free(shn->qlpc);
	-
	- if (shn->backBuffer != NULL)
	- free(shn->backBuffer);
	-
	- if (shn->offset != NULL)
	- free(shn->offset);
	-
	- if (shn->buffer != NULL)
	- free(shn->buffer);
	-
	- if (shn->getbuf != NULL)
	- free(shn->getbuf);
	-
	- free(shn);
	-} /* SHN_close */
	-
	-
	-/* xLaw conversions... */
	-
	-/* adapted by ajr for int input */
	-static Uint8 Slinear2ulaw(int sample)
	-{
	-/*
	-** This routine converts from linear to ulaw.
	-**
	-** Craig Reese: IDA/Supercomputing Research Center
	-** Joe Campbell: Department of Defense
	-** 29 September 1989
	-**
	-** References:
	-** 1) CCITT Recommendation G.711 (very difficult to follow)
	-** 2) "A New Digital Technique for Implementation of Any
	-** Continuous PCM Companding Law," Villeret, Michel,
	-** et al. 1973 IEEE Int. Conf. on Communications, Vol 1,
	-** 1973, pg. 11.12-11.17
	-** 3) MIL-STD-188-113,"Interoperability and Performance Standards
	-** for Analog-to_Digital Conversion Techniques,"
	-** 17 February 1987
	-**
	-** Input: Signed 16 bit linear sample
	-** Output: 8 bit ulaw sample
	-*/
	-
	-#define BIAS 0x84 /* define the add-in bias for 16 bit samples */
	-#define CLIP 32635
	-
	- int sign, exponent, mantissa;
	- Uint8 ulawbyte;
	- static const int exp_lut[256] = {0,0,1,1,2,2,2,2,3,3,3,3,3,3,3,3,
	- 4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,
	- 5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,
	- 5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,
	- 6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,
	- 6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,
	- 6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,
	- 6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,
	- 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
	- 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
	- 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
	- 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
	- 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
	- 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
	- 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
	- 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7};
	-
	- /* Get the sample into sign-magnitude. */
	- if (sample >= 0)
	- sign = 0;
	- else
	- {
	- sign = 0x80;
	- sample = -sample;
	- } /* else */
	-
	- /* clip the magnitude */
	- if (sample > CLIP)
	- sample = CLIP;
	-
	- /* Convert from 16 bit linear to ulaw. */
	- sample = sample + BIAS;
	- exponent = exp_lut[( sample >> 7 ) & 0xFF];
	- mantissa = (sample >> (exponent + 3)) & 0x0F;
	- ulawbyte = ~(sign \| (exponent << 4) \| mantissa);
	-
	- return(ulawbyte);
	-} /* Slinear2ulaw */
	-
	-
	-/* this is derived from the Sun code - it is a bit simpler and has int input */
	-#define QUANT_MASK (0xf) /* Quantization field mask. */
	-#define NSEGS (8) /* Number of A-law segments. */
	-#define SEG_SHIFT (4) /* Left shift for segment number. */
	-
	-
	-static Uint8 Slinear2alaw(Sint32 linear)
	-{
	- int seg;
	- Uint8 aval, mask;
	- static const Sint32 seg_aend[NSEGS] =
	- {
	- 0x1f,0x3f,0x7f,0xff,0x1ff,0x3ff,0x7ff,0xfff
	- };
	-
	- linear >>= 3;
	- if(linear >= 0)
	- mask = 0xd5; /* sign (7th) bit = 1 */
	- else
	- {
	- mask = 0x55; /* sign bit = 0 */
	- linear = -linear - 1;
	- } /* else */
	-
	- /* Convert the scaled magnitude to segment number. */
	- for (seg = 0; (seg < NSEGS) && (linear > seg_aend[seg]); seg++);
	-
	- /* Combine the sign, segment, and quantization bits. */
	- if (seg >= NSEGS) /* out of range, return maximum value. */
	- return((Uint8) (0x7F ^ mask));
	-
	- aval = (Uint8) seg << SEG_SHIFT;
	- if (seg < 2)
	- aval \|= (linear >> 1) & QUANT_MASK;
	- else
	- aval \|= (linear >> seg) & QUANT_MASK;
	-
	- return (aval ^ mask);
	-} /* Slinear2alaw */
	-
	-
	-/* convert from signed ints to a given type and write */
	-static Uint32 put_to_buffers(Sound_Sample *sample, Uint32 bw)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- shn_t shn = (shn_t ) internal->decoder_private;
	- int i, chan;
	- Sint32 *data0 = shn->buffer[0];
	- Sint32 nitem = shn->blocksize;
	- int datasize = ((sample->actual.format & 0xFF) / 8);
	- Uint32 bsiz = shn->nchan * nitem * datasize;
	-
	- assert(shn->backBufLeft == 0);
	-
	- if (shn->backBufferSize < bsiz)
	- {
	- void *rc = realloc(shn->backBuffer, bsiz);
	- if (rc == NULL)
	- {
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- BAIL_MACRO(ERR_OUT_OF_MEMORY, 0);
	- } /* if */
	- shn->backBuffer = (Uint8 *) rc;
	- shn->backBufferSize = bsiz;
	- } /* if */
	-
	- switch (shn->datatype)
	- {
	- case SHN_TYPE_AU1: /* leave the conversion to fix_bitshift() */
	- case SHN_TYPE_AU2:
	- {
	- Uint8 *writebufp = shn->backBuffer;
	- if (shn->nchan == 1)
	- {
	- for (i = 0; i < nitem; i++)
	- *writebufp++ = data0[i];
	- } /* if */
	- else
	- {
	- for (i = 0; i < nitem; i++)
	- {
	- for (chan = 0; chan < shn->nchan; chan++)
	- *writebufp++ = shn->buffer[chan][i];
	- } /* for */
	- } /* else */
	- } /* case */
	- break;
	-
	- case SHN_TYPE_U8:
	- {
	- Uint8 *writebufp = shn->backBuffer;
	- if (shn->nchan == 1)
	- {
	- for (i = 0; i < nitem; i++)
	- *writebufp++ = CAPMAXUCHAR(data0[i]);
	- } /* if */
	- else
	- {
	- for (i = 0; i < nitem; i++)
	- {
	- for (chan = 0; chan < shn->nchan; chan++)
	- *writebufp++ = CAPMAXUCHAR(shn->buffer[chan][i]);
	- } /* for */
	- } /* else */
	- } /* case */
	- break;
	-
	- case SHN_TYPE_S8:
	- {
	- Sint8 writebufp = (Sint8 ) shn->backBuffer;
	- if (shn->nchan == 1)
	- {
	- for(i = 0; i < nitem; i++)
	- *writebufp++ = CAPMAXSCHAR(data0[i]);
	- } /* if */
	- else
	- {
	- for(i = 0; i < nitem; i++)
	- {
	- for(chan = 0; chan < shn->nchan; chan++)
	- *writebufp++ = CAPMAXSCHAR(shn->buffer[chan][i]);
	- } /* for */
	- } /* else */
	- } /* case */
	- break;
	-
	- case SHN_TYPE_S16HL:
	- case SHN_TYPE_S16LH:
	- {
	- Sint16 writebufp = (Sint16 ) shn->backBuffer;
	- if (shn->nchan == 1)
	- {
	- for (i = 0; i < nitem; i++)
	- *writebufp++ = CAPMAXSHORT(data0[i]);
	- } /* if */
	- else
	- {
	- for (i = 0; i < nitem; i++)
	- {
	- for (chan = 0; chan < shn->nchan; chan++)
	- *writebufp++ = CAPMAXSHORT(shn->buffer[chan][i]);
	- } /* for */
	- } /* else */
	- } /* case */
	- break;
	-
	- case SHN_TYPE_U16HL:
	- case SHN_TYPE_U16LH:
	- {
	- Uint16 writebufp = (Uint16 ) shn->backBuffer;
	- if (shn->nchan == 1)
	- {
	- for (i = 0; i < nitem; i++)
	- *writebufp++ = CAPMAXUSHORT(data0[i]);
	- } /* if */
	- else
	- {
	- for (i = 0; i < nitem; i++)
	- {
	- for (chan = 0; chan < shn->nchan; chan++)
	- *writebufp++ = CAPMAXUSHORT(shn->buffer[chan][i]);
	- } /* for */
	- } /* else */
	- } /* case */
	- break;
	-
	- case SHN_TYPE_ULAW:
	- {
	- Uint8 *writebufp = shn->backBuffer;
	- if (shn->nchan == 1)
	- {
	- for(i = 0; i < nitem; i++)
	- *writebufp++ = Slinear2ulaw(CAPMAXSHORT((data0[i] << 3)));
	- } /* if */
	- else
	- {
	- for(i = 0; i < nitem; i++)
	- {
	- for(chan = 0; chan < shn->nchan; chan++)
	- *writebufp++ = Slinear2ulaw(CAPMAXSHORT((shn->buffer[chan][i] << 3)));
	- } /* for */
	- } /* else */
	- } /* case */
	- break;
	-
	- case SHN_TYPE_AU3:
	- {
	- Uint8 *writebufp = shn->backBuffer;
	- if (shn->nchan == 1)
	- {
	- for (i = 0; i < nitem; i++)
	- if(data0[i] < 0)
	- *writebufp++ = (127 - data0[i]) ^ 0xd5;
	- else
	- *writebufp++ = (data0[i] + 128) ^ 0x55;
	- } /* if */
	- else
	- {
	- for (i = 0; i < nitem; i++)
	- {
	- for (chan = 0; chan < shn->nchan; chan++)
	- {
	- if (shn->buffer[chan][i] < 0)
	- *writebufp++ = (127 - shn->buffer[chan][i]) ^ 0xd5;
	- else
	- *writebufp++ = (shn->buffer[chan][i] + 128) ^ 0x55;
	- } /* for */
	- } /* for */
	- } /* else */
	- } /* case */
	- break;
	-
	- case SHN_TYPE_ALAW:
	- {
	- Uint8 *writebufp = shn->backBuffer;
	- if (shn->nchan == 1)
	- {
	- for (i = 0; i < nitem; i++)
	- *writebufp++ = Slinear2alaw(CAPMAXSHORT((data0[i] << 3)));
	- } /* if */
	- else
	- {
	- for (i = 0; i < nitem; i++)
	- {
	- for(chan = 0; chan < shn->nchan; chan++)
	- *writebufp++ = Slinear2alaw(CAPMAXSHORT((shn->buffer[chan][i] << 3)));
	- } /* for */
	- }/* else */
	- } /* case */
	- break;
	- } /* switch */
	-
	- i = MIN_MACRO(internal->buffer_size - bw, bsiz);
	- memcpy((char *)internal->buffer + bw, shn->backBuffer, i);
	- shn->backBufLeft = bsiz - i;
	- memcpy(shn->backBuffer, shn->backBuffer + i, shn->backBufLeft);
	- return(i);
	-} /* put_to_buffers */
	-
	-
	-#define ROUNDEDSHIFTDOWN(x, n) (((n) == 0) ? (x) : ((x) >> ((n) - 1)) >> 1)
	-
	-static Uint32 SHN_read(Sound_Sample *sample)
	-{
	- Uint32 retval = 0;
	- Sint32 chan = 0;
	- Uint32 cpyBytes = 0;
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- SDL_RWops *rw = internal->rw;
	- shn_t shn = (shn_t ) internal->decoder_private;
	- Sint32 cmd;
	-
	- assert(shn->backBufLeft >= 0);
	-
	- /* see if there are leftovers to copy... */
	- if (shn->backBufLeft > 0)
	- {
	- retval = MIN_MACRO(shn->backBufLeft, internal->buffer_size);
	- memcpy(internal->buffer, shn->backBuffer, retval);
	- shn->backBufLeft -= retval;
	- memcpy(shn->backBuffer, shn->backBuffer + retval, shn->backBufLeft);
	- } /* if */
	-
	- assert((shn->backBufLeft == 0) \|\| (retval == internal->buffer_size));
	-
	- /* get commands from file and execute them */
	- while (retval < internal->buffer_size)
	- {
	- if (!uvar_get(SHN_FNSIZE, shn, rw, &cmd))
	- {
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- return(retval);
	- } /* if */
	-
	- if (cmd == SHN_FN_QUIT)
	- {
	- sample->flags \|= SOUND_SAMPLEFLAG_EOF;
	- return(retval);
	- } /* if */
	-
	- switch(cmd)
	- {
	- case SHN_FN_ZERO:
	- case SHN_FN_DIFF0:
	- case SHN_FN_DIFF1:
	- case SHN_FN_DIFF2:
	- case SHN_FN_DIFF3:
	- case SHN_FN_QLPC:
	- {
	- Sint32 i;
	- Sint32 coffset, *cbuffer = shn->buffer[chan];
	- Sint32 resn = 0, nlpc, j;
	-
	- if (cmd != SHN_FN_ZERO)
	- {
	- if (!uvar_get(SHN_ENERGYSIZE, shn, rw, &resn))
	- {
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- return(retval);
	- } /* if */
	-
	- /* version 0 differed in definition of var_get */
	- if (shn->version == 0)
	- resn--;
	- } /* if */
	-
	- /* find mean offset : N.B. this code duplicated */
	- if (shn->nmean == 0)
	- coffset = shn->offset[chan][0];
	- else
	- {
	- Sint32 sum = (shn->version < 2) ? 0 : shn->nmean / 2;
	- for (i = 0; i < shn->nmean; i++)
	- sum += shn->offset[chan][i];
	-
	- if (shn->version < 2)
	- coffset = sum / shn->nmean;
	- else
	- coffset = ROUNDEDSHIFTDOWN(sum / shn->nmean, shn->bitshift);
	- } /* else */
	-
	- switch (cmd)
	- {
	- case SHN_FN_ZERO:
	- for (i = 0; i < shn->blocksize; i++)
	- cbuffer[i] = 0;
	- break;
	-
	- case SHN_FN_DIFF0:
	- for(i = 0; i < shn->blocksize; i++)
	- {
	- if (!var_get(resn, shn, rw, &cbuffer[i]))
	- {
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- return(retval);
	- } /* if */
	- cbuffer[i] += coffset;
	- } /* for */
	- break;
	-
	- case SHN_FN_DIFF1:
	- for(i = 0; i < shn->blocksize; i++)
	- {
	- if (!var_get(resn, shn, rw, &cbuffer[i]))
	- {
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- return(retval);
	- } /* if */
	- cbuffer[i] += cbuffer[i - 1];
	- } /* for */
	- break;
	-
	- case SHN_FN_DIFF2:
	- for (i = 0; i < shn->blocksize; i++)
	- {
	- if (!var_get(resn, shn, rw, &cbuffer[i]))
	- {
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- return(retval);
	- } /* if */
	- cbuffer[i] += (2 * cbuffer[i-1] - cbuffer[i-2]);
	- } /* for */
	- break;
	-
	- case SHN_FN_DIFF3:
	- for (i = 0; i < shn->blocksize; i++)
	- {
	- if (!var_get(resn, shn, rw, &cbuffer[i]))
	- {
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- return(retval);
	- } /* if */
	- cbuffer[i] += 3 * (cbuffer[i - 1] - cbuffer[i - 2]) + cbuffer[i - 3];
	- } /* for */
	- break;
	-
	- case SHN_FN_QLPC:
	- if (!uvar_get(SHN_LPCQSIZE, shn, rw, &nlpc))
	- {
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- return(retval);
	- } /* if */
	-
	- for(i = 0; i < nlpc; i++)
	- {
	- if (!var_get(SHN_LPCQUANT, shn, rw, &shn->qlpc[i]))
	- {
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- return(retval);
	- } /* if */
	- } /* for */
	-
	- for(i = 0; i < nlpc; i++)
	- cbuffer[i - nlpc] -= coffset;
	-
	- for(i = 0; i < shn->blocksize; i++)
	- {
	- Sint32 sum = shn->lpcqoffset;
	-
	- for(j = 0; j < nlpc; j++)
	- sum += shn->qlpc[j] * cbuffer[i - j - 1];
	-
	- if (!var_get(resn, shn, rw, &cbuffer[i]))
	- {
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- return(retval);
	- } /* if */
	- cbuffer[i] += (sum >> SHN_LPCQUANT);
	- } /* for */
	-
	- if (coffset != 0)
	- {
	- for(i = 0; i < shn->blocksize; i++)
	- cbuffer[i] += coffset;
	- } /* if */
	-
	- break;
	- } /* switch */
	-
	- /* store mean value if appropriate : N.B. Duplicated code */
	- if (shn->nmean > 0)
	- {
	- Sint32 sum = (shn->version < 2) ? 0 : shn->blocksize / 2;
	- for (i = 0; i < shn->blocksize; i++)
	- sum += cbuffer[i];
	-
	- for(i = 1; i < shn->nmean; i++)
	- shn->offset[chan][i - 1] = shn->offset[chan][i];
	-
	- if (shn->version < 2)
	- shn->offset[chan][shn->nmean - 1] = sum / shn->blocksize;
	- else
	- shn->offset[chan][shn->nmean - 1] = (sum / shn->blocksize) << shn->bitshift;
	- } /* if */
	-
	- /* do the wrap */
	- for(i = -shn->nwrap; i < 0; i++)
	- cbuffer[i] = cbuffer[i + shn->blocksize];
	-
	- fix_bitshift(cbuffer, shn->blocksize, shn->bitshift, shn->datatype);
	-
	- if (chan == shn->nchan - 1)
	- {
	- retval += put_to_buffers(sample, retval);
	- if (sample->flags & SOUND_SAMPLEFLAG_ERROR)
	- return(retval);
	- } /* if */
	-
	- chan = (chan + 1) % shn->nchan;
	- break;
	- } /* case */
	-
	- case SHN_FN_BLOCKSIZE:
	- if (!uint_get((int) (log((double) shn->blocksize) / M_LN2),
	- shn, rw, &shn->blocksize))
	- {
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- return(retval);
	- } /* if */
	- break;
	-
	- case SHN_FN_BITSHIFT:
	- if (!uvar_get(SHN_BITSHIFTSIZE, shn, rw, &shn->bitshift))
	- {
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- return(retval);
	- } /* if */
	- break;
	-
	- case SHN_FN_VERBATIM:
	- default:
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- BAIL_MACRO("SHN: Unhandled function.", retval);
	- } /* switch */
	- } /* while */
	-
	- return(retval);
	-} /* SHN_read */
	-
	-
	-static int SHN_rewind(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- shn_t shn = (shn_t ) internal->decoder_private;
	-
	-#if 0
	- int rc = SDL_RWseek(internal->rw, shn->start_pos, SEEK_SET);
	- BAIL_IF_MACRO(rc != shn->start_pos, ERR_IO_ERROR, 0);
	- /* !!! FIXME: set state. */
	- return(1);
	-#else
	- /*
	- * !!! FIXME: This is really unacceptable; state should be reset and
	- * !!! FIXME: the RWops should be pointed to the start of the data
	- * !!! FIXME: to decode. The below kludge adds unneeded overhead and
	- * !!! FIXME: risk of failure.
	- */
	- BAIL_IF_MACRO(SDL_RWseek(internal->rw, 0, SEEK_SET) != 0, ERR_IO_ERROR, 0);
	- SHN_close(sample);
	- return(SHN_open(sample, "SHN"));
	-#endif
	-} /* SHN_rewind */
	-
	-
	-static int SHN_seek(Sound_Sample *sample, Uint32 ms)
	-{
	- /*
	- * (This CAN be done for SHNs that have a seek table at the end of the
	- * stream, btw.)
	- */
	- BAIL_MACRO("SHN: Seeking not implemented", 0);
	-} /* SHN_seek */
	-
	-
	-#endif /* defined SOUND_SUPPORTS_SHN */
	-
	-/* end of shn.c ... */
	-
	diff --git a/util/sdl/sound/decoders/smpeg.c b/util/sdl/sound/decoders/smpeg.c
	deleted file mode 100644
	index f4958977..00000000
	--- a/util/sdl/sound/decoders/smpeg.c
	+++ /dev/null
	@@ -1,310 +0,0 @@
	-/*
	- * SDL_sound -- An abstract sound format decoding API.
	- * Copyright (C) 2001 Ryan C. Gordon.
	- *
	- * This library is free software; you can redistribute it and/or
	- * modify it under the terms of the GNU Lesser General Public
	- * License as published by the Free Software Foundation; either
	- * version 2.1 of the License, or (at your option) any later version.
	- *
	- * This library is distributed in the hope that it will be useful,
	- * but WITHOUT ANY WARRANTY; without even the implied warranty of
	- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	- * Lesser General Public License for more details.
	- *
	- * You should have received a copy of the GNU Lesser General Public
	- * License along with this library; if not, write to the Free Software
	- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
	- */
	-
	-/*
	- * MPEG-1 Layer 3, or simply, "MP3", decoder for SDL_sound.
	- *
	- * This driver handles all those highly compressed songs you stole through
	- * Napster. :) It depends on the SMPEG library for decoding, which can
	- * be grabbed from: http://www.lokigames.com/development/smpeg.php3
	- *
	- * This should also be able to extract the audio stream from an MPEG movie.
	- *
	- * There is an alternative MP3 decoder available, called "mpglib", which
	- * doesn't depend on external libraries (the decoder itself is part of
	- * SDL_sound), and may be more efficient, but less flexible than SMPEG. YMMV.
	- *
	- * Please see the file COPYING in the source's root directory.
	- *
	- * This file written by Ryan C. Gordon. (icculus@icculus.org)
	- */
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#ifdef SOUND_SUPPORTS_SMPEG
	-
	-#include <stdio.h>
	-#include <stdlib.h>
	-#include <string.h>
	-
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-#include "smpeg.h"
	-#include "extra_rwops.h"
	-
	-
	-static int _SMPEG_init(void);
	-static void _SMPEG_quit(void);
	-static int _SMPEG_open(Sound_Sample sample, const char ext);
	-static void _SMPEG_close(Sound_Sample *sample);
	-static Uint32 _SMPEG_read(Sound_Sample *sample);
	-static int _SMPEG_rewind(Sound_Sample *sample);
	-static int _SMPEG_seek(Sound_Sample *sample, Uint32 ms);
	-
	-static const char *extensions_smpeg[] = { "MP3", "MPEG", "MPG", NULL };
	-const Sound_DecoderFunctions __Sound_DecoderFunctions_SMPEG =
	-{
	- {
	- extensions_smpeg,
	- "MPEG-1 Layer 3 audio through SMPEG",
	- "Ryan C. Gordon <icculus@icculus.org>",
	- "http://icculus.org/smpeg/"
	- },
	-
	- _SMPEG_init, /* init() method */
	- _SMPEG_quit, /* quit() method */
	- _SMPEG_open, /* open() method */
	- _SMPEG_close, /* close() method */
	- _SMPEG_read, /* read() method */
	- _SMPEG_rewind, /* rewind() method */
	- _SMPEG_seek /* seek() method */
	-};
	-
	-
	-static int _SMPEG_init(void)
	-{
	- return(1); /* always succeeds. */
	-} /* _SMPEG_init */
	-
	-
	-static void _SMPEG_quit(void)
	-{
	- /* it's a no-op. */
	-} /* _SMPEG_quit */
	-
	-
	-static __inline__ void output_version(void)
	-{
	- static int first_time = 1;
	-
	- if (first_time)
	- {
	- SMPEG_version v;
	- SMPEG_VERSION(&v);
	- SNDDBG(("SMPEG: Compiled against SMPEG v%d.%d.%d.\n",
	- v.major, v.minor, v.patch));
	- first_time = 0;
	- } /* if */
	-} /* output_version */
	-
	-
	-static int _SMPEG_open(Sound_Sample sample, const char ext)
	-{
	- SMPEG *smpeg;
	- SMPEG_Info smpeg_info;
	- SDL_AudioSpec spec;
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- SDL_RWops *refCounter;
	- const char *err = NULL;
	-
	- output_version();
	-
	- /*
	- * If I understand things correctly, MP3 files don't really have any
	- * magic header we can check for. The MP3 player is expected to just
	- * pick the first thing that looks like a valid frame and start
	- * playing from there.
	- *
	- * So here's what we do: If the caller insists that this is really
	- * MP3 we'll take his word for it. Otherwise, use the same test as
	- * SDL_mixer does and check if the stream starts with something that
	- * looks like a frame.
	- *
	- * A frame begins with 11 bits of frame sync (all bits must be set),
	- * followed by a two-bit MPEG Audio version ID:
	- *
	- * 00 - MPEG Version 2.5 (later extension of MPEG 2)
	- * 01 - reserved
	- * 10 - MPEG Version 2 (ISO/IEC 13818-3)
	- * 11 - MPEG Version 1 (ISO/IEC 11172-3)
	- *
	- * Apparently we don't handle MPEG Version 2.5.
	- */
	- if (__Sound_strcasecmp(ext, "MP3") != 0)
	- {
	- Uint8 mp3_magic[2];
	-
	- if (SDL_RWread(internal->rw, mp3_magic, sizeof (mp3_magic), 1) != 1)
	- BAIL_MACRO("SMPEG: Could not read MP3 magic.", 0);
	-
	- if (mp3_magic[0] != 0xFF \|\| (mp3_magic[1] & 0xF0) != 0xF0)
	- BAIL_MACRO("SMPEG: Not an MP3 stream.", 0);
	-
	- /* If the seek fails, we'll probably miss a frame, but oh well */
	- SDL_RWseek(internal->rw, -sizeof (mp3_magic), SEEK_CUR);
	- } /* if */
	-
	- refCounter = RWops_RWRefCounter_new(internal->rw);
	- if (refCounter == NULL)
	- {
	- SNDDBG(("SMPEG: Failed to create reference counting RWops.\n"));
	- return(0);
	- } /* if */
	-
	- /* replace original RWops. This is safe. Honest. :) */
	- internal->rw = refCounter;
	-
	- /*
	- * increment the refcount, since SMPEG will nuke the RWops if it can't
	- * accept the contained data...
	- */
	- RWops_RWRefCounter_addRef(refCounter);
	- smpeg = SMPEG_new_rwops(refCounter, &smpeg_info, 0);
	-
	- err = SMPEG_error(smpeg);
	- if (err != NULL)
	- {
	- __Sound_SetError(err); /* make a copy before SMPEG_delete()... */
	- SMPEG_delete(smpeg);
	- return(0);
	- } /* if */
	-
	- if (!smpeg_info.has_audio)
	- {
	- SMPEG_delete(smpeg);
	- BAIL_MACRO("SMPEG: No audio stream found in data.", 0);
	- } /* if */
	-
	- SNDDBG(("SMPEG: Accepting data stream.\n"));
	- SNDDBG(("SMPEG: has_audio == {%s}.\n", smpeg_info.has_audio ? "TRUE" : "FALSE"));
	- SNDDBG(("SMPEG: has_video == {%s}.\n", smpeg_info.has_video ? "TRUE" : "FALSE"));
	- SNDDBG(("SMPEG: width == (%d).\n", smpeg_info.width));
	- SNDDBG(("SMPEG: height == (%d).\n", smpeg_info.height));
	- SNDDBG(("SMPEG: current_frame == (%d).\n", smpeg_info.current_frame));
	- SNDDBG(("SMPEG: current_fps == (%f).\n", smpeg_info.current_fps));
	- SNDDBG(("SMPEG: audio_string == [%s].\n", smpeg_info.audio_string));
	- SNDDBG(("SMPEG: audio_current_frame == (%d).\n", smpeg_info.audio_current_frame));
	- SNDDBG(("SMPEG: current_offset == (%d).\n", smpeg_info.current_offset));
	- SNDDBG(("SMPEG: total_size == (%d).\n", smpeg_info.total_size));
	- SNDDBG(("SMPEG: current_time == (%f).\n", smpeg_info.current_time));
	- SNDDBG(("SMPEG: total_time == (%f).\n", smpeg_info.total_time));
	-
	- SMPEG_enablevideo(smpeg, 0);
	- SMPEG_enableaudio(smpeg, 1);
	- SMPEG_loop(smpeg, 0);
	-
	- SMPEG_wantedSpec(smpeg, &spec);
	-
	- /*
	- * One of the MP3s I tried wouldn't work unless I added this line
	- * to tell SMPEG that yes, it may have the spec it wants.
	- */
	- SMPEG_actualSpec(smpeg, &spec);
	- sample->actual.format = spec.format;
	- sample->actual.rate = spec.freq;
	- sample->actual.channels = spec.channels;
	- sample->flags = SOUND_SAMPLEFLAG_CANSEEK;
	- internal->decoder_private = smpeg;
	-
	- SMPEG_play(smpeg);
	- return(1);
	-} /* _SMPEG_open */
	-
	-
	-static void _SMPEG_close(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- SMPEG_delete((SMPEG *) internal->decoder_private);
	-} /* _SMPEG_close */
	-
	-
	-static Uint32 _SMPEG_read(Sound_Sample *sample)
	-{
	- Uint32 retval;
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- SMPEG smpeg = (SMPEG ) internal->decoder_private;
	-
	- /*
	- * We have to clear the buffer because apparently SMPEG_playAudio()
	- * will mix the decoded audio with whatever's already in it. Nasty.
	- */
	- memset(internal->buffer, '\0', internal->buffer_size);
	- retval = SMPEG_playAudio(smpeg, internal->buffer, internal->buffer_size);
	- if (retval < internal->buffer_size)
	- {
	- char *errMsg = SMPEG_error(smpeg);
	- if (errMsg == NULL)
	- sample->flags \|= SOUND_SAMPLEFLAG_EOF;
	- else
	- {
	- __Sound_SetError(errMsg);
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- } /* else */
	- } /* if */
	-
	- return(retval);
	-} /* _SMPEG_read */
	-
	-
	-static int _SMPEG_rewind(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- SMPEG smpeg = (SMPEG ) internal->decoder_private;
	- SMPEGstatus status;
	-
	- /*
	- * SMPEG_rewind() really means "stop and rewind", so we may have to
	- * restart it afterwards.
	- */
	- status = SMPEG_status(smpeg);
	- SMPEG_rewind(smpeg);
	- /* EW: I think SMPEG_play() has an independent and unrelated meaning
	- * to the flag, "SMPEG_PLAYING". This is why the SMPEG_play() call
	- * is done in the open() function even though the file is not yet
	- * technically playing. I believe SMPEG_play() must always be active
	- * because this seems to be what's causing the:
	- * "Can't rewind after the file has finished playing once" problem,
	- * because always recalling it here seems to make the problem go away.
	- */
	- /*
	- if (status == SMPEG_PLAYING)
	- SMPEG_play(smpeg);
	- */
	- SMPEG_play(smpeg);
	- return(1);
	-} /* _SMPEG_rewind */
	-
	-
	-static int _SMPEG_seek(Sound_Sample *sample, Uint32 ms)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- SMPEG smpeg = (SMPEG ) internal->decoder_private;
	- SMPEGstatus status;
	-
	- /*
	- * SMPEG_rewind() really means "stop and rewind", so we may have to
	- * restart it afterwards.
	- */
	- status = SMPEG_status(smpeg);
	- SMPEG_rewind(smpeg);
	- SMPEG_skip(smpeg, ((float) ms) / 1000.0);
	- if (status == SMPEG_PLAYING)
	- SMPEG_play(smpeg);
	- return(1);
	-} /* _SMPEG_seek */
	-
	-#endif /* SOUND_SUPPORTS_SMPEG */
	-
	-/* end of smpeg.c ... */
	-
	diff --git a/util/sdl/sound/decoders/speex.c b/util/sdl/sound/decoders/speex.c
	deleted file mode 100644
	index 83a2bda3..00000000
	--- a/util/sdl/sound/decoders/speex.c
	+++ /dev/null
	@@ -1,436 +0,0 @@
	-/*
	- * SDL_sound -- An abstract sound format decoding API.
	- * Copyright (C) 2001 Ryan C. Gordon.
	- *
	- * This library is free software; you can redistribute it and/or
	- * modify it under the terms of the GNU Lesser General Public
	- * License as published by the Free Software Foundation; either
	- * version 2.1 of the License, or (at your option) any later version.
	- *
	- * This library is distributed in the hope that it will be useful,
	- * but WITHOUT ANY WARRANTY; without even the implied warranty of
	- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	- * Lesser General Public License for more details.
	- *
	- * You should have received a copy of the GNU Lesser General Public
	- * License along with this library; if not, write to the Free Software
	- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
	- */
	-
	-/*
	- * Speex decoder for SDL_sound.
	- *
	- * This driver handles Speex audio data. Speex is a codec for speech that is
	- * meant to be transmitted over narrowband network connections. Epic Games
	- * estimates that their VoIP solution, built on top of Speex, uses around
	- * 500 bytes per second or less to transmit relatively good sounding speech.
	- *
	- * This decoder processes the .spx files that the speexenc program produces.
	- *
	- * Speex isn't meant for general audio compression. Something like Ogg Vorbis
	- * will give better results in that case.
	- *
	- * Further Speex information can be found at http://www.speex.org/
	- *
	- * This code is based on speexdec.c (see the Speex website).
	- *
	- * Please see the file COPYING in the source's root directory.
	- *
	- * This file written by Ryan C. Gordon. (icculus@icculus.org)
	- */
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#ifdef SOUND_SUPPORTS_SPEEX
	-
	-#include <stdio.h>
	-#include <stdlib.h>
	-#include <string.h>
	-#include <assert.h>
	-
	-#include <ogg/ogg.h>
	-#include <speex/speex.h>
	-#include <speex/speex_header.h>
	-
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-static int SPEEX_init(void);
	-static void SPEEX_quit(void);
	-static int SPEEX_open(Sound_Sample sample, const char ext);
	-static void SPEEX_close(Sound_Sample *sample);
	-static Uint32 SPEEX_read(Sound_Sample *sample);
	-static int SPEEX_rewind(Sound_Sample *sample);
	-static int SPEEX_seek(Sound_Sample *sample, Uint32 ms);
	-
	-static const char *extensions_speex[] = { "spx", NULL };
	-const Sound_DecoderFunctions __Sound_DecoderFunctions_SPEEX =
	-{
	- {
	- extensions_speex,
	- "SPEEX speech compression format",
	- "Ryan C. Gordon <icculus@icculus.org>",
	- "http://www.icculus.org/SDL_sound/"
	- },
	-
	- SPEEX_init, /* init() method */
	- SPEEX_quit, /* quit() method */
	- SPEEX_open, /* open() method */
	- SPEEX_close, /* close() method */
	- SPEEX_read, /* read() method */
	- SPEEX_rewind, /* rewind() method */
	- SPEEX_seek /* seek() method */
	-};
	-
	-#define SPEEX_USE_PERCEPTUAL_ENHANCER 1
	-#define SPEEX_MAGIC 0x5367674F /* "OggS" in ASCII (littleendian) */
	-#define SPEEX_OGG_BUFSIZE 200
	-
	-/* this is what we store in our internal->decoder_private field... */
	-typedef struct
	-{
	- ogg_sync_state oy;
	- ogg_page og;
	- ogg_packet op;
	- ogg_stream_state os;
	- void *state;
	- SpeexBits bits;
	- int header_count;
	- int frame_size;
	- int nframes;
	- int frames_avail;
	- float *decode_buf;
	- int decode_total;
	- int decode_pos;
	- int have_ogg_packet;
	-} speex_t;
	-
	-
	-static int SPEEX_init(void)
	-{
	- return(1); /* no-op. */
	-} /* SPEEX_init */
	-
	-
	-static void SPEEX_quit(void)
	-{
	- /* no-op. */
	-} /* SPEEX_quit */
	-
	-
	-static int process_header(speex_t speex, Sound_Sample sample)
	-{
	- SpeexMode *mode;
	- SpeexHeader *hptr;
	- SpeexHeader header;
	- int enh_enabled = SPEEX_USE_PERCEPTUAL_ENHANCER;
	- int tmp;
	-
	- hptr = speex_packet_to_header((char*) speex->op.packet, speex->op.bytes);
	- BAIL_IF_MACRO(!hptr, "SPEEX: Cannot read header", 0);
	- memcpy(&header, hptr, sizeof (SpeexHeader)); /* move to stack. */
	- free(hptr); /* lame that this forces you to malloc... */
	-
	- BAIL_IF_MACRO(header.mode >= SPEEX_NB_MODES, "SPEEX: Unknown mode", 0);
	- BAIL_IF_MACRO(header.mode < 0, "SPEEX: Unknown mode", 0);
	- mode = speex_mode_list[header.mode];
	- BAIL_IF_MACRO(header.speex_version_id > 1, "SPEEX: Unknown version", 0);
	- BAIL_IF_MACRO(mode->bitstream_version < header.mode_bitstream_version,
	- "SPEEX: Unsupported bitstream version", 0);
	- BAIL_IF_MACRO(mode->bitstream_version > header.mode_bitstream_version,
	- "SPEEX: Unsupported bitstream version", 0);
	-
	- speex->state = speex_decoder_init(mode);
	- BAIL_IF_MACRO(!speex->state, "SPEEX: Decoder initialization error", 0);
	-
	- speex_decoder_ctl(speex->state, SPEEX_SET_ENH, &enh_enabled);
	- speex_decoder_ctl(speex->state, SPEEX_GET_FRAME_SIZE, &speex->frame_size);
	-
	- speex->decode_buf = (float ) malloc(speex->frame_size sizeof (float));
	- BAIL_IF_MACRO(!speex->decode_buf, ERR_OUT_OF_MEMORY, 0);
	-
	- speex->nframes = header.frames_per_packet;
	- if (!speex->nframes)
	- speex->nframes = 1;
	-
	- /* !!! FIXME: Write converters to match desired format.
	- !!! FIXME: We have to convert from Float32 anyhow. */
	- /* !!! FIXME: Is it a performance hit to alter sampling rate?
	- !!! FIXME: If not, try to match desired rate. */
	- /* !!! FIXME: We force mono output, but speexdec.c has code for stereo.
	- !!! FIXME: Use that if sample->desired.channels == 2? */
	- tmp = header.rate;
	- speex_decoder_ctl(speex->state, SPEEX_SET_SAMPLING_RATE, &tmp);
	- speex_decoder_ctl(speex->state, SPEEX_GET_SAMPLING_RATE, &tmp);
	- sample->actual.rate = tmp;
	- sample->actual.channels = 1;
	- sample->actual.format = AUDIO_S16SYS;
	-
	- SNDDBG(("SPEEX: %dHz, mono, %svbr, %s mode.\n",
	- (int) sample->actual.rate,
	- header.vbr ? "" : "not ",
	- mode->modeName));
	-
	- /* plus 2: one for this header, one for the comment header. */
	- speex->header_count = header.extra_headers + 2;
	- return(1);
	-} /* process_header */
	-
	-
	-/* !!! FIXME: this code sucks. */
	-static int SPEEX_open(Sound_Sample sample, const char ext)
	-{
	- int set_error_str = 1;
	- int bitstream_initialized = 0;
	- Uint8 *buffer = NULL;
	- int packet_count = 0;
	- speex_t *speex = NULL;
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- SDL_RWops *rw = internal->rw;
	- Uint32 magic;
	-
	- /* Quick rejection. */
	- /*
	- * !!! FIXME: If (ext) is .spx, ignore bad magic number and assume
	- * !!! FIXME: this is a corrupted file...try to sync up further in
	- * !!! FIXME: stream. But for general purposes we can't read the
	- * !!! FIXME: whole RWops here in case it's not a Speex file at all.
	- */
	- magic = SDL_ReadLE32(rw); /* make sure this is an ogg stream. */
	- BAIL_IF_MACRO(magic != SPEEX_MAGIC, "SPEEX: Not a complete ogg stream", 0);
	- BAIL_IF_MACRO(SDL_RWseek(rw, -4, SEEK_CUR) < 0, ERR_IO_ERROR, 0);
	-
	- speex = (speex_t *) malloc(sizeof (speex_t));
	- BAIL_IF_MACRO(speex == NULL, ERR_OUT_OF_MEMORY, 0);
	- memset(speex, '\0', sizeof (speex_t));
	-
	- speex_bits_init(&speex->bits);
	- if (ogg_sync_init(&speex->oy) != 0) goto speex_open_failed;
	-
	- while (1)
	- {
	- int rc;
	- Uint8 buffer = (Uint8)ogg_sync_buffer(&speex->oy, SPEEX_OGG_BUFSIZE);
	- if (buffer == NULL) goto speex_open_failed;
	- rc = SDL_RWread(rw, buffer, 1, SPEEX_OGG_BUFSIZE);
	- if (rc <= 0) goto speex_open_failed;
	- if (ogg_sync_wrote(&speex->oy, rc) != 0) goto speex_open_failed;
	- while (ogg_sync_pageout(&speex->oy, &speex->og) == 1)
	- {
	- if (!bitstream_initialized)
	- {
	- if (ogg_stream_init(&speex->os, ogg_page_serialno(&speex->og)))
	- goto speex_open_failed;
	- bitstream_initialized = 1;
	- } /* if */
	-
	- if (ogg_stream_pagein(&speex->os, &speex->og) != 0)
	- goto speex_open_failed;
	-
	- while (ogg_stream_packetout(&speex->os, &speex->op) == 1)
	- {
	- if (speex->op.e_o_s)
	- goto speex_open_failed; /* end of stream already?! */
	-
	- packet_count++;
	- if (packet_count == 1) /* need speex header. */
	- {
	- if (!process_header(speex, sample))
	- {
	- set_error_str = 0; /* process_header will set error string. */
	- goto speex_open_failed;
	- } /* if */
	- } /* if */
	-
	- if (packet_count > speex->header_count)
	- {
	- /* if you made it here, you're ready to get a waveform. */
	- SNDDBG(("SPEEX: Accepting data stream.\n"));
	-
	- /* sample->actual is configured in process_header()... */
	- speex->have_ogg_packet = 1;
	- sample->flags = SOUND_SAMPLEFLAG_NONE;
	- internal->decoder_private = speex;
	- return(1); /* we'll handle this data. */
	- } /* if */
	- } /* while */
	-
	- } /* while */
	-
	- } /* while */
	-
	- assert(0); /* shouldn't hit this point. */
	-
	-speex_open_failed:
	- if (speex != NULL)
	- {
	- if (speex->state != NULL)
	- speex_decoder_destroy(speex->state);
	- if (bitstream_initialized)
	- ogg_stream_clear(&speex->os);
	- speex_bits_destroy(&speex->bits);
	- ogg_sync_clear(&speex->oy);
	- free(speex->decode_buf);
	- free(speex);
	- } /* if */
	-
	- if (set_error_str)
	- BAIL_MACRO("SPEEX: decoding error", 0);
	-
	- return(0);
	-} /* SPEEX_open */
	-
	-
	-static void SPEEX_close(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- speex_t speex = (speex_t ) internal->decoder_private;
	- speex_decoder_destroy(speex->state);
	- ogg_stream_clear(&speex->os);
	- speex_bits_destroy(&speex->bits);
	- ogg_sync_clear(&speex->oy);
	- free(speex->decode_buf);
	- free(speex);
	-} /* SPEEX_close */
	-
	-
	-static Uint32 copy_from_decoded(speex_t *speex,
	- Sound_SampleInternal *internal,
	- Uint32 _cpypos)
	-{
	- /*
	- * !!! FIXME: Obviously, this all needs to change if we allow for
	- * !!! FIXME: more than mono, S16SYS data.
	- */
	- Uint32 cpypos = _cpypos >> 1;
	- Sint16 dst = ((Sint16 ) internal->buffer) + cpypos;
	- Sint16 *max;
	- Uint32 maxoutput = (internal->buffer_size >> 1) - cpypos;
	- Uint32 maxavail = speex->decode_total - speex->decode_pos;
	- float *src = speex->decode_buf + speex->decode_pos;
	-
	- if (maxavail < maxoutput)
	- maxoutput = maxavail;
	-
	- speex->decode_pos += maxoutput;
	- cpypos += maxoutput;
	-
	- for (max = dst + maxoutput; dst < max; dst++, src++)
	- {
	- /* !!! FIXME: This screams for vectorization. */
	- register float f = *src;
	- if (f > 32000.0f) /* eh, speexdec.c clamps like this, too. */
	- f = 32000.0f;
	- else if (f < -32000.0f)
	- f = -32000.0f;
	- *dst = (Sint16) (0.5f + f);
	- } /* for */
	-
	- return(cpypos << 1);
	-} /* copy_from_decoded */
	-
	-
	-/* !!! FIXME: this code sucks. */
	-static Uint32 SPEEX_read(Sound_Sample *sample)
	-{
	- Uint32 retval = 0;
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- speex_t speex = (speex_t ) internal->decoder_private;
	- SDL_RWops *rw = internal->rw;
	- Uint8 *buffer;
	- int rc;
	-
	- while (1)
	- {
	- /* see if there's some already-decoded leftovers... */
	- if (speex->decode_total != speex->decode_pos)
	- {
	- retval = copy_from_decoded(speex, internal, retval);
	- if (retval >= internal->buffer_size)
	- return(retval); /* whee. */
	- } /* if */
	-
	- /* okay, decoded buffer is spent. What else do we have? */
	- speex->decode_total = speex->decode_pos = 0;
	-
	- if (speex->frames_avail) /* have more frames to decode? */
	- {
	- rc = speex_decode(speex->state, &speex->bits, speex->decode_buf);
	- if (rc < 0) goto speex_read_failed;
	- if (speex_bits_remaining(&speex->bits) < 0) goto speex_read_failed;
	- speex->frames_avail--;
	- speex->decode_total = speex->frame_size;
	- continue; /* go fill the output buffer... */
	- } /* if */
	-
	- /* need to get more speex frames from available ogg packets... */
	- if (speex->have_ogg_packet)
	- {
	- speex_bits_read_from(&speex->bits,
	- (char *) speex->op.packet,
	- speex->op.bytes);
	-
	- speex->frames_avail += speex->nframes;
	- if (ogg_stream_packetout(&speex->os, &speex->op) <= 0)
	- speex->have_ogg_packet = 0;
	- continue; /* go decode these frames. */
	- } /* if */
	-
	- /* need to get more ogg packets from bitstream... */
	-
	- if (speex->op.e_o_s) /* okay, we're really spent. */
	- {
	- sample->flags \|= SOUND_SAMPLEFLAG_EOF;
	- return(retval);
	- } /* if */
	-
	- while ((!speex->op.e_o_s) && (!speex->have_ogg_packet))
	- {
	- buffer = (Uint8 *) ogg_sync_buffer(&speex->oy, SPEEX_OGG_BUFSIZE);
	- if (buffer == NULL) goto speex_read_failed;
	- rc = SDL_RWread(rw, buffer, 1, SPEEX_OGG_BUFSIZE);
	- if (rc <= 0) goto speex_read_failed;
	- if (ogg_sync_wrote(&speex->oy, rc) != 0) goto speex_read_failed;
	-
	- /* got complete ogg page? */
	- if (ogg_sync_pageout(&speex->oy, &speex->og) == 1)
	- {
	- if (ogg_stream_pagein(&speex->os, &speex->og) != 0)
	- goto speex_read_failed;
	-
	- /* got complete ogg packet? */
	- if (ogg_stream_packetout(&speex->os, &speex->op) == 1)
	- speex->have_ogg_packet = 1;
	- } /* if */
	- } /* while */
	- } /* while */
	-
	- assert(0); /* never hit this. Either return or goto speex_read_failed */
	-
	-speex_read_failed:
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- /* !!! FIXME: "i/o error" is better in some situations. */
	- BAIL_MACRO("SPEEX: Decoding error", retval);
	-} /* SPEEX_read */
	-
	-
	-static int SPEEX_rewind(Sound_Sample *sample)
	-{
	- /* !!! FIXME */ return(0);
	-} /* SPEEX_rewind */
	-
	-
	-static int SPEEX_seek(Sound_Sample *sample, Uint32 ms)
	-{
	- /* !!! FIXME */ return(0);
	-} /* SPEEX_seek */
	-
	-
	-#endif /* SOUND_SUPPORTS_SPEEX */
	-
	-/* end of speex.c ... */
	-
	diff --git a/util/sdl/sound/decoders/timidity/CHANGES b/util/sdl/sound/decoders/timidity/CHANGES
	deleted file mode 100644
	index ab79e993..00000000
	--- a/util/sdl/sound/decoders/timidity/CHANGES
	+++ /dev/null
	@@ -1,77 +0,0 @@
	-This version of TiMidity should contain all the fixes from the
	-September 25 2003 SDL_mixer CVS snapshot. In addition, I've made some
	-changes of my own, e.g.:
	-
	-* All file access is done through SDL_RWops. This means the MIDI
	- stream no longer has to be a file. (The config file and instruments
	- still have to be though.)
	-
	-* Replacing of TiMidity's endian-handling with SDL's.
	-
	-* Removal of much unused or unnecessary code, such as
	-
	- + The "hooks" for putting a user interface onto TiMidity.
	- + The antialias filter. It wasn't active, and even at 4 kHz I
	- couldn't hear any difference when activating it.
	- + Removed all traces of LOOKUP_HACK and LOOKUP_INTERPOLATION.
	- According to the code comments they weren't very good anyway.
	- ("degrades sound quality noticeably"). I also removed the
	- disclaimer about the "8-bit uLaw to 16-bit PCM and the 13-bit-PCM
	- to 8-bit uLaw tables" disclaimer, since I believe those were the
	- tables I removed.
	- + Removed LOOKUP_SINE since it was already commented out. I think we
	- can count on our target audience having math co-processors
	- nowadays.
	- + Removed USE_LDEXP since it wasn't being used and "it doesn't make
	- much of a difference either way".
	- + Removed decompress hack from open_file() since it didn't look very
	- portable.
	- + Removed heaps of unnecessary constants.
	- + Removed unused functions.
	- + Assume that LINEAR_INTERPOLATION is always used, so remove all
	- code dealing with it not being so. It's not that I think the
	- difference in audio quality is that great, but since it wouldn't
	- compile without code changes I assume no one's used it for quite
	- some time...
	- + Assume PRECALC_LOOPS is always defined. Judging by the comments it
	- may not make much of a difference either way, so why maintain two
	- versions of the same code?
	-
	-* Moving several static globals into the MidiSong struct. This
	- includes sample rate, formate, etc. which are now all per-song.
	-
	-* Moved some typedefs (e.g. MidiSong) to timidity.h for easy inclusion
	- into the MIDI decoder.
	-
	-* Added free_pathlist().
	-
	-* Replaced TiMidity's own 8, 16 and 32-bit types with SDL's.
	-
	-* Made TiMidity look for its configuration file in both /etc and
	- /usr/local/lib/timidity. (Windows version remains unchanged.)
	-
	-* Timidity_PlaySome() now takes three arguments. A MidiSong, a decode
	- buffer and decode buffer size in bytes. (MidiSong is a new argument,
	- and buffer size used to be in samples.)
	-
	- In addition, it will return the number of bytes decoded.
	-
	-* Added Timidity_Exit().
	-
	-* Removed Timidity_Stop() and Timidity_Active(). Stopping playback
	- should be handled by SDL_sound, and Timidity_PlaySome() will return
	- 0 when the MIDI stream is finished.
	-
	-* Modified the ToneBank stuff to allow some data to be shared between
	- MidiSongs.
	-
	-* The following files have been removed: controls.c, controls.h,
	- filter.c, filter.h, sdl_a.c, sdl_c.c
	-
	-* config.h has been renamed as options.h to avoid confusion with the
	- automatically generated config.h for SDL_sound.
	-
	-* Added support for loading DLS format instruments:
	- Timidity_LoadDLS(), Timidity_FreeDLS(), Timidity_LoadDLSSong()
	-
	-* Added Timidity_Init_NoConfig()
	diff --git a/util/sdl/sound/decoders/timidity/COPYING b/util/sdl/sound/decoders/timidity/COPYING
	deleted file mode 100644
	index 44bb52fb..00000000
	--- a/util/sdl/sound/decoders/timidity/COPYING
	+++ /dev/null
	@@ -1,519 +0,0 @@
	-Please note that the included source from Timidity, the MIDI decoder, is also
	- licensed under the following terms (GNU LGPL), but can also be used
	- separately under the GNU GPL, or the Perl Artistic License. Those licensing
	- terms are not reprinted here, but can be found on the web easily.
	-
	-If you want to use SDL_sound under a closed-source license, please contact
	- Ryan (icculus@icculus.org), and we can discuss an alternate license for
	- money to be distributed between the contributors to this work, but I'd
	- encourage you to abide by the LGPL, since the usual concern is whether you
	- can use this library without releasing your own source code (you can).
	-
	-
	--------------------
	-
	-
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	- Ty Coon, President of Vice
	-
	-That's all there is to it!
	-
	-
	diff --git a/util/sdl/sound/decoders/timidity/FAQ b/util/sdl/sound/decoders/timidity/FAQ
	deleted file mode 100644
	index 1ee0b77b..00000000
	--- a/util/sdl/sound/decoders/timidity/FAQ
	+++ /dev/null
	@@ -1,100 +0,0 @@
	-----------------------------indented-text-------------------------------
	-
	- TiMidity -- Experimental MIDI to WAVE converter
	- Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi>
	-
	---------------------------------------------------------------------------
	-
	- Frequently Asked Questions with answers:
	-
	---------------------------------------------------------------------------
	-Q: What is it?
	-
	-A: Where? Well Chris, TiMidity is a software-only synthesizer, MIDI
	- renderer, MIDI to WAVE converter, realtime MIDI player for UNIX machines,
	- even (I've heard) a Netscape helper application. It takes a MIDI file
	- and writes a WAVE or raw PCM data or plays it on your digital audio
	- device. It sounds much more realistic than FM synthesis, but you need a
	- ~100Mhz processor to listen to 32kHz stereo music in the background while
	- you work. 11kHz mono can be played on a low-end 486, and, to some, it
	- still sounds better than FM.
	-
	---------------------------------------------------------------------------
	-Q: I don't have a GUS, can I use TiMidity?
	-
	-A: Yes. That's the point. You don't need a Gravis Ultrasound to use
	- TiMidity, you just need GUS-compatible patches, which are freely
	- available on the Internet. See below for pointers.
	-
	---------------------------------------------------------------------------
	-Q: I have a GUS, can I use TiMidity?
	-
	-A: The DOS port doesn't have GUS support, and TiMidity won't be taking
	- advantage of the board's internal synthesizer under other operating
	- systems either. So it kind of defeats the purpose. But you can use it.
	-
	---------------------------------------------------------------------------
	-Q: I tried playing a MIDI file I got off the Net but all I got was a
	- dozen warnings saying "No instrument mapped to tone bank 0, program
	- xx - this instrument will not be heard". What's wrong?
	-
	-A: The General MIDI standard specifies 128 melodic instruments and
	- some sixty percussion sounds. If you wish to play arbitrary General
	- MIDI files, you'll need to get more patch files.
