This commit was manufactured by cvs2svn to create branch 'AMANCIO'.

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diff --git a/sys/i386/isa/sound/README.FREEBSD b/sys/i386/isa/sound/README.FREEBSD
new file mode 100644
index 000000000000..3603bfe040e9
--- /dev/null
+++ b/sys/i386/isa/sound/README.FREEBSD
@@ -0,0 +1,165 @@
+GUSPNP15 8/6//97
+ftp://rah.star-gate.com/pub/guspnp.tar.gz
+
+This sound driver is for FreeBSD 3.0-current .
+
+
+Major enhancement we now support mapping the dma buffer to user
+space for write operations only. This features is useful for 
+games like xquake . So far the mmap features seems to work
+with the GUS PnP Pro and with the SB16. 
+
+For xquake you will need:
+
+1. a few mods to the linux loadable module so just download it and
+ replace yours:
+cd /sys/i386/
+tar -xzf linux_ioctl.tar.gz
+cd /usr/src/lkm/linux
+make 
+make install
+modunload -i 0    (if you have the linux up and running)
+linux
+ftp://rah.star-gate.com/linux_ioctl.tar.gz
+
+2. Install the linux lib 2.4 package:
+pkg_add linux_lib-2.4.tgz
+
+ftp://ftp.freebsd.org/pub/FreeBSD/packages-current/emulators/linux_lib-2.4.tgz
+
+If you don't have xquake you can get it from:
+
+3. ftp://ftp.freebsd.org/pub/idgames/idstuff/unsup/intel_linux_quake101.tgz
+
+
+A simple test program mmap_test.c is provided in the sound driver directory.
+So copy it to your home directory or favorite place and compile it.
+cc -o mmap_test mmap_test.c
+
+record a sample session:
+cat /dev/dsp >smpl
+
+now run mmap_test . You will hear a loud pop thats a test program
+then the sample stream will loop .
+
+
+Phew, now back to sound driver land 8)
+
+
+This is a minor sound driver release for  the GUS PnP Pro , GUS MAX,
+AudioTrix Pro , SB16 and SB16 PnP. There is also an experimental
+AWE sub model.
+
+Included Luigi's clean up work . Tnks!
+
+Randall Hopper's SB16 speed setting fix . Tnks!
+
+Fixed signal handling throught out the sound driver. The problem
+first surfaced with the SB16 not generating an interrupt on the
+last dma request and SB afficionados quickly pointed out the 
+bug in the driver. I had to put back the auto dma feature for
+xquake :(
+
+
+NOTE YOU DON'T NEED THE PNP DRIVER FOR THE GUS PNP SINCE THE DRIVER
+HAS BUILTIN PNP SUPPORT FOR THE GUS.
+
+My P133 Bios supports PnP. To first figured out what my SB16 PnP
+was being configured to I booted win95 and checked out what Win95
+ended up with . Alternatively, if you want to you can also down
+load Sujal Patel's PnP driver:
+
+ ftp://rah.start-gate.com/pub/FreeBSD-ISA_PnP_June8.tar.gz
+
+follow the instructions on how to configure your PnP soundcard or 
+ISA device.
+Lugi also has  Sujal's PnP driver . 
+
+
+---
+
+	http://www.iet.unipi.it/~luigi/pnp.c
+	http://www.iet.unipi.it/~luigi/pnp.h
+
+		the two main files. Follow Sujal Patel's instructions
+		for installing pnp support.
+
+	http://www.iet.unipi.it/~luigi/pnpinfo.tgz
+---
+
+This is my kernel configuration for my SB16 PnP:
+
+controller      snd0
+device sb0      at isa? port 0x220 irq 10 conflicts drq 3 vector sbintr
+device sbxvi0   at isa? port? irq? drq 5 conflicts
+device opl0     at isa? port 0x388 conflicts
+device sbmidi0  at isa? port 0x300 irq? conflicts
+
+----
+
+
+The difference between a GUS PnP and a GUS PnP PRO is that the Pro
+comes with 512kb. I went out an got a 1mb 60ns 30pin simm and installed
+it on my GUS PnP. I own a GUS PnP and a GUS PnP PRO.
+
+
+To unpack:
+cd /
+mv /sys/i386/isa/sound /sys/i386/isa/sound.old
+cd /sys/i386/isa/
+tar -xzf guspnp15.tar.gz 
+
+
+cd /sys/i386/conf
+Edit the kernel config file in /sys/i386/conf
+
+IF YOU HAVE A GUS MAX or GUS add the option NOGUSPNP to the config file
+	OPTIONS NOGUSPNP
+  THEN follow the instructions for NONPNP CONFIGURATION
+
+
+PNP CONFIGURATION
+To configure your guspnp PRO if you have a motherboard which supports
+PnP:
+
+	controller snd0
+	device gus0 at isa?  vector gusintr
+
+NONPNP CONFIGURATION
+To configure your guspnp PRO if your motherboard does NOT support
+PnP:
+
+device gus0 at isa? port  0x220 irq 11 drq 1 flags 0x3 vector gusintr
+
+
+the gus pnp is fully software configurable and the above configuration
+is setup for full duplex audio. The dma channel settings are:
+
+drq 1   	--- DMA channel for playback
+
+flags 0x3	--- DMAN channel for recording
+
+In the event that you have to configure a GUS PnP manually or a GUS MAX:
+
+     Available IRQs for the GUS are:
+     3, 5, 7, 9, 11, 12, 15
+
+     Available DMA channels for the GUS are:
+     1, 3, 5, 6, 7
+
+
+config <kernel-file>
+cd /sys/compile/<kernel-file>
+make 
+make install
+
+THERE IS NO NEED TO BOOT TO DOS TO CONFIGURE YOUR GUS PNP. 
+THE DRIVER HAS BUILTIN SUPPORT FOR PNP WHICH I GOT
+FROM THE  GRAVIS DRIVER DEVELOPMENT KIT
+
+Many thanks to  Brian Litzinger <brian@MediaCity.com> for
+porting the sound driver 3.5 to 2.2-current. Well, thats a while
+ago hence the reference to 2.2. 
+
+	Have fun,
+	Amancio
diff --git a/sys/i386/isa/sound/Readme b/sys/i386/isa/sound/Readme
new file mode 100644
index 000000000000..259414357c7b
--- /dev/null
+++ b/sys/i386/isa/sound/Readme
@@ -0,0 +1,248 @@
+
+VoxWare v3.5-alpha5 release notes
+---------------------------------
+
+IMPORTANT! This version of the driver is compatible only with Linux versions
+	   1.3.33 and later. It may work with earlier ones as a loadable
+	   module but...
+
+	   Also this is an ALPHA test version which has not been tested
+	   with all cards. At least AEDSP16 support will not work. PAS16
+	   and PSS supports have not been tested. /dev/dsp and /dev/audio
+ 	   playback with standard GUS sounds scrambled. 16 bit mode of
+	   SB16 doesn't work.
+
+Please read the SOUND-HOWTO (available from sunsite.unc.edu and other Linux ftp 
+sites). It contains much more information than this file.
+
+*****************************************************************
+* NEW! VoxWare home page is http://personal.eunet.fi/pp/voxware *
+* The file Readme.cards contains card specific instructions     *
+* about configuring various cards.				*
+*****************************************************************
+
+There are some programming information (little bit old) in the
+Hacker's Guide 
+(ftp://nic.funet.fi/pub/OS/Linux/ALPHA/sound/snd-sdk-doc-0.1.ps.gz).
+Believe me: The file is really there. The directory is just hidden and
+you have to cd into it before the file is visible. Note: This directory
+was accidently removed some time ago but it's now back.
+
+I have got many patches from various persons during last year. Some of
+them are still on my mailbox and they should be included in versions
+after v3.0 (I will not add aditional features before v3.0 is ready).
+
+   ====================================================
+-  THIS VERSION ____REQUIRES____ Linux 1.3.33 OR LATER.
+   ====================================================
+
+-  THIS VERSION MAY NOT WORK WITH Linux VERSIONS RELEASED
+   AFTER end of Nov 1995. If this version doesn't compile with
+   your kernel version, please use the sound driver version
+   included in your kernel.
+
+You will need the snd-util-3.0.tar.gz and snd-data-0.1.tar.Z
+packages to use this driver. They should be in the same
+ftp site or BBS from where you got this driver. For
+example at nic.funet.fi:pub/OS/Linux/*.
+
+If you are looking for the installation instructions, please
+look at linux/Readme.
+
+Supported soundcards
+--------------------
+
+Gravis Ultrasound (GUS)
+GUS MAX
+GUS with the 16 bit sampling daughtercard
+PAS16
+Windows Sound System compatible soundcards
+ECHO-PSS (cards based on the PSS architecture by Analog Devices.
+	Including Orchid SW32, Cardinal DSP16 among others).
+		(NOTE! WSS mode may not work (DMA channel setup problem)).
+MediaTriX AudioTriX Pro (OPL4 and the optional effect daughtercard
+	require special initialization. There is a program (setfx) in
+	the snd-util-3.0.tar.gz package which does it).
+Ensoniq SoundScape (works but needs some improvements)
+MV Jazz16 based soundcards (ProSonic, 3D etc).
+SoundMan Wave (recording may not work, mixer support is limited)
+Mozart (OAK OTI-601 interface chip) based soundcards.
+MAD16 (an interface chip by OPTi) based soundcards (TB Tropez ???).
+(NOTE! The MAD16 looks similar to the Mozart chip. It could be a good
+idea to configure MAD16 cards as Mozart ones. The MAD16 driver doesn't set
+up MPU401 which the Mozart one does.
+CS4232 based cards such as AcerMagic S23.
+
+
+In addition all Sound Blaster models and clones (up to AWE32) work if
+you want to use them.
+
+The Emu synthesizer chip of AWE32 is not and will not be supported. The same is
+true with the ASP chip also. Creative Technology will not release detailed
+information about them so it's not possible to support them.
+
+If you want to get support for AWE32 or ASP, please contact Creative Labs.
+Ask _politely_ if they are going to support Linux. Maybe they change
+their policy if there is enough demand.
+
+===========================================================================
+If your card is compatible with SB, MPU401 or Windows Sound System, it
+may work with the driver even if it's not listed in the above list. In this
+case it may require initialization using DOS. Just start DOS and cold
+boot to Linux (etc.) by hitting ctrl-alt-del. 
+===========================================================================
+
+Compatibility with the earlier versions
+---------------------------------------
+
+There have been some changes in soundcard.h after v2.5 of the driver
+(v2.90 is compatible with this one). Binaries compiled with this version
+of soundcard.h will not work with v2.0 and earlier.
+
+Contributors
+------------
+
+This driver contains code by several contributors. In addition several other
+persons have given usefull suggestions. The following is a list of major
+contributors. (I could have forgotten some names.)
+
+	Craig Metz	1/2 of the PAS16 Mixer and PCM support
+	Rob Hooft	Volume computation algorithm for the FM synth.
+	Mika Liljeberg	uLaw encoding and decoding routines
+	Andy Fingerhut	New ulaw conversion tables (ulaw.h)
+	Jeff Tranter	Linux SOUND HOWTO document
+	Greg Lee	Volume computation algorithm for the GUS and
+			lot's of valuable suggestions.
+	Andy Warner	ISC port
+	Jim Lowe,
+	Amancio Hasty Jr	FreeBSD/NetBSD port
+	Anders Baekgaard 	Bughunting and valuable suggestions.
+	Joerg Schubert	SB16 DSP support.
+	Andrew Robinson Improvements to the GUS driver
+	Megens SA	MIDI recording for SB and SB Pro.
+	Mikael Nordqvist  Linear volume support for GUS and
+			  nonblocking /dev/sequencer.
+	Ian Hartas		SVR4.2 port
+	Markus Aroharju	and
+	Risto Kankkunen		Major contributions to the mixer support
+				of GUS v3.7.
+	Hunyue Yau	Mixer support for SG NX Pro.
+	Marc Hoffman	PSS support.
+	Rainer Vranken	Initialization for Jazz16 (ProSonic, MV3D, SM Wave).
+	Peter Trattler	Initial version of loadable module support for Linux.
+	JRA Gibson	16 bit mode for Jazz16
+	Davor Jadrijevic MAD16 support
+	Gregor Hoffleit	Mozart support
+	Riccardo Facchetti Audio Excel DSP 16 (aedsp16) support
+
+There are propably many other names missing. If you have sent me some
+patches and your name is not in the above list, please inform me.
+
+Sponsors etc.
+-------------
+
+The following companies have greatly helped development of this driver 
+in form of a free copy of their product:
+
+Novell, Inc.		UnixWare personal edition + SDK
+The Santa Cruz Operation, Inc. 	A SCO OpenServer + SDK
+Ensoniq Corp,		a SoundScape card and extensive amount of assistance
+MediaTriX Peripherals Inc, a AudioTriX Pro card + SDK
+Acer, Inc.		a pair of AcerMagic S23 cards.
+
+In addition the following companies have provided me sufficial amount
+of technical information at least some of their products (free or $$$):
+
+Advanced Gravis Computer Technology Ltd.
+Media Vision Inc.
+Analog Devices Inc.
+Logitech Inc.
+Aztech Labs Inc.
+Crystal Semiconductor Corporation,
+Integrated Circuit Systems Inc.
+OAK Technology
+OPTi
+Ad Lib Inc. ($$)
+Music Quest Inc. ($$)
+Creative Labs ($$$)
+
+If you have some problems
+=========================
+
+Read the sound HOWTO (sunsite.unc.edu:/pub/Linux/docs/...?).
+Also look at the home page (http://personal.eunet.fi/pp/voxware). It may
+contain info about some recent bug fixes.
+
+It's likely that you have some problems when trying to use the sound driver
+first time. Soundcards don't have standard configuration so there are no
+good default configuration to use. Please try to use same I/O, DMA and IRQ
+values for the soundcard than with DOS.
+
+If you get an error message when trying to use the driver, please look
+at /var/adm/messages for more verbose error message.
+
+
+In general the easiest way to diagnoze problems is to do "cat /dev/sndstat".
+
+If you get an error message, there are some problems with the driver setup:
+
+	- "No such file or directory" tells that the device files for
+	the sound driver are missing. Use the script at the end of
+	linux/drivers/sound/Readme.linux to create them.
+
+	- "No such device" telss that the sound driver is not in the kernel.
+	You have to reconfigure and recompile the kernel to have the sound
+	driver. Compiling the driver doesn't help alone. You have to boot
+	with the newly compiled one before the driver becomes active.
+	The Linux-HOWTO should help in this step.
+
+The following errors are likely with /dev/dsp and /dev/audio.
+
+	- "No such device or address". This error message should not happen
+	with /dev/sndstat but it's possible with the other sound devices.
+	This error indicates that there are no suitable hardware for the
+	device file or the sound driver has been compiled without support for
+	this particular device. For example /dev/audio and /dev/dsp will not
+	work if "digitized voice support" was not enabled during "make config".
+	
+	- "Device or resource busy". Propably the IRQ (or DMA) channel 
+	required by the soundcard is in use by some other device/driver.
+
+	- "I/O error". Almost certainly (99%) it's an IRQ or DMA conflict.
+	Look at the kernel messages in /var/adm/notice for more info.
+
+	- "Invalid argument". The application is calling ioctl()
+	with impossible parameters. Check that the application is
+	for sound driver version 2.X or later.
+
+In general the printout of of /dev/sndstat should tell what is the problem.
+It's possible that there are bugs in the sound driver but 99% of the problems
+reported to me are caused by somehow incorrect setup during "make config".
+
+For owners of TI TM4000M notebooks
+----------------------------------
+
+There appears to be some kind of conflict between the sound support
+(MV Jazz), mouse port and VoxWare. You could try to configure kernel
+with the C&T 82C710 mouse port support disabled.
+
+Hannu
+
+Regards,
+
+Hannu Savolainen
+hannu@voxware.pp.fi
+(or Hannu.Savolainen@cctap.carel.fi in case the above bounces)
+
+Snail mail:	Hannu Savolainen
+		Hiekkalaiturintie 3 A 8
+		00980 Helsinki
+		Finland
+
+NOTE!	I propably don't answer to Snail mail or FAX messages. Sending answer
+	to each of them is simply too expensive and time consuming. However I
+	try to reply every email message I get (within a week). If you don't 
+	get response, please check how your address is written in the message
+	header. I can't answer if I don't have a valid reply address.
+
+VoxWare home page is http://personal.eunet.fi/pp/voxware
diff --git a/sys/i386/isa/sound/Readme.aedsp16 b/sys/i386/isa/sound/Readme.aedsp16
new file mode 100644
index 000000000000..b205a9d0ff56
--- /dev/null
+++ b/sys/i386/isa/sound/Readme.aedsp16
@@ -0,0 +1,6 @@
+Informations about Audio Excel DSP 16 can be found in the source
+file aedsp16.c
+Please, read the head of the source before using it. It contain useful
+informations.
+
+    Riccardo
diff --git a/sys/i386/isa/sound/Readme.cards b/sys/i386/isa/sound/Readme.cards
new file mode 100644
index 000000000000..fe17aa064def
--- /dev/null
+++ b/sys/i386/isa/sound/Readme.cards
@@ -0,0 +1,845 @@
+Configuring VoxWare 3.0 (for Linux) with some most common soundcards
+====================================================================
+
+NOTE! This document may contain some error. Please inform me
+      if you find any mistakes.
+
+Read this before trying to configure the driver
+-----------------------------------------------
+
+There are currently many cards that work with VoxWare. Some of the cards
+have native support while the others work since they emulate some other
+cards (usually SB, MSS/WSS and/or MPU401). The following cards have native
+support in VoxWare. Detailed instructions for configuring these cards
+will be given later in this document.
+
+Pro Audio Spectrum 16 (PAS16) and compatibles:
+	Pro Audio Spectrum 16
+	Pro Audio Studio 16
+	Logitech Sound Man 16
+	NOTE! The original Pro Audio Spectrum as well as the PAS+ are not
+	      and will not be supported by VoxWare.
+
+Media Vision Jazz16 based cards
+	Pro Sonic 16
+	Logitech SoundMan Wave
+	(Other Jazz based cards should work but I don't have any reports
+	about them).
+
+Sound Blasters
+	SB 1.0 to 2.0
+	SB Pro
+	SB 16
+	NOTE!	The ASP chip and the EMU synth of the AWE32 is not supported 
+		since their manufacturer doesn't release information about
+		the card. However both the AB16ASP and the AWE32 work with
+		VoxWare just like a SB16. Also see the comment about some
+		unsupported cards at the end of this file.
+	SB16 compatible cards by other manufacturers than Creative.
+		You have been fooled since there are no SB16 compatible
+		cards in the market (July95). It's likely that your card
+		is compatible just with SB Pro but there is also a non SB
+		compatible 16 bit mode. Usually it's MSS/WSS but could also
+		be a proprietary one like MV Jazz16.
+
+Gravis Ultrasound (GUS)
+	GUS
+	GUS + the 16 bit option
+	GUS MAX
+	GUS ACE (No MIDI port and audio recording)
+
+MPU-401	and compatibles
+	The driver works both with the full (intelligent mode) MPU-401
+	cards (such as MPU IPC-T and MQX-32M) and with the UART only
+	dumb MIDI ports. MPU-401 is currently the most common MIDI
+	interface. Most soundcards are compatible with it. However
+	don't enable MPU401 mode blindly. Many cards having native support
+	in VoxWare have their own MPU401 driver. Enabling the standard one
+	will cause a conflict with these cards. So look if your card is
+	in the list of supported cards before enabling MPU401.
+
+Windows Sound System (MSS/WSS)
+	Even Microsoft has discontinued their own Sound System card, they
+	managed to make a standard. MSS compatible cards are based on a 
+	codec chip which is easily available from at least two manufacturers
+	(AD1848 by Analog Devices and CS4231/CS4248 by Crystal Semiconductor).
+	Currently most soundcards are based on one of the MSS compatible codec
+	chip. The CS4231 is used in the high quality cards such as GUS MAX,
+	MediaTriX AudioTriX Pro and TB Tropez (GUS MAX is not MSS compatible). 
+
+	Having a AD1848, CS4248 or CS4231 codec chip on the card is a good
+	sign. Even if the card is not MSS compatible, it could be easy to write
+	support for it to VoxWare. Note also that most MSS compatible cards
+	require special boot time initialization which may not be present
+	in VoxWare. Also some MSS compatible cards have native support in
+	VoxWare. Enabling the MSS support with these cards is likely to
+	cause a conflict. So check if your card is listed in this file before
+	enabling the MSS support.
+
+6850 UART MIDI
+	This UART chip is used in the MIDI interface of some (rare)
+	soundcards. It's supported by VoxWare in case you need it.
+
+Yamaha FM synthesizers (OPL2, OPL3 and OPL4)
+	Most soundcards have a FM synthesizer chip. The OPL2 is a 2
+	operator chip used in the original AdLib card. Currently it's used
+	only in the cheapest (8 bit mono) cards. The OPL3 is a 4 operator 
+	FM chip which provides better sound quality and/or more available 
+	voices than the OPL2. The OPL4 is a new chip which has a OPL3 and
+	a wave table synthesizer packed on the same chip. VoxWare supports
+	just the OPL3 mode directly. Most cards having a OPL4 (like
+	SM Wave and AudioTriX Pro) support the OPL4 mode using MPU401
+	emulation. Writing a native OPL4 support to VoxWare is difficult
+	since Yamaha doesn't give information about their sample ROM chip.
+
+	Enable the generic OPL2/OPL3 FM synthesizer support if your
+	card has a FM chip made by Yamaha. Don't enable it if your card
+	has a software (TRS) based FM emulator.
+
+PSS based cards (AD1848 + ADSP-2115 + Echo ESC614 ASIC)
+	Analog Devices and Echo Speech have together defined a soundcard
+	architecture based on the above chips. The DSP chip is used
+	for emulation of SB Pro, FM and General MIDI/MT32.
+
+	There are several cards based on this architecture. The most known
+	ones are Orchid SW32 and Cardinal DSP16. 
+
+	VoxWare supports downloading DSP algorithms to these cards.
+
+MediaTriX AudioTriX Pro
+	The ATP card is built around a CS4231 codec and a OPL4 synthesizer
+	chips. The OPL4 mode is supported by a microcontroller running a
+	General MIDI emulator. There is also a SB 1.5 compatible playback mode.
+
+Ensoniq SoundScape and compatibles
+	Ensoniq has designed a soundcard architecture based on the
+	OTTO synthesizer chip used in their professional MIDI synthesizers.
+	Several companies (including Ensoniq, Reveal and Spea) are selling
+	cards based on this architecture.
+
+MAD16 and Mozart based cards
+	The Mozart (OAK OTI-601) and MAD16 Pro (OPTi 82C929) interface
+	chips are used in many different soundcards, including some
+	cards by Reveal and Turtle Beach (Tropez). Purpose of these
+	chips is to connect other audio components to the PC bus. The
+	interface chip performs address decoding for the other chips.
+
+Audio Excell DSP16 
+	Support for this card is made by Riccardo Faccetti
+	(riccardo@cdc8g5.cdc.polimi.it). See aedsp16.c for more info.
+	
+Crystal CS4232 based cards such as AcerMagic S23
+	CS4232 is a PnP multimedia chip which contains a CS3231A codec,
+	SB and MPU401 emulations. There is support for OPL3 too.
+	(Unfortunately the MPU401 mode doesn't work).
+
+Turtle Beach Maui and Tropez
+	VoxWare supports sample, parch and program loading commands
+	described in the Maui/Tropez User's manual. There is no initialization
+	code for Maui so it must be initialized using DOS. Audio side of Tropez
+	is based on the MAD16 chip (see above).
+
+Jumpers and software configuration
+----------------------------------
+
+Some of the earliest soundcards were jumper configurable. You have to
+configure VoxWare to configure VoxWare use I/O, IRQ and DMA settings
+that match the jumpers. Just few 8 bit cards are fully jumper 
+configurable (SB 1.x/2.x, SB Pro and clones).
+Some cards made by Aztech have an EEPROM which contains the 
+config info. These cards behave much like hardware jumpered cards.
+
+Most cards have jumper for the base I/O address but other parameters
+are software configurable. Sometimes there are few other jumpers too.
+
+Latest cards are fully software configurable or they are PnP ISA
+compatible. There are no jumpers on the board.
+
+VoxWare handles software configurable cards automaticly. Just configure
+the driver to use I/O, IRQ and DMA settings which are known to work.
+You could usually use the same values than with DOS and/or Windows.
+Using different settings is possible but not recommended since it may cause
+some trouble (for example when warm booting from an OS to another or
+when installing new hardware to the machine).
+
+VoxWare sets the soft configurable parameters of the card automaticly
+during boot. Usually you don't need to run any extra initialization
+programs when booting Linux but there are some exceptions. See the
+card specific instructions (below) for more info.
+
+The drawback of software configuration is that the driver needs to know
+how the card must be initialized. It cannot initialize unknown cards
+even if they are otherwise compatible with some other cards (like SB,
+MPU401 or Windows Sound System).
+
+What if your card was not listed above?
+---------------------------------------
+
+The first thing to do is to look at the major IC chips on the card.
+Many of the latest soundcards are based on some standard chips. If you
+are lucky, all of them could be supported by VoxWare. The most common ones
+are the OPTi MAD16, Mozart, SoundScape (Ensoniq) and the PSS architectures
+listed above. Also look at the end of this file for list of unsupported
+cards and the ones which could be supported later.
+
+The last resort is to send _exact_ name and model information of the card
+to me together with a list of the major IC chips (manufactured, model) to 
+me. I could then try to check if your card looks like something familiar.
+
+There are much more cards in the word than listed above. The first thing to
+do with these cards is to check if they emulate some other card/interface
+such as SB, MSS and/or MPU401. In this case there is a chance to get the
+card to work by booting DOS before starting Linux (boot DOS, hit ctrl-alt-del
+and boot Linux without hard resetting the machine). In this method the
+DOS based driver initializes the hardware to use a known I/O, IRQ and DMA
+settings. If VoxWare is configured to use the same settings, everything should 
+work OK.
+
+
+Configuring VoxWare (with Linux)
+================================
+
+VoxWare sound driver is currently a part of Linux kernel distribution. The 
+driver files are located in directory /usr/src/linux/drivers/sound. 
+
+****************************************************************************
+*	VoxWare MUST BE CONFIGURED AND COMPILED WITH THE KERNEL. TRYING    *
+*	TO COMPILE IT ALONE WILL _NOT_ WORK.				   *
+*									   *
+*	ALWAYS USE THE SOUND DRIVER VERSION WHICH IS DISTRIBUTED WITH	   *
+*	THE KERNEL SOURCE PACKAGE YOU ARE USING. SOME ALPHA AND BETA TEST  *
+*	VERSIONS CAN BE INSTALLED FROM A SEPARATELY DISTRIBUTED PACKAGE	   *
+*	BUT CHECK THAT THE PACKAGE IS NOT MUCH OLDER (OR NEWER) THAN THE   *
+*	KERNEL YOU ARE USING. IT'S POSSIBLE THAT THE KERNEL/DRIVER 	   *
+*	INTERFACE CHANGES BETWEEN KERNEL RELEASES WHICH MAY CAUSE SOME	   *
+*	INCOMPATIBILITY PROBLEMS.					   *
+*									   *
+*	IN CASE YOU INSTALL A SEPARATELY DISTRIBUTED SOUND DRIVER VERSION, *
+*	BE SURE TO REMOVE OR RENAME THE OLD SOUND DRIVER DIRECTORY BEFORE  *
+*	INSTALLING THE NEW ONE. LEAVING OLD FILES TO THE SOUND DRIVER	   *
+*	DIRECTORY _WILL_ CAUSE PROBLEMS WHEN THE DRIVER IS USED OR	   *
+*	COMPILED.							   *
+****************************************************************************
+
+To configure the driver, run "make config" in the kernel source directory
+(/usr/src/linux). Answer y to the question about Sound card support (after
+questions about mouse, CD-ROM, ftape, etc. supports). Sound config options
+will then be asked after some additional questions.
+
+After configuring the kernel and sound driver, run "make dep" and compile
+the kernel following instructions in the kernel README.
+
+The sound driver configuration dialog
+-------------------------------------
+
+All config information of the sound driver is written to file 
+linux/drivers/sound/local.h. You may save the old version is this file and
+use it again in case you want to use the same config later. In this case
+just answer n to each question made by the sound config program and put
+the original local.h back before running "make dep".
+Don't do this if the version number of the sound driver has changed. In this
+case you have to enter the configuration information again.
+
+If you already have the sound driver installed, consult printout of
+"cat /dev/sndstat" when configuring the driver again. It gives the I/O,
+IRQ and DMA settings you have used earlier.
+
+
+The sound config program (linux/drivers/sound/configure) starts by making
+some yes/no questions. Be careful when answering to these questions since
+answering y to a question may prevent some later ones from being asked. For
+example don't answer y to the first question (PAS16) if you don't really
+have a PAS16. Don't enable more cards than you really need since they
+just consume memory. Also some drivers (like MPU401) may conflict with your
+SCSI controller and prevent kernel from booting. If you card was in the list
+of supported cards (above), please look at the card specific config 
+instructions (later in this file) before starting to configure. Some cards
+must be configured in way which is not obvious.
+
+So here is the beginning of the config dialog. Answer 'y' or 'n' to these
+questions. The default answer is shown so that (y/n) means 'y' by default and
+(n/y) means 'n'. To use the default value, just hit ENTER. But be careful
+since using the default _doesn't_ guarantee anything.
+
+Note also that all questions may not be asked. The configuration program
+may disable some questions dependig on the earlier choices. It may also
+select some options automaticly as well.
+
+  "ProAudioSpectrum 16 support",
+	- Answer 'y'_ONLY_ if you have a Pro Audio Spectrum _16_,
+	  ProAudio Studio 16 or Logitech SoundMan 16 (be sure that
+	  you read the above list correctly). Don't answer 'y' if you
+	  have some other card made by Media Vision or Logitech since they
+	  are not PAS16 compatible.
+  "SoundBlaster support",
+	- Answer 'y' if you have an original SB card made by Creative Labs
+	  or a full 100% hardware compatible clone (like Thunderboard or
+	  SM Games). If your card was in the list of supported cards (above),
+	  please look at the card specific instructions later in this file
+	  before answering this question. For an unknown card you may answer 
+	  'y' if the card claims to be SB compatible.
+
+	 Don't enable SB if you have a MAD16 or Mozart compatible card.
+
+  "Generic OPL2/OPL3 FM synthesizer support",
+	- Answer 'y' if your card has a FM chip made by Yamaha (OPL2/OPL3/OPL4).
+	  Answering 'y' is usually a safe and recommended choice. However some
+	  cards may have software (TSR) FM emulation. Enabling FM support
+	  with these cards may cause trouble. However I don't currently know
+	  such cards.
+  "Gravis Ultrasound support",
+	- Answer 'y' if you have GUS or GUS MAX. Answer 'n' if you don't
+	  have GUS since the GUS driver consumes much memory.
+	  Currently I don't have experiences with the GUS ACE so I don't
+	  know what to answer with it.
+  "MPU-401 support (NOT for SB16)",
+	- Be careful with this question. The MPU401 interface is supported
+	  by almost any soundcard today. However some natively supported cards
+	  have their own driver for MPU401. Enabling the MPU401 option with
+	  these cards wil cause a conflict. Also enabling MPU401 on a system
+	  that doesn't really have a MPU401 could cause some trouble. If your
+	  card was in the list of supported cards (above), please look at
+	  the card specific instructions later in this file.
+	  It's safe to answer 'y' if you have a true MPU401 MIDI interface
+	  card.
+  "6850 UART Midi support",
+	- It's safe to answer 'n' to this question in all cases. The 6850
+	  UART interface is so rarely used.
+  "PSS (ECHO-ADI2111) support",
+	- Answer 'y' only if you have Orchid SW32, Cardinal DSP16 or some
+	  other card based on the PSS chipset (AD1848 codec + ADSP-2115
+	  DSP chip + Echo ESC614 ASIC CHIP).
+  "16 bit sampling option of GUS (_NOT_ GUS MAX)",
+	- Answer 'y' if you have installed the 16 bit sampling daughtercard
+	  to your GUS. Answer 'n' if you have GUS MAX. Enabling this option
+	  disables GUS MAX support.
+  "GUS MAX support",
+	- Answer 'y' only if you have a GUS MAX.
+  "Microsoft Sound System support",
+	- Again think carefully before answering 'y' to this question. It's
+	  safe to answer 'y' in case you have the original Windows Sound 
+	  System card made by Microsoft or Aztech SG 16 Pro (or NX16 Pro).
+	  Also you may answer 'y' in case your card was not listed earlier
+	  in this file. For cards having native support in VoxWare, consult
+	  the card specific instructions later in this file. Some drivers
+	  have their own MSS support and enabling this option wil cause a
+	  conflict.
+  "Ensoniq Soundscape support",
+	- Answer 'y' if you have a soundcard based on the Ensoniq SoundScape
+	  chipset. Suach cards are being manufactured at least by Ensoniq,
+	  Spea and Reveal (note that Reveal makes other cards also).
+  "MediaTriX AudioTriX Pro support",
+	- Answer 'y' if you have the AudioTriX Pro.
+  "Support for MAD16 and/or Mozart based cards",
+	- Answer y if your card has a Mozart (OAK OTI-601) or MAD16
+	  (OPTi 82C928 or 82C929) audio interface chip. These chips are
+	  currently quite common so it's possible that many no-name cards
+	  have one of them. In addition the MAD16 chip is used in some
+	  cards made by known manufacturers such as Turtle Beach (Tropez),
+	  Reveal (some models) and Diamond (latest ones).
+  "SoundBlaster Pro support",
+	- Enable this option if your card is SB Pro or SB16. Enable it
+	  also with any SB Pro clones. Answering 'n' saves some amount of
+	  memory but 'y' is the safe alterative.
+  "SoundBlaster 16 support",
+	- Enable if you have a SB16 (including the AWE32).
+  "Audio Excel DSP 16 initialization support",
+	- Don't know much about this card. Look at aedsp16.c for more info.
+
+Then the configuration program asks some y/n questions about the higher
+level services. It's recommended to answer 'y' to each of these questions.
+Answer 'n' only if you know you will not need the option.
+
+  "/dev/dsp and /dev/audio supports (usually required)",
+	- Answering 'n' disables /dev/dsp and /dev/audio. Answer 'y'.
+  "MIDI interface support",
+	- Answering 'n' disables /dev/midi## devices and access to any
+	  MIDI ports using /dev/sequencer and /dev/music. This option
+	  also affects any MPU401 and/or General MIDI compatible devices.
+  "FM synthesizer (YM3812/OPL-3) support",
+	- Answer 'y' here.
+  "/dev/sequencer support",
+	- Answering 'n' disables /dev/sequencer and /dev/music.
+
+Entering the I/O, IRQ and DMA config parameters
+-----------------------------------------------
+
+After the above questions the configuration program prompts for the
+card specific configuration information. Usually just a set of
+I/O address, IRQ and DMA numbers are asked. With some cards the program
+asks for some files to be used during initialization of the card. For example
+many cards have a DSP chip or microprocessor which must be initialized by
+downloading a program (microcode) file to the card. In some cases this file
+is written to a .h file by the config program and then included to the driver
+during compile.
+
+Instructions for answering these questions are given in the next section.
+
+
+Card specific information
+=========================
+
+This section gives additional instructions about configuring some cards.
+Please refer manual of your card for valid I/O, IRQ and DMA numbers. Using
+the same settings with DOS/Windows and VoxWare is recommended. Using
+different values could cause some problems when switching between
+different operating systems.
+
+SoundBlasters (the original ones by Creative)
+---------------------------------------------
+
+It's possible to configure these cards to use different I/O, IRQ and
+DMA settings. Since the available settings have changed between various
+models, you have to consult manual of your card for the proper ones. It's
+a good idea to use the same values than with DOS/Windows. With SB and SB Pro
+it's the only choice. SB16 has software selectable IRQ and DMA channels but
+using different values with DOS and Linux is likely to cause troubles. The
+DOS driver is not able to reset the card properly after warm boot from Linux
+if Linux has used different IRQ or DMA values.
+
+The original (steam) Sound Blaster (versions 1.x and 2.x) use always
+DMA1. There is no way to change it.
+
+The SB16 needs two DMA channels. A 8 bit one (1 or 3) is required for
+8 bit operation and a 16 bit one (5, 6 or 7) for the 16 bit mode. In theory
+it's possible to use just one (8 bit) DMA channel by answering the 8 bit
+one when the configuration program asks for the 16 bit one. This may work
+in some systems but is likely to cause terrible noise on some other systems.
+
+NOTE!	Don't enable the SM Games option (asked by the configuration program)
+	if you are not 101% sure that your card is a Logitech Soundman Games
+	(not a SM Wave or SM16).
+
+SB Clones
+---------
+
+First of all: There are no SB16 clones. There are SB Pro clones with a
+16 bit mode which is not SB16 compatible. The most likely alternative is that
+the 16 bit mode means MSS/WSS.
+
+There are just few fully 100% hardware SB or SB Pro compatible cards.
+I know just Thunderboard and SM Games. Other cards require some kind of
+hardware initialization before they become SB compatible. Check if your card
+was listed in the beginning of this file. In this case you should follow
+instructions for your card later in this file.
+
+For other not fully SB clones yoy may try initialization using DOS in
+the following way:
+
+	- Boot DOS so that the card specific driver gets run.
+	- Hit ctrl-alt-del (or use loadlin) to boot Linux. Don't
+	  switch off power or press the reset button.
+	- If you use the same I/O, IRQ and DMA settings in Linux, the
+	  card should work.
+
+If your card is both SB and MSS compatible, I recommend using the MSS mode.
+Most cards of this kind are not able to work in the SB and the MSS mode 
+simultaneously. Using the MSS mode provides 16 bit recording and playback.
+
+ProAudioSpectrum 16 and compatibles
+-----------------------------------
+
+There are nothing special with these cards. Just don't enable any
+other cards in case you don't have them also. The PAS16 has
+a SB mode so the driver config program will prompt for the SB settings
+do. Use I/O 0x220 and DMA1 for the SB mode. Ensure that you assign different
+IRQ numbers for the SB and PAS16 modes.
+
+With PAS16 you can use two audio device files at the same time. /dev/dsp (and
+/dev/audio) is connected to the 8/16 bit native codec and the /dev/dsp1 (and
+/dev/audio1) is connected to the SB emulation (8 bit mono only).
+
+Gravis Ultrasound
+-----------------
+
+There are many different revisions of the Ultrasound card (GUS). The
+earliest ones (pre 3.7) don't have a hardware mixer. With these cards
+the driver uses a software emulation for synth and pcm playbacks. It's
+also possible to switch some of the inputs (line in, mic) off by setting
+mixer volume of the channel level below 10%. For recording you have
+to select the channel as a recording source and to use volume above 10%.
+
+GUS 3.7 has a hardware mixer.
+
+GUS MAX and the 16 bit sampling daughtercard have a CS4231 codec chip which 
+also contains a mixer.
+
+Configuring GUS is simple. Just enable the GUS support and GUS MAX or
+the 16 bit daughtercard if you have them. Note that enabling the daughter
+card disables GUS MAX driver.
+
+With just the standard GUS enabled the configuration program prompts
+for the I/O, IRQ and DMA numbers for the card. Use the same values than
+with DOS.
+
+With the daughter card option enabled you will be prompted for the I/O,
+IRQ and DMA numbers for the daughter card. You have to use different I/O
+and DMA values than for the standard GUS. The daughter card permits
+simultaneous recording and playback. Use /dev/dsp (the daughtercard) for
+recording and /dev/dsp1 (GUS GF1) for playback.
+
+GUS MAX uses the same I/O address and IRQ settings than the original GUS
+(GUS MAX = GUS + a CS4231 codec). In addition an extra DMA channel may be used.
+Using two DMA channels permits simultaneous playback using two devices
+(dev/dsp0 and /dev/dsp1). The second DMA channel is required for
+full duplex audio.
+To enable the second DMA channels, give a valid DMA channel when the config
+program asks for the GUS MAX DMA (entering -1 disables the second DMA).
+Using 16 bit DMA channels (5,6 or 7) is recommended.
+
+If you have problems in recording with GUS MAX, you could try to use
+just one 8 bit DMA channel. Recording will not work with one DMA
+channel if it's a 16 bit one.
+
+
+
+MPU401 and Windows Sound System
+-------------------------------
+
+Again. Don't enable these options in case your card is listed
+somewhere else in this file.
+
+Configuring these cards is obvious (or it should be). With MSS
+you should propably enable the OPL3 synth also since
+most MSS compatible cards have it. However check that this is true
+before enabling OPL3.
+
+VoxWare supports more than one MPU401 compatible cards at the same time
+but the config program asks config info for just the first of them.
+Adding the second or third MPU interfaces must be done manually by
+editing sound/local.h (after running the config program). Add defines for
+MPU2_BASE & MPU2_IRQ (and MPU3_BASE & MPU3_IRQ) to the file.
+
+CAUTION!
+
+The default I/O base of Adaptec AHA-1542 SCSI controller is 0x330 which
+is also the default of the MPU401 driver. Don't configure the sound driver to
+use 0x330 as the MPU401 base if you have a AHA1542. The kernel will not boot
+if you make this mistake.
+
+PSS
+---
+
+Even the PSS cards are compatible with SB, MSS and MPU401, you must not
+enable these options when configuring the driver. The configuration
+program handles these options itself. (You may use the SB, MPU and MSS options
+together with PSS if you have another card on the system).
+
+The PSS driver enables MSS and MPU401 modes of the card. SB is not enabled 
+since it doesn't work concurrently with MSS. The driver loads also a
+DSP algorithm which is used to for the general MIDI emulation. The
+algorithm file (.ld) is read by the config program and written to a
+file included when the pss.c is compiled. For this reason the config
+program asks if you want to download the file. Use the genmidi.ld file
+distributed with the DOS/Windows drivers of the card (don't use the mt32.ld).
+With some cards the file is called 'synth.ld'. You must have access to
+the file when configuring the driver. The easiest way is to mount the DOS
+partition containing the file with Linux. 
+
+It's possible to load your own DSP algorithms and run them with the card.
+Look at the directory sound/pss_test for more info (in the VoxWare-3.0.tar.gz)
+package.
+
+AudioTriX Pro
+-------------
+
+You have to enable the OPL3 and SB (not SB Pro or SB16) drivers in addition
+to the native AudioTriX driver. Don't enable MSS or MPU drivers.
+
+Configuring ATP is little bit tricky since it uses so many I/O, IRQ and
+DMA numbers. Using the same values than with DOS/Win is a good idea. Don't
+attemp to use the same IRQ or DMA channels twice.
+
+The SB mode of ATP is implemented so the the ATP driver just enables SB
+in the proper address. The SB driver handles the rest. Yoy have to configure
+both the SB driver and the SB mode of ATP to use the same IRQ, DMA and I/O
+settings.
+
+Also the ATP has a microcontroller for the General MIDI emulation (OPL4).
+For this reason the driver asks for the name of a file containing the
+microcode (TRXPRO.HEX). This file is usually located in the directory
+where the DOS drivers were installed. You must have access to this file
+when configuring the driver. 
+
+If you have the effects daughtercard, it must be initialized by running
+the setfx program of snd-util-3.0.tar.gz package. This step is not required
+when using the (future) binary distribution version of the driver.
+
+Ensoniq SoundScape
+------------------
+
+The SoundScape driver handles initialization of MSS and MPU supports
+itself so you don't need to enable other drivers than SoundScape
+(enable also the /dev/dsp, /dev/sequencer and MIDI supports).
+
+SoundScape driver uses the MSS compatible codec of the card. It's important
+to note that /dev/dsp0 (/dev/dsp is linked to /dev/dsp0 by default)
+doesn't work with SoundScape (yet). The 'ssinit' program needs /dev/dsp0 so
+that's the reason why it's there. It's possible that 'primary' pcm channel
+becomes supported later. Currently the card's firmware doesn't contain
+support for it.
+
+With 3.0 of VoxWare you have to change your system to use /dev/dsp1 by default
+so execute: cd /dev;rm dsp;ln -s dsp1 dsp after you have installed VoxWare
+3.0 (first time).
+
+The configuration program asks two DMA channels and two interrupts. One IRQ
+and one DMA is used by the MSS codec. The second IRQ is required for the
+MPU401 mode (you have to use different IRQs for both purposes).
+The second DMA channel is required for initialization of the microcontroller.
+You have to use separate DMA channels.
+
+The SoundScape card has a Motorola microcontroller which must initialized
+_after_ boot (the driver doesn't initialize it during boot).
+The initialization is done by running the 'ssinit' program which is
+distributed in the snd-util-3.0.tar.gz package. You have to edit two
+defines in the ssinit.c and then compile the program. You may run ssinit 
+manually (after each boot) or add it to /etc/rc.d/rc.local.
+
+The ssinit program needs the microcode file that comes with the DOS/Windows
+driver of the card. You will need to use version 1.30.00 or later
+of the microcode file (sndscape.co0 or sndscape.co1 depending on
+your card model). THE OLD sndscape.cod WILL NOT WORK. IT WILL HANG YOUR
+MACHINE. The only way to get the new microcode file is to download
+and install the DOS/Windows driver from ftp://ftp.ensoniq.com/pub.
+
+Then you have to select the proper microcode file to use: soundscape.co0
+is the right one for most cards and sndscape.co1 is for few (older) cards
+made by Reveal and/or Spea. The driver has capability to detect the card
+version during boot. Look at the boot log messages in /var/adm/messages
+and locate the sound driver initialization message for the SoundScape
+card. If the driver displays string <Ensoniq Soundscape (old)>, you have
+an old card and you will need to use sndscape.co1. For other cards use
+soundscape.co0.
+
+Check /var/adm/messages after running ssinit. The driver prints
+the board version after downloading the microcode file. That version
+number must match the number in the name of the microcode file (extension).
+
+Running ssinit with a wrong version of the sndscape.co? file is not
+dangerous as long as you don't try to use a file called sndscape.cod.
+If you have initialized the card using a wrong microcode file (sounds
+are terrible), just modify ssinit.c to use another microcode file and try
+again. It's possible to use an earlier version of sndscape.co[01] but it
+may sound wierd.
+
+Btw, The driver may complain something about "sscapeintr()" after
+running ssinit. You should just ignore these messages.
+
+MAD16 (Pro) and Mozart
+----------------------
+
+You need to enable just the MAD16 /Mozart support when configuring
+the driver. _Don't_ enable SB, MPU401 or MSS. However you will need the
+/dev/audio, /dev/sequencer and MIDI supports.
+
+Mozart and OPTi 82C928 (the original MAD16) chips don't support
+MPU401 mode so enter just 0 when the configuration program asks the
+MPU/MIDI I/O base. The MAD16 Pro (OPTi 82C929) has MPU401 mode.
+
+TB Tropez is based on the 82C929 chip. It has two MIDI ports.
+The one connected to the MAD16 chip is the second one (there is a second
+MIDI connector/pins somewhere??). If you have not connected the second MIDI
+port, just disable the MIDI port of MAD16. The 'Maui' compatible synth of
+Tropez is jumper configurable and not connected to the MAD16 chip.
+It can be used by enabling the stand alone MPU401 support but you have
+to initialize it by using the MS-DOS SNDSETUP program.
+
+There are some other OPTi chips which may be used in soundcards such as
+82C930 and MAC32. These chips are not supported by VoxWare yet. Please
+contact me if you have a soundcard which uses these chips.
+
+Some MAD16 based cards may cause feedback, whistle or terrible noise if the
+line3 mixer channel is turned too high.
+
+If you have a MAD16 card which have an OPL4 (FM + Wave table) synthesizer
+chip (_not_ an OPL3), you have to apped line containing #define MAD16_OPL4
+to the file linux/dirvers/sound/local.h (after running make config).
+
+MV Jazz (ProSonic)
+------------------
+
+The Jazz16 driver is just a hack made to the SB Pro driver. However it works
+fairly well. You have to enable SB, SB Pro (_not_ SB16) and MPU401 supports
+when configuring the driver. The configuration program asks later if you
+want support for MV Jazz16 based cards (after asking SB base address). Answer
+'y' here and the driver asks the second (16 bit) DMA channel.
+
+The Jazz16 driver uses the MPU401 driver in a way which will cause
+problems if you have another MPU401 compatible card. In this case you must
+give address of the Jazz16 based MPU401 interface when the config
+program prompts for the MPU401 information. Then look at the MPU401
+spesific section for instructions about configuring more than one MPU401 cards.
+
+Logitech Soundman Wave
+----------------------
+
+Read the above MV Jazz spesific instructions first.
+
+The Logitech SoundMan Wave (don't confuse with the SM16 or SM Games) is
+a MV Jazz based card which has an additional OPL4 based wave table
+synthesizer. The OPL4 chip is handled by an on board microcontroller
+which must be initialized during boot. The config program asks if
+you have a SM Wave immediately after asking the second DMA channel of jazz16.
+If you answer 'y', the config program will ask name of the file containing
+code to be loaded to the microcontroller. The file is usually called
+MIDI0001.BIN and it's located in the DOS/Windows driver directory. The file
+may also be called as TSUNAMI.BIN or something else (older cards?).
+
+The OPL4 synth will be inaccessible without loading the microcontroller code.
+Also remember to enable MPU401 support if you want to use the OPL4 mode.
+
+NOTE!	Don't answer 'y' when the driver asks about SM Games support
+	(the next question after the MIDI0001.BIN name). However
+	aneswering 'y' is not dangerous. 
+
+Sound Galaxies
+--------------
+
+There are many different Sound Galaxy cards made by Aztech. The 8 bit
+ones are fully SB or SB Pro compatible and there should be no problems
+with them. 
+
+The older 16 bit cards (SG Pro16, SG NX Pro16, Nova and Lyra) have
+an EEPROM chip for storing the configuration data. There is a microcontroller
+which initializes the card to match the EEPROM settigs when the machine
+is powered on. These cards actually behave just like they have jumpers
+for all of the settings. Configure VoxWare for MSS, MPU, SB/SB Pro  and OPL3 
+supports with these cards. 
+
+The config program asks if you want support for the mixer of
+SG NX Pro. Answer 'y' to these questions if you have one of the above 8 or
+16 bit Aztech cards.
+
+There are some new Sound Galaxies in the market. I have no experience with
+them so read the card's manual carefully.
+
+
+Reveal cards
+------------
+
+There are several different cards made/marketed by Reveal. Some of them
+are compatible with SoundScape and some use the MAD16 chip. You may have
+to look at the card and try to identify origin of the card.
+
+Diamond
+-------
+
+The oldest (Sierra Aria based) soundcards made by Diamond are not supported
+(they may work if the card is initialized using DOS). The recent (LX?)
+models are based on the MAD16 chip which is supported by VoxWare.
+
+Audio Excel DSP16
+-----------------
+
+See comments in aedsp16.c.
+
+
+PCMCIA cards
+------------
+
+Sorry, can't help. Some cards may work and some don't.
+
+TI TM4000M notebooks
+--------------------
+
+These computers have a built in sound support based on the Jazz chipset.
+Look at the instructions for MV Jazz (above). It's also important to note
+that there is something wrong with the mouse port and sound at least on
+some TM models. Don't enable the "C&T 82C710 mouse port support" when
+configuring Linux. Having it enabled is likely to cause mysterious problems
+and kernel failures when sound is used.
+
+Others?
+-------
+
+Since there are so many different soundcards, it's likely that I have 
+forgotten to mention many of them. Please inform me if you know yet another
+card which works with Linux, please inform me (or is anybody else
+willing to maintain a database of supported cards (just like in XF86)?).
+
+Cards not supported yet
+=======================
+
+First of all. There is an easy way to make most soundcards to work
+with Linux. Just use the DOS based driver to initialize the card
+to a _known_ state. Then ctrl-alt-del to Linux. If Linux is configured
+to use the sama I/O, IRQ and DMA numbers than DOS, the card could work.
+
+Don't get fooled with SB compatibility. Most cards are compatible with
+SB but that may require a TSR which is not possible with Linux. If
+the card is compatible with MSS, it's a better choise. Some cards
+don't work in the SB and MSS modes at the same time.
+
+There are some cards which will be supported by VoxWare sooner or later
+(currently at least cards based on the ESS chipset). Such cards are
+so common that there is some idea in writing the driver. Check the
+VoxWare home page (http://personal.eunet.fi/pp/voxware) for latest
+information.
+
+Then there are cards which are no longer manufactured and/or which
+are relatively rarely used (such as the 8 bit ProAudioSpectrum
+models). It's extremely unlikely that such cards never get supported.
+Adding support for a new card requires much work and increases time
+required in maintaining the driver (some changes need to be done
+to all low level drivers and be tested too, maybe with multiple
+operating systems). For this reason I have made a desicion to not support
+obsolete cards. It's possible that someone else makes a separately 
+distributed driver (diffs) for the card. Version v4.0 will be much more
+modular so making separately distributed drivers will be easier with it.
+(The bad news is that v4.0 will not be available before late -96).
+
+Writing a driver for a new card is not possible if there are no 
+programming information available about the card. If you don't
+find your new card from this file, look from the home page 
+(http://personal.eunet.fi/pp/voxware). Then please contact
+manufacturer of the card and ask if they have (or are willing to)
+released technical details of the card. Do this before contacting me. I
+can only answer 'no' if there are no programming information available.
+
+Some companies don't give low level technical information about their
+products to public or at least their require signing a NDA.
+
+I have also made decicion to not accept code based on reverse engineering
+to VoxWare. There are three main reasons: First I don't want to break
+relationships to sound card manufacturers. The second reason is that
+maintaining and supporting a driver withoun any specs will be a pain. The
+third reason is that why shoud we help such companies in selling their
+products to Linux users when they don't want to sell to Linux users
+at all?
+
+Unfortunately many of the leading soundcard manufacturers are not willing
+to co-operate with Linux/Unix community. For example: Creative Technology 
+doesn't give information about the ASP chip and the Emu synth chip of AWE32
+and SB32. Turtle Beach don't give information about any of their
+products. MediaVision requires NDA before they are willing to
+give information about the Jazz16 chip (fortunately Logitech gave
+the info about SM Wave).
+
+So at least the above three companies are out until they are willing to 
+release documentation about their products (the situation is the
+same with many DOS based freeware/shareware games and utilities). If
+you want to use Linux/Unix with their cards, please don't try to push
+me. It's a better idea to contact the manufacturer and explain that
+you want to use your card with Linux/Unix. You could also try to sell
+your card to somebody else and then buy a card that is supported by VoxWare.
+
+However it's possible that things change and a driver gets written
+for some of the banned cards. Please, don't send me messages asking if
+there is any plans to write a driver for the cards mentioned above. I 
+will put any news to the VoxWare www home page (see below).
+
+There are some common audio chipsets that are supported yet. For example
+the ESS chips and Sierra Aria. It's likely that these architectures
+get some support in future but I can't make any promises. Just look
+at the home page for latest info.
+
+Information about unsupported soundcards and chipsets is welcome as well
+as free copies of soundcards, SDKs and operating systems.
+
+If you have any corrections and/or comments, please contact me.
+
+Hannu Savolainen
+hannu@voxware.pp.fi
+VoxWare www home page: http://personal.eunet.fi/pp/voxware
+
diff --git a/sys/i386/isa/sound/Readme.modules b/sys/i386/isa/sound/Readme.modules
new file mode 100644
index 000000000000..315540f510c1
--- /dev/null
+++ b/sys/i386/isa/sound/Readme.modules
@@ -0,0 +1,87 @@
+			Linux sound-driver module
+			(c) Peter Trattler
+			License: GPL (Gnu Public License)
+
+
+Idea:
+
+I've modified the sources for the sound driver to allow simply insert and
+remove the sound driver from the kernel by calling (only available for Linux)
+
+	insmod /usr/src/linux/modules/sound.o
+
+and
+
+	rmmod sound
+
+This may be useful if you are doing one of the following things:
+
+1) Debugging the sound driver
+2) Creating a new device within the sound-driver
+3) You do not the sound driver all the time (as it wastes quite a lot of
+memory for its buffers)
+
+
+Compilation:
+
+Go to /usr/src/linux and make the following steps:
+
+a) configure the sound driver: To do that call "make config" and enable the
+sound-driver -- you will be asked different questions about your
+sound-hardware (remember not to use a too big DMA-Buffer size; you
+should use 16kB, if you have 16Bit devices, otherwise you can use 32kB)
+
+b) disable the sound driver in the kernel: call make config again but answer
+'N' to "Sound card support"
+
+c) run "make modules"; the sound-driver sound.o should end up in
+/usr/src/linux/modules
+
+
+If memory is tight:
+
+I've allocated at about 70kB for the sound-drivers internal tables. If this
+is too much, 'insmod sound.o' will generate the following warning
+...
+use 'insmod memsize=xxxx'
+...
+You can only use this command, if you have (I think) at least
+modules-1.1.87 or up. You can also switch debugging on by running the command
+
+insmod sound.o debugmem=1
+
+
+Files I changed:
+
+I've only changed the files soundcard.c(most changes) and some changes within
+the Makefile, sound_config.h and the Makefile in /usr/src/linux/drivers
+
+
+Bugs:
+
+a) As the kmalloc (..., GFP_DMA) caused some unexpected errors (I don't know if
+it is my fault), I created some code, which is (by default) enabled by
+
+#define KMALLOC_DMA_BROKEN 1 (within soundcard.c).
+
+It trys to allocate a large enough region, so that the complete dma-buffer
+can be occupied in this space. If it does not fit within this region it
+doubles the size of it. But this can cause problems, if the sound-buffer is
+too big (as kmalloc can only handle regions at up to circa 100kB).
+
+So take care to use for 8Bit devices a sound-DMA-buffer of 32kB (maximum)
+and for 16Bit devices a maximum of 16kB. Otherwise the allocation scheme
+might fail.
+
+b) Buffers allocated by the different sound devices via calls to kmalloc are
+not freed, if the sound driver is removed again (these buffers tend to be
+quite small -- so it does not harm a lot)
+
+c) If there is not enough (kernel-) memory available, the installation of
+the sound-driver fails. (This happens quite often, if you did not install the
+driver right after booting -- [PS: I've only got 5MB of Ram, so this might
+be the source for this problem])
+
+
+Author:
+	Peter Trattler (peter@sbox.tu-graz.ac.at)
diff --git a/sys/i386/isa/sound/Readme.v30 b/sys/i386/isa/sound/Readme.v30
new file mode 100644
index 000000000000..cade461d2a6d
--- /dev/null
+++ b/sys/i386/isa/sound/Readme.v30
@@ -0,0 +1,154 @@
+VoxWare v3.0
+------------
+
+This is a PROTOTYPE of the VoxWare v3.0 to be released late 94.
+
+All features of v2.5 should work as earlier. There could be some
+omissions but they are unintentional. I started this version thread
+after v2.3 so all features implemented before it are there.
+
+Even this is a prototype, there should not be any fatal bugs. The
+prototype just means that I don't have implemented all features
+completely. Mainly in the /dev/sequencer2 driver.
+For example recording from /dev/sequencer2 won't work
+with other cards than a full features MPU-401 or clones. As well the
+way how the MIDI controllers are handled will change.
+
+IMPORTANT!!!!!!!!!!!!!!!!!
+
+Don't distribute any binaries compiled with the soundcard.h of this version. 
+They will not work together with older drivers.
+
+New features
+============
+
+There are now two new device interfaces. The /dev/midi## is a raw
+tty like interface to MIDI ports. There is a device file for each MIDI
+port on your system. They are named (/dev/midi00 to /dev/midiNN).
+The second addition is the /dev/sequencer2 which is higher level interface 
+than the old /dev/sequencer. It's intended for writing device independent
+applications like sequencers.
+
+/dev/midi##
+-----------
+
+This interface should be useful for applications like MIDI sysex librarians.
+There are (currently) no timing features so making music could be impossible.
+
+There are as many /dev/midi## devices as there are MIDI ports in the system. 
+The /dev/midi00 is connected to the first one, /dev/midi01 to the second etc.
+
+These devices work like tty devices in raw mode. Everything written to them is
+sent out to the MIDI port. There is currently an extra delay of at most 
+1/100th of sec but it will be removed later.
+
+The reading algorithm is little bit more complicated. There are two different
+cases:
+
+1)	There is at least one byte in the input buffer.
+
+The read returns as many bytes as it can without waiting for more bytes.
+For example when a process reads 100 bytes and there are 10 bytes in the 
+buffer, the read returns just 10 bytes.
+
+2)	The input buffer is empty when the process calls read.
+
+The read waits for the first byte and then continues as in case 1. By
+default it waits infinitely but there is an ioctl for setting a timeout
+for this. The ioctl(fd, SNDCTL_MIDI_PRETIME, &time) changes the timeout.
+The time is given in 1/10th of seconds (10 means one second).
+
+Other ioctl calls:
+
+ioctl(fd, SNDCTL_MIDI_MPUMODE, &mode) is available for full MPU-401
+compatible devices such as MPU-IPC-T, MQ PC Midi Card or MQX-32.
+It's not available for the so called MPU UART ports of some soundcards
+(PAS16, SB16 etc). By default the MIDI port is in UART mode after open. 
+If this ioctl is called with mode=1, the interface is put to the intelligent
+(coprocessor) mode. NOTE! The MIDI port will be reset when this ioctl is called.
+It could have some strange effects if not called immediately after open. This
+call returns EINVAL if the midi port doesn't support the MPU-401 intelligent
+mode.
+
+ioctl(fd, SNDCTL_MIDI_MPUCMD, &cmdstruct) is valid only if the MIDI port
+is put to the coprocessor mode using ioctl(SNDCTL_MIDI_MPUMODE). It's used to
+send commands to a MPU-401 compatible MIDI cards. Please refer to the
+MPU-401 Technical Reference Manual (or Music Quest Technical Reference
+Manual) for descriptions of the commands.
+
+The argument of SNDCTL_MIDI_MPUCOMMAND is of type mpu_command_rec. It
+has the following fields:
+
+typedef struct {
+		unsigned char cmd;
+
+		char nr_args, nr_returns;
+		unsigned char data[30];
+	} mpu_command_rec;
+
+where:
+	cmd		Contains the command number.
+	nr_args		Number of arguments of the command.
+			MUST BE INITIALIZED BEFORE CALL
+	nr_returns	Number of bytes returned by the command.
+			MUST BE INITIALIZED BEFORE CALL
+	data		Buffer for the command arguments and returned
+			data.
+
+Be extremely careful with the nr_args and nr_returns fields. They
+must match the command. An incorrect value will put the card and
+the driver out of sync. Refer to the MPU-401/MQX-32M documentation for further
+details.
+
+
+
+/dev/sequencer2 (if you find a better name, please let me know).
+---------------
+
+This device file works much like the /dev/sequencer which has been present
+since the beginning. The main differences are the following:
+
+- /dev/sequencer makes the MIDI ports to look like the synth devices. In fact
+the result is somewhere between the MIDI specification and the synth devices of
+/dev/sequencer. Both kind of devices are accessed using the SEQ_START_NOTE()
+like macros. The voice number parameters of the API macros have been redefined
+to denote MIDI channels. This means that the driver allocates voices for
+the channels automatically (this is a responsibility/right of an application
+with /dev/sequencer). The result is that a SEQ_START_NOTE() macro has
+similar effects for a synth channel than on a MIDI port. This kind of
+solution provides better device independence than the /dev/sequencer. The
+drawback is that the new interface doesn't permit so low level access to the
+device as the /dev/sequencer does. An application developer must choose between
+these two interfaces. I think the old /dev/sequencer is better for applications
+like module players while the new one is better for making generic sequencer
+programs.
+
+- There are no separate MIDI devices with the /dev/sequencer2. The
+ioctl(SNDCTL_SEQ_NRMIDIS) returns always zero. Instead the MIDI ports are
+shown as synth devices. ioctl(SNDCTL_SEQ_NRSYNTHS) on /dev/sequencer2 will 
+return sum of internal synthesizers (GUS, OPL3) and MIDI ports in the systems.
+
+- The new interface is used much like the ordinary /dev/sequencer. The
+event format is new so you have to use the API macros defined in the 
+sys/soundcard.h. The interface will probably change before the final 3.0
+release but using the API macros should ensure compatibility in source level.
+The new event format is not recognized by version 2.X so don't try to 
+distribute binaries compiled with soundcard.h of v3.X.
+
+- The basic API usage is similar to the current one. There are some new
+macros but the older ones should work as earlier. The most important
+incompatibility is that the /dev/sequencer2 driver allocates voices itself.
+The other one is that the application must send SEQ_START_TIMER() as it's
+first event. Otherwise the timer is not started and the application waits
+infinitely.
+
+
+There are several new features but I don't document them here. There are
+some info in the soundcard.h (near the end). I have also included some
+sample code in the directory v30. Full documentation will
+appear in the Hacker's Guide later.
+
+Don't hesitate to contact me in case you have questions or comments.
+
+Hannu Savolainen
+hannu@voxware.pp.fi
diff --git a/sys/i386/isa/sound/ad1848_mixer.h b/sys/i386/isa/sound/ad1848_mixer.h
new file mode 100644
index 000000000000..9404047ce77d
--- /dev/null
+++ b/sys/i386/isa/sound/ad1848_mixer.h
@@ -0,0 +1,130 @@
+/*
+ * sound/ad1848_mixer.h
+ * 
+ * Definitions for the mixer of AD1848 and compatible codecs.
+ * 
+ * Copyright by Hannu Savolainen 1994
+ * 
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are
+ * met: 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer. 2.
+ * Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED.  IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE FOR
+ * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+ * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
+ * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ */
+/*
+ * The AD1848 codec has generic input lines called Line, Aux1 and Aux2.
+ * Soundcard manufacturers have connected actual inputs (CD, synth, line,
+ * etc) to these inputs in different order. Therefore it's difficult
+ * to assign mixer channels to to these inputs correctly. The following
+ * contains two alternative mappings. The first one is for GUS MAX and
+ * the second is just a generic one (line1, line2 and line3).
+ * (Actually this is not a mapping but rather some kind of interleaving
+ * solution).
+ */
+#ifdef GUSMAX_MIXER
+#define MODE1_REC_DEVICES		(SOUND_MASK_LINE | SOUND_MASK_MIC | \
+					 SOUND_MASK_CD)
+
+#define MODE1_MIXER_DEVICES		(SOUND_MASK_SYNTH | SOUND_MASK_MIC | \
+					 SOUND_MASK_CD | \
+					 SOUND_MASK_IGAIN | \
+					 SOUND_MASK_PCM|SOUND_MASK_IMIX)
+
+#define MODE2_MIXER_DEVICES		(SOUND_MASK_SYNTH | SOUND_MASK_LINE | SOUND_MASK_MIC | \
+					 SOUND_MASK_CD | SOUND_MASK_SPEAKER | \
+					 SOUND_MASK_IGAIN | \
+					 SOUND_MASK_PCM | SOUND_MASK_IMIX)
+#else	/* Generic mapping */
+#define MODE1_REC_DEVICES		(SOUND_MASK_LINE3 | SOUND_MASK_MIC | \
+					 SOUND_MASK_LINE1)
+
+#define MODE1_MIXER_DEVICES		(SOUND_MASK_LINE1 | SOUND_MASK_MIC | \
+					 SOUND_MASK_LINE2 | \
+					 SOUND_MASK_IGAIN | \
+					 SOUND_MASK_PCM | SOUND_MASK_IMIX)
+
+#define MODE2_MIXER_DEVICES		(SOUND_MASK_LINE1 | SOUND_MASK_LINE2 | SOUND_MASK_MIC | \
+					 SOUND_MASK_LINE3 | SOUND_MASK_SPEAKER | \
+					 SOUND_MASK_IGAIN | \
+					 SOUND_MASK_PCM | SOUND_MASK_IMIX)
+#endif
+
+struct mixer_def {
+	unsigned int regno: 7;
+	unsigned int polarity:1;	/* 0=normal, 1=reversed */
+	unsigned int bitpos:4;
+	unsigned int nbits:4;
+};
+
+
+typedef struct mixer_def mixer_ent;
+
+/*
+ * Most of the mixer entries work in backwards. Setting the polarity field
+ * makes them to work correctly.
+ *
+ * The channel numbering used by individual soundcards is not fixed. Some
+ * cards have assigned different meanings for the AUX1, AUX2 and LINE inputs.
+ * The current version doesn't try to compensate this.
+ */
+
+#define MIX_ENT(name, reg_l, pola_l, pos_l, len_l, reg_r, pola_r, pos_r, len_r)	\
+	{{reg_l, pola_l, pos_r, len_l}, {reg_r, pola_r, pos_r, len_r}}
+
+mixer_ent mix_devices[32][2] = {	/* As used in GUS MAX */
+MIX_ENT(SOUND_MIXER_VOLUME,	 0, 0, 0, 0,	 0, 0, 0, 0),
+MIX_ENT(SOUND_MIXER_BASS,	 0, 0, 0, 0,	 0, 0, 0, 0),
+MIX_ENT(SOUND_MIXER_TREBLE,	 0, 0, 0, 0,	 0, 0, 0, 0),
+MIX_ENT(SOUND_MIXER_SYNTH,	 4, 1, 0, 5,	 5, 1, 0, 5),
+MIX_ENT(SOUND_MIXER_PCM,	 6, 1, 0, 6,	 7, 1, 0, 6),
+MIX_ENT(SOUND_MIXER_SPEAKER,	26, 1, 0, 4,	 0, 0, 0, 0),
+MIX_ENT(SOUND_MIXER_LINE,	18, 1, 0, 5,	19, 1, 0, 5),
+MIX_ENT(SOUND_MIXER_MIC,	 0, 1, 5, 1,	 1, 1, 5, 1),
+MIX_ENT(SOUND_MIXER_CD,	 	 2, 1, 0, 5,	 3, 1, 0, 5),
+MIX_ENT(SOUND_MIXER_IMIX,	13, 1, 2, 6,	 0, 0, 0, 0),
+MIX_ENT(SOUND_MIXER_ALTPCM,	 0, 0, 0, 0,	 0, 0, 0, 0),
+MIX_ENT(SOUND_MIXER_RECLEV,	 0, 0, 0, 0,	 0, 0, 0, 0),
+MIX_ENT(SOUND_MIXER_IGAIN,	 0, 0, 0, 4,	 1, 0, 0, 4),
+MIX_ENT(SOUND_MIXER_OGAIN,	 0, 0, 0, 0,	 0, 0, 0, 0),
+MIX_ENT(SOUND_MIXER_LINE1, 	 2, 1, 0, 5,	 3, 1, 0, 5),
+MIX_ENT(SOUND_MIXER_LINE2,	 4, 1, 0, 5,	 5, 1, 0, 5),
+MIX_ENT(SOUND_MIXER_LINE3,	18, 1, 0, 5,	19, 1, 0, 5)
+};
+
+static unsigned short default_mixer_levels[SOUND_MIXER_NRDEVICES] =
+{
+  0x5a5a,			/* Master Volume */
+  0x3232,			/* Bass */
+  0x3232,			/* Treble */
+  0x4b4b,			/* FM */
+  0x6464,			/* PCM */
+  0x4b4b,			/* PC Speaker */
+  0x4b4b,			/* Ext Line */
+  0x1010,			/* Mic */
+  0x4b4b,			/* CD */
+  0x0000,			/* Recording monitor */
+  0x4b4b,			/* SB PCM */
+  0x4b4b,			/* Recording level */
+  0x4b4b,			/* Input gain */
+  0x4b4b,			/* Output gain */
+  0x4b4b,			/* Line1 */
+  0x4b4b,			/* Line2 */
+  0x4b4b			/* Line3 */
+};
+
+#define LEFT_CHN	0
+#define RIGHT_CHN	1
diff --git a/sys/i386/isa/sound/aedsp16.c b/sys/i386/isa/sound/aedsp16.c
new file mode 100644
index 000000000000..b14a24618f0c
--- /dev/null
+++ b/sys/i386/isa/sound/aedsp16.c
@@ -0,0 +1,838 @@
+/*
+   sound/aedsp16.c
+
+   Audio Excel DSP 16 software configuration routines
+
+   Copyright (C) 1995  Riccardo Facchetti (riccardo@cdc8g5.cdc.polimi.it)
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions are
+   met: 1. Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer. 2.
+   Redistributions in binary form must reproduce the above copyright notice,
+   this list of conditions and the following disclaimer in the documentation
+   and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND ANY
+   EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+   WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+   DISCLAIMED.  IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE FOR
+   ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+   DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+   SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
+   CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+   LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+   OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+   SUCH DAMAGE.
+
+   READ THIS
+
+   This module is intended for Audio Excel DSP 16 Sound Card.
+
+   Audio Excel DSP 16 is an SB pro II, Microsoft Sound System
+   and MPU-401 compatible card.
+   It is software-only configurable (no jumpers to hard-set irq/dma/mpu-irq),
+   so before this module, the only way to configure the DSP under linux was
+   boot the MS-BAU loading the sound.sys device driver (this driver soft-
+   configure the sound board hardware by massaging someone of its registers),
+   and then ctrl-alt-del to boot linux with the DSP configured by the DOG
+   driver.
+
+   This module works configuring your Audio Excel DSP 16's
+   irq, dma and mpu-401-irq. The voxware probe routines rely on the
+   fact that if the hardware is there, they can detect it. The problem
+   with AEDSP16 is that no hardware can be found by the probe routines
+   if the sound card is not well configured. Sometimes the kernel probe
+   routines can find an SBPRO even when the card is not configured (this
+   is the standard setup of the card), but the SBPRO emulation don't work
+   well if the card is not properly initialized. For this reason
+
+   InitAEDSP16_...()
+
+   routines are called before the voxware probe routines try to detect the
+   hardware.
+
+   NOTE (READ THE NOTE TOO, IT CONTAIN USEFUL INFORMATIONS)
+
+   The Audio Excel DSP 16 Sound Card emulates both SBPRO and MSS;
+   the voxware sound driver can be configured for SBPRO and MSS cards
+   at the same time, but the aedsp16 can't be two cards!!
+   When we configure it, we have to choose the SBPRO or the MSS emulation
+   for AEDSP16. We also can install a *REAL* card of the other type
+   (see [1], not tested but I can't see any reason for it to fail).
+
+   NOTE: If someone can test the combination AEDSP16+MSS or AEDSP16+SBPRO
+   please let me know if it works.
+
+   The MPU-401 support can be compiled in together with one of the other
+   two operating modes.
+
+   The board configuration calls, are in the probe_...() routines because
+   we have to configure the board before probing it for a particular
+   hardware. After card configuration, we can probe the hardware.
+
+   NOTE: This is something like plug-and-play: we have only to plug
+   the AEDSP16 board in the socket, and then configure and compile
+   a kernel that uses the AEDSP16 software configuration capability.
+   No jumper setting is needed!
+
+   For example, if you want AEDSP16 to be an SBPro, on irq 10, dma 3
+   you have just to make config the voxware package, configuring
+   the SBPro sound card with that parameters, then when configure
+   asks if you have an AEDSP16, answer yes. That's it.
+   Compile the kernel and run it.
+
+   NOTE: This means that you can choose irq and dma, but not the
+   I/O addresses. To change I/O addresses you have to set them
+   with jumpers.
+
+   NOTE: InitAEDSP16_...() routines get as parameter the hw_config,
+   the hardware configuration of the - to be configured - board.
+   The InitAEDSP16() routine, configure the board following our
+   wishes, that are in the hw_config structure.
+
+   You can change the irq/dma/mirq settings WITHOUT THE NEED to open
+   your computer and massage the jumpers (there are no irq/dma/mirq
+   jumpers to be configured anyway, only I/O port ones have to be
+   configured with jumpers)
+
+   For some ununderstandable reason, the card default of irq 7, dma 1,
+   don't work for me. Seems to be an IRQ or DMA conflict. Under heavy
+   HDD work, the kernel start to erupt out a lot of messages like:
+
+   'Sound: DMA timed out - IRQ/DRQ config error?'
+
+   For what I can say, I have NOT any conflict at irq 7 (under linux I'm
+   using the lp polling driver), and dma line 1 is unused as stated by
+   /proc/dma. I can suppose this is a bug of AEDSP16. I know my hardware so
+   I'm pretty sure I have not any conflict, but may be I'm wrong. Who knows!
+   Anyway a setting of irq 10, dma 3 works really fine.
+
+   NOTE: if someone can use AEDSP16 with irq 7, dma 1, please let me know
+   the emulation mode, all the installed hardware and the hardware
+   configuration (irq and dma settings of all the hardware).
+
+   This init module should work with SBPRO+MSS, when one of the two is
+   the AEDSP16 emulation and the other the real card. (see [1])
+   For example:
+
+   AEDSP16 (0x220) in SBPRO emu (0x220) + real MSS + other
+   AEDSP16 (0x220) in MSS emu + real SBPRO (0x240) + other
+
+   MPU401 should work. (see [1])
+
+   [1] Not tested by me for lack of hardware.
+
+   TODO, WISHES AND TECH
+
+   May be there's lot of redundant delays, but for now I want to leave it
+   this way.
+
+   Should be interesting eventually write down a new ioctl for the
+   aedsp16, to let the suser() change the irq/dma/mirq on the fly.
+   The thing is not trivial.
+   In the real world, there's no need to have such an ioctl because
+   when we configure the kernel for compile, we can choose the config
+   parameters. If we change our mind, we can easily re-config the kernel
+   and re-compile.
+   Why let the suser() change the config parameters on the fly ?
+   If anyone have a reasonable answer to this question, I will write down
+   the code to do it.
+
+   More integration with voxware, using voxware low level routines to
+   read-write dsp is not possible because you may want to have MSS
+   support and in that case we can not rely on the functions included
+   in sb_dsp.c to control 0x2yy I/O ports. I will continue to use my
+   own I/O functions.
+
+   - About I/O ports allocation -
+
+   The request_region should be done at device probe in every sound card
+   module. This module is not the best site for requesting regions.
+   When the request_region code will be added to the main modules such as
+   sb, adlib, gus, ad1848, etc, the requesting code in this module should
+   go away.
+
+   I think the request regions should be done this way:
+
+   if (check_region(...))
+   return ERR; // I/O region alredy reserved
+   device_probe(...);
+   device_attach(...);
+   request_region(...); // reserve only when we are sure all is okay
+
+   Request the 2x0h region in any case if we are using this card.
+
+   NOTE: the "(sbpro)" string with which we are requesting the aedsp16 region
+   (see code) does not mean necessarly that we are emulating sbpro.
+   It mean that the region is the sbpro I/O ports region. We use this
+   region to access the control registers of the card, and if emulating
+   sbpro, I/O sbpro registers too. If we are emulating MSS, the sbpro
+   registers are not used, in no way, to emulate an sbpro: they are
+   used only for configuration pourposes.
+
+   Someone pointed out that should be possible use both the SBPRO and MSS
+   modes because the sound card have all the two chipsets, supposing that
+   the card is really two cards. I have tried something to have the two
+   modes work together, but, for some reason unknown to me, without success.
+
+   I think all the soft-config only cards have an init sequence similar to
+   this. If you have a card that is not an aedsp16, you can try to start
+   with this module changing it (mainly in the CMD? I think) to fit your
+   needs.
+
+   Started Fri Mar 17 16:13:18 MET 1995
+
+   v0.1 (ALPHA, was an user-level program called AudioExcelDSP16.c)
+   - Initial code.
+   v0.2 (ALPHA)
+   - Cleanups.
+   - Integrated with Linux voxware v 2.90-2 kernel sound driver.
+   - SoundBlaster Pro mode configuration.
+   - Microsoft Sound System mode configuration.
+   - MPU-401 mode configuration.
+   v0.3 (ALPHA)
+   - Cleanups.
+   - Rearranged the code to let InitAEDSP16 be more general.
+   - Erased the REALLY_SLOW_IO. We don't need it. Erased the linux/io.h
+   inclusion too. We rely on os.h
+   - Used the INB and OUTB #defined in os.h instead of inb and outb.
+   - Corrected the code for GetCardName (DSP Copyright) to get a variable
+   len string (we are not sure about the len of Copyright string).
+   This works with any SB and compatible.
+   - Added the code to request_region at device init (should go in
+   the main body of voxware).
+   v0.4 (BETA)
+   - Better configure.c patch for aedsp16 configuration (better
+   logic of inclusion of AEDSP16 support)
+   - Modified the conditional compilation to better support more than
+   one sound card of the emulated type (read the NOTES above)
+   - Moved the sb init routine from the attach to the very first
+   probe in sb_card.c
+   - Rearrangemens and cleanups
+   - Wiped out some unnecessary code and variables: this is kernel
+   code so it is better save some TEXT and DATA
+   - Fixed the request_region code. We must allocate the aedsp16 (sbpro)
+   I/O ports in any case because they are used to access the DSP
+   configuration registers and we can not allow anyone to get them.
+   v0.5
+   - cleanups on comments
+   - prep for diffs against v3.0-proto-950402
+
+ */
+
+/*
+ * Include the main voxware header file. It include all the os/voxware/etc
+ * headers needed by this source.
+ */
+#include "sound_config.h"
+/*
+ * all but ioport.h :)
+ */
+#include <linux/ioport.h>
+
+#if defined(CONFIGURE_SOUNDCARD) && !defined(EXCLUDE_AEDSP16)
+
+#define VERSION "0.5"		/* Version of Audio Excel DSP 16 driver */
+
+#undef AEDSP16_DEBUG		/* Define this to enable debug code     */
+/* Actually no debug code is activated  */
+
+/*
+ * Hardware related defaults
+ */
+#define IRQ  7			/* 5 7(default) 9 10 11                 */
+#define MIRQ 0			/* 5 7 9 10 0(default), 0 means disable */
+#define DMA  1			/* 0 1(default) 3                       */
+
+/*
+ * Commands of AEDSP16's DSP (SBPRO+special).
+ * For now they are CMDn, in the future they may change.
+ */
+#define CMD1 0xe3		/* Get DSP Copyright                    */
+#define CMD2 0xe1		/* Get DSP Version                      */
+#define CMD3 0x88		/*                                      */
+#define CMD4 0x5c		/*                                      */
+#define CMD5 0x50		/* Set M&I&DRQ mask (the real config)   */
+#define CMD6 0x8c		/* Enable Microsoft Sound System mode   */
+
+/*
+ * Offsets of AEDSP16 DSP I/O ports. The offest is added to portbase
+ * to have the actual I/O port.
+ * Register permissions are:
+ * (wo) == Write Only
+ * (ro) == Read  Only
+ * (w-) == Write
+ * (r-) == Read
+ */
+#define DSP_RESET    0x06	/* offset of DSP RESET             (wo) */
+#define DSP_READ     0x0a	/* offset of DSP READ              (ro) */
+#define DSP_WRITE    0x0c	/* offset of DSP WRITE             (w-) */
+#define DSP_COMMAND  0x0c	/* offset of DSP COMMAND           (w-) */
+#define DSP_STATUS   0x0c	/* offset of DSP STATUS            (r-) */
+#define DSP_DATAVAIL 0x0e	/* offset of DSP DATA AVAILABLE    (ro) */
+
+
+#define RETRY           10	/* Various retry values on I/O opera-   */
+#define STATUSRETRY   1000	/* tions. Sometimes we have to          */
+#define HARDRETRY   500000	/* wait for previous cmd to complete    */
+
+/*
+ * Size of character arrays that store name and version of sound card
+ */
+#define CARDNAMELEN 15		/* Size of the card's name in chars     */
+#define CARDVERLEN  2		/* Size of the card's version in chars  */
+
+/*
+ * Bit mapped flags for calling InitAEDSP16(), and saving the current
+ * emulation mode.
+ */
+#define INIT_NONE   (0   )
+#define INIT_SBPRO  (1<<0)
+#define INIT_MSS    (1<<1)
+#define INIT_MPU401 (1<<2)
+#define RESET_DSP16 (1<<3)
+
+/* Base HW Port for Audio Card          */
+static int      portbase = AEDSP16_BASE;
+static int      irq = IRQ;	/* irq for DSP I/O                      */
+static int      mirq = MIRQ;	/* irq for MPU-401 I/O                  */
+static int      dma = DMA;	/* dma for DSP I/O                      */
+
+/* Init status of the card              */
+static int      ae_init = INIT_NONE;	/* (bitmapped variable)                 */
+static int      oredparams = 0;	/* Will contain or'ed values of params  */
+static int      gc = 0;		/* generic counter (utility counter)    */
+struct orVals
+  {				/* Contain the values to be or'ed       */
+    int             val;	/* irq|mirq|dma                         */
+    int             or;		/* oredparams |= TheStruct.or           */
+  };
+
+/*
+ * Magic values that the DSP will eat when configuring irq/mirq/dma
+ */
+/* DSP IRQ conversion array             */
+static struct orVals orIRQ[] =
+{
+  {0x05, 0x28},
+  {0x07, 0x08},
+  {0x09, 0x10},
+  {0x0a, 0x18},
+  {0x0b, 0x20},
+  {0x00, 0x00}
+};
+
+/* MPU-401 IRQ conversion array         */
+static struct orVals orMIRQ[] =
+{
+  {0x05, 0x04},
+  {0x07, 0x44},
+  {0x09, 0x84},
+  {0x0a, 0xc4},
+  {0x00, 0x00}
+};
+
+/* DMA Channels conversion array        */
+static struct orVals orDMA[] =
+{
+  {0x00, 0x01},
+  {0x01, 0x02},
+  {0x03, 0x03},
+  {0x00, 0x00}
+};
+
+/*
+ * Buffers to store audio card informations
+ */
+static char     AudioExcelName[CARDNAMELEN + 1];
+static char     AudioExcelVersion[CARDVERLEN + 1];
+
+static void
+tenmillisec (void)
+{
+
+  for (gc = 0; gc < 1000; gc++)
+    tenmicrosec ();
+}
+
+static int
+WaitForDataAvail (int port)
+{
+  int             loop = STATUSRETRY;
+  unsigned char   ret = 0;
+
+  do
+    {
+      ret = INB (port + DSP_DATAVAIL);
+      /*
+         * Wait for data available (bit 7 of ret == 1)
+       */
+    }
+  while (!(ret & 0x80) && loop--);
+
+  if (ret & 0x80)
+    return 0;
+
+  return -1;
+}
+
+static int
+ReadData (int port)
+{
+  if (WaitForDataAvail (port))
+    return -1;
+  return INB (port + DSP_READ);
+}
+
+static int
+CheckDSPOkay (int port)
+{
+  return ((ReadData (port) == 0xaa) ? 0 : -1);
+}
+
+static int
+ResetBoard (int port)
+{
+  /*
+     * Reset DSP
+   */
+  OUTB (1, (port + DSP_RESET));
+  tenmicrosec ();
+  OUTB (0, (port + DSP_RESET));
+  tenmicrosec ();
+  tenmicrosec ();
+  return CheckDSPOkay (port);
+}
+
+static int
+WriteDSPCommand (int port, int cmd)
+{
+  unsigned char   ret;
+  int             loop = HARDRETRY;
+
+  do
+    {
+      ret = INB (port + DSP_STATUS);
+      /*
+         * DSP ready to receive data if bit 7 of ret == 0
+       */
+      if (!(ret & 0x80))
+	{
+	  OUTB (cmd, port + DSP_COMMAND);
+	  return 0;
+	}
+    }
+  while (loop--);
+
+  printk ("[aedsp16] DSP Command (0x%x) timeout.\n", cmd);
+  return -1;
+}
+
+int
+InitMSS (int port)
+{
+
+  tenmillisec ();
+
+  if (WriteDSPCommand (port, CMD6))
+    {
+      printk ("[aedsp16] CMD 0x%x: failed!\n", CMD6);
+      return -1;
+    }
+
+  tenmillisec ();
+
+  return 0;
+}
+
+static int
+SetUpBoard (int port)
+{
+  int             loop = RETRY;
+
+  do
+    {
+      if (WriteDSPCommand (portbase, CMD3))
+	{
+	  printk ("[aedsp16] CMD 0x%x: failed!\n", CMD3);
+	  return -1;
+	}
+
+      tenmillisec ();
+
+    }
+  while (WaitForDataAvail (port) && loop--);
+
+#if defined(THIS_SHOULD_GO_AWAY)
+  if (CheckDSPOkay (port))
+    {
+      printk ("[aedsp16]     CheckDSPOkay: failed\n");
+      return -1;
+    }
+#else
+  if (ReadData (port) == -1)
+    {
+      printk ("[aedsp16] ReadData after CMD 0x%x: failed\n", CMD3);
+      return -1;
+    }
+#endif
+
+  if (WriteDSPCommand (portbase, CMD4))
+    {
+      printk ("[aedsp16] CMD 0x%x: failed!\n", CMD4);
+      return -1;
+    }
+
+  if (WriteDSPCommand (portbase, CMD5))
+    {
+      printk ("[aedsp16] CMD 0x%x: failed!\n", CMD5);
+      return -1;
+    }
+
+  if (WriteDSPCommand (portbase, oredparams))
+    {
+      printk ("[aedsp16] Initialization of (M)IRQ and DMA: failed!\n");
+      return -1;
+    }
+  return 0;
+}
+
+static int
+GetCardVersion (int port)
+{
+  int             len = 0;
+  int             ret;
+  int             ver[3];
+
+  do
+    {
+      if ((ret = ReadData (port)) == -1)
+	return -1;
+      /*
+         * We alredy know how many int are stored (2), so we know when the
+         * string is finished.
+       */
+      ver[len++] = ret;
+    }
+  while (len < CARDVERLEN);
+  sprintf (AudioExcelVersion, "%d.%d", ver[0], ver[1]);
+  return 0;
+}
+
+static int
+GetCardName (int port)
+{
+  int             len = 0;
+  int             ret;
+
+  do
+    {
+      if ((ret = ReadData (port)) == -1)
+	/*
+	   * If no more data availabe, return to the caller, no error if len>0.
+	   * We have no other way to know when the string is finished.
+	 */
+	return (len ? 0 : -1);
+
+      AudioExcelName[len++] = ret;
+
+    }
+  while (len < CARDNAMELEN);
+  return 0;
+}
+
+static void
+InitializeHardParams (void)
+{
+
+  memset (AudioExcelName, 0, CARDNAMELEN + 1);
+  memset (AudioExcelVersion, 0, CARDVERLEN + 1);
+
+  for (gc = 0; orIRQ[gc].or; gc++)
+    if (orIRQ[gc].val == irq)
+      oredparams |= orIRQ[gc].or;
+
+  for (gc = 0; orMIRQ[gc].or; gc++)
+    if (orMIRQ[gc].or == mirq)
+      oredparams |= orMIRQ[gc].or;
+
+  for (gc = 0; orDMA[gc].or; gc++)
+    if (orDMA[gc].val == dma)
+      oredparams |= orDMA[gc].or;
+}
+
+static int
+InitAEDSP16 (int which)
+{
+  static char    *InitName = NULL;
+
+  InitializeHardParams ();
+
+  if (ResetBoard (portbase))
+    {
+      printk ("[aedsp16] ResetBoard: failed!\n");
+      return -1;
+    }
+
+#if defined(THIS_SHOULD_GO_AWAY)
+  if (CheckDSPOkay (portbase))
+    {
+      printk ("[aedsp16] CheckDSPOkay: failed!\n");
+      return -1;
+    }
+#endif
+
+  if (WriteDSPCommand (portbase, CMD1))
+    {
+      printk ("[aedsp16] CMD 0x%x: failed!\n", CMD1);
+      return -1;
+    }
+
+  if (GetCardName (portbase))
+    {
+      printk ("[aedsp16] GetCardName: failed!\n");
+      return -1;
+    }
+
+  /*
+     * My AEDSP16 card return SC-6000 in AudioExcelName, so
+     * if we have something different, we have to be warned.
+   */
+  if (strcmp ("SC-6000", AudioExcelName))
+    printk ("[aedsp16] Warning: non SC-6000 audio card!\n");
+
+  if (WriteDSPCommand (portbase, CMD2))
+    {
+      printk ("[aedsp16] CMD 0x%x: failed!\n", CMD2);
+      return -1;
+    }
+
+  if (GetCardVersion (portbase))
+    {
+      printk ("[aedsp16] GetCardVersion: failed!\n");
+      return -1;
+    }
+
+  if (SetUpBoard (portbase))
+    {
+      printk ("[aedsp16] SetUpBoard: failed!\n");
+      return -1;
+    }
+
+  if (which == INIT_MSS)
+    {
+      if (InitMSS (portbase))
+	{
+	  printk ("[aedsp16] Can't initialize Microsoft Sound System mode.\n");
+	  return -1;
+	}
+    }
+
+  /*
+     * If we are resetting, do not print any message because we may be
+     * in playing and we do not want lost too much time.
+   */
+  if (!(which & RESET_DSP16))
+    {
+      if (which & INIT_MPU401)
+	InitName = "MPU401";
+      else if (which & INIT_SBPRO)
+	InitName = "SBPro";
+      else if (which & INIT_MSS)
+	InitName = "MSS";
+      else
+	InitName = "None";
+
+      printk ("Audio Excel DSP 16 init v%s (%s %s) [%s]\n",
+	      VERSION, AudioExcelName,
+	      AudioExcelVersion, InitName);
+    }
+
+  tenmillisec ();
+
+  return 0;
+}
+
+#if defined(AEDSP16_SBPRO)
+
+int
+InitAEDSP16_SBPRO (struct address_info *hw_config)
+{
+  /*
+     * If the card is alredy init'ed MSS, we can not init it to SBPRO too
+     * because the board can not emulate simultaneously MSS and SBPRO.
+   */
+  if (ae_init & INIT_MSS)
+    return -1;
+  if (ae_init & INIT_SBPRO)
+    return 0;
+
+  /*
+     * For now we will leave this
+     * code included only when INCLUDE_AEDSP16 is configured in, but it should
+     * be better include it every time.
+   */
+  if (!(ae_init & INIT_MPU401))
+    {
+      if (check_region (hw_config->io_base, 0x0f))
+	{
+	  printk ("AEDSP16/SBPRO I/O port region is alredy in use.\n");
+	  return -1;
+	}
+    }
+
+  /*
+     * Set up the internal hardware parameters, to let the driver reach
+     * the Sound Card.
+   */
+  portbase = hw_config->io_base;
+  irq = hw_config->irq;
+  dma = hw_config->dma;
+  if (InitAEDSP16 (INIT_SBPRO))
+    return -1;
+
+  if (!(ae_init & INIT_MPU401))
+    request_region (hw_config->io_base, 0x0f, "aedsp16 (sbpro)");
+
+  ae_init |= INIT_SBPRO;
+  return 0;
+}
+
+#endif /* AEDSP16_SBPRO */
+
+#if defined(AEDSP16_MSS)
+
+int
+InitAEDSP16_MSS (struct address_info *hw_config)
+{
+  /*
+     * If the card is alredy init'ed SBPRO, we can not init it to MSS too
+     * because the board can not emulate simultaneously MSS and SBPRO.
+   */
+  if (ae_init & INIT_SBPRO)
+    return -1;
+  if (ae_init & INIT_MSS)
+    return 0;
+
+  /*
+     * For now we will leave this
+     * code included only when INCLUDE_AEDSP16 is configured in, but it should
+     * be better include it every time.
+   */
+  if (check_region (hw_config->io_base, 0x08))
+    {
+      printk ("MSS I/O port region is alredy in use.\n");
+      return -1;
+    }
+
+  /*
+     * We must allocate the AEDSP16 region too because these are the I/O ports
+     * to access card's control registers.
+   */
+  if (!(ae_init & INIT_MPU401))
+    {
+      if (check_region (AEDSP16_BASE, 0x0f))
+	{
+	  printk ("AEDSP16 I/O port region is alredy in use.\n");
+	  return -1;
+	}
+    }
+
+
+  /*
+     * If we are configuring the card for MSS, the portbase for card configuration
+     * is the default one (0x220 unless you have changed the factory default
+     * with board switches), so no need to modify the portbase variable.
+     * The default is AEDSP16_BASE, that is the right value.
+   */
+  irq = hw_config->irq;
+  dma = hw_config->dma;
+  if (InitAEDSP16 (INIT_MSS))
+    return -1;
+
+  request_region (hw_config->io_base, 0x08, "aedsp16 (mss)");
+
+  if (!(ae_init & INIT_MPU401))
+    request_region (AEDSP16_BASE, 0x0f, "aedsp16 (sbpro)");
+
+  ae_init |= INIT_MSS;
+  return 0;
+}
+
+#endif /* AEDSP16_MSS */
+
+#if defined(AEDSP16_MPU401)
+
+int
+InitAEDSP16_MPU401 (struct address_info *hw_config)
+{
+  if (ae_init & INIT_MPU401)
+    return 0;
+
+  /*
+     * For now we will leave this
+     * code included only when INCLUDE_AEDSP16 is configured in, but it should
+     * be better include it every time.
+   */
+  if (check_region (hw_config->io_base, 0x02))
+    {
+      printk ("SB I/O port region is alredy in use.\n");
+      return -1;
+    }
+
+  /*
+     * We must allocate the AEDSP16 region too because these are the I/O ports
+     * to access card's control registers.
+   */
+  if (!(ae_init & (INIT_MSS | INIT_SBPRO)))
+    {
+      if (check_region (AEDSP16_BASE, 0x0f))
+	{
+	  printk ("AEDSP16 I/O port region is alredy in use.\n");
+	  return -1;
+	}
+    }
+
+  /*
+     * If mpu401, the irq and dma are not important, do not touch it
+     * because we may use the default if sbpro is not yet configured,
+     * we may use the sbpro ones if configured, and nothing wrong
+     * should happen.
+     *
+     * The mirq default is 0, but once set it to non-0 value, we should
+     * not touch it anymore (unless I write an ioctl to do it, of course).
+   */
+  mirq = hw_config->irq;
+  if (InitAEDSP16 (INIT_MPU401))
+    return -1;
+
+  request_region (hw_config->io_base, 0x02, "aedsp16 (mpu401)");
+
+  if (!(ae_init & (INIT_MSS | INIT_SBPRO)))
+    request_region (AEDSP16_BASE, 0x0f, "aedsp16 (sbpro)");
+
+  ae_init |= INIT_MPU401;
+  return 0;
+}
+
+#endif /* AEDSP16_MPU401 */
+
+#if 0				/* Leave it out for now. We are not using this portion of code. */
+
+/*
+ * Entry point for a reset function.
+ * May be I will write the infamous ioctl :)
+ */
+int
+ResetAEDSP16 (void)
+{
+#if defined(AEDSP16_DEBUG)
+  printk ("[aedsp16] ResetAEDSP16 called.\n");
+#endif
+  return InitAEDSP16 (RESET_DSP16);
+}
+
+#endif /* 0 */
+
+#endif /* !EXCLUDE_AEDSP16 */
diff --git a/sys/i386/isa/sound/alaw.h b/sys/i386/isa/sound/alaw.h
new file mode 100644
index 000000000000..336bf6b2f776
--- /dev/null
+++ b/sys/i386/isa/sound/alaw.h
@@ -0,0 +1,73 @@
+static unsigned char alaw_linear[] = {
+	45, 	214, 	122, 	133, 	0, 		255, 	107, 	149, 
+	86, 	171, 	126, 	129, 	0, 		255, 	117, 	138, 
+	13, 	246, 	120, 	135, 	0, 		255, 	99, 	157, 
+	70, 	187, 	124, 	131, 	0, 		255, 	113, 	142, 
+	61, 	198, 	123, 	132,  	0, 		255, 	111, 	145, 
+	94, 	163, 	127, 	128, 	0, 		255, 	119, 	136, 
+	29, 	230, 	121, 	134, 	0, 		255, 	103, 	153, 
+	78, 	179, 	125, 	130, 	0, 		255, 	115, 	140, 
+	37, 	222, 	122, 	133, 	0, 		255, 	105, 	151, 
+	82, 	175, 	126, 	129, 	0, 		255, 	116, 	139, 
+	5, 	254, 	120, 	135, 	0, 		255, 	97, 	159, 
+	66, 	191, 	124, 	131, 	0, 		255, 	112,	143, 
+	53, 	206, 	123, 	132, 	0, 		255, 	109, 	147, 
+	90, 	167, 	127, 	128, 	0, 		255,	118, 	137, 
+	21, 	238, 	121, 	134, 	0, 		255, 	101,	155, 
+	74, 	183, 	125, 	130, 	0, 		255, 	114, 	141, 
+	49, 	210, 	123, 	133, 	0, 		255, 	108, 	148, 
+	88, 	169, 	127, 	129, 	0, 		255, 	118, 	138, 
+	17, 	242, 	121, 	135, 	0, 		255, 	100, 	156, 
+	72, 	185, 	125, 	131, 	0, 		255, 	114, 	142, 
+	64, 	194, 	124, 	132, 	0, 		255, 	112, 	144, 
+	96, 	161, 	128, 	128, 	1, 		255, 	120, 	136, 
+	33, 	226, 	122, 	134, 	0, 		255, 	104, 	152, 
+	80, 	177, 	126, 	130, 	0, 		255, 	116, 	140, 
+	41, 	218, 	122, 	133, 	0, 		255, 	106, 	150, 
+	84, 	173, 	126, 	129, 	0, 		255, 	117, 	139, 
+	9, 	250, 	120, 	135, 	0, 		255, 	98, 	158, 
+	68, 	189, 	124, 	131, 	0, 		255, 	113, 	143, 
+	57, 	202, 	123, 	132, 	0, 		255, 	110, 	146, 
+	92, 	165, 	127, 	128, 	0, 		255, 	119, 	137, 
+	25, 	234, 	121, 	134, 	0, 		255, 	102, 	154, 
+	76, 	181, 	125, 	130, 	0, 		255, 	115, 	141, 
+
+};
+
+#ifndef LINEAR_ALAW_NOT_WANTED
+static unsigned char linear_alaw[] = {
+
+	252,	 172,	 172,	 172,	 172,	 80,	 80,	 80,
+	80,	 208,	 208,	 208,	 208,	 16,	 16,	 16,
+	16,	 144,	 144,	 144,	 144,	 112,	 112,	 112,
+	112,	 240,	 240,	 240,	 240,	 48,	 48,	 48,
+	48,	 176,	 176,	 176,	 176,	 64,	 64,	 64,
+	64,	 192,	 192,	 192,	 192,	 0,	 0,	 0,
+	0,	 128,	 128,	 128,	 128,	 96,	 96,	 96,
+	96,	 224,	 224,	 224,	 224,	 32,	 32,	 32,
+	160,	 160,	 88,	 88,	 216,	 216,	 24,	 24,
+	152,	 152,	 120,	 120,	 248,	 248,	 56,	 56,
+	184,	 184,	 72,	 72,	 200,	 200,	 8,	 8,
+	136,	 136,	 104,	 104,	 232,	 232,	 40,	 40,
+	168,	 86,	 214,	 22,	 150,	 118,	 246,	 54,
+	182,	 70,	 198,	 6,	 134,	 102,	 230,	 38,
+	166, 	 222,	 158,	 254,	 190,	 206,	 142,	 238,
+	210,	 242,	 194,	 226,	 218,	 250,	 202,	 234,
+	235,	 203,	 251,	 219,	 227,	 195,	 243,	 211,
+	175,	 239,	 143,	 207,	 191,	 255,	 159,	 223,
+	167,  	 39,	 231,	 103,	 135,	 7,	 199,	 71,
+	183,	 55,	 247,	 119,	 151,	 23,	 215,	 87,
+	87,	 169,	 169,	 41,	 41,	 233,	 233,	 105,
+	105,	 137,	 137,	 9,	 9,	 201,	 201,	 73,
+	73,	 185,	 185,	 57,	 57,	 249,	 249,	 121,
+	121,	 153,	 153,	 25,	 25,	 217,	 217,	 89,
+	89,	 89,	 161,	 161,	 161,	 161,	 33,	 33,
+	33,	 33,	 225,	 225,	 225,	 225,	 97,	 97,
+	97,	 97,	 129,	 129,	 129,	 129,	 1,	 1,
+	1,	 1,	 193,	 193,	 193,	 193,	 65,	 65,
+	65,	 65,	 177,	 177,	 177,	 177,	 49,	 49,
+	49,	 49,	 241,	 241,	 241,	 241,	 113,	 113,
+	113,	 113,	 145,	 145,	 145,	 145,	 17,	 17,
+	17,	 17,	 209,	 209,	 209,	 209,	 81,	 253,
+};
+#endif /* !LINEAR_ALAW_NOT_WANTED */
diff --git a/sys/i386/isa/sound/awe_hw.h b/sys/i386/isa/sound/awe_hw.h
new file mode 100644
index 000000000000..8b6be670c4b8
--- /dev/null
+++ b/sys/i386/isa/sound/awe_hw.h
@@ -0,0 +1,116 @@
+/*
+ * sound/awe_hw.h
+ *
+ * Access routines and definitions for the low level driver for the 
+ * AWE32/Sound Blaster 32 wave table synth.
+ *   version 0.2.0a; Oct. 30, 1996
+ *
+ * (C) 1996 Takashi Iwai
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are
+ * met: 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer. 2.
+ * Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED.  IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE FOR
+ * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+ * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
+ * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ *
+ */
+
+#ifndef AWE_HW_H_DEF
+#define AWE_HW_H_DEF
+
+/*
+ * user configuration:
+ * if auto detection can't work properly, define the following values
+ * for your machine.
+ */
+/*#define AWE_DEFAULT_BASE_ADDR	0x620*/	/* base port address */
+/*#define AWE_DEFAULT_MEM_SIZE	512*/	/* kbytes */
+
+
+/*
+ * maximum size of sample table:
+ * if your data overflow, increase the following values.