	-
	- There's a program called Midia for SGI's, which also plays MIDI
	- files and has a lot more bells and whistles than TiMidity. It uses
	- GUS-compatible patches, too -- so you can get the 8 MB set at
	- ftp://archive.cs.umbc.edu/pub/midia for pretty good GM compatibility.
	-
	- There are also many excellent patches on the Ultrasound FTP sites.
	- I can recommend Dustin McCartney's collections gsdrum*.zip and
	- wow*.zip in the "[.../]sound/patches/files" directory. The huge
	- ProPats series (pp3-*.zip) contains good patches as well. General
	- MIDI files can also be found on these sites.
	-
	- This site list is from the GUS FAQ:
	-
	-> FTP Sites Archive Directories
	-> --------- -------------------
	-> Main N.American Site: archive.orst.edu pub/packages/gravis
	-> wuarchive.wustl.edu systems/ibmpc/ultrasound
	-> Main Asian Site: nctuccca.edu.tw PC/ultrasound
	-> Main European Site: src.doc.ic.ac.uk packages/ultrasound
	-> Main Australian Site: ftp.mpx.com.au /ultrasound/general
	-> /ultrasound/submit
	-> South African Site: ftp.sun.ac.za /pub/packages/ultrasound
	-> Submissions: archive.epas.utoronto.ca pub/pc/ultrasound/submit
	-> Newly Validated Files: archive.epas.utoronto.ca pub/pc/ultrasound
	->
	-> Mirrors: garbo.uwasa.fi mirror/ultrasound
	-> ftp.st.nepean.uws.edu.au pc/ultrasound
	-> ftp.luth.se pub/msdos/ultrasound
	-
	---------------------------------------------------------------------------
	-Q: Some files have awful clicks and pops.
	-
	-A: Find out which patch is responsible for the clicking (try "timidity
	- -P<patch> <midi/test-decay\|midi/test-panning>". Add "strip=tail" in
	- the config file after its name. If this doesn't fix it, mail me the
	- patch.
	-
	---------------------------------------------------------------------------
	-Q: I'm playing Fantasie Impromptu in the background. When I run Netscape,
	- the sound gets choppy and it takes ten minutes to load. What can I do?
	-
	-A: Here are some things to try:
	-
	- - Use a lower sampling rate.
	-
	- - Use mono output. This can improve performance by 10-30%.
	- (Using 8-bit instead of 16-bit output makes no difference.)
	-
	- - Use a smaller number of simultaneous voices.
	-
	- - Make sure you compiled with FAST_DECAY enabled in options.h
	-
	- - Recompile with an Intel-optimized gcc for a 5-15%
	- performance increase.
	-
	---------------------------------------------------------------------------
	diff --git a/util/sdl/sound/decoders/timidity/Makefile.am b/util/sdl/sound/decoders/timidity/Makefile.am
	deleted file mode 100644
	index 7c64b933..00000000
	--- a/util/sdl/sound/decoders/timidity/Makefile.am
	+++ /dev/null
	@@ -1,33 +0,0 @@
	-if USE_TIMIDITY
	-noinst_LTLIBRARIES = libtimidity.la
	-endif
	-
	-INCLUDES = -I$(top_srcdir)
	-
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	- common.c \
	- common.h \
	- dls1.h \
	- dls2.h \
	- instrum.c \
	- instrum.h \
	- instrum_dls.c \
	- instrum_dls.h \
	- mix.c \
	- mix.h \
	- options.h \
	- output.c \
	- output.h \
	- playmidi.c \
	- playmidi.h \
	- readmidi.c \
	- readmidi.h \
	- resample.c \
	- resample.h \
	- tables.c \
	- tables.h \
	- timidity.c \
	- timidity.h
	-
	-EXTRA_DIST = CHANGES COPYING FAQ README TODO Makefile.testmidi testmidi.c
	-
	diff --git a/util/sdl/sound/decoders/timidity/Makefile.in b/util/sdl/sound/decoders/timidity/Makefile.in
	deleted file mode 100644
	index 03b1e427..00000000
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	- $(srcdir)/*) file=`echo "$$file" \| sed "s\|^$$srcdirstrip/\|\|"`;; \
	- $(top_srcdir)/*) file=`echo "$$file" \| sed "s\|^$$topsrcdirstrip/\|$(top_builddir)/\|"`;; \
	- esac; \
	- if test -f $$file \|\| test -d $$file; then d=.; else d=$(srcdir); fi; \
	- dir=`echo "$$file" \| sed -e 's,/[^/]*$$,,'`; \
	- if test "$$dir" != "$$file" && test "$$dir" != "."; then \
	- dir="/$$dir"; \
	- $(mkdir_p) "$(distdir)$$dir"; \
	- else \
	- dir=''; \
	- fi; \
	- if test -d $$d/$$file; then \
	- if test -d $(srcdir)/$$file && test $$d != $(srcdir); then \
	- cp -pR $(srcdir)/$$file $(distdir)$$dir \|\| exit 1; \
	- fi; \
	- cp -pR $$d/$$file $(distdir)$$dir \|\| exit 1; \
	- else \
	- test -f $(distdir)/$$file \
	- \|\| cp -p $$d/$$file $(distdir)/$$file \
	- \|\| exit 1; \
	- fi; \
	- done
	-check-am: all-am
	-check: check-am
	-all-am: Makefile $(LTLIBRARIES)
	-installdirs:
	-install: install-am
	-install-exec: install-exec-am
	-install-data: install-data-am
	-uninstall: uninstall-am
	-
	-install-am: all-am
	- @$(MAKE) $(AM_MAKEFLAGS) install-exec-am install-data-am
	-
	-installcheck: installcheck-am
	-install-strip:
	- $(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
	- install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
	- `test -z '$(STRIP)' \|\| \
	- echo "INSTALL_PROGRAM_ENV=STRIPPROG='$(STRIP)'"` install
	-mostlyclean-generic:
	-
	-clean-generic:
	-
	-distclean-generic:
	- -test -z "$(CONFIG_CLEAN_FILES)" \|\| rm -f $(CONFIG_CLEAN_FILES)
	-
	-maintainer-clean-generic:
	- @echo "This command is intended for maintainers to use"
	- @echo "it deletes files that may require special tools to rebuild."
	-clean: clean-am
	-
	-clean-am: clean-generic clean-libtool clean-noinstLTLIBRARIES \
	- mostlyclean-am
	-
	-distclean: distclean-am
	- -rm -rf ./$(DEPDIR)
	- -rm -f Makefile
	-distclean-am: clean-am distclean-compile distclean-generic \
	- distclean-libtool distclean-tags
	-
	-dvi: dvi-am
	-
	-dvi-am:
	-
	-html: html-am
	-
	-info: info-am
	-
	-info-am:
	-
	-install-data-am:
	-
	-install-exec-am:
	-
	-install-info: install-info-am
	-
	-install-man:
	-
	-installcheck-am:
	-
	-maintainer-clean: maintainer-clean-am
	- -rm -rf ./$(DEPDIR)
	- -rm -f Makefile
	-maintainer-clean-am: distclean-am maintainer-clean-generic
	-
	-mostlyclean: mostlyclean-am
	-
	-mostlyclean-am: mostlyclean-compile mostlyclean-generic \
	- mostlyclean-libtool
	-
	-pdf: pdf-am
	-
	-pdf-am:
	-
	-ps: ps-am
	-
	-ps-am:
	-
	-uninstall-am: uninstall-info-am
	-
	-.PHONY: CTAGS GTAGS all all-am check check-am clean clean-generic \
	- clean-libtool clean-noinstLTLIBRARIES ctags distclean \
	- distclean-compile distclean-generic distclean-libtool \
	- distclean-tags distdir dvi dvi-am html html-am info info-am \
	- install install-am install-data install-data-am install-exec \
	- install-exec-am install-info install-info-am install-man \
	- install-strip installcheck installcheck-am installdirs \
	- maintainer-clean maintainer-clean-generic mostlyclean \
	- mostlyclean-compile mostlyclean-generic mostlyclean-libtool \
	- pdf pdf-am ps ps-am tags uninstall uninstall-am \
	- uninstall-info-am
	-
	-# Tell versions [3.59,3.63) of GNU make to not export all variables.
	-# Otherwise a system limit (for SysV at least) may be exceeded.
	-.NOEXPORT:
	diff --git a/util/sdl/sound/decoders/timidity/Makefile.testmidi b/util/sdl/sound/decoders/timidity/Makefile.testmidi
	deleted file mode 100644
	index 8f03fdab..00000000
	--- a/util/sdl/sound/decoders/timidity/Makefile.testmidi
	+++ /dev/null
	@@ -1,38 +0,0 @@
	-# Silly test makefile
	-
	-CC = gcc
	-
	-# Standard SDL_sound debugging
	-CFLAGS = -g -I../.. -ansi -pedantic -Wall `sdl-config --cflags` -DDEBUG_CHATTER
	-LIBS = `sdl-config --libs`
	-
	-# Electric Fence debugging
	-# CFLAGS = -g -I../.. -ansi -pedantic -Wall `sdl-config --cflags`
	-# LIBS = `sdl-config --libs` -lefence
	-
	-OBJECTS = common.o instrum.o mix.o output.o playmidi.o readmidi.o resample.o \
	- tables.o timidity.o testmidi.o
	-
	-all: testmidi
	-
	-testmidi: $(OBJECTS)
	- $(CC) $(OBJECTS) $(CFLAGS) -o testmidi $(LIBS)
	-
	-clean:
	- $(RM) testmidi .o ~
	-
	-common.o: common.c options.h common.h
	-instrum.o: instrum.c timidity.h options.h common.h instrum.h resample.h \
	- tables.h
	-mix.o: mix.c timidity.h options.h instrum.h playmidi.h output.h tables.h \
	- resample.h mix.h
	-output.o: output.c options.h output.h
	-playmidi.o: playmidi.c timidity.h options.h instrum.h playmidi.h output.h \
	- mix.h tables.h
	-readmidi.o: readmidi.c timidity.h common.h instrum.h playmidi.h
	-resample.o: resample.c timidity.h options.h common.h instrum.h playmidi.h \
	- tables.h resample.h
	-tables.o: tables.c tables.h
	-testmidi.o: testmidi.c common.h timidity.h
	-timidity.o: timidity.c options.h common.h instrum.h playmidi.h readmidi.h \
	- output.h timidity.h tables.h
	diff --git a/util/sdl/sound/decoders/timidity/README b/util/sdl/sound/decoders/timidity/README
	deleted file mode 100644
	index 9c9c55aa..00000000
	--- a/util/sdl/sound/decoders/timidity/README
	+++ /dev/null
	@@ -1,61 +0,0 @@
	-[This version of timidity has been stripped for simplicity in porting to SDL,
	-and then even further for SDL_sound]
	-----------------------------------text----------------------------------
	-
	- From http://www.cgs.fi/~tt/discontinued.html :
	-
	- If you'd like to continue hacking on TiMidity, feel free. I'm
	- hereby extending the TiMidity license agreement: you can now
	- select the most convenient license for your needs from (1) the
	- GNU GPL, (2) the GNU LGPL, or (3) the Perl Artistic License.
	-
	---------------------------------------------------------------------------
	-
	- This is the README file for TiMidity v0.2i
	-
	- TiMidity is a MIDI to WAVE converter that uses Gravis
	-Ultrasound(*)-compatible patch files to generate digital audio data
	-from General MIDI files. The audio data can be played through any
	-sound device or stored on disk. On a fast machine, music can be
	-played in real time. TiMidity runs under Linux, FreeBSD, HP-UX, SunOS, and
	-Win32, and porting to other systems with gcc should be easy.
	-
	- TiMidity Features:
	-
	- * 32 or more dynamically allocated fully independent voices
	- * Compatibility with GUS patch files
	- * Output to 16- or 8-bit PCM or uLaw audio device, file, or
	- stdout at any sampling rate
	- * Optional interactive mode with real-time status display
	- under ncurses and SLang terminal control libraries. Also
	- a user friendly motif interface since version 0.2h
	- * Support for transparent loading of compressed MIDI files and
	- patch files
	-
	- * Support for the following MIDI events:
	- - Program change
	- - Key pressure
	- - Channel main volume
	- - Tempo
	- - Panning
	- - Damper pedal (Sustain)
	- - Pitch wheel
	- - Pitch wheel sensitivity
	- - Change drum set
	-
	-* The GNU General Public License can, as always, be found in the file
	- "../COPYING".
	-
	-* TiMidity requires sampled instruments (patches) to play MIDI files. You
	- should get the file "timidity-lib-0.1.tar.gz" and unpack it in the same
	- directory where you unpacked the source code archive. You'll want more
	- patches later -- read the file "FAQ" for pointers.
	-
	-* Timidity is no longer supported, but can be found by searching the web.
	-
	-
	- Tuukka Toivonen <toivonen@clinet.fi>
	-
	-[(*) Any Registered Trademarks used anywhere in the documentation or
	-source code for TiMidity are acknowledged as belonging to their
	-respective owners.]
	diff --git a/util/sdl/sound/decoders/timidity/TODO b/util/sdl/sound/decoders/timidity/TODO
	deleted file mode 100644
	index 69b37ee2..00000000
	--- a/util/sdl/sound/decoders/timidity/TODO
	+++ /dev/null
	@@ -1,37 +0,0 @@
	-* I don't like the indentation style at all, but for the most part
	- I've left it alone.
	-
	-* Much of the code looks ugly to me.
	-
	-* The return value from SDL_RWread() is checked inconsistenly.
	-
	-* Group the members of MidiSong into logical units, i.e. structs?
	-
	-* The debug messages are probably a bit too noisy. I've removed one
	- particularly annoying one, but...
	-
	- Some of them should be turned into error messages instead.
	-
	-* Can the instrument handling be made more efficient? At the moment
	- different MidiSongs may separately load the same instrument.
	-
	- Note that the MidiSong's audio format affects how the instrument is
	- loaded, so it's not as easy as just letting all MidiSongs share tone
	- and drum banks.
	-
	- At the moment they do share the data that is simply read from the
	- config file, but that's just a quick hack to avoid having to read
	- the config file every time a MIDI song is loaded.
	-
	-* Check if any of MidiStruct's members can safely be made into static
	- globals again.
	-
	-* TiMidity++ adds a number of undocumented (?) extensions to the
	- configuration syntax. These are not implemented here. In particular,
	- the "map" keyword used by the "eawpats".
	-
	-* The other decoders generally only read as much of the file as is
	- necessary. Could we do that in this decoder as well? (Currently it
	- seems to convert the entire file into MIDI events first.)
	-
	-* Can it be optimized?
	diff --git a/util/sdl/sound/decoders/timidity/common.c b/util/sdl/sound/decoders/timidity/common.c
	deleted file mode 100644
	index 81735d65..00000000
	--- a/util/sdl/sound/decoders/timidity/common.c
	+++ /dev/null
	@@ -1,137 +0,0 @@
	-/*
	-
	- TiMidity -- Experimental MIDI to WAVE converter
	- Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi>
	-
	- This program is free software; you can redistribute it and/or modify
	- it under the terms of the GNU General Public License as published by
	- the Free Software Foundation; either version 2 of the License, or
	- (at your option) any later version.
	-
	- This program is distributed in the hope that it will be useful,
	- but WITHOUT ANY WARRANTY; without even the implied warranty of
	- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
	- GNU General Public License for more details.
	-
	- You should have received a copy of the GNU General Public License
	- along with this program; if not, write to the Free Software
	- Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
	-
	- common.c
	-
	- */
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#include <stdio.h>
	-#include <stdlib.h>
	-#include <string.h>
	-
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-#include "options.h"
	-#include "common.h"
	-
	-/* The paths in this list will be tried whenever we're reading a file */
	-static PathList pathlist = NULL; / This is a linked list */
	-
	-/* This is meant to find and open files for reading */
	-SDL_RWops open_file(char name)
	-{
	- SDL_RWops *rw;
	-
	- if (!name \|\| !(*name))
	- {
	- SNDDBG(("Attempted to open nameless file.\n"));
	- return 0;
	- }
	-
	- /* First try the given name */
	-
	- SNDDBG(("Trying to open %s\n", name));
	- if ((rw = SDL_RWFromFile(name, "rb")))
	- return rw;
	-
	- if (name[0] != PATH_SEP)
	- {
	- char current_filename[1024];
	- PathList *plp = pathlist;
	- int l;
	-
	- while (plp) /* Try along the path then */
	- {
	- *current_filename = 0;
	- l = strlen(plp->path);
	- if(l)
	- {
	- strcpy(current_filename, plp->path);
	- if(current_filename[l - 1] != PATH_SEP)
	- {
	- current_filename[l] = PATH_SEP;
	- current_filename[l + 1] = '\0';
	- }
	- }
	- strcat(current_filename, name);
	- SNDDBG(("Trying to open %s\n", current_filename));
	- if ((rw = SDL_RWFromFile(current_filename, "rb")))
	- return rw;
	- plp = plp->next;
	- }
	- }
	-
	- /* Nothing could be opened. */
	- SNDDBG(("Could not open %s\n", name));
	- return 0;
	-}
	-
	-/* This'll allocate memory or die. */
	-void *safe_malloc(size_t count)
	-{
	- void *p;
	-
	- p = malloc(count);
	- if (p == NULL)
	- SNDDBG(("Sorry. Couldn't malloc %d bytes.\n", count));
	-
	- return p;
	-}
	-
	-/* This adds a directory to the path list */
	-void add_to_pathlist(char *s)
	-{
	- PathList *plp = safe_malloc(sizeof(PathList));
	-
	- if (plp == NULL)
	- return;
	-
	- plp->path = safe_malloc(strlen(s) + 1);
	- if (plp->path == NULL)
	- {
	- free(plp);
	- return;
	- }
	-
	- strcpy(plp->path, s);
	- plp->next = pathlist;
	- pathlist = plp;
	-}
	-
	-void free_pathlist(void)
	-{
	- PathList *plp = pathlist;
	- PathList *next;
	-
	- while (plp)
	- {
	- next = plp->next;
	- free(plp->path);
	- free(plp);
	- plp = next;
	- }
	- pathlist = NULL;
	-}
	diff --git a/util/sdl/sound/decoders/timidity/common.h b/util/sdl/sound/decoders/timidity/common.h
	deleted file mode 100644
	index fbcce9b6..00000000
	--- a/util/sdl/sound/decoders/timidity/common.h
	+++ /dev/null
	@@ -1,32 +0,0 @@
	-/*
	-
	- TiMidity -- Experimental MIDI to WAVE converter
	- Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi>
	-
	- This program is free software; you can redistribute it and/or modify
	- it under the terms of the GNU General Public License as published by
	- the Free Software Foundation; either version 2 of the License, or
	- (at your option) any later version.
	-
	- This program is distributed in the hope that it will be useful,
	- but WITHOUT ANY WARRANTY; without even the implied warranty of
	- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
	- GNU General Public License for more details.
	-
	- You should have received a copy of the GNU General Public License
	- along with this program; if not, write to the Free Software
	- Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
	-
	-
	- common.h
	-*/
	-
	-typedef struct {
	- char *path;
	- void *next;
	-} PathList;
	-
	-extern SDL_RWops open_file(char name);
	-extern void add_to_pathlist(char *s);
	-extern void *safe_malloc(size_t count);
	-extern void free_pathlist(void);
	diff --git a/util/sdl/sound/decoders/timidity/dls1.h b/util/sdl/sound/decoders/timidity/dls1.h
	deleted file mode 100644
	index abc2075a..00000000
	--- a/util/sdl/sound/decoders/timidity/dls1.h
	+++ /dev/null
	@@ -1,266 +0,0 @@
	-/*==========================================================================;
	-//
	-// dls1.h
	-//
	-//
	-// Description:
	-//
	-// Interface defines and structures for the Instrument Collection Form
	-// RIFF DLS.
	-//
	-//
	-// Written by Sonic Foundry 1996. Released for public use.
	-//
	-//=========================================================================*/
	-
	-#ifndef _INC_DLS1
	-#define _INC_DLS1
	-
	-/*//////////////////////////////////////////////////////////////////////////
	-//
	-//
	-// Layout of an instrument collection:
	-//
	-//
	-// RIFF [] 'DLS ' [dlid,colh,INSTLIST,WAVEPOOL,INFOLIST]
	-//
	-// INSTLIST
	-// LIST [] 'lins'
	-// LIST [] 'ins ' [dlid,insh,RGNLIST,ARTLIST,INFOLIST]
	-// LIST [] 'ins ' [dlid,insh,RGNLIST,ARTLIST,INFOLIST]
	-// LIST [] 'ins ' [dlid,insh,RGNLIST,ARTLIST,INFOLIST]
	-//
	-// RGNLIST
	-// LIST [] 'lrgn'
	-// LIST [] 'rgn ' [rgnh,wsmp,wlnk,ARTLIST]
	-// LIST [] 'rgn ' [rgnh,wsmp,wlnk,ARTLIST]
	-// LIST [] 'rgn ' [rgnh,wsmp,wlnk,ARTLIST]
	-//
	-// ARTLIST
	-// LIST [] 'lart'
	-// 'art1' level 1 Articulation connection graph
	-// 'art2' level 2 Articulation connection graph
	-// '3rd1' Possible 3rd party articulation structure 1
	-// '3rd2' Possible 3rd party articulation structure 2 .... and so on
	-//
	-// WAVEPOOL
	-// ptbl [] [pool table]
	-// LIST [] 'wvpl'
	-// [path],
	-// [path],
	-// LIST [] 'wave' [dlid,RIFFWAVE]
	-// LIST [] 'wave' [dlid,RIFFWAVE]
	-// LIST [] 'wave' [dlid,RIFFWAVE]
	-// LIST [] 'wave' [dlid,RIFFWAVE]
	-// LIST [] 'wave' [dlid,RIFFWAVE]
	-//
	-// INFOLIST
	-// LIST [] 'INFO'
	-// 'icmt' 'One of those crazy comments.'
	-// 'icop' 'Copyright (C) 1996 Sonic Foundry'
	-//
	-/////////////////////////////////////////////////////////////////////////*/
	-
	-
	-/*/////////////////////////////////////////////////////////////////////////
	-// FOURCC's used in the DLS file
	-/////////////////////////////////////////////////////////////////////////*/
	-
	-#define FOURCC_DLS mmioFOURCC('D','L','S',' ')
	-#define FOURCC_DLID mmioFOURCC('d','l','i','d')
	-#define FOURCC_COLH mmioFOURCC('c','o','l','h')
	-#define FOURCC_WVPL mmioFOURCC('w','v','p','l')
	-#define FOURCC_PTBL mmioFOURCC('p','t','b','l')
	-#define FOURCC_PATH mmioFOURCC('p','a','t','h')
	-#define FOURCC_wave mmioFOURCC('w','a','v','e')
	-#define FOURCC_LINS mmioFOURCC('l','i','n','s')
	-#define FOURCC_INS mmioFOURCC('i','n','s',' ')
	-#define FOURCC_INSH mmioFOURCC('i','n','s','h')
	-#define FOURCC_LRGN mmioFOURCC('l','r','g','n')
	-#define FOURCC_RGN mmioFOURCC('r','g','n',' ')
	-#define FOURCC_RGNH mmioFOURCC('r','g','n','h')
	-#define FOURCC_LART mmioFOURCC('l','a','r','t')
	-#define FOURCC_ART1 mmioFOURCC('a','r','t','1')
	-#define FOURCC_WLNK mmioFOURCC('w','l','n','k')
	-#define FOURCC_WSMP mmioFOURCC('w','s','m','p')
	-#define FOURCC_VERS mmioFOURCC('v','e','r','s')
	-
	-/*/////////////////////////////////////////////////////////////////////////
	-// Articulation connection graph definitions
	-/////////////////////////////////////////////////////////////////////////*/
	-
	-/* Generic Sources */
	-#define CONN_SRC_NONE 0x0000
	-#define CONN_SRC_LFO 0x0001
	-#define CONN_SRC_KEYONVELOCITY 0x0002
	-#define CONN_SRC_KEYNUMBER 0x0003
	-#define CONN_SRC_EG1 0x0004
	-#define CONN_SRC_EG2 0x0005
	-#define CONN_SRC_PITCHWHEEL 0x0006
	-
	-/* Midi Controllers 0-127 */
	-#define CONN_SRC_CC1 0x0081
	-#define CONN_SRC_CC7 0x0087
	-#define CONN_SRC_CC10 0x008a
	-#define CONN_SRC_CC11 0x008b
	-
	-/* Generic Destinations */
	-#define CONN_DST_NONE 0x0000
	-#define CONN_DST_ATTENUATION 0x0001
	-#define CONN_DST_PITCH 0x0003
	-#define CONN_DST_PAN 0x0004
	-
	-/* LFO Destinations */
	-#define CONN_DST_LFO_FREQUENCY 0x0104
	-#define CONN_DST_LFO_STARTDELAY 0x0105
	-
	-/* EG1 Destinations */
	-#define CONN_DST_EG1_ATTACKTIME 0x0206
	-#define CONN_DST_EG1_DECAYTIME 0x0207
	-#define CONN_DST_EG1_RELEASETIME 0x0209
	-#define CONN_DST_EG1_SUSTAINLEVEL 0x020a
	-
	-/* EG2 Destinations */
	-#define CONN_DST_EG2_ATTACKTIME 0x030a
	-#define CONN_DST_EG2_DECAYTIME 0x030b
	-#define CONN_DST_EG2_RELEASETIME 0x030d
	-#define CONN_DST_EG2_SUSTAINLEVEL 0x030e
	-
	-#define CONN_TRN_NONE 0x0000
	-#define CONN_TRN_CONCAVE 0x0001
	-
	-typedef struct _DLSID {
	- ULONG ulData1;
	- USHORT usData2;
	- USHORT usData3;
	- BYTE abData4[8];
	-} DLSID, FAR *LPDLSID;
	-
	-typedef struct _DLSVERSION {
	- DWORD dwVersionMS;
	- DWORD dwVersionLS;
	-} DLSVERSION, FAR *LPDLSVERSION;
	-
	-
	-typedef struct _CONNECTION {
	- USHORT usSource;
	- USHORT usControl;
	- USHORT usDestination;
	- USHORT usTransform;
	- LONG lScale;
	-} CONNECTION, FAR *LPCONNECTION;
	-
	-
	-/* Level 1 Articulation Data */
	-
	-typedef struct _CONNECTIONLIST {
	- ULONG cbSize; /* size of the connection list structure */
	- ULONG cConnections; /* count of connections in the list */
	-} CONNECTIONLIST, FAR *LPCONNECTIONLIST;
	-
	-
	-
	-/*/////////////////////////////////////////////////////////////////////////
	-// Generic type defines for regions and instruments
	-/////////////////////////////////////////////////////////////////////////*/
	-
	-typedef struct _RGNRANGE {
	- USHORT usLow;
	- USHORT usHigh;
	-} RGNRANGE, FAR * LPRGNRANGE;
	-
	-#define F_INSTRUMENT_DRUMS 0x80000000
	-
	-typedef struct _MIDILOCALE {
	- ULONG ulBank;
	- ULONG ulInstrument;
	-} MIDILOCALE, FAR *LPMIDILOCALE;
	-
	-/*/////////////////////////////////////////////////////////////////////////
	-// Header structures found in an DLS file for collection, instruments, and
	-// regions.
	-/////////////////////////////////////////////////////////////////////////*/
	-
	-#define F_RGN_OPTION_SELFNONEXCLUSIVE 0x0001
	-
	-typedef struct _RGNHEADER {
	- RGNRANGE RangeKey; /* Key range */
	- RGNRANGE RangeVelocity; /* Velocity Range */
	- USHORT fusOptions; /* Synthesis options for this range */
	- USHORT usKeyGroup; /* Key grouping for non simultaneous play */
	- /* 0 = no group, 1 up is group */
	- /* for Level 1 only groups 1-15 are allowed */
	-} RGNHEADER, FAR *LPRGNHEADER;
	-
	-typedef struct _INSTHEADER {
	- ULONG cRegions; /* Count of regions in this instrument */
	- MIDILOCALE Locale; /* Intended MIDI locale of this instrument */
	-} INSTHEADER, FAR *LPINSTHEADER;
	-
	-typedef struct _DLSHEADER {
	- ULONG cInstruments; /* Count of instruments in the collection */
	-} DLSHEADER, FAR *LPDLSHEADER;
	-
	-/*////////////////////////////////////////////////////////////////////////////
	-// definitions for the Wave link structure
	-////////////////////////////////////////////////////////////////////////////*/
	-
	-/* ** For level 1 only WAVELINK_CHANNEL_MONO is valid ** */
	-/* ulChannel allows for up to 32 channels of audio with each bit position */
	-/* specifiying a channel of playback */
	-
	-#define WAVELINK_CHANNEL_LEFT 0x0001l
	-#define WAVELINK_CHANNEL_RIGHT 0x0002l
	-
	-#define F_WAVELINK_PHASE_MASTER 0x0001
	-
	-typedef struct _WAVELINK { /* any paths or links are stored right after struct */
	- USHORT fusOptions; /* options flags for this wave */
	- USHORT usPhaseGroup; /* Phase grouping for locking channels */
	- ULONG ulChannel; /* channel placement */
	- ULONG ulTableIndex; /* index into the wave pool table, 0 based */
	-} WAVELINK, FAR *LPWAVELINK;
	-
	-#define POOL_CUE_NULL 0xffffffffl
	-
	-typedef struct _POOLCUE {
	- ULONG ulOffset; /* Offset to the entry in the list */
	-} POOLCUE, FAR *LPPOOLCUE;
	-
	-typedef struct _POOLTABLE {
	- ULONG cbSize; /* size of the pool table structure */
	- ULONG cCues; /* count of cues in the list */
	-} POOLTABLE, FAR *LPPOOLTABLE;
	-
	-/*////////////////////////////////////////////////////////////////////////////
	-// Structures for the "wsmp" chunk
	-////////////////////////////////////////////////////////////////////////////*/
	-
	-#define F_WSMP_NO_TRUNCATION 0x0001l
	-#define F_WSMP_NO_COMPRESSION 0x0002l
	-
	-
	-typedef struct _rwsmp {
	- ULONG cbSize;
	- USHORT usUnityNote; /* MIDI Unity Playback Note */
	- SHORT sFineTune; /* Fine Tune in log tuning */
	- LONG lAttenuation; /* Overall Attenuation to be applied to data */
	- ULONG fulOptions; /* Flag options */
	- ULONG cSampleLoops; /* Count of Sample loops, 0 loops is one shot */
	-} WSMPL, FAR *LPWSMPL;
	-
	-
	-/* This loop type is a normal forward playing loop which is continually */
	-/* played until the envelope reaches an off threshold in the release */
	-/* portion of the volume envelope */
	-
	-#define WLOOP_TYPE_FORWARD 0
	-
	-typedef struct _rloop {
	- ULONG cbSize;
	- ULONG ulType; /* Loop Type */
	- ULONG ulStart; /* Start of loop in samples */
	- ULONG ulLength; /* Length of loop in samples */
	-} WLOOP, FAR *LPWLOOP;
	-
	-#endif /_INC_DLS1 /
	diff --git a/util/sdl/sound/decoders/timidity/dls2.h b/util/sdl/sound/decoders/timidity/dls2.h
	deleted file mode 100644
	index 30cec23a..00000000
	--- a/util/sdl/sound/decoders/timidity/dls2.h
	+++ /dev/null
	@@ -1,130 +0,0 @@
	-/*
	-
	- dls2.h
	-
	- Description:
	-
	- Interface defines and structures for the DLS2 extensions of DLS.
	-
	-
	- Written by Microsoft 1998. Released for public use.
	-
	-*/
	-
	-#ifndef _INC_DLS2
	-#define _INC_DLS2
	-
	-/*
	- FOURCC's used in the DLS2 file, in addition to DLS1 chunks
	-*/
	-
	-#define FOURCC_RGN2 mmioFOURCC('r','g','n','2')
	-#define FOURCC_LAR2 mmioFOURCC('l','a','r','2')
	-#define FOURCC_ART2 mmioFOURCC('a','r','t','2')
	-#define FOURCC_CDL mmioFOURCC('c','d','l',' ')
	-#define FOURCC_DLID mmioFOURCC('d','l','i','d')
	-
	-/*
	- Articulation connection graph definitions. These are in addition to
	- the definitions in the DLS1 header.
	-*/
	-
	-/* Generic Sources (in addition to DLS1 sources. */
	-#define CONN_SRC_POLYPRESSURE 0x0007 /* Polyphonic Pressure */
	-#define CONN_SRC_CHANNELPRESSURE 0x0008 /* Channel Pressure */
	-#define CONN_SRC_VIBRATO 0x0009 /* Vibrato LFO */
	-#define CONN_SRC_MONOPRESSURE 0x000a /* MIDI Mono pressure */
	-
	-
	-/* Midi Controllers */
	-#define CONN_SRC_CC91 0x00db /* Reverb Send */
	-#define CONN_SRC_CC93 0x00dd /* Chorus Send */
	-
	-
	-/* Generic Destinations */
	-#define CONN_DST_GAIN 0x0001 /* Same as CONN_DST_ ATTENUATION, but more appropriate terminology. */
	-#define CONN_DST_KEYNUMBER 0x0005 /* Key Number Generator */
	-
	-/* Audio Channel Output Destinations */
	-#define CONN_DST_LEFT 0x0010 /* Left Channel Send */
	-#define CONN_DST_RIGHT 0x0011 /* Right Channel Send */
	-#define CONN_DST_CENTER 0x0012 /* Center Channel Send */
	-#define CONN_DST_LEFTREAR 0x0013 /* Left Rear Channel Send */
	-#define CONN_DST_RIGHTREAR 0x0014 /* Right Rear Channel Send */
	-#define CONN_DST_LFE_CHANNEL 0x0015 /* LFE Channel Send */
	-#define CONN_DST_CHORUS 0x0080 /* Chorus Send */
	-#define CONN_DST_REVERB 0x0081 /* Reverb Send */
	-
	-/* Vibrato LFO Destinations */
	-#define CONN_DST_VIB_FREQUENCY 0x0114 /* Vibrato Frequency */
	-#define CONN_DST_VIB_STARTDELAY 0x0115 /* Vibrato Start Delay */
	-
	-/* EG1 Destinations */
	-#define CONN_DST_EG1_DELAYTIME 0x020B /* EG1 Delay Time */
	-#define CONN_DST_EG1_HOLDTIME 0x020C /* EG1 Hold Time */
	-#define CONN_DST_EG1_SHUTDOWNTIME 0x020D /* EG1 Shutdown Time */
	-
	-
	-/* EG2 Destinations */
	-#define CONN_DST_EG2_DELAYTIME 0x030F /* EG2 Delay Time */
	-#define CONN_DST_EG2_HOLDTIME 0x0310 /* EG2 Hold Time */
	-
	-
	-/* Filter Destinations */
	-#define CONN_DST_FILTER_CUTOFF 0x0500 /* Filter Cutoff Frequency */
	-#define CONN_DST_FILTER_Q 0x0501 /* Filter Resonance */
	-
	-
	-/* Transforms */
	-#define CONN_TRN_CONVEX 0x0002 /* Convex Transform */
	-#define CONN_TRN_SWITCH 0x0003 /* Switch Transform */
	-
	-
	-/* Conditional chunk operators */
	- #define DLS_CDL_AND 0x0001 /* X = X & Y */
	- #define DLS_CDL_OR 0x0002 /* X = X \| Y */
	- #define DLS_CDL_XOR 0x0003 /* X = X ^ Y */
	- #define DLS_CDL_ADD 0x0004 /* X = X + Y */
	- #define DLS_CDL_SUBTRACT 0x0005 /* X = X - Y */
	- #define DLS_CDL_MULTIPLY 0x0006 /* X = X * Y */
	- #define DLS_CDL_DIVIDE 0x0007 /* X = X / Y */
	- #define DLS_CDL_LOGICAL_AND 0x0008 /* X = X && Y */
	- #define DLS_CDL_LOGICAL_OR 0x0009 /* X = X \|\| Y */
	- #define DLS_CDL_LT 0x000A /* X = (X < Y) */
	- #define DLS_CDL_LE 0x000B /* X = (X <= Y) */
	- #define DLS_CDL_GT 0x000C /* X = (X > Y) */
	- #define DLS_CDL_GE 0x000D /* X = (X >= Y) */
	- #define DLS_CDL_EQ 0x000E /* X = (X == Y) */
	- #define DLS_CDL_NOT 0x000F /* X = !X */
	- #define DLS_CDL_CONST 0x0010 /* 32-bit constant */
	- #define DLS_CDL_QUERY 0x0011 /* 32-bit value returned from query */
	- #define DLS_CDL_QUERYSUPPORTED 0x0012 /* Test to see if query is supported by synth */
	-
	-/*
	- Loop and release
	-*/
	-
	-#define WLOOP_TYPE_RELEASE 1
	-
	-/*
	- WaveLink chunk <wlnk-ck>
	-*/
	-
	-#define F_WAVELINK_MULTICHANNEL 0x0002
	-
	-
	-/*
	- DLSID queries for <cdl-ck>
	-*/
	-
	-DEFINE_GUID(DLSID_GMInHardware, 0x178f2f24, 0xc364, 0x11d1, 0xa7, 0x60, 0x00, 0x00, 0xf8, 0x75, 0xac, 0x12);
	-DEFINE_GUID(DLSID_GSInHardware, 0x178f2f25, 0xc364, 0x11d1, 0xa7, 0x60, 0x00, 0x00, 0xf8, 0x75, 0xac, 0x12);
	-DEFINE_GUID(DLSID_XGInHardware, 0x178f2f26, 0xc364, 0x11d1, 0xa7, 0x60, 0x00, 0x00, 0xf8, 0x75, 0xac, 0x12);
	-DEFINE_GUID(DLSID_SupportsDLS1, 0x178f2f27, 0xc364, 0x11d1, 0xa7, 0x60, 0x00, 0x00, 0xf8, 0x75, 0xac, 0x12);
	-DEFINE_GUID(DLSID_SupportsDLS2, 0xf14599e5, 0x4689, 0x11d2, 0xaf, 0xa6, 0x0, 0xaa, 0x0, 0x24, 0xd8, 0xb6);
	-DEFINE_GUID(DLSID_SampleMemorySize, 0x178f2f28, 0xc364, 0x11d1, 0xa7, 0x60, 0x00, 0x00, 0xf8, 0x75, 0xac, 0x12);
	-DEFINE_GUID(DLSID_ManufacturersID, 0xb03e1181, 0x8095, 0x11d2, 0xa1, 0xef, 0x0, 0x60, 0x8, 0x33, 0xdb, 0xd8);
	-DEFINE_GUID(DLSID_ProductID, 0xb03e1182, 0x8095, 0x11d2, 0xa1, 0xef, 0x0, 0x60, 0x8, 0x33, 0xdb, 0xd8);
	-DEFINE_GUID(DLSID_SamplePlaybackRate, 0x2a91f713, 0xa4bf, 0x11d2, 0xbb, 0xdf, 0x0, 0x60, 0x8, 0x33, 0xdb, 0xd8);
	-
	-#endif /* _INC_DLS2 */
	diff --git a/util/sdl/sound/decoders/timidity/instrum.c b/util/sdl/sound/decoders/timidity/instrum.c
	deleted file mode 100644
	index e46ecd96..00000000
	--- a/util/sdl/sound/decoders/timidity/instrum.c
	+++ /dev/null
	@@ -1,623 +0,0 @@
	-/*
	-
	- TiMidity -- Experimental MIDI to WAVE converter
	- Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi>
	-
	- This program is free software; you can redistribute it and/or modify
	- it under the terms of the GNU General Public License as published by
	- the Free Software Foundation; either version 2 of the License, or
	- (at your option) any later version.
	-
	- This program is distributed in the hope that it will be useful,
	- but WITHOUT ANY WARRANTY; without even the implied warranty of
	- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
	- GNU General Public License for more details.
	-
	- You should have received a copy of the GNU General Public License
	- along with this program; if not, write to the Free Software
	- Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
	-
	- instrum.c
	-
	- Code to load and unload GUS-compatible instrument patches.
	-
	-*/
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#include <stdio.h>
	-#include <string.h>
	-#include <stdlib.h>
	-
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-#include "timidity.h"
	-#include "options.h"
	-#include "common.h"
	-#include "instrum.h"
	-#include "instrum_dls.h"
	-#include "resample.h"
	-#include "tables.h"
	-
	-static void free_instrument(Instrument *ip)
	-{
	- Sample *sp;
	- int i;
	- if (!ip) return;
	- for (i=0; i<ip->samples; i++)
	- {
	- sp=&(ip->sample[i]);
	- free(sp->data);
	- }
	- free(ip->sample);
	- free(ip);
	-}
	-
	-static void free_bank(MidiSong *song, int dr, int b)
	-{
	- int i;
	- ToneBank *bank=((dr) ? song->drumset[b] : song->tonebank[b]);
	- for (i=0; i<128; i++)
	- if (bank->instrument[i])
	- {
	- /* Not that this could ever happen, of course */
	- if (bank->instrument[i] != MAGIC_LOAD_INSTRUMENT)
	- free_instrument(bank->instrument[i]);
	- bank->instrument[i]=0;
	- }
	-}
	-
	-static Sint32 convert_envelope_rate(MidiSong *song, Uint8 rate)
	-{
	- Sint32 r;
	-
	- r = 3 - ((rate >> 6) & 0x3);
	- r *= 3;
	- r = (Sint32) (rate & 0x3f) << r; /* 6.9 fixed point */
	-
	- /* 15.15 fixed point. */
	- r = ((r * 44100) / song->rate) * song->control_ratio;
	-
	-#ifdef FAST_DECAY
	- return r << 10;
	-#else
	- return r << 9;
	-#endif
	-}
	-
	-static Sint32 convert_envelope_offset(Uint8 offset)
	-{
	- /* This is not too good... Can anyone tell me what these values mean?
	- Are they GUS-style "exponential" volumes? And what does that mean? */
	-
	- /* 15.15 fixed point */
	- return offset << (7+15);
	-}
	-
	-static Sint32 convert_tremolo_sweep(MidiSong *song, Uint8 sweep)
	-{
	- if (!sweep)
	- return 0;
	-
	- return
	- ((song->control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) /
	- (song->rate * sweep);
	-}
	-
	-static Sint32 convert_vibrato_sweep(MidiSong *song, Uint8 sweep,
	- Sint32 vib_control_ratio)
	-{
	- if (!sweep)
	- return 0;
	-
	- return
	- (Sint32) (FSCALE((double) (vib_control_ratio) * SWEEP_TUNING, SWEEP_SHIFT)
	- / (double)(song->rate * sweep));
	-
	- /* this was overflowing with seashore.pat
	-
	- ((vib_control_ratio * SWEEP_TUNING) << SWEEP_SHIFT) /
	- (song->rate * sweep); */
	-}
	-
	-static Sint32 convert_tremolo_rate(MidiSong *song, Uint8 rate)
	-{
	- return
	- ((SINE_CYCLE_LENGTH * song->control_ratio * rate) << RATE_SHIFT) /
	- (TREMOLO_RATE_TUNING * song->rate);
	-}
	-
	-static Sint32 convert_vibrato_rate(MidiSong *song, Uint8 rate)
	-{
	- /* Return a suitable vibrato_control_ratio value */
	- return
	- (VIBRATO_RATE_TUNING * song->rate) /
	- (rate * 2 * VIBRATO_SAMPLE_INCREMENTS);
	-}
	-
	-static void reverse_data(Sint16 *sp, Sint32 ls, Sint32 le)
	-{
	- Sint16 s, *ep=sp+le;
	- sp+=ls;
	- le-=ls;
	- le/=2;
	- while (le--)
	- {
	- s=*sp;
	- sp++=ep;
	- *ep--=s;
	- }
	-}
	-
	-/*
	- If panning or note_to_use != -1, it will be used for all samples,
	- instead of the sample-specific values in the instrument file.
	-
	- For note_to_use, any value <0 or >127 will be forced to 0.
	-
	- For other parameters, 1 means yes, 0 means no, other values are
	- undefined.
	-
	- TODO: do reverse loops right */
	-static Instrument load_instrument(MidiSong song, char *name, int percussion,
	- int panning, int amp, int note_to_use,
	- int strip_loop, int strip_envelope,
	- int strip_tail)
	-{
	- Instrument *ip;
	- Sample *sp;
	- SDL_RWops *rw;
	- char tmp[1024];
	- int i,j,noluck=0;
	- static char *patch_ext[] = PATCH_EXT_LIST;
	-
	- if (!name) return 0;
	-
	- /* Open patch file */
	- if ((rw=open_file(name)) == NULL)
	- {
	- noluck=1;
	- /* Try with various extensions */
	- for (i=0; patch_ext[i]; i++)
	- {
	- if (strlen(name)+strlen(patch_ext[i])<1024)
	- {
	- strcpy(tmp, name);
	- strcat(tmp, patch_ext[i]);
	- if ((rw=open_file(tmp)) != NULL)
	- {
	- noluck=0;
	- break;
	- }
	- }
	- }
	- }
	-
	- if (noluck)
	- {
	- SNDDBG(("Instrument `%s' can't be found.\n", name));
	- return 0;
	- }
	-
	- SNDDBG(("Loading instrument %s\n", tmp));
	-
	- /* Read some headers and do cursory sanity checks. There are loads
	- of magic offsets. This could be rewritten... */
	-
	- if ((239 != SDL_RWread(rw, tmp, 1, 239)) \|\|
	- (memcmp(tmp, "GF1PATCH110\0ID#000002", 22) &&
	- memcmp(tmp, "GF1PATCH100\0ID#000002", 22))) /* don't know what the
	- differences are */
	- {
	- SNDDBG(("%s: not an instrument\n", name));
	- return 0;
	- }
	-
	- if (tmp[82] != 1 && tmp[82] != 0) /* instruments. To some patch makers,
	- 0 means 1 */
	- {
	- SNDDBG(("Can't handle patches with %d instruments\n", tmp[82]));
	- return 0;
	- }
	-
	- if (tmp[151] != 1 && tmp[151] != 0) /* layers. What's a layer? */
	- {
	- SNDDBG(("Can't handle instruments with %d layers\n", tmp[151]));
	- return 0;
	- }
	-
	- ip=safe_malloc(sizeof(Instrument));
	- ip->samples = tmp[198];
	- ip->sample = safe_malloc(sizeof(Sample) * ip->samples);
	- for (i=0; i<ip->samples; i++)
	- {
	-
	- Uint8 fractions;
	- Sint32 tmplong;
	- Uint16 tmpshort;
	- Uint8 tmpchar;
	-
	-#define READ_CHAR(thing) \
	- if (1 != SDL_RWread(rw, &tmpchar, 1, 1)) goto fail; \
	- thing = tmpchar;
	-#define READ_SHORT(thing) \
	- if (1 != SDL_RWread(rw, &tmpshort, 2, 1)) goto fail; \
	- thing = SDL_SwapLE16(tmpshort);
	-#define READ_LONG(thing) \
	- if (1 != SDL_RWread(rw, &tmplong, 4, 1)) goto fail; \
	- thing = SDL_SwapLE32(tmplong);
	-
	- SDL_RWseek(rw, 7, SEEK_CUR); /* Skip the wave name */
	-
	- if (1 != SDL_RWread(rw, &fractions, 1, 1))
	- {
	- fail:
	- SNDDBG(("Error reading sample %d\n", i));
	- for (j=0; j<i; j++)
	- free(ip->sample[j].data);
	- free(ip->sample);
	- free(ip);
	- return 0;
	- }
	-
	- sp=&(ip->sample[i]);
	-
	- READ_LONG(sp->data_length);
	- READ_LONG(sp->loop_start);
	- READ_LONG(sp->loop_end);
	- READ_SHORT(sp->sample_rate);
	- READ_LONG(sp->low_freq);
	- READ_LONG(sp->high_freq);
	- READ_LONG(sp->root_freq);
	- sp->low_vel = 0;
	- sp->high_vel = 127;
	- SDL_RWseek(rw, 2, SEEK_CUR); /* Why have a "root frequency" and then
	- * "tuning"?? */
	-
	- READ_CHAR(tmp[0]);
	-
	- if (panning==-1)
	- sp->panning = (tmp[0] * 8 + 4) & 0x7f;
	- else
	- sp->panning=(Uint8)(panning & 0x7F);
	-
	- /* envelope, tremolo, and vibrato */
	- if (18 != SDL_RWread(rw, tmp, 1, 18)) goto fail;
	-
	- if (!tmp[13] \|\| !tmp[14])
	- {
	- sp->tremolo_sweep_increment=
	- sp->tremolo_phase_increment=sp->tremolo_depth=0;
	- SNDDBG((" * no tremolo\n"));
	- }
	- else
	- {
	- sp->tremolo_sweep_increment=convert_tremolo_sweep(song, tmp[12]);
	- sp->tremolo_phase_increment=convert_tremolo_rate(song, tmp[13]);
	- sp->tremolo_depth=tmp[14];
	- SNDDBG((" * tremolo: sweep %d, phase %d, depth %d\n",
	- sp->tremolo_sweep_increment, sp->tremolo_phase_increment,
	- sp->tremolo_depth));
	- }
	-
	- if (!tmp[16] \|\| !tmp[17])
	- {
	- sp->vibrato_sweep_increment=
	- sp->vibrato_control_ratio=sp->vibrato_depth=0;
	- SNDDBG((" * no vibrato\n"));
	- }
	- else
	- {
	- sp->vibrato_control_ratio=convert_vibrato_rate(song, tmp[16]);
	- sp->vibrato_sweep_increment=
	- convert_vibrato_sweep(song, tmp[15], sp->vibrato_control_ratio);
	- sp->vibrato_depth=tmp[17];
	- SNDDBG((" * vibrato: sweep %d, ctl %d, depth %d\n",
	- sp->vibrato_sweep_increment, sp->vibrato_control_ratio,
	- sp->vibrato_depth));
	-
	- }
	-
	- READ_CHAR(sp->modes);
	-
	- SDL_RWseek(rw, 40, SEEK_CUR); /* skip the useless scale frequency, scale
	- factor (what's it mean?), and reserved
	- space */
	-
	- /* Mark this as a fixed-pitch instrument if such a deed is desired. */
	- if (note_to_use!=-1)
	- sp->note_to_use=(Uint8)(note_to_use);
	- else
	- sp->note_to_use=0;
	-
	- /* seashore.pat in the Midia patch set has no Sustain. I don't
	- understand why, and fixing it by adding the Sustain flag to
	- all looped patches probably breaks something else. We do it
	- anyway. */
	-
	- if (sp->modes & MODES_LOOPING)
	- sp->modes \|= MODES_SUSTAIN;
	-
	- /* Strip any loops and envelopes we're permitted to */
	- if ((strip_loop==1) &&
	- (sp->modes & (MODES_SUSTAIN \| MODES_LOOPING \|
	- MODES_PINGPONG \| MODES_REVERSE)))
	- {
	- SNDDBG((" - Removing loop and/or sustain\n"));
	- sp->modes &=~(MODES_SUSTAIN \| MODES_LOOPING \|
	- MODES_PINGPONG \| MODES_REVERSE);
	- }
	-
	- if (strip_envelope==1)
	- {
	- if (sp->modes & MODES_ENVELOPE)
	- SNDDBG((" - Removing envelope\n"));
	- sp->modes &= ~MODES_ENVELOPE;
	- }
	- else if (strip_envelope != 0)
	- {
	- /* Have to make a guess. */
	- if (!(sp->modes & (MODES_LOOPING \| MODES_PINGPONG \| MODES_REVERSE)))
	- {
	- /* No loop? Then what's there to sustain? No envelope needed
	- either... */
	- sp->modes &= ~(MODES_SUSTAIN\|MODES_ENVELOPE);
	- SNDDBG((" - No loop, removing sustain and envelope\n"));
	- }
	- else if (!memcmp(tmp, "??????", 6) \|\| tmp[11] >= 100)
	- {
	- /* Envelope rates all maxed out? Envelope end at a high "offset"?
	- That's a weird envelope. Take it out. */
	- sp->modes &= ~MODES_ENVELOPE;
	- SNDDBG((" - Weirdness, removing envelope\n"));
	- }
	- else if (!(sp->modes & MODES_SUSTAIN))
	- {
	- /* No sustain? Then no envelope. I don't know if this is
	- justified, but patches without sustain usually don't need the
	- envelope either... at least the Gravis ones. They're mostly
	- drums. I think. */
	- sp->modes &= ~MODES_ENVELOPE;
	- SNDDBG((" - No sustain, removing envelope\n"));
	- }
	- }
	-
	- for (j=0; j<6; j++)
	- {
	- sp->envelope_rate[j]=
	- convert_envelope_rate(song, tmp[j]);
	- sp->envelope_offset[j]=
	- convert_envelope_offset(tmp[6+j]);
	- }
	-
	- /* Then read the sample data */
	- sp->data = safe_malloc(sp->data_length);
	- if (1 != SDL_RWread(rw, sp->data, sp->data_length, 1))
	- goto fail;
	-
	- if (!(sp->modes & MODES_16BIT)) /* convert to 16-bit data */
	- {
	- Sint32 i=sp->data_length;
	- Uint8 cp=(Uint8 )(sp->data);
	- Uint16 tmp,new;
	- tmp=new=safe_malloc(sp->data_length*2);
	- while (i--)
	- tmp++ = (Uint16)(cp++) << 8;
	- cp=(Uint8 *)(sp->data);
	- sp->data = (sample_t *)new;
	- free(cp);
	- sp->data_length *= 2;
	- sp->loop_start *= 2;
	- sp->loop_end *= 2;
	- }
	-#if SDL_BYTEORDER == SDL_BIG_ENDIAN
	- else
	- /* convert to machine byte order */
	- {
	- Sint32 i=sp->data_length/2;
	- Sint16 tmp=(Sint16 )sp->data,s;
	- while (i--)
	- {
	- s=SDL_SwapLE16(*tmp);
	- *tmp++=s;
	- }
	- }
	-#endif
	-
	- if (sp->modes & MODES_UNSIGNED) /* convert to signed data */
	- {
	- Sint32 i=sp->data_length/2;
	- Sint16 tmp=(Sint16 )sp->data;
	- while (i--)
	- *tmp++ ^= 0x8000;
	- }
	-
	- /* Reverse reverse loops and pass them off as normal loops */
	- if (sp->modes & MODES_REVERSE)
	- {
	- Sint32 t;
	- /* The GUS apparently plays reverse loops by reversing the
	- whole sample. We do the same because the GUS does not SUCK. */
	-
	- SNDDBG(("Reverse loop in %s\n", name));
	- reverse_data((Sint16 *)sp->data, 0, sp->data_length/2);
	-
	- t=sp->loop_start;
	- sp->loop_start=sp->data_length - sp->loop_end;
	- sp->loop_end=sp->data_length - t;
	-
	- sp->modes &= ~MODES_REVERSE;
	- sp->modes \|= MODES_LOOPING; /* just in case */
	- }
	-
	-#ifdef ADJUST_SAMPLE_VOLUMES
	- if (amp!=-1)
	- sp->volume=(float)((amp) / 100.0);
	- else
	- {
	- /* Try to determine a volume scaling factor for the sample.
	- This is a very crude adjustment, but things sound more
	- balanced with it. Still, this should be a runtime option. */
	- Sint32 i=sp->data_length/2;
	- Sint16 maxamp=0,a;
	- Sint16 tmp=(Sint16 )sp->data;
	- while (i--)
	- {
	- a=*tmp++;
	- if (a<0) a=-a;
	- if (a>maxamp)
	- maxamp=a;
	- }
	- sp->volume=(float)(32768.0 / maxamp);
	- SNDDBG((" * volume comp: %f\n", sp->volume));
	- }
	-#else
	- if (amp!=-1)
	- sp->volume=(double)(amp) / 100.0;
	- else
	- sp->volume=1.0;
	-#endif
	-
	- sp->data_length /= 2; /* These are in bytes. Convert into samples. */
	- sp->loop_start /= 2;
	- sp->loop_end /= 2;
	-
	- /* Then fractional samples */
	- sp->data_length <<= FRACTION_BITS;
	- sp->loop_start <<= FRACTION_BITS;
	- sp->loop_end <<= FRACTION_BITS;
	-
	- /* Adjust for fractional loop points. This is a guess. Does anyone
	- know what "fractions" really stands for? */
	- sp->loop_start \|=
	- (fractions & 0x0F) << (FRACTION_BITS-4);
	- sp->loop_end \|=
	- ((fractions>>4) & 0x0F) << (FRACTION_BITS-4);
	-
	- /* If this instrument will always be played on the same note,
	- and it's not looped, we can resample it now. */
	- if (sp->note_to_use && !(sp->modes & MODES_LOOPING))
	- pre_resample(song, sp);
	-
	- if (strip_tail==1)
	- {
	- /* Let's not really, just say we did. */
	- SNDDBG((" - Stripping tail\n"));
	- sp->data_length = sp->loop_end;
	- }
	- }
	-
	- SDL_RWclose(rw);
	- return ip;
	-}
	-
	-static int fill_bank(MidiSong *song, int dr, int b)
	-{
	- int i, errors=0;
	- ToneBank *bank=((dr) ? song->drumset[b] : song->tonebank[b]);
	- if (!bank)
	- {
	- SNDDBG(("Huh. Tried to load instruments in non-existent %s %d\n",
	- (dr) ? "drumset" : "tone bank", b));
	- return 0;
	- }
	- for (i=0; i<128; i++)
	- {
	- if (bank->instrument[i]==MAGIC_LOAD_INSTRUMENT)
	- {
	- bank->instrument[i]=load_instrument_dls(song, dr, b, i);
	- if (bank->instrument[i])
	- {
	- continue;
	- }
	- if (!(bank->tone[i].name))
	- {
	- SNDDBG(("No instrument mapped to %s %d, program %d%s\n",
	- (dr)? "drum set" : "tone bank", b, i,
	- (b!=0) ? "" : " - this instrument will not be heard"));
	- if (b!=0)
	- {
	- /* Mark the corresponding instrument in the default
	- bank / drumset for loading (if it isn't already) */
	- if (!dr)
	- {
	- if (!(song->tonebank[0]->instrument[i]))
	- song->tonebank[0]->instrument[i] =
	- MAGIC_LOAD_INSTRUMENT;
	- }
	- else
	- {
	- if (!(song->drumset[0]->instrument[i]))
	- song->drumset[0]->instrument[i] =
	- MAGIC_LOAD_INSTRUMENT;
	- }
	- }
	- bank->instrument[i] = 0;
	- errors++;
	- }
	- else if (!(bank->instrument[i] =
	- load_instrument(song,
	- bank->tone[i].name,
	- (dr) ? 1 : 0,
	- bank->tone[i].pan,
	- bank->tone[i].amp,
	- (bank->tone[i].note!=-1) ?
	- bank->tone[i].note :
	- ((dr) ? i : -1),
	- (bank->tone[i].strip_loop!=-1) ?
	- bank->tone[i].strip_loop :
	- ((dr) ? 1 : -1),
	- (bank->tone[i].strip_envelope != -1) ?
	- bank->tone[i].strip_envelope :
	- ((dr) ? 1 : -1),
	- bank->tone[i].strip_tail )))
	- {
	- SNDDBG(("Couldn't load instrument %s (%s %d, program %d)\n",
	- bank->tone[i].name,
	- (dr)? "drum set" : "tone bank", b, i));
	- errors++;
	- }
	- }
	- }
	- return errors;
	-}
	-
	-int load_missing_instruments(MidiSong *song)
	-{
	- int i=128,errors=0;
	- while (i--)
	- {
	- if (song->tonebank[i])
	- errors+=fill_bank(song,0,i);
	- if (song->drumset[i])
	- errors+=fill_bank(song,1,i);
	- }
	- return errors;
	-}
	-
	-void free_instruments(MidiSong *song)
	-{
	- int i=128;
	- while(i--)
	- {
	- if (song->tonebank[i])
	- free_bank(song, 0, i);
	- if (song->drumset[i])
	- free_bank(song, 1, i);
	- }
	-}
	-
	-int set_default_instrument(MidiSong song, char name)
	-{
	- Instrument *ip;
	- if (!(ip=load_instrument(song, name, 0, -1, -1, -1, 0, 0, 0)))
	- return -1;
	- song->default_instrument = ip;
	- song->default_program = SPECIAL_PROGRAM;
	- return 0;
	-}
	diff --git a/util/sdl/sound/decoders/timidity/instrum.h b/util/sdl/sound/decoders/timidity/instrum.h
	deleted file mode 100644
	index e46d2b23..00000000
	--- a/util/sdl/sound/decoders/timidity/instrum.h
	+++ /dev/null
	@@ -1,41 +0,0 @@
	-/*
	-
	- TiMidity -- Experimental MIDI to WAVE converter
	- Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi>
	-
	- This program is free software; you can redistribute it and/or modify
	- it under the terms of the GNU General Public License as published by
	- the Free Software Foundation; either version 2 of the License, or
	- (at your option) any later version.
	-
	- This program is distributed in the hope that it will be useful,
	- but WITHOUT ANY WARRANTY; without even the implied warranty of
	- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
	- GNU General Public License for more details.
	-
	- You should have received a copy of the GNU General Public License
	- along with this program; if not, write to the Free Software
	- Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
	-
	- instrum.h
	-
	- */
	-
	-/* Bits in modes: */
	-#define MODES_16BIT (1<<0)
	-#define MODES_UNSIGNED (1<<1)
	-#define MODES_LOOPING (1<<2)
	-#define MODES_PINGPONG (1<<3)
	-#define MODES_REVERSE (1<<4)
	-#define MODES_SUSTAIN (1<<5)
	-#define MODES_ENVELOPE (1<<6)
	-
	-/* A hack to delay instrument loading until after reading the
	- entire MIDI file. */
	-#define MAGIC_LOAD_INSTRUMENT ((Instrument *) (-1))
	-
	-#define SPECIAL_PROGRAM -1
	-
	-extern int load_missing_instruments(MidiSong *song);
	-extern void free_instruments(MidiSong *song);
	-extern int set_default_instrument(MidiSong song, char name);
	diff --git a/util/sdl/sound/decoders/timidity/instrum_dls.c b/util/sdl/sound/decoders/timidity/instrum_dls.c
	deleted file mode 100644
	index 7b8e15c9..00000000
	--- a/util/sdl/sound/decoders/timidity/instrum_dls.c
	+++ /dev/null
	@@ -1,1269 +0,0 @@
	-/*
	-
	- TiMidity -- Experimental MIDI to WAVE converter
	- Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi>
	-
	- This program is free software; you can redistribute it and/or modify
	- it under the terms of the GNU General Public License as published by
	- the Free Software Foundation; either version 2 of the License, or
	- (at your option) any later version.
	-
	- This program is distributed in the hope that it will be useful,
	- but WITHOUT ANY WARRANTY; without even the implied warranty of
	- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
	- GNU General Public License for more details.
	-
	- You should have received a copy of the GNU General Public License
	- along with this program; if not, write to the Free Software
	- Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
	-
	- instrum.h
	-
	- */
	-
	-#include <stdlib.h>
	-#include <string.h>
	-
	-#include "SDL.h"
	-#include "SDL_endian.h"
	-#include "SDL_rwops.h"
	-
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-#include "timidity.h"
	-#include "options.h"
	-#include "instrum.h"
	-#include "tables.h"
	-#include "common.h"
	-
	-/-------------------------------------------------------------------------/
	-/* * * * * * * * * * * * * * * * * load_riff.h * * * * * * * * * * * * * * */
	-/-------------------------------------------------------------------------/
	-typedef struct _RIFF_Chunk {
	- Uint32 magic;
	- Uint32 length;
	- Uint32 subtype;
	- Uint8 *data;
	- struct _RIFF_Chunk *child;
	- struct _RIFF_Chunk *next;
	-} RIFF_Chunk;
	-
	-extern DECLSPEC RIFF_Chunk* SDLCALL LoadRIFF(SDL_RWops *src);
	-extern DECLSPEC void SDLCALL FreeRIFF(RIFF_Chunk *chunk);
	-extern DECLSPEC void SDLCALL PrintRIFF(RIFF_Chunk *chunk, int level);
	-/* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * */
	-
	-/-------------------------------------------------------------------------/
	-/* * * * * * * * * * * * * * * * * load_riff.c * * * * * * * * * * * * * * */
	-/-------------------------------------------------------------------------/
	-#define RIFF 0x46464952 /* "RIFF" */
	-#define LIST 0x5453494c /* "LIST" */
	-
	-static RIFF_Chunk *AllocRIFFChunk()
	-{
	- RIFF_Chunk chunk = (RIFF_Chunk )malloc(sizeof(*chunk));
	- if ( !chunk ) {
	- __Sound_SetError(ERR_OUT_OF_MEMORY);
	- return NULL;
	- }
	- memset(chunk, 0, sizeof(*chunk));
	- return chunk;
	-}
	-
	-static void FreeRIFFChunk(RIFF_Chunk *chunk)
	-{
	- if ( chunk->child ) {
	- FreeRIFFChunk(chunk->child);
	- }
	- if ( chunk->next ) {
	- FreeRIFFChunk(chunk->next);
	- }
	- free(chunk);
	-}
	-
	-static int ChunkHasSubType(Uint32 magic)
	-{
	- static Uint32 chunk_list[] = {
	- RIFF, LIST
	- };
	- int i;
	- for ( i = 0; i < SDL_TABLESIZE(chunk_list); ++i ) {
	- if ( magic == chunk_list[i] ) {
	- return 1;
	- }
	- }
	- return 0;
	-}
	-
	-static int ChunkHasSubChunks(Uint32 magic)
	-{
	- static Uint32 chunk_list[] = {
	- RIFF, LIST
	- };
	- int i;
	- for ( i = 0; i < SDL_TABLESIZE(chunk_list); ++i ) {
	- if ( magic == chunk_list[i] ) {
	- return 1;
	- }
	- }
	- return 0;
	-}
	-
	-static void LoadSubChunks(RIFF_Chunk chunk, Uint8 data, Uint32 left)
	-{
	- Uint8 *subchunkData;
	- Uint32 subchunkDataLen;
	-
	- while ( left > 8 ) {
	- RIFF_Chunk *child = AllocRIFFChunk();
	- RIFF_Chunk next, prev = NULL;
	- for ( next = chunk->child; next; next = next->next ) {
	- prev = next;
	- }
	- if ( prev ) {
	- prev->next = child;
	- } else {
	- chunk->child = child;
	- }
	-
	- child->magic = (data[0] << 0) \|
	- (data[1] << 8) \|
	- (data[2] << 16) \|
	- (data[3] << 24);
	- data += 4;
	- left -= 4;
	- child->length = (data[0] << 0) \|
	- (data[1] << 8) \|
	- (data[2] << 16) \|
	- (data[3] << 24);
	- data += 4;
	- left -= 4;
	- child->data = data;
	-
	- if ( child->length > left ) {
	- child->length = left;
	- }
	-
	- subchunkData = child->data;
	- subchunkDataLen = child->length;
	- if ( ChunkHasSubType(child->magic) && subchunkDataLen >= 4 ) {
	- child->subtype = (subchunkData[0] << 0) \|
	- (subchunkData[1] << 8) \|
	- (subchunkData[2] << 16) \|
	- (subchunkData[3] << 24);
	- subchunkData += 4;
	- subchunkDataLen -= 4;
	- }
	- if ( ChunkHasSubChunks(child->magic) ) {
	- LoadSubChunks(child, subchunkData, subchunkDataLen);
	- }
	-
	- data += child->length;
	- left -= child->length;
	- }
	-}
	-
	-RIFF_Chunk LoadRIFF(SDL_RWops src)
	-{
	- RIFF_Chunk *chunk;
	- Uint8 *subchunkData;
	- Uint32 subchunkDataLen;
	-
	- /* Allocate the chunk structure */
	- chunk = AllocRIFFChunk();
	-
	- /* Make sure the file is in RIFF format */
	- chunk->magic = SDL_ReadLE32(src);
	- chunk->length = SDL_ReadLE32(src);
	- if ( chunk->magic != RIFF ) {
	- __Sound_SetError("Not a RIFF file");
	- FreeRIFFChunk(chunk);
	- return NULL;
	- }
	- chunk->data = (Uint8 *)malloc(chunk->length);
	- if ( chunk->data == NULL ) {
	- __Sound_SetError(ERR_OUT_OF_MEMORY);
	- FreeRIFFChunk(chunk);
	- return NULL;
	- }
	- if ( SDL_RWread(src, chunk->data, chunk->length, 1) != 1 ) {
	- __Sound_SetError(ERR_IO_ERROR);
	- FreeRIFF(chunk);
	- return NULL;
	- }
	- subchunkData = chunk->data;
	- subchunkDataLen = chunk->length;
	- if ( ChunkHasSubType(chunk->magic) && subchunkDataLen >= 4 ) {
	- chunk->subtype = (subchunkData[0] << 0) \|
	- (subchunkData[1] << 8) \|
	- (subchunkData[2] << 16) \|
	- (subchunkData[3] << 24);
	- subchunkData += 4;
	- subchunkDataLen -= 4;
	- }
	- if ( ChunkHasSubChunks(chunk->magic) ) {
	- LoadSubChunks(chunk, subchunkData, subchunkDataLen);
	- }
	- return chunk;
	-}
	-
	-void FreeRIFF(RIFF_Chunk *chunk)
	-{
	- free(chunk->data);
	- FreeRIFFChunk(chunk);
	-}
	-
	-void PrintRIFF(RIFF_Chunk *chunk, int level)
	-{
	- static char prefix[128];
	-
	- if ( level == sizeof(prefix)-1 ) {
	- return;
	- }
	- if ( level > 0 ) {
	- prefix[(level-1)*2] = ' ';
	- prefix[(level-1)*2+1] = ' ';
	- }
	- prefix[level*2] = '\0';
	- printf("%sChunk: %c%c%c%c (%d bytes)", prefix,
	- ((chunk->magic >> 0) & 0xFF),
	- ((chunk->magic >> 8) & 0xFF),
	- ((chunk->magic >> 16) & 0xFF),
	- ((chunk->magic >> 24) & 0xFF), chunk->length);
	- if ( chunk->subtype ) {
	- printf(" subtype: %c%c%c%c",
	- ((chunk->subtype >> 0) & 0xFF),
	- ((chunk->subtype >> 8) & 0xFF),
	- ((chunk->subtype >> 16) & 0xFF),
	- ((chunk->subtype >> 24) & 0xFF));
	- }
	- printf("\n");
	- if ( chunk->child ) {
	- printf("%s{\n", prefix);
	- PrintRIFF(chunk->child, level + 1);
	- printf("%s}\n", prefix);
	- }
	- if ( chunk->next ) {
	- PrintRIFF(chunk->next, level);
	- }
	- if ( level > 0 ) {
	- prefix[(level-1)*2] = '\0';
	- }
	-}
	-
	-#ifdef TEST_MAIN_RIFF
	-
	-main(int argc, char *argv[])
	-{
	- int i;
	- for ( i = 1; i < argc; ++i ) {
	- RIFF_Chunk *chunk;
	- SDL_RWops *src = SDL_RWFromFile(argv[i], "rb");
	- if ( !src ) {
	- fprintf(stderr, "Couldn't open %s: %s", argv[i], SDL_GetError());
	- continue;
	- }
	- chunk = LoadRIFF(src);
	- if ( chunk ) {
	- PrintRIFF(chunk, 0);
	- FreeRIFF(chunk);
	- } else {
	- fprintf(stderr, "Couldn't load %s: %s\n", argv[i], SDL_GetError());
	- }
	- SDL_RWclose(src);
	- }
	-}
	-
	-#endif // TEST_MAIN
	-/* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * */
	-
	-/-------------------------------------------------------------------------/
	-/* * * * * * * * * * * * * * * * * load_dls.h * * * * * * * * * * * * * * */
	-/-------------------------------------------------------------------------/
	-/* This code is based on the DLS spec version 1.1, available at:
	- http://www.midi.org/about-midi/dls/dlsspec.shtml
	-*/
	-
	-/* Some typedefs so the public dls headers don't need to be modified */
	-#define FAR
	-typedef Uint8 BYTE;
	-typedef Sint16 SHORT;
	-typedef Uint16 USHORT;
	-typedef Uint16 WORD;
	-typedef Sint32 LONG;
	-typedef Uint32 ULONG;
	-typedef Uint32 DWORD;
	-#define mmioFOURCC(A, B, C, D) \
	- (((A) << 0) \| ((B) << 8) \| ((C) << 16) \| ((D) << 24))
	-#define DEFINE_GUID(A, B, C, E, F, G, H, I, J, K, L, M)
	-
	-#include "dls1.h"
	-#include "dls2.h"
	-
	-typedef struct _WaveFMT {
	- WORD wFormatTag;
	- WORD wChannels;
	- DWORD dwSamplesPerSec;
	- DWORD dwAvgBytesPerSec;
	- WORD wBlockAlign;
	- WORD wBitsPerSample;
	-} WaveFMT;
	-
	-typedef struct _DLS_Wave {
	- WaveFMT *format;
	- Uint8 *data;
	- Uint32 length;
	- WSMPL *wsmp;
	- WLOOP *wsmp_loop;
	-} DLS_Wave;
	-
	-typedef struct _DLS_Region {
	- RGNHEADER *header;
	- WAVELINK *wlnk;
	- WSMPL *wsmp;
	- WLOOP *wsmp_loop;
	- CONNECTIONLIST *art;
	- CONNECTION *artList;
	-} DLS_Region;
	-
	-typedef struct _DLS_Instrument {
	- const char *name;
	- INSTHEADER *header;
	- DLS_Region *regions;
	- CONNECTIONLIST *art;
	- CONNECTION *artList;
	-} DLS_Instrument;
	-
	-typedef struct _DLS_Data {
	- struct _RIFF_Chunk *chunk;
	-
	- Uint32 cInstruments;
	- DLS_Instrument *instruments;
	-
	- POOLTABLE *ptbl;
	- POOLCUE *ptblList;
	- DLS_Wave *waveList;
	-
	- const char *name;
	- const char *artist;
	- const char *copyright;
	- const char *comments;
	-} DLS_Data;
	-
	-extern DECLSPEC DLS_Data* SDLCALL LoadDLS(SDL_RWops *src);
	-extern DECLSPEC void SDLCALL FreeDLS(DLS_Data *chunk);
	-/* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * */
	-
	-/-------------------------------------------------------------------------/
	-/* * * * * * * * * * * * * * * * * load_dls.c * * * * * * * * * * * * * * */
	-/-------------------------------------------------------------------------/
	-
	-#define FOURCC_LIST 0x5453494c /* "LIST" */
	-#define FOURCC_FMT 0x20746D66 /* "fmt " */
	-#define FOURCC_DATA 0x61746164 /* "data" */
	-#define FOURCC_INFO mmioFOURCC('I','N','F','O')
	-#define FOURCC_IARL mmioFOURCC('I','A','R','L')
	-#define FOURCC_IART mmioFOURCC('I','A','R','T')
	-#define FOURCC_ICMS mmioFOURCC('I','C','M','S')
	-#define FOURCC_ICMT mmioFOURCC('I','C','M','T')
	-#define FOURCC_ICOP mmioFOURCC('I','C','O','P')
	-#define FOURCC_ICRD mmioFOURCC('I','C','R','D')
	-#define FOURCC_IENG mmioFOURCC('I','E','N','G')
	-#define FOURCC_IGNR mmioFOURCC('I','G','N','R')
	-#define FOURCC_IKEY mmioFOURCC('I','K','E','Y')
	-#define FOURCC_IMED mmioFOURCC('I','M','E','D')
	-#define FOURCC_INAM mmioFOURCC('I','N','A','M')
	-#define FOURCC_IPRD mmioFOURCC('I','P','R','D')
	-#define FOURCC_ISBJ mmioFOURCC('I','S','B','J')
	-#define FOURCC_ISFT mmioFOURCC('I','S','F','T')
	-#define FOURCC_ISRC mmioFOURCC('I','S','R','C')
	-#define FOURCC_ISRF mmioFOURCC('I','S','R','F')
	-#define FOURCC_ITCH mmioFOURCC('I','T','C','H')
	-
	-
	-static void FreeRegions(DLS_Instrument *instrument)
	-{
	- if ( instrument->regions ) {
	- free(instrument->regions);
	- }
	-}
	-
	-static void AllocRegions(DLS_Instrument *instrument)
	-{
	- int datalen = (instrument->header->cRegions * sizeof(DLS_Region));
	- FreeRegions(instrument);
	- instrument->regions = (DLS_Region *)malloc(datalen);
	- if ( instrument->regions ) {
	- memset(instrument->regions, 0, datalen);
	- }
	-}
	-
	-static void FreeInstruments(DLS_Data *data)
	-{
	- if ( data->instruments ) {
	- Uint32 i;
	- for ( i = 0; i < data->cInstruments; ++i ) {
	- FreeRegions(&data->instruments[i]);
	- }
	- free(data->instruments);
	- }
	-}
	-
	-static void AllocInstruments(DLS_Data *data)
	-{
	- int datalen = (data->cInstruments * sizeof(DLS_Instrument));
	- FreeInstruments(data);
	- data->instruments = (DLS_Instrument *)malloc(datalen);
	- if ( data->instruments ) {
	- memset(data->instruments, 0, datalen);
	- }
	-}
	-
	-static void FreeWaveList(DLS_Data *data)
	-{
	- if ( data->waveList ) {
	- free(data->waveList);
	- }
	-}
	-
	-static void AllocWaveList(DLS_Data *data)
	-{
	- int datalen = (data->ptbl->cCues * sizeof(DLS_Wave));
	- FreeWaveList(data);
	- data->waveList = (DLS_Wave *)malloc(datalen);
	- if ( data->waveList ) {
	- memset(data->waveList, 0, datalen);
	- }
	-}
	-
	-static void Parse_colh(DLS_Data data, RIFF_Chunk chunk)
	-{
	- data->cInstruments = SDL_SwapLE32((Uint32 )chunk->data);
	- AllocInstruments(data);
	-}
	-
	-static void Parse_insh(DLS_Data data, RIFF_Chunk chunk, DLS_Instrument *instrument)
	-{
	- INSTHEADER header = (INSTHEADER )chunk->data;
	- header->cRegions = SDL_SwapLE32(header->cRegions);
	- header->Locale.ulBank = SDL_SwapLE32(header->Locale.ulBank);
	- header->Locale.ulInstrument = SDL_SwapLE32(header->Locale.ulInstrument);
	- instrument->header = header;
	- AllocRegions(instrument);
	-}
	-
	-static void Parse_rgnh(DLS_Data data, RIFF_Chunk chunk, DLS_Region *region)
	-{
	- RGNHEADER header = (RGNHEADER )chunk->data;
	- header->RangeKey.usLow = SDL_SwapLE16(header->RangeKey.usLow);
	- header->RangeKey.usHigh = SDL_SwapLE16(header->RangeKey.usHigh);
	- header->RangeVelocity.usLow = SDL_SwapLE16(header->RangeVelocity.usLow);
	- header->RangeVelocity.usHigh = SDL_SwapLE16(header->RangeVelocity.usHigh);
	- header->fusOptions = SDL_SwapLE16(header->fusOptions);
	- header->usKeyGroup = SDL_SwapLE16(header->usKeyGroup);
	- region->header = header;
	-}
	-
	-static void Parse_wlnk(DLS_Data data, RIFF_Chunk chunk, DLS_Region *region)
	-{
	- WAVELINK wlnk = (WAVELINK )chunk->data;
	- wlnk->fusOptions = SDL_SwapLE16(wlnk->fusOptions);
	- wlnk->usPhaseGroup = SDL_SwapLE16(wlnk->usPhaseGroup);
	- wlnk->ulChannel = SDL_SwapLE16(wlnk->ulChannel);
	- wlnk->ulTableIndex = SDL_SwapLE16(wlnk->ulTableIndex);
	- region->wlnk = wlnk;
	-}
	-
	-static void Parse_wsmp(DLS_Data data, RIFF_Chunk chunk, WSMPL wsmp_ptr, WLOOP wsmp_loop_ptr)
	-{
	- Uint32 i;
	- WSMPL wsmp = (WSMPL )chunk->data;
	- WLOOP *loop;
	- wsmp->cbSize = SDL_SwapLE32(wsmp->cbSize);
	- wsmp->usUnityNote = SDL_SwapLE16(wsmp->usUnityNote);
	- wsmp->sFineTune = SDL_SwapLE16(wsmp->sFineTune);
	- wsmp->lAttenuation = SDL_SwapLE32(wsmp->lAttenuation);
	- wsmp->fulOptions = SDL_SwapLE32(wsmp->fulOptions);
	- wsmp->cSampleLoops = SDL_SwapLE32(wsmp->cSampleLoops);
	- loop = (WLOOP )((Uint8 )chunk->data + wsmp->cbSize);
	- *wsmp_ptr = wsmp;
	- *wsmp_loop_ptr = loop;
	- for ( i = 0; i < wsmp->cSampleLoops; ++i ) {
	- loop->cbSize = SDL_SwapLE32(loop->cbSize);
	- loop->ulType = SDL_SwapLE32(loop->ulType);
	- loop->ulStart = SDL_SwapLE32(loop->ulStart);
	- loop->ulLength = SDL_SwapLE32(loop->ulLength);
	- ++loop;
	- }
	-}
	-
	-static void Parse_art(DLS_Data data, RIFF_Chunk chunk, CONNECTIONLIST art_ptr, CONNECTION artList_ptr)
	-{
	- Uint32 i;
	- CONNECTIONLIST art = (CONNECTIONLIST )chunk->data;
	- CONNECTION *artList;
	- art->cbSize = SDL_SwapLE32(art->cbSize);
	- art->cConnections = SDL_SwapLE32(art->cConnections);
	- artList = (CONNECTION )((Uint8 )chunk->data + art->cbSize);
	- *art_ptr = art;
	- *artList_ptr = artList;
	- for ( i = 0; i < art->cConnections; ++i ) {
	- artList->usSource = SDL_SwapLE16(artList->usSource);
	- artList->usControl = SDL_SwapLE16(artList->usControl);
	- artList->usDestination = SDL_SwapLE16(artList->usDestination);
	- artList->usTransform = SDL_SwapLE16(artList->usTransform);
	- artList->lScale = SDL_SwapLE32(artList->lScale);
	- ++artList;
	- }
	-}
	-
	-static void Parse_lart(DLS_Data data, RIFF_Chunk chunk, CONNECTIONLIST conn_ptr, CONNECTION connList_ptr)
	-{
	- /* FIXME: This only supports one set of connections */
	- for ( chunk = chunk->child; chunk; chunk = chunk->next ) {
	- Uint32 magic = (chunk->magic == FOURCC_LIST) ? chunk->subtype : chunk->magic;
	- switch(magic) {
	- case FOURCC_ART1:
	- case FOURCC_ART2:
	- Parse_art(data, chunk, conn_ptr, connList_ptr);
	- return;
	- }
	- }
	-}
	-
	-static void Parse_rgn(DLS_Data data, RIFF_Chunk chunk, DLS_Region *region)
	-{
	- for ( chunk = chunk->child; chunk; chunk = chunk->next ) {
	- Uint32 magic = (chunk->magic == FOURCC_LIST) ? chunk->subtype : chunk->magic;
	- switch(magic) {
	- case FOURCC_RGNH:
	- Parse_rgnh(data, chunk, region);
	- break;
	- case FOURCC_WLNK:
	- Parse_wlnk(data, chunk, region);
	- break;
	- case FOURCC_WSMP:
	- Parse_wsmp(data, chunk, &region->wsmp, &region->wsmp_loop);
	- break;
	- case FOURCC_LART:
	- case FOURCC_LAR2:
	- Parse_lart(data, chunk, &region->art, &region->artList);
	- break;
	- }
	- }
	-}
	-
	-static void Parse_lrgn(DLS_Data data, RIFF_Chunk chunk, DLS_Instrument *instrument)
	-{
	- Uint32 region = 0;
	- for ( chunk = chunk->child; chunk; chunk = chunk->next ) {
	- Uint32 magic = (chunk->magic == FOURCC_LIST) ? chunk->subtype : chunk->magic;
	- switch(magic) {
	- case FOURCC_RGN:
	- case FOURCC_RGN2:
	- if ( region < instrument->header->cRegions ) {
	- Parse_rgn(data, chunk, &instrument->regions[region++]);
	- }
	- break;
	- }
	- }
	-}
	-
	-static void Parse_INFO_INS(DLS_Data data, RIFF_Chunk chunk, DLS_Instrument *instrument)
	-{
	- for ( chunk = chunk->child; chunk; chunk = chunk->next ) {
	- Uint32 magic = (chunk->magic == FOURCC_LIST) ? chunk->subtype : chunk->magic;
	- switch(magic) {
	- case FOURCC_INAM: /* Name */
	- instrument->name = chunk->data;
	- break;
	- }
	- }
	-}
	-
	-static void Parse_ins(DLS_Data data, RIFF_Chunk chunk, DLS_Instrument *instrument)
	-{
	- for ( chunk = chunk->child; chunk; chunk = chunk->next ) {
	- Uint32 magic = (chunk->magic == FOURCC_LIST) ? chunk->subtype : chunk->magic;
	- switch(magic) {
	- case FOURCC_INSH:
	- Parse_insh(data, chunk, instrument);
	- break;
	- case FOURCC_LRGN:
	- Parse_lrgn(data, chunk, instrument);
	- break;
	- case FOURCC_LART:
	- case FOURCC_LAR2:
	- Parse_lart(data, chunk, &instrument->art, &instrument->artList);
	- break;
	- case FOURCC_INFO:
	- Parse_INFO_INS(data, chunk, instrument);
	- break;
	- }
	- }
	-}
	-
	-static void Parse_lins(DLS_Data data, RIFF_Chunk chunk)
	-{
	- Uint32 instrument = 0;
	- for ( chunk = chunk->child; chunk; chunk = chunk->next ) {
	- Uint32 magic = (chunk->magic == FOURCC_LIST) ? chunk->subtype : chunk->magic;
	- switch(magic) {
	- case FOURCC_INS:
	- if ( instrument < data->cInstruments ) {
	- Parse_ins(data, chunk, &data->instruments[instrument++]);
	- }
	- break;
	- }
	- }
	-}
	-
	-static void Parse_ptbl(DLS_Data data, RIFF_Chunk chunk)
	-{
	- Uint32 i;
	- POOLTABLE ptbl = (POOLTABLE )chunk->data;
	- ptbl->cbSize = SDL_SwapLE32(ptbl->cbSize);
	- ptbl->cCues = SDL_SwapLE32(ptbl->cCues);
	- data->ptbl = ptbl;
	- data->ptblList = (POOLCUE )((Uint8 )chunk->data + ptbl->cbSize);
	- for ( i = 0; i < ptbl->cCues; ++i ) {
	- data->ptblList[i].ulOffset = SDL_SwapLE32(data->ptblList[i].ulOffset);
	- }
	- AllocWaveList(data);
	-}
	-
	-static void Parse_fmt(DLS_Data data, RIFF_Chunk chunk, DLS_Wave *wave)
	-{
	- WaveFMT fmt = (WaveFMT )chunk->data;
	- fmt->wFormatTag = SDL_SwapLE16(fmt->wFormatTag);
	- fmt->wChannels = SDL_SwapLE16(fmt->wChannels);
	- fmt->dwSamplesPerSec = SDL_SwapLE32(fmt->dwSamplesPerSec);
	- fmt->dwAvgBytesPerSec = SDL_SwapLE32(fmt->dwAvgBytesPerSec);
	- fmt->wBlockAlign = SDL_SwapLE16(fmt->wBlockAlign);
	- fmt->wBitsPerSample = SDL_SwapLE16(fmt->wBitsPerSample);
	- wave->format = fmt;
	-}
	-
	-static void Parse_data(DLS_Data data, RIFF_Chunk chunk, DLS_Wave *wave)
	-{
	- wave->data = chunk->data;
	- wave->length = chunk->length;
	-}
	-
	-static void Parse_wave(DLS_Data data, RIFF_Chunk chunk, DLS_Wave *wave)
	-{
	- for ( chunk = chunk->child; chunk; chunk = chunk->next ) {
	- Uint32 magic = (chunk->magic == FOURCC_LIST) ? chunk->subtype : chunk->magic;
	- switch(magic) {
	- case FOURCC_FMT:
	- Parse_fmt(data, chunk, wave);
	- break;
	- case FOURCC_DATA:
	- Parse_data(data, chunk, wave);
	- break;
	- case FOURCC_WSMP:
	- Parse_wsmp(data, chunk, &wave->wsmp, &wave->wsmp_loop);
	- break;
	- }
	- }
	-}
	-
	-static void Parse_wvpl(DLS_Data data, RIFF_Chunk chunk)
	-{
	- Uint32 wave = 0;
	- for ( chunk = chunk->child; chunk; chunk = chunk->next ) {
	- Uint32 magic = (chunk->magic == FOURCC_LIST) ? chunk->subtype : chunk->magic;
	- switch(magic) {
	- case FOURCC_wave:
	- if ( wave < data->ptbl->cCues ) {
	- Parse_wave(data, chunk, &data->waveList[wave++]);
	- }
	- break;
	- }
	- }
	-}
	-
	-static void Parse_INFO_DLS(DLS_Data data, RIFF_Chunk chunk)
	-{
	- for ( chunk = chunk->child; chunk; chunk = chunk->next ) {
	- Uint32 magic = (chunk->magic == FOURCC_LIST) ? chunk->subtype : chunk->magic;
	- switch(magic) {
	- case FOURCC_IARL: /* Archival Location */
	- break;
	- case FOURCC_IART: /* Artist */
	- data->artist = chunk->data;
	- break;
	- case FOURCC_ICMS: /* Commisioned */
	- break;
	- case FOURCC_ICMT: /* Comments */
	- data->comments = chunk->data;
	- break;
	- case FOURCC_ICOP: /* Copyright */
	- data->copyright = chunk->data;
	- break;
	- case FOURCC_ICRD: /* Creation Date */
	- break;
	- case FOURCC_IENG: /* Engineer */
	- break;
	- case FOURCC_IGNR: /* Genre */
	- break;
	- case FOURCC_IKEY: /* Keywords */
	- break;
	- case FOURCC_IMED: /* Medium */
	- break;
	- case FOURCC_INAM: /* Name */
	- data->name = chunk->data;
	- break;
	- case FOURCC_IPRD: /* Product */
	- break;
	- case FOURCC_ISBJ: /* Subject */
	- break;
	- case FOURCC_ISFT: /* Software */
	- break;
	- case FOURCC_ISRC: /* Source */
	- break;
	- case FOURCC_ISRF: /* Source Form */
	- break;
	- case FOURCC_ITCH: /* Technician */
	- break;
	- }
	- }
	-}
	-
	-DLS_Data LoadDLS(SDL_RWops src)
	-{
	- RIFF_Chunk *chunk;
	- DLS_Data data = (DLS_Data )malloc(sizeof(*data));
	- if ( !data ) {
	- __Sound_SetError(ERR_OUT_OF_MEMORY);
	- return NULL;
	- }
	- memset(data, 0, sizeof(*data));
	-
	- data->chunk = LoadRIFF(src);
	- if ( !data->chunk ) {
	- FreeDLS(data);
	- return NULL;
	- }
	-
	- for ( chunk = data->chunk->child; chunk; chunk = chunk->next ) {
	- Uint32 magic = (chunk->magic == FOURCC_LIST) ? chunk->subtype : chunk->magic;
	- switch(magic) {
	- case FOURCC_COLH:
	- Parse_colh(data, chunk);
	- break;
	- case FOURCC_LINS:
	- Parse_lins(data, chunk);
	- break;
	- case FOURCC_PTBL:
	- Parse_ptbl(data, chunk);
	- break;
	- case FOURCC_WVPL:
	- Parse_wvpl(data, chunk);
	- break;
	- case FOURCC_INFO:
	- Parse_INFO_DLS(data, chunk);
	- break;
	- }
	- }
	- return data;
	-}
	-
	-void FreeDLS(DLS_Data *data)
	-{
	- if ( data->chunk ) {
	- FreeRIFF(data->chunk);
	- }
	- FreeInstruments(data);
	- FreeWaveList(data);
	- free(data);
	-}
	-
	-static const char *SourceToString(USHORT usSource)
	-{
	- switch(usSource) {
	- case CONN_SRC_NONE:
	- return "NONE";
	- case CONN_SRC_LFO:
	- return "LFO";
	- case CONN_SRC_KEYONVELOCITY:
	- return "KEYONVELOCITY";
	- case CONN_SRC_KEYNUMBER:
	- return "KEYNUMBER";
	- case CONN_SRC_EG1:
	- return "EG1";
	- case CONN_SRC_EG2:
	- return "EG2";
	- case CONN_SRC_PITCHWHEEL:
	- return "PITCHWHEEL";
	- case CONN_SRC_CC1:
	- return "CC1";
	- case CONN_SRC_CC7:
	- return "CC7";
	- case CONN_SRC_CC10:
	- return "CC10";
	- case CONN_SRC_CC11:
	- return "CC11";
	- case CONN_SRC_POLYPRESSURE:
	- return "POLYPRESSURE";
	- case CONN_SRC_CHANNELPRESSURE:
	- return "CHANNELPRESSURE";
	- case CONN_SRC_VIBRATO:
	- return "VIBRATO";
	- case CONN_SRC_MONOPRESSURE:
	- return "MONOPRESSURE";
	- case CONN_SRC_CC91:
	- return "CC91";
	- case CONN_SRC_CC93:
	- return "CC93";
	- default:
	- return "UNKNOWN";
	- }
	-}
	-
	-static const char *TransformToString(USHORT usTransform)
	-{
	- switch (usTransform) {
	- case CONN_TRN_NONE:
	- return "NONE";
	- case CONN_TRN_CONCAVE:
	- return "CONCAVE";
	- case CONN_TRN_CONVEX:
	- return "CONVEX";
	- case CONN_TRN_SWITCH:
	- return "SWITCH";
	- default:
	- return "UNKNOWN";
	- }
	-}
	-
	-static const char *DestinationToString(USHORT usDestination)
	-{
	- switch (usDestination) {
	- case CONN_DST_NONE:
	- return "NONE";
	- case CONN_DST_ATTENUATION:
	- return "ATTENUATION";
	- case CONN_DST_PITCH:
	- return "PITCH";
	- case CONN_DST_PAN:
	- return "PAN";
	- case CONN_DST_LFO_FREQUENCY:
	- return "LFO_FREQUENCY";
	- case CONN_DST_LFO_STARTDELAY:
	- return "LFO_STARTDELAY";
	- case CONN_DST_EG1_ATTACKTIME:
	- return "EG1_ATTACKTIME";
	- case CONN_DST_EG1_DECAYTIME:
	- return "EG1_DECAYTIME";
	- case CONN_DST_EG1_RELEASETIME:
	- return "EG1_RELEASETIME";
	- case CONN_DST_EG1_SUSTAINLEVEL:
	- return "EG1_SUSTAINLEVEL";
	- case CONN_DST_EG2_ATTACKTIME:
	- return "EG2_ATTACKTIME";
	- case CONN_DST_EG2_DECAYTIME:
	- return "EG2_DECAYTIME";
	- case CONN_DST_EG2_RELEASETIME:
	- return "EG2_RELEASETIME";
	- case CONN_DST_EG2_SUSTAINLEVEL:
	- return "EG2_SUSTAINLEVEL";
	- case CONN_DST_KEYNUMBER:
	- return "KEYNUMBER";
	- case CONN_DST_LEFT:
	- return "LEFT";
	- case CONN_DST_RIGHT:
	- return "RIGHT";
	- case CONN_DST_CENTER:
	- return "CENTER";
	- case CONN_DST_LEFTREAR:
	- return "LEFTREAR";
	- case CONN_DST_RIGHTREAR:
	- return "RIGHTREAR";
	- case CONN_DST_LFE_CHANNEL:
	- return "LFE_CHANNEL";
	- case CONN_DST_CHORUS:
	- return "CHORUS";
	- case CONN_DST_REVERB:
	- return "REVERB";
	- case CONN_DST_VIB_FREQUENCY:
	- return "VIB_FREQUENCY";
	- case CONN_DST_VIB_STARTDELAY:
	- return "VIB_STARTDELAY";
	- case CONN_DST_EG1_DELAYTIME:
	- return "EG1_DELAYTIME";
	- case CONN_DST_EG1_HOLDTIME:
	- return "EG1_HOLDTIME";
	- case CONN_DST_EG1_SHUTDOWNTIME:
	- return "EG1_SHUTDOWNTIME";
	- case CONN_DST_EG2_DELAYTIME:
	- return "EG2_DELAYTIME";
	- case CONN_DST_EG2_HOLDTIME:
	- return "EG2_HOLDTIME";
	- case CONN_DST_FILTER_CUTOFF:
	- return "FILTER_CUTOFF";
	- case CONN_DST_FILTER_Q:
	- return "FILTER_Q";
	- default:
	- return "UNKOWN";
	- }
	-}
	-
	-static void PrintArt(const char type, CONNECTIONLIST art, CONNECTION *artList)
	-{
	- Uint32 i;
	- printf("%s Connections:\n", type);
	- for ( i = 0; i < art->cConnections; ++i ) {
	- printf(" Source: %s, Control: %s, Destination: %s, Transform: %s, Scale: %d\n",
	- SourceToString(artList[i].usSource),
	- SourceToString(artList[i].usControl),
	- DestinationToString(artList[i].usDestination),
	- TransformToString(artList[i].usTransform),
	- artList[i].lScale);
	- }
	-}
	-
	-static void PrintWave(DLS_Wave *wave, Uint32 index)
	-{
	- WaveFMT *format = wave->format;
	- if ( format ) {
	- printf(" Wave %u: Format: %hu, %hu channels, %u Hz, %hu bits (length = %u)\n", index, format->wFormatTag, format->wChannels, format->dwSamplesPerSec, format->wBitsPerSample, wave->length);
	- }
	- if ( wave->wsmp ) {
	- Uint32 i;
	- printf(" wsmp->usUnityNote = %hu\n", wave->wsmp->usUnityNote);
	- printf(" wsmp->sFineTune = %hd\n", wave->wsmp->sFineTune);
	- printf(" wsmp->lAttenuation = %d\n", wave->wsmp->lAttenuation);
	- printf(" wsmp->fulOptions = 0x%8.8x\n", wave->wsmp->fulOptions);
	- printf(" wsmp->cSampleLoops = %u\n", wave->wsmp->cSampleLoops);
	- for ( i = 0; i < wave->wsmp->cSampleLoops; ++i ) {
	- WLOOP *loop = &wave->wsmp_loop[i];
	- printf(" Loop %u:\n", i);
	- printf(" ulStart = %u\n", loop->ulStart);
	- printf(" ulLength = %u\n", loop->ulLength);
	- }
	- }
	-}
	-
	-static void PrintRegion(DLS_Region *region, Uint32 index)
	-{
	- printf(" Region %u:\n", index);
	- if ( region->header ) {
	- printf(" RangeKey = { %hu - %hu }\n", region->header->RangeKey.usLow, region->header->RangeKey.usHigh);
	- printf(" RangeVelocity = { %hu - %hu }\n", region->header->RangeVelocity.usLow, region->header->RangeVelocity.usHigh);
	- printf(" fusOptions = 0x%4.4hx\n", region->header->fusOptions);
	- printf(" usKeyGroup = %hu\n", region->header->usKeyGroup);
	- }
	- if ( region->wlnk ) {
	- printf(" wlnk->fusOptions = 0x%4.4hx\n", region->wlnk->fusOptions);
	- printf(" wlnk->usPhaseGroup = %hu\n", region->wlnk->usPhaseGroup);
	- printf(" wlnk->ulChannel = %u\n", region->wlnk->ulChannel);
	- printf(" wlnk->ulTableIndex = %u\n", region->wlnk->ulTableIndex);
	- }
	- if ( region->wsmp ) {
	- Uint32 i;
	- printf(" wsmp->usUnityNote = %hu\n", region->wsmp->usUnityNote);
	- printf(" wsmp->sFineTune = %hd\n", region->wsmp->sFineTune);
	- printf(" wsmp->lAttenuation = %d\n", region->wsmp->lAttenuation);
	- printf(" wsmp->fulOptions = 0x%8.8x\n", region->wsmp->fulOptions);
	- printf(" wsmp->cSampleLoops = %u\n", region->wsmp->cSampleLoops);
	- for ( i = 0; i < region->wsmp->cSampleLoops; ++i ) {
	- WLOOP *loop = &region->wsmp_loop[i];
	- printf(" Loop %u:\n", i);
	- printf(" ulStart = %u\n", loop->ulStart);
	- printf(" ulLength = %u\n", loop->ulLength);
	- }
	- }
	- if ( region->art && region->art->cConnections > 0 ) {
	- PrintArt("Region", region->art, region->artList);
	- }
	-}
	-
	-static void PrintInstrument(DLS_Instrument *instrument, Uint32 index)
	-{
	- printf("Instrument %u:\n", index);
	- if ( instrument->name ) {
	- printf(" Name: %s\n", instrument->name);
	- }
	- if ( instrument->header ) {
	- Uint32 i;
	- printf(" ulBank = 0x%8.8x\n", instrument->header->Locale.ulBank);
	- printf(" ulInstrument = %u\n", instrument->header->Locale.ulInstrument);
	- printf(" Regions: %u\n", instrument->header->cRegions);
	- for ( i = 0; i < instrument->header->cRegions; ++i ) {
	- PrintRegion(&instrument->regions[i], i);
	- }
	- }
	- if ( instrument->art && instrument->art->cConnections > 0 ) {
	- PrintArt("Instrument", instrument->art, instrument->artList);
	- }
	-};
	-
	-void PrintDLS(DLS_Data *data)
	-{
	- printf("DLS Data:\n");
	- printf("cInstruments = %u\n", data->cInstruments);
	- if ( data->instruments ) {
	- Uint32 i;
	- for ( i = 0; i < data->cInstruments; ++i ) {
	- PrintInstrument(&data->instruments[i], i);
	- }
	- }
	- if ( data->ptbl && data->ptbl->cCues > 0 ) {
	- Uint32 i;
	- printf("Cues: ");
	- for ( i = 0; i < data->ptbl->cCues; ++i ) {
	- if ( i > 0 ) {
	- printf(", ");
	- }
	- printf("%u", data->ptblList[i].ulOffset);
	- }
	- printf("\n");
	- }
	- if ( data->waveList ) {
	- Uint32 i;
	- printf("Waves:\n");
	- for ( i = 0; i < data->ptbl->cCues; ++i ) {
	- PrintWave(&data->waveList[i], i);
	- }
	- }
	- if ( data->name ) {
	- printf("Name: %s\n", data->name);
	- }
	- if ( data->artist ) {
	- printf("Artist: %s\n", data->artist);
	- }
	- if ( data->copyright ) {
	- printf("Copyright: %s\n", data->copyright);
	- }
	- if ( data->comments ) {
	- printf("Comments: %s\n", data->comments);
	- }
	-}
	-
	-#ifdef TEST_MAIN_DLS
	-
	-main(int argc, char *argv[])
	-{
	- int i;
	- for ( i = 1; i < argc; ++i ) {
	- DLS_Data *data;
	- SDL_RWops *src = SDL_RWFromFile(argv[i], "rb");
	- if ( !src ) {
	- fprintf(stderr, "Couldn't open %s: %s", argv[i], SDL_GetError());
	- continue;
	- }
	- data = LoadDLS(src);
	- if ( data ) {
	- PrintRIFF(data->chunk, 0);
	- PrintDLS(data);
	- FreeDLS(data);
	- } else {
	- fprintf(stderr, "Couldn't load %s: %s\n", argv[i], SDL_GetError());
	- }
	- SDL_RWclose(src);
	- }
	-}
	-
	-#endif // TEST_MAIN
	-/* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * */
	-
	-/-------------------------------------------------------------------------/
	-/* * * * * * * * * * * * * * * * * instrum_dls.c * * * * * * * * * * * * * */
	-/-------------------------------------------------------------------------/
	-
	-DLS_Data Timidity_LoadDLS(SDL_RWops src)
	-{
	- DLS_Data *patches = LoadDLS(src);
	- if (!patches) {
	- SNDDBG(("%s", SDL_GetError()));
	- }
	- return patches;
	-}
	-
	-void Timidity_FreeDLS(DLS_Data *patches)
	-{
	- FreeDLS(patches);
	-}
	-
	-/* convert timecents to sec */
	-static double to_msec(int timecent)
	-{
	- if (timecent == 0x80000000 \|\| timecent == 0)
	- return 0.0;
	- return 1000.0 * pow(2.0, (double)(timecent / 65536) / 1200.0);
	-}
	-
	-/* convert decipercent to {0..1} */
	-static double to_normalized_percent(int decipercent)
	-{
	- return ((double)(decipercent / 65536)) / 1000.0;
	-}
	-
	-/* convert from 8bit value to fractional offset (15.15) */
	-static Sint32 to_offset(int offset)
	-{
	- return (Sint32)offset << (7+15);
	-}
	-
	-/* calculate ramp rate in fractional unit;
	- * diff = 8bit, time = msec
	- */
	-static Sint32 calc_rate(MidiSong *song, int diff, int sample_rate, double msec)
	-{
	- double rate;
	-
	- if(msec < 6)
	- msec = 6;
	- if(diff == 0)
	- diff = 255;
	- diff <<= (7+15);
	- rate = ((double)diff / song->rate) * song->control_ratio * 1000.0 / msec;
	- return (Sint32)rate;
	-}
	-
	-static int load_connection(ULONG cConnections, CONNECTION *artList, USHORT destination)
	-{
	- ULONG i;
	- int value = 0;
	- for (i = 0; i < cConnections; ++i) {
	- CONNECTION *conn = &artList[i];
	- if(conn->usDestination == destination) {
	- // The formula for the destination is:
	- // usDestination = usDestination + usTransform(usSource * (usControl * lScale))
	- // Since we are only handling source/control of NONE and identity
	- // transform, this simplifies to: usDestination = usDestination + lScale
	- if (conn->usSource == CONN_SRC_NONE &&
	- conn->usControl == CONN_SRC_NONE &&
	- conn->usTransform == CONN_TRN_NONE)
	- value += conn->lScale;
	- }
	- }
	- return value;
	-}
	-
	-static void load_region_dls(MidiSong song, Sample sample, DLS_Instrument *ins, Uint32 index)
	-{
	- DLS_Region *rgn = &ins->regions[index];
	- DLS_Wave *wave = &song->patches->waveList[rgn->wlnk->ulTableIndex];
	-
	- sample->low_freq = freq_table[rgn->header->RangeKey.usLow];
	- sample->high_freq = freq_table[rgn->header->RangeKey.usHigh];
	- sample->root_freq = freq_table[rgn->wsmp->usUnityNote];
	- sample->low_vel = rgn->header->RangeVelocity.usLow;
	- sample->high_vel = rgn->header->RangeVelocity.usHigh;
	-
	- sample->modes = MODES_16BIT;
	- sample->sample_rate = wave->format->dwSamplesPerSec;
	- sample->data_length = wave->length / 2;
	- sample->data = (sample_t *)safe_malloc(wave->length);
	- memcpy(sample->data, wave->data, wave->length);
	- if (rgn->wsmp->cSampleLoops) {
	- sample->modes \|= (MODES_LOOPING\|MODES_SUSTAIN);
	- sample->loop_start = rgn->wsmp_loop->ulStart / 2;
	- sample->loop_end = sample->loop_start + (rgn->wsmp_loop->ulLength / 2);
	- }
	- sample->volume = 1.0f;
	-
	- if (sample->modes & MODES_SUSTAIN) {
	- int value;
	- double attack, hold, decay, release; int sustain;
	- CONNECTIONLIST *art = NULL;
	- CONNECTION *artList = NULL;
	-
	- if (ins->art && ins->art->cConnections > 0 && ins->artList) {
	- art = ins->art;
	- artList = ins->artList;
	- } else {
	- art = rgn->art;
	- artList = rgn->artList;
	- }
	-
	- value = load_connection(art->cConnections, artList, CONN_DST_EG1_ATTACKTIME);
	- attack = to_msec(value);
	- value = load_connection(art->cConnections, artList, CONN_DST_EG1_HOLDTIME);
	- hold = to_msec(value);
	- value = load_connection(art->cConnections, artList, CONN_DST_EG1_DECAYTIME);
	- decay = to_msec(value);
	- value = load_connection(art->cConnections, artList, CONN_DST_EG1_RELEASETIME);
	- release = to_msec(value);
	- value = load_connection(art->cConnections, artList, CONN_DST_EG1_SUSTAINLEVEL);
	- sustain = (int)((1.0 - to_normalized_percent(value)) * 250.0);
	- value = load_connection(art->cConnections, artList, CONN_DST_PAN);
	- sample->panning = (int)((0.5 + to_normalized_percent(value)) * 127.0);
	-
	-/*
	-printf("%d, Rate=%d LV=%d HV=%d Low=%d Hi=%d Root=%d Pan=%d Attack=%f Hold=%f Sustain=%d Decay=%f Release=%f\n", index, sample->sample_rate, rgn->header->RangeVelocity.usLow, rgn->header->RangeVelocity.usHigh, sample->low_freq, sample->high_freq, sample->root_freq, sample->panning, attack, hold, sustain, decay, release);
	-*/
	-
	- sample->envelope_offset[0] = to_offset(255);
	- sample->envelope_rate[0] = calc_rate(song, 255, sample->sample_rate, attack);
	-
	- sample->envelope_offset[1] = to_offset(250);
	- sample->envelope_rate[1] = calc_rate(song, 5, sample->sample_rate, hold);
	-
	- sample->envelope_offset[2] = to_offset(sustain);
	- sample->envelope_rate[2] = calc_rate(song, 255 - sustain, sample->sample_rate, decay);
	-
	- sample->envelope_offset[3] = to_offset(0);
	- sample->envelope_rate[3] = calc_rate(song, 5 + sustain, sample->sample_rate, release);
	-
	- sample->envelope_offset[4] = to_offset(0);
	- sample->envelope_rate[4] = to_offset(1);
	-
	- sample->envelope_offset[5] = to_offset(0);
	- sample->envelope_rate[5] = to_offset(1);
	-
	- sample->modes \|= MODES_ENVELOPE;
	- }
	-
	- sample->data_length <<= FRACTION_BITS;
	- sample->loop_start <<= FRACTION_BITS;
	- sample->loop_end <<= FRACTION_BITS;
	-}
	-
	-Instrument load_instrument_dls(MidiSong song, int drum, int bank, int instrument)
	-{
	- Instrument *inst;
	- Uint32 i;
	- DLS_Instrument *dls_ins;
	-
	- if (!song->patches)
	- return(NULL);
	-
	- drum = drum ? 0x80000000 : 0;
	- for (i = 0; i < song->patches->cInstruments; ++i) {
	- dls_ins = &song->patches->instruments[i];
	- if ((dls_ins->header->Locale.ulBank & 0x80000000) == drum &&
	- ((dls_ins->header->Locale.ulBank >> 8) & 0xFF) == bank &&
	- dls_ins->header->Locale.ulInstrument == instrument)
	- break;
	- }
	- if (i == song->patches->cInstruments && !bank) {
	- for (i = 0; i < song->patches->cInstruments; ++i) {
	- dls_ins = &song->patches->instruments[i];
	- if ((dls_ins->header->Locale.ulBank & 0x80000000) == drum &&
	- dls_ins->header->Locale.ulInstrument == instrument)
	- break;
	- }
	- }
	- if (i == song->patches->cInstruments) {
	- SNDDBG(("Couldn't find %s instrument %d in bank %d\n", drum ? "drum" : "melodic", instrument, bank));
	- return(NULL);
	- }
	-
	- inst = (Instrument )safe_malloc(sizeof(inst));
	- inst->samples = dls_ins->header->cRegions;
	- inst->sample = (Sample )safe_malloc(inst->samples sizeof(*inst->sample));
	- memset(inst->sample, 0, inst->samples * sizeof(*inst->sample));
	-/*
	-printf("Found %s instrument %d in bank %d named %s with %d regions\n", drum ? "drum" : "melodic", instrument, bank, dls_ins->name, inst->samples);
	-*/
	- for (i = 0; i < dls_ins->header->cRegions; ++i) {
	- load_region_dls(song, &inst->sample[i], dls_ins, i);
	- }
	- return(inst);
	-}
	diff --git a/util/sdl/sound/decoders/timidity/instrum_dls.h b/util/sdl/sound/decoders/timidity/instrum_dls.h
	deleted file mode 100644
	index ac3865a0..00000000
	--- a/util/sdl/sound/decoders/timidity/instrum_dls.h
	+++ /dev/null
	@@ -1,24 +0,0 @@
	-/*
	-
	- TiMidity -- Experimental MIDI to WAVE converter
	- Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi>
	-
	- This program is free software; you can redistribute it and/or modify
	- it under the terms of the GNU General Public License as published by
	- the Free Software Foundation; either version 2 of the License, or
	- (at your option) any later version.