+ */
+#define AWE_MAX_SAMPLES		400
+#define AWE_MAX_INFOS		900	/* GS presets has 801 infos! */
+
+
+/*
+ * Emu-8000 control registers
+ * name(channel)	reg, port
+ */
+
+#define awe_cmd_idx(reg,ch)	(((reg)<< 5) | (ch))
+
+#define Data0    0x620	/* doubleword r/w */
+#define Data1    0xA20	/* doubleword r/w */
+#define Data2    0xA22	/* word r/w */
+#define Data3    0xE20	/* word r/w */
+#define Pointer  0xE22	/* register pointer r/w */
+
+#define AWE_CPF(ch)	awe_cmd_idx(0,ch), Data0	/* DW: current pitch and fractional address */
+#define AWE_PTRX(ch)	awe_cmd_idx(1,ch), Data0	/* DW: pitch target and reverb send */
+#define AWE_CVCF(ch)	awe_cmd_idx(2,ch), Data0	/* DW: current volume and filter cutoff */
+#define AWE_VTFT(ch)	awe_cmd_idx(3,ch), Data0	/* DW: volume and filter cutoff targets */
+#define AWE_PSST(ch)	awe_cmd_idx(6,ch), Data0	/* DW: pan send and loop start address */
+#define AWE_CSL(ch)	awe_cmd_idx(7,ch), Data0	/* DW: chorus send and loop end address */
+#define AWE_CCCA(ch)	awe_cmd_idx(0,ch), Data1	/* DW: Q, control bits, and current address */
+#define AWE_HWCF4	awe_cmd_idx(1,9),  Data1	/* DW: config dw 4 */
+#define AWE_HWCF5	awe_cmd_idx(1,10), Data1	/* DW: config dw 5 */
+#define AWE_HWCF6	awe_cmd_idx(1,13), Data1	/* DW: config dw 6 */
+#define AWE_SMALR	awe_cmd_idx(1,20), Data1	/* DW: sound memory address for left read */
+#define AWE_SMARR	awe_cmd_idx(1,21), Data1	/* DW:    for right read */
+#define AWE_SMALW	awe_cmd_idx(1,22), Data1	/* DW: sound memory address for left write */
+#define AWE_SMARW	awe_cmd_idx(1,23), Data1	/* DW:    for right write */
+#define AWE_SMLD	awe_cmd_idx(1,26), Data1	/* W: sound memory left data */
+#define AWE_SMRD	awe_cmd_idx(1,26), Data2	/* W:    right data */
+#define AWE_WC		awe_cmd_idx(1,27), Data2	/* W: sample counter */
+#define AWE_WC_Cmd	awe_cmd_idx(1,27)
+#define AWE_WC_Port	Data2
+#define AWE_HWCF1	awe_cmd_idx(1,29), Data1	/* W: config w 1 */
+#define AWE_HWCF2	awe_cmd_idx(1,30), Data1	/* W: config w 2 */
+#define AWE_HWCF3	awe_cmd_idx(1,31), Data1	/* W: config w 3 */
+#define AWE_INIT1(ch)	awe_cmd_idx(2,ch), Data1	/* W: init array 1 */
+#define AWE_INIT2(ch)	awe_cmd_idx(2,ch), Data2	/* W: init array 2 */
+#define AWE_INIT3(ch)	awe_cmd_idx(3,ch), Data1	/* W: init array 3 */
+#define AWE_INIT4(ch)	awe_cmd_idx(3,ch), Data2	/* W: init array 4 */
+#define AWE_ENVVOL(ch)	awe_cmd_idx(4,ch), Data1	/* W: volume envelope delay */
+#define AWE_DCYSUSV(ch)	awe_cmd_idx(5,ch), Data1	/* W: volume envelope sustain and decay */
+#define AWE_ENVVAL(ch)	awe_cmd_idx(6,ch), Data1	/* W: modulation envelope delay */
+#define AWE_DCYSUS(ch)	awe_cmd_idx(7,ch), Data1	/* W: modulation envelope sustain and decay */
+#define AWE_ATKHLDV(ch)	awe_cmd_idx(4,ch), Data2	/* W: volume envelope attack and hold */
+#define AWE_LFO1VAL(ch)	awe_cmd_idx(5,ch), Data2	/* W: LFO#1 Delay */
+#define AWE_ATKHLD(ch)	awe_cmd_idx(6,ch), Data2	/* W: modulation envelope attack and hold */
+#define AWE_LFO2VAL(ch)	awe_cmd_idx(7,ch), Data2	/* W: LFO#2 Delay */
+#define AWE_IP(ch)	awe_cmd_idx(0,ch), Data3	/* W: initial pitch */
+#define AWE_IFATN(ch)	awe_cmd_idx(1,ch), Data3	/* W: initial filter cutoff and attenuation */
+#define AWE_PEFE(ch)	awe_cmd_idx(2,ch), Data3	/* W: pitch and filter envelope heights */
+#define AWE_FMMOD(ch)	awe_cmd_idx(3,ch), Data3	/* W: vibrato and filter modulation freq */
+#define AWE_TREMFRQ(ch)	awe_cmd_idx(4,ch), Data3	/* W: LFO#1 tremolo amount and freq */
+#define AWE_FM2FRQ2(ch)	awe_cmd_idx(5,ch), Data3	/* W: LFO#2 vibrato amount and freq */
+
+/*  used during detection (returns ROM version ?)                    */
+#define AWE_U1		0xE0, Data3	  /* (R)(W) used in initialization */
+#define AWE_U2(ch)	0xC0+(ch), Data3  /* (W)(W) used in init envelope  */
+
+
+#define AWE_MAX_VOICES		32
+#define AWE_NORMAL_VOICES	30	/*30&31 are reserved for DRAM refresh*/
+
+#define AWE_DRAM_OFFSET		0x200000
+
+#endif
diff --git a/sys/i386/isa/sound/awe_voice.h b/sys/i386/isa/sound/awe_voice.h
new file mode 100644
index 000000000000..934aaa8eba7e
--- /dev/null
+++ b/sys/i386/isa/sound/awe_voice.h
@@ -0,0 +1,268 @@
+/*
+ * sound/awe_voice.h
+ *
+ * Voice information definitions for the low level driver for the 
+ * AWE32/Sound Blaster 32 wave table synth.
+ *   version 0.2.0a; Oct. 30, 1996
+ *
+ * (C) 1996 Takashi Iwai
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are
+ * met: 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer. 2.
+ * Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED.  IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE FOR
+ * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+ * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
+ * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ *
+ */
+
+#ifndef AWE_VOICE_H
+#define AWE_VOICE_H
+
+#ifndef SAMPLE_TYPE_AWE32
+#define SAMPLE_TYPE_AWE32	0x20
+#endif
+
+#ifndef _PATCHKEY
+#define _PATCHKEY(id) ((id<<8)|0xfd)
+#endif
+
+/*----------------------------------------------------------------
+ * patch information record
+ *----------------------------------------------------------------*/
+
+/* patch interface header: 16 bytes */
+typedef struct awe_patch_info {
+	short key;			/* use AWE_PATCH here */
+#define AWE_PATCH	_PATCHKEY(0x07)
+
+	short device_no;		/* synthesizer number */
+	unsigned short sf_id;		/* file id (should be zero) */
+	short sf_version;		/* patch version (not referred) */
+	long len;			/* data length (without this header) */
+
+	short type;			/* following data type */
+#define AWE_LOAD_INFO		0
+#define AWE_LOAD_DATA		1
+
+	short reserved;			/* word alignment data */
+	char data[0];			/* patch data follows here */
+} awe_patch_info;
+
+
+/*----------------------------------------------------------------
+ * raw voice information record
+ *----------------------------------------------------------------*/
+
+/* wave table envelope & effect parameters to control EMU8000 */
+typedef struct _awe_voice_parm {
+	unsigned short moddelay;	/* modulation delay (0x8000) */
+	unsigned short modatkhld;	/* modulation attack & hold time (0x7f7f) */
+	unsigned short moddcysus;	/* modulation decay & sustain (0x7f7f) */
+	unsigned short modrelease;	/* modulation release time (0x807f) */
+	short modkeyhold, modkeydecay;	/* envelope change per key (not used) */
+	unsigned short voldelay;	/* volume delay (0x8000) */
+	unsigned short volatkhld;	/* volume attack & hold time (0x7f7f) */
+	unsigned short voldcysus;	/* volume decay & sustain (0x7f7f) */
+	unsigned short volrelease;	/* volume release time (0x807f) */
+	short volkeyhold, volkeydecay;	/* envelope change per key (not used) */
+	unsigned short lfo1delay;	/* LFO1 delay (0x8000) */
+	unsigned short lfo2delay;	/* LFO2 delay (0x8000) */
+	unsigned short pefe;		/* modulation pitch & cutoff (0x0000) */
+	unsigned short fmmod;		/* LFO1 pitch & cutoff (0x0000) */
+	unsigned short tremfrq;		/* LFO1 volume & freq (0x0000) */
+	unsigned short fm2frq2;		/* LFO2 pitch & freq (0x0000) */
+	unsigned char cutoff;		/* initial cutoff (0xff) */
+	unsigned char filterQ;		/* initial filter Q [0-15] (0x0) */
+	unsigned char chorus;		/* chorus send (0x00) */
+	unsigned char reverb;		/* reverb send (0x00) */
+	unsigned short reserved[4];	/* not used */
+} awe_voice_parm;
+
+/* wave table parameters: 92 bytes */
+typedef struct _awe_voice_info {
+	unsigned short sf_id;		/* file id (should be zero) */
+	unsigned short sample;		/* sample id */
+	long start, end;		/* sample offset correction */
+	long loopstart, loopend;	/* loop offset correction */
+	short rate_offset;		/* sample rate pitch offset */
+	unsigned short mode;		/* sample mode */
+#define AWE_MODE_ROMSOUND		0x8000
+#define AWE_MODE_STEREO			1
+#define AWE_MODE_LOOPING		2
+#define AWE_MODE_NORELEASE		4	/* obsolete */
+#define AWE_MODE_INIT_PARM		8
+
+	short root;			/* midi root key */
+	short tune;			/* pitch tuning (in cents) */
+	char low, high;			/* key note range */
+	char vellow, velhigh;		/* velocity range */
+	char fixkey, fixvel;		/* fixed key, velocity */
+	char pan, fixpan;		/* panning, fixed panning */
+	short exclusiveClass;		/* exclusive class (0 = none) */
+	unsigned char amplitude;	/* sample volume (127 max) */
+	unsigned char attenuation;	/* attenuation (0.375dB) */
+	short scaleTuning;		/* pitch scale tuning(%), normally 100 */
+	awe_voice_parm parm;		/* voice envelope parameters */
+	short index;			/* internal index (set by driver) */
+} awe_voice_info;
+
+/* instrument info header: 4 bytes */
+typedef struct _awe_voice_rec {
+	unsigned char bank;		/* midi bank number */
+	unsigned char instr;		/* midi preset number */
+	short nvoices;			/* number of voices */
+	awe_voice_info info[0];		/* voice information follows here */
+} awe_voice_rec;
+
+
+/*----------------------------------------------------------------
+ * sample wave information
+ *----------------------------------------------------------------*/
+
+/* wave table sample header: 32 bytes */
+typedef struct awe_sample_info {
+	unsigned short sf_id;		/* file id (should be zero) */
+	unsigned short sample;		/* sample id */
+	long start, end;		/* start & end offset */
+	long loopstart, loopend;	/* loop start & end offset */
+	long size;			/* size (0 = ROM) */
+	short checksum_flag;		/* use check sum = 1 */
+	unsigned short mode_flags;	/* mode flags */
+#define AWE_SAMPLE_8BITS	1	/* wave data is 8bits */
+#define AWE_SAMPLE_UNSIGNED	2	/* wave data is unsigned */
+#define AWE_SAMPLE_NO_BLANK	4	/* no blank loop is attached */
+#define AWE_SAMPLE_SINGLESHOT	8	/* single-shot w/o loop */
+#define AWE_SAMPLE_BIDIR_LOOP	16	/* bidirectional looping */
+#define AWE_SAMPLE_STEREO_LEFT	32	/* stereo left sound */
+#define AWE_SAMPLE_STEREO_RIGHT	64	/* stereo right sound */
+	unsigned long checksum;		/* check sum */
+	unsigned short data[0];		/* sample data follows here */
+} awe_sample_info;
+
+
+/*----------------------------------------------------------------
+ * awe hardware controls
+ *----------------------------------------------------------------*/
+
+#define _AWE_DEBUG_MODE			0x00
+#define _AWE_REVERB_MODE		0x01
+#define _AWE_CHORUS_MODE		0x02
+#define _AWE_REMOVE_LAST_SAMPLES	0x03
+#define _AWE_INITIALIZE_CHIP		0x04
+#define _AWE_SEND_EFFECT		0x05
+#define _AWE_TERMINATE_CHANNEL		0x06
+#define _AWE_TERMINATE_ALL		0x07
+#define _AWE_INITIAL_VOLUME		0x08
+#define _AWE_SET_GUS_BANK		0x09
+
+#define _AWE_MODE_FLAG			0x80
+#define _AWE_COOKED_FLAG		0x40	/* not supported */
+#define _AWE_MODE_VALUE_MASK		0x3F
+
+#define _AWE_CMD(chn, voice, cmd, p1, p2) \
+{_SEQ_NEEDBUF(8); _seqbuf[_seqbufptr] = SEQ_PRIVATE;\
+ _seqbuf[_seqbufptr+1] = chn;\
+ _seqbuf[_seqbufptr+2] = _AWE_MODE_FLAG|(cmd);\
+ _seqbuf[_seqbufptr+3] = voice;\
+ *(unsigned short*)&_seqbuf[_seqbufptr+4] = p1;\
+ *(unsigned short*)&_seqbuf[_seqbufptr+6] = p2;\
+ _SEQ_ADVBUF(8);}
+
+#define AWE_DEBUG_MODE(dev,p1)	_AWE_CMD(dev, 0, _AWE_DEBUG_MODE, p1, 0)
+#define AWE_REVERB_MODE(dev,p1)	_AWE_CMD(dev, 0, _AWE_REVERB_MODE, p1, 0)
+#define AWE_CHORUS_MODE(dev,p1)	_AWE_CMD(dev, 0, _AWE_CHORUS_MODE, p1, 0)
+#define AWE_REMOVE_LAST_SAMPLES(dev) _AWE_CMD(dev, 0, _AWE_REMOVE_LAST_SAMPLES, 0, 0)
+#define AWE_INITIALIZE_CHIP(dev) _AWE_CMD(dev, 0, _AWE_INITIALIZE_CHIP, 0, 0)
+#define AWE_SEND_EFFECT(dev,voice,type,value) _AWE_CMD(dev,voice,_AWE_SEND_EFFECT,type,value)
+#define AWE_TERMINATE_CHANNEL(dev,voice) _AWE_CMD(dev,voice,_AWE_TERMINATE_CHANNEL,0,0)
+#define AWE_TERMINATE_ALL(dev) _AWE_CMD(dev, 0, _AWE_TERMINATE_ALL, 0, 0)
+#define AWE_INITIAL_VOLUME(dev,atten) _AWE_CMD(dev, 0, _AWE_INITIAL_VOLUME, atten, 0)
+#define AWE_SET_GUS_BANK(dev,bank) _AWE_CMD(dev, 0, _AWE_SET_GUS_BANK, bank, 0)
+
+/* reverb mode */
+#define	AWE_REVERB_ROOM1	0
+#define AWE_REVERB_ROOM2	1
+#define	AWE_REVERB_ROOM3	2
+#define	AWE_REVERB_HALL1	3
+#define	AWE_REVERB_HALL2	4
+#define	AWE_REVERB_PLATE	5
+#define	AWE_REVERB_DELAY	6
+#define	AWE_REVERB_PANNINGDELAY 7
+
+/* chorus mode */
+#define AWE_CHORUS_1		0
+#define	AWE_CHORUS_2		1
+#define	AWE_CHORUS_3		2
+#define	AWE_CHORUS_4		3
+#define	AWE_CHORUS_FEEDBACK	4
+#define	AWE_CHORUS_FLANGER	5
+#define	AWE_CHORUS_SHORTDELAY	6
+#define	AWE_CHORUS_SHORTDELAY2	7
+
+/* effects */
+enum {
+
+/* modulation envelope parameters */
+/* 0*/	AWE_FX_ENV1_DELAY,	/* WORD: ENVVAL */
+/* 1*/	AWE_FX_ENV1_ATTACK,	/* BYTE: up ATKHLD */
+/* 2*/	AWE_FX_ENV1_HOLD,	/* BYTE: lw ATKHLD */
+/* 3*/	AWE_FX_ENV1_DECAY,	/* BYTE: lw DCYSUS */
+/* 4*/	AWE_FX_ENV1_RELEASE,	/* BYTE: lw DCYSUS */
+/* 5*/	AWE_FX_ENV1_SUSTAIN,	/* BYTE: up DCYSUS */
+/* 6*/	AWE_FX_ENV1_PITCH,	/* BYTE: up PEFE */
+/* 7*/	AWE_FX_ENV1_CUTOFF,	/* BYTE: lw PEFE */
+
+/* volume envelope parameters */
+/* 8*/	AWE_FX_ENV2_DELAY,	/* WORD: ENVVOL */
+/* 9*/	AWE_FX_ENV2_ATTACK,	/* BYTE: up ATKHLDV */
+/*10*/	AWE_FX_ENV2_HOLD,	/* BYTE: lw ATKHLDV */
+/*11*/	AWE_FX_ENV2_DECAY,	/* BYTE: lw DCYSUSV */
+/*12*/	AWE_FX_ENV2_RELEASE,	/* BYTE: lw DCYSUSV */
+/*13*/	AWE_FX_ENV2_SUSTAIN,	/* BYTE: up DCYSUSV */
+	
+/* LFO1 (tremolo & vibrato) parameters */
+/*14*/	AWE_FX_LFO1_DELAY,	/* WORD: LFO1VAL */
+/*15*/	AWE_FX_LFO1_FREQ,	/* BYTE: lo TREMFRQ */
+/*16*/	AWE_FX_LFO1_VOLUME,	/* BYTE: up TREMFRQ */
+/*17*/	AWE_FX_LFO1_PITCH,	/* BYTE: up FMMOD */
+/*18*/	AWE_FX_LFO1_CUTOFF,	/* BYTE: lo FMMOD */
+
+/* LFO2 (vibrato) parameters */
+/*19*/	AWE_FX_LFO2_DELAY,	/* WORD: LFO2VAL */
+/*20*/	AWE_FX_LFO2_FREQ,	/* BYTE: lo FM2FRQ2 */
+/*21*/	AWE_FX_LFO2_PITCH,	/* BYTE: up FM2FRQ2 */
+
+/* Other overall effect parameters */
+/*22*/	AWE_FX_INIT_PITCH,	/* SHORT: pitch offset */
+/*23*/	AWE_FX_CHORUS,		/* BYTE: chorus effects send (0-255) */
+/*24*/	AWE_FX_REVERB,		/* BYTE: reverb effects send (0-255) */
+/*25*/	AWE_FX_CUTOFF,		/* BYTE: up IFATN */
+/*26*/	AWE_FX_FILTERQ,		/* BYTE: up CCCA */
+
+/* Sample / loop offset changes */
+/*27*/	AWE_FX_SAMPLE_START,	/* SHORT: offset */
+/*28*/	AWE_FX_LOOP_START,	/* SHORT: offset */
+/*29*/	AWE_FX_LOOP_END,	/* SHORT: offset */
+/*30*/	AWE_FX_COARSE_SAMPLE_START,	/* SHORT: upper word offset */
+/*31*/	AWE_FX_COARSE_LOOP_START,	/* SHORT: upper word offset */
+/*32*/	AWE_FX_COARSE_LOOP_END,		/* SHORT: upper word offset */
+
+	AWE_FX_END,
+};
+
+
+#endif /* AWE_VOICE_H */
diff --git a/sys/i386/isa/sound/awe_wave.c b/sys/i386/isa/sound/awe_wave.c
new file mode 100644
index 000000000000..47818165a9e4
--- /dev/null
+++ b/sys/i386/isa/sound/awe_wave.c
@@ -0,0 +1,3367 @@
+/*
+ * sound/awe_wave.c
+ *
+ * The low level driver for the AWE32/Sound Blaster 32 wave table synth.
+ *   version 0.2.0a; Oct. 30, 1996
+ *
+ * (C) 1996 Takashi Iwai
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are
+ * met: 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer. 2.
+ * Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED.  IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE FOR
+ * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+ * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
+ * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ *
+ */
+
+/* if you're using obsolete VoxWare 3.0.x on Linux 1.2.x (or FreeBSD),
+ * uncomment the following line
+ */
+#define AWE_OBSOLETE_VOXWARE
+
+
+#ifdef AWE_OBSOLETE_VOXWARE
+
+#ifdef __FreeBSD__
+#  include <i386/isa/sound/sound_config.h>
+#else
+#  include "sound_config.h"
+#endif
+
+#if defined(CONFIGURE_SOUNDCARD) && !defined(EXCLUDE_AWE32)
+#define CONFIG_AWE32_SYNTH
+#endif
+
+#else /* AWE_OBSOLETE_VOXWARE */
+
+#include "../sound_config.h"
+
+#endif /* AWE_OBSOLETE_VOXWARE */
+
+
+/*----------------------------------------------------------------*
+ * compile condition
+ *----------------------------------------------------------------*/
+
+/* initialize FM passthrough even without extended RAM */
+/*#define AWE_ALWAYS_INIT_FM*/
+
+/* debug on */
+#define AWE_DEBUG_ON
+
+/* verify checksum for uploading samples */
+#define AWE_CHECKSUM_DATA
+#define AWE_CHECKSUM_MEMORY
+
+/* disable interruption during sequencer operation */
+/*#define AWE_NEED_DISABLE_INTR*/
+
+/* use buffered access to user wave data */
+#define AWE_USE_BUFFERED_IO
+
+#ifdef linux
+/* i tested this only on my linux */
+#define INLINE  __inline__
+#else
+#define INLINE /**/
+#endif
+
+/*----------------------------------------------------------------*/
+
+#ifdef CONFIG_AWE32_SYNTH
+
+#include <i386/isa/sound/awe_hw.h>
+#include <i386/isa/sound/awe_voice.h>
+
+#ifdef AWE_OBSOLETE_VOXWARE
+#ifdef __FreeBSD__
+#define SEQUENCER_C
+#endif
+#include <i386/isa/sound/tuning.h>
+#else
+#include "../tuning.h"
+#endif
+
+#ifdef linux
+#  include <linux/ultrasound.h>
+#elif defined(__FreeBSD__)
+#  include <machine/ultrasound.h>
+#endif
+
+
+/*----------------------------------------------------------------
+ * debug message
+ *----------------------------------------------------------------*/
+
+#ifdef AWE_DEBUG_ON
+static int debug_mode = 0;
+#define DEBUG(LVL,XXX)	{if (debug_mode > LVL) { XXX; }}
+#define ERRMSG(XXX)	{if (debug_mode) { XXX; }}
+#define FATALERR(XXX)	XXX
+#else
+#define DEBUG(LVL,XXX) /**/
+#define ERRMSG(XXX)	XXX
+#define FATALERR(XXX)	XXX
+#endif
+
+/*----------------------------------------------------------------
+ * bank and voice record
+ *----------------------------------------------------------------*/
+
+/* bank record */
+typedef struct _awe_voice_list {
+	unsigned char bank, instr;
+	awe_voice_info v;
+	struct _awe_voice_list *next_instr;
+	struct _awe_voice_list *next_bank;
+} awe_voice_list;
+
+/* sample and information table */
+static awe_sample_info *samples;
+static awe_voice_list *infos;
+
+#define AWE_MAX_PRESETS		256
+#define AWE_DEFAULT_BANK	0
+
+/* preset table index */
+static awe_voice_list *preset_table[AWE_MAX_PRESETS];
+
+/*----------------------------------------------------------------
+ * voice table
+ *----------------------------------------------------------------*/
+
+#define AWE_FX_BYTES	((AWE_FX_END+7)/8)
+
+typedef struct _voice_info {
+	int state;		/* status (on = 1, off = 0) */
+	int note;		/* midi key (0-127) */
+	int velocity;		/* midi velocity (0-127) */
+	int bender;		/* midi pitchbend (-8192 - 8192) */
+	int bender_range;	/* midi bender range (x100) */
+	int panning;		/* panning (0-127) */
+	int main_vol;		/* channel volume (0-127) */
+	int expression_vol;	/* midi expression (0-127) */
+
+	/* EMU8000 parameters */
+	int apitch;		/* pitch parameter */
+	int avol;		/* volume parameter */
+
+	/* instrument parameters */
+	int bank;		/* current tone bank */
+	int instr;		/* current program */
+	awe_voice_list *vrec;
+	awe_voice_info *sample;
+
+	/* channel effects */
+	unsigned char fx_flags[AWE_FX_BYTES];
+	short fx[AWE_FX_END];
+} voice_info;
+
+static voice_info voices[AWE_MAX_VOICES];
+
+
+/*----------------------------------------------------------------
+ * global variables
+ *----------------------------------------------------------------*/
+
+/* awe32 base address (overwritten at initialization) */
+static int awe_base = 0;
+/* memory byte size (overwritten at initialization) */
+static long awe_mem_size = 0;
+
+/* maximum channels for playing */
+static int awe_max_voices = AWE_MAX_VOICES;
+
+static long free_mem_ptr = 0;		/* free word byte size */
+static int free_info = 0;		/* free info tables */
+static int last_info = 0;		/* last loaded info index */
+static int free_sample = 0;		/* free sample tables */
+static int last_sample = 0;		/* last loaded sample index */
+static int loaded_once = 0;		/* samples are loaded after init? */
+static unsigned short current_sf_id = 0;	/* internal id */
+
+static int reverb_mode = 0;		/* reverb mode */
+static int chorus_mode = 0;		/* chorus mode */
+static unsigned short init_atten = 32;  /* 12dB */
+
+static int awe_present = 0;		/* awe device present? */
+static int awe_busy = 0;		/* awe device opened? */
+
+static int awe_gus_bank = AWE_DEFAULT_BANK;	/* GUS default bank number */
+
+
+static struct synth_info awe_info = {
+	"AWE32 Synth",		/* name */
+	0,			/* device */
+	SYNTH_TYPE_SAMPLE,	/* synth_type */
+	SAMPLE_TYPE_AWE32,	/* synth_subtype */
+	0,			/* perc_mode (obsolete) */
+	AWE_MAX_VOICES,		/* nr_voices */
+	0,			/* nr_drums (obsolete) */
+	AWE_MAX_INFOS		/* instr_bank_size */
+};
+
+
+static struct voice_alloc_info *voice_alloc;	/* set at initialization */
+
+
+/*----------------------------------------------------------------
+ * function prototypes
+ *----------------------------------------------------------------*/
+
+#ifndef AWE_OBSOLETE_VOXWARE
+static int awe_check_port(void);
+static void awe_request_region(void);
+static void awe_release_region(void);
+#endif
+
+static void awe_reset_samples(void);
+/* emu8000 chip i/o access */
+static void awe_poke(unsigned short cmd, unsigned short port, unsigned short data);
+static void awe_poke_dw(unsigned short cmd, unsigned short port, unsigned long data);
+static unsigned short awe_peek(unsigned short cmd, unsigned short port);
+static unsigned long awe_peek_dw(unsigned short cmd, unsigned short port);
+static void awe_wait(unsigned short delay);
+
+/* initialize emu8000 chip */
+static void awe_initialize(void);
+
+/* set voice parameters */
+static void awe_init_voice_info(awe_voice_info *vp);
+static void awe_init_voice_parm(awe_voice_parm *pp);
+static int freq_to_note(int freq);
+static int calc_rate_offset(int Hz);
+/*static int calc_parm_delay(int msec);*/
+static int calc_parm_hold(int msec);
+static int calc_parm_attack(int msec);
+static int calc_parm_decay(int msec);
+static int calc_parm_search(int msec, short *table);
+
+/* turn on/off note */
+static void awe_note_on(int voice);
+static void awe_note_off(int voice);
+static void awe_terminate(int voice);
+static void awe_exclusive_off(int voice);
+
+/* calculate voice parameters */
+static void awe_set_pitch(int voice);
+static void awe_set_volume(int voice);
+static void awe_set_pan(int voice, int forced);
+static void awe_fx_fmmod(int voice);
+static void awe_fx_tremfrq(int voice);
+static void awe_fx_fm2frq2(int voice);
+static void awe_fx_cutoff(int voice);
+static void awe_fx_initpitch(int voice);
+static void awe_calc_pitch(int voice);
+static void awe_calc_pitch_from_freq(int voice, int freq);
+static void awe_calc_volume(int voice);
+static void awe_voice_init(int voice, int inst_only);
+
+/* sequencer interface */
+static int awe_open(int dev, int mode);
+static void awe_close(int dev);
+static int awe_ioctl(int dev, unsigned int cmd, caddr_t arg);
+static int awe_kill_note(int dev, int voice, int note, int velocity);
+static int awe_start_note(int dev, int v, int note_num, int volume);
+static int awe_set_instr(int dev, int voice, int instr_no);
+static void awe_reset(int dev);
+static void awe_hw_control(int dev, unsigned char *event);
+static int awe_load_patch(int dev, int format, const char *addr,
+			  int offs, int count, int pmgr_flag);
+static void awe_aftertouch(int dev, int voice, int pressure);
+static void awe_controller(int dev, int voice, int ctrl_num, int value);
+static void awe_panning(int dev, int voice, int value);
+static void awe_volume_method(int dev, int mode);
+static int awe_patchmgr(int dev, struct patmgr_info *rec);
+static void awe_bender(int dev, int voice, int value);
+static int awe_alloc(int dev, int chn, int note, struct voice_alloc_info *alloc);
+static void awe_setup_voice(int dev, int voice, int chn);
+
+/* hardware controls */
+static void awe_hw_gus_control(int dev, int cmd, unsigned char *event);
+static void awe_hw_awe_control(int dev, int cmd, unsigned char *event);
+
+/* voice search */
+static awe_voice_info *awe_search_voice(int voice, int note);
+static awe_voice_list *awe_search_instr(int bank, int preset);
+
+/* load / remove patches */
+static void awe_check_loaded(void);
+static int awe_load_info(awe_patch_info *patch, const char *addr);
+static int awe_load_data(awe_patch_info *patch, const char *addr);
+static int awe_load_guspatch(const char *addr, int offs, int size, int pmgr_flag);
+static int awe_write_wave_data(const char *addr, long offset, int size);
+static awe_voice_list *awe_get_removed_list(awe_voice_list *curp);
+static void awe_remove_samples(void);
+static short awe_set_sample(awe_voice_info *vp);
+
+/* lowlevel functions */
+static void awe_init_audio(void);
+static void awe_init_dma(void);
+static void awe_init_array(void);
+static void awe_send_array(unsigned short *data);
+static void awe_tweak(void);
+static void awe_init_fm(void);
+static int awe_open_dram_for_write(int offset);
+static int awe_open_dram_for_read(int offset);
+static void awe_open_dram_for_check(void);
+static void awe_close_dram(void);
+static void awe_close_dram_for_read(void);
+static void awe_write_dram(unsigned short c);
+static int awe_detect(void);
+static int awe_check_dram(void);
+static void awe_set_chorus_mode(int mode);
+static void awe_set_reverb_mode(int mode);
+
+#ifdef AWE_OBSOLETE_VOXWARE
+
+#define awe_check_port()	0	/* always false */
+#define awe_request_region()	/* nothing */
+#define awe_release_region()	/* nothing */
+
+#else /* AWE_OBSOLETE_VOXWARE */
+
+/* the following macros are osbolete */
+
+#define PERMANENT_MALLOC(type,var,size,memptr) \
+	var = (type)(sound_mem_blocks[sound_nblocks++] = vmalloc(size))
+#define RET_ERROR(err)			-err
+
+#endif /* AWE_OBSOLETE_VOXWARE */
+
+
+#ifdef AWE_NEED_DISABLE_INTR
+#define DECL_INTR_FLAGS(x)	unsigned long x
+#else
+#undef DISABLE_INTR
+#undef RESTORE_INTR
+#define DECL_INTR_FLAGS(x) /**/
+#define DISABLE_INTR(x) /**/
+#define RESTORE_INTR(x) /**/
+#endif
+
+
+/* macros for Linux and FreeBSD compatibility */
+
+#undef OUTW
+#undef COPY_FROM_USER
+#undef GET_BYTE_FROM_USER
+#undef GET_SHORT_FROM_USER
+#undef IOCTL_TO_USER
+  
+#ifdef linux
+#  define NO_DATA_ERR                 ENODATA
+#  define OUTW(data, addr)            outw(data, addr)
+#  define COPY_FROM_USER(target, source, offs, count) \
+              memcpy_fromfs( ((caddr_t)(target)),(source)+(offs),(count) )
+#  define GET_BYTE_FROM_USER(target, addr, offs)      \
+              *((char  *)&(target)) = get_fs_byte( (addr)+(offs) )
+#  define GET_SHORT_FROM_USER(target, addr, offs)     \
+              *((short *)&(target)) = get_fs_word( (addr)+(offs) )
+#  define IOCTL_TO_USER(target, offs, source, count)  \
+              memcpy_tofs  ( ((caddr_t)(target)),(source)+(offs),(count) )
+#  define BZERO(target,len)                           \
+              memset( (caddr_t)target, '\0', len )
+#  define MEMCPY(dst,src,len) \
+              memcpy((caddr_t)dst, (caddr_t)src, len)
+#elif defined(__FreeBSD__)
+#  define NO_DATA_ERR                 EINVAL
+#  define OUTW(data, addr)            outw(addr, data)
+#  define COPY_FROM_USER(target, source, offs, count) \
+              uiomove( ((caddr_t)(target)),(count),((struct uio *)(source)) )
+#  define GET_BYTE_FROM_USER(target, addr, offs)      \
+              uiomove( ((char*)&(target)), 1, ((struct uio *)(addr)) )
+#  define GET_SHORT_FROM_USER(target, addr, offs)     \
+              uiomove( ((char*)&(target)), 2, ((struct uio *)(addr)) )
+#  define IOCTL_TO_USER(target, offs, source, count)  \
+              memcpy( &((target)[offs]), (source), (count) )
+#  define BZERO(target,len)                           \
+              bzero( (caddr_t)target, len )
+#  define MEMCPY(dst,src,len) \
+              bcopy((caddr_t)src, (caddr_t)dst, len)
+#endif
+
+
+/*----------------------------------------------------------------
+ * synth operation table
+ *----------------------------------------------------------------*/
+
+static struct synth_operations awe_operations =
+{
+	&awe_info,
+	0,
+	SYNTH_TYPE_SAMPLE,
+	SAMPLE_TYPE_AWE32,
+	awe_open,
+	awe_close,
+	awe_ioctl,
+	awe_kill_note,
+	awe_start_note,
+	awe_set_instr,
+	awe_reset,
+	awe_hw_control,
+	awe_load_patch,
+	awe_aftertouch,
+	awe_controller,
+	awe_panning,
+	awe_volume_method,
+	awe_patchmgr,
+	awe_bender,
+	awe_alloc,
+	awe_setup_voice
+};
+
+
+
+/*================================================================
+ * attach / unload interface
+ *================================================================*/
+
+#ifdef AWE_OBSOLETE_VOXWARE
+long attach_awe_obsolete(long mem_start, struct address_info *hw_config)
+#else
+int attach_awe(void)
+#endif
+{
+	/* check presence of AWE32 card */
+	if (! awe_detect()) {
+		printk("AWE32: not detected\n");
+		return 0;
+	}
+
+	/* check AWE32 ports are available */
+	if (awe_check_port()) {
+		printk("AWE32: I/O area already used.\n");
+		return 0;
+	}
+
+	/* allocate sample tables */
+	PERMANENT_MALLOC(awe_sample_info *, samples,
+			 AWE_MAX_SAMPLES * sizeof(awe_sample_info), mem_start);
+	PERMANENT_MALLOC(awe_voice_list *, infos,
+			 AWE_MAX_INFOS * sizeof(awe_voice_list), mem_start);
+	if (samples == NULL || infos == NULL) {
+		printk("AWE32: can't allocate sample tables\n");
+		return 0;
+	}
+
+	if (num_synths >= MAX_SYNTH_DEV)
+		printk("AWE32 Error: too many synthesizers\n");
+	else {
+		voice_alloc = &awe_operations.alloc;
+		voice_alloc->max_voice = awe_max_voices;
+		synth_devs[num_synths++] = &awe_operations;
+	}
+
+	/* reserve I/O ports for awedrv */
+	awe_request_region();
+
+	/* clear all samples */
+	awe_reset_samples();
+
+	/* intialize AWE32 hardware */
+	awe_initialize();
+
+#if 0	/* Drivers shouldn't be this chatty by default */
+	printk("<AWE32 SynthCard (%dk)>\n", (int)awe_mem_size/1024);
+#endif
+	sprintf(awe_info.name, "AWE32 Synth (%dk)", (int)awe_mem_size/1024);
+
+	/* set reverb & chorus modes */
+	awe_set_reverb_mode(reverb_mode);
+	awe_set_chorus_mode(chorus_mode);
+
+	awe_present = 1;
+
+#ifdef AWE_OBSOLETE_VOXWARE
+	return mem_start;
+#else
+	return 1;
+#endif
+}
+
+
+void unload_awe(void)
+{
+	if (awe_present) {
+		awe_reset_samples();
+		awe_release_region();
+	}
+}
+
+
+#ifdef AWE_OBSOLETE_VOXWARE
+int probe_awe_obsolete(struct address_info *hw_config)
+{
+	return 1;
+	/*return awe_detect();*/
+}
+
+#endif
+
+/*================================================================
+ * clear sample tables 
+ *================================================================*/
+
+static void
+awe_reset_samples(void)
+{
+	int i;
+
+	/* free all bank tables */
+	for (i = 0; i < AWE_MAX_PRESETS; i++) {
+		preset_table[i] = NULL;
+	}
+
+	free_mem_ptr = 0;
+	last_sample = free_sample = 0;
+	last_info = free_info = 0;
+	current_sf_id = 0;
+	loaded_once = 0;
+}
+
+
+/*================================================================
+ * EMU register access
+ *================================================================*/
+
+/* select a given AWE32 pointer */
+static int awe_cur_cmd = -1;
+#define awe_set_cmd(cmd) \
+if (awe_cur_cmd != cmd) { OUTW(cmd, awe_base + 0x802); awe_cur_cmd = cmd; }
+#define awe_port(port)		(awe_base - 0x620 + port)
+
+/* write 16bit data */
+INLINE static void
+awe_poke(unsigned short cmd, unsigned short port, unsigned short data)
+{
+	awe_set_cmd(cmd);
+	OUTW(data, awe_port(port));
+}
+
+/* write 32bit data */
+INLINE static void
+awe_poke_dw(unsigned short cmd, unsigned short port, unsigned long data)
+{
+	awe_set_cmd(cmd);
+	OUTW(data, awe_port(port));		/* write lower 16 bits */
+	OUTW(data >> 16, awe_port(port)+2);	/* write higher 16 bits */
+}
+
+/* read 16bit data */
+INLINE static unsigned short
+awe_peek(unsigned short cmd, unsigned short port)
+{
+	unsigned short k;
+	awe_set_cmd(cmd);
+	k = inw(awe_port(port));
+	return k;
+}
+
+/* read 32bit data */
+INLINE static unsigned long
+awe_peek_dw(unsigned short cmd, unsigned short port)
+{
+	unsigned long k1, k2;
+	awe_set_cmd(cmd);
+	k1 = inw(awe_port(port));
+	k2 = inw(awe_port(port)+2);
+	k1 |= k2 << 16;
+	return k1;
+}
+
+/* wait delay number of AWE32 44100Hz clocks */
+static void
+awe_wait(unsigned short delay)
+{
+	unsigned short clock, target;
+	unsigned short port = awe_port(AWE_WC_Port);
+	int counter;
+  
+	/* sample counter */
+	awe_set_cmd(AWE_WC_Cmd);
+	clock = (unsigned short)inw(port);
+	target = clock + delay;
+	counter = 0;
+	if (target < clock) {
+		for (; (unsigned short)inw(port) > target; counter++)
+			if (counter > 65536)
+				break;
+	}
+	for (; (unsigned short)inw(port) < target; counter++)
+		if (counter > 65536)
+			break;
+}
+
+
+#ifndef AWE_OBSOLETE_VOXWARE
+
+/*================================================================
+ * port check / request
+ *  0x620-622, 0xA20-A22, 0xE20-E22
+ *================================================================*/
+
+static int
+awe_check_port(void)
+{
+	return (check_region(awe_port(Data0), 3) ||
+		check_region(awe_port(Data1), 3) ||
+		check_region(awe_port(Data3), 3));
+}
+
+static void
+awe_request_region(void)
+{
+	request_region(awe_port(Data0), 3, "sound driver (AWE32)");
+	request_region(awe_port(Data1), 3, "sound driver (AWE32)");
+	request_region(awe_port(Data3), 3, "sound driver (AWE32)");
+}
+
+static void
+awe_release_region(void)
+{
+	release_region(awe_port(Data0), 3);
+	release_region(awe_port(Data1), 3);
+	release_region(awe_port(Data3), 3);
+}
+
+#endif /* !AWE_OBSOLETE_VOXWARE */
+
+
+/*================================================================
+ * AWE32 initialization
+ *================================================================*/
+static void
+awe_initialize(void)
+{
+	unsigned short data;
+	DECL_INTR_FLAGS(flags);
+
+	DEBUG(0,printk("AWE32: initializing..\n"));
+	DISABLE_INTR(flags);
+
+	/* check for an error condition */
+	data = awe_peek(AWE_U1);
+	if (!(data & 0x000F) == 0x000C) {
+		FATALERR(printk("AWE32: can't initialize AWE32\n"));
+	}
+
+	/* initialize hardware configuration */
+	awe_poke(AWE_HWCF1, 0x0059);
+	awe_poke(AWE_HWCF2, 0x0020);
+
+	/* disable audio output */
+	awe_poke(AWE_HWCF3, 0x0000);
+
+	/* initialize audio channels */
+	awe_init_audio();
+
+	/* initialize init array */
+	awe_init_dma();
+	awe_init_array();
+
+	/* check DRAM memory size */
+	awe_mem_size = awe_check_dram();
+
+	/* initialize the FM section of the AWE32 */
+	awe_init_fm();
+
+	/* set up voice envelopes */
+	awe_tweak();
+
+	/* enable audio */
+	awe_poke(AWE_HWCF3, 0x0004);
+
+	data = awe_peek(AWE_HWCF2);
+	if (~data & 0x40) {
+		FATALERR(printk("AWE32: Unable to initialize AWE32.\n"));
+	}
+
+	RESTORE_INTR(flags);
+}
+
+
+/*================================================================
+ * AWE32 voice parameters
+ *================================================================*/
+
+/* initialize voice_info record */
+static void
+awe_init_voice_info(awe_voice_info *vp)
+{
+	vp->sf_id = 0;
+	vp->sample = 0;
+	vp->rate_offset = 0;
+
+	vp->start = 0;
+	vp->end = 0;
+	vp->loopstart = 0;
+	vp->loopend = 0;
+	vp->mode = 0;
+	vp->root = 60;
+	vp->tune = 0;
+	vp->low = 0;
+	vp->high = 127;
+	vp->vellow = 0;
+	vp->velhigh = 127;
+
+	vp->fixkey = -1;
+	vp->fixvel = -1;
+	vp->fixpan = -1;
+	vp->pan = -1;
+
+	vp->exclusiveClass = 0;
+	vp->amplitude = 127;
+	vp->attenuation = 0;
+	vp->scaleTuning = 100;
+
+	awe_init_voice_parm(&vp->parm);
+}
+
+/* initialize voice_parm record:
+ * Env1/2: delay=0, attack=0, hold=0, sustain=0, decay=0, release=0.
+ * Vibrato and Tremolo effects are zero.
+ * Cutoff is maximum.
+ * Chorus and Reverb effects are zero.
+ */
+static void
+awe_init_voice_parm(awe_voice_parm *pp)
+{
+	pp->moddelay = 0x8000;
+	pp->modatkhld = 0x7f7f;
+	pp->moddcysus = 0x7f7f;
+	pp->modrelease = 0x807f;
+	pp->modkeyhold = 0;
+	pp->modkeydecay = 0;
+
+	pp->voldelay = 0x8000;
+	pp->volatkhld = 0x7f7f;
+	pp->voldcysus = 0x7f7f;
+	pp->volrelease = 0x807f;
+	pp->volkeyhold = 0;
+	pp->volkeydecay = 0;
+
+	pp->lfo1delay = 0x8000;
+	pp->lfo2delay = 0x8000;
+	pp->pefe = 0;
+
+	pp->fmmod = 0;
+	pp->tremfrq = 0;
+	pp->fm2frq2 = 0;
+
+	pp->cutoff = 0xff;
+	pp->filterQ = 0;
+
+	pp->chorus = 0;
+	pp->reverb = 0;
+}	
+
+
+/* convert frequency mHz to abstract cents (= midi key * 100) */
+static int
+freq_to_note(int mHz)
+{
+	/* abscents = log(mHz/8176) / log(2) * 1200 */
+	unsigned long max_val = (unsigned long)0xffffffff / 10000;
+	int i, times;
+	unsigned long base;
+	unsigned long freq;
+	int note, tune;
+
+	if (mHz == 0)
+		return 0;
+	if (mHz < 0)
+		return 12799; /* maximum */
+
+	freq = mHz;
+	note = 0;
+	for (base = 8176 * 2; freq >= base; base *= 2) {
+		note += 12;
+		if (note >= 128) /* over maximum */
+			return 12799;
+	}
+	base /= 2;
+
+	/* to avoid overflow... */
+	times = 10000;
+	while (freq > max_val) {
+		max_val *= 10;
+		times /= 10;
+		base /= 10;
+	}
+
+	freq = freq * times / base;
+	for (i = 0; i < 12; i++) {
+		if (freq < semitone_tuning[i+1])
+			break;
+		note++;
+	}
+
+	tune = 0;
+	freq = freq * 10000 / semitone_tuning[i];
+	for (i = 0; i < 100; i++) {
+		if (freq < cent_tuning[i+1])
+			break;
+		tune++;
+	}
+
+	return note * 100 + tune;
+}
+
+
+/* convert Hz to AWE32 rate offset:
+ * sample pitch offset for the specified sample rate
+ * rate=44100 is no offset, each 4096 is 1 octave (twice).
+ * eg, when rate is 22050, this offset becomes -4096.