	-
	- This program is distributed in the hope that it will be useful,
	- but WITHOUT ANY WARRANTY; without even the implied warranty of
	- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
	- GNU General Public License for more details.
	-
	- You should have received a copy of the GNU General Public License
	- along with this program; if not, write to the Free Software
	- Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
	-
	- instrum.h
	-
	- */
	-
	-extern Instrument load_instrument_dls(MidiSong song, int drum, int bank, int instrument);
	diff --git a/util/sdl/sound/decoders/timidity/mix.c b/util/sdl/sound/decoders/timidity/mix.c
	deleted file mode 100644
	index af8869ae..00000000
	--- a/util/sdl/sound/decoders/timidity/mix.c
	+++ /dev/null
	@@ -1,573 +0,0 @@
	-/*
	-
	- TiMidity -- Experimental MIDI to WAVE converter
	- Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi>
	-
	- Suddenly, you realize that this program is free software; you get
	- an overwhelming urge to redistribute it and/or modify it under the
	- terms of the GNU General Public License as published by the Free
	- Software Foundation; either version 2 of the License, or (at your
	- option) any later version.
	-
	- This program is distributed in the hope that it will be useful,
	- but WITHOUT ANY WARRANTY; without even the implied warranty of
	- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
	- GNU General Public License for more details.
	-
	- You should have received another copy of the GNU General Public
	- License along with this program; if not, write to the Free
	- Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
	- I bet they'll be amazed.
	-
	- mix.c */
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#include <math.h>
	-#include <stdio.h>
	-#include <stdlib.h>
	-
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-#include "timidity.h"
	-#include "options.h"
	-#include "instrum.h"
	-#include "playmidi.h"
	-#include "output.h"
	-#include "tables.h"
	-#include "resample.h"
	-#include "mix.h"
	-
	-/* Returns 1 if envelope runs out */
	-int recompute_envelope(MidiSong *song, int v)
	-{
	- int stage;
	-
	- stage = song->voice[v].envelope_stage;
	-
	- if (stage>5)
	- {
	- /* Envelope ran out. */
	- song->voice[v].status = VOICE_FREE;
	- return 1;
	- }
	-
	- if (song->voice[v].sample->modes & MODES_ENVELOPE)
	- {
	- if (song->voice[v].status==VOICE_ON \|\| song->voice[v].status==VOICE_SUSTAINED)
	- {
	- if (stage>2)
	- {
	- /* Freeze envelope until note turns off. Trumpets want this. */
	- song->voice[v].envelope_increment=0;
	- return 0;
	- }
	- }
	- }
	- song->voice[v].envelope_stage=stage+1;
	-
	- if (song->voice[v].envelope_volume==song->voice[v].sample->envelope_offset[stage])
	- return recompute_envelope(song, v);
	- song->voice[v].envelope_target = song->voice[v].sample->envelope_offset[stage];
	- song->voice[v].envelope_increment = song->voice[v].sample->envelope_rate[stage];
	- if (song->voice[v].envelope_target < song->voice[v].envelope_volume)
	- song->voice[v].envelope_increment = -song->voice[v].envelope_increment;
	- return 0;
	-}
	-
	-void apply_envelope_to_amp(MidiSong *song, int v)
	-{
	- float lamp = song->voice[v].left_amp, ramp;
	- Sint32 la,ra;
	- if (song->voice[v].panned == PANNED_MYSTERY)
	- {
	- ramp = song->voice[v].right_amp;
	- if (song->voice[v].tremolo_phase_increment)
	- {
	- lamp *= song->voice[v].tremolo_volume;
	- ramp *= song->voice[v].tremolo_volume;
	- }
	- if (song->voice[v].sample->modes & MODES_ENVELOPE)
	- {
	- lamp *= (float)vol_table[song->voice[v].envelope_volume>>23];
	- ramp *= (float)vol_table[song->voice[v].envelope_volume>>23];
	- }
	-
	- la = (Sint32)FSCALE(lamp,AMP_BITS);
	-
	- if (la>MAX_AMP_VALUE)
	- la=MAX_AMP_VALUE;
	-
	- ra = (Sint32)FSCALE(ramp,AMP_BITS);
	- if (ra>MAX_AMP_VALUE)
	- ra=MAX_AMP_VALUE;
	-
	- song->voice[v].left_mix = la;
	- song->voice[v].right_mix = ra;
	- }
	- else
	- {
	- if (song->voice[v].tremolo_phase_increment)
	- lamp *= song->voice[v].tremolo_volume;
	- if (song->voice[v].sample->modes & MODES_ENVELOPE)
	- lamp *= (float)vol_table[song->voice[v].envelope_volume>>23];
	-
	- la = (Sint32)FSCALE(lamp,AMP_BITS);
	-
	- if (la>MAX_AMP_VALUE)
	- la=MAX_AMP_VALUE;
	-
	- song->voice[v].left_mix = la;
	- }
	-}
	-
	-static int update_envelope(MidiSong *song, int v)
	-{
	- song->voice[v].envelope_volume += song->voice[v].envelope_increment;
	- /* Why is there no ^^ operator?? */
	- if (((song->voice[v].envelope_increment < 0) &&
	- (song->voice[v].envelope_volume <= song->voice[v].envelope_target)) \|\|
	- ((song->voice[v].envelope_increment > 0) &&
	- (song->voice[v].envelope_volume >= song->voice[v].envelope_target)))
	- {
	- song->voice[v].envelope_volume = song->voice[v].envelope_target;
	- if (recompute_envelope(song, v))
	- return 1;
	- }
	- return 0;
	-}
	-
	-static void update_tremolo(MidiSong *song, int v)
	-{
	- Sint32 depth = song->voice[v].sample->tremolo_depth << 7;
	-
	- if (song->voice[v].tremolo_sweep)
	- {
	- /* Update sweep position */
	-
	- song->voice[v].tremolo_sweep_position += song->voice[v].tremolo_sweep;
	- if (song->voice[v].tremolo_sweep_position >= (1 << SWEEP_SHIFT))
	- song->voice[v].tremolo_sweep=0; /* Swept to max amplitude */
	- else
	- {
	- /* Need to adjust depth */
	- depth *= song->voice[v].tremolo_sweep_position;
	- depth >>= SWEEP_SHIFT;
	- }
	- }
	-
	- song->voice[v].tremolo_phase += song->voice[v].tremolo_phase_increment;
	-
	- /* if (song->voice[v].tremolo_phase >= (SINE_CYCLE_LENGTH<<RATE_SHIFT))
	- song->voice[v].tremolo_phase -= SINE_CYCLE_LENGTH<<RATE_SHIFT; */
	-
	- song->voice[v].tremolo_volume = (float)
	- (1.0 - FSCALENEG((sine(song->voice[v].tremolo_phase >> RATE_SHIFT) + 1.0)
	- * depth * TREMOLO_AMPLITUDE_TUNING,
	- 17));
	-
	- /* I'm not sure about the +1.0 there -- it makes tremoloed voices'
	- volumes on average the lower the higher the tremolo amplitude. */
	-}
	-
	-/* Returns 1 if the note died */
	-static int update_signal(MidiSong *song, int v)
	-{
	- if (song->voice[v].envelope_increment && update_envelope(song, v))
	- return 1;
	-
	- if (song->voice[v].tremolo_phase_increment)
	- update_tremolo(song, v);
	-
	- apply_envelope_to_amp(song, v);
	- return 0;
	-}
	-
	-#define MIXATION(a) lp++ += (a)s;
	-
	-static void mix_mystery_signal(MidiSong song, sample_t sp, Sint32 *lp, int v,
	- int count)
	-{
	- Voice *vp = song->voice + v;
	- final_volume_t
	- left=vp->left_mix,
	- right=vp->right_mix;
	- int cc;
	- sample_t s;
	-
	- if (!(cc = vp->control_counter))
	- {
	- cc = song->control_ratio;
	- if (update_signal(song, v))
	- return; /* Envelope ran out */
	- left = vp->left_mix;
	- right = vp->right_mix;
	- }
	-
	- while (count)
	- if (cc < count)
	- {
	- count -= cc;
	- while (cc--)
	- {
	- s = *sp++;
	- MIXATION(left);
	- MIXATION(right);
	- }
	- cc = song->control_ratio;
	- if (update_signal(song, v))
	- return; /* Envelope ran out */
	- left = vp->left_mix;
	- right = vp->right_mix;
	- }
	- else
	- {
	- vp->control_counter = cc - count;
	- while (count--)
	- {
	- s = *sp++;
	- MIXATION(left);
	- MIXATION(right);
	- }
	- return;
	- }
	-}
	-
	-static void mix_center_signal(MidiSong song, sample_t sp, Sint32 *lp, int v,
	- int count)
	-{
	- Voice *vp = song->voice + v;
	- final_volume_t
	- left=vp->left_mix;
	- int cc;
	- sample_t s;
	-
	- if (!(cc = vp->control_counter))
	- {
	- cc = song->control_ratio;
	- if (update_signal(song, v))
	- return; /* Envelope ran out */
	- left = vp->left_mix;
	- }
	-
	- while (count)
	- if (cc < count)
	- {
	- count -= cc;
	- while (cc--)
	- {
	- s = *sp++;
	- MIXATION(left);
	- MIXATION(left);
	- }
	- cc = song->control_ratio;
	- if (update_signal(song, v))
	- return; /* Envelope ran out */
	- left = vp->left_mix;
	- }
	- else
	- {
	- vp->control_counter = cc - count;
	- while (count--)
	- {
	- s = *sp++;
	- MIXATION(left);
	- MIXATION(left);
	- }
	- return;
	- }
	-}
	-
	-static void mix_single_signal(MidiSong song, sample_t sp, Sint32 *lp, int v,
	- int count)
	-{
	- Voice *vp = song->voice + v;
	- final_volume_t
	- left=vp->left_mix;
	- int cc;
	- sample_t s;
	-
	- if (!(cc = vp->control_counter))
	- {
	- cc = song->control_ratio;
	- if (update_signal(song, v))
	- return; /* Envelope ran out */
	- left = vp->left_mix;
	- }
	-
	- while (count)
	- if (cc < count)
	- {
	- count -= cc;
	- while (cc--)
	- {
	- s = *sp++;
	- MIXATION(left);
	- lp++;
	- }
	- cc = song->control_ratio;
	- if (update_signal(song, v))
	- return; /* Envelope ran out */
	- left = vp->left_mix;
	- }
	- else
	- {
	- vp->control_counter = cc - count;
	- while (count--)
	- {
	- s = *sp++;
	- MIXATION(left);
	- lp++;
	- }
	- return;
	- }
	-}
	-
	-static void mix_mono_signal(MidiSong song, sample_t sp, Sint32 *lp, int v,
	- int count)
	-{
	- Voice *vp = song->voice + v;
	- final_volume_t
	- left=vp->left_mix;
	- int cc;
	- sample_t s;
	-
	- if (!(cc = vp->control_counter))
	- {
	- cc = song->control_ratio;
	- if (update_signal(song, v))
	- return; /* Envelope ran out */
	- left = vp->left_mix;
	- }
	-
	- while (count)
	- if (cc < count)
	- {
	- count -= cc;
	- while (cc--)
	- {
	- s = *sp++;
	- MIXATION(left);
	- }
	- cc = song->control_ratio;
	- if (update_signal(song, v))
	- return; /* Envelope ran out */
	- left = vp->left_mix;
	- }
	- else
	- {
	- vp->control_counter = cc - count;
	- while (count--)
	- {
	- s = *sp++;
	- MIXATION(left);
	- }
	- return;
	- }
	-}
	-
	-static void mix_mystery(MidiSong song, sample_t sp, Sint32 *lp, int v, int count)
	-{
	- final_volume_t
	- left = song->voice[v].left_mix,
	- right = song->voice[v].right_mix;
	- sample_t s;
	-
	- while (count--)
	- {
	- s = *sp++;
	- MIXATION(left);
	- MIXATION(right);
	- }
	-}
	-
	-static void mix_center(MidiSong song, sample_t sp, Sint32 *lp, int v, int count)
	-{
	- final_volume_t
	- left = song->voice[v].left_mix;
	- sample_t s;
	-
	- while (count--)
	- {
	- s = *sp++;
	- MIXATION(left);
	- MIXATION(left);
	- }
	-}
	-
	-static void mix_single(MidiSong song, sample_t sp, Sint32 *lp, int v, int count)
	-{
	- final_volume_t
	- left = song->voice[v].left_mix;
	- sample_t s;
	-
	- while (count--)
	- {
	- s = *sp++;
	- MIXATION(left);
	- lp++;
	- }
	-}
	-
	-static void mix_mono(MidiSong song, sample_t sp, Sint32 *lp, int v, int count)
	-{
	- final_volume_t
	- left = song->voice[v].left_mix;
	- sample_t s;
	-
	- while (count--)
	- {
	- s = *sp++;
	- MIXATION(left);
	- }
	-}
	-
	-/* Ramp a note out in c samples */
	-static void ramp_out(MidiSong song, sample_t sp, Sint32 *lp, int v, Sint32 c)
	-{
	-
	- /* should be final_volume_t, but Uint8 gives trouble. */
	- Sint32 left, right, li, ri;
	-
	- sample_t s=0; /* silly warning about uninitialized s */
	-
	- /* Fix by James Caldwell */
	- if ( c == 0 ) c = 1;
	-
	- left=song->voice[v].left_mix;
	- li=-(left/c);
	- if (!li) li=-1;
	-
	- /* printf("Ramping out: left=%d, c=%d, li=%d\n", left, c, li); */
	-
	- if (!(song->encoding & PE_MONO))
	- {
	- if (song->voice[v].panned==PANNED_MYSTERY)
	- {
	- right=song->voice[v].right_mix;
	- ri=-(right/c);
	- while (c--)
	- {
	- left += li;
	- if (left<0)
	- left=0;
	- right += ri;
	- if (right<0)
	- right=0;
	- s=*sp++;
	- MIXATION(left);
	- MIXATION(right);
	- }
	- }
	- else if (song->voice[v].panned==PANNED_CENTER)
	- {
	- while (c--)
	- {
	- left += li;
	- if (left<0)
	- return;
	- s=*sp++;
	- MIXATION(left);
	- MIXATION(left);
	- }
	- }
	- else if (song->voice[v].panned==PANNED_LEFT)
	- {
	- while (c--)
	- {
	- left += li;
	- if (left<0)
	- return;
	- s=*sp++;
	- MIXATION(left);
	- lp++;
	- }
	- }
	- else if (song->voice[v].panned==PANNED_RIGHT)
	- {
	- while (c--)
	- {
	- left += li;
	- if (left<0)
	- return;
	- s=*sp++;
	- lp++;
	- MIXATION(left);
	- }
	- }
	- }
	- else
	- {
	- /* Mono output. */
	- while (c--)
	- {
	- left += li;
	- if (left<0)
	- return;
	- s=*sp++;
	- MIXATION(left);
	- }
	- }
	-}
	-
	-
	-/************** interface function ****************/
	-
	-void mix_voice(MidiSong song, Sint32 buf, int v, Sint32 c)
	-{
	- Voice *vp = song->voice + v;
	- sample_t *sp;
	- if (vp->status==VOICE_DIE)
	- {
	- if (c>=MAX_DIE_TIME)
	- c=MAX_DIE_TIME;
	- sp=resample_voice(song, v, &c);
	- ramp_out(song, sp, buf, v, c);
	- vp->status=VOICE_FREE;
	- }
	- else
	- {
	- sp=resample_voice(song, v, &c);
	- if (song->encoding & PE_MONO)
	- {
	- /* Mono output. */
	- if (vp->envelope_increment \|\| vp->tremolo_phase_increment)
	- mix_mono_signal(song, sp, buf, v, c);
	- else
	- mix_mono(song, sp, buf, v, c);
	- }
	- else
	- {
	- if (vp->panned == PANNED_MYSTERY)
	- {
	- if (vp->envelope_increment \|\| vp->tremolo_phase_increment)
	- mix_mystery_signal(song, sp, buf, v, c);
	- else
	- mix_mystery(song, sp, buf, v, c);
	- }
	- else if (vp->panned == PANNED_CENTER)
	- {
	- if (vp->envelope_increment \|\| vp->tremolo_phase_increment)
	- mix_center_signal(song, sp, buf, v, c);
	- else
	- mix_center(song, sp, buf, v, c);
	- }
	- else
	- {
	- /* It's either full left or full right. In either case,
	- every other sample is 0. Just get the offset right: */
	- if (vp->panned == PANNED_RIGHT) buf++;
	-
	- if (vp->envelope_increment \|\| vp->tremolo_phase_increment)
	- mix_single_signal(song, sp, buf, v, c);
	- else
	- mix_single(song, sp, buf, v, c);
	- }
	- }
	- }
	-}
	diff --git a/util/sdl/sound/decoders/timidity/mix.h b/util/sdl/sound/decoders/timidity/mix.h
	deleted file mode 100644
	index b94c32b9..00000000
	--- a/util/sdl/sound/decoders/timidity/mix.h
	+++ /dev/null
	@@ -1,27 +0,0 @@
	-/*
	-
	- TiMidity -- Experimental MIDI to WAVE converter
	- Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi>
	-
	- In case you haven't heard, this program is free software;
	- you can redistribute it and/or modify it under the terms of the
	- GNU General Public License as published by the Free Software
	- Foundation; either version 2 of the License, or (at your option)
	- any later version.
	-
	- This program is distributed in the hope that it will be useful,
	- but WITHOUT ANY WARRANTY; without even the implied warranty of
	- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
	- GNU General Public License for more details.
	-
	- You should have received a copy of the GNU General Public License
	- along with this program; if not, write to the Free Software
	- Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
	-
	- mix.h
	-
	-*/
	-
	-extern void mix_voice(MidiSong song, Sint32 buf, int v, Sint32 c);
	-extern int recompute_envelope(MidiSong *song, int v);
	-extern void apply_envelope_to_amp(MidiSong *song, int v);
	diff --git a/util/sdl/sound/decoders/timidity/options.h b/util/sdl/sound/decoders/timidity/options.h
	deleted file mode 100644
	index 4fa2fb7c..00000000
	--- a/util/sdl/sound/decoders/timidity/options.h
	+++ /dev/null
	@@ -1,113 +0,0 @@
	-/*
	- TiMidity -- Experimental MIDI to WAVE converter
	- Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi>
	-
	- This program is free software; you can redistribute it and/or modify
	- it under the terms of the GNU General Public License as published by
	- the Free Software Foundation; either version 2 of the License, or
	- (at your option) any later version.
	-
	- This program is distributed in the hope that it will be useful,
	- but WITHOUT ANY WARRANTY; without even the implied warranty of
	- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
	- GNU General Public License for more details.
	-
	- You should have received a copy of the GNU General Public License
	- along with this program; if not, write to the Free Software
	- Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
	-*/
	-
	-/* When a patch file can't be opened, one of these extensions is
	- appended to the filename and the open is tried again.
	- */
	-#define PATCH_EXT_LIST { ".pat", 0 }
	-
	-/* Acoustic Grand Piano seems to be the usual default instrument. */
	-#define DEFAULT_PROGRAM 0
	-
	-/* 9 here is MIDI channel 10, which is the standard percussion channel.
	- Some files (notably C:\WINDOWS\CANYON.MID) think that 16 is one too.
	- On the other hand, some files know that 16 is not a drum channel and
	- try to play music on it. This is now a runtime option, so this isn't
	- a critical choice anymore. */
	-#define DEFAULT_DRUMCHANNELS ((1<<9) \| (1<<15))
	-
	-/* In percent. */
	-#define DEFAULT_AMPLIFICATION 70
	-
	-/* Default polyphony */
	-#define DEFAULT_VOICES 32
	-
	-/* 1000 here will give a control ratio of 22:1 with 22 kHz output.
	- Higher CONTROLS_PER_SECOND values allow more accurate rendering
	- of envelopes and tremolo. The cost is CPU time. */
	-#define CONTROLS_PER_SECOND 1000
	-
	-/* Make envelopes twice as fast. Saves ~20% CPU time (notes decay
	- faster) and sounds more like a GUS. There is now a command line
	- option to toggle this as well. */
	-#define FAST_DECAY
	-
	-/* How many bits to use for the fractional part of sample positions.
	- This affects tonal accuracy. The entire position counter must fit
	- in 32 bits, so with FRACTION_BITS equal to 12, the maximum size of
	- a sample is 1048576 samples (2 megabytes in memory). The GUS gets
	- by with just 9 bits and a little help from its friends...
	- "The GUS does not SUCK!!!" -- a happy user :) */
	-#define FRACTION_BITS 12
	-
	-/* For some reason the sample volume is always set to maximum in all
	- patch files. Define this for a crude adjustment that may help
	- equalize instrument volumes. */
	-#define ADJUST_SAMPLE_VOLUMES
	-
	-/* The number of samples to use for ramping out a dying note. Affects
	- click removal. */
	-#define MAX_DIE_TIME 20
	-
	-/**************************************************************************/
	-/* Anything below this shouldn't need to be changed unless you're porting
	- to a new machine with other than 32-bit, big-endian words. */
	-/**************************************************************************/
	-
	-/* change FRACTION_BITS above, not these */
	-#define INTEGER_MASK (0xFFFFFFFF << FRACTION_BITS)
	-#define FRACTION_MASK (~ INTEGER_MASK)
	-
	-/* This is enforced by some computations that must fit in an int */
	-#define MAX_CONTROL_RATIO 255
	-
	-#define MAX_AMPLIFICATION 800
	-
	-/* The TiMidity configuration file */
	-#define CONFIG_FILE "timidity.cfg"
	-
	-/* These affect general volume */
	-#define GUARD_BITS 3
	-#define AMP_BITS (15-GUARD_BITS)
	-
	-#define MAX_AMP_VALUE ((1<<(AMP_BITS+1))-1)
	-
	-#define FSCALE(a,b) (float)((a) * (double)(1<<(b)))
	-#define FSCALENEG(a,b) (float)((a) * (1.0L / (double)(1<<(b))))
	-
	-/* Vibrato and tremolo Choices of the Day */
	-#define SWEEP_TUNING 38
	-#define VIBRATO_AMPLITUDE_TUNING 1.0L
	-#define VIBRATO_RATE_TUNING 38
	-#define TREMOLO_AMPLITUDE_TUNING 1.0L
	-#define TREMOLO_RATE_TUNING 38
	-
	-#define SWEEP_SHIFT 16
	-#define RATE_SHIFT 5
	-
	-#ifndef PI
	- #define PI 3.14159265358979323846
	-#endif
	-
	-/* The path separator (D.M.) */
	-#ifdef WIN32
	-# define PATH_SEP '\\'
	-#else
	-# define PATH_SEP '/'
	-#endif
	diff --git a/util/sdl/sound/decoders/timidity/output.c b/util/sdl/sound/decoders/timidity/output.c
	deleted file mode 100644
	index cfe3991c..00000000
	--- a/util/sdl/sound/decoders/timidity/output.c
	+++ /dev/null
	@@ -1,116 +0,0 @@
	-/*
	-
	- TiMidity -- Experimental MIDI to WAVE converter
	- Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi>
	-
	- This program is free software; you can redistribute it and/or modify
	- it under the terms of the GNU General Public License as published by
	- the Free Software Foundation; either version 2 of the License, or
	- (at your option) any later version.
	-
	- This program is distributed in the hope that it will be useful,
	- but WITHOUT ANY WARRANTY; without even the implied warranty of
	- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
	- GNU General Public License for more details.
	-
	- You should have received a copy of the GNU General Public License
	- along with this program; if not, write to the Free Software
	- Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
	-
	- output.c
	-
	- Audio output (to file / device) functions.
	-*/
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-#include "options.h"
	-#include "output.h"
	-
	-/*****************************************************************/
	-/* Some functions to convert signed 32-bit data to other formats */
	-
	-void s32tos8(void dp, Sint32 lp, Sint32 c)
	-{
	- Sint8 cp=(Sint8 )(dp);
	- Sint32 l;
	- while (c--)
	- {
	- l=(*lp++)>>(32-8-GUARD_BITS);
	- if (l>127) l=127;
	- else if (l<-128) l=-128;
	- *cp++ = (Sint8) (l);
	- }
	-}
	-
	-void s32tou8(void dp, Sint32 lp, Sint32 c)
	-{
	- Uint8 cp=(Uint8 )(dp);
	- Sint32 l;
	- while (c--)
	- {
	- l=(*lp++)>>(32-8-GUARD_BITS);
	- if (l>127) l=127;
	- else if (l<-128) l=-128;
	- *cp++ = 0x80 ^ ((Uint8) l);
	- }
	-}
	-
	-void s32tos16(void dp, Sint32 lp, Sint32 c)
	-{
	- Sint16 sp=(Sint16 )(dp);
	- Sint32 l;
	- while (c--)
	- {
	- l=(*lp++)>>(32-16-GUARD_BITS);
	- if (l > 32767) l=32767;
	- else if (l<-32768) l=-32768;
	- *sp++ = (Sint16)(l);
	- }
	-}
	-
	-void s32tou16(void dp, Sint32 lp, Sint32 c)
	-{
	- Uint16 sp=(Uint16 )(dp);
	- Sint32 l;
	- while (c--)
	- {
	- l=(*lp++)>>(32-16-GUARD_BITS);
	- if (l > 32767) l=32767;
	- else if (l<-32768) l=-32768;
	- *sp++ = 0x8000 ^ (Uint16)(l);
	- }
	-}
	-
	-void s32tos16x(void dp, Sint32 lp, Sint32 c)
	-{
	- Sint16 sp=(Sint16 )(dp);
	- Sint32 l;
	- while (c--)
	- {
	- l=(*lp++)>>(32-16-GUARD_BITS);
	- if (l > 32767) l=32767;
	- else if (l<-32768) l=-32768;
	- *sp++ = SDL_Swap16((Sint16)(l));
	- }
	-}
	-
	-void s32tou16x(void dp, Sint32 lp, Sint32 c)
	-{
	- Uint16 sp=(Uint16 )(dp);
	- Sint32 l;
	- while (c--)
	- {
	- l=(*lp++)>>(32-16-GUARD_BITS);
	- if (l > 32767) l=32767;
	- else if (l<-32768) l=-32768;
	- *sp++ = SDL_Swap16(0x8000 ^ (Uint16)(l));
	- }
	-}
	diff --git a/util/sdl/sound/decoders/timidity/output.h b/util/sdl/sound/decoders/timidity/output.h
	deleted file mode 100644
	index 9cbe3326..00000000
	--- a/util/sdl/sound/decoders/timidity/output.h
	+++ /dev/null
	@@ -1,56 +0,0 @@
	-/*
	-
	- TiMidity -- Experimental MIDI to WAVE converter
	- Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi>
	-
	- This program is free software; you can redistribute it and/or modify
	- it under the terms of the GNU General Public License as published by
	- the Free Software Foundation; either version 2 of the License, or
	- (at your option) any later version.
	-
	- This program is distributed in the hope that it will be useful,
	- but WITHOUT ANY WARRANTY; without even the implied warranty of
	- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
	- GNU General Public License for more details.
	-
	- You should have received a copy of the GNU General Public License
	- along with this program; if not, write to the Free Software
	- Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
	-
	- output.h
	-
	-*/
	-
	-/* Data format encoding bits */
	-
	-#define PE_MONO 0x01 /* versus stereo */
	-#define PE_SIGNED 0x02 /* versus unsigned */
	-#define PE_16BIT 0x04 /* versus 8-bit */
	-
	-/* Conversion functions -- These overwrite the Sint32 data in *lp with
	- data in another format */
	-
	-/* 8-bit signed and unsigned*/
	-extern void s32tos8(void dp, Sint32 lp, Sint32 c);
	-extern void s32tou8(void dp, Sint32 lp, Sint32 c);
	-
	-/* 16-bit */
	-extern void s32tos16(void dp, Sint32 lp, Sint32 c);
	-extern void s32tou16(void dp, Sint32 lp, Sint32 c);
	-
	-/* byte-exchanged 16-bit */
	-extern void s32tos16x(void dp, Sint32 lp, Sint32 c);
	-extern void s32tou16x(void dp, Sint32 lp, Sint32 c);
	-
	-/* little-endian and big-endian specific */
	-#if SDL_BYTEORDER == SDL_LIL_ENDIAN
	-#define s32tou16l s32tou16
	-#define s32tou16b s32tou16x
	-#define s32tos16l s32tos16
	-#define s32tos16b s32tos16x
	-#else
	-#define s32tou16l s32tou16x
	-#define s32tou16b s32tou16
	-#define s32tos16l s32tos16x
	-#define s32tos16b s32tos16
	-#endif
	diff --git a/util/sdl/sound/decoders/timidity/playmidi.c b/util/sdl/sound/decoders/timidity/playmidi.c
	deleted file mode 100644
	index cd0b3cda..00000000
	--- a/util/sdl/sound/decoders/timidity/playmidi.c
	+++ /dev/null
	@@ -1,806 +0,0 @@
	-/*
	-
	- TiMidity -- Experimental MIDI to WAVE converter
	- Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi>
	-
	- This program is free software; you can redistribute it and/or modify
	- it under the terms of the GNU General Public License as published by
	- the Free Software Foundation; either version 2 of the License, or
	- (at your option) any later version.
	-
	- This program is distributed in the hope that it will be useful,
	- but WITHOUT ANY WARRANTY; without even the implied warranty of
	- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
	- GNU General Public License for more details.
	-
	- You should have received a copy of the GNU General Public License
	- along with this program; if not, write to the Free Software
	- Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
	-
	- playmidi.c -- random stuff in need of rearrangement
	-
	-*/
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#include <stdio.h>
	-#include <stdlib.h>
	-#include <string.h>
	-
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-#include "timidity.h"
	-#include "options.h"
	-#include "instrum.h"
	-#include "playmidi.h"
	-#include "output.h"
	-#include "mix.h"
	-#include "tables.h"
	-
	-static void adjust_amplification(MidiSong *song)
	-{
	- song->master_volume = (float)(song->amplification) / (float)100.0;
	-}
	-
	-static void reset_voices(MidiSong *song)
	-{
	- int i;
	- for (i=0; i<MAX_VOICES; i++)
	- song->voice[i].status=VOICE_FREE;
	-}
	-
	-/* Process the Reset All Controllers event */
	-static void reset_controllers(MidiSong *song, int c)
	-{
	- song->channel[c].volume=90; /* Some standard says, although the SCC docs say 0. */
	- song->channel[c].expression=127; /* SCC-1 does this. */
	- song->channel[c].sustain=0;
	- song->channel[c].pitchbend=0x2000;
	- song->channel[c].pitchfactor=0; /* to be computed */
	-}
	-
	-static void reset_midi(MidiSong *song)
	-{
	- int i;
	- for (i=0; i<16; i++)
	- {
	- reset_controllers(song, i);
	- /* The rest of these are unaffected by the Reset All Controllers event */
	- song->channel[i].program=song->default_program;
	- song->channel[i].panning=NO_PANNING;
	- song->channel[i].pitchsens=2;
	- song->channel[i].bank=0; /* tone bank or drum set */
	- }
	- reset_voices(song);
	-}
	-
	-static void select_sample(MidiSong song, int v, Instrument ip, int vel)
	-{
	- Sint32 f, cdiff, diff;
	- int s,i;
	- Sample sp, closest;
	-
	- s=ip->samples;
	- sp=ip->sample;
	-
	- if (s==1)
	- {
	- song->voice[v].sample=sp;
	- return;
	- }
	-
	- f=song->voice[v].orig_frequency;
	- for (i=0; i<s; i++)
	- {
	- if (sp->low_vel <= vel && sp->high_vel >= vel &&
	- sp->low_freq <= f && sp->high_freq >= f)
	- {
	- song->voice[v].sample=sp;
	- return;
	- }
	- sp++;
	- }
	-
	- /*
	- No suitable sample found! We'll select the sample whose root
	- frequency is closest to the one we want. (Actually we should
	- probably convert the low, high, and root frequencies to MIDI note
	- values and compare those.) */
	-
	- cdiff=0x7FFFFFFF;
	- closest=sp=ip->sample;
	- for(i=0; i<s; i++)
	- {
	- diff=sp->root_freq - f;
	- if (diff<0) diff=-diff;
	- if (diff<cdiff)
	- {
	- cdiff=diff;
	- closest=sp;
	- }
	- sp++;
	- }
	- song->voice[v].sample=closest;
	- return;
	-}
	-
	-static void recompute_freq(MidiSong *song, int v)
	-{
	- int
	- sign=(song->voice[v].sample_increment < 0), /* for bidirectional loops */
	- pb=song->channel[song->voice[v].channel].pitchbend;
	- double a;
	-
	- if (!song->voice[v].sample->sample_rate)
	- return;
	-
	- if (song->voice[v].vibrato_control_ratio)
	- {
	- /* This instrument has vibrato. Invalidate any precomputed
	- sample_increments. */
	-
	- int i=VIBRATO_SAMPLE_INCREMENTS;
	- while (i--)
	- song->voice[v].vibrato_sample_increment[i]=0;
	- }
	-
	- if (pb==0x2000 \|\| pb<0 \|\| pb>0x3FFF)
	- song->voice[v].frequency = song->voice[v].orig_frequency;
	- else
	- {
	- pb-=0x2000;
	- if (!(song->channel[song->voice[v].channel].pitchfactor))
	- {
	- /* Damn. Somebody bent the pitch. */
	- Sint32 i=pb*song->channel[song->voice[v].channel].pitchsens;
	- if (pb<0)
	- i=-i;
	- song->channel[song->voice[v].channel].pitchfactor=
	- (float)(bend_fine[(i>>5) & 0xFF] * bend_coarse[i>>13]);
	- }
	- if (pb>0)
	- song->voice[v].frequency=
	- (Sint32)(song->channel[song->voice[v].channel].pitchfactor *
	- (double)(song->voice[v].orig_frequency));
	- else
	- song->voice[v].frequency=
	- (Sint32)((double)(song->voice[v].orig_frequency) /
	- song->channel[song->voice[v].channel].pitchfactor);
	- }
	-
	- a = FSCALE(((double)(song->voice[v].sample->sample_rate) *
	- (double)(song->voice[v].frequency)) /
	- ((double)(song->voice[v].sample->root_freq) *
	- (double)(song->rate)),
	- FRACTION_BITS);
	-
	- if (sign)
	- a = -a; /* need to preserve the loop direction */
	-
	- song->voice[v].sample_increment = (Sint32)(a);
	-}
	-
	-static void recompute_amp(MidiSong *song, int v)
	-{
	- Sint32 tempamp;
	-
	- /* TODO: use fscale */
	-
	- tempamp= (song->voice[v].velocity *
	- song->channel[song->voice[v].channel].volume *
	- song->channel[song->voice[v].channel].expression); /* 21 bits */
	-
	- if (!(song->encoding & PE_MONO))
	- {
	- if (song->voice[v].panning > 60 && song->voice[v].panning < 68)
	- {
	- song->voice[v].panned=PANNED_CENTER;
	-
	- song->voice[v].left_amp=
	- FSCALENEG((double)(tempamp) * song->voice[v].sample->volume * song->master_volume,
	- 21);
	- }
	- else if (song->voice[v].panning<5)
	- {
	- song->voice[v].panned = PANNED_LEFT;
	-
	- song->voice[v].left_amp=
	- FSCALENEG((double)(tempamp) * song->voice[v].sample->volume * song->master_volume,
	- 20);
	- }
	- else if (song->voice[v].panning>123)
	- {
	- song->voice[v].panned = PANNED_RIGHT;
	-
	- song->voice[v].left_amp= /* left_amp will be used */
	- FSCALENEG((double)(tempamp) * song->voice[v].sample->volume * song->master_volume,
	- 20);
	- }
	- else
	- {
	- song->voice[v].panned = PANNED_MYSTERY;
	-
	- song->voice[v].left_amp=
	- FSCALENEG((double)(tempamp) * song->voice[v].sample->volume * song->master_volume,
	- 27);
	- song->voice[v].right_amp = song->voice[v].left_amp * (song->voice[v].panning);
	- song->voice[v].left_amp *= (float)(127 - song->voice[v].panning);
	- }
	- }
	- else
	- {
	- song->voice[v].panned = PANNED_CENTER;
	-
	- song->voice[v].left_amp=
	- FSCALENEG((double)(tempamp) * song->voice[v].sample->volume * song->master_volume,
	- 21);
	- }
	-}
	-
	-static void start_note(MidiSong song, MidiEvent e, int i)
	-{
	- Instrument *ip;
	- int j;
	-
	- if (ISDRUMCHANNEL(song, e->channel))
	- {
	- if (!(ip=song->drumset[song->channel[e->channel].bank]->instrument[e->a]))
	- {
	- if (!(ip=song->drumset[0]->instrument[e->a]))
	- return; /* No instrument? Then we can't play. */
	- }
	- if (ip->samples != 1)
	- {
	- SNDDBG(("Strange: percussion instrument with %d samples!",
	- ip->samples));
	- }
	-
	- if (ip->sample->note_to_use) /* Do we have a fixed pitch? */
	- song->voice[i].orig_frequency = freq_table[(int)(ip->sample->note_to_use)];
	- else
	- song->voice[i].orig_frequency = freq_table[e->a & 0x7F];
	-
	- /* drums are supposed to have only one sample */
	- song->voice[i].sample = ip->sample;
	- }
	- else
	- {
	- if (song->channel[e->channel].program == SPECIAL_PROGRAM)
	- ip=song->default_instrument;
	- else if (!(ip=song->tonebank[song->channel[e->channel].bank]->
	- instrument[song->channel[e->channel].program]))
	- {
	- if (!(ip=song->tonebank[0]->instrument[song->channel[e->channel].program]))
	- return; /* No instrument? Then we can't play. */
	- }
	-
	- if (ip->sample->note_to_use) /* Fixed-pitch instrument? */
	- song->voice[i].orig_frequency = freq_table[(int)(ip->sample->note_to_use)];
	- else
	- song->voice[i].orig_frequency = freq_table[e->a & 0x7F];
	- select_sample(song, i, ip, e->b);
	- }
	-
	- song->voice[i].status = VOICE_ON;
	- song->voice[i].channel = e->channel;
	- song->voice[i].note = e->a;
	- song->voice[i].velocity = e->b;
	- song->voice[i].sample_offset = 0;
	- song->voice[i].sample_increment = 0; /* make sure it isn't negative */
	-
	- song->voice[i].tremolo_phase = 0;
	- song->voice[i].tremolo_phase_increment = song->voice[i].sample->tremolo_phase_increment;
	- song->voice[i].tremolo_sweep = song->voice[i].sample->tremolo_sweep_increment;
	- song->voice[i].tremolo_sweep_position = 0;
	-
	- song->voice[i].vibrato_sweep = song->voice[i].sample->vibrato_sweep_increment;
	- song->voice[i].vibrato_sweep_position = 0;
	- song->voice[i].vibrato_control_ratio = song->voice[i].sample->vibrato_control_ratio;
	- song->voice[i].vibrato_control_counter = song->voice[i].vibrato_phase = 0;
	- for (j=0; j<VIBRATO_SAMPLE_INCREMENTS; j++)
	- song->voice[i].vibrato_sample_increment[j] = 0;
	-
	- if (song->channel[e->channel].panning != NO_PANNING)
	- song->voice[i].panning = song->channel[e->channel].panning;
	- else
	- song->voice[i].panning = song->voice[i].sample->panning;
	-
	- recompute_freq(song, i);
	- recompute_amp(song, i);
	- if (song->voice[i].sample->modes & MODES_ENVELOPE)
	- {
	- /* Ramp up from 0 */
	- song->voice[i].envelope_stage = 0;
	- song->voice[i].envelope_volume = 0;
	- song->voice[i].control_counter = 0;
	- recompute_envelope(song, i);
	- apply_envelope_to_amp(song, i);
	- }
	- else
	- {
	- song->voice[i].envelope_increment = 0;
	- apply_envelope_to_amp(song, i);
	- }
	-}
	-
	-static void kill_note(MidiSong *song, int i)
	-{
	- song->voice[i].status = VOICE_DIE;
	-}
	-
	-/* Only one instance of a note can be playing on a single channel. */
	-static void note_on(MidiSong *song)
	-{
	- int i = song->voices, lowest=-1;
	- Sint32 lv=0x7FFFFFFF, v;
	- MidiEvent *e = song->current_event;
	-
	- while (i--)
	- {
	- if (song->voice[i].status == VOICE_FREE)
	- lowest=i; /* Can't get a lower volume than silence */
	- else if (song->voice[i].channel==e->channel &&
	- (song->voice[i].note==e->a \|\| song->channel[song->voice[i].channel].mono))
	- kill_note(song, i);
	- }
	-
	- if (lowest != -1)
	- {
	- /* Found a free voice. */
	- start_note(song,e,lowest);
	- return;
	- }
	-
	- /* Look for the decaying note with the lowest volume */
	- i = song->voices;
	- while (i--)
	- {
	- if ((song->voice[i].status != VOICE_ON) &&
	- (song->voice[i].status != VOICE_DIE))
	- {
	- v = song->voice[i].left_mix;
	- if ((song->voice[i].panned == PANNED_MYSTERY)
	- && (song->voice[i].right_mix > v))
	- v = song->voice[i].right_mix;
	- if (v<lv)
	- {
	- lv=v;
	- lowest=i;
	- }
	- }
	- }
	-
	- if (lowest != -1)
	- {
	- /* This can still cause a click, but if we had a free voice to
	- spare for ramping down this note, we wouldn't need to kill it
	- in the first place... Still, this needs to be fixed. Perhaps
	- we could use a reserve of voices to play dying notes only. */
	-
	- song->cut_notes++;
	- song->voice[lowest].status=VOICE_FREE;
	- start_note(song,e,lowest);
	- }
	- else
	- song->lost_notes++;
	-}
	-
	-static void finish_note(MidiSong *song, int i)
	-{
	- if (song->voice[i].sample->modes & MODES_ENVELOPE)
	- {
	- /* We need to get the envelope out of Sustain stage */
	- song->voice[i].envelope_stage = 3;
	- song->voice[i].status = VOICE_OFF;
	- recompute_envelope(song, i);
	- apply_envelope_to_amp(song, i);
	- }
	- else
	- {
	- /* Set status to OFF so resample_voice() will let this voice out
	- of its loop, if any. In any case, this voice dies when it
	- hits the end of its data (ofs>=data_length). */
	- song->voice[i].status = VOICE_OFF;
	- }
	-}
	-
	-static void note_off(MidiSong *song)
	-{
	- int i = song->voices;
	- MidiEvent *e = song->current_event;
	-
	- while (i--)
	- if (song->voice[i].status == VOICE_ON &&
	- song->voice[i].channel == e->channel &&
	- song->voice[i].note == e->a)
	- {
	- if (song->channel[e->channel].sustain)
	- {
	- song->voice[i].status = VOICE_SUSTAINED;
	- }
	- else
	- finish_note(song, i);
	- return;
	- }
	-}
	-
	-/* Process the All Notes Off event */
	-static void all_notes_off(MidiSong *song)
	-{
	- int i = song->voices;
	- int c = song->current_event->channel;
	-
	- SNDDBG(("All notes off on channel %d", c));
	- while (i--)
	- if (song->voice[i].status == VOICE_ON &&
	- song->voice[i].channel == c)
	- {
	- if (song->channel[c].sustain)
	- song->voice[i].status = VOICE_SUSTAINED;
	- else
	- finish_note(song, i);
	- }
	-}
	-
	-/* Process the All Sounds Off event */
	-static void all_sounds_off(MidiSong *song)
	-{
	- int i = song->voices;
	- int c = song->current_event->channel;
	-
	- while (i--)
	- if (song->voice[i].channel == c &&
	- song->voice[i].status != VOICE_FREE &&
	- song->voice[i].status != VOICE_DIE)
	- {
	- kill_note(song, i);
	- }
	-}
	-
	-static void adjust_pressure(MidiSong *song)
	-{
	- MidiEvent *e = song->current_event;
	- int i = song->voices;
	-
	- while (i--)
	- if (song->voice[i].status == VOICE_ON &&
	- song->voice[i].channel == e->channel &&
	- song->voice[i].note == e->a)
	- {
	- song->voice[i].velocity = e->b;
	- recompute_amp(song, i);
	- apply_envelope_to_amp(song, i);
	- return;
	- }
	-}
	-
	-static void drop_sustain(MidiSong *song)
	-{
	- int i = song->voices;
	- int c = song->current_event->channel;
	-
	- while (i--)
	- if (song->voice[i].status == VOICE_SUSTAINED && song->voice[i].channel == c)
	- finish_note(song, i);
	-}
	-
	-static void adjust_pitchbend(MidiSong *song)
	-{
	- int c = song->current_event->channel;
	- int i = song->voices;
	-
	- while (i--)
	- if (song->voice[i].status != VOICE_FREE && song->voice[i].channel == c)
	- {
	- recompute_freq(song, i);
	- }
	-}
	-
	-static void adjust_volume(MidiSong *song)
	-{
	- int c = song->current_event->channel;
	- int i = song->voices;
	-
	- while (i--)
	- if (song->voice[i].channel == c &&
	- (song->voice[i].status==VOICE_ON \|\| song->voice[i].status==VOICE_SUSTAINED))
	- {
	- recompute_amp(song, i);
	- apply_envelope_to_amp(song, i);
	- }
	-}
	-
	-static void seek_forward(MidiSong *song, Sint32 until_time)
	-{
	- reset_voices(song);
	- while (song->current_event->time < until_time)
	- {
	- switch(song->current_event->type)
	- {
	- /* All notes stay off. Just handle the parameter changes. */
	-
	- case ME_PITCH_SENS:
	- song->channel[song->current_event->channel].pitchsens =
	- song->current_event->a;
	- song->channel[song->current_event->channel].pitchfactor = 0;
	- break;
	-
	- case ME_PITCHWHEEL:
	- song->channel[song->current_event->channel].pitchbend =
	- song->current_event->a + song->current_event->b * 128;
	- song->channel[song->current_event->channel].pitchfactor = 0;
	- break;
	-
	- case ME_MAINVOLUME:
	- song->channel[song->current_event->channel].volume =
	- song->current_event->a;
	- break;
	-
	- case ME_PAN:
	- song->channel[song->current_event->channel].panning =
	- song->current_event->a;
	- break;
	-
	- case ME_EXPRESSION:
	- song->channel[song->current_event->channel].expression =
	- song->current_event->a;
	- break;
	-
	- case ME_PROGRAM:
	- if (ISDRUMCHANNEL(song, song->current_event->channel))
	- /* Change drum set */
	- song->channel[song->current_event->channel].bank =
	- song->current_event->a;
	- else
	- song->channel[song->current_event->channel].program =
	- song->current_event->a;
	- break;
	-
	- case ME_SUSTAIN:
	- song->channel[song->current_event->channel].sustain =
	- song->current_event->a;
	- break;
	-
	- case ME_RESET_CONTROLLERS:
	- reset_controllers(song, song->current_event->channel);
	- break;
	-
	- case ME_TONE_BANK:
	- song->channel[song->current_event->channel].bank =
	- song->current_event->a;
	- break;
	-
	- case ME_EOT:
	- song->current_sample = song->current_event->time;
	- return;
	- }
	- song->current_event++;
	- }
	- /song->current_sample=song->current_event->time;/
	- if (song->current_event != song->events)
	- song->current_event--;
	- song->current_sample=until_time;
	-}
	-
	-static void skip_to(MidiSong *song, Sint32 until_time)
	-{
	- if (song->current_sample > until_time)
	- song->current_sample = 0;
	-
	- reset_midi(song);
	- song->buffered_count = 0;
	- song->buffer_pointer = song->common_buffer;
	- song->current_event = song->events;
	-
	- if (until_time)
	- seek_forward(song, until_time);
	-}
	-
	-static void do_compute_data(MidiSong *song, Sint32 count)
	-{
	- int i;
	- memset(song->buffer_pointer, 0,
	- (song->encoding & PE_MONO) ? (count * 4) : (count * 8));
	- for (i = 0; i < song->voices; i++)
	- {
	- if(song->voice[i].status != VOICE_FREE)
	- mix_voice(song, song->buffer_pointer, i, count);
	- }
	- song->current_sample += count;
	-}
	-
	-/* count=0 means flush remaining buffered data to output device, then
	- flush the device itself */
	-static void compute_data(MidiSong song, void stream, Sint32 count)
	-{
	- int channels;
	-
	- if ( song->encoding & PE_MONO )
	- channels = 1;
	- else
	- channels = 2;
	-
	- if (!count)
	- {
	- if (song->buffered_count)
	- song->write(stream, song->common_buffer, channels * song->buffered_count);
	- song->buffer_pointer = song->common_buffer;
	- song->buffered_count = 0;
	- return;
	- }
	-
	- while ((count + song->buffered_count) >= song->buffer_size)
	- {
	- do_compute_data(song, song->buffer_size - song->buffered_count);
	- count -= song->buffer_size - song->buffered_count;
	- song->write(stream, song->common_buffer, channels * song->buffer_size);
	- song->buffer_pointer = song->common_buffer;
	- song->buffered_count = 0;
	- }
	- if (count>0)
	- {
	- do_compute_data(song, count);
	- song->buffered_count += count;
	- song->buffer_pointer += (song->encoding & PE_MONO) ? count : count*2;
	- }
	-}
	-
	-void Timidity_Start(MidiSong *song)
	-{
	- song->playing = 1;
	- adjust_amplification(song);
	- skip_to(song, 0);
	-}
	-
	-void Timidity_Seek(MidiSong *song, Uint32 ms)
	-{
	- skip_to(song, (ms * song->rate) / 1000);
	-}
	-
	-int Timidity_PlaySome(MidiSong song, void stream, Sint32 len)
	-{
	- Sint32 start_sample, end_sample, samples;
	- int bytes_per_sample;
	-
	- if (!song->playing)
	- return 0;
	-
	- bytes_per_sample =
	- ((song->encoding & PE_MONO) ? 1 : 2)
	- * ((song->encoding & PE_16BIT) ? 2 : 1);
	- samples = len / bytes_per_sample;
	-
	- start_sample = song->current_sample;
	- end_sample = song->current_sample+samples;
	- while ( song->current_sample < end_sample ) {
	- /* Handle all events that should happen at this time */
	- while (song->current_event->time <= song->current_sample) {
	- switch(song->current_event->type) {
	-
	- /* Effects affecting a single note */
	-
	- case ME_NOTEON:
	- if (!(song->current_event->b)) /* Velocity 0? */
	- note_off(song);
	- else
	- note_on(song);
	- break;
	-
	- case ME_NOTEOFF:
	- note_off(song);
	- break;
	-
	- case ME_KEYPRESSURE:
	- adjust_pressure(song);
	- break;
	-
	- /* Effects affecting a single channel */
	-
	- case ME_PITCH_SENS:
	- song->channel[song->current_event->channel].pitchsens =
	- song->current_event->a;
	- song->channel[song->current_event->channel].pitchfactor = 0;
	- break;
	-
	- case ME_PITCHWHEEL:
	- song->channel[song->current_event->channel].pitchbend =
	- song->current_event->a + song->current_event->b * 128;
	- song->channel[song->current_event->channel].pitchfactor = 0;
	- /* Adjust pitch for notes already playing */
	- adjust_pitchbend(song);
	- break;
	-
	- case ME_MAINVOLUME:
	- song->channel[song->current_event->channel].volume =
	- song->current_event->a;
	- adjust_volume(song);
	- break;
	-
	- case ME_PAN:
	- song->channel[song->current_event->channel].panning =
	- song->current_event->a;
	- break;
	-
	- case ME_EXPRESSION:
	- song->channel[song->current_event->channel].expression =
	- song->current_event->a;
	- adjust_volume(song);
	- break;
	-
	- case ME_PROGRAM:
	- if (ISDRUMCHANNEL(song, song->current_event->channel)) {
	- /* Change drum set */
	- song->channel[song->current_event->channel].bank =
	- song->current_event->a;
	- }
	- else
	- song->channel[song->current_event->channel].program =
	- song->current_event->a;
	- break;
	-
	- case ME_SUSTAIN:
	- song->channel[song->current_event->channel].sustain =
	- song->current_event->a;
	- if (!song->current_event->a)
	- drop_sustain(song);
	- break;
	-
	- case ME_RESET_CONTROLLERS:
	- reset_controllers(song, song->current_event->channel);
	- break;
	-
	- case ME_ALL_NOTES_OFF:
	- all_notes_off(song);
	- break;
	-
	- case ME_ALL_SOUNDS_OFF:
	- all_sounds_off(song);
	- break;
	-
	- case ME_TONE_BANK:
	- song->channel[song->current_event->channel].bank =
	- song->current_event->a;
	- break;
	-
	- case ME_EOT:
	- /* Give the last notes a couple of seconds to decay */
	- SNDDBG(("Playing time: ~%d seconds\n",
	- song->current_sample/song->rate+2));
	- SNDDBG(("Notes cut: %d\n", song->cut_notes));
	- SNDDBG(("Notes lost totally: %d\n", song->lost_notes));
	- song->playing = 0;
	- return (song->current_sample - start_sample) * bytes_per_sample;
	- }
	- song->current_event++;
	- }
	- if (song->current_event->time > end_sample)
	- compute_data(song, stream, end_sample-song->current_sample);
	- else
	- compute_data(song, stream, song->current_event->time-song->current_sample);
	- }
	- return samples * bytes_per_sample;
	-}
	-
	-void Timidity_SetVolume(MidiSong *song, int volume)
	-{
	- int i;
	- if (volume > MAX_AMPLIFICATION)
	- song->amplification = MAX_AMPLIFICATION;
	- else
	- if (volume < 0)
	- song->amplification = 0;
	- else
	- song->amplification = volume;
	- adjust_amplification(song);
	- for (i = 0; i < song->voices; i++)
	- if (song->voice[i].status != VOICE_FREE)
	- {
	- recompute_amp(song, i);
	- apply_envelope_to_amp(song, i);
	- }
	-}
	diff --git a/util/sdl/sound/decoders/timidity/playmidi.h b/util/sdl/sound/decoders/timidity/playmidi.h
	deleted file mode 100644
	index b4545ab2..00000000
	--- a/util/sdl/sound/decoders/timidity/playmidi.h
	+++ /dev/null
	@@ -1,64 +0,0 @@
	-/*
	-
	- TiMidity -- Experimental MIDI to WAVE converter
	- Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi>
	-
	- This program is free software; you can redistribute it and/or modify
	- it under the terms of the GNU General Public License as published by
	- the Free Software Foundation; either version 2 of the License, or
	- (at your option) any later version.
	-
	- This program is distributed in the hope that it will be useful,
	- but WITHOUT ANY WARRANTY; without even the implied warranty of
	- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
	- GNU General Public License for more details.
	-
	- You should have received a copy of the GNU General Public License
	- along with this program; if not, write to the Free Software
	- Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
	-
	- playmidi.h
	-
	- */
	-
	-/* Midi events */
	-#define ME_NONE 0
	-#define ME_NOTEON 1
	-#define ME_NOTEOFF 2
	-#define ME_KEYPRESSURE 3
	-#define ME_MAINVOLUME 4
	-#define ME_PAN 5
	-#define ME_SUSTAIN 6
	-#define ME_EXPRESSION 7
	-#define ME_PITCHWHEEL 8
	-#define ME_PROGRAM 9
	-#define ME_TEMPO 10
	-#define ME_PITCH_SENS 11
	-
	-#define ME_ALL_SOUNDS_OFF 12
	-#define ME_RESET_CONTROLLERS 13
	-#define ME_ALL_NOTES_OFF 14
	-#define ME_TONE_BANK 15
	-
	-#define ME_LYRIC 16
	-
	-#define ME_EOT 99
	-
	-/* Causes the instrument's default panning to be used. */
	-#define NO_PANNING -1
	-
	-/* Voice status options: */
	-#define VOICE_FREE 0
	-#define VOICE_ON 1
	-#define VOICE_SUSTAINED 2
	-#define VOICE_OFF 3
	-#define VOICE_DIE 4
	-
	-/* Voice panned options: */
	-#define PANNED_MYSTERY 0
	-#define PANNED_LEFT 1
	-#define PANNED_RIGHT 2
	-#define PANNED_CENTER 3
	-/* Anything but PANNED_MYSTERY only uses the left volume */
	-
	-#define ISDRUMCHANNEL(s, c) (((s)->drumchannels & (1<<(c))))
	diff --git a/util/sdl/sound/decoders/timidity/readmidi.c b/util/sdl/sound/decoders/timidity/readmidi.c
	deleted file mode 100644
	index f3435f79..00000000
	--- a/util/sdl/sound/decoders/timidity/readmidi.c
	+++ /dev/null
	@@ -1,584 +0,0 @@
	-/*
	-
	- TiMidity -- Experimental MIDI to WAVE converter
	- Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi>
	-
	- This program is free software; you can redistribute it and/or modify
	- it under the terms of the GNU General Public License as published by
	- the Free Software Foundation; either version 2 of the License, or
	- (at your option) any later version.
	-
	- This program is distributed in the hope that it will be useful,
	- but WITHOUT ANY WARRANTY; without even the implied warranty of
	- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
	- GNU General Public License for more details.
	-
	- You should have received a copy of the GNU General Public License
	- along with this program; if not, write to the Free Software
	- Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
	-
	-*/
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#include <stdio.h>
	-#include <stdlib.h>
	-#include <string.h>
	-
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-#include "timidity.h"
	-#include "common.h"
	-#include "instrum.h"
	-#include "playmidi.h"
	-
	-/* Computes how many (fractional) samples one MIDI delta-time unit contains */
	-static void compute_sample_increment(MidiSong *song, Sint32 tempo,
	- Sint32 divisions)
	-{
	- double a;
	- a = (double) (tempo) * (double) (song->rate) * (65536.0/1000000.0) /
	- (double)(divisions);
	-
	- song->sample_correction = (Sint32)(a) & 0xFFFF;
	- song->sample_increment = (Sint32)(a) >> 16;
	-
	- SNDDBG(("Samples per delta-t: %d (correction %d)",
	- song->sample_increment, song->sample_correction));
	-}
	-
	-/* Read variable-length number (7 bits per byte, MSB first) */
	-static Sint32 getvl(SDL_RWops *rw)
	-{
	- Sint32 l=0;
	- Uint8 c;
	- for (;;)
	- {
	- SDL_RWread(rw, &c, 1, 1);
	- l += (c & 0x7f);
	- if (!(c & 0x80)) return l;
	- l<<=7;
	- }
	-}
	-
	-/* Print a string from the file, followed by a newline. Any non-ASCII
	- or unprintable characters will be converted to periods. */
	-static int dumpstring(SDL_RWops rw, Sint32 len, char label)
	-{
	- signed char *s=safe_malloc(len+1);
	- if (len != (Sint32) SDL_RWread(rw, s, 1, len))
	- {
	- free(s);
	- return -1;
	- }
	- s[len]='\0';
	- while (len--)
	- {
	- if (s[len]<32)
	- s[len]='.';
	- }
	- SNDDBG(("%s%s", label, s));
	- free(s);
	- return 0;
	-}
	-
	-#define MIDIEVENT(at,t,ch,pa,pb) \
	- new=safe_malloc(sizeof(MidiEventList)); \
	- new->event.time=at; new->event.type=t; new->event.channel=ch; \
	- new->event.a=pa; new->event.b=pb; new->next=0;\
	- return new;
	-
	-#define MAGIC_EOT ((MidiEventList *)(-1))
	-
	-/* Read a MIDI event, returning a freshly allocated element that can
	- be linked to the event list */
	-static MidiEventList read_midi_event(MidiSong song)
	-{
	- static Uint8 laststatus, lastchan;
	- static Uint8 nrpn=0, rpn_msb[16], rpn_lsb[16]; /* one per channel */
	- Uint8 me, type, a,b,c;
	- Sint32 len;
	- MidiEventList *new;
	-
	- for (;;)
	- {
	- song->at += getvl(song->rw);
	- if (SDL_RWread(song->rw, &me, 1, 1) != 1)
	- {
	- SNDDBG(("read_midi_event: SDL_RWread() failure\n"));
	- return 0;
	- }
	-
	- if(me==0xF0 \|\| me == 0xF7) /* SysEx event */
	- {
	- len=getvl(song->rw);
	- SDL_RWseek(song->rw, len, SEEK_CUR);
	- }
	- else if(me==0xFF) /* Meta event */
	- {
	- SDL_RWread(song->rw, &type, 1, 1);
	- len=getvl(song->rw);
	- if (type>0 && type<16)
	- {
	- static char *label[]={
	- "Text event: ", "Text: ", "Copyright: ", "Track name: ",
	- "Instrument: ", "Lyric: ", "Marker: ", "Cue point: "};
	- dumpstring(song->rw, len, label[(type>7) ? 0 : type]);
	- }
	- else
	- switch(type)
	- {
	- case 0x2F: /* End of Track */
	- return MAGIC_EOT;
	-
	- case 0x51: /* Tempo */
	- SDL_RWread(song->rw, &a, 1, 1);
	- SDL_RWread(song->rw, &b, 1, 1);
	- SDL_RWread(song->rw, &c, 1, 1);
	- MIDIEVENT(song->at, ME_TEMPO, c, a, b);
	-
	- default:
	- SNDDBG(("(Meta event type 0x%02x, length %d)\n", type, len));
	- SDL_RWseek(song->rw, len, SEEK_CUR);
	- break;
	- }
	- }
	- else
	- {
	- a=me;
	- if (a & 0x80) /* status byte */
	- {
	- lastchan=a & 0x0F;
	- laststatus=(a>>4) & 0x07;
	- SDL_RWread(song->rw, &a, 1, 1);
	- a &= 0x7F;
	- }
	- switch(laststatus)
	- {
	- case 0: /* Note off */
	- SDL_RWread(song->rw, &b, 1, 1);
	- b &= 0x7F;
	- MIDIEVENT(song->at, ME_NOTEOFF, lastchan, a,b);
	-
	- case 1: /* Note on */
	- SDL_RWread(song->rw, &b, 1, 1);
	- b &= 0x7F;
	- MIDIEVENT(song->at, ME_NOTEON, lastchan, a,b);
	-
	- case 2: /* Key Pressure */
	- SDL_RWread(song->rw, &b, 1, 1);
	- b &= 0x7F;
	- MIDIEVENT(song->at, ME_KEYPRESSURE, lastchan, a, b);
	-
	- case 3: /* Control change */
	- SDL_RWread(song->rw, &b, 1, 1);
	- b &= 0x7F;
	- {
	- int control=255;
	- switch(a)
	- {
	- case 7: control=ME_MAINVOLUME; break;
	- case 10: control=ME_PAN; break;
	- case 11: control=ME_EXPRESSION; break;
	- case 64: control=ME_SUSTAIN; break;
	- case 120: control=ME_ALL_SOUNDS_OFF; break;
	- case 121: control=ME_RESET_CONTROLLERS; break;
	- case 123: control=ME_ALL_NOTES_OFF; break;
	-
	- /* These should be the SCC-1 tone bank switch
	- commands. I don't know why there are two, or
	- why the latter only allows switching to bank 0.
	- Also, some MIDI files use 0 as some sort of
	- continuous controller. This will cause lots of
	- warnings about undefined tone banks. */
	- case 0: control=ME_TONE_BANK; break;
	- case 32:
	- if (b!=0)
	- SNDDBG(("(Strange: tone bank change 0x20%02x)\n", b));
	- else
	- control=ME_TONE_BANK;
	- break;
	-
	- case 100: nrpn=0; rpn_msb[lastchan]=b; break;
	- case 101: nrpn=0; rpn_lsb[lastchan]=b; break;
	- case 99: nrpn=1; rpn_msb[lastchan]=b; break;
	- case 98: nrpn=1; rpn_lsb[lastchan]=b; break;
	-
	- case 6:
	- if (nrpn)
	- {
	- SNDDBG(("(Data entry (MSB) for NRPN %02x,%02x: %d)\n",
	- rpn_msb[lastchan], rpn_lsb[lastchan], b));
	- break;
	- }
	-
	- switch((rpn_msb[lastchan]<<8) \| rpn_lsb[lastchan])
	- {
	- case 0x0000: /* Pitch bend sensitivity */
	- control=ME_PITCH_SENS;
	- break;
	-
	- case 0x7F7F: /* RPN reset */
	- /* reset pitch bend sensitivity to 2 */
	- MIDIEVENT(song->at, ME_PITCH_SENS, lastchan, 2, 0);
	-
	- default:
	- SNDDBG(("(Data entry (MSB) for RPN %02x,%02x: %d)\n",
	- rpn_msb[lastchan], rpn_lsb[lastchan], b));
	- break;
	- }
	- break;
	-
	- default:
	- SNDDBG(("(Control %d: %d)\n", a, b));
	- break;
	- }
	- if (control != 255)
	- {
	- MIDIEVENT(song->at, control, lastchan, b, 0);
	- }
	- }
	- break;
	-
	- case 4: /* Program change */
	- a &= 0x7f;
	- MIDIEVENT(song->at, ME_PROGRAM, lastchan, a, 0);
	-
	- case 5: /* Channel pressure - NOT IMPLEMENTED */
	- break;
	-
	- case 6: /* Pitch wheel */
	- SDL_RWread(song->rw, &b, 1, 1);
	- b &= 0x7F;
	- MIDIEVENT(song->at, ME_PITCHWHEEL, lastchan, a, b);
	-
	- default:
	- SNDDBG(("*** Can't happen: status 0x%02X, channel 0x%02X\n",
	- laststatus, lastchan));
	- break;
	- }
	- }
	- }
	-
	- return new;
	-}
	-
	-#undef MIDIEVENT
	-
	-/* Read a midi track into the linked list, either merging with any previous
	- tracks or appending to them. */
	-static int read_track(MidiSong *song, int append)
	-{
	- MidiEventList *meep;
	- MidiEventList next, new;
	- Sint32 len;
	- char tmp[4];
	-
	- meep = song->evlist;
	- if (append && meep)
	- {
	- /* find the last event in the list */
	- for (; meep->next; meep=meep->next)
	- ;
	- song->at = meep->event.time;
	- }
	- else
	- song->at=0;
	-
	- /* Check the formalities */
	-
	- if (SDL_RWread(song->rw, tmp, 1, 4) != 4 \|\| SDL_RWread(song->rw, &len, 4, 1) != 1)
	- {
	- SNDDBG(("Can't read track header.\n"));
	- return -1;
	- }
	- len=SDL_SwapBE32(len);
	- if (memcmp(tmp, "MTrk", 4))
	- {
	- SNDDBG(("Corrupt MIDI file.\n"));
	- return -2;
	- }
	-
	- for (;;)
	- {
	- if (!(new=read_midi_event(song))) /* Some kind of error */
	- return -2;
	-
	- if (new==MAGIC_EOT) /* End-of-track Hack. */
	- {
	- return 0;
	- }
	-
	- next=meep->next;
	- while (next && (next->event.time < new->event.time))
	- {
	- meep=next;
	- next=meep->next;
	- }
	-
	- new->next=next;
	- meep->next=new;
	-
	- song->event_count++; /* Count the event. (About one?) */
	- meep=new;
	- }
	-}
	-
	-/* Free the linked event list from memory. */
	-static void free_midi_list(MidiSong *song)
	-{
	- MidiEventList meep, next;
	- if (!(meep = song->evlist)) return;
	- while (meep)
	- {
	- next=meep->next;
	- free(meep);
	- meep=next;
	- }
	- song->evlist=0;
	-}
	-
	-/* Allocate an array of MidiEvents and fill it from the linked list of
	- events, marking used instruments for loading. Convert event times to
	- samples: handle tempo changes. Strip unnecessary events from the list.
	- Free the linked list. */
	-static MidiEvent groom_list(MidiSong song, Sint32 divisions,Sint32 *eventsp,
	- Sint32 *samplesp)
	-{
	- MidiEvent groomed_list, lp;
	- MidiEventList *meep;
	- Sint32 i, our_event_count, tempo, skip_this_event, new_value;
	- Sint32 sample_cum, samples_to_do, at, st, dt, counting_time;
	-
	- int current_bank[16], current_set[16], current_program[16];
	- /* Or should each bank have its own current program? */
	-
	- for (i=0; i<16; i++)
	- {
	- current_bank[i]=0;
	- current_set[i]=0;
	- current_program[i]=song->default_program;
	- }
	-
	- tempo=500000;
	- compute_sample_increment(song, tempo, divisions);
	-
	- /* This may allocate a bit more than we need */
	- groomed_list=lp=safe_malloc(sizeof(MidiEvent) * (song->event_count+1));
	- meep=song->evlist;
	-
	- our_event_count=0;
	- st=at=sample_cum=0;
	- counting_time=2; /* We strip any silence before the first NOTE ON. */
	-
	- for (i = 0; i < song->event_count; i++)
	- {
	- skip_this_event=0;
	-
	- if (meep->event.type==ME_TEMPO)
	- {
	- tempo=
	- meep->event.channel + meep->event.b * 256 + meep->event.a * 65536;
	- compute_sample_increment(song, tempo, divisions);
	- skip_this_event=1;
	- }
	- else switch (meep->event.type)
	- {
	- case ME_PROGRAM:
	- if (ISDRUMCHANNEL(song, meep->event.channel))
	- {
	- if (song->drumset[meep->event.a]) /* Is this a defined drumset? */
	- new_value=meep->event.a;
	- else
	- {
	- SNDDBG(("Drum set %d is undefined\n", meep->event.a));
	- new_value=meep->event.a=0;
	- }
	- if (current_set[meep->event.channel] != new_value)
	- current_set[meep->event.channel]=new_value;
	- else
	- skip_this_event=1;
	- }
	- else
	- {
	- new_value=meep->event.a;
	- if ((current_program[meep->event.channel] != SPECIAL_PROGRAM)
	- && (current_program[meep->event.channel] != new_value))
	- current_program[meep->event.channel] = new_value;
	- else
	- skip_this_event=1;
	- }
	- break;
	-
	- case ME_NOTEON:
	- if (counting_time)
	- counting_time=1;
	- if (ISDRUMCHANNEL(song, meep->event.channel))
	- {
	- /* Mark this instrument to be loaded */
	- if (!(song->drumset[current_set[meep->event.channel]]
	- ->instrument[meep->event.a]))
	- song->drumset[current_set[meep->event.channel]]
	- ->instrument[meep->event.a] = MAGIC_LOAD_INSTRUMENT;
	- }
	- else
	- {
	- if (current_program[meep->event.channel]==SPECIAL_PROGRAM)
	- break;
	- /* Mark this instrument to be loaded */
	- if (!(song->tonebank[current_bank[meep->event.channel]]
	- ->instrument[current_program[meep->event.channel]]))
	- song->tonebank[current_bank[meep->event.channel]]
	- ->instrument[current_program[meep->event.channel]] =
	- MAGIC_LOAD_INSTRUMENT;
	- }
	- break;
	-
	- case ME_TONE_BANK:
	- if (ISDRUMCHANNEL(song, meep->event.channel))
	- {
	- skip_this_event=1;
	- break;
	- }
	- if (song->tonebank[meep->event.a]) /* Is this a defined tone bank? */
	- new_value=meep->event.a;
	- else
	- {
	- SNDDBG(("Tone bank %d is undefined\n", meep->event.a));
	- new_value=meep->event.a=0;
	- }
	- if (current_bank[meep->event.channel]!=new_value)
	- current_bank[meep->event.channel]=new_value;
	- else
	- skip_this_event=1;
	- break;
	- }
	-
	- /* Recompute time in samples*/
	- if ((dt=meep->event.time - at) && !counting_time)
	- {
	- samples_to_do = song->sample_increment * dt;
	- sample_cum += song->sample_correction * dt;
	- if (sample_cum & 0xFFFF0000)
	- {
	- samples_to_do += ((sample_cum >> 16) & 0xFFFF);
	- sample_cum &= 0x0000FFFF;
	- }
	- st += samples_to_do;
	- }
	- else if (counting_time==1) counting_time=0;
	- if (!skip_this_event)
	- {
	- /* Add the event to the list */
	- *lp=meep->event;
	- lp->time=st;
	- lp++;
	- our_event_count++;
	- }
	- at=meep->event.time;
	- meep=meep->next;
	- }
	- /* Add an End-of-Track event */
	- lp->time=st;
	- lp->type=ME_EOT;
	- our_event_count++;
	- free_midi_list(song);
	-
	- *eventsp=our_event_count;
	- *samplesp=st;
	- return groomed_list;
	-}
	-
	-MidiEvent read_midi_file(MidiSong song, Sint32 count, Sint32 sp)
	-{
	- Sint32 len, divisions;
	- Sint16 format, tracks, divisions_tmp;
	- int i;
	- char tmp[4];
	-
	- song->event_count=0;
	- song->at=0;
	- song->evlist=0;
	-
	- if (SDL_RWread(song->rw, tmp, 1, 4) != 4 \|\| SDL_RWread(song->rw, &len, 4, 1) != 1)
	- {
	- SNDDBG(("Not a MIDI file!\n"));
	- return 0;
	- }
	- len=SDL_SwapBE32(len);
	- if (memcmp(tmp, "MThd", 4) \|\| len < 6)
	- {
	- SNDDBG(("Not a MIDI file!\n"));
	- return 0;
	- }
	-
	- SDL_RWread(song->rw, &format, 2, 1);
	- SDL_RWread(song->rw, &tracks, 2, 1);
	- SDL_RWread(song->rw, &divisions_tmp, 2, 1);
	- format=SDL_SwapBE16(format);
	- tracks=SDL_SwapBE16(tracks);
	- divisions_tmp=SDL_SwapBE16(divisions_tmp);
	-
	- if (divisions_tmp<0)
	- {
	- /* SMPTE time -- totally untested. Got a MIDI file that uses this? */
	- divisions=
	- (Sint32)(-(divisions_tmp/256)) * (Sint32)(divisions_tmp & 0xFF);
	- }
	- else divisions=(Sint32)(divisions_tmp);
	-
	- if (len > 6)
	- {
	- SNDDBG(("MIDI file header size %u bytes", len));
	- SDL_RWseek(song->rw, len-6, SEEK_CUR); /* skip the excess */
	- }
	- if (format<0 \|\| format >2)
	- {
	- SNDDBG(("Unknown MIDI file format %d\n", format));
	- return 0;
	- }
	- SNDDBG(("Format: %d Tracks: %d Divisions: %d\n",
	- format, tracks, divisions));
	-
	- /* Put a do-nothing event first in the list for easier processing */
	- song->evlist=safe_malloc(sizeof(MidiEventList));
	- song->evlist->event.time=0;
	- song->evlist->event.type=ME_NONE;
	- song->evlist->next=0;
	- song->event_count++;
	-
	- switch(format)
	- {
	- case 0:
	- if (read_track(song, 0))
	- {
	- free_midi_list(song);
	- return 0;
	- }
	- break;
	-
	- case 1:
	- for (i=0; i<tracks; i++)
	- if (read_track(song, 0))
	- {
	- free_midi_list(song);
	- return 0;
	- }
	- break;
	-
	- case 2: /* We simply play the tracks sequentially */
	- for (i=0; i<tracks; i++)
	- if (read_track(song, 1))
	- {
	- free_midi_list(song);
	- return 0;
	- }
	- break;
	- }
	- return groom_list(song, divisions, count, sp);
	-}
	diff --git a/util/sdl/sound/decoders/timidity/readmidi.h b/util/sdl/sound/decoders/timidity/readmidi.h
	deleted file mode 100644
	index 0d129a05..00000000
	--- a/util/sdl/sound/decoders/timidity/readmidi.h
	+++ /dev/null
	@@ -1,24 +0,0 @@
	-/*
	-
	- TiMidity -- Experimental MIDI to WAVE converter
	- Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi>
	-
	- This program is free software; you can redistribute it and/or modify
	- it under the terms of the GNU General Public License as published by
	- the Free Software Foundation; either version 2 of the License, or
	- (at your option) any later version.
	-
	- This program is distributed in the hope that it will be useful,
	- but WITHOUT ANY WARRANTY; without even the implied warranty of
	- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
	- GNU General Public License for more details.
	-
	- You should have received a copy of the GNU General Public License
	- along with this program; if not, write to the Free Software
	- Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
	-
	- readmidi.h
	-
	- */
	-
	-extern MidiEvent read_midi_file(MidiSong song, Sint32 count, Sint32 sp);
	diff --git a/util/sdl/sound/decoders/timidity/resample.c b/util/sdl/sound/decoders/timidity/resample.c
	deleted file mode 100644
	index 31c739ca..00000000
	--- a/util/sdl/sound/decoders/timidity/resample.c
	+++ /dev/null
	@@ -1,612 +0,0 @@
	-/*
	-
	- TiMidity -- Experimental MIDI to WAVE converter
	- Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi>
	-
	- This program is free software; you can redistribute it and/or modify
	- it under the terms of the GNU General Public License as published by
	- the Free Software Foundation; either version 2 of the License, or
	- (at your option) any later version.
	-
	- This program is distributed in the hope that it will be useful,
	- but WITHOUT ANY WARRANTY; without even the implied warranty of
	- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
	- GNU General Public License for more details.
	-
	- You should have received a copy of the GNU General Public License
	- along with this program; if not, write to the Free Software
	- Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
	-
	- resample.c
	-*/
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#include <math.h>
	-#include <stdio.h>
	-#include <stdlib.h>
	-
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-#include "timidity.h"
	-#include "options.h"
	-#include "common.h"
	-#include "instrum.h"
	-#include "playmidi.h"
	-#include "tables.h"
	-#include "resample.h"
	-
	-/************* resampling with fixed increment ***************/
	-
	-static sample_t rs_plain(MidiSong song, int v, Sint32 *countptr)
	-{
	-
	- /* Play sample until end, then free the voice. */
	-
	- sample_t v1, v2;
	- Voice
	- *vp=&(song->voice[v]);
	- sample_t
	- *dest=song->resample_buffer,
	- *src=vp->sample->data;
	- Sint32
	- ofs=vp->sample_offset,
	- incr=vp->sample_increment,
	- le=vp->sample->data_length,
	- count=*countptr;
	- Sint32 i;
	-
	- if (incr<0) incr = -incr; /* In case we're coming out of a bidir loop */
	-
	- /* Precalc how many times we should go through the loop.
	- NOTE: Assumes that incr > 0 and that ofs <= le */
	- i = (le - ofs) / incr + 1;
	-
	- if (i > count)
	- {
	- i = count;
	- count = 0;
	- }
	- else count -= i;
	-
	- while (i--)
	- {
	- v1 = src[ofs >> FRACTION_BITS];
	- v2 = src[(ofs >> FRACTION_BITS)+1];
	- dest++ = v1 + (((v2 - v1) (ofs & FRACTION_MASK)) >> FRACTION_BITS);
	- ofs += incr;
	- }
	-
	- if (ofs >= le)
	- {
	- if (ofs == le)
	- *dest++ = src[ofs >> FRACTION_BITS];
	- vp->status=VOICE_FREE;
	- *countptr-=count+1;
	- }
	-
	- vp->sample_offset=ofs; /* Update offset */
	- return song->resample_buffer;
	-}
	-
	-static sample_t rs_loop(MidiSong song, Voice *vp, Sint32 count)
	-{
	-
	- /* Play sample until end-of-loop, skip back and continue. */
	-
	- sample_t v1, v2;
	- Sint32
	- ofs=vp->sample_offset,
	- incr=vp->sample_increment,
	- le=vp->sample->loop_end,
	- ll=le - vp->sample->loop_start;
	- sample_t
	- *dest=song->resample_buffer,
	- *src=vp->sample->data;
	- Sint32 i;
	-
	- while (count)
	- {
	- if (ofs >= le)
	- /* NOTE: Assumes that ll > incr and that incr > 0. */
	- ofs -= ll;
	- /* Precalc how many times we should go through the loop */
	- i = (le - ofs) / incr + 1;
	- if (i > count)
	- {
	- i = count;
	- count = 0;
	- }
	- else count -= i;
	- while (i--)
	- {
	- v1 = src[ofs >> FRACTION_BITS];
	- v2 = src[(ofs >> FRACTION_BITS)+1];
	- dest++ = v1 + (((v2 - v1) (ofs & FRACTION_MASK)) >> FRACTION_BITS);
	- ofs += incr;
	- }
	- }
	-
	- vp->sample_offset=ofs; /* Update offset */
	- return song->resample_buffer;
	-}
	-
	-static sample_t rs_bidir(MidiSong song, Voice *vp, Sint32 count)
	-{
	- sample_t v1, v2;
	- Sint32
	- ofs=vp->sample_offset,
	- incr=vp->sample_increment,
	- le=vp->sample->loop_end,
	- ls=vp->sample->loop_start;
	- sample_t
	- *dest=song->resample_buffer,
	- *src=vp->sample->data;
	- Sint32
	- le2 = le<<1,
	- ls2 = ls<<1,
	- i;
	- /* Play normally until inside the loop region */
	-
	- if (ofs <= ls)
	- {
	- /* NOTE: Assumes that incr > 0, which is NOT always the case
	- when doing bidirectional looping. I have yet to see a case
	- where both ofs <= ls AND incr < 0, however. */
	- i = (ls - ofs) / incr + 1;
	- if (i > count)
	- {
	- i = count;
	- count = 0;
	- }
	- else count -= i;
	- while (i--)
	- {
	- v1 = src[ofs >> FRACTION_BITS];
	- v2 = src[(ofs >> FRACTION_BITS)+1];
	- dest++ = v1 + (((v2 - v1) (ofs & FRACTION_MASK)) >> FRACTION_BITS);
	- ofs += incr;
	- }
	- }
	-
	- /* Then do the bidirectional looping */
	-
	- while(count)
	- {
	- /* Precalc how many times we should go through the loop */
	- i = ((incr > 0 ? le : ls) - ofs) / incr + 1;
	- if (i > count)
	- {
	- i = count;
	- count = 0;
	- }
	- else count -= i;
	- while (i--)
	- {
	- v1 = src[ofs >> FRACTION_BITS];
	- v2 = src[(ofs >> FRACTION_BITS)+1];
	- dest++ = v1 + (((v2 - v1) (ofs & FRACTION_MASK)) >> FRACTION_BITS);
	- ofs += incr;
	- }
	- if (ofs>=le)
	- {
	- /* fold the overshoot back in */
	- ofs = le2 - ofs;
	- incr *= -1;
	- }
	- else if (ofs <= ls)
	- {
	- ofs = ls2 - ofs;
	- incr *= -1;
	- }
	- }
	-
	- vp->sample_increment=incr;
	- vp->sample_offset=ofs; /* Update offset */
	- return song->resample_buffer;
	-}
	-
	-/********************* vibrato versions *************************/
	-
	-/* We only need to compute one half of the vibrato sine cycle */
	-static int vib_phase_to_inc_ptr(int phase)
	-{
	- if (phase < VIBRATO_SAMPLE_INCREMENTS/2)
	- return VIBRATO_SAMPLE_INCREMENTS/2-1-phase;
	- else if (phase >= 3*VIBRATO_SAMPLE_INCREMENTS/2)
	- return 5*VIBRATO_SAMPLE_INCREMENTS/2-1-phase;
	- else
	- return phase-VIBRATO_SAMPLE_INCREMENTS/2;
	-}
	-
	-static Sint32 update_vibrato(MidiSong song, Voice vp, int sign)
	-{
	- Sint32 depth;
	- int phase, pb;
	- double a;
	-
	- if (vp->vibrato_phase++ >= 2*VIBRATO_SAMPLE_INCREMENTS-1)
	- vp->vibrato_phase=0;
	- phase=vib_phase_to_inc_ptr(vp->vibrato_phase);
	-
	- if (vp->vibrato_sample_increment[phase])
	- {
	- if (sign)
	- return -vp->vibrato_sample_increment[phase];
	- else
	- return vp->vibrato_sample_increment[phase];
	- }
	-
	- /* Need to compute this sample increment. */
	-
	- depth=vp->sample->vibrato_depth<<7;
	-
	- if (vp->vibrato_sweep)
	- {
	- /* Need to update sweep */
	- vp->vibrato_sweep_position += vp->vibrato_sweep;
	- if (vp->vibrato_sweep_position >= (1<<SWEEP_SHIFT))
	- vp->vibrato_sweep=0;
	- else
	- {
	- /* Adjust depth */
	- depth *= vp->vibrato_sweep_position;
	- depth >>= SWEEP_SHIFT;
	- }
	- }
	-
	- a = FSCALE(((double)(vp->sample->sample_rate) *
	- (double)(vp->frequency)) /
	- ((double)(vp->sample->root_freq) *
	- (double)(song->rate)),
	- FRACTION_BITS);
	-
	- pb=(int)((sine(vp->vibrato_phase *
	- (SINE_CYCLE_LENGTH/(2*VIBRATO_SAMPLE_INCREMENTS)))
	- * (double)(depth) * VIBRATO_AMPLITUDE_TUNING));
	-
	- if (pb<0)
	- {
	- pb=-pb;
	- a /= bend_fine[(pb>>5) & 0xFF] * bend_coarse[pb>>13];
	- }
	- else
	- a = bend_fine[(pb>>5) & 0xFF] bend_coarse[pb>>13];
	-
	- /* If the sweep's over, we can store the newly computed sample_increment */
	- if (!vp->vibrato_sweep)
	- vp->vibrato_sample_increment[phase]=(Sint32) a;
	-
	- if (sign)
	- a = -a; /* need to preserve the loop direction */
	-
	- return (Sint32) a;
	-}
	-
	-static sample_t rs_vib_plain(MidiSong song, int v, Sint32 *countptr)
	-{
	-
	- /* Play sample until end, then free the voice. */
	-
	- sample_t v1, v2;
	- Voice *vp=&(song->voice[v]);
	- sample_t
	- *dest=song->resample_buffer,
	- *src=vp->sample->data;
	- Sint32
	- le=vp->sample->data_length,
	- ofs=vp->sample_offset,
	- incr=vp->sample_increment,
	- count=*countptr;
	- int
	- cc=vp->vibrato_control_counter;
	-
	- /* This has never been tested */
	-
	- if (incr<0) incr = -incr; /* In case we're coming out of a bidir loop */
	-
	- while (count--)
	- {
	- if (!cc--)
	- {
	- cc=vp->vibrato_control_ratio;
	- incr=update_vibrato(song, vp, 0);
	- }
	- v1 = src[ofs >> FRACTION_BITS];
	- v2 = src[(ofs >> FRACTION_BITS)+1];
	- dest++ = v1 + (((v2 - v1) (ofs & FRACTION_MASK)) >> FRACTION_BITS);
	- ofs += incr;
	- if (ofs >= le)
	- {
	- if (ofs == le)
	- *dest++ = src[ofs >> FRACTION_BITS];
	- vp->status=VOICE_FREE;
	- *countptr-=count+1;
	- break;
	- }
	- }
	-
	- vp->vibrato_control_counter=cc;
	- vp->sample_increment=incr;
	- vp->sample_offset=ofs; /* Update offset */
	- return song->resample_buffer;
	-}
	-
	-static sample_t rs_vib_loop(MidiSong song, Voice *vp, Sint32 count)
	-{
	-
	- /* Play sample until end-of-loop, skip back and continue. */
	-
	- sample_t v1, v2;
	- Sint32
	- ofs=vp->sample_offset,
	- incr=vp->sample_increment,
	- le=vp->sample->loop_end,
	- ll=le - vp->sample->loop_start;
	- sample_t
	- *dest=song->resample_buffer,
	- *src=vp->sample->data;
	- int
	- cc=vp->vibrato_control_counter;
	- Sint32 i;
	- int
	- vibflag=0;
	-
	- while (count)
	- {
	- /* Hopefully the loop is longer than an increment */
	- if(ofs >= le)
	- ofs -= ll;
	- /* Precalc how many times to go through the loop, taking
	- the vibrato control ratio into account this time. */
	- i = (le - ofs) / incr + 1;
	- if(i > count) i = count;
	- if(i > cc)
	- {
	- i = cc;
	- vibflag = 1;
	- }
	- else cc -= i;
	- count -= i;
	- while(i--)
	- {
	- v1 = src[ofs >> FRACTION_BITS];
	- v2 = src[(ofs >> FRACTION_BITS)+1];
	- dest++ = v1 + (((v2 - v1) (ofs & FRACTION_MASK)) >> FRACTION_BITS);
	- ofs += incr;
	- }
	- if(vibflag)
	- {
	- cc = vp->vibrato_control_ratio;
	- incr = update_vibrato(song, vp, 0);
	- vibflag = 0;
	- }
	- }
	-
	- vp->vibrato_control_counter=cc;
	- vp->sample_increment=incr;
	- vp->sample_offset=ofs; /* Update offset */
	- return song->resample_buffer;
	-}
	-
	-static sample_t rs_vib_bidir(MidiSong song, Voice *vp, Sint32 count)
	-{
	- sample_t v1, v2;
	- Sint32
	- ofs=vp->sample_offset,
	- incr=vp->sample_increment,
	- le=vp->sample->loop_end,
	- ls=vp->sample->loop_start;
	- sample_t
	- *dest=song->resample_buffer,
	- *src=vp->sample->data;
	- int
	- cc=vp->vibrato_control_counter;
	- Sint32
	- le2=le<<1,
	- ls2=ls<<1,
	- i;
	- int
	- vibflag = 0;
	-
	- /* Play normally until inside the loop region */
	- while (count && (ofs <= ls))
	- {
	- i = (ls - ofs) / incr + 1;
	- if (i > count) i = count;
	- if (i > cc)
	- {
	- i = cc;
	- vibflag = 1;
	- }
	- else cc -= i;
	- count -= i;
	- while (i--)
	- {
	- v1 = src[ofs >> FRACTION_BITS];
	- v2 = src[(ofs >> FRACTION_BITS)+1];
	- dest++ = v1 + (((v2 - v1) (ofs & FRACTION_MASK)) >> FRACTION_BITS);
	- ofs += incr;
	- }
	- if (vibflag)
	- {
	- cc = vp->vibrato_control_ratio;
	- incr = update_vibrato(song, vp, 0);
	- vibflag = 0;
	- }
	- }
	-
	- /* Then do the bidirectional looping */
	-
	- while (count)
	- {
	- /* Precalc how many times we should go through the loop */
	- i = ((incr > 0 ? le : ls) - ofs) / incr + 1;
	- if(i > count) i = count;
	- if(i > cc)
	- {
	- i = cc;
	- vibflag = 1;
	- }
	- else cc -= i;
	- count -= i;
	- while (i--)
	- {
	- v1 = src[ofs >> FRACTION_BITS];
	- v2 = src[(ofs >> FRACTION_BITS)+1];
	- dest++ = v1 + (((v2 - v1) (ofs & FRACTION_MASK)) >> FRACTION_BITS);
	- ofs += incr;
	- }
	- if (vibflag)
	- {
	- cc = vp->vibrato_control_ratio;
	- incr = update_vibrato(song, vp, (incr < 0));
	- vibflag = 0;
	- }
	- if (ofs >= le)
	- {
	- /* fold the overshoot back in */
	- ofs = le2 - ofs;
	- incr *= -1;
	- }
	- else if (ofs <= ls)
	- {
	- ofs = ls2 - ofs;
	- incr *= -1;
	- }
	- }
	-
	- vp->vibrato_control_counter=cc;
	- vp->sample_increment=incr;
	- vp->sample_offset=ofs; /* Update offset */
	- return song->resample_buffer;
	-}
	-
	-sample_t resample_voice(MidiSong song, int v, Sint32 *countptr)
	-{
	- Sint32 ofs;
	- Uint8 modes;
	- Voice *vp=&(song->voice[v]);
	-
	- if (!(vp->sample->sample_rate))
	- {
	- /* Pre-resampled data -- just update the offset and check if
	- we're out of data. */
	- ofs=vp->sample_offset >> FRACTION_BITS; /* Kind of silly to use
	- FRACTION_BITS here... */
	- if (*countptr >= (vp->sample->data_length>>FRACTION_BITS) - ofs)
	- {
	- /* Note finished. Free the voice. */
	- vp->status = VOICE_FREE;
	-
	- /* Let the caller know how much data we had left */
	- *countptr = (vp->sample->data_length>>FRACTION_BITS) - ofs;
	- }
	- else
	- vp->sample_offset += *countptr << FRACTION_BITS;
	-
	- return vp->sample->data+ofs;
	- }
	-
	- /* Need to resample. Use the proper function. */
	- modes=vp->sample->modes;
	-
	- if (vp->vibrato_control_ratio)
	- {
	- if ((modes & MODES_LOOPING) &&
	- ((modes & MODES_ENVELOPE) \|\|
	- (vp->status==VOICE_ON \|\| vp->status==VOICE_SUSTAINED)))
	- {
	- if (modes & MODES_PINGPONG)
	- return rs_vib_bidir(song, vp, *countptr);
	- else
	- return rs_vib_loop(song, vp, *countptr);
	- }
	- else
	- return rs_vib_plain(song, v, countptr);
	- }
	- else
	- {
	- if ((modes & MODES_LOOPING) &&
	- ((modes & MODES_ENVELOPE) \|\|
	- (vp->status==VOICE_ON \|\| vp->status==VOICE_SUSTAINED)))
	- {
	- if (modes & MODES_PINGPONG)
	- return rs_bidir(song, vp, *countptr);
	- else
	- return rs_loop(song, vp, *countptr);
	- }
	- else
	- return rs_plain(song, v, countptr);
	- }
	-}
	-
	-void pre_resample(MidiSong song, Sample sp)
	-{
	- double a, xdiff;
	- Sint32 incr, ofs, newlen, count;
	- Sint16 newdata, dest, src = (Sint16 ) sp->data;
	- Sint16 v1, v2, v3, v4, *vptr;
	-#ifdef DEBUG_CHATTER
	- static const char note_name[12][3] =
	- {
	- "C", "C#", "D", "D#", "E", "F", "F#", "G", "G#", "A", "A#", "B"
	- };
	-#endif
	-
	- SNDDBG((" * pre-resampling for note %d (%s%d)\n",
	- sp->note_to_use,
	- note_name[sp->note_to_use % 12], (sp->note_to_use & 0x7F) / 12));
	-
	- a = ((double) (sp->sample_rate) * freq_table[(int) (sp->note_to_use)]) /
	- ((double) (sp->root_freq) * song->rate);
	- newlen = (Sint32)(sp->data_length / a);
	- dest = newdata = safe_malloc(newlen >> (FRACTION_BITS - 1));
	-
	- count = (newlen >> FRACTION_BITS) - 1;
	- ofs = incr = (sp->data_length - (1 << FRACTION_BITS)) / count;
	-
	- if (--count)
	- *dest++ = src[0];
	-
	- /* Since we're pre-processing and this doesn't have to be done in
	- real-time, we go ahead and do the full sliding cubic interpolation. */
	- while (--count)
	- {
	- vptr = src + (ofs >> FRACTION_BITS);
	- /*
	- * Electric Fence to the rescue: Accessing *(vptr - 1) is not a
	- * good thing to do when vptr <= src. (TiMidity++ has a similar
	- * safe-guard here.)
	- */
	- v1 = (vptr == src) ? vptr : (vptr - 1);
	- v2 = *vptr;
	- v3 = *(vptr + 1);
	- v4 = *(vptr + 2);
	- xdiff = FSCALENEG(ofs & FRACTION_MASK, FRACTION_BITS);
	- dest++ = (Sint16)(v2 + (xdiff / 6.0) (-2 * v1 - 3 * v2 + 6 * v3 - v4 +
	- xdiff * (3 * (v1 - 2 * v2 + v3) + xdiff * (-v1 + 3 * (v2 - v3) + v4))));
	- ofs += incr;
	- }
	-
	- if (ofs & FRACTION_MASK)
	- {
	- v1 = src[ofs >> FRACTION_BITS];
	- v2 = src[(ofs >> FRACTION_BITS) + 1];
	- dest++ = v1 + (((v2 - v1) (ofs & FRACTION_MASK)) >> FRACTION_BITS);
	- }
	- else
	- *dest++ = src[ofs >> FRACTION_BITS];
	-
	- sp->data_length = newlen;
	- sp->loop_start = (Sint32)(sp->loop_start / a);
	- sp->loop_end = (Sint32)(sp->loop_end / a);
	- free(sp->data);
	- sp->data = (sample_t *) newdata;
	- sp->sample_rate = 0;
	-}
	diff --git a/util/sdl/sound/decoders/timidity/resample.h b/util/sdl/sound/decoders/timidity/resample.h
	deleted file mode 100644
	index 152cb386..00000000
	--- a/util/sdl/sound/decoders/timidity/resample.h
	+++ /dev/null
	@@ -1,24 +0,0 @@
	-/*
	-
	- TiMidity -- Experimental MIDI to WAVE converter
	- Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi>
	-
	- This program is free software; you can redistribute it and/or modify
	- it under the terms of the GNU General Public License as published by
	- the Free Software Foundation; either version 2 of the License, or
	- (at your option) any later version.
	-
	- This program is distributed in the hope that it will be useful,
	- but WITHOUT ANY WARRANTY; without even the implied warranty of
	- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
	- GNU General Public License for more details.
	-
	- You should have received a copy of the GNU General Public License
	- along with this program; if not, write to the Free Software
	- Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
	-
	- resample.h
	-*/
	-
	-extern sample_t resample_voice(MidiSong song, int v, Sint32 *countptr);
	-extern void pre_resample(MidiSong song, Sample sp);
	diff --git a/util/sdl/sound/decoders/timidity/tables.c b/util/sdl/sound/decoders/timidity/tables.c
	deleted file mode 100644
	index 6c092add..00000000
	--- a/util/sdl/sound/decoders/timidity/tables.c
	+++ /dev/null
	@@ -1,218 +0,0 @@
	-/*
	-
	- TiMidity -- Experimental MIDI to WAVE converter
	- Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi>
	-
	- This program is free software; you can redistribute it and/or modify
	- it under the terms of the GNU General Public License as published by
	- the Free Software Foundation; either version 2 of the License, or
	- (at your option) any later version.
	-
	- This program is distributed in the hope that it will be useful,
	- but WITHOUT ANY WARRANTY; without even the implied warranty of
	- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
	- GNU General Public License for more details.