+ */
+static int
+calc_rate_offset(int Hz)
+{
+	/* offset = log(Hz / 44100) / log(2) * 4096 */
+	int freq, base, i;
+
+	/* maybe smaller than max (44100Hz) */
+	if (Hz <= 0 || Hz >= 44100) return 0;
+
+	base = 0;
+	for (freq = Hz * 2; freq < 44100; freq *= 2)
+		base++;
+	base *= 1200;
+
+	freq = 44100 * 10000 / (freq/2);
+	for (i = 0; i < 12; i++) {
+		if (freq < semitone_tuning[i+1])
+			break;
+		base += 100;
+	}
+	freq = freq * 10000 / semitone_tuning[i];
+	for (i = 0; i < 100; i++) {
+		if (freq < cent_tuning[i+1])
+			break;
+		base++;
+	}
+	return -base * 4096 / 1200;
+}
+
+
+/*----------------------------------------------------------------
+ * convert envelope time parameter to AWE32 raw parameter
+ *----------------------------------------------------------------*/
+
+/* attack & decay/release time table (mHz) */
+static short attack_time_tbl[128] = {
+32767, 5939, 3959, 2969, 2375, 1979, 1696, 1484, 1319, 1187, 1079, 989, 913, 848, 791, 742,
+ 698, 659, 625, 593, 565, 539, 516, 494, 475, 456, 439, 424, 409, 395, 383, 371,
+ 359, 344, 330, 316, 302, 290, 277, 266, 255, 244, 233, 224, 214, 205, 196, 188,
+ 180, 173, 165, 158, 152, 145, 139, 133, 127, 122, 117, 112, 107, 103, 98, 94,
+ 90, 86, 83, 79, 76, 73, 69, 67, 64, 61, 58, 56, 54, 51, 49, 47,
+ 45, 43, 41, 39, 38, 36, 35, 33, 32, 30, 29, 28, 27, 25, 24, 23,
+ 22, 21, 20, 20, 19, 18, 17, 16, 16, 15, 14, 14, 13, 13, 12, 11,
+ 11, 10, 10, 10, 9, 9, 8, 8, 8, 7, 7, 7, 6, 6, 6, 0,
+};
+
+static short decay_time_tbl[128] = {
+32767, 3651, 3508, 3371, 3239, 3113, 2991, 2874, 2761, 2653, 2550, 2450, 2354, 2262, 2174, 2089,
+ 2007, 1928, 1853, 1781, 1711, 1644, 1580, 1518, 1459, 1401, 1347, 1294, 1243, 1195, 1148, 1103,
+ 1060, 1018, 979, 940, 904, 868, 834, 802, 770, 740, 711, 683, 657, 631, 606, 582,
+ 560, 538, 517, 496, 477, 458, 440, 423, 407, 391, 375, 361, 347, 333, 320, 307,
+ 295, 284, 273, 262, 252, 242, 232, 223, 215, 206, 198, 190, 183, 176, 169, 162,
+ 156, 150, 144, 138, 133, 128, 123, 118, 113, 109, 104, 100, 96, 93, 89, 85,
+ 82, 79, 76, 73, 70, 67, 64, 62, 60, 57, 55, 53, 51, 49, 47, 45,
+ 43, 41, 40, 38, 37, 35, 34, 32, 31, 30, 29, 28, 27, 25, 24, 0,
+};
+
+/*
+static int
+calc_parm_delay(int msec)
+{
+	return (0x8000 - msec * 1000 / 725);
+}
+*/
+
+static int
+calc_parm_hold(int msec)
+{
+	int val = 0x7f - (unsigned char)(msec / 92);
+	if (val < 1) val = 1;
+	if (val > 127) val = 127;
+	return val;
+}
+
+static int
+calc_parm_attack(int msec)
+{
+	return calc_parm_search(msec, attack_time_tbl);
+}
+
+static int
+calc_parm_decay(int msec)
+{
+	return calc_parm_search(msec, decay_time_tbl);
+}
+
+static int
+calc_parm_search(int msec, short *table)
+{
+	int left = 0, right = 127, mid;
+	while (left < right) {
+		mid = (left + right) / 2;
+		if (msec < (int)table[mid])
+			left = mid + 1;
+		else
+			right = mid;
+	}
+	return left;
+}
+
+
+/*================================================================
+ * effects table
+ *================================================================*/
+
+/* set an effect value */
+#define FX_SET(v,type,value) \
+(voices[v].fx_flags[(type)/8] |= (1<<((type)%8)),\
+ voices[v].fx[type] = (value))
+/* check the effect value is set */
+#define FX_ON(v,type)	(voices[v].fx_flags[(type)/8] & (1<<((type)%8)))
+
+#if 0
+#define FX_BYTE(v,type,value)\
+	(FX_ON(v,type) ? (unsigned char)voices[v].fx[type] :\
+	 (unsigned char)(value))
+#define FX_WORD(v,type,value)\
+	(FX_ON(v,type) ? (unsigned short)voices[v].fx[type] :\
+	 (unsigned short)(value))
+
+#else
+
+/* get byte effect value */
+static unsigned char FX_BYTE(int v, int type, unsigned char value)
+{
+	unsigned char tmp;
+	if (FX_ON(v,type))
+		tmp = (unsigned char)voices[v].fx[type];
+	else
+		tmp = value;
+	DEBUG(4,printk("AWE32: [-- byte(%d) = %x]\n", type, tmp));
+	return tmp;
+}
+
+/* get word effect value */
+static unsigned short FX_WORD(int v, int type, unsigned short value)
+{
+	unsigned short tmp;
+	if (FX_ON(v,type))
+		tmp = (unsigned short)voices[v].fx[type];
+	else
+		tmp = value;
+	DEBUG(4,printk("AWE32: [-- word(%d) = %x]\n", type, tmp));
+	return tmp;
+}
+
+#endif
+
+/* get word (upper=type1/lower=type2) effect value */
+static unsigned short FX_COMB(int v, int type1, int type2, unsigned short value)
+{
+	unsigned short tmp;
+	if (FX_ON(v, type1))
+		tmp = (unsigned short)(voices[v].fx[type1]) << 8;
+	else
+		tmp = value & 0xff00;
+	if (FX_ON(v, type2))
+		tmp |= (unsigned short)(voices[v].fx[type2]) & 0xff;
+	else
+		tmp |= value & 0xff;
+	DEBUG(4,printk("AWE32: [-- comb(%d/%d) = %x]\n", type1, type2, tmp));
+	return tmp;
+}
+
+/* address offset */
+static long
+FX_OFFSET(int voice, int lo, int hi)
+{
+	awe_voice_info *vp;
+	long addr;
+	if ((vp = voices[voice].sample) == NULL || vp->index < 0)
+		return 0;
+
+	addr = 0;
+	if (FX_ON(voice, hi)) {
+		addr = (short)voices[voice].fx[hi];
+		addr = addr << 15;
+	}
+	if (FX_ON(voice, lo))
+		addr += (short)voices[voice].fx[lo];
+	if (!(vp->mode & (AWE_SAMPLE_8BITS<<6)))
+		addr /= 2;
+	return addr;
+}
+
+/* converter function table for realtime paramter change */
+
+typedef void (*fx_affect_func)(int voice);
+static fx_affect_func fx_realtime[] = {
+	/* env1: delay, attack, hold, decay, release, sustain, pitch, cutoff*/
+	NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL,
+	/* env2: delay, attack, hold, decay, release, sustain */
+	NULL, NULL, NULL, NULL, NULL, NULL,
+	/* lfo1: delay, freq, volume, pitch, cutoff */
+	NULL, awe_fx_tremfrq, awe_fx_tremfrq, awe_fx_fmmod, awe_fx_fmmod,
+	/* lfo2: delay, freq, pitch */
+	NULL, awe_fx_fm2frq2, awe_fx_fm2frq2,
+	/* global: initpitch, chorus, reverb, cutoff, filterQ */
+	awe_fx_initpitch, NULL, NULL, awe_fx_cutoff, NULL,
+	/* sample: start, loopstart, loopend */
+	NULL, NULL, NULL,
+};
+
+
+/*================================================================
+ * turn on/off sample
+ *================================================================*/
+
+static void
+awe_note_on(int voice)
+{
+	unsigned long temp;
+	long addr;
+	unsigned short tmp2;
+	awe_voice_info *vp;
+
+	/* A voice sample must assigned before calling */
+	if ((vp = voices[voice].sample) == NULL || vp->index < 0)
+		return;
+
+	/* channel to be silent and idle */
+	awe_poke(AWE_DCYSUSV(voice), 0x0080);
+	awe_poke(AWE_VTFT(voice), 0);
+	awe_poke(AWE_CVCF(voice), 0);
+	awe_poke(AWE_PTRX(voice), 0);
+	awe_poke(AWE_CPF(voice), 0);
+
+	/* modulation & volume envelope */
+	awe_poke(AWE_ENVVAL(voice),
+		 FX_WORD(voice, AWE_FX_ENV1_DELAY, vp->parm.moddelay));
+	awe_poke(AWE_ATKHLD(voice),
+		 FX_COMB(voice, AWE_FX_ENV1_ATTACK, AWE_FX_ENV1_HOLD,
+			 vp->parm.modatkhld));
+	awe_poke(AWE_DCYSUS(voice),
+		 FX_COMB(voice, AWE_FX_ENV1_SUSTAIN, AWE_FX_ENV1_DECAY,
+			  vp->parm.moddcysus));
+	awe_poke(AWE_ENVVOL(voice),
+		 FX_WORD(voice, AWE_FX_ENV2_DELAY, vp->parm.voldelay));
+	awe_poke(AWE_ATKHLDV(voice),
+		 FX_COMB(voice, AWE_FX_ENV2_ATTACK, AWE_FX_ENV2_HOLD,
+			 vp->parm.volatkhld));
+	/* decay/sustain parameter for volume envelope must be set at last */
+
+	/* pitch offset */
+	awe_poke(AWE_IP(voice), voices[voice].apitch);
+	DEBUG(3,printk("AWE32: [-- pitch=%x]\n", voices[voice].apitch));
+
+	/* cutoff and volume */
+	tmp2 = FX_BYTE(voice, AWE_FX_CUTOFF, vp->parm.cutoff);
+	tmp2 = (tmp2 << 8) | voices[voice].avol;
+	awe_poke(AWE_IFATN(voice), tmp2);
+
+	/* modulation envelope heights */
+	awe_poke(AWE_PEFE(voice),
+		 FX_COMB(voice, AWE_FX_ENV1_PITCH, AWE_FX_ENV1_CUTOFF,
+			 vp->parm.pefe));
+
+	/* lfo1/2 delay */
+	awe_poke(AWE_LFO1VAL(voice),
+		 FX_WORD(voice, AWE_FX_LFO1_DELAY, vp->parm.lfo1delay));
+	awe_poke(AWE_LFO2VAL(voice),
+		 FX_WORD(voice, AWE_FX_LFO2_DELAY, vp->parm.lfo2delay));
+
+	/* lfo1 pitch & cutoff shift */
+	awe_poke(AWE_FMMOD(voice),
+		 FX_COMB(voice, AWE_FX_LFO1_PITCH, AWE_FX_LFO1_CUTOFF,
+			 vp->parm.fmmod));
+	/* lfo1 volume & freq */
+	awe_poke(AWE_TREMFRQ(voice),
+		 FX_COMB(voice, AWE_FX_LFO1_VOLUME, AWE_FX_LFO1_FREQ,
+			 vp->parm.tremfrq));
+	/* lfo2 pitch & freq */
+	awe_poke(AWE_FM2FRQ2(voice),
+		 FX_COMB(voice, AWE_FX_LFO2_PITCH, AWE_FX_LFO2_FREQ,
+			 vp->parm.fm2frq2));
+
+	/* pan & loop start */
+	 awe_set_pan(voice, 1);
+
+	/* chorus & loop end (chorus 8bit, MSB) */
+	addr = vp->loopend - 1;
+	addr += FX_OFFSET(voice, AWE_FX_LOOP_END,
+			  AWE_FX_COARSE_LOOP_END);
+	temp = FX_BYTE(voice, AWE_FX_CHORUS, vp->parm.chorus);
+	temp = (temp <<24) | (unsigned long)addr;
+	awe_poke_dw(AWE_CSL(voice), temp);
+
+	/* Q & current address (Q 4bit value, MSB) */
+	addr = vp->start - 1;
+	addr += FX_OFFSET(voice, AWE_FX_SAMPLE_START,
+			  AWE_FX_COARSE_SAMPLE_START);
+	temp = FX_BYTE(voice, AWE_FX_FILTERQ, vp->parm.filterQ);
+	temp = (temp<<28) | (unsigned long)addr;
+	awe_poke_dw(AWE_CCCA(voice), temp);
+
+	/* reset volume */
+	awe_poke_dw(AWE_VTFT(voice), 0x0000FFFF);
+	awe_poke_dw(AWE_CVCF(voice), 0x0000FFFF);
+
+	/* turn on envelope */
+	awe_poke(AWE_DCYSUSV(voice),
+		 FX_COMB(voice, AWE_FX_ENV2_SUSTAIN, AWE_FX_ENV2_DECAY,
+			  vp->parm.voldcysus));
+	/* set chorus */
+	temp = FX_BYTE(voice, AWE_FX_REVERB, vp->parm.reverb);
+	temp = (awe_peek_dw(AWE_PTRX(voice)) & 0xffff0000) | (temp<<8);
+	awe_poke_dw(AWE_PTRX(voice), temp);
+	awe_poke_dw(AWE_CPF(voice), 0x40000000);
+
+	DEBUG(3,printk("AWE32: [-- start=%x loop=%x]\n",
+		       (int)vp->start, (int)vp->loopstart));
+}
+
+/* turn off the voice */
+static void
+awe_note_off(int voice)
+{
+	awe_voice_info *vp;
+	unsigned short tmp;
+	if ((vp = voices[voice].sample) == NULL || !voices[voice].state)
+		return;
+	if (FX_ON(voice, AWE_FX_ENV1_RELEASE))
+		tmp = 0x8000 | voices[voice].fx[AWE_FX_ENV1_RELEASE];
+	else
+		tmp = vp->parm.modrelease;
+	awe_poke(AWE_DCYSUS(voice), tmp);
+	if (FX_ON(voice, AWE_FX_ENV2_RELEASE))
+		tmp = 0x8000 | voices[voice].fx[AWE_FX_ENV2_RELEASE];
+	else
+		tmp = vp->parm.volrelease;
+	awe_poke(AWE_DCYSUSV(voice), tmp);
+}
+
+/* force to terminate the voice (no releasing echo) */
+static void
+awe_terminate(int voice)
+{
+	awe_poke(AWE_DCYSUSV(voice), 0x807F);
+}
+
+
+/* turn off other voices with the same exclusive class (for drums) */
+static void
+awe_exclusive_off(int voice)
+{
+	int i, excls;
+
+	if (voices[voice].sample == NULL) /* no sample */
+		return;
+	excls = voices[voice].sample->exclusiveClass;
+	if (excls == 0)	/* not exclusive */
+		return;
+
+	/* turn off voices with the same class */
+	for (i = 0; i < awe_max_voices; i++) {
+		if (i != voice && voices[voice].state &&
+		    voices[i].sample &&
+		    voices[i].sample->exclusiveClass == excls) {
+			DEBUG(4,printk("AWE32: [exoff(%d)]\n", i));
+			awe_note_off(i);
+			awe_voice_init(i, 1);
+		}
+	}
+}
+
+
+/*================================================================
+ * change the parameters of an audible voice
+ *================================================================*/
+
+/* change pitch */
+static void
+awe_set_pitch(int voice)
+{
+	if (!voices[voice].state) return;
+	awe_poke(AWE_IP(voice), voices[voice].apitch);
+}
+
+/* change volume */
+static void
+awe_set_volume(int voice)
+{
+	awe_voice_info *vp;
+	unsigned short tmp2;
+	if (!voices[voice].state) return;
+	if ((vp = voices[voice].sample) == NULL || vp->index < 0)
+		return;
+	tmp2 = FX_BYTE(voice, AWE_FX_CUTOFF, vp->parm.cutoff);
+	tmp2 = (tmp2 << 8) | voices[voice].avol;
+	awe_poke(AWE_IFATN(voice), tmp2);
+}
+
+/* change pan; this could make a click noise.. */
+static void
+awe_set_pan(int voice, int forced)
+{
+	unsigned long temp;
+	long addr;
+	awe_voice_info *vp;
+
+	if (!voices[voice].state && !forced) return;
+	if ((vp = voices[voice].sample) == NULL || vp->index < 0)
+		return;
+
+	/* pan & loop start (pan 8bit, MSB, 0:right, 0xff:left) */
+	if (vp->fixpan > 0)	/* 0-127 */
+		temp = 255 - (int)vp->fixpan * 2;
+	else {
+		int pos = 0;
+		if (vp->pan >= 0) /* 0-127 */
+			pos = (int)vp->pan * 2 - 128;
+		pos += voices[voice].panning; /* -128 - 127 */
+		pos = 127 - pos;
+		if (pos < 0)
+			temp = 0;
+		else if (pos > 255)
+			temp = 255;
+		else
+			temp = pos;
+	}
+	addr = vp->loopstart - 1;
+	addr += FX_OFFSET(voice, AWE_FX_LOOP_START,
+			  AWE_FX_COARSE_LOOP_START);
+	temp = (temp<<24) | (unsigned long)addr;
+	awe_poke_dw(AWE_PSST(voice), temp);
+}
+
+/* effects change during playing */
+static void
+awe_fx_fmmod(int voice)
+{
+	awe_voice_info *vp;
+	if (!voices[voice].state) return;
+	if ((vp = voices[voice].sample) == NULL || vp->index < 0)
+		return;
+	awe_poke(AWE_FMMOD(voice),
+		 FX_COMB(voice, AWE_FX_LFO1_PITCH, AWE_FX_LFO1_CUTOFF,
+			 vp->parm.fmmod));
+}
+
+static void
+awe_fx_tremfrq(int voice)
+{
+	awe_voice_info *vp;
+	if (!voices[voice].state) return;
+	if ((vp = voices[voice].sample) == NULL || vp->index < 0)
+		return;
+	awe_poke(AWE_TREMFRQ(voice),
+		 FX_COMB(voice, AWE_FX_LFO1_VOLUME, AWE_FX_LFO1_FREQ,
+			 vp->parm.tremfrq));
+}
+
+static void
+awe_fx_fm2frq2(int voice)
+{
+	awe_voice_info *vp;
+	if (!voices[voice].state) return;
+	if ((vp = voices[voice].sample) == NULL || vp->index < 0)
+		return;
+	awe_poke(AWE_FM2FRQ2(voice),
+		 FX_COMB(voice, AWE_FX_LFO2_PITCH, AWE_FX_LFO2_FREQ,
+			 vp->parm.fm2frq2));
+}
+
+static void
+awe_fx_cutoff(int voice)
+{
+	unsigned short tmp2;
+	awe_voice_info *vp;
+	if (!voices[voice].state) return;
+	if ((vp = voices[voice].sample) == NULL || vp->index < 0)
+		return;
+	tmp2 = FX_BYTE(voice, AWE_FX_CUTOFF, vp->parm.cutoff);
+	tmp2 = (tmp2 << 8) | voices[voice].avol;
+	awe_poke(AWE_IFATN(voice), tmp2);
+}
+
+static void
+awe_fx_initpitch(int voice)
+{
+	if (!voices[voice].state) return;
+	if (FX_ON(voice, AWE_FX_INIT_PITCH)) {
+		DEBUG(3,printk("AWE32: initpitch ok\n"));
+	} else {
+		DEBUG(3,printk("AWE32: BAD initpitch %d\n", AWE_FX_INIT_PITCH));
+	}
+	awe_calc_pitch(voice);
+	awe_poke(AWE_IP(voice), voices[voice].apitch);
+}
+
+
+/*================================================================
+ * calculate pitch offset
+ *----------------------------------------------------------------
+ * 0xE000 is no pitch offset at 44100Hz sample.
+ * Every 4096 is one octave.
+ *================================================================*/
+
+static void
+awe_calc_pitch(int voice)
+{
+	voice_info *vp = &voices[voice];
+	awe_voice_info *ap;
+	int offset;
+
+	/* search voice information */
+	if ((ap = vp->sample) == NULL)
+			return;
+	if (ap->index < 0) {
+		if (awe_set_sample(ap) < 0)
+			return;
+	}
+
+	/* calculate offset */
+	if (ap->fixkey >= 0) {
+		DEBUG(3,printk("AWE32: p-> fixkey(%d) tune(%d)\n", ap->fixkey, ap->tune));
+		offset = (ap->fixkey - ap->root) * 4096 / 12;
+	} else {
+		DEBUG(3,printk("AWE32: p(%d)-> root(%d) tune(%d)\n", vp->note, ap->root, ap->tune));
+		offset = (vp->note - ap->root) * 4096 / 12;
+		DEBUG(4,printk("AWE32: p-> ofs=%d\n", offset));
+	}
+	offset += ap->tune * 4096 / 1200;
+	DEBUG(4,printk("AWE32: p-> tune+ ofs=%d\n", offset));
+	if (vp->bender != 0) {
+		DEBUG(3,printk("AWE32: p-> bend(%d) %d\n", voice, vp->bender));
+		/* (819200: 1 semitone) ==> (4096: 12 semitones) */
+		offset += vp->bender * vp->bender_range / 2400;
+	}
+	offset = (offset * ap->scaleTuning) / 100;
+	DEBUG(4,printk("AWE32: p-> scale* ofs=%d\n", offset));
+
+	/* add initial pitch correction */
+	if (FX_ON(voice, AWE_FX_INIT_PITCH)) {
+		DEBUG(3,printk("AWE32: fx_pitch(%d) %d\n", voice, vp->fx[AWE_FX_INIT_PITCH]));
+		offset += vp->fx[AWE_FX_INIT_PITCH];
+	}
+
+	/* 0xe000: root pitch */
+	vp->apitch = 0xe000 + ap->rate_offset + offset;
+	DEBUG(4,printk("AWE32: p-> sum aofs=%x, rate_ofs=%d\n", vp->apitch, ap->rate_offset));
+	if (vp->apitch > 0xffff)
+		vp->apitch = 0xffff;
+	if (vp->apitch < 0)
+		vp->apitch = 0;
+}
+
+
+static void
+awe_calc_pitch_from_freq(int voice, int freq)
+{
+	voice_info *vp = &voices[voice];
+	awe_voice_info *ap;
+	int offset;
+	int note;
+
+	/* search voice information */
+	if ((ap = vp->sample) == NULL)
+		return;
+	if (ap->index < 0) {
+		if (awe_set_sample(ap) < 0)
+			return;
+	}
+	note = freq_to_note(freq);
+	offset = (note - ap->root * 100 + ap->tune) * 4096 / 1200;
+	offset = (offset * ap->scaleTuning) / 100;
+	if (FX_ON(voice, AWE_FX_INIT_PITCH))
+		offset += vp->fx[AWE_FX_INIT_PITCH];
+	vp->apitch = 0xe000 + ap->rate_offset + offset;
+	if (vp->apitch > 0xffff)
+		vp->apitch = 0xffff;
+	if (vp->apitch < 0)
+		vp->apitch = 0;
+}
+
+/*================================================================
+ * calculate volume attenuation
+ *----------------------------------------------------------------
+ * Voice volume is controlled by volume attenuation parameter.
+ * So volume becomes maximum when avol is 0 (no attenuation), and
+ * minimum when 255 (-96dB or silence).
+ *================================================================*/
+
+static int vol_table[128] = {
+	255,111,95,86,79,74,70,66,63,61,58,56,54,52,50,49,
+	47,46,45,43,42,41,40,39,38,37,36,35,34,34,33,32,
+	31,31,30,29,29,28,27,27,26,26,25,24,24,23,23,22,
+	22,21,21,21,20,20,19,19,18,18,18,17,17,16,16,16,
+	15,15,15,14,14,14,13,13,13,12,12,12,11,11,11,10,
+	10,10,10,9,9,9,8,8,8,8,7,7,7,7,6,6,
+	6,6,5,5,5,5,5,4,4,4,4,3,3,3,3,3,
+	2,2,2,2,2,1,1,1,1,1,0,0,0,0,0,0,
+};
+
+static void
+awe_calc_volume(int voice)
+{
+	voice_info *vp = &voices[voice];
+	awe_voice_info *ap;
+	int vol;
+
+	/* search voice information */
+	if ((ap = vp->sample) == NULL)
+		return;
+
+	ap = vp->sample;
+	if (ap->index < 0) {
+		if (awe_set_sample(ap) < 0)
+			return;
+	}
+	
+	if (vp->velocity < ap->vellow)
+		vp->velocity = ap->vellow;
+	else if (vp->velocity > ap->velhigh)
+		vp->velocity = ap->velhigh;
+
+	/* 0 - 127 */
+	vol = (vp->velocity * vp->main_vol * vp->expression_vol) / (127*127);
+	vol = vol * ap->amplitude / 127;
+	if (vol < 0) vol = 0;
+	if (vol > 127) vol = 127;
+
+	/* calc to attenuation */
+	vol = vol_table[vol];
+	vol = vol + (int)ap->attenuation + init_atten;
+	if (vol > 255) vol = 255;
+
+	vp->avol = vol;
+	DEBUG(3,printk("AWE32: [-- voice(%d) vol=%x]\n", voice, vol));
+}
+
+
+/*================================================================
+ * synth operation routines
+ *================================================================*/
+
+/* initialize the voice */
+static void
+awe_voice_init(int voice, int inst_only)
+{
+	if (! inst_only) {
+		/* clear voice parameters */
+		voices[voice].note = -1;
+		voices[voice].velocity = 0;
+		voices[voice].panning = 0; /* zero center */
+		voices[voice].bender = 0; /* zero tune skew */
+		voices[voice].bender_range = 200; /* sense * 100 */
+		voices[voice].main_vol = 127;
+		voices[voice].expression_vol = 127;
+		voices[voice].bank = AWE_DEFAULT_BANK;
+		voices[voice].instr = -1;
+		voices[voice].vrec = NULL;
+		voices[voice].sample = NULL;
+	}
+
+	/* clear voice mapping */
+	voices[voice].state = 0;
+	voice_alloc->map[voice] = 0;
+
+	/* emu8000 parameters */
+	voices[voice].apitch = 0;
+	voices[voice].avol = 255;
+
+	/* clear effects */
+	BZERO(voices[voice].fx_flags, sizeof(voices[voice].fx_flags));
+}
+
+
+/*----------------------------------------------------------------
+ * device open / close
+ *----------------------------------------------------------------*/
+
+/* open device:
+ *   reset status of all voices, and clear sample position flag
+ */
+static int
+awe_open(int dev, int mode)
+{
+	if (awe_busy)
+		return RET_ERROR(EBUSY);
+
+	awe_busy = 1;
+	awe_reset(dev);
+
+	/* clear sample position flag */
+	loaded_once = 0;
+
+	/* set GUS bank to default */
+	awe_gus_bank = AWE_DEFAULT_BANK;
+	return 0;
+}
+
+
+/* close device:
+ *   reset all voices again (terminate sounds)
+ */
+static void
+awe_close(int dev)
+{
+	awe_reset(dev);
+	awe_busy = 0;
+}
+
+
+/* sequencer I/O control:
+ */
+static int
+awe_ioctl(int dev, unsigned int cmd, caddr_t arg)
+{
+	switch (cmd) {
+	case SNDCTL_SYNTH_INFO:
+		awe_info.nr_voices = awe_max_voices;
+		IOCTL_TO_USER((char*)arg, 0, &awe_info, sizeof(awe_info));
+		return 0;
+		break;
+
+	case SNDCTL_SEQ_RESETSAMPLES:
+		awe_reset_samples();
+		awe_reset(dev); /* better to reset emu8k chip... */
+		return 0;
+		break;
+
+	case SNDCTL_SEQ_PERCMODE:
+		/* what's this? */
+		return 0;
+		break;
+
+	case SNDCTL_SYNTH_MEMAVL:
+		DEBUG(0,printk("AWE32: [ioctl memavl = %d]\n", (int)free_mem_ptr));
+		return awe_mem_size - free_mem_ptr*2;
+
+	default:
+		ERRMSG(printk("AWE32: unsupported ioctl %d\n", cmd));
+		return RET_ERROR(EINVAL);
+	}
+}
+
+
+/* kill a voice:
+ *   not terminate, just release the voice.
+ */
+static int
+awe_kill_note(int dev, int voice, int note, int velocity)
+{
+	awe_voice_info *vp;
+	DECL_INTR_FLAGS(flags);
+
+	DEBUG(2,printk("AWE32: [off(%d)]\n", voice));
+	if (voice < 0 || voice >= awe_max_voices)
+		      return RET_ERROR(EINVAL);
+	if ((vp = voices[voice].sample) == NULL)
+		return 0;
+       
+	if (!(vp->mode & AWE_MODE_NORELEASE)) {
+		DISABLE_INTR(flags);
+		awe_note_off(voice);
+		RESTORE_INTR(flags);
+	}
+	awe_voice_init(voice, 1);
+	return 0;
+}
+
+
+/* search the note with the specified key range */
+static awe_voice_info *
+awe_search_voice(int voice, int note)
+{
+	awe_voice_list *rec;
+	int maxc;
+
+	for (rec = voices[voice].vrec, maxc = AWE_MAX_INFOS;
+	     rec && maxc; rec = rec->next_instr, maxc--) {
+		if (rec->v.low <= note && note <= rec->v.high)
+			return &rec->v;
+	}
+	return NULL;
+}
+
+/* start a voice:
+ *   if note is 255, identical with aftertouch function.
+ *   Otherwise, start a voice with specified not and volume.
+ */
+static int
+awe_start_note(int dev, int v, int note_num, int volume)
+{
+	DECL_INTR_FLAGS(flags);
+
+	DEBUG(2,printk("AWE32: [on(%d) nt=%d vl=%d]\n", v, note_num, volume));
+	if (v < 0 || v >= awe_max_voices)
+		      return RET_ERROR(EINVAL);
+	/* an instrument must be set before starting a note */
+	if (voices[v].vrec == NULL) {
+		DEBUG(1,printk("AWE32: [-- vrec is null]\n"));
+		return 0;
+	}
+
+	if (note_num == 255) {
+		/* dynamic volume change; sample is already assigned */
+		if (! voices[v].state || voices[v].sample == NULL)
+			return 0;
+		/* calculate volume parameter */
+		voices[v].velocity = volume;
+		awe_calc_volume(v);
+		DISABLE_INTR(flags);
+		awe_set_volume(v);
+		RESTORE_INTR(flags);
+		return 0;
+	}
+	/* assign a sample with the corresponding note */
+	if ((voices[v].sample = awe_search_voice(v, note_num)) == NULL) {
+		DEBUG(1,printk("AWE32: [-- sample is null]\n"));
+		return 0;
+	}
+	/* calculate pitch & volume parameters */
+	voices[v].note = note_num;
+	voices[v].velocity = volume;
+	awe_calc_pitch(v);
+	awe_calc_volume(v);
+
+	DISABLE_INTR(flags);
+	/* turn off other voices (for drums) */
+	awe_exclusive_off(v);
+	/* turn on the voice */
+	awe_note_on(v);
+	voices[v].state = 1;	/* flag up */
+	RESTORE_INTR(flags);
+
+	return 0;
+}
+
+
+/* search instrument from preset table with the specified bank */
+static awe_voice_list *
+awe_search_instr(int bank, int preset)
+{
+	awe_voice_list *p;
+	int maxc;
+
+	for (maxc = AWE_MAX_INFOS, p = preset_table[preset];
+	     p && maxc; p = p->next_bank, maxc--) {
+		if (p->bank == bank)
+			return p;
+	}
+	return NULL;
+}
+
+
+/* assign the instrument to a voice */
+static int
+awe_set_instr(int dev, int voice, int instr_no)
+{
+	awe_voice_list *rec;
+
+	if (voice < 0 || voice >= awe_max_voices)
+		return RET_ERROR(EINVAL);
+
+	if (instr_no < 0 || instr_no >= AWE_MAX_PRESETS)
+		return RET_ERROR(EINVAL);
+
+	if ((rec = awe_search_instr(voices[voice].bank, instr_no)) == NULL) {
+		/* if bank is not defined, use the default bank 0 */
+		if (voices[voice].bank != AWE_DEFAULT_BANK &&
+		    (rec = awe_search_instr(AWE_DEFAULT_BANK, instr_no)) == NULL) {
+			DEBUG(1,printk("AWE32 Warning: can't find instrument %d\n", instr_no));
+			return 0;
+		}
+	}
+
+	voices[voice].instr = instr_no;
+	voices[voice].vrec = rec;
+	voices[voice].sample = NULL;  /* not set yet */
+
+	return 0;
+}
+
+
+/* reset all voices; terminate sounds and initialize parameters */
+static void
+awe_reset(int dev)
+{
+	int i;
+	/* don't turn off voice 31 and 32.  they are used also for FM voices */
+	for (i = 0; i < AWE_NORMAL_VOICES; i++) {
+		awe_terminate(i);
+		awe_voice_init(i, 0);
+	}
+	awe_init_fm();
+	awe_tweak();
+}
+
+
+/* hardware specific control:
+ *   GUS specific and AWE32 specific controls are available.
+ */
+static void
+awe_hw_control(int dev, unsigned char *event)
+{
+	int cmd = event[2];
+	if (cmd & _AWE_MODE_FLAG)
+		awe_hw_awe_control(dev, cmd & _AWE_MODE_VALUE_MASK, event);
+	else
+		awe_hw_gus_control(dev, cmd & _AWE_MODE_VALUE_MASK, event);
+}
+
+/* GUS compatible controls */
+static void
+awe_hw_gus_control(int dev, int cmd, unsigned char *event)
+{
+	int voice;
+	unsigned short p1;
+	short p2;
+	int plong;
+	DECL_INTR_FLAGS(flags);
+
+	voice = event[3];
+	p1 = *(unsigned short *) &event[4];
+	p2 = *(short *) &event[6];
+	plong = *(int*) &event[4];
+
+	switch (cmd) {
+	case _GUS_NUMVOICES:
+		if (p1 >= awe_max_voices)
+			printk("AWE32: num_voices: voices out of range %d\n", p1);
+		break;
+	case _GUS_VOICESAMPLE:
+		if (voice < awe_max_voices)
+			awe_set_instr(dev, voice, p1);
+		break;
+
+	case _GUS_VOICEON:
+		if (voice < awe_max_voices) {
+			DISABLE_INTR(flags);
+			awe_note_on(voice);
+			RESTORE_INTR(flags);
+		}
+		break;
+		
+	case _GUS_VOICEOFF:
+		if (voice < awe_max_voices) {
+			DISABLE_INTR(flags);
+			awe_note_off(voice);
+			RESTORE_INTR(flags);
+		}
+		break;
+		
+	case _GUS_VOICEMODE:
+		/* not supported */
+		break;
+
+	case _GUS_VOICEBALA:
+		/* -128 to 127 */
+		if (voice < awe_max_voices)
+			awe_panning(dev, voice, (short)p1);
+		break;
+
+	case _GUS_VOICEFREQ:
+		if (voice < awe_max_voices)
+			awe_calc_pitch_from_freq(voice, plong);
+		break;
+		
+	case _GUS_VOICEVOL:
+	case _GUS_VOICEVOL2:
+		/* not supported yet */
+		break;
+
+	case _GUS_RAMPRANGE:
+	case _GUS_RAMPRATE:
+	case _GUS_RAMPMODE:
+	case _GUS_RAMPON:
+	case _GUS_RAMPOFF:
+		/* volume ramping not supported */
+		break;
+
+	case _GUS_VOLUME_SCALE:
+		break;
+
+	case _GUS_VOICE_POS:
+		if (voice < awe_max_voices) {
+			FX_SET(voice, AWE_FX_SAMPLE_START, (short)(plong & 0x7fff));
+			FX_SET(voice, AWE_FX_COARSE_SAMPLE_START, (plong >> 15) & 0xffff);
+		}
+		break;
+	}
+}
+
+
+/* AWE32 specific controls */
+static void
+awe_hw_awe_control(int dev, int cmd, unsigned char *event)
+{
+	int voice;
+	unsigned short p1;
+	short p2;
+	int chn;
+
+	chn = event[1];
+	voice = event[3];
+	p1 = *(unsigned short *) &event[4];
+	p2 = *(short *) &event[6];
+	
+
+#ifdef AWE_DEBUG_ON
+	switch (cmd) {
+	case _AWE_DEBUG_MODE:
+		debug_mode = p1;
+		printk("AWE32: debug mode = %d\n", debug_mode);
+		break;
+#endif
+	case _AWE_REVERB_MODE:
+		if (p1 <= 7) {
+			reverb_mode = p1;
+			DEBUG(0,printk("AWE32: reverb mode %d\n", reverb_mode));
+			awe_set_reverb_mode(reverb_mode);
+		}
+		break;
+
+	case _AWE_CHORUS_MODE:
+		if (p1 <= 7) {
+			chorus_mode = p1;
+			DEBUG(0,printk("AWE32: chorus mode %d\n", chorus_mode));
+			awe_set_chorus_mode(chorus_mode);
+		}
+		break;
+		      
+	case _AWE_REMOVE_LAST_SAMPLES:
+		DEBUG(0,printk("AWE32: remove last samples\n"));
+		awe_remove_samples();
+		break;
+
+	case _AWE_INITIALIZE_CHIP:
+		awe_initialize();
+		break;
+
+	case _AWE_SEND_EFFECT:
+		if (voice < awe_max_voices && p1 < AWE_FX_END) {
+			FX_SET(voice, p1, p2);
+			DEBUG(0,printk("AWE32: effects (%d) %d %d\n", voice, p1, voices[voice].fx[p1]));
+			if (fx_realtime[p1]) {
+				DEBUG(0,printk("AWE32: fx_realtime (%d)\n", voice));
+				fx_realtime[p1](voice);
+			}
+		}
+		break;
+
+	case _AWE_TERMINATE_CHANNEL:
+		if (voice < awe_max_voices) {
+			DEBUG(0,printk("AWE32: terminate (%d)\n", voice));
+			awe_terminate(voice);
+			awe_voice_init(voice, 1);
+		}
+		break;
+
+	case _AWE_TERMINATE_ALL:
+		DEBUG(0,printk("AWE32: terminate all\n"));
+		awe_reset(0);
+		break;
+
+	case _AWE_INITIAL_VOLUME:
+		DEBUG(0,printk("AWE32: init attenuation %d\n", p1));
+		init_atten = p1;
+		break;
+
+	case _AWE_SET_GUS_BANK:
+		DEBUG(0,printk("AWE32: set gus bank %d\n", p1));
+		awe_gus_bank = p1;
+		break;
+		
+	default:
+		DEBUG(0,printk("AWE32: hw control cmd=%d voice=%d\n", cmd, voice));
+		break;
+	}
+}
+
+
+/*----------------------------------------------------------------
+ * load a sound patch:
+ *   three types of patches are accepted: AWE, GUS, and SYSEX.
+ *----------------------------------------------------------------*/
+
+static int
+awe_load_patch(int dev, int format, const char *addr,
+	       int offs, int count, int pmgr_flag)
+{
+	awe_patch_info patch;
+	int rc = 0;
+
+	if (format == GUS_PATCH) {
+		return awe_load_guspatch(addr, offs, count, pmgr_flag);
+	} else if (format == SYSEX_PATCH) {
+		/* no system exclusive message supported yet */
+		return 0;
+	} else if (format != AWE_PATCH) {
+		FATALERR(printk("AWE32 Error: Invalid patch format (key) 0x%x\n", format));
+		return RET_ERROR(EINVAL);
+	}
+	
+	if (count < sizeof(awe_patch_info)) {
+		FATALERR(printk("AWE32 Error: Patch header too short\n"));
+		return RET_ERROR(EINVAL);
+	}
+	COPY_FROM_USER(((char*)&patch) + offs, addr, offs, 
+		       sizeof(awe_patch_info) - offs);
+
+	count -= sizeof(awe_patch_info);
+	if (count < patch.len) {
+		FATALERR(printk("AWE32 Warning: Patch record too short (%d<%d)\n",
+		       count, (int)patch.len));
+		patch.len = count;
+	}
+	
+	switch (patch.type) {
+	case AWE_LOAD_INFO:
+		rc = awe_load_info(&patch, addr);
+		break;
+
+	case AWE_LOAD_DATA:
+		rc = awe_load_data(&patch, addr);
+		/*
+		if (!pmgr_flag && rc == 0)
+			pmgr_inform(dev, PM_E_PATCH_LOADED, instr, free_sample, 0, 0);
+		*/
+		break;
+
+	default:
+		FATALERR(printk("AWE32 Error: unknown patch format type %d\n",
+		       patch.type));
+		rc = RET_ERROR(EINVAL);
+	}
+
+	return rc;
+}
+
+
+/* load voice information data */
+static int
+awe_load_info(awe_patch_info *patch, const char *addr)
+{
+	awe_voice_list *rec, *curp;
+	long offset;
+	short i, nvoices;
+	unsigned char bank, instr;
+	int total_size;
+
+	if (patch->len < sizeof(awe_voice_rec)) {
+		FATALERR(printk("AWE32 Error: invalid patch info length\n"));
+		return RET_ERROR(EINVAL);
+	}
+
+	offset = sizeof(awe_patch_info);
+	GET_BYTE_FROM_USER(bank, addr, offset); offset++;
+	GET_BYTE_FROM_USER(instr, addr, offset); offset++;
+	GET_SHORT_FROM_USER(nvoices, addr, offset); offset+=2;
+
+	if (nvoices <= 0 || nvoices >= 100) {
+		FATALERR(printk("AWE32 Error: Illegal voice number %d\n", nvoices));
+		return RET_ERROR(EINVAL);
+	}
+	if (free_info + nvoices > AWE_MAX_INFOS) {
+		ERRMSG(printk("AWE32 Error: Too many voice informations\n"));
+		return RET_ERROR(ENOSPC);
+	}
+
+	total_size = sizeof(awe_voice_rec) + sizeof(awe_voice_info) * nvoices;
+	if (patch->len < total_size) {
+		ERRMSG(printk("AWE32 Error: patch length(%d) is smaller than nvoices(%d)\n",
+		       (int)patch->len, nvoices));
+		return RET_ERROR(EINVAL);
+	}
+
+	curp = awe_search_instr(bank, instr);
+	for (i = 0; i < nvoices; i++) {
+		rec = &infos[free_info + i];
+
+		rec->bank = bank;
+		rec->instr = instr;
+		if (i < nvoices - 1)
+			rec->next_instr = rec + 1;
+		else
+			rec->next_instr = curp;
+		rec->next_bank = NULL;
+
+		/* copy awe_voice_info parameters */
+		COPY_FROM_USER(&rec->v, addr, offset, sizeof(awe_voice_info));
+		offset += sizeof(awe_voice_info);
+		rec->v.sf_id = current_sf_id;
+		if (rec->v.mode & AWE_MODE_INIT_PARM)
+			awe_init_voice_parm(&rec->v.parm);
+		awe_set_sample(&rec->v);
+	}
+
+	/* prepend to top of the list */
+	infos[free_info].next_bank = preset_table[instr];
+	preset_table[instr] = &infos[free_info];
+	free_info += nvoices;
+
+	return 0;
+}
+
+
+/* load wave sample data */
+static int
+awe_load_data(awe_patch_info *patch, const char *addr)
+{
+	long offset;
+	int size;
+	int rc;
+
+	if (free_sample >= AWE_MAX_SAMPLES) {
+		ERRMSG(printk("AWE32 Error: Sample table full\n"));
+		return RET_ERROR(ENOSPC);
+	}
+
+	size = (patch->len - sizeof(awe_sample_info)) / 2;
+	offset = sizeof(awe_patch_info);
+	COPY_FROM_USER(&samples[free_sample], addr, offset,
+		       sizeof(awe_sample_info));
+	offset += sizeof(awe_sample_info);
+	if (size != samples[free_sample].size) {
+		ERRMSG(printk("AWE32 Warning: sample size differed (%d != %d)\n",
+		       (int)samples[free_sample].size, (int)size));
+		samples[free_sample].size = size;
+	}
+	if (samples[free_sample].size > 0)
+		if ((rc = awe_write_wave_data(addr, offset, size)) != 0)
+			return rc;
+
+	awe_check_loaded();
+	samples[free_sample].sf_id = current_sf_id;
+
+	free_sample++;
+	return 0;
+}
+
+/* check the other samples are already loaded */
+static void
+awe_check_loaded(void)
+{
+	if (!loaded_once) {
+		/* it's the first time */
+		last_sample = free_sample;
+		last_info = free_info;
+		current_sf_id++;
+		loaded_once = 1;
+	}
+}
+
+
+/*----------------------------------------------------------------*/
+
+static const char *readbuf_addr;
+static long readbuf_offs;
+static int readbuf_flags;
+
+#ifdef AWE_USE_BUFFERED_IO
+
+#define TMP_WAVBUF_SIZE		4096
+static unsigned short readbuf[TMP_WAVBUF_SIZE];
+static int readbuf_size, readbuf_cur, readbuf_left;
+
+/* read through temporary buffer */
+static unsigned short
+awe_readbuf_word(int pos)
+{
+	if (readbuf_left <= 0) {
+		int i;
+		if (readbuf_size - pos < TMP_WAVBUF_SIZE)
+			readbuf_left = readbuf_size - pos;
+		else
+			readbuf_left = TMP_WAVBUF_SIZE;
+		/* read from user buffer */
+		if (readbuf_flags & AWE_SAMPLE_8BITS) {
+			unsigned char *pbuf = (unsigned char *)readbuf;
+			COPY_FROM_USER(pbuf, readbuf_addr,
+				       readbuf_offs + pos, readbuf_left);
+			/* convert 8bit -> 16bit */
+			for (i = readbuf_left - 1; i >= 0; i--)
+				readbuf[i] = pbuf[i] << 8;
+		} else {
+			COPY_FROM_USER(readbuf, readbuf_addr,
+				       readbuf_offs + pos * 2, readbuf_left*2);
+		}
+
+		if (readbuf_flags & AWE_SAMPLE_UNSIGNED) {
+			/* unsigned -> signed */
+			for (i = 0; i < readbuf_left; i++)
+				readbuf[i] ^= 0x8000;
+		}
+		readbuf_cur = 0;
+	}
+
+	readbuf_left--;
+	return readbuf[readbuf_cur++];
+}
+
+#else  /* AWE_USE_BUFFERED_IO */
+
+#define awe_readbuf_word(pos)  awe_read_word(pos)
+
+#endif  /* AWE_USE_BUFFERED_IO */
+
+
+/* initialize read buffer */
+static void
+awe_init_readbuf(const char *addr, long offset, int size, int mode_flags)
+{
+	readbuf_addr = addr;
+	readbuf_offs = offset;
+	readbuf_flags = mode_flags;
+#ifdef AWE_USE_BUFFERED_IO
+	readbuf_size = size;
+	readbuf_left = 0;
+	readbuf_cur = 0;
+#endif
+}
+
+/* read directly from user buffer */
+static unsigned short
+awe_read_word(int pos)
+{
+	unsigned short c;
+	/* read from user buffer */
+	if (readbuf_flags & AWE_SAMPLE_8BITS) {
+		unsigned char cc;
+		GET_BYTE_FROM_USER(cc, readbuf_addr, readbuf_offs + pos);
+		c = cc << 8; /* convert 8bit -> 16bit */
+	} else {
+		GET_SHORT_FROM_USER(c, readbuf_addr, readbuf_offs + pos * 2);
+	}
+	if (readbuf_flags & AWE_SAMPLE_UNSIGNED)
+		c ^= 0x8000; /* unsigned -> signed */
+	return c;
+}
+
+
+
+#define BLANK_LOOP_START	8
+#define BLANK_LOOP_END		40
+#define BLANK_LOOP_SIZE		48
+
+/* loading onto memory */
+static int 
+awe_write_wave_data(const char *addr, long offset, int size)
+{
+	awe_sample_info *sp = &samples[free_sample];
+	int i, truesize;
+	int rc;
+	unsigned long csum1, csum2;
+	DECL_INTR_FLAGS(flags);
+
+	/* be sure loop points start < end */
+	if (sp->loopstart > sp->loopend) {
+		long tmp = sp->loopstart;
+		sp->loopstart = sp->loopend;
+		sp->loopend = tmp;
+	}
+
+	/* compute true data size to be loaded */
+	truesize = size;
+	if (sp->mode_flags & AWE_SAMPLE_BIDIR_LOOP)
+		truesize += sp->loopend - sp->loopstart;
+	if (sp->mode_flags & AWE_SAMPLE_NO_BLANK)
+		truesize += BLANK_LOOP_SIZE;
+	if (size > 0 && free_mem_ptr + truesize >= awe_mem_size/2) {
+		ERRMSG(printk("AWE32 Error: Sample memory full\n"));
+		return RET_ERROR(ENOSPC);
+	}
+
+	/* recalculate address offset */
+	sp->end -= sp->start;
+	sp->loopstart -= sp->start;
+	sp->loopend -= sp->start;
+	sp->size = truesize;
+
+	sp->start = free_mem_ptr + AWE_DRAM_OFFSET;
+	sp->end += free_mem_ptr + AWE_DRAM_OFFSET;
+	sp->loopstart += free_mem_ptr + AWE_DRAM_OFFSET;
+	sp->loopend += free_mem_ptr + AWE_DRAM_OFFSET;
+
+	DISABLE_INTR(flags);
+	if ((rc = awe_open_dram_for_write(free_mem_ptr)) != 0) {
+		RESTORE_INTR(flags);
+		return rc;
+	}
+
+	awe_init_readbuf(addr, offset, size, sp->mode_flags);
+	csum1 = 0;
+	for (i = 0; i < size; i++) {
+		unsigned short c;
+		c = awe_readbuf_word(i);
+		csum1 += c;
+		awe_write_dram(c);
+		if (i == sp->loopend &&
+		    (sp->mode_flags & AWE_SAMPLE_BIDIR_LOOP)) {
+			int looplen = sp->loopend - sp->loopstart;
+			/* copy reverse loop */
+			int k;
+			for (k = 0; k < looplen; k++) {
+				/* non-buffered data */
+				c = awe_read_word(i - k);
+				awe_write_dram(c);
+			}
+		}
+	}
+
+	/* if no blank loop is attached in the sample, add it */
+	if (sp->mode_flags & AWE_SAMPLE_NO_BLANK) {
+		for (i = 0; i < BLANK_LOOP_SIZE; i++)
+			awe_write_dram(0);
+		if (sp->mode_flags & AWE_SAMPLE_SINGLESHOT) {
+			sp->loopstart = sp->end + BLANK_LOOP_START;
+			sp->loopend = sp->end + BLANK_LOOP_END;
+		}
+		sp->size += BLANK_LOOP_SIZE;
+	}
+
+	awe_close_dram();
+	RESTORE_INTR(flags);
+	if (sp->checksum_flag) {
+#ifdef AWE_CHECKSUM_DATA
+		if (sp->checksum_flag != 2 && csum1 != sp->checksum) {
+			ERRMSG(printk("AWE32: [%d] checksum mismatch on data %x:%x\n",
+			       free_sample,
+			       (int)samples[free_sample].checksum,
+			       (int)csum1));
+			return RET_ERROR(NO_DATA_ERR);
+		}
+#endif /* AWE_CHECKSUM_DATA */
+#ifdef AWE_CHECKSUM_MEMORY
+		DISABLE_INTR(flags);
+		if (awe_open_dram_for_read(free_mem_ptr) == 0) {
+			csum2 = 0;
+			for (i = 0; i < size; i++) {
+				unsigned short c;
+				c = awe_peek(AWE_SMLD);
+				csum2 += c;
+			}
+			awe_close_dram_for_read();
+			if (csum2 != samples[free_sample].checksum) {
+				RESTORE_INTR(flags);
+				ERRMSG(printk("AWE32: [%d] checksum mismatch on DRAM %x:%x\n",
+					      free_sample,
+					      (int)samples[free_sample].checksum,
+					      (int)csum2));
+				return RET_ERROR(NO_DATA_ERR);
+			}
+		}
+		RESTORE_INTR(flags);
+#endif /* AWE_CHECKSUM_MEMORY */
+	}
+	free_mem_ptr += sp->size;
+
+	/* re-initialize FM passthrough */
+	DISABLE_INTR(flags);
+	awe_init_fm();
+	awe_tweak();
+	RESTORE_INTR(flags);
+
+	return 0;
+}
+
+
+/* calculate GUS envelope time:
+ * is this correct?  i have no idea..