	-
	- You should have received a copy of the GNU General Public License
	- along with this program; if not, write to the Free Software
	- Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
	-
	-*/
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#include <stdio.h>
	-
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-#include "tables.h"
	-
	-const Sint32 freq_table[128]=
	-{
	- 8176, 8662, 9177, 9723,
	- 10301, 10913, 11562, 12250,
	- 12978, 13750, 14568, 15434,
	-
	- 16352, 17324, 18354, 19445,
	- 20602, 21827, 23125, 24500,
	- 25957, 27500, 29135, 30868,
	-
	- 32703, 34648, 36708, 38891,
	- 41203, 43654, 46249, 48999,
	- 51913, 55000, 58270, 61735,
	-
	- 65406, 69296, 73416, 77782,
	- 82407, 87307, 92499, 97999,
	- 103826, 110000, 116541, 123471,
	-
	- 130813, 138591, 146832, 155563,
	- 164814, 174614, 184997, 195998,
	- 207652, 220000, 233082, 246942,
	-
	- 261626, 277183, 293665, 311127,
	- 329628, 349228, 369994, 391995,
	- 415305, 440000, 466164, 493883,
	-
	- 523251, 554365, 587330, 622254,
	- 659255, 698456, 739989, 783991,
	- 830609, 880000, 932328, 987767,
	-
	- 1046502, 1108731, 1174659, 1244508,
	- 1318510, 1396913, 1479978, 1567982,
	- 1661219, 1760000, 1864655, 1975533,
	-
	- 2093005, 2217461, 2349318, 2489016,
	- 2637020, 2793826, 2959955, 3135963,
	- 3322438, 3520000, 3729310, 3951066,
	-
	- 4186009, 4434922, 4698636, 4978032,
	- 5274041, 5587652, 5919911, 6271927,
	- 6644875, 7040000, 7458620, 7902133,
	-
	- 8372018, 8869844, 9397273, 9956063,
	- 10548082, 11175303, 11839822, 12543854
	-};
	-
	-/* v=2.^((x/127-1) * 6) */
	-const double vol_table[128] =
	-{
	- 0.015625, 0.016145143728351113, 0.016682602624583379, 0.017237953096759438,
	- 0.017811790741104401, 0.01840473098076444, 0.019017409725829021, 0.019650484055324921,
	- 0.020304632921913132, 0.020980557880044631, 0.021678983838355849, 0.02240065983711079,
	- 0.023146359851523596, 0.023916883621822989, 0.024713057510949051, 0.025535735390801884,
	- 0.026385799557992876, 0.027264161680080529, 0.028171763773305786, 0.029109579212875332,
	- 0.030078613776876421, 0.031079906724942836, 0.032114531912828696, 0.033183598944085631,
	- 0.034288254360078256, 0.035429682869614412, 0.036609108619508737, 0.037827796507442342,
	- 0.039087053538526394, 0.040388230227024875, 0.041732722044739302, 0.043121970917609151,
	- 0.044557466772132896, 0.046040749133268132, 0.047573408775524545, 0.049157089429020417,
	- 0.050793489542332405, 0.05248436410402918, 0.054231526524842463, 0.056036850582493913,
	- 0.057902272431264008, 0.059829792678457581, 0.061821478529993396, 0.063879466007418645,
	- 0.066005962238725971, 0.068203247825430205, 0.070473679288442961, 0.072819691595368496,
	- 0.075243800771931268, 0.077748606600335793, 0.080336795407452768, 0.083011142945821612,
	- 0.085774517370559328, 0.088629882315368294, 0.091580300070941839, 0.094628934869176312,
	- 0.097779056276712184, 0.10103404270144323, 0.1043973850157546, 0.1078726903003755,
	- 0.11146368571286204, 0.11517422248485852, 0.11900828005242428, 0.12296997032385605,
	- 0.12706354208958254, 0.13129338557886089, 0.13566403716816194, 0.14018018424629392,
	- 0.14484667024148207, 0.14966849981579558, 0.15465084423249356, 0.15979904690204472,
	- 0.16511862911277009, 0.17061529595225433, 0.17629494242587571, 0.18216365977901747,
	- 0.18822774202974024, 0.19449369271892172, 0.20096823188510385, 0.20765830327152621,
	- 0.21457108177307616, 0.22171398113114205, 0.2290946618846218, 0.23672103958561411,
	- 0.2446012932886038, 0.25274387432224471, 0.26115751535314891, 0.26985123975140174,
	- 0.27883437126784744, 0.28811654403352405, 0.29770771289197112, 0.30761816407549192,
	- 0.31785852623682015, 0.32843978184802081, 0.33937327897885317, 0.3506707434672246,
	- 0.36234429149478936, 0.37440644258117928, 0.38687013301080181, 0.39974872970660535,
	- 0.41305604456569134, 0.42680634927214656, 0.44101439060298442, 0.45569540624360722,
	- 0.47086514112975281, 0.48653986433345225, 0.50273638651110641, 0.51947207793239625,
	- 0.53676488710936021, 0.55463336004561792, 0.57309666012638816, 0.59217458867062556,
	- 0.61188760616732485, 0.63225685421876243, 0.65330417821421161, 0.67505215075844849,
	- 0.69752409588017272, 0.72074411404630734, 0.74473710800900605, 0.76952880951308478,
	- 0.79514580689252357, 0.82161557358563286, 0.84896649759946774, 0.87722791195508854,
	- 0.90643012614631979, 0.93660445864574493, 0.96778327049280244, 1
	-};
	-
	-const double bend_fine[256] = {
	- 1, 1.0002256593050698, 1.0004513695322617, 1.0006771306930664,
	- 1.0009029427989777, 1.0011288058614922, 1.0013547198921082, 1.0015806849023274,
	- 1.0018067009036538, 1.002032767907594, 1.0022588859256572, 1.0024850549693551,
	- 1.0027112750502025, 1.0029375461797159, 1.0031638683694153, 1.0033902416308227,
	- 1.0036166659754628, 1.0038431414148634, 1.0040696679605541, 1.0042962456240678,
	- 1.0045228744169397, 1.0047495543507072, 1.0049762854369111, 1.0052030676870944,
	- 1.0054299011128027, 1.0056567857255843, 1.00588372153699, 1.006110708558573,
	- 1.0063377468018897, 1.0065648362784985, 1.0067919769999607, 1.0070191689778405,
	- 1.0072464122237039, 1.0074737067491204, 1.0077010525656616, 1.0079284496849015,
	- 1.0081558981184175, 1.008383397877789, 1.008610948974598, 1.0088385514204294,
	- 1.0090662052268706, 1.0092939104055114, 1.0095216669679448, 1.0097494749257656,
	- 1.009977334290572, 1.0102052450739643, 1.0104332072875455, 1.0106612209429215,
	- 1.0108892860517005, 1.0111174026254934, 1.0113455706759138, 1.0115737902145781,
	- 1.0118020612531047, 1.0120303838031153, 1.0122587578762337, 1.012487183484087,
	- 1.0127156606383041, 1.0129441893505169, 1.0131727696323602, 1.0134014014954713,
	- 1.0136300849514894, 1.0138588200120575, 1.0140876066888203, 1.0143164449934257,
	- 1.0145453349375237, 1.0147742765327674, 1.0150032697908125, 1.0152323147233171,
	- 1.015461411341942, 1.0156905596583505, 1.0159197596842091, 1.0161490114311862,
	- 1.0163783149109531, 1.0166076701351838, 1.0168370771155553, 1.0170665358637463,
	- 1.0172960463914391, 1.0175256087103179, 1.0177552228320703, 1.0179848887683858,
	- 1.0182146065309567, 1.0184443761314785, 1.0186741975816487, 1.0189040708931674,
	- 1.0191339960777379, 1.0193639731470658, 1.0195940021128593, 1.0198240829868295,
	- 1.0200542157806898, 1.0202844005061564, 1.0205146371749483, 1.0207449257987866,
	- 1.0209752663893958, 1.0212056589585028, 1.0214361035178368, 1.0216666000791297,
	- 1.0218971486541166, 1.0221277492545349, 1.0223584018921241, 1.0225891065786274,
	- 1.0228198633257899, 1.0230506721453596, 1.023281533049087, 1.0235124460487257,
	- 1.0237434111560313, 1.0239744283827625, 1.0242054977406807, 1.0244366192415495,
	- 1.0246677928971357, 1.0248990187192082, 1.025130296719539, 1.0253616269099028,
	- 1.0255930093020766, 1.0258244439078401, 1.0260559307389761, 1.0262874698072693,
	- 1.0265190611245079, 1.0267507047024822, 1.0269824005529853, 1.027214148687813,
	- 1.0274459491187637, 1.0276778018576387, 1.0279097069162415, 1.0281416643063788,
	- 1.0283736740398595, 1.0286057361284953, 1.0288378505841009, 1.0290700174184932,
	- 1.0293022366434921, 1.0295345082709197, 1.0297668323126017, 1.0299992087803651,
	- 1.030231637686041, 1.0304641190414621, 1.0306966528584645, 1.0309292391488862,
	- 1.0311618779245688, 1.0313945691973556, 1.0316273129790936, 1.0318601092816313,
	- 1.0320929581168212, 1.0323258594965172, 1.0325588134325767, 1.0327918199368598,
	- 1.0330248790212284, 1.0332579906975481, 1.0334911549776868, 1.033724371873515,
	- 1.0339576413969056, 1.0341909635597348, 1.0344243383738811, 1.0346577658512259,
	- 1.034891246003653, 1.0351247788430489, 1.0353583643813031, 1.0355920026303078,
	- 1.0358256936019572, 1.0360594373081489, 1.0362932337607829, 1.0365270829717617,
	- 1.0367609849529913, 1.0369949397163791, 1.0372289472738365, 1.0374630076372766,
	- 1.0376971208186156, 1.0379312868297725, 1.0381655056826686, 1.0383997773892284,
	- 1.0386341019613787, 1.0388684794110492, 1.0391029097501721, 1.0393373929906822,
	- 1.0395719291445176, 1.0398065182236185, 1.0400411602399278, 1.0402758552053915,
	- 1.0405106031319582, 1.0407454040315787, 1.0409802579162071, 1.0412151647977996,
	- 1.0414501246883161, 1.0416851375997183, 1.0419202035439705, 1.0421553225330404,
	- 1.042390494578898, 1.042625719693516, 1.0428609978888699, 1.043096329176938,
	- 1.0433317135697009, 1.0435671510791424, 1.0438026417172486, 1.0440381854960086,
	- 1.0442737824274138, 1.044509432523459, 1.044745135796141, 1.0449808922574599,
	- 1.0452167019194181, 1.0454525647940205, 1.0456884808932754, 1.0459244502291931,
	- 1.0461604728137874, 1.0463965486590741, 1.046632677777072, 1.0468688601798024,
	- 1.0471050958792898, 1.047341384887561, 1.0475777272166455, 1.047814122878576,
	- 1.048050571885387, 1.0482870742491166, 1.0485236299818055, 1.0487602390954964,
	- 1.0489969016022356, 1.0492336175140715, 1.0494703868430555, 1.0497072096012419,
	- 1.0499440858006872, 1.0501810154534512, 1.050417998571596, 1.0506550351671864,
	- 1.0508921252522903, 1.0511292688389782, 1.0513664659393229, 1.0516037165654004,
	- 1.0518410207292894, 1.0520783784430709, 1.0523157897188296, 1.0525532545686513,
	- 1.0527907730046264, 1.0530283450388465, 1.0532659706834067, 1.0535036499504049,
	- 1.0537413828519411, 1.0539791694001188, 1.0542170096070436, 1.0544549034848243,
	- 1.0546928510455722, 1.0549308523014012, 1.0551689072644284, 1.0554070159467728,
	- 1.0556451783605572, 1.0558833945179062, 1.0561216644309479, 1.0563599881118126,
	- 1.0565983655726334, 1.0568367968255465, 1.0570752818826903, 1.0573138207562065,
	- 1.057552413458239, 1.0577910600009348, 1.0580297603964437, 1.058268514656918,
	- 1.0585073227945128, 1.0587461848213857, 1.058985100749698, 1.0592240705916123
	-};
	-
	-const double bend_coarse[128] = {
	- 1, 1.0594630943592953, 1.122462048309373, 1.189207115002721,
	- 1.2599210498948732, 1.3348398541700344, 1.4142135623730951, 1.4983070768766815,
	- 1.5874010519681994, 1.681792830507429, 1.7817974362806785, 1.8877486253633868,
	- 2, 2.1189261887185906, 2.244924096618746, 2.3784142300054421,
	- 2.5198420997897464, 2.6696797083400687, 2.8284271247461903, 2.996614153753363,
	- 3.1748021039363992, 3.363585661014858, 3.5635948725613571, 3.7754972507267741,
	- 4, 4.2378523774371812, 4.4898481932374912, 4.7568284600108841,
	- 5.0396841995794928, 5.3393594166801366, 5.6568542494923806, 5.993228307506727,
	- 6.3496042078727974, 6.727171322029716, 7.1271897451227151, 7.5509945014535473,
	- 8, 8.4757047548743625, 8.9796963864749824, 9.5136569200217682,
	- 10.079368399158986, 10.678718833360273, 11.313708498984761, 11.986456615013454,
	- 12.699208415745595, 13.454342644059432, 14.25437949024543, 15.101989002907095,
	- 16, 16.951409509748721, 17.959392772949972, 19.027313840043536,
	- 20.158736798317967, 21.357437666720553, 22.627416997969522, 23.972913230026901,
	- 25.398416831491197, 26.908685288118864, 28.508758980490853, 30.203978005814196,
	- 32, 33.902819019497443, 35.918785545899944, 38.054627680087073,
	- 40.317473596635935, 42.714875333441107, 45.254833995939045, 47.945826460053802,
	- 50.796833662982394, 53.817370576237728, 57.017517960981706, 60.407956011628393,
	- 64, 67.805638038994886, 71.837571091799887, 76.109255360174146,
	- 80.63494719327187, 85.429750666882214, 90.509667991878089, 95.891652920107603,
	- 101.59366732596479, 107.63474115247546, 114.03503592196341, 120.81591202325679,
	- 128, 135.61127607798977, 143.67514218359977, 152.21851072034829,
	- 161.26989438654374, 170.85950133376443, 181.01933598375618, 191.78330584021521,
	- 203.18733465192958, 215.26948230495091, 228.07007184392683, 241.63182404651357,
	- 256, 271.22255215597971, 287.35028436719938, 304.43702144069658,
	- 322.53978877308765, 341.71900266752868, 362.03867196751236, 383.56661168043064,
	- 406.37466930385892, 430.53896460990183, 456.14014368785394, 483.26364809302686,
	- 512, 542.44510431195943, 574.70056873439876, 608.87404288139317,
	- 645.0795775461753, 683.43800533505737, 724.07734393502471, 767.13322336086128,
	- 812.74933860771785, 861.07792921980365, 912.28028737570787, 966.52729618605372,
	- 1024, 1084.8902086239189, 1149.4011374687975, 1217.7480857627863,
	- 1290.1591550923506, 1366.8760106701147, 1448.1546878700494, 1534.2664467217226
	-};
	diff --git a/util/sdl/sound/decoders/timidity/tables.h b/util/sdl/sound/decoders/timidity/tables.h
	deleted file mode 100644
	index 6b84a382..00000000
	--- a/util/sdl/sound/decoders/timidity/tables.h
	+++ /dev/null
	@@ -1,30 +0,0 @@
	-/*
	-
	- TiMidity -- Experimental MIDI to WAVE converter
	- Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi>
	-
	- This program is free software; you can redistribute it and/or modify
	- it under the terms of the GNU General Public License as published by
	- the Free Software Foundation; either version 2 of the License, or
	- (at your option) any later version.
	-
	- This program is distributed in the hope that it will be useful,
	- but WITHOUT ANY WARRANTY; without even the implied warranty of
	- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
	- GNU General Public License for more details.
	-
	- You should have received a copy of the GNU General Public License
	- along with this program; if not, write to the Free Software
	- Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
	-
	- tables.h
	-*/
	-
	-#include <math.h>
	-#define sine(x) (sin((2PI/1024.0) (x)))
	-
	-#define SINE_CYCLE_LENGTH 1024
	-extern const Sint32 freq_table[];
	-extern const double vol_table[];
	-extern const double bend_fine[];
	-extern const double bend_coarse[];
	diff --git a/util/sdl/sound/decoders/timidity/testmidi.c b/util/sdl/sound/decoders/timidity/testmidi.c
	deleted file mode 100644
	index d71fa557..00000000
	--- a/util/sdl/sound/decoders/timidity/testmidi.c
	+++ /dev/null
	@@ -1,105 +0,0 @@
	-/*
	- * SDL_sound -- An abstract sound format decoding API.
	- * Copyright (C) 2001 Ryan C. Gordon.
	- *
	- * This library is free software; you can redistribute it and/or
	- * modify it under the terms of the GNU Lesser General Public
	- * License as published by the Free Software Foundation; either
	- * version 2.1 of the License, or (at your option) any later version.
	- *
	- * This library is distributed in the hope that it will be useful,
	- * but WITHOUT ANY WARRANTY; without even the implied warranty of
	- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	- * Lesser General Public License for more details.
	- *
	- * You should have received a copy of the GNU Lesser General Public
	- * License along with this library; if not, write to the Free Software
	- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
	- */
	-
	-/**
	- * Program to test the TiMidity core, without having to worry about decoder
	- * and/or playsound bugs. It's not meant to be robust or user-friendly.
	- */
	-
	-#include <stdio.h>
	-#include <stdlib.h>
	-#include "SDL.h"
	-#include "timidity.h"
	-
	-int done_flag = 0;
	-MidiSong *song;
	-
	-static void audio_callback(void userdata, Uint8 stream, int len)
	-{
	- if (Timidity_PlaySome(song, stream, len) == 0)
	- done_flag = 1;
	-}
	-
	-int main(int argc, char *argv[])
	-{
	- SDL_AudioSpec audio;
	- SDL_RWops *rw;
	-
	- if (SDL_Init(SDL_INIT_AUDIO) < 0)
	- {
	- fprintf(stderr, "Couldn't initialize SDL: %s\n", SDL_GetError());
	- return 1;
	- }
	-
	- atexit(SDL_Quit);
	-
	- if (argc != 2)
	- {
	- fprintf(stderr, "Usage: %s midifile\n", argv[0]);
	- return 1;
	- }
	-
	- audio.freq = 44100;
	- audio.format = AUDIO_S16SYS;
	- audio.channels = 2;
	- audio.samples = 4096;
	- audio.callback = audio_callback;
	-
	- if (SDL_OpenAudio(&audio, NULL) < 0)
	- {
	- fprintf(stderr, "Couldn't open audio device. %s\n", SDL_GetError());
	- return 1;
	- }
	-
	- if (Timidity_Init() < 0)
	- {
	- fprintf(stderr, "Could not initialise TiMidity.\n");
	- return 1;
	- }
	-
	- rw = SDL_RWFromFile(argv[1], "rb");
	- if (rw == NULL)
	- {
	- fprintf(stderr, "Could not create RWops from MIDI file.\n");
	- return 1;
	- }
	-
	- song = Timidity_LoadSong(rw, &audio);
	- SDL_RWclose(rw);
	-
	- if (song == NULL)
	- {
	- fprintf(stderr, "Could not open MIDI file.\n");
	- return 1;
	- }
	-
	- Timidity_SetVolume(song, 100);
	- Timidity_Start(song);
	-
	- SDL_PauseAudio(0);
	- while (!done_flag)
	- {
	- SDL_Delay(10);
	- }
	- SDL_PauseAudio(1);
	- Timidity_FreeSong(song);
	- Timidity_Exit();
	-
	- return 0;
	-}
	diff --git a/util/sdl/sound/decoders/timidity/timidity.c b/util/sdl/sound/decoders/timidity/timidity.c
	deleted file mode 100644
	index 244d6b1e..00000000
	--- a/util/sdl/sound/decoders/timidity/timidity.c
	+++ /dev/null
	@@ -1,602 +0,0 @@
	-/*
	-
	- TiMidity -- Experimental MIDI to WAVE converter
	- Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi>
	-
	- This program is free software; you can redistribute it and/or modify
	- it under the terms of the GNU General Public License as published by
	- the Free Software Foundation; either version 2 of the License, or
	- (at your option) any later version.
	-
	- This program is distributed in the hope that it will be useful,
	- but WITHOUT ANY WARRANTY; without even the implied warranty of
	- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
	- GNU General Public License for more details.
	-
	- You should have received a copy of the GNU General Public License
	- along with this program; if not, write to the Free Software
	- Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
	-
	-*/
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#include <stdio.h>
	-#include <stdlib.h>
	-#include <string.h>
	-
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-#include "timidity.h"
	-
	-#include "options.h"
	-#include "common.h"
	-#include "instrum.h"
	-#include "playmidi.h"
	-#include "readmidi.h"
	-#include "output.h"
	-
	-#include "tables.h"
	-
	-ToneBank master_tonebank[128], master_drumset[128];
	-
	-static char def_instr_name[256] = "";
	-
	-#define MAXWORDS 10
	-
	-/* Quick-and-dirty fgets() replacement. */
	-
	-static char RWgets(SDL_RWops rw, char *s, int size)
	-{
	- int num_read = 0;
	- int newline = 0;
	-
	- while (num_read < size && !newline)
	- {
	- if (SDL_RWread(rw, &s[num_read], 1, 1) != 1)
	- break;
	-
	- /* Unlike fgets(), don't store newline. Under Windows/DOS we'll
	- * probably get an extra blank line for every line that's being
	- * read, but that should be ok.
	- */
	- if (s[num_read] == '\n' \|\| s[num_read] == '\r')
	- {
	- s[num_read] = '\0';
	- newline = 1;
	- }
	-
	- num_read++;
	- }
	-
	- s[num_read] = '\0';
	-
	- return (num_read != 0) ? s : NULL;
	-}
	-
	-static int read_config_file(char *name)
	-{
	- SDL_RWops *rw;
	- char tmp[1024], w[MAXWORDS], cp;
	- ToneBank *bank=0;
	- int i, j, k, line=0, words;
	- static int rcf_count=0;
	-
	- if (rcf_count>50)
	- {
	- SNDDBG(("Probable source loop in configuration files\n"));
	- return (-1);
	- }
	-
	- if (!(rw=open_file(name)))
	- return -1;
	-
	- while (RWgets(rw, tmp, sizeof(tmp)))
	- {
	- line++;
	- w[words=0]=strtok(tmp, " \t\240");
	- if (!w[0]) continue;
	-
	- /* Originally the TiMidity++ extensions were prefixed like this */
	- if (strcmp(w[0], "#extension") == 0)
	- words = -1;
	- else if (*w[0] == '#')
	- continue;
	-
	- while (w[words] && *w[words] != '#' && (words < MAXWORDS))
	- w[++words]=strtok(0," \t\240");
	-
	- /*
	- * TiMidity++ adds a number of extensions to the config file format.
	- * Many of them are completely irrelevant to SDL_sound, but at least
	- * we shouldn't choke on them.
	- *
	- * Unfortunately the documentation for these extensions is often quite
	- * vague, gramatically strange or completely absent.
	- */
	- if (
	- !strcmp(w[0], "comm") /* "comm" program second */
	- \|\| !strcmp(w[0], "HTTPproxy") /* "HTTPproxy" hostname:port */
	- \|\| !strcmp(w[0], "FTPproxy") /* "FTPproxy" hostname:port */
	- \|\| !strcmp(w[0], "mailaddr") /* "mailaddr" your-mail-address */
	- \|\| !strcmp(w[0], "opt") /* "opt" timidity-options */
	- )
	- {
	- /*
	- * + "comm" sets some kind of comment -- the documentation is too
	- * vague for me to understand at this time.
	- * + "HTTPproxy", "FTPproxy" and "mailaddr" are for reading data
	- * over a network, rather than from the file system.
	- * + "opt" specifies default options for TiMidity++.
	- *
	- * These are all quite useless for our version of TiMidity, so
	- * they can safely remain no-ops.
	- */
	- } else if (!strcmp(w[0], "timeout")) /* "timeout" program second */
	- {
	- /*
	- * Specifies a timeout value of the program. A number of seconds
	- * before TiMidity kills the note. This may be useful to implement
	- * later, but I don't see any urgent need for it.
	- */
	- SNDDBG(("FIXME: Implement \"timeout\" in TiMidity config.\n"));
	- } else if (!strcmp(w[0], "copydrumset") /* "copydrumset" drumset */
	- \|\| !strcmp(w[0], "copybank")) /* "copybank" bank */
	- {
	- /*
	- * Copies all the settings of the specified drumset or bank to
	- * the current drumset or bank. May be useful later, but not a
	- * high priority.
	- */
	- SNDDBG(("FIXME: Implement \"%s\" in TiMidity config.\n", w[0]));
	- } else if (!strcmp(w[0], "undef")) /* "undef" progno */
	- {
	- /*
	- * Undefines the tone "progno" of the current tone bank (or
	- * drum set?). Not a high priority.
	- */
	- SNDDBG(("FIXME: Implement \"undef\" in TiMidity config.\n"));
	- } else if (!strcmp(w[0], "altassign")) /* "altassign" prog1 prog2 ... */
	- {
	- /*
	- * Sets the alternate assign for drum set. Whatever that's
	- * supposed to mean.
	- */
	- SNDDBG(("FIXME: Implement \"altassign\" in TiMidity config.\n"));
	- } else if (!strcmp(w[0], "soundfont")
	- \|\| !strcmp(w[0], "font"))
	- {
	- /*
	- * I can't find any documentation for these, but I guess they're
	- * an alternative way of loading/unloading instruments.
	- *
	- * "soundfont" sf_file "remove"
	- * "soundfont" sf_file ["order=" order] ["cutoff=" cutoff]
	- * ["reso=" reso] ["amp=" amp]
	- * "font" "exclude" bank preset keynote
	- * "font" "order" order bank preset keynote
	- */
	- SNDDBG(("FIXME: Implmement \"%s\" in TiMidity config.\n", w[0]));
	- } else if (!strcmp(w[0], "progbase"))
	- {
	- /*
	- * The documentation for this makes absolutely no sense to me, but
	- * apparently it sets some sort of base offset for tone numbers.
	- * Why anyone would want to do this is beyond me.
	- */
	- SNDDBG(("FIXME: Implement \"progbase\" in TiMidity config.\n"));
	- } else if (!strcmp(w[0], "map")) /* "map" name set1 elem1 set2 elem2 */
	- {
	- /*
	- * This extension is the one we will need to implement, as it is
	- * used by the "eawpats". Unfortunately I cannot find any
	- * documentation whatsoever for it, but it looks like it's used
	- * for remapping one instrument to another somehow.
	- */
	- SNDDBG(("FIXME: Implement \"map\" in TiMidity config.\n"));
	- }
	-
	- /* Standard TiMidity config */
	-
	- else if (!strcmp(w[0], "dir"))
	- {
	- if (words < 2)
	- {
	- SNDDBG(("%s: line %d: No directory given\n", name, line));
	- return -2;
	- }
	- for (i=1; i<words; i++)
	- add_to_pathlist(w[i]);
	- }
	- else if (!strcmp(w[0], "source"))
	- {
	- if (words < 2)
	- {
	- SNDDBG(("%s: line %d: No file name given\n", name, line));
	- return -2;
	- }
	- for (i=1; i<words; i++)
	- {
	- rcf_count++;
	- read_config_file(w[i]);
	- rcf_count--;
	- }
	- }
	- else if (!strcmp(w[0], "default"))
	- {
	- if (words != 2)
	- {
	- SNDDBG(("%s: line %d: Must specify exactly one patch name\n",
	- name, line));
	- return -2;
	- }
	- strncpy(def_instr_name, w[1], 255);
	- def_instr_name[255]='\0';
	- }
	- else if (!strcmp(w[0], "drumset"))
	- {
	- if (words < 2)
	- {
	- SNDDBG(("%s: line %d: No drum set number given\n", name, line));
	- return -2;
	- }
	- i=atoi(w[1]);
	- if (i<0 \|\| i>127)
	- {
	- SNDDBG(("%s: line %d: Drum set must be between 0 and 127\n",
	- name, line));
	- return -2;
	- }
	- if (!master_drumset[i])
	- {
	- master_drumset[i] = safe_malloc(sizeof(ToneBank));
	- memset(master_drumset[i], 0, sizeof(ToneBank));
	- master_drumset[i]->tone = safe_malloc(128 * sizeof(ToneBankElement));
	- memset(master_drumset[i]->tone, 0, 128 * sizeof(ToneBankElement));
	- }
	- bank=master_drumset[i];
	- }
	- else if (!strcmp(w[0], "bank"))
	- {
	- if (words < 2)
	- {
	- SNDDBG(("%s: line %d: No bank number given\n", name, line));
	- return -2;
	- }
	- i=atoi(w[1]);
	- if (i<0 \|\| i>127)
	- {
	- SNDDBG(("%s: line %d: Tone bank must be between 0 and 127\n",
	- name, line));
	- return -2;
	- }
	- if (!master_tonebank[i])
	- {
	- master_tonebank[i] = safe_malloc(sizeof(ToneBank));
	- memset(master_tonebank[i], 0, sizeof(ToneBank));
	- master_tonebank[i]->tone = safe_malloc(128 * sizeof(ToneBankElement));
	- memset(master_tonebank[i]->tone, 0, 128 * sizeof(ToneBankElement));
	- }
	- bank=master_tonebank[i];
	- }
	- else
	- {
	- if ((words < 2) \|\| (w[0] < '0' \|\| w[0] > '9'))
	- {
	- SNDDBG(("%s: line %d: syntax error\n", name, line));
	- return -2;
	- }
	- i=atoi(w[0]);
	- if (i<0 \|\| i>127)
	- {
	- SNDDBG(("%s: line %d: Program must be between 0 and 127\n",
	- name, line));
	- return -2;
	- }
	- if (!bank)
	- {
	- SNDDBG(("%s: line %d: Must specify tone bank or drum set before assignment\n",
	- name, line));
	- return -2;
	- }
	- if (bank->tone[i].name)
	- free(bank->tone[i].name);
	- strcpy((bank->tone[i].name=safe_malloc(strlen(w[1])+1)),w[1]);
	- bank->tone[i].note=bank->tone[i].amp=bank->tone[i].pan=
	- bank->tone[i].strip_loop=bank->tone[i].strip_envelope=
	- bank->tone[i].strip_tail=-1;
	-
	- for (j=2; j<words; j++)
	- {
	- if (!(cp=strchr(w[j], '=')))
	- {
	- SNDDBG(("%s: line %d: bad patch option %s\n", name, line, w[j]));
	- return -2;
	- }
	- *cp++=0;
	- if (!strcmp(w[j], "amp"))
	- {
	- k=atoi(cp);
	- if ((k<0 \|\| k>MAX_AMPLIFICATION) \|\| (cp < '0' \|\| cp > '9'))
	- {
	- SNDDBG(("%s: line %d: amplification must be between 0 and %d\n",
	- name, line, MAX_AMPLIFICATION));
	- return -2;
	- }
	- bank->tone[i].amp=k;
	- }
	- else if (!strcmp(w[j], "note"))
	- {
	- k=atoi(cp);
	- if ((k<0 \|\| k>127) \|\| (cp < '0' \|\| cp > '9'))
	- {
	- SNDDBG(("%s: line %d: note must be between 0 and 127\n",
	- name, line));
	- return -2;
	- }
	- bank->tone[i].note=k;
	- }
	- else if (!strcmp(w[j], "pan"))
	- {
	- if (!strcmp(cp, "center"))
	- k=64;
	- else if (!strcmp(cp, "left"))
	- k=0;
	- else if (!strcmp(cp, "right"))
	- k=127;
	- else
	- k=((atoi(cp)+100) * 100) / 157;
	- if ((k<0 \|\| k>127) \|\| (k==0 && cp!='-' && (cp < '0' \|\| *cp > '9')))
	- {
	- SNDDBG(("%s: line %d: panning must be left, right, center, or between -100 and 100\n",
	- name, line));
	- return -2;
	- }
	- bank->tone[i].pan=k;
	- }
	- else if (!strcmp(w[j], "keep"))
	- {
	- if (!strcmp(cp, "env"))
	- bank->tone[i].strip_envelope=0;
	- else if (!strcmp(cp, "loop"))
	- bank->tone[i].strip_loop=0;
	- else
	- {
	- SNDDBG(("%s: line %d: keep must be env or loop\n", name, line));
	- return -2;
	- }
	- }
	- else if (!strcmp(w[j], "strip"))
	- {
	- if (!strcmp(cp, "env"))
	- bank->tone[i].strip_envelope=1;
	- else if (!strcmp(cp, "loop"))
	- bank->tone[i].strip_loop=1;
	- else if (!strcmp(cp, "tail"))
	- bank->tone[i].strip_tail=1;
	- else
	- {
	- SNDDBG(("%s: line %d: strip must be env, loop, or tail\n",
	- name, line));
	- return -2;
	- }
	- }
	- else
	- {
	- SNDDBG(("%s: line %d: bad patch option %s\n", name, line, w[j]));
	- return -2;
	- }
	- }
	- }
	- }
	- SDL_RWclose(rw);
	- return 0;
	-}
	-
	-int Timidity_Init_NoConfig()
	-{
	- /* Allocate memory for the standard tonebank and drumset */
	- master_tonebank[0] = safe_malloc(sizeof(ToneBank));
	- memset(master_tonebank[0], 0, sizeof(ToneBank));
	- master_tonebank[0]->tone = safe_malloc(128 * sizeof(ToneBankElement));
	- memset(master_tonebank[0]->tone, 0, 128 * sizeof(ToneBankElement));
	-
	- master_drumset[0] = safe_malloc(sizeof(ToneBank));
	- memset(master_drumset[0], 0, sizeof(ToneBank));
	- master_drumset[0]->tone = safe_malloc(128 * sizeof(ToneBankElement));
	- memset(master_drumset[0]->tone, 0, 128 * sizeof(ToneBankElement));
	-
	- return 0;
	-}
	-
	-int Timidity_Init()
	-{
	- /* !!! FIXME: This may be ugly, but slightly less so than requiring the
	- * default search path to have only one element. I think.
	- *
	- * We only need to include the likely locations for the config
	- * file itself since that file should contain any other directory
	- * that needs to be added to the search path.
	- */
	-#ifdef WIN32
	- add_to_pathlist("\\TIMIDITY");
	-#else
	- add_to_pathlist("/usr/local/lib/timidity");
	- add_to_pathlist("/etc");
	-#endif
	-
	- Timidity_Init_NoConfig();
	-
	- return read_config_file(CONFIG_FILE);
	-}
	-
	-MidiSong Timidity_LoadDLSSong(SDL_RWops rw, DLS_Patches patches, SDL_AudioSpec audio)
	-{
	- MidiSong *song;
	- Sint32 events;
	- int i;
	-
	- if (rw == NULL)
	- return NULL;
	-
	- /* Allocate memory for the song */
	- song = (MidiSong )safe_malloc(sizeof(song));
	- memset(song, 0, sizeof(*song));
	- song->patches = patches;
	-
	- for (i = 0; i < 128; i++)
	- {
	- if (master_tonebank[i])
	- {
	- song->tonebank[i] = safe_malloc(sizeof(ToneBank));
	- memset(song->tonebank[i], 0, sizeof(ToneBank));
	- song->tonebank[i]->tone = master_tonebank[i]->tone;
	- }
	- if (master_drumset[i])
	- {
	- song->drumset[i] = safe_malloc(sizeof(ToneBank));
	- memset(song->drumset[i], 0, sizeof(ToneBank));
	- song->drumset[i]->tone = master_drumset[i]->tone;
	- }
	- }
	-
	- song->amplification = DEFAULT_AMPLIFICATION;
	- song->voices = DEFAULT_VOICES;
	- song->drumchannels = DEFAULT_DRUMCHANNELS;
	-
	- song->rw = rw;
	-
	- song->rate = audio->freq;
	- song->encoding = 0;
	- if ((audio->format & 0xFF) == 16)
	- song->encoding \|= PE_16BIT;
	- if (audio->format & 0x8000)
	- song->encoding \|= PE_SIGNED;
	- if (audio->channels == 1)
	- song->encoding \|= PE_MONO;
	- switch (audio->format) {
	- case AUDIO_S8:
	- song->write = s32tos8;
	- break;
	- case AUDIO_U8:
	- song->write = s32tou8;
	- break;
	- case AUDIO_S16LSB:
	- song->write = s32tos16l;
	- break;
	- case AUDIO_S16MSB:
	- song->write = s32tos16b;
	- break;
	- case AUDIO_U16LSB:
	- song->write = s32tou16l;
	- break;
	- default:
	- SNDDBG(("Unsupported audio format"));
	- song->write = s32tou16l;
	- break;
	- }
	-
	- song->buffer_size = audio->samples;
	- song->resample_buffer = safe_malloc(audio->samples * sizeof(sample_t));
	- song->common_buffer = safe_malloc(audio->samples * 2 * sizeof(Sint32));
	-
	- song->control_ratio = audio->freq / CONTROLS_PER_SECOND;
	- if (song->control_ratio < 1)
	- song->control_ratio = 1;
	- else if (song->control_ratio > MAX_CONTROL_RATIO)
	- song->control_ratio = MAX_CONTROL_RATIO;
	-
	- song->lost_notes = 0;
	- song->cut_notes = 0;
	-
	- song->events = read_midi_file(song, &events, &song->samples);
	-
	- /* The RWops can safely be closed at this point, but let's make that the
	- * responsibility of the caller.
	- */
	-
	- /* Make sure everything is okay */
	- if (!song->events) {
	- free(song);
	- return(NULL);
	- }
	-
	- song->default_instrument = 0;
	- song->default_program = DEFAULT_PROGRAM;
	-
	- if (*def_instr_name)
	- set_default_instrument(song, def_instr_name);
	-
	- load_missing_instruments(song);
	-
	- return(song);
	-}
	-
	-MidiSong Timidity_LoadSong(SDL_RWops rw, SDL_AudioSpec *audio)
	-{
	- return Timidity_LoadDLSSong(rw, NULL, audio);
	-}
	-
	-void Timidity_FreeSong(MidiSong *song)
	-{
	- int i;
	-
	- free_instruments(song);
	-
	- for (i = 0; i < 128; i++)
	- {
	- if (song->tonebank[i])
	- free(song->tonebank[i]);
	- if (song->drumset[i])
	- free(song->drumset[i]);
	- }
	-
	- free(song->common_buffer);
	- free(song->resample_buffer);
	- free(song->events);
	- free(song);
	-}
	-
	-void Timidity_Exit(void)
	-{
	- int i, j;
	-
	- for (i = 0; i < 128; i++)
	- {
	- if (master_tonebank[i])
	- {
	- ToneBankElement *e = master_tonebank[i]->tone;
	- if (e != NULL)
	- {
	- for (j = 0; j < 128; j++)
	- {
	- if (e[j].name != NULL)
	- free(e[j].name);
	- }
	- free(e);
	- }
	- free(master_tonebank[i]);
	- }
	- if (master_drumset[i])
	- {
	- ToneBankElement *e = master_drumset[i]->tone;
	- if (e != NULL)
	- {
	- for (j = 0; j < 128; j++)
	- {
	- if (e[j].name != NULL)
	- free(e[j].name);
	- }
	- free(e);
	- }
	- free(master_drumset[i]);
	- }
	- }
	-
	- free_pathlist();
	-}
	diff --git a/util/sdl/sound/decoders/timidity/timidity.h b/util/sdl/sound/decoders/timidity/timidity.h
	deleted file mode 100644
	index 53ca825f..00000000
	--- a/util/sdl/sound/decoders/timidity/timidity.h
	+++ /dev/null
	@@ -1,176 +0,0 @@
	-/*
	-
	- TiMidity -- Experimental MIDI to WAVE converter
	- Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi>
	-
	- This program is free software; you can redistribute it and/or modify
	- it under the terms of the GNU General Public License as published by
	- the Free Software Foundation; either version 2 of the License, or
	- (at your option) any later version.
	-
	- This program is distributed in the hope that it will be useful,
	- but WITHOUT ANY WARRANTY; without even the implied warranty of
	- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
	- GNU General Public License for more details.
	-
	- You should have received a copy of the GNU General Public License
	- along with this program; if not, write to the Free Software
	- Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
	-
	-*/
	-
	-#ifndef TIMIDITY_H
	-#define TIMIDITY_H
	-#ifdef __cplusplus
	-extern "C" {
	-#endif
	-
	-typedef Sint16 sample_t;
	-typedef Sint32 final_volume_t;
	-
	-#define VIBRATO_SAMPLE_INCREMENTS 32
	-
	-/* Maximum polyphony. */
	-#define MAX_VOICES 48
	-
	-typedef struct {
	- Sint32
	- loop_start, loop_end, data_length,
	- sample_rate, low_vel, high_vel, low_freq, high_freq, root_freq;
	- Sint32
	- envelope_rate[6], envelope_offset[6];
	- float
	- volume;
	- sample_t *data;
	- Sint32
	- tremolo_sweep_increment, tremolo_phase_increment,
	- vibrato_sweep_increment, vibrato_control_ratio;
	- Uint8
	- tremolo_depth, vibrato_depth,
	- modes;
	- Sint8
	- panning, note_to_use;
	-} Sample;
	-
	-typedef struct {
	- int
	- bank, program, volume, sustain, panning, pitchbend, expression,
	- mono, /* one note only on this channel -- not implemented yet */
	- pitchsens;
	- /* chorus, reverb... Coming soon to a 300-MHz, eight-way superscalar
	- processor near you */
	- float
	- pitchfactor; /* precomputed pitch bend factor to save some fdiv's */
	-} Channel;
	-
	-typedef struct {
	- Uint8
	- status, channel, note, velocity;
	- Sample *sample;
	- Sint32
	- orig_frequency, frequency,
	- sample_offset, sample_increment,
	- envelope_volume, envelope_target, envelope_increment,
	- tremolo_sweep, tremolo_sweep_position,
	- tremolo_phase, tremolo_phase_increment,
	- vibrato_sweep, vibrato_sweep_position;
	-
	- final_volume_t left_mix, right_mix;
	-
	- float
	- left_amp, right_amp, tremolo_volume;
	- Sint32
	- vibrato_sample_increment[VIBRATO_SAMPLE_INCREMENTS];
	- int
	- vibrato_phase, vibrato_control_ratio, vibrato_control_counter,
	- envelope_stage, control_counter, panning, panned;
	-
	-} Voice;
	-
	-typedef struct {
	- int samples;
	- Sample *sample;
	-} Instrument;
	-
	-/* Shared data */
	-typedef struct {
	- char *name;
	- int note, amp, pan, strip_loop, strip_envelope, strip_tail;
	-} ToneBankElement;
	-
	-typedef struct {
	- ToneBankElement *tone;
	- Instrument *instrument[128];
	-} ToneBank;
	-
	-typedef struct {
	- Sint32 time;
	- Uint8 channel, type, a, b;
	-} MidiEvent;
	-
	-typedef struct {
	- MidiEvent event;
	- void *next;
	-} MidiEventList;
	-
	-struct _DLS_Data;
	-typedef struct _DLS_Data DLS_Patches;
	-
	-typedef struct {
	- int playing;
	- SDL_RWops *rw;
	- Sint32 rate;
	- Sint32 encoding;
	- float master_volume;
	- Sint32 amplification;
	- DLS_Patches *patches;
	- ToneBank *tonebank[128];
	- ToneBank *drumset[128];
	- Instrument *default_instrument;
	- int default_program;
	- void (write)(void dp, Sint32 *lp, Sint32 c);
	- int buffer_size;
	- sample_t *resample_buffer;
	- Sint32 *common_buffer;
	- Sint32 *buffer_pointer;
	- /* These would both fit into 32 bits, but they are often added in
	- large multiples, so it's simpler to have two roomy ints */
	- /* samples per MIDI delta-t */
	- Sint32 sample_increment;
	- Sint32 sample_correction;
	- Channel channel[16];
	- Voice voice[MAX_VOICES];
	- int voices;
	- Sint32 drumchannels;
	- Sint32 buffered_count;
	- Sint32 control_ratio;
	- Sint32 lost_notes;
	- Sint32 cut_notes;
	- Sint32 samples;
	- MidiEvent *events;
	- MidiEvent *current_event;
	- MidiEventList *evlist;
	- Sint32 current_sample;
	- Sint32 event_count;
	- Sint32 at;
	-} MidiSong;
	-
	-/* Some of these are not defined in timidity.c but are here for convenience */
	-
	-extern int Timidity_Init(void);
	-extern int Timidity_Init_NoConfig(void);
	-extern void Timidity_SetVolume(MidiSong *song, int volume);
	-extern int Timidity_PlaySome(MidiSong song, void stream, Sint32 len);
	-extern DLS_Patches Timidity_LoadDLS(SDL_RWops rw);
	-extern void Timidity_FreeDLS(DLS_Patches *patches);
	-extern MidiSong Timidity_LoadDLSSong(SDL_RWops rw, DLS_Patches patches, SDL_AudioSpec audio);
	-extern MidiSong Timidity_LoadSong(SDL_RWops rw, SDL_AudioSpec *audio);
	-extern void Timidity_Start(MidiSong *song);
	-extern void Timidity_Seek(MidiSong *song, Uint32 ms);
	-extern void Timidity_FreeSong(MidiSong *song);
	-extern void Timidity_Exit(void);
	-
	-#ifdef __cplusplus
	-}
	-#endif
	-#endif /* TIMIDITY_H */
	diff --git a/util/sdl/sound/decoders/voc.c b/util/sdl/sound/decoders/voc.c
	deleted file mode 100644
	index d7c2795c..00000000
	--- a/util/sdl/sound/decoders/voc.c
	+++ /dev/null
	@@ -1,569 +0,0 @@
	-/*
	- * SDL_sound -- An abstract sound format decoding API.
	- * Copyright (C) 2001 Ryan C. Gordon.
	- *
	- * This library is free software; you can redistribute it and/or
	- * modify it under the terms of the GNU Lesser General Public
	- * License as published by the Free Software Foundation; either
	- * version 2.1 of the License, or (at your option) any later version.
	- *
	- * This library is distributed in the hope that it will be useful,
	- * but WITHOUT ANY WARRANTY; without even the implied warranty of
	- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	- * Lesser General Public License for more details.
	- *
	- * You should have received a copy of the GNU Lesser General Public
	- * License along with this library; if not, write to the Free Software
	- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
	- */
	-
	-/*
	- * VOC decoder for SDL_sound.
	- *
	- * This driver handles Creative Labs VOC audio data...this is a legacy format,
	- * but there's some game ports that could make use of such a decoder. Plus,
	- * VOC is fairly straightforward to decode, so this is a more complex, but
	- * still palatable example of an SDL_sound decoder. Y'know, in case the
	- * RAW decoder didn't do it for you. :)
	- *
	- * This code was ripped from a decoder I had written for SDL_mixer, which was
	- * largely ripped from sox v12.17.1's voc.c.
	- *
	- * SDL_mixer: http://www.libsdl.org/projects/SDL_mixer/
	- * sox: http://www.freshmeat.net/projects/sox/
	- *
	- * Please see the file COPYING in the source's root directory.
	- *
	- * This file written by Ryan C. Gordon. (icculus@icculus.org)
	- */
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#ifdef SOUND_SUPPORTS_VOC
	-
	-#include <stdio.h>
	-#include <stdlib.h>
	-#include <string.h>
	-
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-static int VOC_init(void);
	-static void VOC_quit(void);
	-static int VOC_open(Sound_Sample sample, const char ext);
	-static void VOC_close(Sound_Sample *sample);
	-static Uint32 VOC_read(Sound_Sample *sample);
	-static int VOC_rewind(Sound_Sample *sample);
	-static int VOC_seek(Sound_Sample *sample, Uint32 ms);
	-
	-static const char *extensions_voc[] = { "VOC", NULL };
	-const Sound_DecoderFunctions __Sound_DecoderFunctions_VOC =
	-{
	- {
	- extensions_voc,
	- "Creative Labs Voice format",
	- "Ryan C. Gordon <icculus@icculus.org>",
	- "http://www.icculus.org/SDL_sound/"
	- },
	-
	- VOC_init, /* init() method */
	- VOC_quit, /* quit() method */
	- VOC_open, /* open() method */
	- VOC_close, /* close() method */
	- VOC_read, /* read() method */
	- VOC_rewind, /* rewind() method */
	- VOC_seek /* seek() method */
	-};
	-
	-
	-/* Private data for VOC file */
	-typedef struct vocstuff {
	- Uint32 rest; /* bytes remaining in current block */
	- Uint32 rate; /* rate code (byte) of this chunk */
	- int silent; /* sound or silence? */
	- Uint32 srate; /* rate code (byte) of silence */
	- Uint32 blockseek; /* start of current output block */
	- Uint32 samples; /* number of samples output */
	- Uint32 size; /* word length of data */
	- Uint8 channels; /* number of sound channels */
	- int extended; /* Has an extended block been read? */
	- Uint32 bufpos; /* byte position in internal->buffer. */
	- Uint32 start_pos; /* offset to seek to in stream when rewinding. */
	- int error; /* error condition (as opposed to EOF). */
	-} vs_t;
	-
	-
	-/* Size field */
	-/* SJB: note that the 1st 3 are sometimes used as sizeof(type) */
	-#define ST_SIZE_BYTE 1
	-#define ST_SIZE_8BIT 1
	-#define ST_SIZE_WORD 2
	-#define ST_SIZE_16BIT 2
	-#define ST_SIZE_DWORD 4
	-#define ST_SIZE_32BIT 4
	-#define ST_SIZE_FLOAT 5
	-#define ST_SIZE_DOUBLE 6
	-#define ST_SIZE_IEEE 7 /* IEEE 80-bit floats. */
	-
	-/* Style field */
	-#define ST_ENCODING_UNSIGNED 1 /* unsigned linear: Sound Blaster */
	-#define ST_ENCODING_SIGN2 2 /* signed linear 2's comp: Mac */
	-#define ST_ENCODING_ULAW 3 /* U-law signed logs: US telephony, SPARC */
	-#define ST_ENCODING_ALAW 4 /* A-law signed logs: non-US telephony */
	-#define ST_ENCODING_ADPCM 5 /* Compressed PCM */
	-#define ST_ENCODING_IMA_ADPCM 6 /* Compressed PCM */
	-#define ST_ENCODING_GSM 7 /* GSM 6.10 33-byte frame lossy compression */
	-
	-#define VOC_TERM 0
	-#define VOC_DATA 1
	-#define VOC_CONT 2
	-#define VOC_SILENCE 3
	-#define VOC_MARKER 4
	-#define VOC_TEXT 5
	-#define VOC_LOOP 6
	-#define VOC_LOOPEND 7
	-#define VOC_EXTENDED 8
	-#define VOC_DATA_16 9
	-
	-
	-static int VOC_init(void)
	-{
	- return(1); /* always succeeds. */
	-} /* VOC_init */
	-
	-
	-static void VOC_quit(void)
	-{
	- /* it's a no-op. */
	-} /* VOC_quit */
	-
	-
	-static __inline__ int voc_readbytes(SDL_RWops src, vs_t v, void *p, int size)
	-{
	- if (SDL_RWread(src, p, size, 1) != 1)
	- {
	- v->error = 1;
	- BAIL_MACRO("VOC: i/o error", 0);
	- } /* if */
	-
	- return(1);
	-} /* voc_readbytes */
	-
	-
	-static __inline__ int voc_check_header(SDL_RWops *src)
	-{
	- /* VOC magic header */
	- Uint8 signature[20]; /* "Creative Voice File\032" */
	- Uint16 datablockofs;
	- vs_t v; /* dummy struct for voc_readbytes */
	-
	- if (!voc_readbytes(src, &v, signature, sizeof (signature)))
	- return(0);
	-
	- if (memcmp(signature, "Creative Voice File\032", sizeof (signature)) != 0)
	- {
	- BAIL_MACRO("VOC: Wrong signature; not a VOC file.", 0);
	- } /* if */
	-
	- /* get the offset where the first datablock is located */
	- if (!voc_readbytes(src, &v, &datablockofs, sizeof (Uint16)))
	- return(0);
	-
	- datablockofs = SDL_SwapLE16(datablockofs);
	-
	- if (SDL_RWseek(src, datablockofs, SEEK_SET) != datablockofs)
	- {
	- BAIL_MACRO("VOC: Failed to seek to data block.", 0);
	- } /* if */
	-
	- return(1); /* success! */
	-} /* voc_check_header */
	-
	-
	-/* Read next block header, save info, leave position at start of data */
	-static int voc_get_block(Sound_Sample sample, vs_t v)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- SDL_RWops *src = internal->rw;
	- Uint8 bits24[3];
	- Uint8 uc, block;
	- Uint32 sblen;
	- Uint16 new_rate_short;
	- Uint32 new_rate_long;
	- Uint8 trash[6];
	- Uint16 period;
	- int i;
	-
	- v->silent = 0;
	- while (v->rest == 0)
	- {
	- if (SDL_RWread(src, &block, sizeof (block), 1) != 1)
	- return 1; /* assume that's the end of the file. */
	-
	- if (block == VOC_TERM)
	- return 1;
	-
	- if (SDL_RWread(src, bits24, sizeof (bits24), 1) != 1)
	- return 1; /* assume that's the end of the file. */
	-
	- /* Size is an 24-bit value. Ugh. */
	- sblen = ( (bits24[0]) \| (bits24[1] << 8) \| (bits24[2] << 16) );
	-
	- switch(block)
	- {
	- case VOC_DATA:
	- if (!voc_readbytes(src, v, &uc, sizeof (uc)))
	- return 0;
	-
	- /* When DATA block preceeded by an EXTENDED */
	- /* block, the DATA blocks rate value is invalid */
	- if (!v->extended)
	- {
	- BAIL_IF_MACRO(uc == 0, "VOC: Sample rate is zero?", 0);
	-
	- if ((v->rate != -1) && (uc != v->rate))
	- BAIL_MACRO("VOC sample rate codes differ", 0);
	-
	- v->rate = uc;
	- v->channels = 1;
	- sample->actual.rate = 1000000.0/(256 - v->rate);
	- sample->actual.channels = 1;
	- } /* if */
	-
	- if (!voc_readbytes(src, v, &uc, sizeof (uc)))
	- return(0);
	-
	- BAIL_IF_MACRO(uc != 0, "VOC: only supports 8-bit data", 0);
	-
	- v->extended = 0;
	- v->rest = sblen - 2;
	- v->size = ST_SIZE_BYTE;
	- return 1;
	-
	- case VOC_DATA_16:
	- if (!voc_readbytes(src, v, &new_rate_long, sizeof (Uint32)))
	- return 0;
	-
	- new_rate_long = SDL_SwapLE32(new_rate_long);
	- BAIL_IF_MACRO(!new_rate_long, "VOC: Sample rate is zero?", 0);
	-
	- if ((v->rate != -1) && (new_rate_long != v->rate))
	- BAIL_MACRO("VOC: sample rate codes differ", 0);
	-
	- v->rate = new_rate_long;
	- sample->actual.rate = new_rate_long;
	-
	- if (!voc_readbytes(src, v, &uc, sizeof (uc)))
	- return 0;
	-
	- switch (uc)
	- {
	- case 8: v->size = ST_SIZE_BYTE; break;
	- case 16: v->size = ST_SIZE_WORD; break;
	- default:
	- BAIL_MACRO("VOC: unknown data size", 0);
	- } /* switch */
	-
	- if (!voc_readbytes(src, v, &v->channels, sizeof (Uint8)))
	- return 0;
	-
	- if (!voc_readbytes(src, v, trash, sizeof (Uint8) * 6))
	- return 0;
	-
	- v->rest = sblen - 12;
	- return 1;
	-
	- case VOC_CONT:
	- v->rest = sblen;
	- return 1;
	-
	- case VOC_SILENCE:
	- if (!voc_readbytes(src, v, &period, sizeof (period)))
	- return 0;
	-
	- period = SDL_SwapLE16(period);
	-
	- if (!voc_readbytes(src, v, &uc, sizeof (uc)))
	- return 0;
	-
	- BAIL_IF_MACRO(uc == 0, "VOC: silence sample rate is zero", 0);
	-
	- /*
	- * Some silence-packed files have gratuitously
	- * different sample rate codes in silence.
	- * Adjust period.
	- */
	- if ((v->rate != -1) && (uc != v->rate))
	- period = (period * (256 - uc))/(256 - v->rate);
	- else
	- v->rate = uc;
	- v->rest = period;
	- v->silent = 1;
	- return 1;
	-
	- case VOC_LOOP:
	- case VOC_LOOPEND:
	- for(i = 0; i < sblen; i++) /* skip repeat loops. */
	- {
	- if (!voc_readbytes(src, v, trash, sizeof (Uint8)))
	- return 0;
	- } /* for */
	- break;
	-
	- case VOC_EXTENDED:
	- /* An Extended block is followed by a data block */
	- /* Set this byte so we know to use the rate */
	- /* value from the extended block and not the */
	- /* data block. */
	- v->extended = 1;
	- if (!voc_readbytes(src, v, &new_rate_short, sizeof (Uint16)))
	- return 0;
	-
	- new_rate_short = SDL_SwapLE16(new_rate_short);
	- BAIL_IF_MACRO(!new_rate_short, "VOC: sample rate is zero", 0);
	-
	- if ((v->rate != -1) && (new_rate_short != v->rate))
	- BAIL_MACRO("VOC: sample rate codes differ", 0);
	-
	- v->rate = new_rate_short;
	-
	- if (!voc_readbytes(src, v, &uc, sizeof (uc)))
	- return 0;
	-
	- BAIL_IF_MACRO(uc != 0, "VOC: only supports 8-bit data", 0);
	-
	- if (!voc_readbytes(src, v, &uc, sizeof (uc)))
	- return 0;
	-
	- if (uc)
	- sample->actual.channels = 2; /* Stereo */
	-
	- /* Needed number of channels before finishing
	- compute for rate */
	- sample->actual.rate =
	- (256000000L/(65536L - v->rate)) / sample->actual.channels;
	- /* An extended block must be followed by a data */
	- /* block to be valid so loop back to top so it */
	- /* can be grabed. */
	- continue;
	-
	- case VOC_MARKER:
	- if (!voc_readbytes(src, v, trash, sizeof (Uint8) * 2))
	- return 0;
	-
	- /* Falling! Falling! */
	-
	- default: /* text block or other krapola. */
	- for(i = 0; i < sblen; i++) /* skip repeat loops. */
	- {
	- if (!voc_readbytes(src, v, trash, sizeof (Uint8)))
	- return 0;
	- } /* for */
	-
	- if (block == VOC_TEXT)
	- continue; /* get next block */
	- } /* switch */
	- } /* while */
	-
	- return 1;
	-} /* voc_get_block */
	-
	-
	-static int voc_read_waveform(Sound_Sample *sample, int fill_buf, Uint32 max)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- SDL_RWops *src = internal->rw;
	- vs_t v = (vs_t ) internal->decoder_private;
	- int done = 0;
	- Uint8 silence = 0x80;
	- Uint8 *buf = internal->buffer;
	-
	- if (v->rest == 0)
	- {
	- if (!voc_get_block(sample, v))
	- return 0;
	- } /* if */
	-
	- if (v->rest == 0)
	- return 0;
	-
	- max = (v->rest < max) ? v->rest : max;
	-
	- if (v->silent)
	- {
	- if (v->size == ST_SIZE_WORD)
	- silence = 0x00;
	-
	- /* Fill in silence */
	- if (fill_buf)
	- memset(buf + v->bufpos, silence, max);
	-
	- done = max;
	- v->rest -= done;
	- } /* if */
	-
	- else
	- {
	- if (fill_buf)
	- {
	- done = SDL_RWread(src, buf + v->bufpos, 1, max);
	- if (done < max)
	- {
	- __Sound_SetError("VOC: i/o error");
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- } /* if */
	- } /* if */
	-
	- else
	- {
	- int cur, rc;
	- cur = SDL_RWtell(src);
	- if (cur >= 0)
	- {
	- rc = SDL_RWseek(src, max, SEEK_CUR);
	- if (rc >= 0)
	- done = rc - cur;
	- else
	- {
	- __Sound_SetError("VOC: seek error");
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- } /* else */
	- } /* if */
	- } /* else */
	-
	- v->rest -= done;
	- v->bufpos += done;
	- } /* else */
	-
	- return(done);
	-} /* voc_read_waveform */
	-
	-
	-static int VOC_open(Sound_Sample sample, const char ext)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- vs_t *v = NULL;
	-
	- if (!voc_check_header(internal->rw))
	- return(0);
	-
	- v = (vs_t *) malloc(sizeof (vs_t));
	- BAIL_IF_MACRO(v == NULL, ERR_OUT_OF_MEMORY, 0);
	- memset(v, '\0', sizeof (vs_t));
	-
	- v->start_pos = SDL_RWtell(internal->rw);
	- v->rate = -1;
	- if (!voc_get_block(sample, v))
	- {
	- free(v);
	- return(0);
	- } /* if */
	-
	- if (v->rate == -1)
	- {
	- free(v);
	- BAIL_MACRO("VOC: data had no sound!", 0);
	- } /* if */
	-
	- SNDDBG(("VOC: Accepting data stream.\n"));
	- sample->actual.format = (v->size == ST_SIZE_WORD) ? AUDIO_S16LSB:AUDIO_U8;
	- sample->actual.channels = v->channels;
	- sample->flags = SOUND_SAMPLEFLAG_CANSEEK;
	- internal->decoder_private = v;
	- return(1);
	-} /* VOC_open */
	-
	-
	-static void VOC_close(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- free(internal->decoder_private);
	-} /* VOC_close */
	-
	-
	-static Uint32 VOC_read(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- vs_t v = (vs_t ) internal->decoder_private;
	-
	- v->bufpos = 0;
	- while (v->bufpos < internal->buffer_size)
	- {
	- Uint32 rc = voc_read_waveform(sample, 1, internal->buffer_size);
	- if (rc == 0)
	- {
	- sample->flags \|= (v->error) ?