+ */
+static int
+calc_gus_envelope_time(int rate, int start, int end)
+{
+	int r, p, t;
+	r = (3 - ((rate >> 6) & 3)) * 3;
+	p = rate & 0x3f;
+	t = end - start;
+	if (t < 0) t = -t;
+	if (13 > r)
+		t = t << (13 - r);
+	else
+		t = t >> (r - 13);
+	return (t * 10) / (p * 441);
+}
+
+#define calc_gus_sustain(val)  (0x7f - vol_table[(val)/2])
+#define calc_gus_attenuation(val)	vol_table[(val)/2]
+
+/* load GUS patch */
+static int
+awe_load_guspatch(const char *addr, int offs, int size, int pmgr_flag)
+{
+	struct patch_info patch;
+	awe_voice_list *rec, *curp;
+	long sizeof_patch;
+	int note;
+	int rc;
+
+	sizeof_patch = (long)&patch.data[0] - (long)&patch; /* header size */
+	if (free_sample >= AWE_MAX_SAMPLES) {
+		ERRMSG(printk("AWE32 Error: Sample table full\n"));
+		return RET_ERROR(ENOSPC);
+	}
+	if (free_info >= AWE_MAX_INFOS) {
+		ERRMSG(printk("AWE32 Error: Too many voice informations\n"));
+		return RET_ERROR(ENOSPC);
+	}
+	if (size < sizeof_patch) {
+		ERRMSG(printk("AWE32 Error: Patch header too short\n"));
+		return RET_ERROR(EINVAL);
+	}
+	COPY_FROM_USER(((char*)&patch) + offs, addr, offs, sizeof_patch - offs);
+	size -= sizeof_patch;
+	if (size < patch.len) {
+		FATALERR(printk("AWE32 Warning: Patch record too short (%d<%d)\n",
+		       size, (int)patch.len));
+		patch.len = size;
+	}
+
+	samples[free_sample].sf_id = 0;
+	samples[free_sample].sample = free_sample;
+	samples[free_sample].start = 0;
+	samples[free_sample].end = patch.len;
+	samples[free_sample].loopstart = patch.loop_start;
+	samples[free_sample].loopend = patch.loop_end;
+	samples[free_sample].size = patch.len;
+
+	/* set up mode flags */
+	samples[free_sample].mode_flags = 0;
+	if (!(patch.mode & WAVE_16_BITS))
+		samples[free_sample].mode_flags |= AWE_SAMPLE_8BITS;
+	if (patch.mode & WAVE_UNSIGNED)
+		samples[free_sample].mode_flags |= AWE_SAMPLE_UNSIGNED;
+	samples[free_sample].mode_flags |= AWE_SAMPLE_NO_BLANK;
+	if (!(patch.mode & (WAVE_LOOPING|WAVE_BIDIR_LOOP)))
+		samples[free_sample].mode_flags |= AWE_SAMPLE_SINGLESHOT;
+	if (patch.mode & WAVE_BIDIR_LOOP)
+		samples[free_sample].mode_flags |= AWE_SAMPLE_BIDIR_LOOP;
+
+	DEBUG(0,printk("AWE32: [sample %d mode %x]\n", patch.instr_no,
+		       samples[free_sample].mode_flags));
+	if (patch.mode & WAVE_16_BITS) {
+		/* convert to word offsets */
+		samples[free_sample].size /= 2;
+		samples[free_sample].end /= 2;
+		samples[free_sample].loopstart /= 2;
+		samples[free_sample].loopend /= 2;
+	}
+	samples[free_sample].checksum_flag = 0;
+	samples[free_sample].checksum = 0;
+
+	if ((rc = awe_write_wave_data(addr, sizeof_patch,
+				      samples[free_sample].size)) != 0)
+		return rc;
+
+	awe_check_loaded();
+	samples[free_sample].sf_id = current_sf_id;
+	free_sample++;
+
+	/* set up voice info */
+	rec = &infos[free_info];
+	awe_init_voice_info(&rec->v);
+	rec->v.sf_id = current_sf_id;
+	rec->v.sample = free_sample - 1; /* the last sample */
+	rec->v.rate_offset = calc_rate_offset(patch.base_freq);
+	note = freq_to_note(patch.base_note);
+	rec->v.root = note / 100;
+	rec->v.tune = -(note % 100);
+	rec->v.low = freq_to_note(patch.low_note) / 100;
+	rec->v.high = freq_to_note(patch.high_note) / 100;
+	DEBUG(1,printk("AWE32: [gus base offset=%d, note=%d, range=%d-%d(%d-%d)]\n",
+		       rec->v.rate_offset, note,
+		       rec->v.low, rec->v.high,
+	      patch.low_note, patch.high_note));
+	/* panning position; -128 - 127 => 0-127 */
+	rec->v.pan = (patch.panning + 128) / 2;
+
+	/* detuning is ignored */
+	/* 6points volume envelope */
+	if (patch.mode & WAVE_ENVELOPES) {
+		int attack, hold, decay, release;
+		attack = calc_gus_envelope_time
+			(patch.env_rate[0], 0, patch.env_offset[0]);
+		hold = calc_gus_envelope_time
+			(patch.env_rate[1], patch.env_offset[0],
+			 patch.env_offset[1]);
+		decay = calc_gus_envelope_time
+			(patch.env_rate[2], patch.env_offset[1],
+			 patch.env_offset[2]);
+		release = calc_gus_envelope_time
+			(patch.env_rate[3], patch.env_offset[1],
+			 patch.env_offset[4]);
+		release += calc_gus_envelope_time
+			(patch.env_rate[4], patch.env_offset[3],
+			 patch.env_offset[4]);
+		release += calc_gus_envelope_time
+			(patch.env_rate[5], patch.env_offset[4],
+			 patch.env_offset[5]);
+		rec->v.parm.volatkhld = (calc_parm_attack(attack) << 8) |
+			calc_parm_hold(hold);
+		rec->v.parm.voldcysus = (calc_gus_sustain(patch.env_offset[2]) << 8) |
+			calc_parm_decay(decay);
+		rec->v.parm.volrelease = 0x8000 | calc_parm_decay(release);
+		DEBUG(2,printk("AWE32: [gusenv atk=%d, hld=%d, dcy=%d, rel=%d]\n", attack, hold, decay, release));
+		rec->v.attenuation = calc_gus_attenuation(patch.env_offset[0]);
+	}
+
+	/* tremolo effect */
+	if (patch.mode & WAVE_TREMOLO) {
+		int rate = (patch.tremolo_rate * 1000 / 38) / 42;
+		rec->v.parm.tremfrq = ((patch.tremolo_depth / 2) << 8) | rate;
+		DEBUG(2,printk("AWE32: [gusenv tremolo rate=%d, dep=%d, tremfrq=%x]\n",
+			       patch.tremolo_rate, patch.tremolo_depth,
+			       rec->v.parm.tremfrq));
+	}
+	/* vibrato effect */
+	if (patch.mode & WAVE_VIBRATO) {
+		int rate = (patch.vibrato_rate * 1000 / 38) / 42;
+		rec->v.parm.fm2frq2 = ((patch.vibrato_depth / 6) << 8) | rate;
+		DEBUG(2,printk("AWE32: [gusenv vibrato rate=%d, dep=%d, tremfrq=%x]\n",
+			       patch.tremolo_rate, patch.tremolo_depth,
+			       rec->v.parm.tremfrq));
+	}
+	
+	/* scale_freq, scale_factor, volume, and fractions not implemented */
+
+	/* set the voice index */
+	awe_set_sample(&rec->v);
+
+	/* prepend to top of the list */
+	curp = awe_search_instr(awe_gus_bank, patch.instr_no);
+	rec->bank = awe_gus_bank;
+	rec->instr = patch.instr_no;
+	rec->next_instr = curp;
+	rec->next_bank = preset_table[rec->instr];
+	preset_table[rec->instr] = rec;
+	free_info++;
+
+	return 0;
+}
+
+
+/* remove samples with current sf_id from instrument list */
+static awe_voice_list *
+awe_get_removed_list(awe_voice_list *curp)
+{
+	awe_voice_list *lastp, **prevp;
+	int maxc;
+	lastp = curp;
+	prevp = &lastp;
+	for (maxc = AWE_MAX_INFOS;
+	     curp && maxc; curp = curp->next_instr, maxc--) {
+		if (curp->v.sf_id == current_sf_id)
+			*prevp = curp->next_instr;
+		else
+			prevp = &curp->next_instr;
+	}
+	return lastp;
+}
+
+
+/* remove last loaded samples */
+static void
+awe_remove_samples(void)
+{
+	awe_voice_list **prevp, *p, *nextp;
+	int maxc;
+	int i;
+
+	if (last_sample == free_sample && last_info == free_info)
+		return;
+
+	/* remove the records from preset table */
+	for (i = 0; i < AWE_MAX_PRESETS; i++) {
+		prevp = &preset_table[i];
+		for (maxc = AWE_MAX_INFOS, p = preset_table[i];
+		     p && maxc; p = nextp, maxc--) {
+			nextp = p->next_bank;
+			p = awe_get_removed_list(p);
+			if (p == NULL)
+				*prevp = nextp;
+			else {
+				*prevp = p;
+				prevp = &p->next_bank;
+			}
+		}
+	}
+
+	for (i = last_sample; i < free_sample; i++)
+		free_mem_ptr -= samples[i].size;
+
+	free_sample = last_sample;
+	free_info = last_info;
+	current_sf_id--;
+	loaded_once = 0;
+}
+
+
+/* search the specified sample */
+static short
+awe_set_sample(awe_voice_info *vp)
+{
+	int i;
+	for (i = 0; i < free_sample; i++) {
+		if (samples[i].sf_id == vp->sf_id &&
+		    samples[i].sample == vp->sample) {
+			/* set the actual sample offsets */
+			vp->start += samples[i].start;
+			vp->end += samples[i].end;
+			vp->loopstart += samples[i].loopstart;
+			vp->loopend += samples[i].loopend;
+			/* copy mode flags */
+			vp->mode |= (samples[i].mode_flags << 6);
+			/* set index */
+			vp->index = i;
+			return i;
+		}
+	}
+	return -1;
+}
+
+
+/* voice pressure change */
+static void
+awe_aftertouch(int dev, int voice, int pressure)
+{
+	DECL_INTR_FLAGS(flags);
+
+	DEBUG(2,printk("AWE32: [after(%d) %d]\n", voice, pressure));
+	if (voice < 0 || voice >= awe_max_voices)
+		return;
+	voices[voice].velocity = pressure;
+	awe_calc_volume(voice);
+	DISABLE_INTR(flags);
+	awe_set_volume(voice);
+	RESTORE_INTR(flags);
+}
+
+
+/* voice control change */
+static void
+awe_controller(int dev, int voice, int ctrl_num, int value)
+{
+	DECL_INTR_FLAGS(flags);
+
+	if (voice < 0 || voice >= awe_max_voices)
+		return;
+
+	switch (ctrl_num) {
+	case CTL_BANK_SELECT:
+		DEBUG(2,printk("AWE32: [bank(%d) %d]\n", voice, value));
+		voices[voice].bank = value;
+		break;
+
+	case CTRL_PITCH_BENDER:
+		DEBUG(2,printk("AWE32: [bend(%d) %d]\n", voice, value));
+		/* zero centered */
+		voices[voice].bender = value;
+		awe_calc_pitch(voice);
+		DISABLE_INTR(flags);
+		awe_set_pitch(voice);
+		RESTORE_INTR(flags);
+		break;
+
+	case CTRL_PITCH_BENDER_RANGE:
+		DEBUG(2,printk("AWE32: [range(%d) %d]\n", voice, value));
+		/* sense x 100 */
+		voices[voice].bender_range = value;
+		/* no audible pitch change yet.. */
+		break;
+
+	case CTL_EXPRESSION:
+		value /= 128;
+	case CTRL_EXPRESSION:
+		DEBUG(2,printk("AWE32: [expr(%d) %d]\n", voice, value));
+		/* 0 - 127 */
+		voices[voice].expression_vol = value;
+		awe_calc_volume(voice);
+		DISABLE_INTR(flags);
+		awe_set_volume(voice);
+		RESTORE_INTR(flags);
+		break;
+
+	case CTL_PAN:
+		DEBUG(2,printk("AWE32: [pan(%d) %d]\n", voice, value));
+		/* (0-127) -> signed 8bit */
+		voices[voice].panning = value * 2 - 128;
+		DISABLE_INTR(flags);
+		awe_set_pan(voice, 0);
+		RESTORE_INTR(flags);
+		break;
+
+	case CTL_MAIN_VOLUME:
+		value = (value * 127) / 16383;
+	case CTRL_MAIN_VOLUME:
+		DEBUG(2,printk("AWE32: [mainvol(%d) %d]\n", voice, value));
+		/* 0 - 127 */
+		voices[voice].main_vol = value;
+		awe_calc_volume(voice);
+		DISABLE_INTR(flags);
+		awe_set_volume(voice);
+		RESTORE_INTR(flags);
+		break;
+
+	case CTL_EXT_EFF_DEPTH: /* reverb effects: 0-127 */
+		DEBUG(2,printk("AWE32: [reverb(%d) %d]\n", voice, value));
+		FX_SET(voice, AWE_FX_REVERB, value * 2);
+		break;		
+
+	case CTL_CHORUS_DEPTH: /* chorus effects: 0-127 */
+		DEBUG(2,printk("AWE32: [chorus(%d) %d]\n", voice, value));
+		FX_SET(voice, AWE_FX_CHORUS, value * 2);
+		break;		
+
+
+	default:
+		DEBUG(0,printk("AWE32: [control(%d) ctrl=%d val=%d]\n",
+			   voice, ctrl_num, value));
+		break;
+	}
+}
+
+
+/* voice pan change (value = -128 - 127) */
+static void
+awe_panning(int dev, int voice, int value)
+{
+	DECL_INTR_FLAGS(flags);
+	if (voice >= 0 || voice < awe_max_voices) {
+		voices[voice].panning = value;
+		DEBUG(2,printk("AWE32: [pan(%d) %d]\n", voice, voices[voice].panning));
+		DISABLE_INTR(flags);
+		awe_set_pan(voice, 0);
+		RESTORE_INTR(flags);
+	}
+}
+
+
+/* volume mode change */
+static void
+awe_volume_method(int dev, int mode)
+{
+	/* not impremented */
+	DEBUG(0,printk("AWE32: [volmethod mode=%d]\n", mode));
+}
+
+
+/* patch manager */
+static int
+awe_patchmgr(int dev, struct patmgr_info *rec)
+{
+	FATALERR(printk("AWE32 Warning: patch manager control not supported\n"));
+	return 0;
+}
+
+
+/* pitch wheel change: 0-16384 */
+static void
+awe_bender(int dev, int voice, int value)
+{
+	DECL_INTR_FLAGS(flags);
+
+	if (voice < 0 || voice >= awe_max_voices)
+		return;
+	/* convert to zero centered value */
+	voices[voice].bender = value - 8192;
+	DEBUG(2,printk("AWE32: [bend(%d) %d]\n", voice, voices[voice].bender));
+	awe_calc_pitch(voice);
+	DISABLE_INTR(flags);
+	awe_set_pitch(voice);
+	RESTORE_INTR(flags);
+}
+
+
+/* search an empty voice; used by sequencer2 */
+static int
+awe_alloc(int dev, int chn, int note, struct voice_alloc_info *alloc)
+{
+	int i, p, best = -1, best_time = 0x7fffffff;
+
+	p = alloc->ptr;
+	/* First look for a completely stopped voice */
+
+	for (i = 0; i < alloc->max_voice; i++) {
+		if (alloc->map[p] == 0) {
+			alloc->ptr = p;
+			return p;
+		}
+		if (alloc->alloc_times[p] < best_time) {
+			best = p;
+			best_time = alloc->alloc_times[p];
+		}
+		p = (p + 1) % alloc->max_voice;
+	}
+
+	/* Then look for a releasing voice */
+	for (i = 0; i < alloc->max_voice; i++) {
+		if (alloc->map[p] == 0xffff) {
+			alloc->ptr = p;
+			return p;
+		}
+		p = (p + 1) % alloc->max_voice;
+	}
+
+	if (best >= 0)
+		p = best;
+
+	/* terminate the voice */
+	if (voices[p].state)
+		awe_terminate(p);
+
+	alloc->ptr = p;
+	return p;
+}
+
+
+/* set up voice; used by sequencer2 */
+static void
+awe_setup_voice(int dev, int voice, int chn)
+{
+	struct channel_info *info;
+	if (synth_devs[dev] == NULL ||
+	    (info = &synth_devs[dev]->chn_info[chn]) == NULL)
+		return;
+	if (voice < 0 || voice >= awe_max_voices)
+		return;
+
+	DEBUG(2,printk("AWE32: [setup(%d) ch=%d]\n", voice, chn));
+	voices[voice].expression_vol = info->controllers[CTL_EXPRESSION];
+	voices[voice].main_vol =
+		(info->controllers[CTL_MAIN_VOLUME] * 100) / 128; /* 0 - 127 */
+	voices[voice].panning =
+		info->controllers[CTL_PAN] * 2 - 128; /* signed 8bit */
+	voices[voice].bender = info->bender_value; /* zero center */
+	voices[voice].bank = info->controllers[CTL_BANK_SELECT];
+	awe_set_instr(dev, voice, info->pgm_num);
+}
+
+
+/*================================================================
+ * initialization of AWE32
+ *================================================================*/
+
+/* intiailize audio channels */
+static void
+awe_init_audio(void)
+{
+	int ch;
+
+	/* turn off envelope engines */
+	for (ch = 0; ch < AWE_MAX_VOICES; ch++) {
+		awe_poke(AWE_DCYSUSV(ch), 0x0080);
+	}
+  
+	for (ch = 0; ch < AWE_MAX_VOICES; ch++) {
+		awe_poke(AWE_ENVVOL(ch), 0);
+		awe_poke(AWE_ENVVAL(ch), 0);
+		awe_poke(AWE_DCYSUS(ch), 0);
+		awe_poke(AWE_ATKHLDV(ch), 0);
+		awe_poke(AWE_LFO1VAL(ch), 0);
+		awe_poke(AWE_ATKHLD(ch), 0);
+		awe_poke(AWE_LFO2VAL(ch), 0);
+		awe_poke(AWE_IP(ch), 0);
+		awe_poke(AWE_IFATN(ch), 0);
+		awe_poke(AWE_PEFE(ch), 0);
+		awe_poke(AWE_FMMOD(ch), 0);
+		awe_poke(AWE_TREMFRQ(ch), 0);
+		awe_poke(AWE_FM2FRQ2(ch), 0);
+		awe_poke_dw(AWE_PTRX(ch), 0);
+		awe_poke_dw(AWE_VTFT(ch), 0);
+		awe_poke_dw(AWE_PSST(ch), 0);
+		awe_poke_dw(AWE_CSL(ch), 0);
+		awe_poke_dw(AWE_CCCA(ch), 0);
+	}
+
+	for (ch = 0; ch < AWE_MAX_VOICES; ch++) {
+		awe_poke_dw(AWE_CPF(ch), 0);
+		awe_poke_dw(AWE_CVCF(ch), 0);
+	}
+}
+
+
+/* initialize DMA address */
+static void
+awe_init_dma(void)
+{
+	awe_poke_dw(AWE_SMALR, 0x00000000);
+	awe_poke_dw(AWE_SMARR, 0x00000000);
+	awe_poke_dw(AWE_SMALW, 0x00000000);
+	awe_poke_dw(AWE_SMARW, 0x00000000);
+}
+
+
+/* initialization arrays */
+
+static unsigned short init1[128] = {
+	0x03ff, 0x0030,  0x07ff, 0x0130, 0x0bff, 0x0230,  0x0fff, 0x0330,
+	0x13ff, 0x0430,  0x17ff, 0x0530, 0x1bff, 0x0630,  0x1fff, 0x0730,
+	0x23ff, 0x0830,  0x27ff, 0x0930, 0x2bff, 0x0a30,  0x2fff, 0x0b30,
+	0x33ff, 0x0c30,  0x37ff, 0x0d30, 0x3bff, 0x0e30,  0x3fff, 0x0f30,
+
+	0x43ff, 0x0030,  0x47ff, 0x0130, 0x4bff, 0x0230,  0x4fff, 0x0330,
+	0x53ff, 0x0430,  0x57ff, 0x0530, 0x5bff, 0x0630,  0x5fff, 0x0730,
+	0x63ff, 0x0830,  0x67ff, 0x0930, 0x6bff, 0x0a30,  0x6fff, 0x0b30,
+	0x73ff, 0x0c30,  0x77ff, 0x0d30, 0x7bff, 0x0e30,  0x7fff, 0x0f30,
+
+	0x83ff, 0x0030,  0x87ff, 0x0130, 0x8bff, 0x0230,  0x8fff, 0x0330,
+	0x93ff, 0x0430,  0x97ff, 0x0530, 0x9bff, 0x0630,  0x9fff, 0x0730,
+	0xa3ff, 0x0830,  0xa7ff, 0x0930, 0xabff, 0x0a30,  0xafff, 0x0b30,
+	0xb3ff, 0x0c30,  0xb7ff, 0x0d30, 0xbbff, 0x0e30,  0xbfff, 0x0f30,
+
+	0xc3ff, 0x0030,  0xc7ff, 0x0130, 0xcbff, 0x0230,  0xcfff, 0x0330,
+	0xd3ff, 0x0430,  0xd7ff, 0x0530, 0xdbff, 0x0630,  0xdfff, 0x0730,
+	0xe3ff, 0x0830,  0xe7ff, 0x0930, 0xebff, 0x0a30,  0xefff, 0x0b30,
+	0xf3ff, 0x0c30,  0xf7ff, 0x0d30, 0xfbff, 0x0e30,  0xffff, 0x0f30,
+};
+
+static unsigned short init2[128] = {
+	0x03ff, 0x8030, 0x07ff, 0x8130, 0x0bff, 0x8230, 0x0fff, 0x8330,
+	0x13ff, 0x8430, 0x17ff, 0x8530, 0x1bff, 0x8630, 0x1fff, 0x8730,
+	0x23ff, 0x8830, 0x27ff, 0x8930, 0x2bff, 0x8a30, 0x2fff, 0x8b30,
+	0x33ff, 0x8c30, 0x37ff, 0x8d30, 0x3bff, 0x8e30, 0x3fff, 0x8f30,
+
+	0x43ff, 0x8030, 0x47ff, 0x8130, 0x4bff, 0x8230, 0x4fff, 0x8330,
+	0x53ff, 0x8430, 0x57ff, 0x8530, 0x5bff, 0x8630, 0x5fff, 0x8730,
+	0x63ff, 0x8830, 0x67ff, 0x8930, 0x6bff, 0x8a30, 0x6fff, 0x8b30,
+	0x73ff, 0x8c30, 0x77ff, 0x8d30, 0x7bff, 0x8e30, 0x7fff, 0x8f30,
+
+	0x83ff, 0x8030, 0x87ff, 0x8130, 0x8bff, 0x8230, 0x8fff, 0x8330,
+	0x93ff, 0x8430, 0x97ff, 0x8530, 0x9bff, 0x8630, 0x9fff, 0x8730,
+	0xa3ff, 0x8830, 0xa7ff, 0x8930, 0xabff, 0x8a30, 0xafff, 0x8b30,
+	0xb3ff, 0x8c30, 0xb7ff, 0x8d30, 0xbbff, 0x8e30, 0xbfff, 0x8f30,
+	0xc3ff, 0x8030, 0xc7ff, 0x8130, 0xcbff, 0x8230, 0xcfff, 0x8330,
+	0xd3ff, 0x8430, 0xd7ff, 0x8530, 0xdbff, 0x8630, 0xdfff, 0x8730,
+	0xe3ff, 0x8830, 0xe7ff, 0x8930, 0xebff, 0x8a30, 0xefff, 0x8b30,
+	0xf3ff, 0x8c30, 0xf7ff, 0x8d30, 0xfbff, 0x8e30, 0xffff, 0x8f30,
+};
+
+static unsigned short init3[128] = {
+	0x0C10, 0x8470, 0x14FE, 0xB488, 0x167F, 0xA470, 0x18E7, 0x84B5,
+	0x1B6E, 0x842A, 0x1F1D, 0x852A, 0x0DA3, 0x8F7C, 0x167E, 0xF254,
+	0x0000, 0x842A, 0x0001, 0x852A, 0x18E6, 0x8BAA, 0x1B6D, 0xF234,
+	0x229F, 0x8429, 0x2746, 0x8529, 0x1F1C, 0x86E7, 0x229E, 0xF224,
+
+	0x0DA4, 0x8429, 0x2C29, 0x8529, 0x2745, 0x87F6, 0x2C28, 0xF254,
+	0x383B, 0x8428, 0x320F, 0x8528, 0x320E, 0x8F02, 0x1341, 0xF264,
+	0x3EB6, 0x8428, 0x3EB9, 0x8528, 0x383A, 0x8FA9, 0x3EB5, 0xF294,
+	0x3EB7, 0x8474, 0x3EBA, 0x8575, 0x3EB8, 0xC4C3, 0x3EBB, 0xC5C3,
+
+	0x0000, 0xA404, 0x0001, 0xA504, 0x141F, 0x8671, 0x14FD, 0x8287,
+	0x3EBC, 0xE610, 0x3EC8, 0x8C7B, 0x031A, 0x87E6, 0x3EC8, 0x86F7,
+	0x3EC0, 0x821E, 0x3EBE, 0xD208, 0x3EBD, 0x821F, 0x3ECA, 0x8386,
+	0x3EC1, 0x8C03, 0x3EC9, 0x831E, 0x3ECA, 0x8C4C, 0x3EBF, 0x8C55,
+
+	0x3EC9, 0xC208, 0x3EC4, 0xBC84, 0x3EC8, 0x8EAD, 0x3EC8, 0xD308,
+	0x3EC2, 0x8F7E, 0x3ECB, 0x8219, 0x3ECB, 0xD26E, 0x3EC5, 0x831F,
+	0x3EC6, 0xC308, 0x3EC3, 0xB2FF, 0x3EC9, 0x8265, 0x3EC9, 0x8319,
+	0x1342, 0xD36E, 0x3EC7, 0xB3FF, 0x0000, 0x8365, 0x1420, 0x9570,
+};
+
+static unsigned short init4[128] = {
+	0x0C10, 0x8470, 0x14FE, 0xB488, 0x167F, 0xA470, 0x18E7, 0x84B5,
+	0x1B6E, 0x842A, 0x1F1D, 0x852A, 0x0DA3, 0x0F7C, 0x167E, 0x7254,
+	0x0000, 0x842A, 0x0001, 0x852A, 0x18E6, 0x0BAA, 0x1B6D, 0x7234,
+	0x229F, 0x8429, 0x2746, 0x8529, 0x1F1C, 0x06E7, 0x229E, 0x7224,
+
+	0x0DA4, 0x8429, 0x2C29, 0x8529, 0x2745, 0x07F6, 0x2C28, 0x7254,
+	0x383B, 0x8428, 0x320F, 0x8528, 0x320E, 0x0F02, 0x1341, 0x7264,
+	0x3EB6, 0x8428, 0x3EB9, 0x8528, 0x383A, 0x0FA9, 0x3EB5, 0x7294,
+	0x3EB7, 0x8474, 0x3EBA, 0x8575, 0x3EB8, 0x44C3, 0x3EBB, 0x45C3,
+
+	0x0000, 0xA404, 0x0001, 0xA504, 0x141F, 0x0671, 0x14FD, 0x0287,
+	0x3EBC, 0xE610, 0x3EC8, 0x0C7B, 0x031A, 0x07E6, 0x3EC8, 0x86F7,
+	0x3EC0, 0x821E, 0x3EBE, 0xD208, 0x3EBD, 0x021F, 0x3ECA, 0x0386,
+	0x3EC1, 0x0C03, 0x3EC9, 0x031E, 0x3ECA, 0x8C4C, 0x3EBF, 0x0C55,
+
+	0x3EC9, 0xC208, 0x3EC4, 0xBC84, 0x3EC8, 0x0EAD, 0x3EC8, 0xD308,
+	0x3EC2, 0x8F7E, 0x3ECB, 0x0219, 0x3ECB, 0xD26E, 0x3EC5, 0x031F,
+	0x3EC6, 0xC308, 0x3EC3, 0x32FF, 0x3EC9, 0x0265, 0x3EC9, 0x8319,
+	0x1342, 0xD36E, 0x3EC7, 0x33FF, 0x0000, 0x8365, 0x1420, 0x9570,
+};
+
+
+/* send initialization arrays to start up */
+static void
+awe_init_array(void)
+{
+	awe_send_array(init1);
+	awe_wait(1024);
+	awe_send_array(init2);
+	awe_send_array(init3);
+	awe_poke_dw(AWE_HWCF4, 0);
+	awe_poke_dw(AWE_HWCF5, 0x83);
+	awe_poke_dw(AWE_HWCF6, 0x8000);
+	awe_send_array(init4);
+}
+
+/* send an initialization array */
+static void
+awe_send_array(unsigned short *data)
+{
+	int i;
+	unsigned short *p;
+
+	p = data;
+	for (i = 0; i < AWE_MAX_VOICES; i++, p++)
+		awe_poke(AWE_INIT1(i), *p);
+	for (i = 0; i < AWE_MAX_VOICES; i++, p++)
+		awe_poke(AWE_INIT2(i), *p);
+	for (i = 0; i < AWE_MAX_VOICES; i++, p++)
+		awe_poke(AWE_INIT3(i), *p);
+	for (i = 0; i < AWE_MAX_VOICES; i++, p++)
+		awe_poke(AWE_INIT4(i), *p);
+}
+
+
+/*
+ * set up awe32 channels to some known state.
+ */
+
+static void
+awe_tweak(void)
+{
+	int i;
+
+	/* Set the envelope engine parameters to the "default" values for
+	   simply playing back unarticulated audio at 44.1kHz.  Set all
+	   of the channels: */
+
+	for (i = 0; i < AWE_MAX_VOICES; i++) {
+		awe_poke(AWE_ENVVOL(i)   , 0x8000);
+		awe_poke(AWE_ENVVAL(i)   , 0x8000);
+		awe_poke(AWE_DCYSUS(i)   , 0x7F7F);
+		awe_poke(AWE_ATKHLDV(i)  , 0x7F7F);
+		awe_poke(AWE_LFO1VAL(i)  , 0x8000);
+		awe_poke(AWE_ATKHLD(i)   , 0x7F7F);
+		awe_poke(AWE_LFO2VAL(i)  , 0x8000);
+		awe_poke(AWE_IP(i)       , 0xE000);
+		awe_poke(AWE_IFATN(i)    , 0xFF00);
+		awe_poke(AWE_PEFE(i)     , 0x0000);
+		awe_poke(AWE_FMMOD(i)    , 0x0000);
+		awe_poke(AWE_TREMFRQ(i)  , 0x0010);
+		awe_poke(AWE_FM2FRQ2(i)  , 0x0010);
+	}
+}
+
+
+/*
+ *  initializes the FM section of AWE32
+ */
+
+static void
+awe_init_fm(void)
+{
+#ifndef AWE_ALWAYS_INIT_FM
+	/* if no extended memory is on board.. */
+	if (awe_mem_size <= 0)
+		return;
+#endif
+	DEBUG(0,printk("AWE32: initializing FM\n"));
+
+	/* Initialize the last two channels for DRAM refresh and producing
+	   the reverb and chorus effects for Yamaha OPL-3 synthesizer */
+
+	awe_poke(   AWE_DCYSUSV(30)   , 0x0080);
+	awe_poke_dw(AWE_PSST(30)      , 0xFFFFFFE0);
+	awe_poke_dw(AWE_CSL(30)       , 0xFFFFFFE8);
+	awe_poke_dw(AWE_PTRX(30)      , 0x00FFFF00);
+	awe_poke_dw(AWE_CPF(30)       , 0x00000000);
+	awe_poke_dw(AWE_CCCA(30)      , 0x00FFFFE3);
+
+	awe_poke(   AWE_DCYSUSV(31)   , 0x0080);
+	awe_poke_dw(AWE_PSST(31)      , 0x00FFFFE0);
+	awe_poke_dw(AWE_CSL(31)       , 0xFFFFFFE8);
+	awe_poke_dw(AWE_PTRX(31)      , 0x00FFFF00);
+	awe_poke_dw(AWE_CPF(31)       , 0x00000000);
+	awe_poke_dw(AWE_CCCA(31)      , 0x00FFFFE3);
+
+	/* Timing loop */
+
+	/* PTRX is 32 bit long but do not write to the MS byte */
+	awe_poke(AWE_PTRX(30)      , 0x0000);
+
+	while(! (inw(awe_base-0x620+Pointer) & 0x1000));
+	while(   inw(awe_base-0x620+Pointer) & 0x1000);
+
+	/* now write the MS byte of PTRX */
+	OUTW(0x4828, awe_base-0x620+Data0+0x002);
+
+	awe_poke(   AWE_IFATN(28)     , 0x0000);
+	awe_poke_dw(AWE_VTFT(30)      , 0x8000FFFF);
+	awe_poke_dw(AWE_VTFT(31)      , 0x8000FFFF);
+
+	/* change maximum channels to 30 */
+	awe_max_voices = AWE_NORMAL_VOICES;
+	awe_info.nr_voices = awe_max_voices;
+	voice_alloc->max_voice = awe_max_voices;
+}
+
+/*
+ *  AWE32 DRAM access routines
+ */
+
+/* open DRAM write accessing mode */
+static int
+awe_open_dram_for_write(int offset)
+{
+	int i;
+
+	/* use all channels for DMA transfer */
+	for (i = 0; i < AWE_NORMAL_VOICES; i++) {
+		awe_poke(AWE_DCYSUSV(i), 0x80);
+		awe_poke_dw(AWE_VTFT(i), 0);
+		awe_poke_dw(AWE_CVCF(i), 0);
+		awe_poke_dw(AWE_PTRX(i), 0x40000000);
+		awe_poke_dw(AWE_CPF(i), 0x40000000);
+		awe_poke_dw(AWE_PSST(i), 0);
+		awe_poke_dw(AWE_CSL(i), 0);
+		awe_poke_dw(AWE_CCCA(i), 0x06000000);
+	}
+	/* point channels 31 & 32 to ROM samples for DRAM refresh */
+	awe_poke_dw(AWE_VTFT(30), 0);
+	awe_poke_dw(AWE_PSST(30), 0x1d8);
+	awe_poke_dw(AWE_CSL(30), 0x1e0);
+	awe_poke_dw(AWE_CCCA(30), 0x1d8);
+	awe_poke_dw(AWE_VTFT(31), 0);
+	awe_poke_dw(AWE_PSST(31), 0x1d8);
+	awe_poke_dw(AWE_CSL(31), 0x1e0);
+	awe_poke_dw(AWE_CCCA(31), 0x1d8);
+
+	/* if full bit is on, not ready to write on */
+	if (awe_peek_dw(AWE_SMALW) & 0x80000000) {
+		for (i = 0; i < AWE_NORMAL_VOICES; i++)
+			awe_poke_dw(AWE_CCCA(i), 0);
+		return RET_ERROR(ENOSPC);
+	}
+
+	/* set address to write */
+	awe_poke_dw(AWE_SMALW, offset + AWE_DRAM_OFFSET);
+
+	return 0;
+}
+
+/* open DRAM for RAM size detection */
+static void
+awe_open_dram_for_check(void)
+{
+	int k;
+	unsigned long scratch;
+
+	awe_poke(AWE_HWCF2 , 0x0020);
+
+	for (k = 0; k < AWE_NORMAL_VOICES; k++) {
+		awe_poke(AWE_DCYSUSV(k), 0x0080);
+		awe_poke_dw(AWE_VTFT(k), 0x00000000);
+		awe_poke_dw(AWE_CVCF(k), 0x00000000);
+		awe_poke_dw(AWE_PTRX(k), 0x40000000);
+		awe_poke_dw(AWE_CPF(k), 0x40000000);
+		awe_poke_dw(AWE_PSST(k), 0x00000000);
+		awe_poke_dw(AWE_CSL(k), 0x00000000);
+		scratch = (((k&1) << 9) + 0x400);
+		scratch = scratch << 16;
+		awe_poke_dw(AWE_CCCA(k), scratch);
+	}
+}
+
+
+/* close dram access */
+static void
+awe_close_dram(void)
+{
+	int i;
+	/* wait until FULL bit in SMAxW register be false */
+	for (i = 0; i < 10000; i++) {
+		if (!(awe_peek_dw(AWE_SMALW) & 0x80000000))
+			break;
+		awe_wait(10);
+	}
+
+	for (i = 0; i < AWE_NORMAL_VOICES; i++) {
+		awe_poke_dw(AWE_CCCA(i), 0);
+		awe_poke(AWE_DCYSUSV(i), 0x807F);
+	}
+}
+
+
+#ifdef AWE_CHECKSUM_MEMORY
+/* open DRAM read accessing mode */
+static int
+awe_open_dram_for_read(int offset)
+{
+	int i;
+
+	/* use all channels for DMA transfer */
+	for (i = 0; i < AWE_NORMAL_VOICES; i++) {
+		awe_poke(AWE_DCYSUSV(i), 0x80);
+		awe_poke_dw(AWE_VTFT(i), 0);
+		awe_poke_dw(AWE_CVCF(i), 0);
+		awe_poke_dw(AWE_PTRX(i), 0x40000000);
+		awe_poke_dw(AWE_CPF(i), 0x40000000);
+		awe_poke_dw(AWE_PSST(i), 0);
+		awe_poke_dw(AWE_CSL(i), 0);
+		awe_poke_dw(AWE_CCCA(i), 0x04000000);
+	}
+	/* point channels 31 & 32 to ROM samples for DRAM refresh */
+	awe_poke_dw(AWE_VTFT(30), 0);
+	awe_poke_dw(AWE_PSST(30), 0x1d8);
+	awe_poke_dw(AWE_CSL(30), 0x1e0);
+	awe_poke_dw(AWE_CCCA(30), 0x1d8);
+	awe_poke_dw(AWE_VTFT(31), 0);
+	awe_poke_dw(AWE_PSST(31), 0x1d8);
+	awe_poke_dw(AWE_CSL(31), 0x1e0);
+	awe_poke_dw(AWE_CCCA(31), 0x1d8);
+
+	/* if empty flag is on, not ready to read */
+	if (awe_peek_dw(AWE_SMALR) & 0x80000000) {
+		for (i = 0; i < AWE_NORMAL_VOICES; i++)
+			awe_poke_dw(AWE_CCCA(i), 0);
+		return RET_ERROR(ENOSPC);
+	}
+
+	/* set address to read */
+	awe_poke_dw(AWE_SMALR, offset + AWE_DRAM_OFFSET);
+	/* drop stale data */
+	awe_peek(AWE_SMLD);
+	return 0;
+}
+
+/* close dram access for read */
+static void
+awe_close_dram_for_read(void)
+{
+	int i;
+	/* wait until FULL bit in SMAxW register be false */
+	for (i = 0; i < 10000; i++) {
+		if (!(awe_peek_dw(AWE_SMALR) & 0x80000000))
+			break;
+		awe_wait(10);
+	}
+	for (i = 0; i < AWE_NORMAL_VOICES; i++) {
+		awe_poke_dw(AWE_CCCA(i), 0);
+		awe_poke(AWE_DCYSUSV(i), 0x807F);
+	}
+}
+#endif /* AWE_CHECKSUM_MEMORY */
+
+
+/* write a word data */
+static void
+awe_write_dram(unsigned short c)
+{
+	int k;
+	/* wait until FULL bit in SMAxW register be false */
+	for (k = 0; k < 10000; k++) {
+		if (!(awe_peek_dw(AWE_SMALW) & 0x80000000))
+			break;
+		awe_wait(10);
+	}
+	awe_poke(AWE_SMLD, c);
+}
+
+/*================================================================
+ * detect presence of AWE32 and check memory size
+ *================================================================*/
+
+static int
+awe_detect(void)
+{
+#ifdef AWE_DEFAULT_BASE_ADDR
+	awe_base = AWE_DEFAULT_BASE_ADDR;
+	if (((awe_peek(AWE_U1) & 0x000F) == 0x000C) &&
+	    ((awe_peek(AWE_HWCF1) & 0x007E) == 0x0058) &&
+	    ((awe_peek(AWE_HWCF2) & 0x0003) == 0x0003))
+		return 1;
+#endif
+	if (awe_base == 0) {
+		for (awe_base = 0x620; awe_base <= 0x680; awe_base += 0x20) {
+			if ((awe_peek(AWE_U1) & 0x000F) != 0x000C)
+				continue;
+			if ((awe_peek(AWE_HWCF1) & 0x007E) != 0x0058)
+				continue;
+			if ((awe_peek(AWE_HWCF2) & 0x0003) != 0x0003)
+			        continue;
+		        DEBUG(0,printk("AWE32 found at %x\n", awe_base));
+			return 1;
+		}
+	}
+	FATALERR(printk("AWE32 not found\n"));
+	awe_base = 0;
+	return 0;
+}
+
+
+/*================================================================
+ * check dram size on AWE board
+ *================================================================*/
+static int
+awe_check_dram(void)
+{
+	awe_open_dram_for_check();
+
+	awe_poke_dw(AWE_SMALW    , 0x00200000);     /* DRAM start address */
+	awe_poke(   AWE_SMLD     , 0x1234);
+	awe_poke(   AWE_SMLD     , 0x7777);
+
+	awe_mem_size = 0;
+	while (awe_mem_size < 28*1024) {     /* 28 MB is max onboard memory */
+		awe_wait(2);
+		awe_poke_dw(AWE_SMALR, 0x00200000); /* Address for reading */
+		awe_peek(AWE_SMLD);		/* Discard stale data  */
+		if (awe_peek(AWE_SMLD) != 0x1234)
+			break;
+		if (awe_peek(AWE_SMLD) != 0x7777)
+			break;
+		awe_mem_size += 32;
+		/* Address for writing */
+		awe_poke_dw(AWE_SMALW, 0x00200000+awe_mem_size*512L);
+		awe_poke(AWE_SMLD, 0xFFFF);
+	}
+	awe_close_dram();
+
+	DEBUG(0,printk("AWE32: %d Kbytes memory detected\n", (int)awe_mem_size));
+#ifdef AWE_DEFAULT_MEM_SIZE
+	if (awe_mem_size == 0)
+		awe_mem_size = AWE_DEFAULT_MEM_SIZE;
+#endif
+	/* convert to Kbytes */
+	awe_mem_size *= 1024;
+	return awe_mem_size;
+}
+
+
+/*================================================================
+ * chorus and reverb controls
+ *================================================================*/
+
+static unsigned short ChorusEffects[24] =
+{
+  0xE600,0x03F6,0xBC2C,0xE608,0x031A,0xBC6E,0xE610,0x031A,
+  0xBC84,0xE620,0x0269,0xBC6E,0xE680,0x04D3,0xBCA6,0xE6E0,
+  0x044E,0xBC37,0xE600,0x0B06,0xBC00,0xE6C0,0x0B06,0xBC00
+};
+
+static unsigned long ChorusEffects2[] =
+{
+  0x0000 ,0x006D,0x8000,0x0000,0x0000 ,0x017C,0x8000,0x0000,
+  0x0000 ,0x0083,0x8000,0x0000,0x0000 ,0x017C,0x8000,0x0000,
+  0x0000 ,0x005B,0x8000,0x0000,0x0000 ,0x0026,0x8000,0x0000,
+  0x6E000,0x0083,0x8000,0x0000,0x6E000,0x0083,0x8000,0x0000
+};
+
+static unsigned short ChorusCommand[14] =
+{
+  0x69,0xA20,0x6C,0xA20,0x63,0xA22,0x29,
+  0xA20,0x2A,0xA20,0x2D,0xA20,0x2E,0xA20
+};
+
+static unsigned short ReverbEffects[224] =
+{
+  /* Room 1 */
+  0xB488,0xA450,0x9550,0x84B5,0x383A,0x3EB5,0x72F4,
+  0x72A4,0x7254,0x7204,0x7204,0x7204,0x4416,0x4516,
+  0xA490,0xA590,0x842A,0x852A,0x842A,0x852A,0x8429,
+  0x8529,0x8429,0x8529,0x8428,0x8528,0x8428,0x8528,
+  /* Room 2 */
+  0xB488,0xA458,0x9558,0x84B5,0x383A,0x3EB5,0x7284,
+  0x7254,0x7224,0x7224,0x7254,0x7284,0x4448,0x4548,
+  0xA440,0xA540,0x842A,0x852A,0x842A,0x852A,0x8429,
+  0x8529,0x8429,0x8529,0x8428,0x8528,0x8428,0x8528,
+  /* Room 3 */
+  0xB488,0xA460,0x9560,0x84B5,0x383A,0x3EB5,0x7284,
+  0x7254,0x7224,0x7224,0x7254,0x7284,0x4416,0x4516,
+  0xA490,0xA590,0x842C,0x852C,0x842C,0x852C,0x842B,
+  0x852B,0x842B,0x852B,0x842A,0x852A,0x842A,0x852A,
+  /* Hall 1 */
+  0xB488,0xA470,0x9570,0x84B5,0x383A,0x3EB5,0x7284,
+  0x7254,0x7224,0x7224,0x7254,0x7284,0x4448,0x4548,
+  0xA440,0xA540,0x842B,0x852B,0x842B,0x852B,0x842A,
+  0x852A,0x842A,0x852A,0x8429,0x8529,0x8429,0x8529,
+  /* Hall 2 */
+  0xB488,0xA470,0x9570,0x84B5,0x383A,0x3EB5,0x7254,
+  0x7234,0x7224,0x7254,0x7264,0x7294,0x44C3,0x45C3,
+  0xA404,0xA504,0x842A,0x852A,0x842A,0x852A,0x8429,
+  0x8529,0x8429,0x8529,0x8428,0x8528,0x8428,0x8528,
+  /* Plate */
+  0xB4FF,0xA470,0x9570,0x84B5,0x383A,0x3EB5,0x7234,
+  0x7234,0x7234,0x7234,0x7234,0x7234,0x4448,0x4548,
+  0xA440,0xA540,0x842A,0x852A,0x842A,0x852A,0x8429,
+  0x8529,0x8429,0x8529,0x8428,0x8528,0x8428,0x8528,
+  /* Delay */
+  0xB4FF,0xA470,0x9500,0x84B5,0x333A,0x39B5,0x7204,
+  0x7204,0x7204,0x7204,0x7204,0x72F4,0x4400,0x4500,
+  0xA4FF,0xA5FF,0x8420,0x8520,0x8420,0x8520,0x8420,
+  0x8520,0x8420,0x8520,0x8420,0x8520,0x8420,0x8520,
+  /* Panning Delay */
+  0xB4FF,0xA490,0x9590,0x8474,0x333A,0x39B5,0x7204,
+  0x7204,0x7204,0x7204,0x7204,0x72F4,0x4400,0x4500,
+  0xA4FF,0xA5FF,0x8420,0x8520,0x8420,0x8520,0x8420,
+  0x8520,0x8420,0x8520,0x8420,0x8520,0x8420,0x8520
+};
+
+static unsigned short ReverbCommand[56] =
+{
+  0x43,0xA20,0x45,0xA20,0x7F,0xA22,0x47,0xA20,
+  0x54,0xA22,0x56,0xA22,0x4F,0xA20,0x57,0xA20,
+  0x5F,0xA20,0x47,0xA22,0x4F,0xA22,0x57,0xA22,
+  0x5D,0xA22,0x5F,0xA22,0x61,0xA20,0x63,0xA20,
+  0x49,0xA20,0x4B,0xA20,0x51,0xA20,0x53,0xA20,
+  0x59,0xA20,0x5B,0xA20,0x41,0xA22,0x43,0xA22,
+  0x49,0xA22,0x4B,0xA22,0x51,0xA22,0x53,0xA22
+};
+
+static void awe_set_chorus_mode(int effect)
+{
+	int k;
+	DECL_INTR_FLAGS(flags);
+
+	DISABLE_INTR(flags);
+	for (k = 0; k < 3; k++)
+		awe_poke(ChorusCommand[k*2],
+			 ChorusCommand[k*2+1],
+			 ChorusEffects[k+effect*3]);
+	for (k = 0; k < 4; k++)
+		awe_poke_dw(ChorusCommand[6+k*2],
+			    ChorusCommand[6+k*2+1],
+			    ChorusEffects2[k+effect*4]);
+	RESTORE_INTR(flags);
+}
+
+static void awe_set_reverb_mode(int effect)
+{
+	int k;
+	DECL_INTR_FLAGS(flags);
+
+	DISABLE_INTR(flags);
+	for (k = 0; k < 28; k++)
+		awe_poke(ReverbCommand[k*2],
+			 ReverbCommand[k*2+1],
+			 ReverbEffects[k+effect*28]);
+	RESTORE_INTR(flags);
+}
+
+#endif /* CONFIG_AWE32_SYNTH */
diff --git a/sys/i386/isa/sound/coproc.h b/sys/i386/isa/sound/coproc.h
new file mode 100644
index 000000000000..f9023821d3f2
--- /dev/null
+++ b/sys/i386/isa/sound/coproc.h
@@ -0,0 +1,12 @@
+/*
+ * Definitions for various on board processors on the soundcards. For
+ * example DSP processors.
+ */
+
+/*
+ * Coprocessor access types 
+ */
+#define COPR_CUSTOM		0x0001	/* Custom applications */
+#define COPR_MIDI		0x0002	/* MIDI (MPU-401) emulation */
+#define COPR_PCM		0x0004	/* Digitized voice applications */
+#define COPR_SYNTH		0x0008	/* Music synthesis */
diff --git a/sys/i386/isa/sound/cs4232.c b/sys/i386/isa/sound/cs4232.c
new file mode 100644
index 000000000000..4b2c38d2e51c
--- /dev/null
+++ b/sys/i386/isa/sound/cs4232.c
@@ -0,0 +1,206 @@
+/*
+ * sound/cs4232.c
+ * 
+ * The low level driver for Crystal CS4232 based cards. The CS4232 is a PnP
+ * compatible chip which contains a CS4231A codec, SB emulation, a MPU401
+ * compatible MIDI port, joystick and synthesizer and IDE CD-ROM interfaces.
+ * This is just a temporary driver until full PnP support gets inplemented.
+ * Just the WSS codec, FM synth and the MIDI ports are supported. Other
+ * interfaces are left uninitialized.
+ * 
+ * Copyright by Hannu Savolainen 1995
+ * 
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are
+ * met: 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer. 2.
+ * Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED.  IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE FOR
+ * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+ * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
+ * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ * 
+ */
+
+#include <i386/isa/sound/sound_config.h>
+
+#if defined(CONFIG_CS4232)
+
+#define KEY_PORT	0x279	/* Same as LPT1 status port */
+#define CSN_NUM		0x99	/* Just a random number */
+
+#define CS_OUT(a) 		outb( KEY_PORT,  a)
+#define CS_OUT2(a, b)		{CS_OUT(a);CS_OUT(b);}
+#define CS_OUT3(a, b, c)	{CS_OUT(a);CS_OUT(b);CS_OUT(c);}
+
+static int      mpu_base = 0, mpu_irq = 0;
+static int      mpu_detected = 0;
+
+int
+probe_cs4232_mpu(struct address_info * hw_config)
+{
+    /*
+     * Just write down the config values.
+     */
+
+    mpu_base = hw_config->io_base;
+    mpu_irq = hw_config->irq;
+    return 0;
+}
+
+void
+attach_cs4232_mpu(struct address_info * hw_config)
+{
+}
+
+static unsigned char crystal_key[] =	/* A 32 byte magic key sequence */
+{
+    0x96, 0x35, 0x9a, 0xcd, 0xe6, 0xf3, 0x79, 0xbc,
+    0x5e, 0xaf, 0x57, 0x2b, 0x15, 0x8a, 0xc5, 0xe2,
+    0xf1, 0xf8, 0x7c, 0x3e, 0x9f, 0x4f, 0x27, 0x13,
+    0x09, 0x84, 0x42, 0xa1, 0xd0, 0x68, 0x34, 0x1a
+};
+
+int
+probe_cs4232(struct address_info * hw_config)
+{
+    int             i;
+    int             base = hw_config->io_base, irq = hw_config->irq;
+    int             dma1 = hw_config->dma, dma2 = hw_config->dma2;
+
+    /*
+     * Verify that the I/O port range is free.
+     */
+
+    if (0) {
+	printf("cs4232.c: I/O port 0x%03x not free\n", base);
+	return 0;
+    }
+    /*
+     * This version of the driver doesn't use the PnP method when
+     * configuring the card but a simplified method defined by Crystal.
+     * This means that just one CS4232 compatible device can exist on the
+     * system. Also this method conflicts with possible PnP support in
+     * the OS. For this reason driver is just a temporary kludge.
+     */
+
+    /*
+     * Wake up the card by sending a 32 byte Crystal key to the key port.
+     */
+    for (i = 0; i < 32; i++)
+	CS_OUT(crystal_key[i]);
+
+    /*
+     * Now set the CSN (Card Select Number).
+     */
+
+    CS_OUT2(0x06, CSN_NUM);
+
+    /*
+     * Ensure that there is no other codec using the same address.
+     */
+
+    CS_OUT2(0x15, 0x00);	/* Select logical device 0 (WSS/SB/FM) */
+    CS_OUT2(0x33, 0x00);	/* Inactivate logical dev 0 */
+    if (ad1848_detect(hw_config->io_base, NULL, hw_config->osp))
+	return 0;
+
+    /*
+     * Then set some config bytes. First logical device 0
+     */
+
+    CS_OUT2(0x15, 0x00);	/* Select logical device 0 (WSS/SB/FM) */
+    CS_OUT3(0x47, (base >> 8) & 0xff, base & 0xff);	/* WSSbase */
+
+    if (0)			/* Not free */
+	CS_OUT3(0x48, 0x00, 0x00)	/* FMbase off */
+    else
+	CS_OUT3(0x48, 0x03, 0x88);	/* FMbase 0x388 */
+
+    CS_OUT3(0x42, 0x00, 0x00);	/* SBbase off */
+    CS_OUT2(0x22, irq);	/* SB+WSS IRQ */
+    CS_OUT2(0x2a, dma1);	/* SB+WSS DMA */
+
+    if (dma2 != -1)
+	CS_OUT2(0x25, dma2)	/* WSS DMA2 */
+    else
+	CS_OUT2(0x25, 4);	/* No WSS DMA2 */
+
+    CS_OUT2(0x33, 0x01);	/* Activate logical dev 0 */
+
+    /*
+     * Initialize logical device 3 (MPU)
+     */
+
+#if (defined(CONFIG_MPU401) || defined(CONFIG_MPU_EMU)) && defined(CONFIG_MIDI)
+    if (mpu_base != 0 && mpu_irq != 0) {
+	CS_OUT2(0x15, 0x03);	/* Select logical device 3 (MPU) */
+	CS_OUT3(0x47, (mpu_base >> 8) & 0xff, mpu_base & 0xff);	/* MPUbase */
+	CS_OUT2(0x22, mpu_irq);	/* MPU IRQ */
+	CS_OUT2(0x33, 0x01);	/* Activate logical dev 3 */
+    }
+#endif
+
+    /*
+     * Finally activate the chip
+     */
+    CS_OUT(0x79);
+
+    /*
+     * Then try to detect the codec part of the chip
+     */
+
+    return ad1848_detect(hw_config->io_base, NULL, hw_config->osp);
+}
+
+void
+attach_cs4232(struct address_info * hw_config)
+{
+    int             base = hw_config->io_base, irq = hw_config->irq;
+    int             dma1 = hw_config->dma, dma2 = hw_config->dma2;
+
+    if (dma2 == -1)
+	dma2 = dma1;
+
+    ad1848_init("CS4232", base,
+	irq,
+	dma1,	/* Playback DMA */
+	dma2,	/* Capture DMA */
+	0,
+	hw_config->osp);
+
+#if (defined(CONFIG_MPU401) || defined(CONFIG_MPU_EMU)) && defined(CONFIG_MIDI)
+    if (mpu_base != 0 && mpu_irq != 0) {
+	static struct address_info hw_config2 =
+	{0};		/* Ensure it's initialized */
+
+	hw_config2.io_base = mpu_base;
+	hw_config2.irq = mpu_irq;
+	hw_config2.dma = -1;
+	hw_config2.dma2 = -1;
+	hw_config2.always_detect = 0;
+	hw_config2.name = NULL;
+	hw_config2.card_subtype = 0;
+	hw_config2.osp = hw_config->osp;
+
+	if (probe_mpu401(&hw_config2)) {
+	    mpu_detected = 1;
+	    attach_mpu401(&hw_config2);
+	} else {
+	    mpu_base = mpu_irq = 0;
+	}
+    }
+#endif
+}
+
+#endif
diff --git a/sys/i386/isa/sound/hex2hex.h b/sys/i386/isa/sound/hex2hex.h
new file mode 100644
index 000000000000..ecd7b4c4239c
--- /dev/null
+++ b/sys/i386/isa/sound/hex2hex.h
@@ -0,0 +1,97 @@
+/*
+ * This file is a part of configure.c
+ *
+ * hex2hex reads an input file in Intel HEX format and produces
+ * an (unsigned char) array which contains the bytes and writes it to the
+ * output file using C syntax
+ */
+
+#define MAX_SIZE (256*1024)
+#define ABANDON(why) { \
+		fprintf(stderr, "%s: " why "\n", source); \
+		fclose(inf);fclose(outf);return 0; \
+		}
+
+int hex2hex(char *source, char *target, char *varline)
+{
+	FILE *inf, *outf;
+
+	int i,l, c;
+	unsigned char buf[MAX_SIZE];
+
+	if ((inf=fopen(source, "r"))==NULL)
+	{
+		perror(source);
+		return 0;
+	}
+
+	if ((outf=fopen(target, "w"))==NULL)
+	{
+		perror(target);
+		fclose(inf);
+		return 0;
+	}
+
+	l=0;
+
+	while ((c=getc(inf))!=EOF)
+	{
+		if (c == ':')	/* Sync with beginning of line */
+		{
+			int n, check;
+			unsigned char sum;
+			int addr;
+			int linetype;
+
+			if (fscanf(inf, "%02x", &n) != 1)
+			   ABANDON("File format error");
+			sum = n;
+
+			if (fscanf(inf, "%04x", &addr) != 1)
+			   ABANDON("File format error");
+			sum += addr/256;
+			sum += addr%256;
+
+			if (fscanf(inf, "%02x", &linetype) != 1)
+			   ABANDON("File format error");
+			sum += linetype;
+
+			if (linetype != 0)
+			   continue;
+
+			for (i=0;i<n;i++)
+			{
+				if (fscanf(inf, "%02x", &c) != 1)
+			   	   ABANDON("File format error");
+				if (addr >= MAX_SIZE)
+				   ABANDON("File too large");
+				buf[addr++] = c;
+				if (addr > l)
+				   l = addr;
+				sum += c;
+			}
+
+			if (fscanf(inf, "%02x", &check) != 1)
+			   ABANDON("File format error");
+
+			sum = ~sum + 1;
+			if (check != sum)
+			   ABANDON("Line checksum error");
+		}
+	}
+
+	fprintf(outf, "/*\n *\t Computer generated file. Do not edit.\n */\n");
+	fprintf(outf, "%s[] = {\n", varline);
+
+	for (i=0;i<l;i++)
+	{
+		if (i) fprintf(outf, ",");
+		if (i && !(i % 16)) fprintf(outf, "\n");
+		fprintf(outf, "0x%02x", buf[i]);
+	}
+
+	fprintf(outf, "\n};\n\n");
+	fclose(inf);
+	fclose(outf);
+	return 1;
+}
diff --git a/sys/i386/isa/sound/iwdefs.h b/sys/i386/isa/sound/iwdefs.h
new file mode 100644
index 000000000000..60f50fc50028
--- /dev/null
+++ b/sys/i386/isa/sound/iwdefs.h
@@ -0,0 +1,712 @@
+/*#######################################################################*/
+/*  FILE   : iwdefs.h*/
+/**/
+/*  REMARKS: This file contains all defines used by DDK functions. It*/
+/*           should be included by all applications and should be*/
+/*           referenced by programmers to make their code easy to read*/
+/*           and understand.*/
+/**/
+/*  UPDATE: 3/21/95*/
+/*          7/10/95 --- added #def DRAM_HOLES*/
+/*#######################################################################*/
+#ifndef IWDEFS
+#define IWDEFS
+/*#######################################################################*/
+/**/
+/* Macros for use in loading Synth Addr Regs*/
+/**/
+/*#######################################################################*/
+#define ADDR_HIGH(x)    (short)(x>>7)
+#define ADDR_LOW(x)     (short)(x<<9)
+/*#######################################################################*/
+/**/
+/*     Defines for DMA Controllers*/
+/**/
+/*#######################################################################*/
+/* DMA Controler #1 (8-bit controller) */
+#define DMA1_STAT         0x08          /* read status register */
+#define DMA1_WCMD         0x08          /* write command register */
+#define DMA1_WREQ         0x09          /* write request register */
+#define DMA1_SNGL         0x0A          /* write single bit register */
+#define DMA1_MODE         0x0B          /* write mode register */
+#define DMA1_CLRFF        0x0C          /* clear byte ptr flip/flop */
+#define DMA1_MCLR         0x0D          /* master clear register */
+#define DMA1_CLRM         0x0E          /* clear mask register */
+#define DMA1_WRTALL       0x0F          /* write all mask register */
+
+/* DMA Controler #2 (16-bit controller) */
+#define DMA2_STAT         0xD0          /* read status register */
+#define DMA2_WCMD         0xD0          /* write command register */
+#define DMA2_WREQ         0xD2          /* write request register */
+#define DMA2_SNGL         0xD4          /* write single bit register */
+#define DMA2_MODE         0xD6          /* write mode register */
+#define DMA2_CLRFF        0xD8          /* clear byte ptr flip/flop */
+#define DMA2_MCLR         0xDA          /* master clear register */
+#define DMA2_CLRM         0xDC          /* clear mask register */
+#define DMA2_WRTALL       0xDE          /* write all mask register */
+
+#define DMA0_ADDR         0x00          /* chan 0 base adddress */
+#define DMA0_CNT          0x01          /* chan 0 base count */
+#define DMA1_ADDR         0x02          /* chan 1 base adddress */
+#define DMA1_CNT          0x03          /* chan 1 base count */
+#define DMA2_ADDR         0x04          /* chan 2 base adddress */
+#define DMA2_CNT          0x05          /* chan 2 base count */
+#define DMA3_ADDR         0x06          /* chan 3 base adddress */
+#define DMA3_CNT          0x07          /* chan 3 base count */
+#define DMA4_ADDR         0xC0          /* chan 4 base adddress */
+#define DMA4_CNT          0xC2          /* chan 4 base count */
+#define DMA5_ADDR         0xC4          /* chan 5 base adddress */
+#define DMA5_CNT          0xC6          /* chan 5 base count */
+#define DMA6_ADDR         0xC8          /* chan 6 base adddress */
+#define DMA6_CNT          0xCA          /* chan 6 base count */
+#define DMA7_ADDR         0xCC          /* chan 7 base adddress */
+#define DMA7_CNT          0xCE          /* chan 7 base count */
+
+#define DMA0_PAGE         0x87          /* chan 0 page register (refresh)*/
+#define DMA1_PAGE         0x83          /* chan 1 page register */
+#define DMA2_PAGE         0x81          /* chan 2 page register */
+#define DMA3_PAGE         0x82          /* chan 3 page register */
+#define DMA4_PAGE         0x8F          /* chan 4 page register (unusable)*/
+#define DMA5_PAGE         0x8B          /* chan 5 page register */
+#define DMA6_PAGE         0x89          /* chan 6 page register */
+#define DMA7_PAGE         0x8A          /* chan 7 page register */
+/*#######################################################################*/
+/**/
+/* Defines for register UISR (Interrupt Status)*/
+/**/
+/*#######################################################################*/
+#define MIDI_TX_IRQ       0x01
+#define MIDI_RX_IRQ       0x02
+#define ALIB_TIMER1_IRQ   0x04
+#define ALIB_TIMER2_IRQ   0x08
+#define _UASBCI           0x45          /* UASBCI index */
+#define SAMPLE_CONTROL    0x49          /* Not used by IW */
+#define SET_VOICES        0x0E
+#define WAVETABLE_IRQ     0x20
+#define ENVELOPE_IRQ      0x40
+#define DMA_TC_IRQ        0x80
+/*#######################################################################*/
+/**/
+/*            Synthesizer-related defines*/
+/**/
+/*#######################################################################*/
+#define GEN_INDEX         0x03           /* IGIDX offset into p3xr */
+#define VOICE_SELECT      0x02           /* SVSR offset into p3xr */
+#define VOICE_IRQS        0x8F           /* SVII index (read) */
+#define _URSTI            0x4C           /* URSTI index */
+#define GF1_SET           0x01           /* URSTI[0] */
+#define GF1_OUT_ENABLE    0x02           /* URSTI[1] */
+#define GF1_IRQ_ENABLE    0x04           /* URSTI[2] */
+#define GF1_RESET         0xFE           /* URSTI[0]=0 */
+#define VOICE_VOLUME_IRQ  0x04           /* SVII[2] */
+#define VOICE_WAVE_IRQ    0x08           /* SVII[3] */
+#define VC_IRQ_ENABLE     0x20           /* SACI[5] or SVCI[5]*/
+#define VOICE_NUMBER      0x1F           /* Mask for SVII[4:0] */
+#define VC_IRQ_PENDING    0x80           /* SACI[7] or SVCI[7] */
+#define VC_DIRECT         0x40           /* SACI[6] or SVCI[6]*/
+#define VC_DATA_WIDTH     0x04           /* SACI[2] */
+#define VOICE_STOP        0x02           /* SACI[1] */
+#define VOICE_STOPPED     0x01           /* SACI[0] */
+#define VOLUME_STOP       0x02           /* SVCI[1] */
+#define VOLUME_STOPPED    0x01           /* SVCI[0] */
+#define VC_ROLLOVER       0x04           /* SVCI[2] */
+#define VC_LOOP_ENABLE    0x08           /* SVCI[3] or SACI[3]*/
+#define VC_BI_LOOP        0x10           /* SVCI[4] or SACI[4]*/
+#define VOICE_OFFSET      0x20           /* SMSI[5] */
+#define VOLUME_RATE0      0x00           /* SVRI[7:6]=(0,0) */
+#define VOLUME_RATE1      0x40           /* SVRI[7:6]=(0,1) */
+#define VOLUME_RATE2      0x80           /* SVRI[7:6]=(1,0) */
+#define VOLUME_RATE3      0xC0           /* SVRI[7:6]=(1,1) */
+/*#######################################################################*/
+/**/
+/*         Power-Mode Control Defines*/
+/**/
+/*#######################################################################*/
+#define SHUT_DOWN         0x7E           /* shuts InterWave down */
+#define POWER_UP          0xFE           /* enables all modules */
+#define CODEC_PWR_UP      0x81           /* enables Codec Analog Ckts */
+#define CODEC_PWR_DOWN    0x01           /* disables Codec Analog Ckts */
+#define CODEC_REC_UP      0x82           /* Enables Record Path */
+#define CODEC_REC_DOWN    0x02           /* Disables Record Path */
+#define CODEC_PLAY_UP     0x84           /* Enables Playback Path */
+#define CODEC_PLAY_DOWN   0x04           /* Disables Playback Path */
+#define CODEC_IRQ_ENABLE  0x02           /* CEXTI[2] */
+#define CODEC_TIMER_IRQ   0x40           /* CSR3I[6] */
+#define CODEC_REC_IRQ     0x20           /* CSR3I[5] */
+#define CODEC_PLAY_IRQ    0x10           /* CSR3I[4] */
+#define CODEC_INT         0x01           /* CSR1R[0] */
+#define MONO_INPUT        0x80           /* CMONOI[7] */
+#define MONO_OUTPUT       0x40           /* CMONOI[6] */
+#define MIDI_UP           0x88           /* Enables MIDI ports */
+#define MIDI_DOWN         0x08           /* Disables MIDI ports */
+#define SYNTH_UP          0x90           /* Enables Synthesizer */
+#define SYNTH_DOWN        0x10           /* Disables Synthesizer */
+#define LMC_UP            0xA0           /* Enables LM Module */
+#define LMC_DOWN          0x20           /* Disbales LM Module */
+#define XTAL24_UP         0xC0           /* Enables 24MHz Osc */
+#define XTAL24_DOWN       0x40           /* Disables 24MHz Osc */
+#define _PPWRI            0xF2           /* PPWRI index */
+#define PLAY              0x0F
+#define REC               0x1F
+#define LEFT_AUX1_INPUT   0x02
+#define RIGHT_AUX1_INPUT  0x03
+#define LEFT_AUX2_INPUT   0x04
+#define RIGHT_AUX2_INPUT  0x05
+#define LEFT_LINE_IN      0x12
+#define RIGHT_LINE_IN     0x13
+#define LEFT_LINE_OUT     0x19
+#define RIGHT_LINE_OUT    0x1B
+#define LEFT_SOURCE       0x00
+#define RIGHT_SOURCE      0x01
+#define LINE_IN           0x00
+#define AUX1_IN           0x40
+#define MIC_IN            0x80
+#define MIX_IN            0xC0
+#define LEFT_DAC          0x06
+#define RIGHT_DAC         0x07
+#define LEFT_MIC_IN       0x16
+#define RIGHT_MIC_IN      0x17
+#define _CUPCTI           0x0E
+#define _CLPCTI           0x0F
+#define _CURCTI           0x1E
+#define _CLRCTI           0x1F
+#define _CLAX1I           0x02
+#define _CRAX1I           0x03
+#define _CLAX2I           0x04
+#define _CRAX2I           0x05
+#define _CLLICI           0x12
+#define _CRLICI           0x13
+#define _CLOAI            0x19
+#define _CROAI            0x1B
+#define _CLICI            0x00
+#define _CRICI            0x01
+#define _CLDACI           0x06
+#define _CRDACI           0x07
+#define _CPVFI            0x1D
+/*#######################################################################*/
+/**/
+/* Defines for DMA transfer related operations*/
+/**/
+/*#######################################################################*/
+#define MAX_DMA         0x07
+#define DMA_DECREMENT   0x20
+#define AUTO_INIT       0x10
+#define DMA_READ        0x01
+#define DMA_WRITE       0x02
+#define AUTO_READ       0x03
+#define AUTO_WRITE      0x04
+#define IDMA_INV        0x0400
+#define IDMA_WIDTH_16   0x0100
+/*#######################################################################*/
+/**/
+/* Bits for dma flags within a DMA structure.*/
+/**/
+/*#######################################################################*/
+#define DMA_USED        0x0001
+
+#define DMA_SPLIT       0x0004    /* DMA Controller Page Crossover*/
+#define CODEC_DMA       0x0008    /* Indicates a Codec DMA*/
+#define DMA_WAIT        0x0020    /* Wait for DMA xfer to complete*/
+#define DMA_DOWN        0x0040    /* DMA xfer from PC to InterWave*/
+#define DRAM_HOLES      0x8000    /* Indicates Non-contiguous RAM configuration*/
+#define DMA_UP          0xFFBF    /* DMA xfer from InterWave to PC */
+/*#######################################################################*/
+/**/
+/* Bits for DMA Control Register (LDMACI)*/
+/**/
+/*#######################################################################*/
+#define _LDMACI         0x41  /* Index */
+#define DMA_INV         0x80
+#define DMA_IRQ_ENABLE  0x20
+#define DMA_IRQ_PENDING 0x40  /* on reads of LDMACI[6] */
+#define DMA_DATA_16     0x40  /* on writes to LDMACI[6] */
+#define DMA_WIDTH_16    0x04  /* 1=16-bit, 0=8-bit (DMA channel) */
+#define DMA_RATE        0x18  /* 00=fastest,...,11=slowest */
+#define DMA_UPLOAD      0x02  /* From LM to PC */
+#define DMA_ENABLE      0x01
+/*#######################################################################*/
+/**/
+/*    DMA Transfer Rates*/
+/**/
+/*#######################################################################*/
+#define DMA_R0          0xE7    /* Fastest (use ANDing to set) */
+#define DMA_R1          0x08
+#define DMA_R2          0x10
+#define DMA_R3          0x18    /* Slowest */
+/*#######################################################################*/
+/**/
+/*    Interrupt Controller Defines*/
+/**/
+/*#######################################################################*/
+#define IW_HANDLERS_ON  0x80    /* Flag for when IVT is modified */
+#define EOI             0x20
+#define OCR1            0x20    /* 8259-1 Operation Control Reg. */
+#define IMR1            0x21    /* 8259-1 Interrupt Mask Reg. */
+#define OCR2            0xA0    /* 8259-2 Operation Control Reg. */
+#define IMR2            0xA1    /* 8259-2 Interrupt Mask Reg. */
+#define IRQ0_UNMASK     0xFE    /* Mask to clear bit 0 in IMR */
+#define IRQ1_UNMASK     0xFD
+#define IRQ2_UNMASK     0xFB
+#define IRQ3_UNMASK     0xF7
+#define IRQ4_UNMASK     0xEF
+#define IRQ5_UNMASK     0xDF
+#define IRQ6_UNMASK     0xBF
+#define IRQ7_UNMASK     0x7F
+#define IRQ8_UNMASK     0xFE    /* Mask to clear bit 0 in IMR */
+#define IRQ9_UNMASK     0xFD
+#define IRQ10_UNMASK    0xFB
+#define IRQ11_UNMASK    0xF7
+#define IRQ12_UNMASK    0xEF
+#define IRQ13_UNMASK    0xDF
+#define IRQ14_UNMASK    0xBF
+#define IRQ15_UNMASK    0x7F
+#define IRQ0_EOI        0x60    /* Spec EOI for IRQ0 */
+#define IRQ1_EOI        0x61
+#define IRQ2_EOI        0x62
+#define IRQ3_EOI        0x63
+#define IRQ4_EOI        0x64
+#define IRQ5_EOI        0x65
+#define IRQ6_EOI        0x66
+#define IRQ7_EOI        0x67
+#define IRQ8_EOI        0x60    /* Spec EOI for IRQ8 */
+#define IRQ9_EOI        0x61
+#define IRQ10_EOI       0x62
+#define IRQ11_EOI       0x63
+#define IRQ12_EOI       0x64
+#define IRQ13_EOI       0x65
+#define IRQ14_EOI       0x66
+#define IRQ15_EOI       0x67
+/*#######################################################################*/
+/**/
+/*     Generic defines*/
+/**/
+/*#######################################################################*/
+/**/
+#define MEMBANK0      0L                 /* Addr of Memory Bank 0*/
+#define MEMBANK1      4194304L           /* Addr of Memory Bank 1*/
+#define MEMBANK2      8388608L           /* Addr of Memory Bank 2*/
+#define MEMBANK3     12582912L           /* Addr of Memory Bank 3*/
+#define IRQ_UNAVAIL       0x0000
+#define IRQ_AVAIL         0x0001
+#define IRQ_USED          0x0002
+#define MAX_IRQ           16
+#define NEXT_OFFSET       0L
+#define PREV_OFFSET       4L
+#define SIZE_OFFSET       8L
+#define MEM_HEADER_SIZE  12L
+#define GF1_POOL         (usigned long)(256L*1024L)
+#define GUS_MODE          0x00           /* SGMI[0]=0*/
+#define ENH_MODE          0x01           /* SGMI[0]=1*/
+#define ENABLE_LFOS       0x02           /* SGMI[1]*/
+#define NO_WAVETABLE      0x04           /* SGMI[2]*/
+#define RAM_TEST          0x08           /* SGMI[3]*/
+#define TRUE              1
+#define FALSE             0
+#define ON                1
+#define OFF               0
+#define AUDIO             0
+#define EXT               1
+#define GAME              2
+#define EMULATION         3
+#define MPU401            4
+#define AUDIO_EXT         2
+#define ALLOC_FAILURE     0xFFFFFFFFL
+#define MEM_EXHAUSTED     0xFFFFFFFFL
+#define RAM_MAX           16777216L
+#define RAM_STEP          65536L
+#define BANK_MAX          4194304L
+#define ILLEGAL_SIZE      -1
+#define MEM_INIT           1
+#define NO_NEXT           0xFFFFFFFFL
+#define NO_PREV           NO_NEXT
+#define DMA_BAD_ADDR     -1
+#define DMA_ON           -1
+#define DMA_OK            1
+#define MIDI_TX_IRQ       0x01
+#define MIDI_RX_IRQ       0x02
+#define ALIB_TIMER1_IRQ   0x04
+#define ALIB_TIMER2_IRQ   0x08
+#define WAVETABLE_IRQ     0x20
+#define ENVELOPE_IRQ      0x40
+#define DMA_TC_IRQ        0x80
+#define DMA_SET_MASK      0x04
+#define PNP_DATA_RDY       1          /* PRESSI[0] */
+#define IWAVE_ABSENT      2
+#define IWAVE_OPEN        4
+#define IWAVE_OK          5
+#define BAD_VOICES        -1
+#define PNP_ABSENT        0xFF        /* No PNP cards in system */
+#define DPMI_INT          0x31
+#define _PCCCI            0x02
+#define _PCSNI            0x06
+#define _PIDXR            0x279
+#define _PNPWRP           0xA79
+#define _LDSALI           0x42
+#define _LDSAHI           0x50
+#define _LMALI            0x43
+#define _LMAHI            0x44
+#define _LMCFI            0x52
+#define _LMCI             0x53
+#define _LDIBI            0x58
+#define _LDICI            0x57
+#define _LMSBAI           0x51
+#define _SVCI_RD          0x8D
+#define _SVCI_WR          0x0D
+#define _SACI_RD          0x80
+#define _SACI_WR          0x00
+#define _SALI_RD          0x8B
+#define _SALI_WR          0x0B
+#define _SAHI_RD          0x8A
+#define _SAHI_WR          0x0A
+#define _SASHI_RD         0x82
+#define _SASHI_WR         0x02
+#define _SASLI_RD         0x83
+#define _SASLI_WR         0x03
+#define _SAEHI_RD         0x84
+#define _SAEHI_WR         0x04
+#define _SAELI_RD         0x85
+#define _SAELI_WR         0x05
+#define _SVRI_RD          0x86
+#define _SVRI_WR          0x06
+#define _SVSI_RD          0x87
+#define _SVSI_WR          0x07
+#define _SVEI_RD          0x88
+#define _SVEI_WR          0x08
+#define _SVLI_RD          0x89
+#define _SVLI_WR          0x09
+#define _SROI_RD          0x8C
+#define _SROI_WR          0x0C
+#define _SLOI_RD          0x93
+#define _SLOI_WR          0x13
+#define _SMSI_RD          0x95
+#define _SMSI_WR          0x15
+#define _SGMI_RD          0x99
+#define _SGMI_WR          0x19
+#define _SFCI_RD          0x81
+#define _SFCI_WR          0x01
+#define _SUAI_RD          0x90
+#define _SUAI_WR          0x10
+#define _SVII             0x8F          
+#define _CMODEI           0x0C        /* index for CMODEI */
+#define _CFIG3I           0x11
+#define _CFIG2I           0x10
+#define _CLTIMI           0x14
+#define _CUTIMI           0x15
+#define _CSR3I            0x18        /* Index to CSR3I (Interrupt Status) */
+#define _CEXTI            0x0A        /* Index to External Control Register */
+#define _CFIG1I           0x09        /* Index to Codec Conf Reg 1 */
+#define _CSR2I            0x0B        /* Index to Codec Stat Reg 2 */
+#define _CPDFI            0x08        /* Index to Play Data Format Reg */
+#define _CRDFI            0x1C        /* Index to Rec Data Format Reg */
+#define _CLMICI           0x16        /* Index to Left Mic Input Ctrl Register */
+#define _CRMICI           0x17        /* Index to Right Mic Input Ctrl Register */
+#define _CLCI             0x0D        /* Index to Loopback Ctrl Register */
+#define _IVERI            0x5B        /* Index to register IVERI */
+#define CODEC_MODE1       0x00
+#define CODEC_MODE2       0x40
+#define CODEC_MODE3       0x6C        /* Enhanced Mode */
+#define CODEC_STATUS1     0x01
+#define CODEC_STATUS2     0x0B        /* Index to CSR2I */
+#define CODEC_STATUS3     0x18        /* Index to CSR3I */
+#define PLAYBACK          0x01        /* Enable playback path CFIG1I[0]=1*/
+#define RECORD            0x02        /* Enable Record path CFIG1I[1]=1*/
+#define TIMER_ENABLE      0x40        /* CFIG2I[6] */
+#define CODEC_MCE         0x40        /* CIDXR[6] */
+#define CALIB_IN_PROGRESS 0x20        /* CSR2I[5] */
+#define CODEC_INIT        0x80        /* CIDXR[7] */
+#define BIT16_BIG         0xC0        /* 16-bit signed, big endian */
+#define IMA_ADPCM         0xA0        /* IMA-compliant ADPCM */
+#define BIT8_ALAW         0x60        /* 8-bit A-law */
+#define BIT16_LITTLE      0x40        /* 16-bit signed, lillte endian */
+#define BIT8_ULAW         0x20        /* 8-bit u-law */
+#define BIT8_LINEAR       0x00        /* 8-bit unsigned */
+#define REC_DFORMAT       0x1C
+#define PLAY_DFORMAT      0x08
+#define DMA_ACCESS        0x00
+#define PIO_ACCESS        0xC0
+#define DMA_SIMPLEX       0x04
+#define STEREO            0x10        /* CxDFI[4] */
+#define XTAL1             0x00        /* CxDFI[4]=0 selects 24.5Mhz XTAL */
+#define XTAL2             0x01        /* CxDFI[4]=1 selects 16.9Mhz XTAL */
+#define AUTOCALIB         0x08        /* CFIG1I[3] */
+#define ROM_IO            0x02        /* ROM I/O cycles - LMCI[1]=1 */
+#define DRAM_IO           0x4D        /* DRAM I/O cycles - LMCI[1]=0 */
+#define AUTOI             0x01        /* LMCI[0]=1 */
+#define _PLDNI            0x07
+#define ACTIVATE_DEV      0x30
+#define _PWAKEI           0x03        /* Index for PWAKEI */
+#define _PISOCI           0x01        /* Index for PISOCI */
+#define _PSECI            0xF1        /* Index for PSECI */
+#define RANGE_IOCHK       0x31        /* PURCI or PRRCI Index */
+#define MIDI_RESET        0x03
+#define IO_OK             5           /* No IO conflict flag */
+#define IO_CONFLICT       6           /* IO Conflict detected */
+#define IO_0x55           0x01
+#define IO_0xAA           0xFE          
+/*#######################################################################*/
+/**/
+/* Defines for Sound Handlers in "iw".*/
+/**/
+/*#######################################################################*/
+#define PLAY_DMA_HANDLER     0x01
+#define REC_DMA_HANDLER      0x02
+#define MIDI_TX_HANDLER      0x03
+#define MIDI_RX_HANDLER      0x04
+#define TIMER1_HANDLER       0x05
+#define TIMER2_HANDLER       0x06
+#define WAVE_HANDLER         0x07
+#define VOLUME_HANDLER       0x08
+#define CODEC_TIMER_HANDLER  0x09
+#define CODEC_PLAY_HANDLER   0x0A
+#define CODEC_REC_HANDLER    0x0B
+#define AUX_HANDLER          0x0C
+/*#######################################################################*/
+/**/
+/*     Mapping for System Control Regs.*/
+/**/
+/*#######################################################################*/
+#define UMCR            0x00010000    /* Mix Control Reg.*/
+#define UISR            0x00020006    /* IRQ Stat Reg. (read) */
+#define U2X6R           0x00030006    /* SB 2X6 reg */
+#define UACWR           0x00040008    /* AdLib Command Write Reg */
+#define UASRR           0x00050008    /* AdLib Stat Read Reg */
+#define UADR            0x00060009    /* AdLib Data Register */
+#define UACRR           0x0007000A    /* AdLib Cmd Read Reg */
+#define UASWR           0x0008000A    /* AdLib Stat Write Reg */
+#define UHRDP           0x0009000B    /* Hidden Reg Data Port */
+#define UI2XCR          0x000A000C    /* SB IRQ 2xC Reg */
+#define U2XCR           0x000B000D    /* SB 2xC Reg. (No IRQ) */
+#define U2XER           0x000C000E    /* SB 2xE Reg. */
+#define URCR            0x000D000F    /* Reg Control Register */
+#define USRR            0x000E000F    /* Status Read Register */
+#define UDCI            0x000F000B    /* DMA Channel Control Reg */
+#define UICI            0x0010000B    /* Interrupt Ctrl Reg */
+#define UGP1I           0x0011010B    /* GP Reg 1 (Back Door) */
+#define UGP2I           0x0012020B    /* GP Reg 2 (Back Door) */
+#define UGPA1I          0x0013030B    /* GP reg 1 Address */
+#define UGPA2I          0x0014040B    /* GP reg 2 Address */
+#define UCLRII          0x0015050B    /* Clear Interrupt Reg */
+#define UJMPI           0x0016060B    /* Jumper Register */
+#define UGP1II          0x0017000B    /* Gen. Purp Reg 1(Emulation) */
+#define UGP2II          0x0018000B    /* Gen. Purp Reg 2(Emulation) */
+#define GGCR            0x00190201    /* Game Control Register */
+#define GMCR            0x001A0000    /* MIDI Control Register */
+#define GMSR            0x001B0000    /* MIDI Status Reg. */
+#define GMTDR           0x001C0001    /* MIDI xmit data reg */
+#define GMRDR           0x001D0001    /* MIDI rcv data reg */
+#define SVSR            0x001E0002    /* Synth Voice Select Reg */
+#define IGIDXR          0x001F0003    /* General Index Register */
+#define I16DP           0x00200004    /* General 16-bit Data Port */
+#define I8DP            0x00210005    /* General 8-bit Data Port */
+/*#######################################################################*/
+/**/
+/*             Synth defines*/
+/**/
+/*#######################################################################*/
+#define SACI            0x00220005    /* Synth Addr Control */
+#define SFCI            0x00230104    /* Synth Freq Control */
+#define SASHI           0x00240204    /* Synth Addr Start High */
+#define SASLI           0x00250304    /* Synth Addr Start Low */
+#define SAEHI           0x00260404    /* Synth Addr End High */
+#define SAELI           0x00270504    /* Synth Addr End Low */
+#define SVRI            0x00280605    /* Synth Volume Rate */
+#define SVSI            0x00290705    /* Synth Volume Start */
+#define SVEI            0x002A0805    /* Synth Volume End */
+#define SVLI            0x002B0904    /* Synth Volume Level */
+#define SAHI            0x002C0A04    /* Synth Address High */
+#define SALI            0x002D0B04    /* Synth Address Low */
+#define SROI            0x002E0C04    /* Synth Right Offset */
+#define SVCI            0x002F0D05    /* Synth Volume Control */
+#define SAVI            0x00300E05    /* Synth Active Voices */
+#define SVII            0x00318F05    /* Synth Voice IRQ */
+#define SUAI            0x00321005    /* Synth Upper Addr */
+#define SEAHI           0x00331104    /* Synth Effect Addr High */
+#define SEALI           0x00341204    /* Synth Effect Addr Low */
+#define SLOI            0x00351304    /* Synth Left Offset */
+#define SEASI           0x00361405    /* Synth Effects Accum Sel */
+#define SMSI            0x00371505    /* Synth Mode Select */
+#define SEVI            0x00381604    /* Synth Effect Volume */
+#define SFLFOI          0x00391705    /* Synth Freq LFO */
+#define SVLFOI          0x003A1805    /* Synth Vol LFO */
+#define SGMI            0x003B1905    /* Synth Global Mode */
+#define SLFOBI          0x003C1A04    /* Synth LFO Base Address */
+#define SROFI           0x003D1B04
+#define SLOFI           0x003E1C04
+#define SEVFI           0x003F1D04
+#define SVIRI           0x00409F05    /* Synth Voice Read IRQ */
+#define LDMACI          0x00414105    /* DMA Control Reg. */
+#define LDSALI          0x00424204    /* LMC DMA Start Addr. Low Reg. */
+#define LMALI           0x00434304    /* LMC Addr Low (I/O) */
+#define LMAHI           0x00444405    /* LMC Addr High (I/O) */
+#define UASBCI          0x00454505    /* Adlib-SB Control */
+#define UAT1I           0x00464605    /* AdLib Timer 1 Count */
+#define UAT2I           0x00474705    /* AdLib Timer 2 Count */
+#define USCI            0x00484905    /* Sample Control Reg */
+#define GJTDI           0x00494B05
+#define URSTI           0x004A4C05
+#define LDSAHI          0x004B5005
+#define LMSBAI          0x004C5104
+#define LMCFI           0x004D5204
+#define LMCI            0x004E5305
+#define LMRFAI          0x004F5404
+#define LMPFAI          0x00505504
+#define LMSFI           0x00515604
+#define LDICI           0x00525704
+#define LDIBI           0x00535804
+#define ICMPTI          0x00545905
+#define IDECI           0x00555A05
+#define IVERI           0x00565B05
+#define IEMUAI          0x00575C05
+#define IEMUBI          0x00585D05
+#define GMRFAI          0x00595E05
+#define ITCI            0x005A5F05
+#define IEIRQI          0x005B6005
+#define LMBDR           0x005C0007
+/*##########################################################*/
+/* Mnemonics for Codec Registers*/
+/*##########################################################*/
+#define CIDXR           0x005D0000
+#define CDATAP          0x005E0001
+#define CSR1R           0x005F0002
+#define CPDR            0x00600003
+#define CRDR            0x00610003
+#define CLICI           0x00620001
+#define CRICI           0x00630101
+#define CLAX1I          0x00640201
+#define CRAX1I          0x00650301
+#define CLAX2I          0x00660401
+#define CRAX2I          0x00670501
+#define CLDACI          0x00680601
+#define CRDACI          0x00690701
+#define CPDFI           0x006A0801
+#define CFIG1I          0x006B0901
+#define CEXTI           0x006C0A01
+#define CSR2I           0x006D0B01
+#define CMODEI          0x006E0C01
+#define CLCI            0x006F0D01
+#define CUPCTI          0x00700E01
+#define CLPCTI          0x00710F01
+#define CFIG2I          0x00721001
+#define CFIG3I          0x00731101
+#define CLLICI          0x00741201
+#define CRLICI          0x00751301
+#define CLTIMI          0x00761401
+#define CUTIMI          0x00771501
+#define CLMICI          0x00781601
+#define CRMICI          0x00791701
+#define CSR3I           0x007A1801
+#define CLOAI           0x007B1901
+#define CMONOI          0x007C1A01
+#define CROAI           0x007D1B01
+#define CRDFI           0x007E1C01
+#define CPVFI           0x007F1D01
+#define CURCTI          0x00801E01
+#define CLRCTI          0x00811F01
+/*##########################################################*/
+/* Mnemonics for PnP Registers*/
+/*##########################################################*/
+#define PCSNBR          0x00820201
+#define PIDXR           0x00830279
+#define PNPWRP          0x00840A79
+#define PNPRDP          0x00850000
+#define PSRPAI          0x00860000
+#define PISOCI          0x00870100
+#define PCCCI           0x00880200
+#define PWAKEI          0x00890300
+#define PRESDI          0x008A0400
+#define PRESSI          0x008B0500
+#define PCSNI           0x008C0600
+#define PLDNI           0x008D0700
+#define PUACTI          0x008E3000
+#define PURCI           0x008F3100
+#define P2X0HI          0x00906000
+#define P2X0LI          0x00916100
+#define P3X0HI          0x00926200
+#define P3X0LI          0x00936300
+#define PHCAI           0x00946400
+#define PLCAI           0x00956500
+#define PUI1SI          0x00967000
+#define PUI1TI          0x00977100
+#define PUI2SI          0x00987200
+#define PUI2TI          0x00997300
+#define PUD1SI          0x009A7400
+#define PUD2SI          0x009B7500
+#define PSEENI          0x009CF000
+#define PSECI           0x009DF100
+#define PPWRI           0x009EF200
+#define PRACTI          0x009F3001
+#define PRRCI           0x00A03101
+#define PRAHI           0x00A16001
+#define PRALI           0x00A26101
+#define PATAHI          0x00A36201
+#define PATALI          0x00A46301
+#define PRISI           0x00A57001
+#define PRITI           0x00A67101
+#define PRDSI           0x00A77401
+#define PGACTI          0x00A83002
+#define PGRCI           0x00A93102
+#define P201HI          0x00AA6002
+#define P201LI          0x00AB6102
+#define PSACTI          0x00AC3003
+#define PSRCI           0x00AD3103
+#define P388HI          0x00AE6003
+#define P388LI          0x00AF6103
+#define PSBISI          0x00B07003
+#define PSBITI          0x00B17103
+#define PMACTI          0x00B23004
+#define PMRCI           0x00B33104
+#define P401HI          0x00B46004
+#define P401LI          0x00B56104
+#define PMISI           0x00B67004
+#define PMITI           0x00B77104
+
+typedef unsigned char  BYTE;
+typedef unsigned short WORD;
+typedef unsigned short PORT;
+typedef unsigned long  DWORD;
+typedef unsigned long  ADDRESS;
+typedef int BOOL;
+typedef int FLAG;
+
+typedef struct
+ {
+	short flags;              /* InterWave stat flags */
+	PORT pcodar;                      /* Base Port for Codec */
+	PORT pcdrar;                      /* Base Port for Ext Device */
+	PORT p2xr;                        /* Compatibility Base Port */
+	PORT p3xr;                        /* MIDI and Synth Base Port */
+	PORT p401ar;                      /* Gen Purpose Reg. 1 address */
+	PORT p201ar;                      /* Game Ctrl normally at 0x201 */
+	PORT pataar;                      /* Base Address for ATAPI I/O Space */
+	PORT p388ar;                      /* Base Port for AdLib. It should be 388h */
+	PORT pnprdp;                      /* PNP read data port */
+	PORT igidxr;                      /* Gen Index Reg at P3XR+0x03 */
+	PORT i16dp;                       /* 16-bit data port at P3XR+0x04  */
+	PORT i8dp;                        /* 8-bit data port at P3XR+0x05 */
+	PORT svsr;                        /* Synth Voice Select at P3XR+0x02 */
+	PORT cdatap;                      /* Codec Indexed Data Port at PCODAR+0x01 */
+	PORT csr1r;                       /* Codec Stat Reg 1 at PCODAR+0x02 */
+	PORT cxdr;                        /* Play or Record Data Reg at PCODAR+0x03 */
+	PORT gmxr;                        /* GMCR or GMSR at P3XR+0x00 */
+	PORT gmxdr;                       /* GMTDR or GMRDR at P3XR+0x01 */
+	PORT lmbdr;                       /* LMBDR at P3XR+0x07 */
+	BYTE csn;                         /* Card Select Number */
+	BYTE cmode;                       /* Codec Operation Mode */
+	int dma1_chan;                    /* DMA channel 1 (local DMA & codec rec) */
+	int dma2_chan;                    /* DMA channel 2 (codec play) */
+	int ext_chan;                     /* Ext Dev DMA channel */
+	BYTE voices;                      /* Number of active voices */
+	DWORD vendor;                     /* Vendor ID and Product Identifier */
+	int synth_irq;                    /* Synth IRQ number */
+	int midi_irq;                     /* MIDI IRQ number */
+	int ext_irq;                      /* Ext Dev IRQ  */
+	int mpu_irq;                      /* MPU401 Dev IRQ */
+	int emul_irq;                     /* Sound Blaster/AdLib Dev IRQ */
+	ADDRESS  free_mem;                /* Address of First Free LM Block */
+	DWORD reserved_mem;               /* Amount of LM reserved by app. */
+	BYTE smode;                       /* Synth Mode */
+	WORD  size_mem;                   /* Total LM in Kbytes */
+
+ } IWAVE;
+
+#endif
diff --git a/sys/i386/isa/sound/mad16.c b/sys/i386/isa/sound/mad16.c
new file mode 100644
index 000000000000..0e2064a97b8a
--- /dev/null
+++ b/sys/i386/isa/sound/mad16.c
@@ -0,0 +1,524 @@
+/*
+ * sound/mad16.c
+ * 
+ * Initialization code for OPTi MAD16 compatible audio chips. Including
+ * 
+ * OPTi 82C928     MAD16           (replaced by C929) OAK OTI-601D    Mozart
+ * OPTi 82C929     MAD16 Pro
+ * 
+ * These audio interface chips don't prduce sound themselves. They just connect
+ * some other components (OPL-[234] and a WSS compatible codec) to the PC bus
+ * and perform I/O, DMA and IRQ address decoding. There is also a UART for
+ * the MPU-401 mode (not 82C928/Mozart). The Mozart chip appears to be
+ * compatible with the 82C928 (can anybody confirm this?).
+ * 
+ * NOTE! If you want to set CD-ROM address and/or joystick enable, define
+ * MAD16_CONF in local.h as combination of the following bits:
+ * 
+ * 0x01    - joystick disabled
+ * 
+ * CD-ROM type selection (select just one): 0x00    - none 0x02    - Sony 31A
+ * 0x04    - Mitsumi 0x06    - Panasonic (type "LaserMate", not
+ * "SoundBlaster") 0x08    - Secondary IDE (address 0x170) 0x0a    - Primary
+ * IDE (address 0x1F0)
+ * 
+ * For example Mitsumi with joystick disabled = 0x04|0x01 = 0x05 For example
+ * LaserMate (for use with sbpcd) plus joystick = 0x06
+ * 
+ * MAD16_CDSEL: This defaults to CD I/O 0x340, no IRQ and DMA3 (DMA5 with
+ * Mitsumi or IDE). If you like to change these, define MAD16_CDSEL with the
+ * following bits:
+ * 
+ * CD-ROM port: 0x00=340, 0x40=330, 0x80=360 or 0xc0=320 OPL4 select: 0x20=OPL4,
+ * 0x00=OPL3 CD-ROM irq: 0x00=disabled, 0x04=IRQ5, 0x08=IRQ7, 0x0a=IRQ3,
+ * 0x10=IRQ9, 0x14=IRQ10 and 0x18=IRQ11.
+ * 
+ * CD-ROM DMA (Sony or Panasonic): 0x00=DMA3, 0x01=DMA2, 0x02=DMA1 or
+ * 0x03=disabled or CD-ROM DMA (Mitsumi or IDE):    0x00=DMA5, 0x01=DMA6,
+ * 0x02=DMA7 or 0x03=disabled
+ * 
+ * For use with sbpcd, address 0x340, set MAD16_CDSEL to 0x03 or 0x23.
+ * 
+ * Copyright by Hannu Savolainen 1995
+ * 
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are
+ * met: 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer. 2.
+ * Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED.  IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE FOR
+ * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+ * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
+ * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ * 
+ */
+
+#include <i386/isa/sound/sound_config.h>
+
+#if defined(CONFIG_MAD16)
+
+static int      already_initialized = 0;
+
+#define C928	1
+#define MOZART	2
+#define C929	3
+
+/*
+ * Registers
+ * 
+ * The MAD16 occupies I/O ports 0xf8d to 0xf93 (fixed locations). All ports are
+ * inactive by default. They can be activated by writing 0xE2 or 0xE3 to the
+ * password register. The password is valid only until the next I/O read or
+ * write.
+ */
+
+#define MC1_PORT	0xf8d	/* SB address, CDROM interface type, joystick */
+#define MC2_PORT	0xf8e	/* CDROM address, IRQ, DMA, plus OPL4 bit */
+#define MC3_PORT	0xf8f
+#define PASSWD_REG	0xf8f
+#define MC4_PORT	0xf90
+#define MC5_PORT	0xf91
+#define MC6_PORT	0xf92
+#define MC7_PORT	0xf93
+
+static int      board_type = C928;
+
+static sound_os_info *mad16_osp;
+
+#ifndef DDB
+#define DDB(x)
+#endif
+
+static unsigned char
+mad_read(int port)
+{
+	unsigned long   flags;
+	unsigned char   tmp;
+
+	flags = splhigh();
+
+	switch (board_type) {	/* Output password */
+	case C928:
+	case MOZART:
+		outb(PASSWD_REG, 0xE2);
+		break;
+
+	case C929:
+		outb(PASSWD_REG, 0xE3);
+		break;
+	}
+
+	tmp = inb(port);
+	splx(flags);
+
+	return tmp;
+}
+
+static void
+mad_write(int port, int value)
+{
+	unsigned long   flags;
+
+	flags = splhigh();
+
+	switch (board_type) {	/* Output password */
+	case C928:
+	case MOZART:
+		outb(PASSWD_REG, 0xE2);
+		break;
+
+	case C929:
+		outb(PASSWD_REG, 0xE3);
+		break;
+	}
+
+	outb(port, (unsigned char) (value & 0xff));
+	splx(flags);
+}
+
+static int
+detect_mad16(void)
+{
+	unsigned char   tmp, tmp2;
+
+	/*
+	 * Check that reading a register doesn't return bus float (0xff) when
+	 * the card is accessed using password. This may fail in case the
+	 * card is in low power mode. Normally at least the power saving mode
+	 * bit should be 0.
+	 */
+	if ((tmp = mad_read(MC1_PORT)) == 0xff) {
+		DDB(printf("MC1_PORT returned 0xff\n"));
+		return 0;
+	}
+	/*
+	 * Now check that the gate is closed on first I/O after writing the
+	 * password. (This is how a MAD16 compatible card works).
+	 */
+
+	if ((tmp2 = inb(MC1_PORT)) == tmp) {	/* It didn't close */
+		DDB(printf("MC1_PORT didn't close after read (0x%02x)\n", tmp2));
+		return 0;
+	}
+	mad_write(MC1_PORT, tmp ^ 0x80);	/* Togge a bit */
+
+	if ((tmp2 = mad_read(MC1_PORT)) != (tmp ^ 0x80)) {	/* Compare the bit */
+		mad_write(MC1_PORT, tmp);	/* Restore */
+		DDB(printf("Bit revert test failed (0x%02x, 0x%02x)\n", tmp, tmp2));
+		return 0;
+	}
+	mad_write(MC1_PORT, tmp);	/* Restore */
+	return 1;		/* Bingo */
+
+}
+
+int
+probe_mad16(struct address_info * hw_config)
+{
+	int             i;
+	static int      valid_ports[] =
+	{0x530, 0xe80, 0xf40, 0x604};
+	unsigned char   tmp;
+	unsigned char   cs4231_mode = 0;
+
+	int             ad_flags = 0;
+
+	if (already_initialized)
+		return 0;
+
+	mad16_osp = hw_config->osp;
+	/*
+	 * Check that all ports return 0xff (bus float) when no password is
+	 * written to the password register.
+	 */
+
+	DDB(printf("--- Detecting MAD16 / Mozart ---\n"));
+
+
+	/*
+	 * Then try to detect with the old password
+	 */
+	board_type = C928;
+
+	DDB(printf("Detect using password = 0xE2\n"));
+
+	if (!detect_mad16()) {	/* No luck. Try different model */
+		board_type = C929;
+
+		DDB(printf("Detect using password = 0xE3\n"));
+
+		if (!detect_mad16())
+			return 0;
+
+		DDB(printf("mad16.c: 82C929 detected\n"));
+	} else {
+		unsigned char   model;
+
+		if (((model = mad_read(MC3_PORT)) & 0x03) == 0x03) {
+			DDB(printf("mad16.c: Mozart detected\n"));
+			board_type = MOZART;
+		} else {
+			DDB(printf("mad16.c: 82C928 detected???\n"));
+			board_type = C928;
+		}
+	}
+
+	for (i = 0xf8d; i <= 0xf93; i++)
+		DDB(printf("port %03x = %03x\n", i, mad_read(i)));
+
+	/*
+	 * Set the WSS address
+	 */
+
+	tmp = 0x80;		/* Enable WSS, Disable SB */
+
+	for (i = 0; i < 5; i++) {
+		if (i > 3) {	/* Not a valid port */
+			printf("MAD16/Mozart: Bad WSS base address 0x%x\n", hw_config->io_base);
+			return 0;
+		}
+		if (valid_ports[i] == hw_config->io_base) {
+			tmp |= i << 4;	/* WSS port select bits */
+			break;
+		}
+	}
+
+	/*
+	 * Set optional CD-ROM and joystick settings.
+	 */
+
+#ifdef MAD16_CONF
+	tmp |= ((MAD16_CONF) & 0x0f);	/* CD-ROM and joystick bits */
+#endif
+	mad_write(MC1_PORT, tmp);
+
+#if defined(MAD16_CONF) && defined(MAD16_CDSEL)
+	tmp = MAD16_CDSEL;
+#else
+	tmp = 0x03;
+#endif
+
+#ifdef MAD16_OPL4
+	tmp |= 0x20;		/* Enable OPL4 access */
+#endif
+
+	mad_write(MC2_PORT, tmp);
+	mad_write(MC3_PORT, 0xf0);	/* Disable SB */
+
+	if (!ad1848_detect(hw_config->io_base + 4, &ad_flags, mad16_osp))
+		return 0;
+
+	if (ad_flags & (AD_F_CS4231 | AD_F_CS4248))
+		cs4231_mode = 0x02;	/* CS4248/CS4231 sync delay switch */
+
+	if (board_type == C929) {
+		mad_write(MC4_PORT, 0xa2);
+		mad_write(MC5_PORT, 0xA5 | cs4231_mode);
+		mad_write(MC6_PORT, 0x03);	/* Disable MPU401 */
+	} else {
+		mad_write(MC4_PORT, 0x02);
+		mad_write(MC5_PORT, 0x30 | cs4231_mode);
+	}
+
+	for (i = 0xf8d; i <= 0xf93; i++)
+		DDB(printf("port %03x after init = %03x\n", i, mad_read(i)));
+
+	/*
+	 * Verify the WSS parameters
+	 */
+
+	if (0) {
+		printf("MSS: I/O port conflict\n");
+		return 0;
+	}
+	/*
+	 * Check if the IO port returns valid signature. The original MS
+	 * Sound system returns 0x04 while some cards (AudioTriX Pro for
+	 * example) return 0x00.
+	 */
+
+	if ((inb(hw_config->io_base + 3) & 0x3f) != 0x04 &&
+	    (inb(hw_config->io_base + 3) & 0x3f) != 0x00) {
+		DDB(printf("No MSS signature detected on port 0x%x (0x%x)\n",
+			   hw_config->io_base, inb(hw_config->io_base + 3)));
+		return 0;
+	}
+	if (hw_config->irq > 11) {
+		printf("MSS: Bad IRQ %d\n", hw_config->irq);
+		return 0;
+	}
+	if (hw_config->dma != 0 && hw_config->dma != 1 && hw_config->dma != 3) {
+		printf("MSS: Bad DMA %d\n", hw_config->dma);
+		return 0;
+	}
+	/*
+	 * Check that DMA0 is not in use with a 8 bit board.
+	 */
+
+	if (hw_config->dma == 0 && inb(hw_config->io_base + 3) & 0x80) {
+		printf("MSS: Can't use DMA0 with a 8 bit card/slot\n");
+		return 0;
+	}
+	if (hw_config->irq > 7 && hw_config->irq != 9 && inb(hw_config->io_base + 3) & 0x80) {
+		printf("MSS: Can't use IRQ%d with a 8 bit card/slot\n", hw_config->irq);
+		return 0;
+	}
+	return 1;
+}
+
+void
+attach_mad16(struct address_info * hw_config)
+{
+
+	static char     interrupt_bits[12] =
+	{
+		-1, -1, -1, -1, -1, -1, -1, 0x08, -1, 0x10, 0x18, 0x20
+	};
+	char            bits;
+
+	static char     dma_bits[4] =
+	{
+		1, 2, 0, 3
+	};
+
+	int             config_port = hw_config->io_base + 0, version_port = hw_config->io_base + 3;
+	int             ad_flags = 0, dma = hw_config->dma, dma2 = hw_config->dma2;
+	unsigned char   dma2_bit = 0;
+
+	already_initialized = 1;
+
+	if (!ad1848_detect(hw_config->io_base + 4, &ad_flags, mad16_osp))
+		return;
+
+	/*
+	 * Set the IRQ and DMA addresses.
+	 */
+
+	bits = interrupt_bits[hw_config->irq];
+	if (bits == -1)
+		return;
+
+	outb(config_port, bits | 0x40);
+	if ((inb(version_port) & 0x40) == 0)
+		printf("[IRQ Conflict?]");
+
+	/*
+	 * Handle the capture DMA channel
+	 */
+
+	if (ad_flags & AD_F_CS4231 && dma2 != -1 && dma2 != dma) {
+		if ((dma == 0 && dma2 == 1) ||
+		    (dma == 1 && dma2 == 0) ||
+		    (dma == 3 && dma2 == 0)) {
+			dma2_bit = 0x04;	/* Enable capture DMA */
+		} else {
+			printf("MAD16: Invalid capture DMA\n");
+			dma2 = dma;
+		}
+	} else
+		dma2 = dma;
+
+	outb(config_port, bits | dma_bits[dma] | dma2_bit);	/* Write IRQ+DMA setup */
+
+	ad1848_init("MAD16 WSS", hw_config->io_base + 4,
+		    hw_config->irq,
+		    dma,
+		    dma2, 0,
+		    hw_config->osp);
+}
+
+void
+attach_mad16_mpu(struct address_info * hw_config)
+{
+	if (board_type < C929) {/* Early chip. No MPU support. Just SB MIDI */
+#ifdef CONFIG_MIDI
+
+		if (mad_read(MC1_PORT) & 0x20)
+			hw_config->io_base = 0x240;
+		else
+			hw_config->io_base = 0x220;
+
+		return mad16_sb_dsp_init(hw_config);
+#else
+		return 0;
+#endif
+	}
+#if (defined(CONFIG_MPU401) || defined(CONFIG_MPU_EMU)) && defined(CONFIG_MIDI)
+	if (!already_initialized)
+		return;
+
+	attach_mpu401(hw_config);
+#endif
+}
+
+int
+probe_mad16_mpu(struct address_info * hw_config)
+{
+#if (defined(CONFIG_MPU401) || defined(CONFIG_MPU_EMU)) && defined(CONFIG_MIDI)
+	static int      mpu_attached = 0;
+	static int      valid_ports[] =
+	{0x330, 0x320, 0x310, 0x300};
+	static short    valid_irqs[] =
+	{9, 10, 5, 7};
+	unsigned char   tmp;
+
+	int             i;	/* A variable with secret power */
+
+	if (!already_initialized)	/* The MSS port must be initialized
+					 * first */
+		return 0;
+
+	if (mpu_attached)	/* Don't let them call this twice */
+		return 0;
+	mpu_attached = 1;
+
+	if (board_type < C929) {/* Early chip. No MPU support. Just SB MIDI */
+
+#ifdef CONFIG_MIDI
+		unsigned char   tmp;
+
+		tmp = mad_read(MC3_PORT);
+
+		/*
+		 * MAD16 SB base is defined by the WSS base. It cannot be
+		 * changed alone. Ignore configured I/O base. Use the active
+		 * setting.
+		 */
+
+		if (mad_read(MC1_PORT) & 0x20)
+			hw_config->io_base = 0x240;
+		else
+			hw_config->io_base = 0x220;
+
+		switch (hw_config->irq) {
+		case 5:
+			tmp = (tmp & 0x3f) | 0x80;
+			break;
+		case 7:
+			tmp = (tmp & 0x3f);
+			break;
+		case 11:
+			tmp = (tmp & 0x3f) | 0x40;
+			break;
+		default:
+			printf("mad16/Mozart: Invalid MIDI IRQ\n");
+			return 0;
+		}
+
+		mad_write(MC3_PORT, tmp | 0x04);
+		return mad16_sb_dsp_detect(hw_config);
+#else
+		return 0;
+#endif
+	}
+	tmp = 0x83;		/* MPU-401 enable */
+
+	/*
+	 * Set the MPU base bits
+	 */
+
+	for (i = 0; i < 5; i++) {
+		if (i > 3) {	/* Out of array bounds */
+			printf("MAD16 / Mozart: Invalid MIDI port 0x%x\n", hw_config->io_base);
+			return 0;
+		}
+		if (valid_ports[i] == hw_config->io_base) {
+			tmp |= i << 5;
+			break;
+		}
+	}
+
+	/*
+	 * Set the MPU IRQ bits
+	 */
+
+	for (i = 0; i < 5; i++) {
+		if (i > 3) {	/* Out of array bounds */
+			printf("MAD16 / Mozart: Invalid MIDI IRQ %d\n", hw_config->irq);
+			return 0;
+		}
+		if (valid_irqs[i] == hw_config->irq) {
+			tmp |= i << 3;
+			break;
+		}
+	}
+	mad_write(MC6_PORT, tmp);	/* Write MPU401 config */
+
+	return probe_mpu401(hw_config);
+#else
+	return 0;
+#endif
+}
+
+/* That's all folks */
+#endif
diff --git a/sys/i386/isa/sound/mad16.h b/sys/i386/isa/sound/mad16.h
new file mode 100644
index 000000000000..0370973667a4
--- /dev/null
+++ b/sys/i386/isa/sound/mad16.h
@@ -0,0 +1,91 @@
+
+/*
+ *	Initialization code for OPTI MAD16 interface chip by
+ *	Davor Jadrijevic <davor@emard.pub.hr>
+ *	(Included by ad1848.c when MAD16 support is enabled)
+ *
+ * It looks like MAD16 is similar than the Mozart chip (OAK OTI-601).
+ * It could be even possible that these chips are exactly the same. Can
+ * anybody confirm this?
+ */
+
+static void wr_a_mad16(int base, int v, int a)
+{
+ OUTB(a, base + 0xf);
+ OUTB(v, base + 0x11);
+}
+
+static void wr_b_mad16(int base, int v, int a)
+{
+ OUTB(a, base + 0xf);
+ OUTB(v, base + 0xd);
+}
+
+/*
+static int rd_a_mad16(int base, int a)
+{
+ OUTB(a, base + 0xf);
+ return INB(base + 0x11);
+}
+*/
+
+static int rd_b_mad16(int base, int a)
+{
+ OUTB(a, base + 0xf);
+ return INB(base + 0xd);
+}
+
+/*
+static int rd_0_mad16(int base, int a)
+{
+ OUTB(a, base + 0xf);
+ return INB(base + 0xf);
+}
+
+static void wr_ad(int base, int v, int a)
+{
+ OUTB(a, base + 4);
+ OUTB(v, base + 5);
+}
+
+static int rd_ad(int base, int a)
+{
+ OUTB(a, base + 4);
+ return INB(base + 5);
+}
+*/
+
+static int mad16init(int adr)
+{
+ int j;
+ long i;
+
+ static int ad1848_bases[] = 
+{ 0x220, -1, -1, 0x240, -1, -1, -1, -1, 0x530, 0xE80, 0xF40, 0x604, 0 };
+
+ int mad16_base = 0xf80, ad1848_base;
+
+
+ for(j = 0; (j < 16) && (ad1848_bases[j] != 0); j++)
+  if(adr == ad1848_bases[j])
+   break;
+
+ if( (ad1848_base = ad1848_bases[j]) < 0x530)
+ {
+  printk("Unknown MAD16 setting 0x%3X\n", adr);
+  return -1;
+ }
+
+ /* printk("OPTi MAD16 WSS at 0x%3X\n", ad1848_base); */
+
+ rd_b_mad16(mad16_base, 0xe2);
+ wr_a_mad16(mad16_base, 0x1a, 0xe2);
+ wr_b_mad16(mad16_base, j * 16 + 1, 0xe2);
+ wr_a_mad16(mad16_base, 0x1a, 0xe2);
+ for( i = 0; i < 10000; i++)
+  if( (INB(ad1848_base+4) & 0x80) == 0 )
+   break;
+
+ return 0;
+};
+
diff --git a/sys/i386/isa/sound/mad16_sb_midi.c b/sys/i386/isa/sound/mad16_sb_midi.c
new file mode 100644
index 000000000000..d1b3ad0167cc
--- /dev/null
+++ b/sys/i386/isa/sound/mad16_sb_midi.c
@@ -0,0 +1,285 @@
+/*
+ * sound/mad16_sb_midi.c
+ * 
+ * The low level driver for MAD16 SoundBlaster-DS-chip-based MIDI.
+ * 
+ * Copyright by Hannu Savolainen 1993, Aaron Ucko 1995
+ * 
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are
+ * met: 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer. 2.
+ * Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED.  IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE FOR
+ * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+ * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
+ * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ * 
+ */
+
+#include <i386/isa/sound/sound_config.h>
+
+#if defined(CONFIG_MAD16) && defined(CONFIG_MIDI)
+
+#define sbc_base mad16_sb_base
+#include <i386/isa/sound/sb_card.h>
+
+static int      input_opened = 0;
+static int      my_dev;
+static int      mad16_sb_base = 0x220;
+static int      mad16_sb_irq = 0;
+static int      mad16_sb_dsp_ok = 0;
+static sound_os_info *midi_osp;
+
+int             mad16_sb_midi_mode = NORMAL_MIDI;
+int             mad16_sb_midi_busy = 0;
+
+int             mad16_sb_duplex_midi = 0;
+volatile int    mad16_sb_intr_active = 0;
+
+void            (*midi_input_intr) (int dev, unsigned char data);
+
+static void     mad16_sb_midi_init(int model);
+
+static int
+mad16_sb_dsp_command(unsigned char val)
+{
+	int             i;
+	unsigned long   limit;
+
+	limit = get_time() + hz / 10;	/* The timeout is 0.1 secods */
+
+	/*
+	 * Note! the i<500000 is an emergency exit. The
+	 * mad16_sb_dsp_command() is sometimes called while interrupts are
+	 * disabled. This means that the timer is disabled also. However the
+	 * timeout situation is a abnormal condition. Normally the DSP should
+	 * be ready to accept commands after just couple of loops.