	- SOUND_SAMPLEFLAG_ERROR :
	- SOUND_SAMPLEFLAG_EOF;
	- break;
	- } /* if */
	-
	- if (!voc_get_block(sample, v))
	- {
	- sample->flags \|= (v->error) ?
	- SOUND_SAMPLEFLAG_ERROR :
	- SOUND_SAMPLEFLAG_EOF;
	- break;
	- } /* if */
	- } /* while */
	-
	- return(v->bufpos);
	-} /* VOC_read */
	-
	-
	-static int VOC_rewind(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- vs_t v = (vs_t ) internal->decoder_private;
	- int rc = SDL_RWseek(internal->rw, v->start_pos, SEEK_SET);
	- BAIL_IF_MACRO(rc != v->start_pos, ERR_IO_ERROR, 0);
	- v->rest = 0;
	- return(1);
	-} /* VOC_rewind */
	-
	-
	-static int VOC_seek(Sound_Sample *sample, Uint32 ms)
	-{
	- /*
	- * VOCs don't lend themselves well to seeking, since you have to
	- * parse each section, which is an arbitrary size. The best we can do
	- * is rewind, set a flag saying not to write the waveforms to a buffer,
	- * and decode to the point that we want. Ugh. Fortunately, there's
	- * really no such thing as a large VOC, due to the era and hardware that
	- * spawned them, so even though this is inefficient, this is still a
	- * relatively fast operation in most cases.
	- */
	-
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- vs_t v = (vs_t ) internal->decoder_private;
	- int offset = __Sound_convertMsToBytePos(&sample->actual, ms);
	- int origpos = SDL_RWtell(internal->rw);
	- int origrest = v->rest;
	-
	- BAIL_IF_MACRO(!VOC_rewind(sample), NULL, 0);
	-
	- v->bufpos = 0;
	-
	- while (offset > 0)
	- {
	- Uint32 rc = voc_read_waveform(sample, 0, offset);
	- if ( (rc == 0) \|\| (!voc_get_block(sample, v)) )
	- {
	- SDL_RWseek(internal->rw, origpos, SEEK_SET);
	- v->rest = origrest;
	- return(0);
	- } /* if */
	-
	- offset -= rc;
	- } /* while */
	-
	- return(1);
	-} /* VOC_seek */
	-
	-#endif /* SOUND_SUPPORTS_VOC */
	-
	-/* end of voc.c ... */
	diff --git a/util/sdl/sound/decoders/wav.c b/util/sdl/sound/decoders/wav.c
	deleted file mode 100644
	index cf652f72..00000000
	--- a/util/sdl/sound/decoders/wav.c
	+++ /dev/null
	@@ -1,800 +0,0 @@
	-/*
	- * SDL_sound -- An abstract sound format decoding API.
	- * Copyright (C) 2001 Ryan C. Gordon.
	- *
	- * This library is free software; you can redistribute it and/or
	- * modify it under the terms of the GNU Lesser General Public
	- * License as published by the Free Software Foundation; either
	- * version 2.1 of the License, or (at your option) any later version.
	- *
	- * This library is distributed in the hope that it will be useful,
	- * but WITHOUT ANY WARRANTY; without even the implied warranty of
	- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	- * Lesser General Public License for more details.
	- *
	- * You should have received a copy of the GNU Lesser General Public
	- * License along with this library; if not, write to the Free Software
	- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
	- */
	-
	-/*
	- * WAV decoder for SDL_sound.
	- *
	- * This driver handles Microsoft .WAVs, in as many of the thousands of
	- * variations as we can.
	- *
	- * Please see the file COPYING in the source's root directory.
	- *
	- * This file written by Ryan C. Gordon. (icculus@icculus.org)
	- */
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#ifdef SOUND_SUPPORTS_WAV
	-
	-#include <stdio.h>
	-#include <stdlib.h>
	-#include <string.h>
	-
	-#include "SDL_sound.h"
	-
	-#define __SDL_SOUND_INTERNAL__
	-#include "SDL_sound_internal.h"
	-
	-static int WAV_init(void);
	-static void WAV_quit(void);
	-static int WAV_open(Sound_Sample sample, const char ext);
	-static void WAV_close(Sound_Sample *sample);
	-static Uint32 WAV_read(Sound_Sample *sample);
	-static int WAV_rewind(Sound_Sample *sample);
	-static int WAV_seek(Sound_Sample *sample, Uint32 ms);
	-
	-static const char *extensions_wav[] = { "WAV", NULL };
	-const Sound_DecoderFunctions __Sound_DecoderFunctions_WAV =
	-{
	- {
	- extensions_wav,
	- "Microsoft WAVE audio format",
	- "Ryan C. Gordon <icculus@icculus.org>",
	- "http://www.icculus.org/SDL_sound/"
	- },
	-
	- WAV_init, /* init() method */
	- WAV_quit, /* quit() method */
	- WAV_open, /* open() method */
	- WAV_close, /* close() method */
	- WAV_read, /* read() method */
	- WAV_rewind, /* rewind() method */
	- WAV_seek /* seek() method */
	-};
	-
	-
	-/* Better than SDL_ReadLE16, since you can detect i/o errors... */
	-static __inline__ int read_le16(SDL_RWops rw, Uint16 ui16)
	-{
	- int rc = SDL_RWread(rw, ui16, sizeof (Uint16), 1);
	- BAIL_IF_MACRO(rc != 1, ERR_IO_ERROR, 0);
	- ui16 = SDL_SwapLE16(ui16);
	- return(1);
	-} /* read_le16 */
	-
	-
	-/* Better than SDL_ReadLE32, since you can detect i/o errors... */
	-static __inline__ int read_le32(SDL_RWops rw, Uint32 ui32)
	-{
	- int rc = SDL_RWread(rw, ui32, sizeof (Uint32), 1);
	- BAIL_IF_MACRO(rc != 1, ERR_IO_ERROR, 0);
	- ui32 = SDL_SwapLE32(ui32);
	- return(1);
	-} /* read_le32 */
	-
	-
	-/* This is just cleaner on the caller's end... */
	-static __inline__ int read_uint8(SDL_RWops rw, Uint8 ui8)
	-{
	- int rc = SDL_RWread(rw, ui8, sizeof (Uint8), 1);
	- BAIL_IF_MACRO(rc != 1, ERR_IO_ERROR, 0);
	- return(1);
	-} /* read_uint8 */
	-
	-
	- /* Chunk management code... */
	-
	-#define riffID 0x46464952 /* "RIFF", in ascii. */
	-#define waveID 0x45564157 /* "WAVE", in ascii. */
	-#define factID 0x74636166 /* "fact", in ascii. */
	-
	-
	-/*****************************************************************************
	- * The FORMAT chunk... *
	- *****************************************************************************/
	-
	-#define fmtID 0x20746D66 /* "fmt ", in ascii. */
	-
	-#define FMT_NORMAL 0x0001 /* Uncompressed waveform data. */
	-#define FMT_ADPCM 0x0002 /* ADPCM compressed waveform data. */
	-
	-typedef struct
	-{
	- Sint16 iCoef1;
	- Sint16 iCoef2;
	-} ADPCMCOEFSET;
	-
	-typedef struct
	-{
	- Uint8 bPredictor;
	- Uint16 iDelta;
	- Sint16 iSamp1;
	- Sint16 iSamp2;
	-} ADPCMBLOCKHEADER;
	-
	-typedef struct S_WAV_FMT_T
	-{
	- Uint32 chunkID;
	- Sint32 chunkSize;
	- Sint16 wFormatTag;
	- Uint16 wChannels;
	- Uint32 dwSamplesPerSec;
	- Uint32 dwAvgBytesPerSec;
	- Uint16 wBlockAlign;
	- Uint16 wBitsPerSample;
	-
	- Uint32 next_chunk_offset;
	-
	- Uint32 sample_frame_size;
	- Uint32 data_starting_offset;
	- Uint32 total_bytes;
	-
	- void (free)(struct S_WAV_FMT_T fmt);
	- Uint32 (read_sample)(Sound_Sample sample);
	- int (rewind_sample)(Sound_Sample sample);
	- int (seek_sample)(Sound_Sample sample, Uint32 ms);
	-
	- union
	- {
	- struct
	- {
	- Uint16 cbSize;
	- Uint16 wSamplesPerBlock;
	- Uint16 wNumCoef;
	- ADPCMCOEFSET *aCoef;
	- ADPCMBLOCKHEADER *blockheaders;
	- Uint32 samples_left_in_block;
	- int nibble_state;
	- Sint8 nibble;
	- } adpcm;
	-
	- /* put other format-specific data here... */
	- } fmt;
	-} fmt_t;
	-
	-
	-/*
	- * Read in a fmt_t from disk. This makes this process safe regardless of
	- * the processor's byte order or how the fmt_t structure is packed.
	- * Note that the union "fmt" is not read in here; that is handled as
	- * needed in the read_fmt_* functions.
	- */
	-static int read_fmt_chunk(SDL_RWops rw, fmt_t fmt)
	-{
	- /* skip reading the chunk ID, since it was already read at this point... */
	- fmt->chunkID = fmtID;
	-
	- BAIL_IF_MACRO(!read_le32(rw, &fmt->chunkSize), NULL, 0);
	- BAIL_IF_MACRO(fmt->chunkSize < 16, "WAV: Invalid chunk size", 0);
	- fmt->next_chunk_offset = SDL_RWtell(rw) + fmt->chunkSize;
	-
	- BAIL_IF_MACRO(!read_le16(rw, &fmt->wFormatTag), NULL, 0);
	- BAIL_IF_MACRO(!read_le16(rw, &fmt->wChannels), NULL, 0);
	- BAIL_IF_MACRO(!read_le32(rw, &fmt->dwSamplesPerSec), NULL, 0);
	- BAIL_IF_MACRO(!read_le32(rw, &fmt->dwAvgBytesPerSec), NULL, 0);
	- BAIL_IF_MACRO(!read_le16(rw, &fmt->wBlockAlign), NULL, 0);
	- BAIL_IF_MACRO(!read_le16(rw, &fmt->wBitsPerSample), NULL, 0);
	-
	- return(1);
	-} /* read_fmt_chunk */
	-
	-
	-
	-/*****************************************************************************
	- * The DATA chunk... *
	- *****************************************************************************/
	-
	-#define dataID 0x61746164 /* "data", in ascii. */
	-
	-typedef struct
	-{
	- Uint32 chunkID;
	- Sint32 chunkSize;
	- /* Then, (chunkSize) bytes of waveform data... */
	-} data_t;
	-
	-
	-/*
	- * Read in a data_t from disk. This makes this process safe regardless of
	- * the processor's byte order or how the fmt_t structure is packed.
	- */
	-static int read_data_chunk(SDL_RWops rw, data_t data)
	-{
	- /* skip reading the chunk ID, since it was already read at this point... */
	- data->chunkID = dataID;
	- BAIL_IF_MACRO(!read_le32(rw, &data->chunkSize), NULL, 0);
	- return(1);
	-} /* read_data_chunk */
	-
	-
	-
	-
	-/*****************************************************************************
	- * this is what we store in our internal->decoder_private field... *
	- *****************************************************************************/
	-
	-typedef struct
	-{
	- fmt_t *fmt;
	- Sint32 bytesLeft;
	-} wav_t;
	-
	-
	-
	-
	-/*****************************************************************************
	- * Normal, uncompressed waveform handler... *
	- *****************************************************************************/
	-
	-/*
	- * Sound_Decode() lands here for uncompressed WAVs...
	- */
	-static Uint32 read_sample_fmt_normal(Sound_Sample *sample)
	-{
	- Uint32 retval;
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- wav_t w = (wav_t ) internal->decoder_private;
	- Uint32 max = (internal->buffer_size < (Uint32) w->bytesLeft) ?
	- internal->buffer_size : (Uint32) w->bytesLeft;
	-
	- assert(max > 0);
	-
	- /*
	- * We don't actually do any decoding, so we read the wav data
	- * directly into the internal buffer...
	- */
	- retval = SDL_RWread(internal->rw, internal->buffer, 1, max);
	-
	- w->bytesLeft -= retval;
	-
	- /* Make sure the read went smoothly... */
	- if ((retval == 0) \|\| (w->bytesLeft == 0))
	- sample->flags \|= SOUND_SAMPLEFLAG_EOF;
	-
	- else if (retval == -1)
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	-
	- /* (next call this EAGAIN may turn into an EOF or error.) */
	- else if (retval < internal->buffer_size)
	- sample->flags \|= SOUND_SAMPLEFLAG_EAGAIN;
	-
	- return(retval);
	-} /* read_sample_fmt_normal */
	-
	-
	-static int seek_sample_fmt_normal(Sound_Sample *sample, Uint32 ms)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- wav_t w = (wav_t ) internal->decoder_private;
	- fmt_t *fmt = w->fmt;
	- int offset = __Sound_convertMsToBytePos(&sample->actual, ms);
	- int pos = (int) (fmt->data_starting_offset + offset);
	- int rc = SDL_RWseek(internal->rw, pos, SEEK_SET);
	- BAIL_IF_MACRO(rc != pos, ERR_IO_ERROR, 0);
	- w->bytesLeft = fmt->total_bytes - offset;
	- return(1); /* success. */
	-} /* seek_sample_fmt_normal */
	-
	-
	-static int rewind_sample_fmt_normal(Sound_Sample *sample)
	-{
	- /* no-op. */
	- return(1);
	-} /* rewind_sample_fmt_normal */
	-
	-
	-static int read_fmt_normal(SDL_RWops rw, fmt_t fmt)
	-{
	- /* (don't need to read more from the RWops...) */
	- fmt->free = NULL;
	- fmt->read_sample = read_sample_fmt_normal;
	- fmt->rewind_sample = rewind_sample_fmt_normal;
	- fmt->seek_sample = seek_sample_fmt_normal;
	- return(1);
	-} /* read_fmt_normal */
	-
	-
	-
	-/*****************************************************************************
	- * ADPCM compression handler... *
	- *****************************************************************************/
	-
	-#define FIXED_POINT_COEF_BASE 256
	-#define FIXED_POINT_ADAPTION_BASE 256
	-#define SMALLEST_ADPCM_DELTA 16
	-
	-
	-static __inline__ int read_adpcm_block_headers(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- SDL_RWops *rw = internal->rw;
	- wav_t w = (wav_t ) internal->decoder_private;
	- fmt_t *fmt = w->fmt;
	- ADPCMBLOCKHEADER *headers = fmt->fmt.adpcm.blockheaders;
	- int i;
	- int max = fmt->wChannels;
	-
	- if (w->bytesLeft < fmt->wBlockAlign)
	- {
	- sample->flags \|= SOUND_SAMPLEFLAG_EOF;
	- return(0);
	- } /* if */
	-
	- w->bytesLeft -= fmt->wBlockAlign;
	-
	- for (i = 0; i < max; i++)
	- BAIL_IF_MACRO(!read_uint8(rw, &headers[i].bPredictor), NULL, 0);
	-
	- for (i = 0; i < max; i++)
	- BAIL_IF_MACRO(!read_le16(rw, &headers[i].iDelta), NULL, 0);
	-
	- for (i = 0; i < max; i++)
	- BAIL_IF_MACRO(!read_le16(rw, &headers[i].iSamp1), NULL, 0);
	-
	- for (i = 0; i < max; i++)
	- BAIL_IF_MACRO(!read_le16(rw, &headers[i].iSamp2), NULL, 0);
	-
	- fmt->fmt.adpcm.samples_left_in_block = fmt->fmt.adpcm.wSamplesPerBlock;
	- fmt->fmt.adpcm.nibble_state = 0;
	- return(1);
	-} /* read_adpcm_block_headers */
	-
	-
	-static __inline__ void do_adpcm_nibble(Uint8 nib,
	- ADPCMBLOCKHEADER *header,
	- Sint32 lPredSamp)
	-{
	- static const Sint32 max_audioval = ((1<<(16-1))-1);
	- static const Sint32 min_audioval = -(1<<(16-1));
	- static const Sint32 AdaptionTable[] =
	- {
	- 230, 230, 230, 230, 307, 409, 512, 614,
	- 768, 614, 512, 409, 307, 230, 230, 230
	- };
	-
	- Sint32 lNewSamp;
	- Sint32 delta;
	-
	- if (nib & 0x08)
	- lNewSamp = lPredSamp + (header->iDelta * (nib - 0x10));
	- else
	- lNewSamp = lPredSamp + (header->iDelta * nib);
	-
	- /* clamp value... */
	- if (lNewSamp < min_audioval)
	- lNewSamp = min_audioval;
	- else if (lNewSamp > max_audioval)
	- lNewSamp = max_audioval;
	-
	- delta = ((Sint32) header->iDelta * AdaptionTable[nib]) /
	- FIXED_POINT_ADAPTION_BASE;
	-
	- if (delta < SMALLEST_ADPCM_DELTA)
	- delta = SMALLEST_ADPCM_DELTA;
	-
	- header->iDelta = delta;
	- header->iSamp2 = header->iSamp1;
	- header->iSamp1 = lNewSamp;
	-} /* do_adpcm_nibble */
	-
	-
	-static __inline__ int decode_adpcm_sample_frame(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- wav_t w = (wav_t ) internal->decoder_private;
	- fmt_t *fmt = w->fmt;
	- ADPCMBLOCKHEADER *headers = fmt->fmt.adpcm.blockheaders;
	- SDL_RWops *rw = internal->rw;
	- int i;
	- int max = fmt->wChannels;
	- Sint32 delta;
	- Uint8 nib = fmt->fmt.adpcm.nibble;
	-
	- for (i = 0; i < max; i++)
	- {
	- Uint8 byte;
	- Sint16 iCoef1 = fmt->fmt.adpcm.aCoef[headers[i].bPredictor].iCoef1;
	- Sint16 iCoef2 = fmt->fmt.adpcm.aCoef[headers[i].bPredictor].iCoef2;
	- Sint32 lPredSamp = ((headers[i].iSamp1 * iCoef1) +
	- (headers[i].iSamp2 * iCoef2)) /
	- FIXED_POINT_COEF_BASE;
	-
	- if (fmt->fmt.adpcm.nibble_state == 0)
	- {
	- BAIL_IF_MACRO(!read_uint8(rw, &nib), NULL, 0);
	- fmt->fmt.adpcm.nibble_state = 1;
	- do_adpcm_nibble(nib >> 4, &headers[i], lPredSamp);
	- } /* if */
	- else
	- {
	- fmt->fmt.adpcm.nibble_state = 0;
	- do_adpcm_nibble(nib & 0x0F, &headers[i], lPredSamp);
	- } /* else */
	- } /* for */
	-
	- fmt->fmt.adpcm.nibble = nib;
	- return(1);
	-} /* decode_adpcm_sample_frame */
	-
	-
	-static __inline__ void put_adpcm_sample_frame1(void _buf, fmt_t fmt)
	-{
	- Uint16 buf = (Uint16 ) _buf;
	- ADPCMBLOCKHEADER *headers = fmt->fmt.adpcm.blockheaders;
	- int i;
	- for (i = 0; i < fmt->wChannels; i++)
	- *(buf++) = headers[i].iSamp1;
	-} /* put_adpcm_sample_frame1 */
	-
	-
	-static __inline__ void put_adpcm_sample_frame2(void _buf, fmt_t fmt)
	-{
	- Uint16 buf = (Uint16 ) _buf;
	- ADPCMBLOCKHEADER *headers = fmt->fmt.adpcm.blockheaders;
	- int i;
	- for (i = 0; i < fmt->wChannels; i++)
	- *(buf++) = headers[i].iSamp2;
	-} /* put_adpcm_sample_frame2 */
	-
	-
	-/*
	- * Sound_Decode() lands here for ADPCM-encoded WAVs...
	- */
	-static Uint32 read_sample_fmt_adpcm(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- wav_t w = (wav_t ) internal->decoder_private;
	- fmt_t *fmt = w->fmt;
	- Uint32 bw = 0;
	-
	- while (bw < internal->buffer_size)
	- {
	- /* write ongoing sample frame before reading more data... */
	- switch (fmt->fmt.adpcm.samples_left_in_block)
	- {
	- case 0: /* need to read a new block... */
	- if (!read_adpcm_block_headers(sample))
	- {
	- if ((sample->flags & SOUND_SAMPLEFLAG_EOF) == 0)
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- return(bw);
	- } /* if */
	-
	- /* only write first sample frame for now. */
	- put_adpcm_sample_frame2((Uint8 *) internal->buffer + bw, fmt);
	- fmt->fmt.adpcm.samples_left_in_block--;
	- bw += fmt->sample_frame_size;
	- break;
	-
	- case 1: /* output last sample frame of block... */
	- put_adpcm_sample_frame1((Uint8 *) internal->buffer + bw, fmt);
	- fmt->fmt.adpcm.samples_left_in_block--;
	- bw += fmt->sample_frame_size;
	- break;
	-
	- default: /* output latest sample frame and read a new one... */
	- put_adpcm_sample_frame1((Uint8 *) internal->buffer + bw, fmt);
	- fmt->fmt.adpcm.samples_left_in_block--;
	- bw += fmt->sample_frame_size;
	-
	- if (!decode_adpcm_sample_frame(sample))
	- {
	- sample->flags \|= SOUND_SAMPLEFLAG_ERROR;
	- return(bw);
	- } /* if */
	- } /* switch */
	- } /* while */
	-
	- return(bw);
	-} /* read_sample_fmt_adpcm */
	-
	-
	-/*
	- * Sound_FreeSample() lands here for ADPCM-encoded WAVs...
	- */
	-static void free_fmt_adpcm(fmt_t *fmt)
	-{
	- if (fmt->fmt.adpcm.aCoef != NULL)
	- free(fmt->fmt.adpcm.aCoef);
	-
	- if (fmt->fmt.adpcm.blockheaders != NULL)
	- free(fmt->fmt.adpcm.blockheaders);
	-} /* free_fmt_adpcm */
	-
	-
	-static int rewind_sample_fmt_adpcm(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- wav_t w = (wav_t ) internal->decoder_private;
	- w->fmt->fmt.adpcm.samples_left_in_block = 0;
	- return(1);
	-} /* rewind_sample_fmt_adpcm */
	-
	-
	-static int seek_sample_fmt_adpcm(Sound_Sample *sample, Uint32 ms)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- wav_t w = (wav_t ) internal->decoder_private;
	- fmt_t *fmt = w->fmt;
	- Uint32 origsampsleft = fmt->fmt.adpcm.samples_left_in_block;
	- int origpos = SDL_RWtell(internal->rw);
	- int offset = __Sound_convertMsToBytePos(&sample->actual, ms);
	- int bpb = (fmt->fmt.adpcm.wSamplesPerBlock * fmt->sample_frame_size);
	- int skipsize = (offset / bpb) * fmt->wBlockAlign;
	- int pos = skipsize + fmt->data_starting_offset;
	- int rc = SDL_RWseek(internal->rw, pos, SEEK_SET);
	- BAIL_IF_MACRO(rc != pos, ERR_IO_ERROR, 0);
	-
	- /* The offset we need is in this block, so we need to decode to there. */
	- skipsize += (offset % bpb);
	- rc = (offset % bpb); /* bytes into this block we need to decode */
	- if (!read_adpcm_block_headers(sample))
	- {
	- SDL_RWseek(internal->rw, origpos, SEEK_SET); /* try to make sane. */
	- return(0);
	- } /* if */
	-
	- /* first sample frame of block is a freebie. :) */
	- fmt->fmt.adpcm.samples_left_in_block--;
	- rc -= fmt->sample_frame_size;
	- while (rc > 0)
	- {
	- if (!decode_adpcm_sample_frame(sample))
	- {
	- SDL_RWseek(internal->rw, origpos, SEEK_SET);
	- fmt->fmt.adpcm.samples_left_in_block = origsampsleft;
	- return(0);
	- } /* if */
	-
	- fmt->fmt.adpcm.samples_left_in_block--;
	- rc -= fmt->sample_frame_size;
	- } /* while */
	-
	- w->bytesLeft = fmt->total_bytes - skipsize;
	- return(1); /* success. */
	-} /* seek_sample_fmt_adpcm */
	-
	-
	-/*
	- * Read in the adpcm-specific info from disk. This makes this process
	- * safe regardless of the processor's byte order or how the fmt_t
	- * structure is packed.
	- */
	-static int read_fmt_adpcm(SDL_RWops rw, fmt_t fmt)
	-{
	- size_t i;
	-
	- memset(&fmt->fmt.adpcm, '\0', sizeof (fmt->fmt.adpcm));
	- fmt->free = free_fmt_adpcm;
	- fmt->read_sample = read_sample_fmt_adpcm;
	- fmt->rewind_sample = rewind_sample_fmt_adpcm;
	- fmt->seek_sample = seek_sample_fmt_adpcm;
	-
	- BAIL_IF_MACRO(!read_le16(rw, &fmt->fmt.adpcm.cbSize), NULL, 0);
	- BAIL_IF_MACRO(!read_le16(rw, &fmt->fmt.adpcm.wSamplesPerBlock), NULL, 0);
	- BAIL_IF_MACRO(!read_le16(rw, &fmt->fmt.adpcm.wNumCoef), NULL, 0);
	-
	- /* fmt->free() is always called, so these malloc()s will be cleaned up. */
	-
	- i = sizeof (ADPCMCOEFSET) * fmt->fmt.adpcm.wNumCoef;
	- fmt->fmt.adpcm.aCoef = (ADPCMCOEFSET *) malloc(i);
	- BAIL_IF_MACRO(fmt->fmt.adpcm.aCoef == NULL, ERR_OUT_OF_MEMORY, 0);
	-
	- for (i = 0; i < fmt->fmt.adpcm.wNumCoef; i++)
	- {
	- BAIL_IF_MACRO(!read_le16(rw, &fmt->fmt.adpcm.aCoef[i].iCoef1), NULL, 0);
	- BAIL_IF_MACRO(!read_le16(rw, &fmt->fmt.adpcm.aCoef[i].iCoef2), NULL, 0);
	- } /* for */
	-
	- i = sizeof (ADPCMBLOCKHEADER) * fmt->wChannels;
	- fmt->fmt.adpcm.blockheaders = (ADPCMBLOCKHEADER *) malloc(i);
	- BAIL_IF_MACRO(fmt->fmt.adpcm.blockheaders == NULL, ERR_OUT_OF_MEMORY, 0);
	-
	- return(1);
	-} /* read_fmt_adpcm */
	-
	-
	-
	-/*****************************************************************************
	- * Everything else... *
	- *****************************************************************************/
	-
	-static int WAV_init(void)
	-{
	- return(1); /* always succeeds. */
	-} /* WAV_init */
	-
	-
	-static void WAV_quit(void)
	-{
	- /* it's a no-op. */
	-} /* WAV_quit */
	-
	-
	-static int read_fmt(SDL_RWops rw, fmt_t fmt)
	-{
	- /* if it's in this switch statement, we support the format. */
	- switch (fmt->wFormatTag)
	- {
	- case FMT_NORMAL:
	- SNDDBG(("WAV: Appears to be uncompressed audio.\n"));
	- return(read_fmt_normal(rw, fmt));
	-
	- case FMT_ADPCM:
	- SNDDBG(("WAV: Appears to be ADPCM compressed audio.\n"));
	- return(read_fmt_adpcm(rw, fmt));
	-
	- /* add other types here. */
	-
	- default:
	- SNDDBG(("WAV: Format 0x%X is unknown.\n",
	- (unsigned int) fmt->wFormatTag));
	- BAIL_MACRO("WAV: Unsupported format", 0);
	- } /* switch */
	-
	- assert(0); /* shouldn't hit this point. */
	- return(0);
	-} /* read_fmt */
	-
	-
	-/*
	- * Locate a specific chunk in the WAVE file by ID...
	- */
	-static int find_chunk(SDL_RWops *rw, Uint32 id)
	-{
	- Sint32 siz = 0;
	- Uint32 _id = 0;
	- Uint32 pos = SDL_RWtell(rw);
	-
	- while (1)
	- {
	- BAIL_IF_MACRO(!read_le32(rw, &_id), NULL, 0);
	- if (_id == id)
	- return(1);
	-
	- /* skip ahead and see what next chunk is... */
	- BAIL_IF_MACRO(!read_le32(rw, &siz), NULL, 0);
	- assert(siz >= 0);
	- pos += (sizeof (Uint32) * 2) + siz;
	- if (siz > 0)
	- BAIL_IF_MACRO(SDL_RWseek(rw, pos, SEEK_SET) != pos, NULL, 0);
	- } /* while */
	-
	- return(0); /* shouldn't hit this, but just in case... */
	-} /* find_chunk */
	-
	-
	-static int WAV_open_internal(Sound_Sample sample, const char ext, fmt_t *fmt)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- SDL_RWops *rw = internal->rw;
	- data_t d;
	- wav_t *w;
	- Uint32 pos;
	-
	- BAIL_IF_MACRO(SDL_ReadLE32(rw) != riffID, "WAV: Not a RIFF file.", 0);
	- SDL_ReadLE32(rw); /* throw the length away; we get this info later. */
	- BAIL_IF_MACRO(SDL_ReadLE32(rw) != waveID, "WAV: Not a WAVE file.", 0);
	- BAIL_IF_MACRO(!find_chunk(rw, fmtID), "WAV: No format chunk.", 0);
	- BAIL_IF_MACRO(!read_fmt_chunk(rw, fmt), "WAV: Can't read format chunk.", 0);
	-
	- sample->actual.channels = (Uint8) fmt->wChannels;
	- sample->actual.rate = fmt->dwSamplesPerSec;
	- if ((fmt->wBitsPerSample == 4) /\|\| (fmt->wBitsPerSample == 0) / )
	- sample->actual.format = AUDIO_S16SYS;
	- else if (fmt->wBitsPerSample == 8)
	- sample->actual.format = AUDIO_U8;
	- else if (fmt->wBitsPerSample == 16)
	- sample->actual.format = AUDIO_S16LSB;
	- else
	- {
	- SNDDBG(("WAV: %d bits per sample!?\n", (int) fmt->wBitsPerSample));
	- BAIL_MACRO("WAV: Unsupported sample size.", 0);
	- } /* else */
	-
	- BAIL_IF_MACRO(!read_fmt(rw, fmt), NULL, 0);
	- SDL_RWseek(rw, fmt->next_chunk_offset, SEEK_SET);
	- BAIL_IF_MACRO(!find_chunk(rw, dataID), "WAV: No data chunk.", 0);
	- BAIL_IF_MACRO(!read_data_chunk(rw, &d), "WAV: Can't read data chunk.", 0);
	-
	- w = (wav_t *) malloc(sizeof(wav_t));
	- BAIL_IF_MACRO(w == NULL, ERR_OUT_OF_MEMORY, 0);
	- w->fmt = fmt;
	- fmt->total_bytes = w->bytesLeft = d.chunkSize;
	- fmt->data_starting_offset = SDL_RWtell(rw);
	- fmt->sample_frame_size = ( ((sample->actual.format & 0xFF) / 8) *
	- sample->actual.channels );
	-
	- internal->decoder_private = (void *) w;
	-
	- sample->flags = SOUND_SAMPLEFLAG_NONE;
	- if (fmt->seek_sample != NULL)
	- sample->flags \|= SOUND_SAMPLEFLAG_CANSEEK;
	-
	- SNDDBG(("WAV: Accepting data stream.\n"));
	- return(1); /* we'll handle this data. */
	-} /* WAV_open_internal */
	-
	-
	-static int WAV_open(Sound_Sample sample, const char ext)
	-{
	- int rc;
	-
	- fmt_t fmt = (fmt_t ) malloc(sizeof (fmt_t));
	- BAIL_IF_MACRO(fmt == NULL, ERR_OUT_OF_MEMORY, 0);
	- memset(fmt, '\0', sizeof (fmt_t));
	-
	- rc = WAV_open_internal(sample, ext, fmt);
	- if (!rc)
	- {
	- if (fmt->free != NULL)
	- fmt->free(fmt);
	- free(fmt);
	- } /* if */
	-
	- return(rc);
	-} /* WAV_open */
	-
	-
	-static void WAV_close(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- wav_t w = (wav_t ) internal->decoder_private;
	-
	- if (w->fmt->free != NULL)
	- w->fmt->free(w->fmt);
	-
	- free(w->fmt);
	- free(w);
	-} /* WAV_close */
	-
	-
	-static Uint32 WAV_read(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- wav_t w = (wav_t ) internal->decoder_private;
	- return(w->fmt->read_sample(sample));
	-} /* WAV_read */
	-
	-
	-static int WAV_rewind(Sound_Sample *sample)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- wav_t w = (wav_t ) internal->decoder_private;
	- fmt_t *fmt = w->fmt;
	- int rc = SDL_RWseek(internal->rw, fmt->data_starting_offset, SEEK_SET);
	- BAIL_IF_MACRO(rc != fmt->data_starting_offset, ERR_IO_ERROR, 0);
	- w->bytesLeft = fmt->total_bytes;
	- return(fmt->rewind_sample(sample));
	-} /* WAV_rewind */
	-
	-
	-static int WAV_seek(Sound_Sample *sample, Uint32 ms)
	-{
	- Sound_SampleInternal internal = (Sound_SampleInternal ) sample->opaque;
	- wav_t w = (wav_t ) internal->decoder_private;
	- return(w->fmt->seek_sample(sample, ms));
	-} /* WAV_seek */
	-
	-#endif /* SOUND_SUPPORTS_WAV */
	-
	-/* end of wav.c ... */
	-
	diff --git a/util/sdl/sound/docs/README b/util/sdl/sound/docs/README
	deleted file mode 100644
	index 4f9d16f4..00000000
	--- a/util/sdl/sound/docs/README
	+++ /dev/null
	@@ -1,3 +0,0 @@
	-Docs are generated with the program Doxygen (http://www.doxygen.org/),
	- or can be read online at http://icculus.org/SDL_sound/docs/
	-
	diff --git a/util/sdl/sound/extra_rwops.c b/util/sdl/sound/extra_rwops.c
	deleted file mode 100644
	index 6ea92c30..00000000
	--- a/util/sdl/sound/extra_rwops.c
	+++ /dev/null
	@@ -1,135 +0,0 @@
	-/*
	- * SDL_sound -- An abstract sound format decoding API.
	- * Copyright (C) 2001 Ryan C. Gordon.
	- *
	- * This library is free software; you can redistribute it and/or
	- * modify it under the terms of the GNU Lesser General Public
	- * License as published by the Free Software Foundation; either
	- * version 2.1 of the License, or (at your option) any later version.
	- *
	- * This library is distributed in the hope that it will be useful,
	- * but WITHOUT ANY WARRANTY; without even the implied warranty of
	- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	- * Lesser General Public License for more details.
	- *
	- * You should have received a copy of the GNU Lesser General Public
	- * License along with this library; if not, write to the Free Software
	- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
	- */
	-
	-/*
	- * Some extra RWops that are needed or are just handy to have.
	- *
	- * Please see the file COPYING in the source's root directory.
	- *
	- * This file written by Ryan C. Gordon. (icculus@icculus.org)
	- */
	-
	-#include <stdio.h>
	-#include <stdlib.h>
	-#include "SDL.h"
	-
	-
	- /*
	- * The Reference Counter RWops...
	- */
	-
	-
	-typedef struct
	-{
	- SDL_RWops rw; / The actual RWops we're refcounting... */
	- int refcount; /* The refcount; starts at 1. If goes to 0, delete. */
	-} RWRefCounterData;
	-
	-
	-/* Just pass through to the actual SDL_RWops's method... */
	-static int refcounter_seek(SDL_RWops *rw, int offset, int whence)
	-{
	- RWRefCounterData data = (RWRefCounterData ) rw->hidden.unknown.data1;
	- return(data->rw->seek(data->rw, offset, whence));
	-} /* refcounter_seek */
	-
	-
	-/* Just pass through to the actual SDL_RWops's method... */
	-static int refcounter_read(SDL_RWops rw, void ptr, int size, int maxnum)
	-{
	- RWRefCounterData data = (RWRefCounterData ) rw->hidden.unknown.data1;
	- return(data->rw->read(data->rw, ptr, size, maxnum));
	-} /* refcounter_read */
	-
	-
	-/* Just pass through to the actual SDL_RWops's method... */
	-static int refcounter_write(SDL_RWops rw, const void ptr, int size, int num)
	-{
	- RWRefCounterData data = (RWRefCounterData ) rw->hidden.unknown.data1;
	- return(data->rw->write(data->rw, ptr, size, num));
	-} /* refcounter_write */
	-
	-
	-/*
	- * Decrement the reference count. If there are no more references, pass
	- * through to the actual SDL_RWops's method, and then clean ourselves up.
	- */
	-static int refcounter_close(SDL_RWops *rw)
	-{
	- int retval = 0;
	- RWRefCounterData data = (RWRefCounterData ) rw->hidden.unknown.data1;
	- data->refcount--;
	- if (data->refcount <= 0)
	- {
	- retval = data->rw->close(data->rw);
	- free(data);
	- SDL_FreeRW(rw);
	- } /* if */
	-
	- return(retval);
	-} /* refcounter_close */
	-
	-
	-void RWops_RWRefCounter_addRef(SDL_RWops *rw)
	-{
	- RWRefCounterData data = (RWRefCounterData ) rw->hidden.unknown.data1;
	- data->refcount++;
	-} /* RWops_RWRefCounter_addRef */
	-
	-
	-SDL_RWops RWops_RWRefCounter_new(SDL_RWops rw)
	-{
	- SDL_RWops *retval = NULL;
	-
	- if (rw == NULL)
	- {
	- SDL_SetError("NULL argument to RWops_RWRefCounter_new().");
	- return(NULL);
	- } /* if */
	-
	- retval = SDL_AllocRW();
	- if (retval != NULL)
	- {
	- RWRefCounterData *data;
	- data = (RWRefCounterData *) malloc(sizeof (RWRefCounterData));
	- if (data == NULL)
	- {
	- SDL_SetError("Out of memory.");
	- SDL_FreeRW(retval);
	- retval = NULL;
	- } /* if */
	- else
	- {
	- data->rw = rw;
	- data->refcount = 1;
	- retval->hidden.unknown.data1 = data;
	- retval->seek = refcounter_seek;
	- retval->read = refcounter_read;
	- retval->write = refcounter_write;
	- retval->close = refcounter_close;
	- } /* else */
	- } /* if */
	-
	- return(retval);
	-} /* RWops_RWRefCounter_new */
	-
	-
	-/* end of extra_rwops.c ... */
	-
	-
	diff --git a/util/sdl/sound/extra_rwops.h b/util/sdl/sound/extra_rwops.h
	deleted file mode 100644
	index f86b5564..00000000
	--- a/util/sdl/sound/extra_rwops.h
	+++ /dev/null
	@@ -1,71 +0,0 @@
	-/*
	- * SDL_sound -- An abstract sound format decoding API.
	- * Copyright (C) 2001 Ryan C. Gordon.
	- *
	- * This library is free software; you can redistribute it and/or
	- * modify it under the terms of the GNU Lesser General Public
	- * License as published by the Free Software Foundation; either
	- * version 2.1 of the License, or (at your option) any later version.
	- *
	- * This library is distributed in the hope that it will be useful,
	- * but WITHOUT ANY WARRANTY; without even the implied warranty of
	- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	- * Lesser General Public License for more details.
	- *
	- * You should have received a copy of the GNU Lesser General Public
	- * License along with this library; if not, write to the Free Software
	- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
	- */
	-
	-/*
	- * Some extra RWops that are needed or are just handy to have.
	- *
	- * Please see the file COPYING in the source's root directory.
	- *
	- * This file written by Ryan C. Gordon. (icculus@icculus.org)
	- */
	-
	-#ifndef _INCLUDE_EXTRA_RWOPS_H_
	-#define _INCLUDE_EXTRA_RWOPS_H_
	-
	-#include "SDL.h"
	-
	-#ifdef __cplusplus
	-extern "C" {
	-#endif
	-
	-/*
	- * The Reference Counter RWops...
	- *
	- * This wraps another RWops with a reference counter. When you create a
	- * reference counter RWops, it sets a counter to one. Everytime you call
	- * RWops_RWRefCounter_new(), that's RWops's counter increments by one.
	- * Everytime you call that RWops's close() method, the counter decrements
	- * by one. If the counter hits zero, the original RWops's close() method
	- * is called, and the reference counting wrapper deletes itself. The read,
	- * write, and seek methods just pass through to the original.
	- *
	- * This is handy if you have two libraries (in the original case, SDL_sound
	- * and SMPEG), who both want an SDL_RWops, and both want to close it when
	- * they are finished. This resolves that contention. The user creates a
	- * RWops, passes it to SDL_sound, which wraps it in a reference counter and
	- * increments the number of references, and passes the wrapped RWops to
	- * SMPEG. SMPEG "closes" this wrapped RWops when the MP3 has finished
	- * playing, and SDL_sound then closes it, too. This second closing removes
	- * the last reference, and the RWops is smoothly destructed.
	- */
	-
	-/* Return a SDL_RWops that is a reference counting wrapper of (rw). */
	-SDL_RWops RWops_RWRefCounter_new(SDL_RWops rw);
	-
	-/* Increment a reference counting RWops's refcount by one. */
	-void RWops_RWRefCounter_addRef(SDL_RWops *rw);
	-
	-#ifdef __cplusplus
	-}
	-#endif
	-
	-#endif /* !defined _INCLUDE_EXTRA_RWOPS_H_ */
	-
	-/* end of extra_rwops.h ... */
	-
	diff --git a/util/sdl/sound/install-sh b/util/sdl/sound/install-sh
	deleted file mode 100755
	index 4d4a9519..00000000
	--- a/util/sdl/sound/install-sh
	+++ /dev/null
	@@ -1,323 +0,0 @@
	-#!/bin/sh
	-# install - install a program, script, or datafile
	-
	-scriptversion=2005-05-14.22
	-
	-# This originates from X11R5 (mit/util/scripts/install.sh), which was
	-# later released in X11R6 (xc/config/util/install.sh) with the
	-# following copyright and license.
	-#
	-# Copyright (C) 1994 X Consortium
	-#
	-# Permission is hereby granted, free of charge, to any person obtaining a copy
	-# of this software and associated documentation files (the "Software"), to
	-# deal in the Software without restriction, including without limitation the
	-# rights to use, copy, modify, merge, publish, distribute, sublicense, and/or
	-# sell copies of the Software, and to permit persons to whom the Software is
	-# furnished to do so, subject to the following conditions:
	-#
	-# The above copyright notice and this permission notice shall be included in
	-# all copies or substantial portions of the Software.
	-#
	-# THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
	-# IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
	-# FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
	-# X CONSORTIUM BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN
	-# AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNEC-
	-# TION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
	-#
	-# Except as contained in this notice, the name of the X Consortium shall not
	-# be used in advertising or otherwise to promote the sale, use or other deal-
	-# ings in this Software without prior written authorization from the X Consor-
	-# tium.
	-#
	-#
	-# FSF changes to this file are in the public domain.
	-#
	-# Calling this script install-sh is preferred over install.sh, to prevent
	-# `make' implicit rules from creating a file called install from it
	-# when there is no Makefile.
	-#
	-# This script is compatible with the BSD install script, but was written
	-# from scratch. It can only install one file at a time, a restriction
	-# shared with many OS's install programs.
	-
	-# set DOITPROG to echo to test this script
	-
	-# Don't use :- since 4.3BSD and earlier shells don't like it.
	-doit="${DOITPROG-}"
	-
	-# put in absolute paths if you don't have them in your path; or use env. vars.
	-
	-mvprog="${MVPROG-mv}"
	-cpprog="${CPPROG-cp}"
	-chmodprog="${CHMODPROG-chmod}"
	-chownprog="${CHOWNPROG-chown}"
	-chgrpprog="${CHGRPPROG-chgrp}"
	-stripprog="${STRIPPROG-strip}"
	-rmprog="${RMPROG-rm}"
	-mkdirprog="${MKDIRPROG-mkdir}"
	-
	-chmodcmd="$chmodprog 0755"
	-chowncmd=
	-chgrpcmd=
	-stripcmd=
	-rmcmd="$rmprog -f"
	-mvcmd="$mvprog"
	-src=
	-dst=
	-dir_arg=
	-dstarg=
	-no_target_directory=
	-
	-usage="Usage: $0 [OPTION]... [-T] SRCFILE DSTFILE
	- or: $0 [OPTION]... SRCFILES... DIRECTORY
	- or: $0 [OPTION]... -t DIRECTORY SRCFILES...
	- or: $0 [OPTION]... -d DIRECTORIES...
	-
	-In the 1st form, copy SRCFILE to DSTFILE.
	-In the 2nd and 3rd, copy all SRCFILES to DIRECTORY.
	-In the 4th, create DIRECTORIES.
	-
	-Options:
	--c (ignored)
	--d create directories instead of installing files.
	--g GROUP $chgrpprog installed files to GROUP.
	--m MODE $chmodprog installed files to MODE.
	--o USER $chownprog installed files to USER.
	--s $stripprog installed files.
	--t DIRECTORY install into DIRECTORY.
	--T report an error if DSTFILE is a directory.
	---help display this help and exit.
	---version display version info and exit.
	-
	-Environment variables override the default commands:
	- CHGRPPROG CHMODPROG CHOWNPROG CPPROG MKDIRPROG MVPROG RMPROG STRIPPROG
	-"
	-
	-while test -n "$1"; do
	- case $1 in
	- -c) shift
	- continue;;
	-
	- -d) dir_arg=true
	- shift
	- continue;;
	-
	- -g) chgrpcmd="$chgrpprog $2"
	- shift
	- shift
	- continue;;
	-
	- --help) echo "$usage"; exit $?;;
	-
	- -m) chmodcmd="$chmodprog $2"
	- shift
	- shift
	- continue;;
	-
	- -o) chowncmd="$chownprog $2"
	- shift
	- shift
	- continue;;
	-
	- -s) stripcmd=$stripprog
	- shift
	- continue;;
	-
	- -t) dstarg=$2
	- shift
	- shift
	- continue;;
	-
	- -T) no_target_directory=true
	- shift
	- continue;;
	-
	- --version) echo "$0 $scriptversion"; exit $?;;
	-
	- *) # When -d is used, all remaining arguments are directories to create.
	- # When -t is used, the destination is already specified.
	- test -n "$dir_arg$dstarg" && break
	- # Otherwise, the last argument is the destination. Remove it from $@.
	- for arg
	- do
	- if test -n "$dstarg"; then
	- # $@ is not empty: it contains at least $arg.
	- set fnord "$@" "$dstarg"
	- shift # fnord
	- fi
	- shift # arg
	- dstarg=$arg
	- done
	- break;;
	- esac
	-done
	-
	-if test -z "$1"; then
	- if test -z "$dir_arg"; then
	- echo "$0: no input file specified." >&2
	- exit 1
	- fi
	- # It's OK to call `install-sh -d' without argument.
	- # This can happen when creating conditional directories.
	- exit 0
	-fi
	-
	-for src
	-do
	- # Protect names starting with `-'.
	- case $src in
	- -*) src=./$src ;;
	- esac
	-
	- if test -n "$dir_arg"; then
	- dst=$src
	- src=
	-
	- if test -d "$dst"; then
	- mkdircmd=:
	- chmodcmd=
	- else
	- mkdircmd=$mkdirprog
	- fi
	- else
	- # Waiting for this to be detected by the "$cpprog $src $dsttmp" command
	- # might cause directories to be created, which would be especially bad
	- # if $src (and thus $dsttmp) contains '*'.
	- if test ! -f "$src" && test ! -d "$src"; then
	- echo "$0: $src does not exist." >&2
	- exit 1
	- fi
	-
	- if test -z "$dstarg"; then
	- echo "$0: no destination specified." >&2
	- exit 1
	- fi
	-
	- dst=$dstarg
	- # Protect names starting with `-'.
	- case $dst in
	- -*) dst=./$dst ;;
	- esac
	-
	- # If destination is a directory, append the input filename; won't work
	- # if double slashes aren't ignored.
	- if test -d "$dst"; then
	- if test -n "$no_target_directory"; then
	- echo "$0: $dstarg: Is a directory" >&2
	- exit 1
	- fi
	- dst=$dst/`basename "$src"`
	- fi
	- fi
	-
	- # This sed command emulates the dirname command.
	- dstdir=`echo "$dst" \| sed -e 's,/$,,;s,[^/]$,,;s,/*$,,;s,^$,.,'`
	-
	- # Make sure that the destination directory exists.
	-
	- # Skip lots of stat calls in the usual case.
	- if test ! -d "$dstdir"; then
	- defaultIFS='
	- '
	- IFS="${IFS-$defaultIFS}"
	-
	- oIFS=$IFS
	- # Some sh's can't handle IFS=/ for some reason.
	- IFS='%'
	- set x `echo "$dstdir" \| sed -e 's@/@%@g' -e 's@^%@/@'`
	- shift
	- IFS=$oIFS
	-
	- pathcomp=
	-
	- while test $# -ne 0 ; do
	- pathcomp=$pathcomp$1
	- shift
	- if test ! -d "$pathcomp"; then
	- $mkdirprog "$pathcomp"
	- # mkdir can fail with a `File exist' error in case several
	- # install-sh are creating the directory concurrently. This
	- # is OK.
	- test -d "$pathcomp" \|\| exit
	- fi
	- pathcomp=$pathcomp/
	- done
	- fi
	-
	- if test -n "$dir_arg"; then
	- $doit $mkdircmd "$dst" \
	- && { test -z "$chowncmd" \|\| $doit $chowncmd "$dst"; } \
	- && { test -z "$chgrpcmd" \|\| $doit $chgrpcmd "$dst"; } \
	- && { test -z "$stripcmd" \|\| $doit $stripcmd "$dst"; } \
	- && { test -z "$chmodcmd" \|\| $doit $chmodcmd "$dst"; }
	-
	- else
	- dstfile=`basename "$dst"`
	-
	- # Make a couple of temp file names in the proper directory.
	- dsttmp=$dstdir/_inst.$$_
	- rmtmp=$dstdir/_rm.$$_
	-
	- # Trap to clean up those temp files at exit.
	- trap 'ret=$?; rm -f "$dsttmp" "$rmtmp" && exit $ret' 0
	- trap '(exit $?); exit' 1 2 13 15
	-
	- # Copy the file name to the temp name.
	- $doit $cpprog "$src" "$dsttmp" &&
	-
	- # and set any options; do chmod last to preserve setuid bits.
	- #
	- # If any of these fail, we abort the whole thing. If we want to
	- # ignore errors from any of these, just make sure not to ignore
	- # errors from the above "$doit $cpprog $src $dsttmp" command.
	- #
	- { test -z "$chowncmd" \|\| $doit $chowncmd "$dsttmp"; } \
	- && { test -z "$chgrpcmd" \|\| $doit $chgrpcmd "$dsttmp"; } \
	- && { test -z "$stripcmd" \|\| $doit $stripcmd "$dsttmp"; } \
	- && { test -z "$chmodcmd" \|\| $doit $chmodcmd "$dsttmp"; } &&
	-
	- # Now rename the file to the real destination.
	- { $doit $mvcmd -f "$dsttmp" "$dstdir/$dstfile" 2>/dev/null \
	- \|\| {
	- # The rename failed, perhaps because mv can't rename something else
	- # to itself, or perhaps because mv is so ancient that it does not
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	- END { for (i in files) print i; }'`; \
	- mkid -fID $$unique
	-tags: TAGS
	-
	-TAGS: $(HEADERS) $(SOURCES) $(TAGS_DEPENDENCIES) \
	- $(TAGS_FILES) $(LISP)
	- tags=; \
	- here=`pwd`; \
	- list='$(SOURCES) $(HEADERS) $(LISP) $(TAGS_FILES)'; \
	- unique=`for i in $$list; do \
	- if test -f "$$i"; then echo $$i; else echo $(srcdir)/$$i; fi; \
	- done \| \
	- $(AWK) ' { files[$$0] = 1; } \
	- END { for (i in files) print i; }'`; \
	- if test -z "$(ETAGS_ARGS)$$tags$$unique"; then :; else \
	- test -n "$$unique" \|\| unique=$$empty_fix; \
	- $(ETAGS) $(ETAGSFLAGS) $(AM_ETAGSFLAGS) $(ETAGS_ARGS) \
	- $$tags $$unique; \
	- fi
	-ctags: CTAGS
	-CTAGS: $(HEADERS) $(SOURCES) $(TAGS_DEPENDENCIES) \
	- $(TAGS_FILES) $(LISP)
	- tags=; \
	- here=`pwd`; \
	- list='$(SOURCES) $(HEADERS) $(LISP) $(TAGS_FILES)'; \
	- unique=`for i in $$list; do \
	- if test -f "$$i"; then echo $$i; else echo $(srcdir)/$$i; fi; \
	- done \| \
	- $(AWK) ' { files[$$0] = 1; } \
	- END { for (i in files) print i; }'`; \
	- test -z "$(CTAGS_ARGS)$$tags$$unique" \
	- \|\| $(CTAGS) $(CTAGSFLAGS) $(AM_CTAGSFLAGS) $(CTAGS_ARGS) \
	- $$tags $$unique
	-
	-GTAGS:
	- here=`$(am__cd) $(top_builddir) && pwd` \
	- && cd $(top_srcdir) \
	- && gtags -i $(GTAGS_ARGS) $$here
	-
	-distclean-tags:
	- -rm -f TAGS ID GTAGS GRTAGS GSYMS GPATH tags
	-
	-distdir: $(DISTFILES)
	- @srcdirstrip=`echo "$(srcdir)" \| sed 's\|.\|.\|g'`; \
	- topsrcdirstrip=`echo "$(top_srcdir)" \| sed 's\|.\|.\|g'`; \
	- list='$(DISTFILES)'; for file in $$list; do \
	- case $$file in \
	- $(srcdir)/*) file=`echo "$$file" \| sed "s\|^$$srcdirstrip/\|\|"`;; \
	- $(top_srcdir)/*) file=`echo "$$file" \| sed "s\|^$$topsrcdirstrip/\|$(top_builddir)/\|"`;; \
	- esac; \
	- if test -f $$file \|\| test -d $$file; then d=.; else d=$(srcdir); fi; \
	- dir=`echo "$$file" \| sed -e 's,/[^/]*$$,,'`; \
	- if test "$$dir" != "$$file" && test "$$dir" != "."; then \
	- dir="/$$dir"; \
	- $(mkdir_p) "$(distdir)$$dir"; \
	- else \
	- dir=''; \
	- fi; \
	- if test -d $$d/$$file; then \
	- if test -d $(srcdir)/$$file && test $$d != $(srcdir); then \
	- cp -pR $(srcdir)/$$file $(distdir)$$dir \|\| exit 1; \
	- fi; \
	- cp -pR $$d/$$file $(distdir)$$dir \|\| exit 1; \
	- else \
	- test -f $(distdir)/$$file \
	- \|\| cp -p $$d/$$file $(distdir)/$$file \
	- \|\| exit 1; \
	- fi; \
	- done
	-check-am: all-am
	-check: check-am
	-all-am: Makefile $(PROGRAMS)
	-installdirs:
	- for dir in "$(DESTDIR)$(bindir)"; do \
	- test -z "$$dir" \|\| $(mkdir_p) "$$dir"; \
	- done
	-install: install-am
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	-
	-install-am: all-am
	- @$(MAKE) $(AM_MAKEFLAGS) install-exec-am install-data-am
	-
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	-install-strip:
	- $(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
	- install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
	- `test -z '$(STRIP)' \|\| \
	- echo "INSTALL_PROGRAM_ENV=STRIPPROG='$(STRIP)'"` install
	-mostlyclean-generic:
	-
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	-
	-distclean-generic:
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	-
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	- @echo "it deletes files that may require special tools to rebuild."