+	 */
+
+	for (i = 0; i < 500000 && get_time() < limit; i++) {
+		if ((inb(DSP_STATUS) & 0x80) == 0) {
+			outb(DSP_COMMAND, val);
+			return 1;
+		}
+	}
+
+	printf("MAD16 (SBP mode): DSP Command(%x) Timeout.\n", val);
+	printf("IRQ conflict???\n");
+	return 0;
+}
+
+void
+mad16_sbintr(int irq)
+{
+	int             status;
+
+	unsigned long   flags;
+	unsigned char   data;
+
+	status = inb(DSP_DATA_AVAIL);	/* Clear interrupt */
+
+	flags = splhigh();
+
+	data = inb(DSP_READ);
+	if (input_opened)
+		midi_input_intr(my_dev, data);
+
+	splx(flags);
+}
+
+static int
+mad16_sb_reset_dsp(void)
+{
+	int             loopc;
+
+	outb(DSP_RESET, 1);
+	DELAY(10);
+	outb(DSP_RESET, 0);
+	DELAY(30);
+
+	for (loopc = 0; loopc < 100 && !(inb(DSP_DATA_AVAIL) & 0x80); loopc++)
+		DELAY(10);
+		/* Wait for data available status */
+
+	if (inb(DSP_READ) != 0xAA)
+		return 0;	/* Sorry */
+
+	return 1;
+}
+
+int
+mad16_sb_dsp_detect(struct address_info * hw_config)
+{
+	mad16_sb_base = hw_config->io_base;
+	mad16_sb_irq = hw_config->irq;
+	midi_osp = hw_config->osp;
+
+	if (mad16_sb_dsp_ok)
+		return 0;	/* Already initialized */
+	if (!mad16_sb_reset_dsp())
+		return 0;
+
+	return 1;		/* Detected */
+}
+
+void
+mad16_sb_dsp_init(struct address_info * hw_config)
+/*
+ * this function now just verifies the reported version and calls
+ * mad16_sb_midi_init -- everything else is done elsewhere
+ */
+{
+
+	midi_osp = hw_config->osp;
+	if (snd_set_irq_handler(mad16_sb_irq, mad16_sbintr, midi_osp) < 0) {
+		printf("MAD16 SB MIDI: IRQ not free\n");
+		return;
+	}
+
+	conf_printf("MAD16 MIDI (SB mode)", hw_config);
+	mad16_sb_midi_init(2);
+
+	mad16_sb_dsp_ok = 1;
+	return;
+}
+
+static int
+mad16_sb_midi_open(int dev, int mode,
+		   void (*input) (int dev, unsigned char data),
+		   void (*output) (int dev)
+)
+{
+
+	if (!mad16_sb_dsp_ok) {
+		printf("MAD16_SB Error: MIDI hardware not installed\n");
+		return -(ENXIO);
+	}
+	if (mad16_sb_midi_busy)
+		return -(EBUSY);
+
+	if (mode != OPEN_WRITE && !mad16_sb_duplex_midi) {
+		if (num_midis == 1)
+			printf("MAD16 (SBP mode): Midi input not currently supported\n");
+		return -(EPERM);
+	}
+	mad16_sb_midi_mode = NORMAL_MIDI;
+	if (mode != OPEN_WRITE) {
+		if (mad16_sb_intr_active)
+			return -(EBUSY);
+		mad16_sb_midi_mode = UART_MIDI;
+	}
+	if (mad16_sb_midi_mode == UART_MIDI) {
+		mad16_sb_reset_dsp();
+
+		if (!mad16_sb_dsp_command(0x35))
+			return -(EIO);	/* Enter the UART mode */
+		mad16_sb_intr_active = 1;
+
+		input_opened = 1;
+		midi_input_intr = input;
+	}
+	mad16_sb_midi_busy = 1;
+
+	return 0;
+}
+
+static void
+mad16_sb_midi_close(int dev)
+{
+	if (mad16_sb_midi_mode == UART_MIDI) {
+		mad16_sb_reset_dsp();	/* The only way to kill the UART mode */
+	}
+	mad16_sb_intr_active = 0;
+	mad16_sb_midi_busy = 0;
+	input_opened = 0;
+}
+
+static int
+mad16_sb_midi_out(int dev, unsigned char midi_byte)
+{
+	unsigned long   flags;
+
+	if (mad16_sb_midi_mode == NORMAL_MIDI) {
+		flags = splhigh();
+		if (mad16_sb_dsp_command(0x38))
+			mad16_sb_dsp_command(midi_byte);
+		else
+			printf("MAD16_SB Error: Unable to send a MIDI byte\n");
+		splx(flags);
+	} else
+		mad16_sb_dsp_command(midi_byte);	/* UART write */
+
+	return 1;
+}
+
+static int
+mad16_sb_midi_start_read(int dev)
+{
+	if (mad16_sb_midi_mode != UART_MIDI) {
+		printf("MAD16 (SBP mode): MIDI input not implemented.\n");
+		return -(EPERM);
+	}
+	return 0;
+}
+
+static int
+mad16_sb_midi_end_read(int dev)
+{
+	if (mad16_sb_midi_mode == UART_MIDI) {
+		mad16_sb_reset_dsp();
+		mad16_sb_intr_active = 0;
+	}
+	return 0;
+}
+
+static int
+mad16_sb_midi_ioctl(int dev, unsigned cmd, ioctl_arg arg)
+{
+	return -(EPERM);
+}
+
+#define MIDI_SYNTH_NAME	"pseudo-SoundBlaster Midi"
+#define MIDI_SYNTH_CAPS	0
+#include <i386/isa/sound/midi_synth.h>
+
+static struct midi_operations mad16_sb_midi_operations =
+{
+	{"MAD16 (SBP mode)", 0, 0, SNDCARD_MAD16},
+	&std_midi_synth,
+	{0},
+	mad16_sb_midi_open,
+	mad16_sb_midi_close,
+	mad16_sb_midi_ioctl,
+	mad16_sb_midi_out,
+	mad16_sb_midi_start_read,
+	mad16_sb_midi_end_read,
+	NULL,			/* Kick */
+	NULL,			/* command */
+	NULL,			/* buffer_status */
+	NULL
+};
+
+static void
+mad16_sb_midi_init(int model)
+{
+	if (num_midis >= MAX_MIDI_DEV) {
+		printf("Sound: Too many midi devices detected\n");
+		return;
+	}
+	std_midi_synth.midi_dev = num_midis;
+	my_dev = num_midis;
+	midi_devs[num_midis++] = &mad16_sb_midi_operations;
+}
+
+#endif
diff --git a/sys/i386/isa/sound/maui.c b/sys/i386/isa/sound/maui.c
new file mode 100644
index 000000000000..e92b48a4bff2
--- /dev/null
+++ b/sys/i386/isa/sound/maui.c
@@ -0,0 +1,225 @@
+/*
+ * sound/maui.c
+ * 
+ * The low level driver for Turtle Beach Maui and Tropez.
+ * 
+ * Copyright by Hannu Savolainen 1995
+ * 
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are
+ * met: 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer. 2.
+ * Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED.  IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE FOR
+ * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+ * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
+ * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ * 
+ */
+
+#define USE_SEQ_MACROS
+#define USE_SIMPLE_MACROS
+
+#include <i386/isa/sound/sound_config.h>
+
+
+#if defined(CONFIG_MAUI)
+
+static int      maui_base = 0x330;
+
+static volatile int irq_ok = 0;
+static sound_os_info *maui_osp;
+
+#define HOST_DATA_PORT	(maui_base + 2)
+#define HOST_STAT_PORT	(maui_base + 3)
+#define HOST_CTRL_PORT	(maui_base + 3)
+
+#define STAT_TX_INTR	0x40
+#define STAT_TX_AVAIL	0x20
+#define STAT_TX_IENA	0x10
+#define STAT_RX_INTR	0x04
+#define STAT_RX_AVAIL	0x02
+#define STAT_RX_IENA	0x01
+
+static int      (*orig_load_patch) (int dev, int format, snd_rw_buf * addr,
+		                 int offs, int count, int pmgr_flag) = NULL;
+
+static int
+maui_read(void)
+{
+    int             timeout;
+
+    for (timeout = 0; timeout < 1000000; timeout++)
+	if (inb(HOST_STAT_PORT) & STAT_RX_AVAIL)
+	    return inb(HOST_DATA_PORT);
+
+    printf("Maui: Receive timeout\n");
+
+    return -1;
+}
+
+static int
+maui_write(u_char data)
+{
+    int             timeout;
+
+    for (timeout = 0; timeout < 10000000; timeout++) {
+	if (inb(HOST_STAT_PORT) & STAT_TX_AVAIL) {
+	    outb(HOST_DATA_PORT, data);
+	    return 1;
+	}
+    }
+
+    printf("Maui: Write timeout\n");
+
+    return 0;
+}
+
+void
+mauiintr(int irq)
+{
+    irq_ok = 1;
+}
+
+
+int
+maui_load_patch(int dev, int format, snd_rw_buf * addr,
+		int offs, int count, int pmgr_flag)
+{
+
+    struct sysex_info header;
+    u_long   left, src_offs;
+    int             hdr_size = (u_long) &header.data[0] - (u_long) &header;
+    int             i;
+
+    if (format == SYSEX_PATCH)	/* Handled by midi_synth.c */
+	return orig_load_patch(dev, format, addr, offs, count, pmgr_flag);
+
+    if (format != MAUI_PATCH) {
+	printf("Maui: Unknown patch format\n");
+    }
+    if (count < hdr_size) {
+	printf("Maui error: Patch header too short\n");
+	return -(EINVAL);
+    }
+    count -= hdr_size;
+
+    /*
+     * Copy the header from user space but ignore the first bytes which
+     * have been transferred already.
+     */
+
+    if (uiomove(&((char *) &header)[offs], hdr_size - offs, addr)) {
+	printf("sb: Bad copyin()!\n");
+    };
+
+    if (count < header.len) {
+	printf("Maui warning: Host command record too short (%d<%d)\n",
+	       count, (int) header.len);
+	header.len = count;
+    }
+    left = header.len;
+    src_offs = 0;
+
+    for (i = 0; i < left; i++) {
+	u_char   data;
+	uiomove((char *) &(data), 1, addr);
+	if (i == 0 && !(data & 0x80))
+	    return -(EINVAL);
+
+	if (maui_write(data) == -1)
+	    return -(EIO);
+    }
+
+    if ((i = maui_read()) != 0x80) {
+	if (i != -1)
+	    printf("Maui: Error status %02x\n", i);
+
+	return -(EIO);
+    }
+    return 0;
+}
+
+int
+probe_maui(struct address_info * hw_config)
+{
+    int             i;
+    int             tmp1, tmp2;
+
+    maui_base = hw_config->io_base;
+    maui_osp = hw_config->osp;
+
+    if (snd_set_irq_handler(hw_config->irq, mauiintr, maui_osp) < 0)
+	return 0;
+
+    if (!maui_write(0xCF)) {/* Report hardware version */
+	/* snd_release_irq(hw_config->irq); */
+	return 0;
+    }
+    if ((tmp1 = maui_read()) == -1 || (tmp2 = maui_read()) == -1) {
+	/* snd_release_irq(hw_config->irq); */
+	return 0;
+    }
+    printf("WaveFront hardware version %d.%d\n", tmp1, tmp2);
+
+    if (!maui_write(0x9F))	/* Report firmware version */
+	return 0;
+    if ((tmp1 = maui_read()) == -1 || (tmp2 = maui_read()) == -1)
+	return 0;
+    printf("WaveFront firmware version %d.%d\n", tmp1, tmp2);
+
+    if (!maui_write(0x85))	/* Report free DRAM */
+	return 0;
+    tmp1 = 0;
+    for (i = 0; i < 4; i++) {
+	tmp1 |= maui_read() << (7 * i);
+    }
+    printf("Available DRAM %dk\n", tmp1 / 1024);
+
+    for (i = 0; i < 1000; i++)
+	if (probe_mpu401(hw_config))
+	    break;
+
+    return probe_mpu401(hw_config);
+}
+
+void
+attach_maui(struct address_info * hw_config)
+{
+    int             this_dev = num_midis;
+
+    conf_printf("Maui", hw_config);
+
+    hw_config->irq *= -1;
+    attach_mpu401(hw_config);
+
+    if (num_midis > this_dev) {	/* The MPU401 driver installed itself */
+	struct synth_operations *synth;
+
+	/*
+	 * Intercept patch loading calls so that they canbe handled
+	 * by the Maui driver.
+	 */
+
+	synth = midi_devs[this_dev]->converter;
+
+	if (synth != NULL) {
+	    orig_load_patch = synth->load_patch;
+	    synth->load_patch = &maui_load_patch;
+	} else
+	    printf("Maui: Can't install patch loader\n");
+    }
+    return;
+}
+
+#endif
diff --git a/sys/i386/isa/sound/mmap_test.c b/sys/i386/isa/sound/mmap_test.c
new file mode 100644
index 000000000000..229144939301
--- /dev/null
+++ b/sys/i386/isa/sound/mmap_test.c
@@ -0,0 +1,274 @@
+/*
+ * This is a simple program which demonstrates use of mmapped DMA buffer
+ * of the sound driver directly from application program.
+ *
+ * This sample program works (currently) only with Linux, FreeBSD and BSD/OS
+ * (FreeBSD and BSD/OS require OSS version 3.8-beta16 or later.
+ *
+ * Note! Don't use mmapped DMA buffers (direct audio) unless you have
+ * very good reasons to do it. Programs using this feature will not
+ * work with all soundcards. GUS (GF1) is one of them (GUS MAX works).
+ *
+ * This program requires version 3.5-beta7 or later of OSS
+ * (3.8-beta16 or later in FreeBSD and BSD/OS).
+ */
+
+#include <stdio.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/types.h>
+#include <sys/mman.h>
+#include <machine/soundcard.h>
+#include <sys/time.h>
+
+main()
+{
+	int fd, sz, fsz, i, tmp, n, l, have_data=0, nfrag;
+        int caps;
+
+	int sd, sl=0, sp;
+
+	unsigned char data[500000], *dp = data;
+
+	struct buffmem_desc imemd, omemd;
+        caddr_t buf;
+	struct timeval tim;
+
+	unsigned char *op;
+	
+        struct audio_buf_info info;
+
+	int frag = 0xffff000c;	/* Max # fragments of 2^13=8k bytes */
+
+	fd_set writeset;
+
+	close(0);
+	if ((fd=open("/dev/dsp", O_RDWR, 0))==-1)
+	{
+		perror("/dev/dsp");
+		exit(-1);
+	}
+/*
+ * Then setup sampling parameters. Just sampling rate in this case.
+ */
+
+	tmp = 8000;
+	ioctl(fd, SNDCTL_DSP_SPEED, &tmp);
+
+/*
+ * Load some test data.
+ */
+
+  sl = sp = 0;
+  if ((sd=open("smpl", O_RDONLY, 0))!=-1)
+  {
+	sl = read(sd, data, sizeof(data));
+	printf("%d bytes read from file.\n", sl);
+	close(sd);
+  }
+  else perror("smpl");
+
+	if (ioctl(fd, SNDCTL_DSP_GETCAPS, &caps)==-1)
+	{
+		perror("/dev/dsp");
+		fprintf(stderr, "Sorry but your sound driver is too old\n");
+		exit(-1);
+	}
+
+/*
+ * Check that the device has capability to do this. Currently just
+ * CS4231 based cards will work.
+ *
+ * The application should also check for DSP_CAP_MMAP bit but this
+ * version of driver doesn't have it yet.
+ */
+/*	ioctl(fd, SNDCTL_DSP_SETSYNCRO, 0); */
+
+/*
+ * You need version 3.5-beta7 or later of the sound driver before next
+ * two lines compile. There is no point to modify this program to
+ * compile with older driver versions since they don't have working
+ * mmap() support.
+ */
+	if (!(caps & DSP_CAP_TRIGGER) ||
+	    !(caps & DSP_CAP_MMAP))
+	{
+		fprintf(stderr, "Sorry but your soundcard can't do this\n");
+		exit(-1);
+	}
+
+/*
+ * Select the fragment size. This is propably important only when
+ * the program uses select(). Fragment size defines how often
+ * select call returns.
+ */
+
+	ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &frag);
+
+/*
+ * Compute total size of the buffer. It's important to use this value
+ * in mmap() call.
+ */
+
+	if (ioctl(fd, SNDCTL_DSP_GETOSPACE, &info)==-1)
+	{
+		perror("GETOSPACE");
+		exit(-1);
+	}
+
+	sz = info.fragstotal * info.fragsize;
+	fsz = info.fragsize;
+/*
+ * Call mmap().
+ * 
+ * IMPORTANT NOTE!!!!!!!!!!!
+ *
+ * Full duplex audio devices have separate input and output buffers. 
+ * It is not possible to map both of them at the same mmap() call. The buffer
+ * is selected based on the prot argument in the following way:
+ *
+ * - PROT_READ (alone) selects the input buffer.
+ * - PROT_WRITE (alone) selects the output buffer.
+ * - PROT_WRITE|PROT_READ together select the output buffer. This combination
+ *   is required in BSD to make the buffer accessible. With just PROT_WRITE
+ *   every attempt to access the returned buffer will result in segmentation/bus
+ *   error. PROT_READ|PROT_WRITE is also permitted in Linux with OSS version
+ *   3.8-beta16 and later (earlier versions don't accept it).
+ *
+ * Non duplex devices have just one buffer. When an application wants to do both
+ * input and output it's recommended that the device is closed and re-opened when
+ * switching between modes. PROT_READ|PROT_WRITE can be used to open the buffer
+ * for both input and output (with OSS 3.8-beta16 and later) but the result may be
+ * unpredictable.
+ */
+
+	if ((buf=mmap(NULL, sz, PROT_WRITE | PROT_READ, MAP_FILE|MAP_SHARED, fd, 0))==(caddr_t)-1)
+  	{
+		perror("mmap (write)");
+		exit(-1);
+	}
+	printf("mmap (out) returned %08x\n", buf);
+	op=buf;
+
+/*
+ * op contains now a pointer to the DMA buffer
+ */
+
+/*
+ * Then it's time to start the engine. The driver doesn't allow read() and/or
+ * write() when the buffer is mapped. So the only way to start operation is
+ * to togle device's enable bits. First set them off. Setting them on enables
+ * recording and/or playback.
+ */
+
+	tmp = 0;
+	ioctl(fd, SNDCTL_DSP_SETTRIGGER, &tmp);  
+
+/*
+ * It might be usefull to write some data to the buffer before starting.
+ */
+
+	tmp = PCM_ENABLE_OUTPUT;
+	ioctl(fd, SNDCTL_DSP_SETTRIGGER, &tmp);
+
+/*
+ * The machine is up and running now. Use SNDCTL_DSP_GETOPTR to get the
+ * buffer status.
+ *
+ * NOTE! The driver empties each buffer fragmen after they have been
+ * played. This prevents looping sound if there are some performance problems
+ * in the application side. For similar reasons it recommended that the
+ * application uses some amout of play ahead. It can rewrite the unplayed
+ * data later if necessary.
+ */
+
+	nfrag = 0;
+	while (1)
+	{
+		struct count_info count;
+		int p, l, extra;
+
+		FD_ZERO(&writeset);
+		FD_SET(fd, &writeset);
+
+		tim.tv_sec = 10;
+		tim.tv_usec= 0;
+
+		select(fd+1, &writeset, &writeset, NULL, NULL);
+/*
+ * SNDCTL_DSP_GETOPTR (and GETIPTR as well) return three items. The
+ * bytes field returns number of bytes played since start. It can be used
+ * as a real time clock.
+ *
+ * The blocks field returns number of fragment transitions (interrupts) since
+ * previous GETOPTR call. It can be used as a method to detect underrun 
+ * situations.
+ *
+ * The ptr field is the DMA pointer inside the buffer area (in bytes from
+ * the beginning of total buffer area).
+ */
+
+		if (ioctl(fd, SNDCTL_DSP_GETOPTR, &count)==-1)
+		{
+			perror("GETOPTR");
+			exit(-1);
+		}
+                if (count.ptr < 0 ) count.ptr = 0;
+		nfrag += count.blocks;
+
+
+#ifdef VERBOSE
+
+		printf("\rTotal: %09d, Fragment: %03d, Ptr: %06d",
+			count.bytes, nfrag, count.ptr);
+		fflush(stdout);
+#endif
+
+/*
+ * Caution! This version doesn't check for bounds of the DMA
+ * memory area. It's possible that the returned pointer value is not aligned
+ * to fragment boundaries. It may be several samples behind the boundary
+ * in case there was extra delay between the actual hardware interrupt and
+ * the time when DSP_GETOPTR was called.
+ *
+ * Don't just call memcpy() with length set to 'fragment_size' without
+ * first checking that the transfer really fits to the buffer area.
+ * A mistake of just one byte causes seg fault. It may be easiest just
+ * to align the returned pointer value to fragment boundary before using it.
+ *
+ * It would be very good idea to write few extra samples to next fragment
+ * too. Otherwise several (uninitialized) samples from next fragment
+ * will get played before your program gets chance to initialize them.
+ * Take in count the fact thaat there are other processes batling about
+ * the same CPU. This effect is likely to be very annoying if fragment
+ * size is decreased too much.
+ */
+
+/*
+ * Just a minor clarification to the above. The following line alings
+ * the pointer to fragment boundaries. Note! Don't trust that fragment
+ * size is always a power of 2. It may not be so in future.
+ */
+		count.ptr = ((count.ptr+16)/fsz )*fsz;
+#ifdef VERBOSE
+		printf(" memcpy(%6d, %4d)", (dp-data), fsz);
+		fflush(stdout);
+#endif
+
+/*
+ * Set few bytes in the beginning of next fragment too.
+ */
+
+		if ((count.ptr+fsz+16) < sz)	/* Last fragment? */
+		   extra = 16;
+		else
+		   extra = 0;
+		memcpy(op+count.ptr, dp, (fsz+extra));
+		dp += fsz;
+		if (dp > (data+sl-fsz))
+		   dp = data;
+
+	}
+
+	exit(0);
+}
diff --git a/sys/i386/isa/sound/pas_defs.h b/sys/i386/isa/sound/pas_defs.h
new file mode 100644
index 000000000000..c4af2d499db4
--- /dev/null
+++ b/sys/i386/isa/sound/pas_defs.h
@@ -0,0 +1,250 @@
+/* 																*/
+/*	Port addresses and bit fields for the Media Vision Pro AudioSpectrum second generation sound cards.			*/
+/* 																*/
+/* 	Feel free to use this header file in any application you create that has support for the Media Vision			*/
+/*	Pro AudioSpectrum second generation sound cards. Other uses prohibited without prior permission.			*/
+/* 																*/
+/*										- cmetz@thor.tjhsst.edu				*/
+/* 																*/
+/*	Notes:															*/
+/* 																*/
+/*	*	All of these ports go into the MVD101 multimedia controller chip, which	then signals the other chips to do	*/
+/* 		the actual work. Many ports like the FM ones functionally attach directly to the destination chip though 	*/
+/* 		they don't actually have a direct connection.									*/
+/* 																*/
+/* 	*	The PAS2 series cards have an MVD101 multimedia controller chip, the original PAS cards don't. The original	*/
+/* 		PAS cards are pretty defunct now, so no attempt is made here to support them. 					*/
+/*																*/
+/*	*	The PAS2 series cards are all really different at the hardware level, though the MVD101 hides some of the 	*/
+/*		incompatibilities, there still are differences that need to be accounted for.					*/
+/*																*/
+/*		Card		CD-ROM interface	PCM chip		Mixer chip		FM chip			*/
+/* 		PAS Plus	Sony proprietary	(Crystal?) 8-bit DAC	National 		OPL3			*/
+/* 		PAS 16		Zilog SCSI		MVA416 16-bit Codec	MVA508 			OPL3			*/
+/* 		CDPC		Sony proprietary	Sony 16-bit Codec	National		OPL3			*/
+/*		Fusion CD 16	Sony proprietary	MVA416 16-bit Codec	MVA508			OPL3			*/
+/*		Fusion CD	Sony proprietary	(Crystal?) 8-bit DAC	National		OPL3			*/
+/* 																*/
+#define PAS_DEFAULT_BASE		0x388
+
+/*      Symbolic Name			Value 		   R W  Subsystem	Description					*/
+#define SPEAKER_CONTROL			0x61		/*   W	PC speaker 	Control register 				*/
+#define SPEAKER_CONTROL_GHOST		0x738B		/* R W	PC speaker 	Control ghost register				*/
+#define SPEAKER_TIMER_CONTROL		0x43		/*   W  PC speaker 	Timer control register				*/
+#define SPEAKER_TIMER_CONTROL_GHOST	0x778B		/* R W  PC speaker 	Timer control register ghost			*/
+#define SPEAKER_TIMER_DATA		0x42		/*   W  PC speaker 	Timer data register				*/
+#define SPEAKER_TIMER_DATA_GHOST	0x138A		/* R W  PC speaker	Timer data register ghost			*/
+
+#define WARM_BOOT			0x41		/*   W  Control		Used to detect system warm boot	  		*/
+#define WARM_BOOT_GHOST			0x7789		/* ? W  Control		Use to get the card to fake warm boot		*/
+#define MASTER_DECODE			0x9A01		/*   W  Control		Address >> 2 of card base address		*/
+#define PRESCALE_DIVIDER		0xBF8A		/* R W	PCM		Ration between Codec clock and master clock	*/
+#define WAIT_STATE			0xBF88		/* R W	Control		Four-bit bus wait-state count (~140ns ea.)	*/
+#define BOARD_REV_ID			0x2789		/* R	Control		Extended Board Revision ID			*/
+
+#define CHIP_REV			0xFF88		/* R	0=PAS, 1=PAS+, 2=CDPC, 3=PAS16C, 4=PAS16D */
+
+#define SYSTEM_CONFIGURATION_1		0x8388		/* R W	Control								*/
+	#define S_C_1_PCS_ENABLE	0x01		/* R W  PC speaker	1=enable, 0=disable PC speaker emulation	*/
+	#define S_C_1_PCM_CLOCK_SELECT	0x02		/* R W  PCM		1=14.31818Mhz/12, 0=28.224Mhz master clock	*/ 
+	#define S_C_1_FM_EMULATE_CLOCK	0x04		/* R W  FM		1=use 28.224Mhz/2, 0=use 14.31818Mhz clock	*/
+	#define S_C_1_PCS_STEREO	0x10 		/* R W  PC speaker	1=enable PC speaker stereo effect, 0=disable	*/
+	#define S_C_1_PCS_REALSOUND	0x20		/* R W  PC speaker	1=enable RealSound enhancement, 0=disable	*/
+	#define S_C_1_FORCE_EXT_RESET	0x40		/* R W  Control		Force external reset				*/
+	#define S_C_1_FORCE_INT_RESET	0x80		/* R W  Control		Force internal reset				*/
+#define SYSTEM_CONFIGURATION_2		0x8389		/* R W  Control								*/
+	#define S_C_2_PCM_OVERSAMPLING	0x03		/* R W  PCM		00=0x, 01=2x, 10=4x, 11=reserved		*/
+	#define S_C_2_PCM_16_BIT	0x04		/* R W	PCM		1=16-bit, 0=8-bit samples			*/
+#define SYSTEM_CONFIGURATION_3		0x838A		/* R W	Control								*/
+	#define S_C_3_PCM_CLOCK_SELECT	0x02		/* R W	PCM		1=use 1.008Mhz clock for PCM, 0=don't		*/
+#define SYSTEM_CONFIGURATION_4		0x838B		/* R W  Control		CD-ROM interface controls			*/
+
+#define IO_CONFIGURATION_1		0xF388		/* R W	Control								*/
+	#define I_C_1_BOOT_RESET_ENABLE	0x80		/* R W  Control		1=reset board on warm boot, 0=don't		*/
+	#define I_C_1_JOYSTICK_ENABLE	0x40		/* R W  Control		1=enable joystick port, 0=don't		        */
+#define IO_CONFIGURATION_2		0xF389		/* R W  Control								*/
+	#define	I_C_2_PCM_DMA_DISABLED	0x00		/* R W  PCM		PCM DMA disabled				*/
+#define IO_CONFIGURATION_3		0xF38A		/* R W	Control								*/
+	#define I_C_3_PCM_IRQ_DISABLED	0x00		/* R W  PCM		PCM IRQ disabled				*/
+
+#define COMPATIBILITY_ENABLE		0xF788		/* R W  Control								*/
+	#define C_E_MPU401_ENABLE	0x01		/* R W	MIDI		1=enable, 0=disable MPU401 MIDI emulation	*/
+	#define C_E_SB_ENABLE		0x02		/* R W  PCM		1=enable, 0=disable Sound Blaster emulation	*/
+	#define C_E_SB_ACTIVE		0x04		/* R    PCM		"Sound Blaster Interrupt active"		*/
+	#define C_E_MPU401_ACTIVE	0x08		/* R	MIDI		"MPU UART mode active"				*/
+	#define C_E_PCM_COMPRESSION	0x10		/* R W  PCM		1=enable, 0=disabled compression		*/
+#define EMULATION_ADDRESS		0xF789		/* R W  Control								*/
+	#define E_A_SB_BASE		0x0f		/* R W  PCM		bits A4-A7 for SB base port			*/
+	#define E_A_MPU401_BASE		0xf0		/* R W	MIDI		bits A4-A7 for MPU401 base port 		*/
+#define EMULATION_CONFIGURATION		0xFB8A		/* R W			***** Only valid on newer PAS2 cards (?) *****	*/
+	#define E_C_MPU401_IRQ		0x07		/* R W	MIDI		MPU401 emulation IRQ				*/
+	#define E_C_SB_IRQ		0x38		/* R W  PCM		SB emulation IRQ				*/
+	#define E_C_SB_DMA		0xC0		/* R W	PCM		SB emulation DMA				*/
+
+#define OPERATION_MODE_1		0xEF8B		/* R	Control								*/
+	#define	O_M_1_CDROM_TYPE	0x03		/* R	CD-ROM		3=SCSI, 2=Sony, 0=no CD-ROM interface		*/
+	#define O_M_1_FM_TYPE		0x04		/* R	FM		1=sterero, 0=mono FM chip			*/
+	#define O_M_1_PCM_TYPE 		0x08		/* R	PCM		1=16-bit Codec, 0=8-bit DAC			*/
+#define OPERATION_MODE_2		0xFF8B		/* R	Control								*/
+	#define O_M_2_PCS_ENABLED	0x02		/* R	PC speaker	PC speaker emulation 1=enabled, 0=disabled	*/
+	#define O_M_2_BUS_TIMING	0x10		/* R	Control		1=AT bus timing, 0=XT bus timing		*/
+	#define O_M_2_BOARD_REVISION	0xe0		/* R	Control		Board revision					*/
+
+#define INTERRUPT_MASK			0x0B8B		/* R W	Control								*/
+	#define I_M_FM_LEFT_IRQ_ENABLE	0x01		/* R W	FM		Enable FM left interrupt			*/
+	#define I_M_FM_RIGHT_IRQ_ENABLE	0x02		/* R W	FM		Enable FM right interrupt			*/
+	#define I_M_PCM_RATE_IRQ_ENABLE	0x04		/* R W	PCM		Enable Sample Rate interrupt			*/
+	#define I_M_PCM_BUFFER_IRQ_ENABLE 0x08		/* R W	PCM		Enable Sample Buffer interrupt			*/
+	#define I_M_MIDI_IRQ_ENABLE	0x10		/* R W	MIDI		Enable MIDI interrupt				*/
+	#define I_M_BOARD_REV		0xE0		/* R	Control		Board revision					*/
+
+#define INTERRUPT_STATUS		0x0B89		/* R W	Control								*/
+	#define I_S_FM_LEFT_IRQ		0x01		/* R W	FM		Left FM Interrupt Pending			*/
+	#define I_S_FM_RIGHT_IRQ	0x02		/* R W	FM		Right FM Interrupt Pending			*/
+	#define I_S_PCM_SAMPLE_RATE_IRQ	0x04		/* R W	PCM		Sample Rate Interrupt Pending			*/
+	#define I_S_PCM_SAMPLE_BUFFER_IRQ 0x08		/* R W	PCM		Sample Buffer Interrupt Pending			*/
+	#define I_S_MIDI_IRQ		0x10		/* R W	MIDI		MIDI Interrupt Pending				*/
+	#define I_S_PCM_CHANNEL		0x20		/* R W	PCM		1=right, 0=left					*/
+	#define I_S_RESET_ACTIVE	0x40		/* R W	Control		Reset is active (Timed pulse not finished)	*/
+	#define I_S_PCM_CLIPPING	0x80		/* R W	PCM		Clipping has occurred				*/
+
+#define FILTER_FREQUENCY		0x0B8A		/* R W	Control								*/
+	#define F_F_FILTER_DISABLED	0x00		/* R W 	Mixer		No filter					*/
+#if 0
+	struct {					/* R W	Mixer		Filter translation				*/
+		unsigned int freq:24;
+		unsigned int value:8;
+	} F_F_FILTER_translate[] = 
+	{ { 73500, 0x01 },	/* 73500Hz - divide by  16 */
+	  { 65333, 0x02 },	/* 65333Hz - divide by  18 */
+	  { 49000, 0x09 },	/* 49000Hz - divide by  24 */
+	  { 36750, 0x11 },	/* 36750Hz - divide by  32 */
+	  { 24500, 0x19 },	/* 24500Hz - divide by  48 */
+	  { 18375, 0x07 },	/* 18375Hz - divide by  64 */
+	  { 12783, 0x0f }, 	/* 12783Hz - divide by  92 */
+	  { 12250, 0x04 },	/* 12250Hz - divide by  96 */
+	  {  9188, 0x17 }, 	/*  9188Hz - divide by 128 */
+	  {  6125, 0x1f },	/*  6125Hz - divide by 192 */
+	};
+#endif
+	#define F_F_MIXER_UNMUTE	0x20		/* R W	Mixer		1=disable, 0=enable board mute			*/
+	#define F_F_PCM_RATE_COUNTER	0x40		/* R W	PCM		1=enable, 0=disable sample rate counter		*/
+	#define F_F_PCM_BUFFER_COUNTER	0x80		/* R W 	PCM		1=enable, 0=disable sample buffer counter	*/
+
+#define PAS_NONE	0
+#define PAS_PLUS	1
+#define PAS_CDPC	2
+#define PAS_16		3
+#define PAS_16D		4
+
+#ifdef DEFINE_TRANSLATIONS
+static	unsigned char I_C_2_PCM_DMA_translate[] = 		/* R W  PCM		PCM DMA channel value translations		*/
+			{ 4, 1, 2, 3, 0, 5, 6, 7 };
+static	unsigned char I_C_3_PCM_IRQ_translate[] = 		/* R W	PCM		PCM IRQ level value translation			*/
+		{ 0,  0,  1,  2,  3,  4,  5,  6, 0,  1,  7,  8,  9,  0, 10, 11 };  
+#ifdef unused
+static	unsigned char E_C_MPU401_IRQ_translate[] = 		/* R W	MIDI		MPU401 emulation IRQ value translation		*/
+		{ 0x00, 0x00, 0x01, 0x02, 0x00, 0x03, 0x00, 0x04, 0x00, 0x01, 0x05, 0x06, 0x07 };
+#endif
+static	unsigned char E_C_SB_IRQ_translate[] = 			/* R W	PCM		SB emulation IRQ translate			*/
+		{ 0x00, 0x00, 0x08, 0x10, 0x00, 0x18, 0x00, 0x20, 0x00, 0x08, 0x28, 0x30, 0x38, 0, 0 };
+static	unsigned char E_C_SB_DMA_translate[] = 			/* R W	PCM		SB emulation DMA translate			*/
+		{ 0x00, 0x40, 0x80, 0xC0, 0, 0, 0, 0 };
+#ifdef unused
+static	unsigned char O_M_1_to_card[] = 				/* R W	Control		Translate (OM1 & 0x0f) to card type		*/
+		{ 0, 0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 0, 4, 0, 2, 3 };   
+#endif
+#endif
+
+#define PARALLEL_MIXER			0x078B		/*   W	Mixer		Documented for MVD101 as FM Mono Right decode?? */
+	#define P_M_MV508_ADDRESS	0x80		/*   W	Mixer		MVD508	Address/mixer select			*/
+	#define P_M_MV508_DATA		0x00
+	#define P_M_MV508_LEFT		0x20		/*   W	Mixer		MVD508	Left channel select			*/
+	#define P_M_MV508_RIGHT		0x40		/*   W	Mixer		MVD508	Right channel select			*/
+	#define P_M_MV508_BOTH		0x00		/*   W	Mixer		MVD508	Both channel select			*/
+	#define P_M_MV508_MIXER		0x10		/*   W	Mixer		MVD508	Select a mixer (rather than a volume) 	*/
+	#define P_M_MV508_VOLUME	0x00
+
+	#define P_M_MV508_INPUTMIX	0x20		/*   W  Mixer		MVD508	Select mixer A				*/
+	#define P_M_MV508_OUTPUTMIX	0x00		/*   W  Mixer		MVD508	Select mixer B				*/
+
+	#define P_M_MV508_MASTER_A	0x01		/*   W	Mixer		MVD508	Master volume control A (output)	*/
+	#define P_M_MV508_MASTER_B	0x02		/*   W	Mixer		MVD508	Master volume control B (DSP input)	*/
+	#define P_M_MV508_BASS		0x03		/*   W	Mixer		MVD508	Bass control				*/
+	#define P_M_MV508_TREBLE	0x04		/*   W	Mixer		MVD508	Treble control				*/
+	#define P_M_MV508_MODE		0x05		/*   W	Mixer		MVD508	Master mode control			*/
+
+	#define P_M_MV508_LOUDNESS	0x04		/*   W	Mixer		MVD508	Mode control - Loudness filter 		*/
+	#define P_M_MV508_ENHANCE_BITS	0x03
+	#define P_M_MV508_ENHANCE_NONE	0x00		/*   W	Mixer		MVD508	Mode control - No stereo enhancement	*/
+	#define P_M_MV508_ENHANCE_40	0x01		/*   W	Mixer		MVD508	Mode control - 40% stereo enhancement	*/
+	#define P_M_MV508_ENHANCE_60	0x02		/*   W	Mixer		MVD508	Mode control - 60% stereo enhancement	*/
+	#define P_M_MV508_ENHANCE_80	0x03		/*   W	Mixer		MVD508	Mode control - 80% stereo enhancement	*/
+
+	#define P_M_MV508_FM		0x00		/*   W	Mixer		MVD508	Channel 0 - FM				*/
+	#define P_M_MV508_IMIXER	0x01		/*   W	Mixer		MVD508	Channel 1 - Input mixer (rec monitor)	*/
+	#define P_M_MV508_LINE		0x02		/*   W	Mixer		MVD508	Channel 2 - Line in			*/
+	#define P_M_MV508_CDROM		0x03		/*   W	Mixer		MVD508	Channel 3 - CD-ROM			*/
+	#define P_M_MV508_MIC		0x04		/*   W	Mixer		MVD508	Channel 4 - Microphone			*/
+	#define P_M_MV508_PCM		0x05		/*   W	Mixer		MVD508	Channel 5 - PCM				*/
+	#define P_M_MV508_SPEAKER	0x06		/*   W	Mixer		MVD508	Channel 6 - PC Speaker			*/
+	#define P_M_MV508_SB		0x07		/*   W	Mixer		MVD508	Channel 7 - SB DSP			*/
+
+#define SERIAL_MIXER			0xB88		/* R W	Control		Serial mixer control (used other ways)		*/
+	#define S_M_PCM_RESET		0x01		/* R W	PCM		Codec/DSP reset					*/
+	#define S_M_FM_RESET		0x02		/* R W	FM		FM chip reset					*/
+	#define S_M_SB_RESET		0x04		/* R W	PCM		SB emulation chip reset				*/
+	#define S_M_MIXER_RESET		0x10		/* R W	Mixer		Mixer chip reset				*/
+	#define S_M_INTEGRATOR_ENABLE	0x40		/* R W	Speaker		Enable PC speaker integrator (FORCE RealSound)	*/
+	#define S_M_OPL3_DUAL_MONO	0x80		/* R W  FM		Set the OPL-3 to dual mono mode			*/
+
+#define PCM_CONTROL			0xF8A		/* R W	PCM		PCM Control Register				*/
+        #define P_C_MIXER_CROSS_FIELD	0x0f
+	#define P_C_MIXER_CROSS_R_TO_R	0x01		/* R W	Mixer		Connect Right to Right				*/
+	#define P_C_MIXER_CROSS_L_TO_R	0x02		/* R W	Mixer		Connect Left  to Right				*/
+	#define P_C_MIXER_CROSS_R_TO_L	0x04		/* R W	Mixer		Connect Right to Left				*/
+	#define P_C_MIXER_CROSS_L_TO_L	0x08		/* R W	Mixer		Connect Left  to Left				*/
+	#define P_C_PCM_DAC_MODE	0x10		/* R W	PCM		Playback (DAC) mode				*/
+	#define P_C_PCM_ADC_MODE	0x00		/* R W	PCM		Record (ADC) mode				*/
+	#define P_C_PCM_MONO		0x20		/* R W	PCM		Mono mode					*/
+	#define P_C_PCM_STEREO		0x00		/* R W	PCM		Stereo mode					*/
+	#define P_C_PCM_ENABLE		0x40		/* R W	PCM		Enable PCM engine				*/
+	#define P_C_PCM_DMA_ENABLE	0x80		/* R W	PCM		Enable DRQ					*/
+
+#define SAMPLE_COUNTER_CONTROL		0x138B		/* R W	PCM		Sample counter control register			*/
+	#define S_C_C_SQUARE_WAVE	0x04		/* R W	PCM		Square wave generator (use for sample rate)	*/
+	#define S_C_C_RATE		0x06		/* R W	PCM		Rate generator (use for sample buffer count)	*/
+	#define S_C_C_LSB_THEN_MSB	0x30		/* R W	PCM		Change all 16 bits, LSB first, then MSB		*/
+
+	/* MVD101 and SDK documentations have S_C_C_SAMPLE_RATE and S_C_C_SAMPLE_BUFFER transposed. Only one works :-) */
+	#define S_C_C_SAMPLE_RATE	0x00		/* R W	PCM		Select sample rate timer			*/
+	#define S_C_C_SAMPLE_BUFFER	0x40		/* R W	PCM		Select sample buffer counter			*/
+
+	#define S_C_C_PC_SPEAKER	0x80		/* R W	PCM		Select PC speaker counter			*/
+
+#define SAMPLE_RATE_TIMER		0x1388		/*   W	PCM		Sample rate timer register (PCM wait interval)	*/
+#define SAMPLE_BUFFER_COUNTER		0x1389		/* R W	PCM		Sample buffer counter (DMA buffer size)		*/
+
+#define MIDI_CONTROL			0x178b		/* R W  MIDI		Midi control register				*/
+	#define M_C_ENA_TSTAMP_IRQ	0x01		/* R W	MIDI		Enable Time Stamp Interrupts			*/
+	#define M_C_ENA_TME_COMP_IRQ	0x02		/* R W  MIDI		Enable time compare interrupts			*/
+	#define M_C_ENA_INPUT_IRQ	0x04		/* R W  MIDI		Enable input FIFO interrupts			*/
+	#define M_C_ENA_OUTPUT_IRQ	0x08		/* R W  MIDI		Enable output FIFO interrupts			*/
+	#define M_C_ENA_OUTPUT_HALF_IRQ	0x10		/* R W  MIDI		Enable output FIFO half full interrupts		*/
+	#define M_C_RESET_INPUT_FIFO	0x20		/* R W  MIDI		Reset input FIFO pointer			*/
+	#define M_C_RESET_OUTPUT_FIFO	0x40		/* R W  MIDI		Reset output FIFO pointer			*/
+	#define M_C_ENA_THRU_MODE	0x80		/* R W  MIDI		Echo input to output (THRU)			*/
+
+#define MIDI_STATUS			0x1B88		/* R W  MIDI		Midi (interrupt) status register		*/
+	#define M_S_TIMESTAMP		0x01		/* R W  MIDI		Midi time stamp interrupt occurred		*/
+	#define M_S_COMPARE		0x02		/* R W  MIDI		Midi compare time interrupt occurred		*/
+	#define M_S_INPUT_AVAIL		0x04		/* R W  MIDI		Midi input data available interrupt occurred	*/
+	#define M_S_OUTPUT_EMPTY	0x08		/* R W  MIDI		Midi output FIFO empty interrupt occurred	*/
+	#define M_S_OUTPUT_HALF_EMPTY	0x10		/* R W  MIDI		Midi output FIFO half empty interrupt occurred	*/
+	#define M_S_INPUT_OVERRUN	0x20		/* R W  MIDI		Midi input overrun error occurred		*/
+	#define M_S_OUTPUT_OVERRUN	0x40		/* R W  MIDI 		Midi output overrun error occurred		*/
+	#define M_S_FRAMING_ERROR	0x80		/* R W  MIDI		Midi input framing error occurred		*/
+
+#define MIDI_FIFO_STATUS		0x1B89		/* R W  MIDI		Midi fifo status				*/
+#define MIDI_DATA			0x178A		/* R W  MIDI		Midi data register				*/
+#define MIDI_INPUT_AVAILABLE		0x0f		/* RW   MIDI */ 
diff --git a/sys/i386/isa/sound/pas_hw.h b/sys/i386/isa/sound/pas_hw.h
new file mode 100644
index 000000000000..5f823287fa90
--- /dev/null
+++ b/sys/i386/isa/sound/pas_hw.h
@@ -0,0 +1,236 @@
+/* 																*/
+/*	Port addresses and bit fields for the Media Vision Pro AudioSpectrum second generation sound cards.			*/
+/* 																*/
+/* 	Feel free to use this header file in any application you create that has support for the Media Vision			*/
+/*	Pro AudioSpectrum second generation sound cards. Other uses prohibited without prior permission.			*/
+/* 																*/
+/*										- cmetz@thor.tjhsst.edu				*/
+/* 																*/
+/*	Notes:															*/
+/* 																*/
+/*	*	All of these ports go into the MVD101 multimedia controller chip, which	then signals the other chips to do	*/
+/* 		the actual work. Many ports like the FM ones functionally attach directly to the destination chip though 	*/
+/* 		they don't actually have a direct connection.									*/
+/* 																*/
+/* 	*	The PAS2 series cards have an MVD101 multimedia controller chip, the original PAS cards don't. The original	*/
+/* 		PAS cards are pretty defunct now, so no attempt is made here to support them. 					*/
+/*																*/
+/*	*	The PAS2 series cards are all really different at the hardware level, though the MVD101 hides some of the 	*/
+/*		incompatibilities, there still are differences that need to be accounted for.					*/
+/*																*/
+/*		Card		CD-ROM interface	PCM chip		Mixer chip		FM chip			*/
+/* 		PAS Plus	Sony proprietary	(Crystal?) 8-bit DAC	National 		OPL3			*/
+/* 		PAS 16		Zilog SCSI		MVA416 16-bit Codec	MVA508 			OPL3			*/
+/* 		CDPC		Sony proprietary	Sony 16-bit Codec	National		OPL3			*/
+/*		Fusion CD 16	Sony proprietary	MVA416 16-bit Codec	MVA508			OPL3			*/
+/*		Fusion CD	Sony proprietary	(Crystal?) 8-bit DAC	National		OPL3			*/
+/* 																*/
+#define PAS_DEFAULT_BASE		0x388
+
+/*      Symbolic Name			Value 		   R W  Subsystem	Description					*/
+#define SPEAKER_CONTROL			0x61		/*   W	PC speaker 	Control register 				*/
+#define SPEAKER_CONTROL_GHOST		0x738B		/* R W	PC speaker 	Control ghost register				*/
+#define SPEAKER_TIMER_CONTROL		0x43		/*   W  PC speaker 	Timer control register				*/
+#define SPEAKER_TIMER_CONTROL_GHOST	0x778B		/* R W  PC speaker 	Timer control register ghost			*/
+#define SPEAKER_TIMER_DATA		0x42		/*   W  PC speaker 	Timer data register				*/
+#define SPEAKER_TIMER_DATA_GHOST	0x138A		/* R W  PC speaker	Timer data register ghost			*/
+
+#define WARM_BOOT			0x41		/*   W  Control		Used to detect system warm boot	  		*/
+#define WARM_BOOT_GHOST			0x7789		/* ? W  Control		Use to get the card to fake warm boot		*/
+#define MASTER_DECODE			0x9A01		/*   W  Control		Address >> 2 of card base address		*/
+#define PRESCALE_DIVIDER		0xBF8A		/* R W	PCM		Ration between Codec clock and master clock	*/
+#define WAIT_STATE			0xBF88		/* R W	Control		Four-bit bus wait-state count (~140ns ea.)	*/
+#define BOARD_REV_ID			0x2789		/* R	Control		Extended Board Revision ID			*/
+
+#define CHIP_REV			0xFF88		/* R	0=PAS, 1=PAS+, 2=CDPC, 3=PAS16C, 4=PAS16D */
+
+#define SYSTEM_CONFIGURATION_1		0x8388		/* R W	Control								*/
+#	define S_C_1_PCS_ENABLE	0x01		/* R W  PC speaker	1=enable, 0=disable PC speaker emulation	*/
+#	define S_C_1_PCM_CLOCK_SELECT	0x02		/* R W  PCM		1=14.31818Mhz/12, 0=28.224Mhz master clock	*/ 
+#	define S_C_1_FM_EMULATE_CLOCK	0x04		/* R W  FM		1=use 28.224Mhz/2, 0=use 14.31818Mhz clock	*/
+#	define S_C_1_PCS_STEREO	0x10 		/* R W  PC speaker	1=enable PC speaker stereo effect, 0=disable	*/
+#	define S_C_1_PCS_REALSOUND	0x20		/* R W  PC speaker	1=enable RealSound enhancement, 0=disable	*/
+#	define S_C_1_FORCE_EXT_RESET	0x40		/* R W  Control		Force external reset				*/
+#	define S_C_1_FORCE_INT_RESET	0x80		/* R W  Control		Force internal reset				*/
+#define SYSTEM_CONFIGURATION_2		0x8389		/* R W  Control								*/
+#	define S_C_2_PCM_OVERSAMPLING	0x03		/* R W  PCM		00=0x, 01=2x, 10=4x, 11=reserved		*/
+#	define S_C_2_PCM_16_BIT	0x04		/* R W	PCM		1=16-bit, 0=8-bit samples			*/
+#define SYSTEM_CONFIGURATION_3		0x838A		/* R W	Control								*/
+#	define S_C_3_PCM_CLOCK_SELECT	0x02		/* R W	PCM		1=use 1.008Mhz clock for PCM, 0=don't		*/
+#define SYSTEM_CONFIGURATION_4		0x838B		/* R W  Control		CD-ROM interface controls			*/
+
+#define IO_CONFIGURATION_1		0xF388		/* R W	Control								*/
+#	define I_C_1_BOOT_RESET_ENABLE	0x80		/* R W  Control		1=reset board on warm boot, 0=don't		*/
+#	define I_C_1_JOYSTICK_ENABLE	0x40		/* R W  Control		1=enable joystick port, 0=don't		        */
+#define IO_CONFIGURATION_2		0xF389		/* R W  Control								*/
+#	define	I_C_2_PCM_DMA_DISABLED	0x00		/* R W  PCM		PCM DMA disabled				*/
+#define IO_CONFIGURATION_3		0xF38A		/* R W	Control								*/
+#	define I_C_3_PCM_IRQ_DISABLED	0x00		/* R W  PCM		PCM IRQ disabled				*/
+
+#define COMPATIBILITY_ENABLE		0xF788		/* R W  Control								*/
+#	define C_E_MPU401_ENABLE	0x01		/* R W	MIDI		1=enable, 0=disable MPU401 MIDI emulation	*/
+#	define C_E_SB_ENABLE		0x02		/* R W  PCM		1=enable, 0=disable Sound Blaster emulation	*/
+#	define C_E_SB_ACTIVE		0x04		/* R    PCM		"Sound Blaster Interrupt active"		*/
+#	define C_E_MPU401_ACTIVE	0x08		/* R	MIDI		"MPU UART mode active"				*/
+#	define C_E_PCM_COMPRESSION	0x10		/* R W  PCM		1=enable, 0=disabled compression		*/
+#define EMULATION_ADDRESS		0xF789		/* R W  Control								*/
+#	define E_A_SB_BASE		0x0f		/* R W  PCM		bits A4-A7 for SB base port			*/
+#	define E_A_MPU401_BASE		0xf0		/* R W	MIDI		bits A4-A7 for MPU401 base port 		*/
+#define EMULATION_CONFIGURATION		0xFB8A		/* R W			***** Only valid on newer PAS2 cards (?) *****	*/
+#	define E_C_MPU401_IRQ		0x07		/* R W	MIDI		MPU401 emulation IRQ				*/
+#	define E_C_SB_IRQ		0x38		/* R W  PCM		SB emulation IRQ				*/
+#	define E_C_SB_DMA		0xC0		/* R W	PCM		SB emulation DMA				*/
+
+#define OPERATION_MODE_1		0xEF8B		/* R	Control								*/
+#	define	O_M_1_CDROM_TYPE	0x03		/* R	CD-ROM		3=SCSI, 2=Sony, 0=no CD-ROM interface		*/
+#	define O_M_1_FM_TYPE		0x04		/* R	FM		1=sterero, 0=mono FM chip			*/
+#	define O_M_1_PCM_TYPE 		0x08		/* R	PCM		1=16-bit Codec, 0=8-bit DAC			*/
+#define OPERATION_MODE_2		0xFF8B		/* R	Control								*/
+#	define O_M_2_PCS_ENABLED	0x02		/* R	PC speaker	PC speaker emulation 1=enabled, 0=disabled	*/
+#	define O_M_2_BUS_TIMING	0x10		/* R	Control		1=AT bus timing, 0=XT bus timing		*/
+#	define O_M_2_BOARD_REVISION	0xe0		/* R	Control		Board revision					*/
+
+#define INTERRUPT_MASK			0x0B8B		/* R W	Control								*/
+#	define I_M_FM_LEFT_IRQ_ENABLE	0x01		/* R W	FM		Enable FM left interrupt			*/
+#	define I_M_FM_RIGHT_IRQ_ENABLE	0x02		/* R W	FM		Enable FM right interrupt			*/
+#	define I_M_PCM_RATE_IRQ_ENABLE	0x04		/* R W	PCM		Enable Sample Rate interrupt			*/
+#	define I_M_PCM_BUFFER_IRQ_ENABLE 0x08		/* R W	PCM		Enable Sample Buffer interrupt			*/
+#	define I_M_MIDI_IRQ_ENABLE	0x10		/* R W	MIDI		Enable MIDI interrupt				*/
+#	define I_M_BOARD_REV		0xE0		/* R	Control		Board revision					*/
+
+#define INTERRUPT_STATUS		0x0B89		/* R W	Control								*/
+#	define I_S_FM_LEFT_IRQ		0x01		/* R W	FM		Left FM Interrupt Pending			*/
+#	define I_S_FM_RIGHT_IRQ	0x02		/* R W	FM		Right FM Interrupt Pending			*/
+#	define I_S_PCM_SAMPLE_RATE_IRQ	0x04		/* R W	PCM		Sample Rate Interrupt Pending			*/
+#	define I_S_PCM_SAMPLE_BUFFER_IRQ 0x08		/* R W	PCM		Sample Buffer Interrupt Pending			*/
+#	define I_S_MIDI_IRQ		0x10		/* R W	MIDI		MIDI Interrupt Pending				*/
+#	define I_S_PCM_CHANNEL		0x20		/* R W	PCM		1=right, 0=left					*/
+#	define I_S_RESET_ACTIVE	0x40		/* R W	Control		Reset is active (Timed pulse not finished)	*/
+#	define I_S_PCM_CLIPPING	0x80		/* R W	PCM		Clipping has occurred				*/
+
+#define FILTER_FREQUENCY		0x0B8A		/* R W	Control								*/
+#	define F_F_FILTER_DISABLED	0x00		/* R W 	Mixer		No filter					*/
+#	define F_F_MIXER_UNMUTE	0x20		/* R W	Mixer		1=disable, 0=enable board mute			*/
+#	define F_F_PCM_RATE_COUNTER	0x40		/* R W	PCM		1=enable, 0=disable sample rate counter		*/
+#	define F_F_PCM_BUFFER_COUNTER	0x80		/* R W 	PCM		1=enable, 0=disable sample buffer counter	*/
+
+#define PAS_NONE	0
+#define PAS_PLUS	1
+#define PAS_CDPC	2
+#define PAS_16		3
+#define PAS_16D		4
+
+#ifdef DEFINE_TRANSLATIONS
+	unsigned char I_C_2_PCM_DMA_translate[] = 		/* R W  PCM		PCM DMA channel value translations		*/
+			{ 4, 1, 2, 3, 0, 5, 6, 7 };
+	unsigned char I_C_3_PCM_IRQ_translate[] = 		/* R W	PCM		PCM IRQ level value translation			*/
+		{ 0,  0,  1,  2,  3,  4,  5,  6, 0,  1,  7,  8,  9,  0, 10, 11 };  
+	unsigned char E_C_MPU401_IRQ_translate[] = 		/* R W	MIDI		MPU401 emulation IRQ value translation		*/
+		{ 0x00, 0x00, 0x01, 0x02, 0x00, 0x03, 0x00, 0x04, 0x00, 0x01, 0x05, 0x06, 0x07 };
+	unsigned char E_C_SB_IRQ_translate[] = 			/* R W	PCM		SB emulation IRQ translate			*/
+		{ 0x00, 0x00, 0x08, 0x10, 0x00, 0x18, 0x00, 0x20, 0x00, 0x08, 0x28, 0x30, 0x38, 0, 0 };
+	unsigned char E_C_SB_DMA_translate[] = 			/* R W	PCM		SB emulation DMA translate			*/
+		{ 0x00, 0x40, 0x80, 0xC0, 0, 0, 0, 0 };
+	unsigned char O_M_1_to_card[] = 				/* R W	Control		Translate (OM1 & 0x0f) to card type		*/
+		{ 0, 0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 0, 4, 0, 2, 3 };   
+#else
+	extern unsigned char I_C_2_PCM_DMA_translate[];		/* R W  PCM		PCM DMA channel value translations		*/
+	extern unsigned char I_C_3_PCM_IRQ_translate[];		/* R W	PCM		PCM IRQ level value translation			*/
+	extern unsigned char E_C_MPU401_IRQ_translate[];		/* R W	MIDI		MPU401 emulation IRQ value translation		*/
+	extern unsigned char E_C_SB_IRQ_translate[];		/* R W	PCM		SB emulation IRQ translate			*/
+	extern unsigned char E_C_SB_DMA_translate[];		/* R W	PCM		SB emulation DMA translate			*/
+	extern unsigned char O_M_1_to_card[];			/* R W	Control		Translate (OM1 & 0x0f) to card type		*/
+#endif
+
+#define PARALLEL_MIXER			0x078B		/*   W	Mixer		Documented for MVD101 as FM Mono Right decode?? */
+#	define P_M_MV508_ADDRESS	0x80		/*   W	Mixer		MVD508	Address/mixer select			*/
+#	define P_M_MV508_DATA		0x00
+#	define P_M_MV508_LEFT		0x20		/*   W	Mixer		MVD508	Left channel select			*/
+#	define P_M_MV508_RIGHT		0x40		/*   W	Mixer		MVD508	Right channel select			*/
+#	define P_M_MV508_BOTH		0x00		/*   W	Mixer		MVD508	Both channel select			*/
+#	define P_M_MV508_MIXER		0x10		/*   W	Mixer		MVD508	Select a mixer (rather than a volume) 	*/
+#	define P_M_MV508_VOLUME	0x00
+
+#	define P_M_MV508_INPUTMIX	0x20		/*   W  Mixer		MVD508	Select mixer A				*/
+#	define P_M_MV508_OUTPUTMIX	0x00		/*   W  Mixer		MVD508	Select mixer B				*/
+
+#	define P_M_MV508_MASTER_A	0x01		/*   W	Mixer		MVD508	Master volume control A (output)	*/
+#	define P_M_MV508_MASTER_B	0x02		/*   W	Mixer		MVD508	Master volume control B (DSP input)	*/
+#	define P_M_MV508_BASS		0x03		/*   W	Mixer		MVD508	Bass control				*/
+#	define P_M_MV508_TREBLE	0x04		/*   W	Mixer		MVD508	Treble control				*/
+#	define P_M_MV508_MODE		0x05		/*   W	Mixer		MVD508	Master mode control			*/
+
+#	define P_M_MV508_LOUDNESS	0x04		/*   W	Mixer		MVD508	Mode control - Loudness filter 		*/
+#	define P_M_MV508_ENHANCE_BITS	0x03
+#	define P_M_MV508_ENHANCE_NONE	0x00		/*   W	Mixer		MVD508	Mode control - No stereo enhancement	*/
+#	define P_M_MV508_ENHANCE_40	0x01		/*   W	Mixer		MVD508	Mode control - 40% stereo enhancement	*/
+#	define P_M_MV508_ENHANCE_60	0x02		/*   W	Mixer		MVD508	Mode control - 60% stereo enhancement	*/
+#	define P_M_MV508_ENHANCE_80	0x03		/*   W	Mixer		MVD508	Mode control - 80% stereo enhancement	*/
+
+#	define P_M_MV508_FM		0x00		/*   W	Mixer		MVD508	Channel 0 - FM				*/
+#	define P_M_MV508_IMIXER	0x01		/*   W	Mixer		MVD508	Channel 1 - Input mixer (rec monitor)	*/
+#	define P_M_MV508_LINE		0x02		/*   W	Mixer		MVD508	Channel 2 - Line in			*/
+#	define P_M_MV508_CDROM		0x03		/*   W	Mixer		MVD508	Channel 3 - CD-ROM			*/
+#	define P_M_MV508_MIC		0x04		/*   W	Mixer		MVD508	Channel 4 - Microphone			*/
+#	define P_M_MV508_PCM		0x05		/*   W	Mixer		MVD508	Channel 5 - PCM				*/
+#	define P_M_MV508_SPEAKER	0x06		/*   W	Mixer		MVD508	Channel 6 - PC Speaker			*/
+#	define P_M_MV508_SB		0x07		/*   W	Mixer		MVD508	Channel 7 - SB DSP			*/
+
+#define SERIAL_MIXER			0xB88		/* R W	Control		Serial mixer control (used other ways)		*/
+#	define S_M_PCM_RESET		0x01		/* R W	PCM		Codec/DSP reset					*/
+#	define S_M_FM_RESET		0x02		/* R W	FM		FM chip reset					*/
+#	define S_M_SB_RESET		0x04		/* R W	PCM		SB emulation chip reset				*/
+#	define S_M_MIXER_RESET		0x10		/* R W	Mixer		Mixer chip reset				*/
+#	define S_M_INTEGRATOR_ENABLE	0x40		/* R W	Speaker		Enable PC speaker integrator (FORCE RealSound)	*/
+#	define S_M_OPL3_DUAL_MONO	0x80		/* R W  FM		Set the OPL-3 to dual mono mode			*/
+
+#define PCM_CONTROL			0xF8A		/* R W	PCM		PCM Control Register				*/
+#	define P_C_MIXER_CROSS_FIELD	0x0f
+#	define P_C_MIXER_CROSS_R_TO_R	0x01		/* R W	Mixer		Connect Right to Right				*/
+#	define P_C_MIXER_CROSS_L_TO_R	0x02		/* R W	Mixer		Connect Left  to Right				*/
+#	define P_C_MIXER_CROSS_R_TO_L	0x04		/* R W	Mixer		Connect Right to Left				*/
+#	define P_C_MIXER_CROSS_L_TO_L	0x08		/* R W	Mixer		Connect Left  to Left				*/
+#	define P_C_PCM_DAC_MODE	0x10		/* R W	PCM		Playback (DAC) mode				*/
+#	define P_C_PCM_ADC_MODE	0x00		/* R W	PCM		Record (ADC) mode				*/
+#	define P_C_PCM_MONO		0x20		/* R W	PCM		Mono mode					*/
+#	define P_C_PCM_STEREO		0x00		/* R W	PCM		Stereo mode					*/
+#	define P_C_PCM_ENABLE		0x40		/* R W	PCM		Enable PCM engine				*/
+#	define P_C_PCM_DMA_ENABLE	0x80		/* R W	PCM		Enable DRQ					*/
+
+#define SAMPLE_COUNTER_CONTROL		0x138B		/* R W	PCM		Sample counter control register			*/
+#	define S_C_C_SQUARE_WAVE	0x04		/* R W	PCM		Square wave generator (use for sample rate)	*/
+#	define S_C_C_RATE		0x06		/* R W	PCM		Rate generator (use for sample buffer count)	*/
+#	define S_C_C_LSB_THEN_MSB	0x30		/* R W	PCM		Change all 16 bits, LSB first, then MSB		*/
+
+	/* MVD101 and SDK documentations have S_C_C_SAMPLE_RATE and S_C_C_SAMPLE_BUFFER transposed. Only one works :-) */
+#	define S_C_C_SAMPLE_RATE	0x00		/* R W	PCM		Select sample rate timer			*/
+#	define S_C_C_SAMPLE_BUFFER	0x40		/* R W	PCM		Select sample buffer counter			*/
+
+#	define S_C_C_PC_SPEAKER	0x80		/* R W	PCM		Select PC speaker counter			*/
+
+#define SAMPLE_RATE_TIMER		0x1388		/*   W	PCM		Sample rate timer register (PCM wait interval)	*/
+#define SAMPLE_BUFFER_COUNTER		0x1389		/* R W	PCM		Sample buffer counter (DMA buffer size)		*/
+
+#define MIDI_CONTROL			0x178b		/* R W  MIDI		Midi control register				*/
+#	define M_C_ENA_TSTAMP_IRQ	0x01		/* R W	MIDI		Enable Time Stamp Interrupts			*/
+#	define M_C_ENA_TME_COMP_IRQ	0x02		/* R W  MIDI		Enable time compare interrupts			*/
+#	define M_C_ENA_INPUT_IRQ	0x04		/* R W  MIDI		Enable input FIFO interrupts			*/
+#	define M_C_ENA_OUTPUT_IRQ	0x08		/* R W  MIDI		Enable output FIFO interrupts			*/
+#	define M_C_ENA_OUTPUT_HALF_IRQ	0x10		/* R W  MIDI		Enable output FIFO half full interrupts		*/
+#	define M_C_RESET_INPUT_FIFO	0x20		/* R W  MIDI		Reset input FIFO pointer			*/
+#	define M_C_RESET_OUTPUT_FIFO	0x40		/* R W  MIDI		Reset output FIFO pointer			*/
+#	define M_C_ENA_THRU_MODE	0x80		/* R W  MIDI		Echo input to output (THRU)			*/
+
+#define MIDI_STATUS			0x1B88		/* R W  MIDI		Midi (interrupt) status register		*/
+#	define M_S_TIMESTAMP		0x01		/* R W  MIDI		Midi time stamp interrupt occurred		*/
+#	define M_S_COMPARE		0x02		/* R W  MIDI		Midi compare time interrupt occurred		*/
+#	define M_S_INPUT_AVAIL		0x04		/* R W  MIDI		Midi input data available interrupt occurred	*/
+#	define M_S_OUTPUT_EMPTY	0x08		/* R W  MIDI		Midi output FIFO empty interrupt occurred	*/
+#	define M_S_OUTPUT_HALF_EMPTY	0x10		/* R W  MIDI		Midi output FIFO half empty interrupt occurred	*/
+#	define M_S_INPUT_OVERRUN	0x20		/* R W  MIDI		Midi input overrun error occurred		*/
+#	define M_S_OUTPUT_OVERRUN	0x40		/* R W  MIDI 		Midi output overrun error occurred		*/
+#	define M_S_FRAMING_ERROR	0x80		/* R W  MIDI		Midi input framing error occurred		*/
+
+#define MIDI_FIFO_STATUS		0x1B89		/* R W  MIDI		Midi fifo status				*/
+#define MIDI_DATA			0x178A		/* R W  MIDI		Midi data register				*/
+#define MIDI_INPUT_AVAILABLE		0x0f		/* RW   MIDI */ 
diff --git a/sys/i386/isa/sound/pcm86.c b/sys/i386/isa/sound/pcm86.c
new file mode 100644
index 000000000000..aa7e3d77ef21
--- /dev/null
+++ b/sys/i386/isa/sound/pcm86.c
@@ -0,0 +1,2189 @@
+/*
+ * PC-9801-86 PCM driver for FreeBSD(98).