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	-
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	- -rm -rf ./$(DEPDIR)
	- -rm -f Makefile
	-distclean-am: clean-am distclean-compile distclean-generic \
	- distclean-libtool distclean-tags
	-
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	- -rm -rf ./$(DEPDIR)
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	- distclean-generic distclean-libtool distclean-tags distdir dvi \
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	- pdf pdf-am ps ps-am tags uninstall uninstall-am \
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	-
	-# Tell versions [3.59,3.63) of GNU make to not export all variables.
	-# Otherwise a system limit (for SysV at least) may be exceeded.
	-.NOEXPORT:
	diff --git a/util/sdl/sound/playsound/physfsrwops.c b/util/sdl/sound/playsound/physfsrwops.c
	deleted file mode 100644
	index 42377fe5..00000000
	--- a/util/sdl/sound/playsound/physfsrwops.c
	+++ /dev/null
	@@ -1,195 +0,0 @@
	-/*
	- * This code provides a glue layer between PhysicsFS and Simple Directmedia
	- * Layer's (SDL) RWops i/o abstraction.
	- *
	- * License: this code is public domain. I make no warranty that it is useful,
	- * correct, harmless, or environmentally safe.
	- *
	- * This particular file may be used however you like, including copying it
	- * verbatim into a closed-source project, exploiting it commercially, and
	- * removing any trace of my name from the source (although I hope you won't
	- * do that). I welcome enhancements and corrections to this file, but I do
	- * not require you to send me patches if you make changes.
	- *
	- * Unless otherwise stated, the rest of PhysicsFS falls under the GNU Lesser
	- * General Public License: http://www.gnu.org/licenses/lgpl.txt
	- *
	- * SDL falls under the LGPL, too. You can get SDL at http://www.libsdl.org/
	- *
	- * This file was written by Ryan C. Gordon. (icculus@icculus.org).
	- */
	-
	-#if SUPPORT_PHYSFS
	-
	-#include <stdio.h> /* used for SEEK_SET, SEEK_CUR, SEEK_END ... */
	-#include "physfsrwops.h"
	-
	-static int physfsrwops_seek(SDL_RWops *rw, int offset, int whence)
	-{
	- PHYSFS_file handle = (PHYSFS_file ) rw->hidden.unknown.data1;
	- int pos = 0;
	-
	- if (whence == SEEK_SET)
	- {
	- pos = offset;
	- } /* if */
	-
	- else if (whence == SEEK_CUR)
	- {
	- PHYSFS_sint64 current = PHYSFS_tell(handle);
	- if (current == -1)
	- {
	- SDL_SetError("Can't find position in file: %s",
	- PHYSFS_getLastError());
	- return(-1);
	- } /* if */
	-
	- pos = (int) current;
	- if ( ((PHYSFS_sint64) pos) != current )
	- {
	- SDL_SetError("Can't fit current file position in an int!");
	- return(-1);
	- } /* if */
	-
	- if (offset == 0) /* this is a "tell" call. We're done. */
	- return(pos);
	-
	- pos += offset;
	- } /* else if */
	-
	- else if (whence == SEEK_END)
	- {
	- PHYSFS_sint64 len = PHYSFS_fileLength(handle);
	- if (len == -1)
	- {
	- SDL_SetError("Can't find end of file: %s", PHYSFS_getLastError());
	- return(-1);
	- } /* if */
	-
	- pos = (int) len;
	- if ( ((PHYSFS_sint64) pos) != len )
	- {
	- SDL_SetError("Can't fit end-of-file position in an int!");
	- return(-1);
	- } /* if */
	-
	- pos += offset;
	- } /* else if */
	-
	- else
	- {
	- SDL_SetError("Invalid 'whence' parameter.");
	- return(-1);
	- } /* else */
	-
	- if ( pos < 0 )
	- {
	- SDL_SetError("Attempt to seek past start of file.");
	- return(-1);
	- } /* if */
	-
	- if (!PHYSFS_seek(handle, (PHYSFS_uint64) pos))
	- {
	- SDL_SetError("PhysicsFS error: %s", PHYSFS_getLastError());
	- return(-1);
	- } /* if */
	-
	- return(pos);
	-} /* physfsrwops_seek */
	-
	-
	-static int physfsrwops_read(SDL_RWops rw, void ptr, int size, int maxnum)
	-{
	- PHYSFS_file handle = (PHYSFS_file ) rw->hidden.unknown.data1;
	- PHYSFS_sint64 rc = PHYSFS_read(handle, ptr, size, maxnum);
	- if (rc != maxnum)
	- {
	- if (!PHYSFS_eof(handle)) /* not EOF? Must be an error. */
	- SDL_SetError("PhysicsFS error: %s", PHYSFS_getLastError());
	- } /* if */
	-
	- return((int) rc);
	-} /* physfsrwops_read */
	-
	-
	-static int physfsrwops_write(SDL_RWops rw, const void ptr, int size, int num)
	-{
	- PHYSFS_file handle = (PHYSFS_file ) rw->hidden.unknown.data1;
	- PHYSFS_sint64 rc = PHYSFS_write(handle, ptr, size, num);
	- if (rc != num)
	- SDL_SetError("PhysicsFS error: %s", PHYSFS_getLastError());
	-
	- return((int) rc);
	-} /* physfsrwops_write */
	-
	-
	-static int physfsrwops_close(SDL_RWops *rw)
	-{
	- PHYSFS_file handle = (PHYSFS_file ) rw->hidden.unknown.data1;
	- if (!PHYSFS_close(handle))
	- {
	- SDL_SetError("PhysicsFS error: %s", PHYSFS_getLastError());
	- return(-1);
	- } /* if */
	-
	- SDL_FreeRW(rw);
	- return(0);
	-} /* physfsrwops_close */
	-
	-
	-static SDL_RWops create_rwops(PHYSFS_file handle)
	-{
	- SDL_RWops *retval = NULL;
	-
	- if (handle == NULL)
	- SDL_SetError("PhysicsFS error: %s", PHYSFS_getLastError());
	- else
	- {
	- retval = SDL_AllocRW();
	- if (retval != NULL)
	- {
	- retval->seek = physfsrwops_seek;
	- retval->read = physfsrwops_read;
	- retval->write = physfsrwops_write;
	- retval->close = physfsrwops_close;
	- retval->hidden.unknown.data1 = handle;
	- } /* if */
	- } /* else */
	-
	- return(retval);
	-} /* create_rwops */
	-
	-
	-SDL_RWops PHYSFSRWOPS_makeRWops(PHYSFS_file handle)
	-{
	- SDL_RWops *retval = NULL;
	- if (handle == NULL)
	- SDL_SetError("NULL pointer passed to PHYSFSRWOPS_makeRWops().");
	- else
	- retval = create_rwops(handle);
	-
	- return(retval);
	-} /* PHYSFSRWOPS_makeRWops */
	-
	-
	-SDL_RWops PHYSFSRWOPS_openRead(const char fname)
	-{
	- return(create_rwops(PHYSFS_openRead(fname)));
	-} /* PHYSFSRWOPS_openRead */
	-
	-
	-SDL_RWops PHYSFSRWOPS_openWrite(const char fname)
	-{
	- return(create_rwops(PHYSFS_openWrite(fname)));
	-} /* PHYSFSRWOPS_openWrite */
	-
	-
	-SDL_RWops PHYSFSRWOPS_openAppend(const char fname)
	-{
	- return(create_rwops(PHYSFS_openAppend(fname)));
	-} /* PHYSFSRWOPS_openAppend */
	-
	-#endif
	-
	-/* end of physfsrwops.c ... */
	-
	diff --git a/util/sdl/sound/playsound/physfsrwops.h b/util/sdl/sound/playsound/physfsrwops.h
	deleted file mode 100644
	index 5ff519a1..00000000
	--- a/util/sdl/sound/playsound/physfsrwops.h
	+++ /dev/null
	@@ -1,87 +0,0 @@
	-/*
	- * This code provides a glue layer between PhysicsFS and Simple Directmedia
	- * Layer's (SDL) RWops i/o abstraction.
	- *
	- * License: this code is public domain. I make no warranty that it is useful,
	- * correct, harmless, or environmentally safe.
	- *
	- * This particular file may be used however you like, including copying it
	- * verbatim into a closed-source project, exploiting it commercially, and
	- * removing any trace of my name from the source (although I hope you won't
	- * do that). I welcome enhancements and corrections to this file, but I do
	- * not require you to send me patches if you make changes.
	- *
	- * Unless otherwise stated, the rest of PhysicsFS falls under the GNU Lesser
	- * General Public License: http://www.gnu.org/licenses/lgpl.txt
	- *
	- * SDL falls under the LGPL, too. You can get SDL at http://www.libsdl.org/
	- *
	- * This file was written by Ryan C. Gordon. (icculus@icculus.org).
	- */
	-
	-#ifndef _INCLUDE_PHYSFSRWOPS_H_
	-#define _INCLUDE_PHYSFSRWOPS_H_
	-
	-#include "physfs.h"
	-#include "SDL.h"
	-
	-#ifdef __cplusplus
	-extern "C" {
	-#endif
	-
	-/**
	- * Open a platform-independent filename for reading, and make it accessible
	- * via an SDL_RWops structure. The file will be closed in PhysicsFS when the
	- * RWops is closed. PhysicsFS should be configured to your liking before
	- * opening files through this method.
	- *
	- * @param filename File to open in platform-independent notation.
	- * @return A valid SDL_RWops structure on success, NULL on error. Specifics
	- * of the error can be gleaned from PHYSFS_getLastError().
	- */
	-__EXPORT__ SDL_RWops PHYSFSRWOPS_openRead(const char fname);
	-
	-/**
	- * Open a platform-independent filename for writing, and make it accessible
	- * via an SDL_RWops structure. The file will be closed in PhysicsFS when the
	- * RWops is closed. PhysicsFS should be configured to your liking before
	- * opening files through this method.
	- *
	- * @param filename File to open in platform-independent notation.
	- * @return A valid SDL_RWops structure on success, NULL on error. Specifics
	- * of the error can be gleaned from PHYSFS_getLastError().
	- */
	-__EXPORT__ SDL_RWops PHYSFSRWOPS_openWrite(const char fname);
	-
	-/**
	- * Open a platform-independent filename for appending, and make it accessible
	- * via an SDL_RWops structure. The file will be closed in PhysicsFS when the
	- * RWops is closed. PhysicsFS should be configured to your liking before
	- * opening files through this method.
	- *
	- * @param filename File to open in platform-independent notation.
	- * @return A valid SDL_RWops structure on success, NULL on error. Specifics
	- * of the error can be gleaned from PHYSFS_getLastError().
	- */
	-__EXPORT__ SDL_RWops PHYSFSRWOPS_openAppend(const char fname);
	-
	-/**
	- * Make a SDL_RWops from an existing PhysicsFS file handle. You should
	- * dispose of any references to the handle after successful creation of
	- * the RWops. The actual PhysicsFS handle will be destroyed when the
	- * RWops is closed.
	- *
	- * @param handle a valid PhysicsFS file handle.
	- * @return A valid SDL_RWops structure on success, NULL on error. Specifics
	- * of the error can be gleaned from PHYSFS_getLastError().
	- */
	-__EXPORT__ SDL_RWops PHYSFSRWOPS_makeRWops(PHYSFS_file handle);
	-
	-#ifdef __cplusplus
	-}
	-#endif
	-
	-#endif /* include-once blocker */
	-
	-/* end of physfsrwops.h ... */
	-
	diff --git a/util/sdl/sound/playsound/playsound.c b/util/sdl/sound/playsound/playsound.c
	deleted file mode 100644
	index d160e053..00000000
	--- a/util/sdl/sound/playsound/playsound.c
	+++ /dev/null
	@@ -1,1062 +0,0 @@
	-/*
	- * SDL_sound -- An abstract sound format decoding API.
	- * Copyright (C) 2001 Ryan C. Gordon.
	- *
	- * This library is free software; you can redistribute it and/or
	- * modify it under the terms of the GNU Lesser General Public
	- * License as published by the Free Software Foundation; either
	- * version 2.1 of the License, or (at your option) any later version.
	- *
	- * This library is distributed in the hope that it will be useful,
	- * but WITHOUT ANY WARRANTY; without even the implied warranty of
	- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	- * Lesser General Public License for more details.
	- *
	- * You should have received a copy of the GNU Lesser General Public
	- * License along with this library; if not, write to the Free Software
	- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
	- */
	-
	-/**
	- * This is a quick and dirty test of SDL_sound.
	- *
	- * Please see the file COPYING in the source's root directory.
	- *
	- * This file written by Ryan C. Gordon. (icculus@icculus.org)
	- */
	-
	-#if HAVE_CONFIG_H
	-# include <config.h>
	-#endif
	-
	-#include <stdio.h>
	-#include <stdlib.h>
	-#include <string.h>
	-
	-#if HAVE_ASSERT_H
	-# include <assert.h>
	-#elif (!defined assert)
	-# define assert(x)
	-#endif
	-
	-#if HAVE_SIGNAL_H
	-# include <signal.h>
	-#endif
	-
	-#include "SDL.h"
	-#include "SDL_sound.h"
	-
	-#if SUPPORT_PHYSFS
	-#include "physfs.h"
	-#include "physfsrwops.h"
	-#endif
	-
	-#define DEFAULT_DECODEBUF 16384
	-#define DEFAULT_AUDIOBUF 4096
	-
	-#define PLAYSOUND_VER_MAJOR 0
	-#define PLAYSOUND_VER_MINOR 1
	-#define PLAYSOUND_VER_PATCH 5
	-
	-
	-static const char *option_list[] =
	-{
	- "--rate", "n Playback at sample rate of n HZ.",
	- "--format", "fmt Playback in fmt format (see below).",
	- "--channels", "n Playback on n channels (1 or 2).",
	- "--decodebuf", "n Buffer n decoded bytes at a time (default 16384).",
	- "--audiobuf", "n Buffer n samples to audio device (default 4096).",
	- "--volume", "n Playback volume multiplier (default 1.0).",
	- "--stdin", "[ext] Read from stdin (treat data as format [ext])",
	- "--version", " Display version information and exit.",
	- "--decoders", " List supported data formats and exit.",
	- "--predecode", " Decode entire sample before playback.",
	- "--loop", "n Loop playback n times.",
	- "--seek", "list List of seek points and playback durations.",
	- "--credits", " Shameless promotion.",
	- "--help", " Display this information and exit.",
	- NULL, NULL
	-};
	-
	-
	-static void output_versions(const char *argv0)
	-{
	- Sound_Version compiled;
	- Sound_Version linked;
	- SDL_version sdl_compiled;
	- const SDL_version *sdl_linked;
	-
	- SOUND_VERSION(&compiled);
	- Sound_GetLinkedVersion(&linked);
	- SDL_VERSION(&sdl_compiled);
	- sdl_linked = SDL_Linked_Version();
	-
	- fprintf(stdout,
	- "%s version %d.%d.%d\n"
	- "Copyright 2001 Ryan C. Gordon\n"
	- "This program is free software, covered by the GNU Lesser General\n"
	- "Public License, and you are welcome to change it and/or\n"
	- "distribute copies of it under certain conditions. There is\n"
	- "absolutely NO WARRANTY for this program.\n"
	- "\n"
	- " Compiled against SDL_sound version %d.%d.%d,\n"
	- " and linked against %d.%d.%d.\n"
	- " Compiled against SDL version %d.%d.%d,\n"
	- " and linked against %d.%d.%d.\n\n",
	- argv0,
	- PLAYSOUND_VER_MAJOR, PLAYSOUND_VER_MINOR, PLAYSOUND_VER_PATCH,
	- compiled.major, compiled.minor, compiled.patch,
	- linked.major, linked.minor, linked.patch,
	- sdl_compiled.major, sdl_compiled.minor, sdl_compiled.patch,
	- sdl_linked->major, sdl_linked->minor, sdl_linked->patch);
	-} /* output_versions */
	-
	-
	-static void output_decoders(void)
	-{
	- const Sound_DecoderInfo **rc = Sound_AvailableDecoders();
	- const Sound_DecoderInfo **i;
	- const char **ext;
	-
	- fprintf(stdout, "Supported sound formats:\n");
	- if (rc == NULL)
	- fprintf(stdout, " * Apparently, NONE!\n");
	- else
	- {
	- for (i = rc; *i != NULL; i++)
	- {
	- fprintf(stdout, " * %s\n", (*i)->description);
	-
	- for (ext = (i)->extensions; ext != NULL; ext++)
	- fprintf(stdout, " File extension \"%s\"\n", *ext);
	-
	- fprintf(stdout, " Written by %s.\n %s\n\n",
	- (i)->author, (i)->url);
	- } /* for */
	- } /* else */
	-
	- fprintf(stdout, "\n");
	-} /* output_decoders */
	-
	-
	-static void output_usage(const char *argv0)
	-{
	- const char **i = option_list;
	-
	- fprintf(stderr,
	- "USAGE: %s [...options...] [soundFile1] ... [soundFileN]\n"
	- "\n"
	- " Options:\n",
	- argv0);
	-
	- while (*i != NULL)
	- {
	- const char option = (i++);
	- const char optiondesc = (i++);
	- fprintf(stderr, " %s %s\n", option, optiondesc);
	- } /* while */
	-
	- fprintf(stderr,
	- "\n"
	- " Valid arguments to the --format option are:\n"
	- " U8 Unsigned 8-bit.\n"
	- " S8 Signed 8-bit.\n"
	- " U16LSB Unsigned 16-bit (least significant byte first).\n"
	- " U16MSB Unsigned 16-bit (most significant byte first).\n"
	- " S16LSB Signed 16-bit (least significant byte first).\n"
	- " S16MSB Signed 16-bit (most significant byte first).\n"
	- "\n"
	- " Valid arguments to the --seek options look like:\n"
	- " --seek \"mm:SS:ss;mm:SS:ss;mm:SS:ss\"\n"
	- " Where the first \"mm:SS:ss\" is the position, in minutes,\n"
	- " seconds and milliseconds to seek to at start of playback. The\n"
	- " next mm:SS:ss is how long to play audio from that point.\n"
	- " The third mm:SS:ss is another seek after the duration of\n"
	- " playback has completed. If the final playback duration is\n"
	- " omitted, playback continues until the end of the file.\n"
	- " The \"mm\" and \"SS\" portions may be omitted. --loop\n"
	- " and --seek can coexist.\n"
	- "\n");
	-} /* output_usage */
	-
	-
	-static void output_credits(void)
	-{
	- fprintf(stdout,
	- "playsound version %d.%d.%d\n"
	- "Copyright 2001 Ryan C. Gordon\n"
	- "playsound is free software, covered by the GNU Lesser General\n"
	- "Public License, and you are welcome to change it and/or\n"
	- "distribute copies of it under certain conditions. There is\n"
	- "absolutely NO WARRANTY for playsound.\n"
	- "\n"
	- " Written by Ryan C. Gordon, Torbjörn Andersson, Max Horn,\n"
	- " Tsuyoshi Iguchi, Tyler Montbriand, Darrell Walisser,\n"
	- " and a cast of thousands.\n"
	- "\n"
	- " Website and source code: http://icculus.org/SDL_sound/\n"
	- "\n",
	- PLAYSOUND_VER_MAJOR, PLAYSOUND_VER_MINOR, PLAYSOUND_VER_PATCH);
	-} /* output_credits */
	-
	-
	-
	-/* archive stuff... */
	-
	-static int init_archive(const char *argv0)
	-{
	- int retval = 1;
	-
	-#if SUPPORT_PHYSFS
	- retval = PHYSFS_init(argv0);
	- if (!retval)
	- {
	- fprintf(stderr, "Couldn't init PhysicsFS: %s\n",
	- PHYSFS_getLastError());
	- } /* if */
	-#endif
	-
	- return(retval);
	-} /* init_archive */
	-
	-
	-#if SUPPORT_PHYSFS
	-static SDL_RWops rwops_from_physfs(const char filename)
	-{
	- SDL_RWops *retval = NULL;
	-
	- char path = (char ) malloc(strlen(filename) + 1);
	- char *archive;
	-
	- if (path == NULL)
	- {
	- fprintf(stderr, "Out of memory!\n");
	- return(NULL);
	- } /* if */
	-
	- strcpy(path, filename);
	- archive = strchr(path, '@');
	- if (archive != NULL)
	- {
	- (archive++) = '\0'; / blank '@', point to archive name. */
	- if (!PHYSFS_addToSearchPath(archive, 0))
	- {
	- fprintf(stderr, "Couldn't open archive: %s\n",
	- PHYSFS_getLastError());
	- free(path);
	- return(NULL);
	- } /* if */
	-
	- retval = PHYSFSRWOPS_openRead(path);
	- } /* if */
	-
	- free(path);
	- return(retval);
	-} /* rwops_from_physfs */
	-#endif
	-
	-
	-static Sound_Sample sample_from_archive(const char fname,
	- Sound_AudioInfo *desired,
	- Uint32 decode_buffersize)
	-{
	- Sound_Sample *retval = NULL;
	-
	-#if SUPPORT_PHYSFS
	- SDL_RWops *rw = rwops_from_physfs(fname);
	- if (rw != NULL)
	- {
	- char path = (char ) malloc(strlen(fname) + 1);
	- char *ptr;
	- strcpy(path, fname);
	- ptr = strchr(path, '@');
	- *ptr = '\0';
	- ptr = strrchr(path, '.');
	- if (ptr != NULL)
	- ptr++;
	-
	- retval = Sound_NewSample(rw, ptr, desired, decode_buffersize);
	- free(path);
	- } /* if */
	-#endif
	-
	- return(retval);
	-} /* sample_from_archive */
	-
	-
	-static void close_archive(const char *filename)
	-{
	-#if SUPPORT_PHYSFS
	- char *archive_name = strchr(filename, '@');
	- if (archive_name != NULL)
	- PHYSFS_removeFromSearchPath(archive_name + 1);
	-#endif
	-} /* close_archive */
	-
	-
	-static void deinit_archive(void)
	-{
	-#if SUPPORT_PHYSFS
	- PHYSFS_deinit();
	-#endif
	-} /* deinit_archive */
	-
	-
	-
	-static volatile int done_flag = 0;
	-
	-#if HAVE_SIGNAL_H
	-void sigint_catcher(int signum)
	-{
	- static Uint32 last_sigint = 0;
	- Uint32 ticks = SDL_GetTicks();
	-
	- assert(signum == SIGINT);
	-
	- if ((last_sigint != 0) && (ticks - last_sigint < 500))
	- {
	- SDL_PauseAudio(1);
	- SDL_CloseAudio();
	- Sound_Quit();
	- SDL_Quit();
	- deinit_archive();
	- exit(1);
	- } /* if */
	-
	- else
	- {
	- last_sigint = ticks;
	- done_flag = 1;
	- } /* else */
	-} /* sigint_catcher */
	-#endif
	-
	-
	-/* global decoding state. */
	-typedef struct
	-{
	- Uint8 *decoded_ptr;
	- Uint32 decoded_bytes;
	- int predecode;
	- int looping;
	- int wants_volume_change;
	- float volume;
	- Uint32 total_seeks;
	- Uint32 *seek_list;
	- Uint32 seek_index;
	- Sint32 bytes_before_next_seek;
	-} playsound_global_state;
	-
	-static volatile playsound_global_state global_state;
	-
	-
	-static Uint32 cvtMsToBytePos(Sound_AudioInfo *info, Uint32 ms)
	-{
	- /* "frames" == "sample frames" */
	- float frames_per_ms = ((float) info->rate) / 1000.0;
	- Uint32 frame_offset = (Uint32) (frames_per_ms * ((float) ms));
	- Uint32 frame_size = (Uint32) ((info->format & 0xFF) / 8) * info->channels;
	- return(frame_offset * frame_size);
	-} /* cvtMsToBytePos */
	-
	-
	-static void do_seek(Sound_Sample *sample)
	-{
	- Uint32 *seek_list = global_state.seek_list;
	- Uint32 seek_index = global_state.seek_index;
	- Uint32 total_seeks = global_state.total_seeks;
	-
	- fprintf(stdout, "Seeking to %.2d:%.2d:%.4d...\n",
	- (int) ((seek_list[seek_index] / 1000) / 60),
	- (int) ((seek_list[seek_index] / 1000) % 60),
	- (int) ((seek_list[seek_index] % 1000)));
	-
	- if (global_state.predecode)
	- {
	- Uint32 pos = cvtMsToBytePos(&sample->desired, seek_list[seek_index]);
	- if (pos > sample->buffer_size)
	- {
	- fprintf(stderr, "Seek past end of predecoded buffer.\n");
	- done_flag = 1;
	- } /* if */
	- else
	- {
	- global_state.decoded_ptr = (((Uint8 *) sample->buffer) + pos);
	- global_state.decoded_bytes = sample->buffer_size - pos;
	- } /* else */
	- } /* if */
	- else
	- {
	- if (!Sound_Seek(sample, seek_list[seek_index]))
	- {
	- fprintf(stderr, "Sound_Seek() failed: %s\n", Sound_GetError());
	- done_flag = 1;
	- } /* if */
	- } /* else */
	-
	- seek_index++;
	- if (seek_index >= total_seeks)
	- global_state.bytes_before_next_seek = -1; /* no more seeks. */
	- else
	- {
	- global_state.bytes_before_next_seek = cvtMsToBytePos(&sample->desired,
	- seek_list[seek_index]);
	- seek_index++;
	- } /* else */
	-
	- global_state.seek_index = seek_index;
	-} /* do_seek */
	-
	-
	-/*
	- * This updates (decoded_bytes) and (decoded_ptr) with more audio data,
	- * taking into account potential looping, seeking and predecoding.
	- */
	-static int read_more_data(Sound_Sample *sample)
	-{
	- if (done_flag) /* probably a sigint; stop trying to read. */
	- {
	- global_state.decoded_bytes = 0;
	- return(0);
	- } /* if */
	-
	- if ((global_state.bytes_before_next_seek >= 0) &&
	- (global_state.decoded_bytes > global_state.bytes_before_next_seek))
	- {
	- global_state.decoded_bytes = global_state.bytes_before_next_seek;
	- } /* if */
	-
	- if (global_state.decoded_bytes > 0) /* don't need more data; just return. */
	- return(global_state.decoded_bytes);
	-
	- /* Need more audio data. See if we're supposed to seek... */
	- if ((global_state.bytes_before_next_seek == 0) &&
	- (global_state.seek_index < global_state.total_seeks))
	- {
	- do_seek(sample); /* do it, baby! */
	- return(read_more_data(sample)); /* handle loops conditions. */
	- } /* if */
	-
	- /* See if there's more to be read... */
	- if ( (global_state.bytes_before_next_seek != 0) &&
	- (!(sample->flags & (SOUND_SAMPLEFLAG_ERROR \| SOUND_SAMPLEFLAG_EOF))) )
	- {
	- global_state.decoded_bytes = Sound_Decode(sample);
	- if (sample->flags & SOUND_SAMPLEFLAG_ERROR)
	- {
	- fprintf(stderr, "Error in decoding sound file!\n"
	- " reason: [%s].\n", Sound_GetError());
	- } /* if */
	-
	- global_state.decoded_ptr = sample->buffer;
	- return(read_more_data(sample)); /* handle loops conditions. */
	- } /* if */
	-
	- /* No more to be read from stream, but we may want to loop the sample. */
	-
	- if (!global_state.looping)
	- return(0);
	-
	- global_state.looping--;
	-
	- global_state.seek_index = 0;
	- global_state.bytes_before_next_seek =
	- (global_state.total_seeks > 0) ? 0 : -1;
	-
	- /* we just need to point predecoded samples to the start of the buffer. */
	- if (global_state.predecode)
	- {
	- global_state.decoded_bytes = sample->buffer_size;
	- global_state.decoded_ptr = sample->buffer;
	- } /* if */
	- else
	- {
	- Sound_Rewind(sample); /* error is checked in recursion. */
	- } /* else */
	-
	- return(read_more_data(sample));
	-} /* read_more_data */
	-
	-
	-static void memcpy_with_volume(Sound_Sample *sample,
	- Uint8 dst, Uint8 src, int len)
	-{
	- int i;
	- Uint16 *u16src = NULL;
	- Uint16 *u16dst = NULL;
	- Sint16 *s16src = NULL;
	- Sint16 *s16dst = NULL;
	- float volume = global_state.volume;
	-
	- if (!global_state.wants_volume_change)
	- {
	- memcpy(dst, src, len);
	- return;
	- } /* if */
	-
	- /* !!! FIXME: This would be more efficient with a lookup table. */
	- switch (sample->desired.format)
	- {
	- case AUDIO_U8:
	- for (i = 0; i < len; i++, src++, dst++)
	- dst = (Uint8) (((float) (src)) * volume);
	- break;
	-
	- case AUDIO_S8:
	- for (i = 0; i < len; i++, src++, dst++)
	- dst = (Sint8) (((float) (src)) * volume);
	- break;
	-
	- case AUDIO_U16LSB:
	- u16src = (Uint16 *) src;
	- u16dst = (Uint16 *) dst;
	- for (i = 0; i < len; i += sizeof (Uint16), u16src++, u16dst++)
	- {
	- u16dst = (Uint16) (((float) (SDL_SwapLE16(u16src))) * volume);
	- u16dst = SDL_SwapLE16(u16dst);
	- } /* for */
	- break;
	-
	- case AUDIO_S16LSB:
	- s16src = (Sint16 *) src;
	- s16dst = (Sint16 *) dst;
	- for (i = 0; i < len; i += sizeof (Sint16), s16src++, s16dst++)
	- {
	- s16dst = (Sint16) (((float) (SDL_SwapLE16(s16src))) * volume);
	- s16dst = SDL_SwapLE16(s16dst);
	- } /* for */
	- break;
	-
	- case AUDIO_U16MSB:
	- u16src = (Uint16 *) src;
	- u16dst = (Uint16 *) dst;
	- for (i = 0; i < len; i += sizeof (Uint16), u16src++, u16dst++)
	- {
	- u16dst = (Uint16) (((float) (SDL_SwapBE16(u16src))) * volume);
	- u16dst = SDL_SwapBE16(u16dst);
	- } /* for */
	- break;
	-
	- case AUDIO_S16MSB:
	- s16src = (Sint16 *) src;
	- s16dst = (Sint16 *) dst;
	- for (i = 0; i < len; i += sizeof (Sint16), s16src++, s16dst++)
	- {
	- s16dst = (Sint16) (((float) (SDL_SwapBE16(s16src))) * volume);
	- s16dst = SDL_SwapBE16(s16dst);
	- } /* for */
	- break;
	- } /* switch */
	-} /* memcpy_with_volume */
	-
	-
	-static void audio_callback(void userdata, Uint8 stream, int len)
	-{
	- Sound_Sample sample = (Sound_Sample ) userdata;
	- int bw = 0; /* bytes written to stream this time through the callback */
	-
	- while (bw < len)
	- {
	- int cpysize; /* bytes to copy on this iteration of the loop. */
	-
	- if (!read_more_data(sample)) /* read more data, if needed. */
	- {
	- /* ...there isn't any more data to read! */
	- memset(stream + bw, '\0', len - bw);
	- done_flag = 1;
	- return;
	- } /* if */
	-
	- /* decoded_bytes and decoder_ptr are updated as necessary... */
	-
	- cpysize = len - bw;
	- if (cpysize > global_state.decoded_bytes)
	- cpysize = global_state.decoded_bytes;
	-
	- if (cpysize > 0)
	- {
	- memcpy_with_volume(sample, stream + bw,
	- (Uint8 *) global_state.decoded_ptr,
	- cpysize);
	-
	- bw += cpysize;
	- global_state.decoded_ptr += cpysize;
	- global_state.decoded_bytes -= cpysize;
	- if (global_state.bytes_before_next_seek >= 0)
	- global_state.bytes_before_next_seek -= cpysize;
	- } /* if */
	- } /* while */
	-} /* audio_callback */
	-
	-
	-static int count_seek_list(const char *list)
	-{
	- const char *ptr;
	- int retval = 0;
	-
	- for (ptr = list; ptr != NULL; ptr = strchr(ptr + 1, ';'))
	- retval++;
	-
	- return(retval);
	-} /* count_seek_list */
	-
	-
	-static Uint32 parse_time_str(char *str)
	-{
	- Uint32 minutes = 0;
	- Uint32 seconds = 0;
	- Uint32 ms = 0;
	-
	- char *ptr = strchr(str, ':');
	- if (ptr != NULL)
	- {
	- char *ptr2;
	-
	- *ptr = '\0';
	- ptr2 = strchr(ptr + 1, ':');
	- if (ptr2 != NULL)
	- {
	- *ptr2 = '\0';
	- minutes = atoi(str);
	- str = ptr + 1;
	- ptr = ptr2;
	- } /* if */
	-
	- seconds = atoi(str);
	- str = ptr + 1;
	- } /* if */
	-
	- ms = atoi(str);
	- return( (((minutes * 60) + seconds) * 1000) + ms );
	-} /* parse_time_str */
	-
	-
	-static void parse_seek_list(const char *_list)
	-{
	- Uint32 i;
	-
	- char list = (char) malloc(strlen(_list) + 1);
	- char *save_list = list;
	- if (list == NULL)
	- {
	- fprintf(stderr, "malloc() failed. Skipping seek list.\n");
	- return;
	- } /* if */
	-
	- strcpy(list, _list);
	-
	- if (global_state.seek_list != NULL)
	- free((void *) global_state.seek_list);
	-
	- global_state.total_seeks = count_seek_list(list);
	-
	- global_state.seek_list =
	- (Uint32 ) malloc(global_state.total_seeks sizeof (Uint32));
	-
	- if (global_state.seek_list == NULL)
	- {
	- fprintf(stderr, "malloc() failed. Skipping seek list.\n");
	- global_state.total_seeks = 0;
	- return;
	- } /* if */
	-
	- for (i = 0; i < global_state.total_seeks; i++)
	- {
	- char *ptr = strchr(list, ';');
	- if (ptr != NULL)
	- *ptr = '\0';
	- global_state.seek_list[i] = parse_time_str(list);
	- list = ptr + 1;
	- } /* for */
	-
	- global_state.bytes_before_next_seek = 0;
	-
	- free(save_list);
	-} /* parse_seek_list */
	-
	-
	-static int str_to_fmt(char *str)
	-{
	- if (strcmp(str, "U8") == 0)
	- return AUDIO_U8;
	- if (strcmp(str, "S8") == 0)
	- return AUDIO_S8;
	- if (strcmp(str, "U16LSB") == 0)
	- return AUDIO_U16LSB;
	- if (strcmp(str, "S16LSB") == 0)
	- return AUDIO_S16LSB;
	- if (strcmp(str, "U16MSB") == 0)
	- return AUDIO_U16MSB;
	- if (strcmp(str, "S16MSB") == 0)
	- return AUDIO_S16MSB;
	- return 0;
	-} /* str_to_fmt */
	-
	-
	-static int valid_cmdline(int argc, char **argv)
	-{
	- int i;
	-
	- if (argc < 2) /* no command line? Show help text and quit. */
	- {
	- output_usage(argv[0]);
	- return(0);
	- } /* if */
	-
	- /* Make sure all command line options are valid. */
	- for (i = 1; i < argc; i++)
	- {
	- const char **opts = option_list;
	-
	- if (strncmp(argv[i], "--", 2) != 0) /* not an option; skip it. */
	- continue;
	-
	- while (*opts != NULL)
	- {
	- if (strcmp(argv[i], *(opts++)) == 0)
	- break;
	-
	- opts++; /* skip option description. */
	- } /* else */
	-
	- if (opts == NULL) / didn't find it in option_list... */
	- {
	- fprintf(stderr, "unknown option: \"%s\"\n", argv[i]);
	- return(0);
	- } /* if */
	- } /* for */
	-
	- return(1); /* everything appears to be in order. */
	-} /* valid_cmdline */
	-
	-
	-int main(int argc, char **argv)
	-{
	- Sound_AudioInfo sound_desired;
	- SDL_AudioSpec sdl_desired;
	- Uint32 audio_buffersize;
	- Uint32 decode_buffersize;
	- Sound_Sample *sample;
	- int use_specific_audiofmt = 0;
	- int i;
	- int delay;
	- int new_sample = 1;
	- Uint32 sdl_init_flags = SDL_INIT_AUDIO;
	-
	- #if ENABLE_EVENTS
	- SDL_Surface *screen = NULL;
	- SDL_Event event;
	-
	- sdl_init_flags \|= SDL_INIT_VIDEO;
	- #endif
	-
	- #ifdef HAVE_SETBUF
	- setbuf(stdout, NULL);
	- setbuf(stderr, NULL);
	- #endif
	-
	- if (!valid_cmdline(argc, argv))
	- return(42);
	-
	- /* Handle some command lines upfront. */
	- for (i = 0; i < argc; i++)
	- {
	- if (strncmp(argv[i], "--", 2) != 0)
	- continue;
	-
	- if (strcmp(argv[i], "--version") == 0)
	- {
	- output_versions(argv[0]);
	- return(42);
	- } /* if */
	-
	- if (strcmp(argv[i], "--credits") == 0)
	- {
	- output_credits();
	- return(42);
	- } /* if */
	-
	- else if (strcmp(argv[i], "--help") == 0)
	- {
	- output_usage(argv[0]);
	- return(42);
	- } /* if */
	-
	- else if (strcmp(argv[i], "--decoders") == 0)
	- {
	- if (!Sound_Init())
	- {
	- fprintf(stderr, "Sound_Init() failed!\n"
	- " reason: [%s].\n", Sound_GetError());
	- SDL_Quit();
	- return(42);
	- } /* if */
	-
	- output_decoders();
	- Sound_Quit();
	- return(0);
	- } /* else if */
	- } /* for */
	-
	- if (!init_archive(argv[0]))
	- return(42);
	-
	- if (SDL_Init(sdl_init_flags) == -1)
	- {
	- fprintf(stderr, "SDL_Init() failed!\n"
	- " reason: [%s].\n", SDL_GetError());
	- return(42);
	- } /* if */
	-
	- if (!Sound_Init())
	- {
	- fprintf(stderr, "Sound_Init() failed!\n"
	- " reason: [%s].\n", Sound_GetError());
	- SDL_Quit();
	- return(42);
	- } /* if */
	-
	- #if HAVE_SIGNAL_H
	- signal(SIGINT, sigint_catcher);
	- #endif
	-
	- #if ENABLE_EVENTS
	- screen = SDL_SetVideoMode(320, 240, 8, 0);
	- assert(screen != NULL);
	- #endif
	-
	- for (i = 1; i < argc; i++)
	- {
	- char *filename = NULL;
	-
	- /* set variables back to defaults for next file... */
	- if (new_sample)
	- {
	- if (global_state.seek_list != NULL)
	- free((void *) global_state.seek_list);
	-
	- memset((void *) &global_state, '\0', sizeof (global_state));
	- memset(&sdl_desired, '\0', sizeof (SDL_AudioSpec));
	- global_state.volume = 1.0;
	- global_state.bytes_before_next_seek = -1;
	- audio_buffersize = DEFAULT_AUDIOBUF;
	- decode_buffersize = DEFAULT_DECODEBUF;
	- new_sample = 0;
	- } /* if */
	-
	- if (strcmp(argv[i], "--rate") == 0 && argc > i + 1)
	- {
	- use_specific_audiofmt = 1;
	- sound_desired.rate = atoi(argv[++i]);
	- if (sound_desired.rate <= 0)
	- {
	- fprintf(stderr, "Bad argument to --rate!\n");
	- return(42);
	- } /* if */
	- } /* else if */
	-
	- else if (strcmp(argv[i], "--format") == 0 && argc > i + 1)
	- {
	- use_specific_audiofmt = 1;
	- sound_desired.format = str_to_fmt(argv[++i]);
	- if (sound_desired.format == 0)
	- {
	- fprintf(stderr, "Bad argument to --format! Try one of:\n"
	- "U8, S8, U16LSB, S16LSB, U16MSB, S16MSB\n");
	- return(42);
	- } /* if */
	- } /* else if */
	-
	- else if (strcmp(argv[i], "--channels") == 0 && argc > i + 1)
	- {
	- use_specific_audiofmt = 1;
	- sound_desired.channels = atoi(argv[++i]);
	- if (sound_desired.channels < 1 \|\| sound_desired.channels > 2)
	- {
	- fprintf(stderr,
	- "Bad argument to --channels! Try 1 (mono) or 2 "
	- "(stereo).\n");
	- return(42);
	- } /* if */
	- } /* else if */
	-
	- else if (strcmp(argv[i], "--audiobuf") == 0 && argc > i + 1)
	- {
	- audio_buffersize = atoi(argv[++i]);
	- } /* else if */
	-
	- else if (strcmp(argv[i], "--decodebuf") == 0 && argc > i + 1)
	- {
	- decode_buffersize = atoi(argv[++i]);
	- } /* else if */
	-
	- else if (strcmp(argv[i], "--volume") == 0 && argc > i + 1)
	- {
	- global_state.volume = atof(argv[++i]);
	- if (global_state.volume != 1.0)
	- global_state.wants_volume_change = 1;
	- } /* else if */
	-
	- else if (strcmp(argv[i], "--predecode") == 0)
	- {
	- global_state.predecode = 1;
	- } /* else if */
	-
	- else if (strcmp(argv[i], "--loop") == 0)
	- {
	- global_state.looping = atoi(argv[++i]);
	- } /* else if */
	-
	- else if (strcmp(argv[i], "--seek") == 0)
	- {
	- parse_seek_list(argv[++i]);
	- } /* else if */
	-
	- else if (strcmp(argv[i], "--stdin") == 0)
	- {
	- SDL_RWops *rw = SDL_RWFromFP(stdin, 1);
	- filename = "...from stdin...";
	-
	- /*
	- * The second argument will be NULL if --stdin is the last
	- * thing on the command line. This is correct behaviour.
	- */
	- sample = Sound_NewSample(rw, argv[++i],
	- use_specific_audiofmt ? &sound_desired : NULL,
	- decode_buffersize);
	- } /* if */
	-
	- else if (strncmp(argv[i], "--", 2) == 0)
	- {
	- /* ignore it, since it was handled at startup. */
	- } /* else if */
	-
	- else
	- {
	- filename = argv[i];
	- sample = sample_from_archive(filename,
	- use_specific_audiofmt ? &sound_desired : NULL,
	- decode_buffersize);
	-
	- if (sample == NULL)
	- {
	- sample = Sound_NewSampleFromFile(filename,
	- use_specific_audiofmt ? &sound_desired : NULL,
	- decode_buffersize);
	- } /* if */
	- } /* else */
	-
	- if (filename == NULL) /* still parsing command line stuff? */
	- continue;
	-
	- new_sample = 1;
	-
	- if (sample == NULL)
	- {
	- fprintf(stderr, "Couldn't load \"%s\"!\n"
	- " reason: [%s].\n",
	- filename, Sound_GetError());
	- continue;
	- } /* if */
	-
	- if (global_state.total_seeks > 0)
	- {
	- if ((!global_state.predecode) &&
	- (!(sample->flags & SOUND_SAMPLEFLAG_CANSEEK)))
	- {
	- fprintf(stderr, "Want seeks, but sample cannot handle it!\n");
	- Sound_FreeSample(sample);
	- close_archive(filename);
	- continue;
	- } /* if */
	- } /* if */
	-
	- /*
	- * Unless explicitly specified, pick the format from the sound
	- * to be played.
	- */
	- if (use_specific_audiofmt)
	- {
	- sdl_desired.freq = sample->desired.rate;
	- sdl_desired.format = sample->desired.format;
	- sdl_desired.channels = sample->desired.channels;
	- } /* if */
	- else
	- {
	- sdl_desired.freq = sample->actual.rate;
	- sdl_desired.format = sample->actual.format;
	- sdl_desired.channels = sample->actual.channels;
	- } /* else */
	-
	- sdl_desired.samples = audio_buffersize;
	- sdl_desired.callback = audio_callback;
	- sdl_desired.userdata = sample;
	-
	- if (SDL_OpenAudio(&sdl_desired, NULL) < 0)
	- {
	- fprintf(stderr, "Couldn't open audio device!\n"
	- " reason: [%s].\n", SDL_GetError());
	- Sound_Quit();
	- SDL_Quit();
	- return(42);
	- } /* if */
	-
	- fprintf(stdout, "Now playing [%s]...\n", filename);
	-
	- if (global_state.predecode)
	- {
	- fprintf(stdout, " predecoding...");
	- global_state.decoded_bytes = Sound_DecodeAll(sample);
	- global_state.decoded_ptr = sample->buffer;
	- if (sample->flags & SOUND_SAMPLEFLAG_ERROR)
	- {
	- fprintf(stderr,
	- "Couldn't fully decode \"%s\"!\n"
	- " reason: [%s].\n"
	- " (playing first %lu bytes of decoded data...)\n",
	- filename, Sound_GetError(), global_state.decoded_bytes);
	- } /* if */
	- else
	- {
	- fprintf(stdout, "done.\n");
	- } /* else */
	- } /* if */
	-
	- SDL_PauseAudio(0);
	-
	- done_flag = 0; /* the audio callback will flip this flag. */
	- while (!done_flag)
	- {
	- #if ENABLE_EVENTS
	- SDL_PollEvent(&event);
	- if ((event.type == SDL_KEYDOWN) \|\| (event.type == SDL_QUIT))
	- done_flag = 1;
	- #endif
	-
	- SDL_Delay(10);
	- } /* while */
	-
	- SDL_PauseAudio(1);
	-
	- /*
	- * Sleep two buffers' worth of audio before closing, in order
	- * to allow the playback to finish. This isn't always enough;
	- * perhaps SDL needs a way to explicitly wait for device drain?
	- */
	- delay = 2 * 1000 * sdl_desired.samples / sdl_desired.freq;
	- SDL_Delay(delay);
	-
	- SDL_CloseAudio(); /* reopen with next sample's format if possible */
	- Sound_FreeSample(sample);
	-
	- close_archive(filename);
	- } /* for */
	-
	- Sound_Quit();
	- SDL_Quit();
	- deinit_archive();
	- return(0);
	-} /* main */
	-
	-/* end of playsound.c ... */
	-
	diff --git a/util/sdl/sound/playsound/playsound_simple.c b/util/sdl/sound/playsound/playsound_simple.c
	deleted file mode 100644
	index 16ce506b..00000000
	--- a/util/sdl/sound/playsound/playsound_simple.c
	+++ /dev/null
	@@ -1,197 +0,0 @@
	-/*
	- * SDL_sound -- An abstract sound format decoding API.
	- * Copyright (C) 2001 Ryan C. Gordon.
	- *
	- * This library is free software; you can redistribute it and/or
	- * modify it under the terms of the GNU Lesser General Public
	- * License as published by the Free Software Foundation; either
	- * version 2.1 of the License, or (at your option) any later version.
	- *
	- * This library is distributed in the hope that it will be useful,
	- * but WITHOUT ANY WARRANTY; without even the implied warranty of
	- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	- * Lesser General Public License for more details.
	- *
	- * You should have received a copy of the GNU Lesser General Public
	- * License along with this library; if not, write to the Free Software
	- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
	- */
	-
	-/**
	- * This is just a simple "decode sound, play it through SDL" example.
	- * The much more complex, fancy, and robust code is playsound.c.
	- *
	- * Please see the file COPYING in the source's root directory.
	- *
	- * This file written by Ryan C. Gordon. (icculus@icculus.org)
	- */
	-
	-#include <stdio.h>
	-#include <stdlib.h>
	-#include <string.h>
	-
	-#include "SDL.h"
	-#include "SDL_sound.h"
	-
	-/* global decoding state. */
	-typedef struct
	-{
	- Sound_Sample *sample;
	- SDL_AudioSpec devformat;
	- Uint8 *decoded_ptr;
	- Uint32 decoded_bytes;
	-} PlaysoundAudioCallbackData;
	-
	-/*
	- * This variable is flipped to non-zero when the audio callback has
	- * finished playing the whole file.
	- */
	-static volatile int global_done_flag = 0;
	-
	-
	-/*
	- * The audio callback. SDL calls this frequently to feed the audio device.
	- * We decode the audio file being played in here in small chunks and feed
	- * the device as necessary. Other solutions may want to predecode more
	- * (or all) of the file, since this needs to run fast and frequently,
	- * but since we're only sitting here and waiting for the file to play,
	- * the only real requirement is that we can decode a given audio file
	- * faster than realtime, which isn't really a problem with any modern format
	- * on even pretty old hardware at this point.
	- */
	-static void audio_callback(void userdata, Uint8 stream, int len)
	-{
	- PlaysoundAudioCallbackData data = (PlaysoundAudioCallbackData ) userdata;
	- Sound_Sample *sample = data->sample;
	- int bw = 0; /* bytes written to stream this time through the callback */
	-
	- while (bw < len)
	- {
	- int cpysize; /* bytes to copy on this iteration of the loop. */
	-
	- if (data->decoded_bytes == 0) /* need more data! */
	- {
	- /* if there wasn't previously an error or EOF, read more. */
	- if ( ((sample->flags & SOUND_SAMPLEFLAG_ERROR) == 0) &&
	- ((sample->flags & SOUND_SAMPLEFLAG_EOF) == 0) )
	- {
	- data->decoded_bytes = Sound_Decode(sample);
	- data->decoded_ptr = sample->buffer;
	- } /* if */
	-
	- if (data->decoded_bytes == 0)
	- {
	- /* ...there isn't any more data to read! */
	- memset(stream + bw, '\0', len - bw); /* write silence. */
	- global_done_flag = 1;
	- return; /* we're done playback, one way or another. */
	- } /* if */
	- } /* if */
	-
	- /* we have data decoded and ready to write to the device... */
	- cpysize = len - bw; /* len - bw == amount device still wants. */
	- if (cpysize > data->decoded_bytes)
	- cpysize = data->decoded_bytes; /* clamp to what we have left. */
	-
	- /* if it's 0, next iteration will decode more or decide we're done. */
	- if (cpysize > 0)
	- {
	- /* write this iteration's data to the device. */
	- memcpy(stream + bw, (Uint8 *) data->decoded_ptr, cpysize);
	-
	- /* update state for next iteration or callback */
	- bw += cpysize;
	- data->decoded_ptr += cpysize;
	- data->decoded_bytes -= cpysize;
	- } /* if */
	- } /* while */
	-} /* audio_callback */
	-
	-
	-
	-static void playOneSoundFile(const char *fname)
	-{
	- PlaysoundAudioCallbackData data;
	-
	- memset(&data, '\0', sizeof (PlaysoundAudioCallbackData));
	- data.sample = Sound_NewSampleFromFile(fname, NULL, 65536);
	- if (data.sample == NULL)
	- {
	- fprintf(stderr, "Couldn't load '%s': %s.\n", fname, Sound_GetError());
	- return;
	- } /* if */
	-
	- /*
	- * Open device in format of the the sound to be played.
	- * We open and close the device for each sound file, so that SDL
	- * handles the data conversion to hardware format; this is the
	- * easy way out, but isn't practical for most apps. Usually you'll
	- * want to pick one format for all the data or one format for the
	- * audio device and convert the data when needed. This is a more
	- * complex issue than I can describe in a source code comment, though.
	- */
	- data.devformat.freq = data.sample->actual.rate;
	- data.devformat.format = data.sample->actual.format;
	- data.devformat.channels = data.sample->actual.channels;
	- data.devformat.samples = 4096; /* I just picked a largish number here. */
	- data.devformat.callback = audio_callback;
	- data.devformat.userdata = &data;
	- if (SDL_OpenAudio(&data.devformat, NULL) < 0)
	- {
	- fprintf(stderr, "Couldn't open audio device: %s.\n", SDL_GetError());
	- Sound_FreeSample(data.sample);
	- return;
	- } /* if */
	-
	- printf("Now playing [%s]...\n", fname);
	- SDL_PauseAudio(0); /* SDL audio device is "paused" right after opening. */
	-
	- global_done_flag = 0; /* the audio callback will flip this flag. */
	- while (!global_done_flag)
	- SDL_Delay(10); /* just wait for the audio callback to finish. */
	-
	- /* at this point, we've played the entire audio file. */
	- SDL_PauseAudio(1); /* so stop the device. */
	-
	- /*
	- * Sleep two buffers' worth of audio before closing, in order
	- * to allow the playback to finish. This isn't always enough;
	- * perhaps SDL needs a way to explicitly wait for device drain?
	- * Most apps don't have this issue, since they aren't explicitly
	- * closing the device as soon as a sound file is done playback.
	- * As an alternative for this app, you could also change the callback
	- * to write silence for a call or two before flipping global_done_flag.
	- */
	- SDL_Delay(2 * 1000 * data.devformat.samples / data.devformat.freq);
	-
	- /* if there was an error, tell the user. */
	- if (data.sample->flags & SOUND_SAMPLEFLAG_ERROR)
	- fprintf(stderr, "Error decoding file: %s\n", Sound_GetError());
	-
	- Sound_FreeSample(data.sample); /* clean up SDL_Sound resources... */
	- SDL_CloseAudio(); /* will reopen with next file's format. */
	-} /* playOneSoundFile */
	-
	-
	-int main(int argc, char **argv)
	-{
	- int i;
	-
	- if (!Sound_Init()) /* this calls SDL_Init(SDL_INIT_AUDIO) ... */
	- {
	- fprintf(stderr, "Sound_Init() failed: %s.\n", Sound_GetError());
	- SDL_Quit();
	- return(42);
	- } /* if */
	-
	- for (i = 1; i < argc; i++) /* each arg is an audio file to play. */
	- playOneSoundFile(argv[i]);
	-
	- /* Shutdown the libraries... */
	- Sound_Quit();
	- SDL_Quit();
	- return(0);
	-} /* main */
	-
	-/* end of playsound-simple.c ... */
	-

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