+ *
+ * Copyright (c) 1995  NAGAO Tadaaki (ABTK)
+ * All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ *    notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ *    notice, this list of conditions and the following disclaimer in the
+ *    documentation and/or other materials provided with the distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND
+ * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ * ARE DISCLAIMED.  IN NO EVENT SHALL THE AUTHOR AND CONTRIBUTORS BE LIABLE
+ * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+ * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+ * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ *
+ * $Id: pcm86.c,v 2.4 1996/01/24 19:53:34 abtk Exp $
+ */
+
+/*
+ * !! NOTE !! :
+ *   This file DOES NOT belong to the VoxWare distribution though it works
+ *   as part of the VoxWare drivers.  It is FreeBSD(98) original.
+ *   -- Nagao (nagao@cs.titech.ac.jp)
+ */
+
+
+#include <i386/isa/sound/sound_config.h>
+
+#ifdef CONFIGURE_SOUNDCARD
+
+#if !defined(EXCLUDE_PCM86) && !defined(EXCLUDE_AUDIO)
+
+
+/*
+ * Constants
+ */
+
+#define YES		1
+#define NO		0
+
+#define IMODE_NONE	0
+#define IMODE_INPUT	1
+#define IMODE_OUTPUT	2
+
+/* PC-9801-86 specific constants */
+#define	PCM86_IOBASE	0xa460	/* PCM I/O ports */
+#define PCM86_FIFOSIZE	32768	/* There is a 32kB FIFO buffer on 86-board */
+
+/* XXX -- These values should be chosen appropriately. */
+#define PCM86_INTRSIZE_OUT	1024
+#define PCM86_INTRSIZE_IN	(PCM86_FIFOSIZE / 2 - 128)
+#define DEFAULT_VOLUME		15	/* 0(min) - 15(max) */
+
+
+/*
+ * Switches for debugging and experiments
+ */
+
+/* #define PCM86_DEBUG */
+
+#ifdef PCM86_DEBUG
+# ifdef DEB
+#  undef DEB
+# endif
+# define DEB(x) x
+#endif
+
+
+/*
+ * Private variables and types
+ */
+
+typedef unsigned char pcm_data;
+
+enum board_type {
+    NO_SUPPORTED_BOARD = 0,
+    PC980186_FAMILY = 1,
+    PC980173_FAMILY = 2
+};
+
+static char *board_name[] = {
+    /* Each must be of the length less than 32 bytes. */
+    "No supported board",
+    "PC-9801-86 soundboard",
+    "PC-9801-73 soundboard"
+};
+
+/* Current status of the driver */
+static struct {
+    int			iobase;
+    int			irq;
+    enum board_type	board_type;
+    int			opened;
+    int			format;
+    int			bytes;
+    int			chipspeedno;
+    int			chipspeed;
+    int			speed;
+    int			stereo;
+    int			volume;
+    int			intr_busy;
+    int			intr_size;
+    int			intr_mode;
+    int			intr_last;
+    int			intr_trailer;
+    pcm_data *		pdma_buf;
+    int			pdma_count;
+    int			pdma_chunkcount;
+    int			acc;
+    int			last_l;
+    int			last_r;
+} pcm_s;
+
+static struct {
+    pcm_data		buff[4];
+    int			size;
+} tmpbuf;
+
+static int my_dev = 0;
+static char pcm_initialized = NO;
+
+/* 86-board supports only the following rates. */
+static int rates_tbl[8] = {
+#ifndef WAVEMASTER_FREQ
+    44100, 33075, 22050, 16538, 11025, 8269, 5513, 4134
+#else
+    /*
+     * It is said that Q-Vision's WaveMaster of some earlier lot(s?) has
+     * sampling rates incompatible with PC-9801-86.
+     * But I'm not sure whether the following rates are correct, especially
+     * 4000Hz.
+     */
+    44100, 33075, 22050, 16000, 11025, 8000, 5510, 4000
+#endif
+};
+
+/* u-law to linear translation table */
+static pcm_data ulaw2linear[256] = {
+    130, 134, 138, 142, 146, 150, 154, 158, 
+    162, 166, 170, 174, 178, 182, 186, 190, 
+    193, 195, 197, 199, 201, 203, 205, 207, 
+    209, 211, 213, 215, 217, 219, 221, 223, 
+    225, 226, 227, 228, 229, 230, 231, 232, 
+    233, 234, 235, 236, 237, 238, 239, 240, 
+    240, 241, 241, 242, 242, 243, 243, 244, 
+    244, 245, 245, 246, 246, 247, 247, 248, 
+    248, 248, 249, 249, 249, 249, 250, 250, 
+    250, 250, 251, 251, 251, 251, 252, 252, 
+    252, 252, 252, 252, 253, 253, 253, 253, 
+    253, 253, 253, 253, 254, 254, 254, 254, 
+    254, 254, 254, 254, 254, 254, 254, 254, 
+    255, 255, 255, 255, 255, 255, 255, 255, 
+    255, 255, 255, 255, 255, 255, 255, 255, 
+    255, 255, 255, 255, 255, 255, 255, 255, 
+    125, 121, 117, 113, 109, 105, 101,  97, 
+     93,  89,  85,  81,  77,  73,  69,  65, 
+     62,  60,  58,  56,  54,  52,  50,  48, 
+     46,  44,  42,  40,  38,  36,  34,  32, 
+     30,  29,  28,  27,  26,  25,  24,  23, 
+     22,  21,  20,  19,  18,  17,  16,  15, 
+     15,  14,  14,  13,  13,  12,  12,  11, 
+     11,  10,  10,   9,   9,   8,   8,   7, 
+      7,   7,   6,   6,   6,   6,   5,   5, 
+      5,   5,   4,   4,   4,   4,   3,   3, 
+      3,   3,   3,   3,   2,   2,   2,   2, 
+      2,   2,   2,   2,   1,   1,   1,   1, 
+      1,   1,   1,   1,   1,   1,   1,   1, 
+      0,   0,   0,   0,   0,   0,   0,   0, 
+      0,   0,   0,   0,   0,   0,   0,   0, 
+      0,   0,   0,   0,   0,   0,   0,   0
+};
+
+/* linear to u-law translation table */
+static pcm_data linear2ulaw[256] = {
+    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, 159, 158, 158, 157, 157, 156, 156, 
+    155, 155, 154, 154, 153, 153, 152, 152, 
+    151, 151, 150, 150, 149, 149, 148, 148, 
+    147, 147, 146, 146, 145, 145, 144, 144, 
+    143, 143, 143, 143, 142, 142, 142, 142, 
+    141, 141, 141, 141, 140, 140, 140, 140, 
+    139, 139, 139, 139, 138, 138, 138, 138, 
+    137, 137, 137, 137, 136, 136, 136, 136, 
+    135, 135, 135, 135, 134, 134, 134, 134, 
+    133, 133, 133, 133, 132, 132, 132, 132, 
+    131, 131, 131, 131, 130, 130, 130, 130, 
+    129, 129, 129, 129, 128, 128, 128, 128, 
+      0,   0,   0,   0,   0,   1,   1,   1, 
+      1,   2,   2,   2,   2,   3,   3,   3, 
+      3,   4,   4,   4,   4,   5,   5,   5, 
+      5,   6,   6,   6,   6,   7,   7,   7, 
+      7,   8,   8,   8,   8,   9,   9,   9, 
+      9,  10,  10,  10,  10,  11,  11,  11, 
+     11,  12,  12,  12,  12,  13,  13,  13, 
+     13,  14,  14,  14,  14,  15,  15,  15, 
+     15,  16,  16,  17,  17,  18,  18,  19, 
+     19,  20,  20,  21,  21,  22,  22,  23, 
+     23,  24,  24,  25,  25,  26,  26,  27, 
+     27,  28,  28,  29,  29,  30,  30,  31, 
+     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
+};
+
+
+/*
+ * Prototypes
+ */
+
+static int pcm86_detect(struct address_info *);
+
+static int pcm86_open(int, int);
+static void pcm86_close(int);
+static void pcm86_output_block(int, unsigned long, int, int, int);
+static void pcm86_start_input(int, unsigned long, int, int, int);
+static int pcm86_ioctl(int, unsigned int, unsigned int, int);
+static int pcm86_prepare_for_input(int, int, int);
+static int pcm86_prepare_for_output(int, int, int);
+static void pcm86_reset(int);
+static void pcm86_halt_xfer(int);
+
+static void dsp73_send_command(unsigned char);
+static void dsp73_send_data(unsigned char);
+static void dsp73_init(void);
+static int set_format(int);
+static int set_speed(int);
+static int set_stereo(int);
+static void set_volume(int);
+static void fifo_start(int);
+static void fifo_stop(void);
+static void fifo_reset(void);
+static void fifo_output_block(void);
+static int fifo_send(pcm_data *, int);
+static void fifo_sendtrailer(int);
+static void fifo_send_stereo(pcm_data *, int);
+static void fifo_send_monoral(pcm_data *, int);
+static void fifo_send_stereo_ulaw(pcm_data *, int);
+static void fifo_send_stereo_8(pcm_data *, int, int);
+static void fifo_send_stereo_16le(pcm_data *, int, int);
+static void fifo_send_stereo_16be(pcm_data *, int, int);
+static void fifo_send_mono_ulaw(pcm_data *, int);
+static void fifo_send_mono_8(pcm_data *, int, int);
+static void fifo_send_mono_16le(pcm_data *, int, int);
+static void fifo_send_mono_16be(pcm_data *, int, int);
+static void fifo_input_block(void);
+static void fifo_recv(pcm_data *, int);
+static void fifo_recv_stereo(pcm_data *, int);
+static void fifo_recv_monoral(pcm_data *, int);
+static void fifo_recv_stereo_ulaw(pcm_data *, int);
+static void fifo_recv_stereo_8(pcm_data *, int, int);
+static void fifo_recv_stereo_16le(pcm_data *, int, int);
+static void fifo_recv_stereo_16be(pcm_data *, int, int);
+static void fifo_recv_mono_ulaw(pcm_data *, int);
+static void fifo_recv_mono_8(pcm_data *, int, int);
+static void fifo_recv_mono_16le(pcm_data *, int, int);
+static void fifo_recv_mono_16be(pcm_data *, int, int);
+static void pcm_stop(void);
+static void pcm_init(void);
+
+
+/*
+ * Identity
+ */
+
+static struct audio_operations pcm86_operations =
+{
+    "PC-9801-86 SoundBoard", /* filled in properly by auto configuration */
+    NOTHING_SPECIAL,
+    ( AFMT_MU_LAW |
+      AFMT_U8 | AFMT_S16_LE | AFMT_S16_BE |
+      AFMT_S8 | AFMT_U16_LE | AFMT_U16_BE ),
+    NULL,
+    pcm86_open,
+    pcm86_close,
+    pcm86_output_block,
+    pcm86_start_input,
+    pcm86_ioctl,
+    pcm86_prepare_for_input,
+    pcm86_prepare_for_output,
+    pcm86_reset,
+    pcm86_halt_xfer,
+    NULL,
+    NULL
+};
+
+
+/*
+ * Codes for internal use
+ */
+
+static void
+dsp73_send_command(unsigned char command)
+{
+    /* wait for RDY */
+    while ((inb(pcm_s.iobase + 2) & 0x48) != 8);
+
+    /* command mode */
+    outb(pcm_s.iobase + 2, (inb(pcm_s.iobase + 2) & 0x20) | 3);
+
+    /* wait for RDY */
+    while ((inb(pcm_s.iobase + 2) & 0x48) != 8);
+
+    /* send command */
+    outb(pcm_s.iobase + 4, command);
+}
+
+
+static void
+dsp73_send_data(unsigned char data)
+{
+    /* wait for RDY */
+    while ((inb(pcm_s.iobase + 2) & 0x48) != 8);
+
+    /* data mode */
+    outb(pcm_s.iobase + 2, (inb(pcm_s.iobase + 2) & 0x20) | 0x83);
+
+    /* wait for RDY */
+    while ((inb(pcm_s.iobase + 2) & 0x48) != 8);
+
+    /* send command */
+    outb(pcm_s.iobase + 4, data);
+}
+
+
+static void
+dsp73_init(void)
+{
+    const unsigned char dspinst[15] = {
+	0x00, 0x00, 0x27,
+	0x3f, 0xe0, 0x01,
+	0x00, 0x00, 0x27,
+	0x36, 0x5a, 0x0d,
+	0x3e, 0x60, 0x04
+    };
+    unsigned char t;
+    int i;
+
+    /* reset DSP */
+    t = inb(pcm_s.iobase + 2);
+    outb(pcm_s.iobase + 2, (t & 0x80) | 0x23);
+
+    /* mute on */
+    dsp73_send_command(0x04);
+    dsp73_send_data(0x6f);
+    dsp73_send_data(0x3c);
+
+    /* write DSP instructions */
+    dsp73_send_command(0x01);
+    dsp73_send_data(0x00);
+    for (i = 0; i < 16; i++)
+	dsp73_send_data(dspinst[i]);
+
+    /* mute off */
+    dsp73_send_command(0x04);
+    dsp73_send_data(0x6f);
+    dsp73_send_data(0x30);
+
+    /* wait for RDY */
+    while ((inb(pcm_s.iobase + 2) & 0x48) != 8);
+
+    outb(pcm_s.iobase + 2, 3);
+}
+
+
+static int
+set_format(int format)
+{
+    switch (format) {
+    case AFMT_MU_LAW:
+    case AFMT_S8:
+    case AFMT_U8:
+	pcm_s.format = format;
+	pcm_s.bytes = 1;	/* 8bit */
+	break;
+    case AFMT_S16_LE:
+    case AFMT_U16_LE:
+    case AFMT_S16_BE:
+    case AFMT_U16_BE:
+	pcm_s.format = format;
+	pcm_s.bytes = 2;	/* 16bit */
+	break;
+    case AFMT_QUERY:
+	break;
+    default:
+	return -1;
+    }
+
+    return pcm_s.format;
+}
+
+
+static int
+set_speed(int speed)
+{
+    int i;
+
+    if (speed < 4000)	/* Minimum 4000Hz */
+	speed = 4000;
+    if (speed > 44100)	/* Maximum 44100Hz */
+	speed = 44100;
+    for (i = 7; i >= 0; i--) {
+	if (speed <= rates_tbl[i]) {
+	    pcm_s.chipspeedno = i;
+	    pcm_s.chipspeed = rates_tbl[i];
+	    break;
+	}
+    }
+    pcm_s.speed = speed;
+
+    return speed;
+}
+
+
+static int
+set_stereo(int stereo)
+{
+    pcm_s.stereo = stereo ? YES : NO;
+
+    return pcm_s.stereo;
+}
+
+
+static void
+set_volume(int volume)
+{
+    if (volume < 0)
+	volume = 0;
+    if (volume > 15)
+	volume = 15;
+    pcm_s.volume = volume;
+
+    outb(pcm_s.iobase + 6, 0xaf - volume);	/* D/A -> LINE OUT */
+    outb(0x5f,0);
+    outb(0x5f,0);
+    outb(0x5f,0);
+    outb(0x5f,0);
+    outb(pcm_s.iobase + 6, 0x20);		/* FM -> A/D */
+    outb(0x5f,0);
+    outb(0x5f,0);
+    outb(0x5f,0);
+    outb(0x5f,0);
+    outb(pcm_s.iobase + 6, 0x60);		/* LINE IN -> A/D */
+    outb(0x5f,0);
+    outb(0x5f,0);
+    outb(0x5f,0);
+    outb(0x5f,0);
+}
+
+
+static void
+fifo_start(int mode)
+{
+    unsigned char tmp;
+
+    /* Set frame length & panpot(LR). */
+    tmp = inb(pcm_s.iobase + 10) & 0x88;
+    outb(pcm_s.iobase + 10, tmp | ((pcm_s.bytes == 1) ? 0x72 : 0x32));
+
+    tmp = pcm_s.chipspeedno;
+    if (mode == IMODE_INPUT)
+	tmp |= 0x40;
+
+    /* Reset intr. flag. */
+    outb(pcm_s.iobase + 8, tmp);
+    outb(pcm_s.iobase + 8, tmp | 0x10);
+
+    /* Enable FIFO intr. */
+    outb(pcm_s.iobase + 8, tmp | 0x30);
+
+    /* Set intr. interval. */
+    outb(pcm_s.iobase + 10, pcm_s.intr_size / 128 - 1);
+
+    /* Start intr. */
+    outb(pcm_s.iobase + 8, tmp | 0xb0);
+}
+
+
+static void
+fifo_stop(void)
+{
+    unsigned char tmp;
+
+    /* Reset intr. flag, and disable FIFO intr. */
+    tmp = inb(pcm_s.iobase + 8) & 0x0f;
+    outb(pcm_s.iobase + 8, tmp);
+}
+
+
+static void
+fifo_reset(void)
+{
+    unsigned char tmp;
+
+    /* Reset FIFO. */
+    tmp = inb(pcm_s.iobase + 8) & 0x77;
+    outb(pcm_s.iobase + 8, tmp | 0x8);
+    outb(pcm_s.iobase + 8, tmp);
+}
+
+
+static void
+fifo_output_block(void)
+{
+    int chunksize, count;
+
+    if (pcm_s.pdma_chunkcount) {
+	/* Update chunksize and then send the next chunk to FIFO. */
+	chunksize = pcm_s.pdma_count / pcm_s.pdma_chunkcount--;
+	count = fifo_send(pcm_s.pdma_buf, chunksize);
+    } else {
+	/* ??? something wrong... */
+	printk("pcm0: chunkcount overrun\n");
+	chunksize = count = 0;
+    }
+
+    if (((audio_devs[my_dev]->dmap->qlen < 2) && (pcm_s.pdma_chunkcount == 0))
+	|| (count < pcm_s.intr_size)) {
+	/* The sent chunk seems to be the last one. */
+	fifo_sendtrailer(pcm_s.intr_size);
+	pcm_s.intr_last = YES;
+    }
+
+    pcm_s.pdma_buf += chunksize;
+    pcm_s.pdma_count -= chunksize;
+}
+
+
+static int
+fifo_send(pcm_data *buf, int count)
+{
+    int i, length, r, cnt, rslt;
+    pcm_data *p;
+
+    /* Calculate the length of PCM frames. */
+    cnt = count + tmpbuf.size;
+    length = pcm_s.bytes << pcm_s.stereo;
+    r = cnt % length;
+    cnt -= r;
+
+    if (cnt > 0) {
+	if (pcm_s.stereo)
+	    fifo_send_stereo(buf, cnt);
+	else
+	    fifo_send_monoral(buf, cnt);
+	/* Carry over extra data which doesn't seem to be a full PCM frame. */
+	p = (pcm_data *)buf + count - r;
+	for (i = 0; i < r; i++)
+	    tmpbuf.buff[i] = *p++;
+    } else {
+	/* Carry over extra data which doesn't seem to be a full PCM frame. */
+	p = (pcm_data *)buf;
+	for (i = tmpbuf.size; i < r; i++)
+	    tmpbuf.buff[i] = *p++;
+    }
+    tmpbuf.size = r;
+
+    rslt = ((cnt / length) * pcm_s.chipspeed / pcm_s.speed) * pcm_s.bytes * 2;
+#ifdef PCM86_DEBUG
+    printk("fifo_send(): %d bytes sent\n", rslt);
+#endif
+    return rslt;
+}
+
+
+static void
+fifo_sendtrailer(int count)
+{
+    /* Send trailing zeros to the FIFO buffer. */
+    int i;
+
+    for (i = 0; i < count; i++)
+	outb(pcm_s.iobase + 12, 0);
+    pcm_s.intr_trailer = YES;
+
+#ifdef PCM86_DEBUG
+    printk("fifo_sendtrailer(): %d bytes sent\n", count);
+#endif
+}
+
+
+static void
+fifo_send_stereo(pcm_data *buf, int count)
+{
+    /* Convert format and sampling speed. */
+    switch (pcm_s.format) {
+    case AFMT_MU_LAW:
+	fifo_send_stereo_ulaw(buf, count);
+	break;
+    case AFMT_S8:
+	fifo_send_stereo_8(buf, count, NO);
+	break;
+    case AFMT_U8:
+	fifo_send_stereo_8(buf, count, YES);
+	break;
+    case AFMT_S16_LE:
+	fifo_send_stereo_16le(buf, count, NO);
+	break;
+    case AFMT_U16_LE:
+	fifo_send_stereo_16le(buf, count, YES);
+	break;
+    case AFMT_S16_BE:
+	fifo_send_stereo_16be(buf, count, NO);
+	break;
+    case AFMT_U16_BE:
+	fifo_send_stereo_16be(buf, count, YES);
+	break;
+    }
+}
+
+
+static void
+fifo_send_monoral(pcm_data *buf, int count)
+{
+    /* Convert format and sampling speed. */
+    switch (pcm_s.format) {
+    case AFMT_MU_LAW:
+	fifo_send_mono_ulaw(buf, count);
+	break;
+    case AFMT_S8:
+	fifo_send_mono_8(buf, count, NO);
+	break;
+    case AFMT_U8:
+	fifo_send_mono_8(buf, count, YES);
+	break;
+    case AFMT_S16_LE:
+	fifo_send_mono_16le(buf, count, NO);
+	break;
+    case AFMT_U16_LE:
+	fifo_send_mono_16le(buf, count, YES);
+	break;
+    case AFMT_S16_BE:
+	fifo_send_mono_16be(buf, count, NO);
+	break;
+    case AFMT_U16_BE:
+	fifo_send_mono_16be(buf, count, YES);
+	break;
+    }
+}
+
+
+static void
+fifo_send_stereo_ulaw(pcm_data *buf, int count)
+{
+    int i;
+    signed char dl, dl0, dl1, dr, dr0, dr1;
+    pcm_data t[2];
+
+    if (tmpbuf.size > 0)
+	t[0] = ulaw2linear[tmpbuf.buff[0]];
+    else
+	t[0] = ulaw2linear[*buf++];
+    t[1] = ulaw2linear[*buf++];
+
+    if (pcm_s.speed == pcm_s.chipspeed) {
+	/* No reason to convert the pcm speed. */
+	outb(pcm_s.iobase + 12, t[0]);
+	outb(pcm_s.iobase + 12, t[1]);
+	count -= 2;
+	for (i = 0; i < count; i++)
+	    outb(pcm_s.iobase + 12, ulaw2linear[*buf++]);
+    } else {
+	/* Speed conversion with linear interpolation method. */
+	dl0 = pcm_s.last_l;
+	dr0 = pcm_s.last_r;
+	dl1 = t[0];
+	dr1 = t[1];
+	i = 0;
+	count /= 2;
+	while (i < count) {
+	    while (pcm_s.acc >= pcm_s.chipspeed) {
+		pcm_s.acc -= pcm_s.chipspeed;
+		i++;
+		dl0 = dl1;
+		dr0 = dr1;
+		if (i < count) {
+		    dl1 = ulaw2linear[*buf++];
+		    dr1 = ulaw2linear[*buf++];
+		} else
+		    dl1 = dr1 = 0;
+	    }
+	    dl = ((dl0 * (pcm_s.chipspeed - pcm_s.acc)) + (dl1 * pcm_s.acc))
+		/ pcm_s.chipspeed;
+	    dr = ((dr0 * (pcm_s.chipspeed - pcm_s.acc)) + (dr1 * pcm_s.acc))
+		/ pcm_s.chipspeed;
+	    outb(pcm_s.iobase + 12, dl);
+	    outb(pcm_s.iobase + 12, dr);
+	    pcm_s.acc += pcm_s.speed;
+	}
+
+	pcm_s.last_l = dl0;
+	pcm_s.last_r = dr0;
+    }
+}
+
+
+static void
+fifo_send_stereo_8(pcm_data *buf, int count, int uflag)
+{
+    int i;
+    signed char dl, dl0, dl1, dr, dr0, dr1, zlev;
+    pcm_data t[2];
+
+    zlev = uflag ? -128 : 0;
+
+    if (tmpbuf.size > 0)
+	t[0] = tmpbuf.buff[0] + zlev;
+    else
+	t[0] = *buf++ + zlev;
+    t[1] = *buf++ + zlev;
+
+    if (pcm_s.speed == pcm_s.chipspeed) {
+	/* No reason to convert the pcm speed. */
+	outb(pcm_s.iobase + 12, t[0]);
+	outb(pcm_s.iobase + 12, t[1]);
+	count -= 2;
+	for (i = 0; i < count; i++)
+	    outb(pcm_s.iobase + 12, *buf++ + zlev);
+    } else {
+	/* Speed conversion with linear interpolation method. */
+	dl0 = pcm_s.last_l;
+	dr0 = pcm_s.last_r;
+	dl1 = t[0];
+	dr1 = t[1];
+	i = 0;
+	count /= 2;
+	while (i < count) {
+	    while (pcm_s.acc >= pcm_s.chipspeed) {
+		pcm_s.acc -= pcm_s.chipspeed;
+		i++;
+		dl0 = dl1;
+		dr0 = dr1;
+		if (i < count) {
+		    dl1 = *buf++ + zlev;
+		    dr1 = *buf++ + zlev;
+		} else
+		    dl1 = dr1 = 0;
+	    }
+	    dl = ((dl0 * (pcm_s.chipspeed - pcm_s.acc)) + (dl1 * pcm_s.acc))
+		/ pcm_s.chipspeed;
+	    dr = ((dr0 * (pcm_s.chipspeed - pcm_s.acc)) + (dr1 * pcm_s.acc))
+		/ pcm_s.chipspeed;
+	    outb(pcm_s.iobase + 12, dl);
+	    outb(pcm_s.iobase + 12, dr);
+	    pcm_s.acc += pcm_s.speed;
+	}
+
+	pcm_s.last_l = dl0;
+	pcm_s.last_r = dr0;
+    }
+}
+
+
+static void
+fifo_send_stereo_16le(pcm_data *buf, int count, int uflag)
+{
+    int i;
+    short dl, dl0, dl1, dr, dr0, dr1, zlev;
+    pcm_data t[4];
+
+    zlev = uflag ? -128 : 0;
+
+    for (i = 0; i < 4; i++)
+	t[i] = (tmpbuf.size > i) ? tmpbuf.buff[i] : *buf++;
+
+    if (pcm_s.speed == pcm_s.chipspeed) {
+	/* No reason to convert the pcm speed. */
+	outb(pcm_s.iobase + 12, t[1] + zlev);
+	outb(pcm_s.iobase + 12, t[0]);
+	outb(pcm_s.iobase + 12, t[3] + zlev);
+	outb(pcm_s.iobase + 12, t[2]);
+	count = count / 2 - 2;
+	for (i = 0; i < count; i++) {
+	    outb(pcm_s.iobase + 12, *(buf + 1) + zlev);
+	    outb(pcm_s.iobase + 12, *buf);
+	    buf += 2;
+	}
+    } else {
+	/* Speed conversion with linear interpolation method. */
+	dl0 = pcm_s.last_l;
+	dr0 = pcm_s.last_r;
+	dl1 = t[0] + ((t[1] + zlev) << 8);
+	dr1 = t[2] + ((t[3] + zlev) << 8);
+	i = 0;
+	count /= 4;
+	while (i < count) {
+	    while (pcm_s.acc >= pcm_s.chipspeed) {
+		pcm_s.acc -= pcm_s.chipspeed;
+		i++;
+		dl0 = dl1;
+		dr0 = dr1;
+		if (i < count) {
+		    dl1 = *buf + ((*(buf + 1) + zlev) << 8);
+		    buf += 2;
+		    dr1 = *buf + ((*(buf + 1) + zlev) << 8);
+		    buf += 2;
+		} else
+		    dl1 = dr1 = 0;
+	    }
+	    dl = ((dl0 * (pcm_s.chipspeed - pcm_s.acc)) + (dl1 * pcm_s.acc))
+		/ pcm_s.chipspeed;
+	    dr = ((dr0 * (pcm_s.chipspeed - pcm_s.acc)) + (dr1 * pcm_s.acc))
+		/ pcm_s.chipspeed;
+	    outb(pcm_s.iobase + 12, (dl >> 8) & 0xff);
+	    outb(pcm_s.iobase + 12, dl & 0xff);
+	    outb(pcm_s.iobase + 12, (dr >> 8) & 0xff);
+	    outb(pcm_s.iobase + 12, dr & 0xff);
+	    pcm_s.acc += pcm_s.speed;
+	}
+
+	pcm_s.last_l = dl0;
+	pcm_s.last_r = dr0;
+    }
+}
+
+
+static void
+fifo_send_stereo_16be(pcm_data *buf, int count, int uflag)
+{
+    int i;
+    short dl, dl0, dl1, dr, dr0, dr1, zlev;
+    pcm_data t[4];
+
+    zlev = uflag ? -128 : 0;
+
+    for (i = 0; i < 4; i++)
+	t[i] = (tmpbuf.size > i) ? tmpbuf.buff[i] : *buf++;
+
+    if (pcm_s.speed == pcm_s.chipspeed) {
+	/* No reason to convert the pcm speed. */
+	outb(pcm_s.iobase + 12, t[0] + zlev);
+	outb(pcm_s.iobase + 12, t[1]);
+	outb(pcm_s.iobase + 12, t[2] + zlev);
+	outb(pcm_s.iobase + 12, t[3]);
+	count = count / 2 - 2;
+	for (i = 0; i < count; i++) {
+	    outb(pcm_s.iobase + 12, *buf + zlev);
+	    outb(pcm_s.iobase + 12, *(buf + 1));
+	    buf += 2;
+	}
+    } else {
+	/* Speed conversion with linear interpolation method. */
+	dl0 = pcm_s.last_l;
+	dr0 = pcm_s.last_r;
+	dl1 = ((t[0] + zlev) << 8) + t[1];
+	dr1 = ((t[2] + zlev) << 8) + t[3];
+	i = 0;
+	count /= 4;
+	while (i < count) {
+	    while (pcm_s.acc >= pcm_s.chipspeed) {
+		pcm_s.acc -= pcm_s.chipspeed;
+		i++;
+		dl0 = dl1;
+		dr0 = dr1;
+		if (i < count) {
+		    dl1 = ((*buf + zlev) << 8) + *(buf + 1);
+		    buf += 2;
+		    dr1 = ((*buf + zlev) << 8) + *(buf + 1);
+		    buf += 2;
+		} else
+		    dl1 = dr1 = 0;
+	    }
+	    dl = ((dl0 * (pcm_s.chipspeed - pcm_s.acc)) + (dl1 * pcm_s.acc))
+		/ pcm_s.chipspeed;
+	    dr = ((dr0 * (pcm_s.chipspeed - pcm_s.acc)) + (dr1 * pcm_s.acc))
+		/ pcm_s.chipspeed;
+	    outb(pcm_s.iobase + 12, (dl >> 8) & 0xff);
+	    outb(pcm_s.iobase + 12, dl & 0xff);
+	    outb(pcm_s.iobase + 12, (dr >> 8) & 0xff);
+	    outb(pcm_s.iobase + 12, dr & 0xff);
+	    pcm_s.acc += pcm_s.speed;
+	}
+
+	pcm_s.last_l = dl0;
+	pcm_s.last_r = dr0;
+    }
+}
+
+
+static void
+fifo_send_mono_ulaw(pcm_data *buf, int count)
+{
+    int i;
+    signed char d, d0, d1;
+
+    if (pcm_s.speed == pcm_s.chipspeed)
+	/* No reason to convert the pcm speed. */
+	for (i = 0; i < count; i++) {
+	    d = ulaw2linear[*buf++];
+	    outb(pcm_s.iobase + 12, d);
+	    outb(pcm_s.iobase + 12, d);
+	}
+    else {
+	/* Speed conversion with linear interpolation method. */
+	d0 = pcm_s.last_l;
+	d1 = ulaw2linear[*buf++];
+	i = 0;
+	while (i < count) {
+	    while (pcm_s.acc >= pcm_s.chipspeed) {
+		pcm_s.acc -= pcm_s.chipspeed;
+		i++;
+		d0 = d1;
+		d1 = (i < count) ? ulaw2linear[*buf++] : 0;
+	    }
+	    d = ((d0 * (pcm_s.chipspeed - pcm_s.acc)) + (d1 * pcm_s.acc))
+		/ pcm_s.chipspeed;
+	    outb(pcm_s.iobase + 12, d);
+	    outb(pcm_s.iobase + 12, d);
+	    pcm_s.acc += pcm_s.speed;
+	}
+
+	pcm_s.last_l = d0;
+    }
+}
+
+
+static void
+fifo_send_mono_8(pcm_data *buf, int count, int uflag)
+{
+    int i;
+    signed char d, d0, d1, zlev;
+
+    zlev = uflag ? -128 : 0;
+
+    if (pcm_s.speed == pcm_s.chipspeed)
+	/* No reason to convert the pcm speed. */
+	for (i = 0; i < count; i++) {
+	    d = *buf++ + zlev;
+	    outb(pcm_s.iobase + 12, d);
+	    outb(pcm_s.iobase + 12, d);
+	}
+    else {
+	/* Speed conversion with linear interpolation method. */
+	d0 = pcm_s.last_l;
+	d1 = *buf++ + zlev;
+	i = 0;
+	while (i < count) {
+	    while (pcm_s.acc >= pcm_s.chipspeed) {
+		pcm_s.acc -= pcm_s.chipspeed;
+		i++;
+		d0 = d1;
+		d1 = (i < count) ? *buf++ + zlev : 0;
+	    }
+	    d = ((d0 * (pcm_s.chipspeed - pcm_s.acc)) + (d1 * pcm_s.acc))
+		/ pcm_s.chipspeed;
+	    outb(pcm_s.iobase + 12, d);
+	    outb(pcm_s.iobase + 12, d);
+	    pcm_s.acc += pcm_s.speed;
+	}
+
+	pcm_s.last_l = d0;
+    }
+}
+
+
+static void
+fifo_send_mono_16le(pcm_data *buf, int count, int uflag)
+{
+    int i;
+    short d, d0, d1, zlev;
+    pcm_data t[2];
+
+    zlev = uflag ? -128 : 0;
+
+    for (i = 0; i < 2; i++)
+	t[i] = (tmpbuf.size > i) ? tmpbuf.buff[i] : *buf++;
+
+    if (pcm_s.speed == pcm_s.chipspeed) {
+	/* No reason to convert the pcm speed. */
+	outb(pcm_s.iobase + 12, t[1] + zlev);
+	outb(pcm_s.iobase + 12, t[0]);
+	outb(pcm_s.iobase + 12, t[1] + zlev);
+	outb(pcm_s.iobase + 12, t[0]);
+	count = count / 2 - 1;
+	for (i = 0; i < count; i++) {
+	    outb(pcm_s.iobase + 12, *(buf + 1) + zlev);
+	    outb(pcm_s.iobase + 12, *buf);
+	    outb(pcm_s.iobase + 12, *(buf + 1) + zlev);
+	    outb(pcm_s.iobase + 12, *buf);
+	    buf += 2;
+	}
+    } else {
+	/* Speed conversion with linear interpolation method. */
+	d0 = pcm_s.last_l;
+	d1 = t[0] + ((t[1] + zlev) << 8);
+	i = 0;
+	count /= 2;
+	while (i < count) {
+	    while (pcm_s.acc >= pcm_s.chipspeed) {
+		pcm_s.acc -= pcm_s.chipspeed;
+		i++;
+		d0 = d1;
+		if (i < count) {
+		    d1 = *buf + ((*(buf + 1) + zlev) << 8);
+		    buf += 2;
+		} else
+		    d1 = 0;
+	    }
+	    d = ((d0 * (pcm_s.chipspeed - pcm_s.acc)) + (d1 * pcm_s.acc))
+		/ pcm_s.chipspeed;
+	    outb(pcm_s.iobase + 12, (d >> 8) & 0xff);
+	    outb(pcm_s.iobase + 12, d & 0xff);
+	    outb(pcm_s.iobase + 12, (d >> 8) & 0xff);
+	    outb(pcm_s.iobase + 12, d & 0xff);
+	    pcm_s.acc += pcm_s.speed;
+	}
+
+	pcm_s.last_l = d0;
+    }
+}
+
+
+static void
+fifo_send_mono_16be(pcm_data *buf, int count, int uflag)
+{
+    int i;
+    short d, d0, d1, zlev;
+    pcm_data t[2];
+
+    zlev = uflag ? -128 : 0;
+
+    for (i = 0; i < 2; i++)
+	t[i] = (tmpbuf.size > i) ? tmpbuf.buff[i] : *buf++;
+
+    if (pcm_s.speed == pcm_s.chipspeed) {
+	/* No reason to convert the pcm speed. */
+	outb(pcm_s.iobase + 12, t[0] + zlev);
+	outb(pcm_s.iobase + 12, t[1]);
+	outb(pcm_s.iobase + 12, t[0] + zlev);
+	outb(pcm_s.iobase + 12, t[1]);
+	count = count / 2 - 1;
+	for (i = 0; i < count; i++) {
+	    outb(pcm_s.iobase + 12, *buf + zlev);
+	    outb(pcm_s.iobase + 12, *(buf + 1));
+	    outb(pcm_s.iobase + 12, *buf + zlev);
+	    outb(pcm_s.iobase + 12, *(buf + 1));
+	    buf += 2;
+	}
+    } else {
+	/* Speed conversion with linear interpolation method. */
+	d0 = pcm_s.last_l;
+	d1 = ((t[0] + zlev) << 8) + t[1];
+	i = 0;
+	count /= 2;
+	while (i < count) {
+	    while (pcm_s.acc >= pcm_s.chipspeed) {
+		pcm_s.acc -= pcm_s.chipspeed;
+		i++;
+		d0 = d1;
+		if (i < count) {
+		    d1 = ((*buf + zlev) << 8) + *(buf + 1);
+		    buf += 2;
+		} else
+		    d1 = 0;
+	    }
+	    d = ((d0 * (pcm_s.chipspeed - pcm_s.acc)) + (d1 * pcm_s.acc))
+		/ pcm_s.chipspeed;
+	    outb(pcm_s.iobase + 12, d & 0xff);
+	    outb(pcm_s.iobase + 12, (d >> 8) & 0xff);
+	    outb(pcm_s.iobase + 12, d & 0xff);
+	    outb(pcm_s.iobase + 12, (d >> 8) & 0xff);
+	    pcm_s.acc += pcm_s.speed;
+	}
+
+	pcm_s.last_l = d0;
+    }
+}
+
+
+static void
+fifo_input_block(void)
+{
+    int chunksize;
+
+    if (pcm_s.pdma_chunkcount) {
+	/* Update chunksize and then receive the next chunk from FIFO. */
+	chunksize = pcm_s.pdma_count / pcm_s.pdma_chunkcount--;
+	fifo_recv(pcm_s.pdma_buf, chunksize);
+	pcm_s.pdma_buf += chunksize;
+	pcm_s.pdma_count -= chunksize;
+    } else
+	/* ??? something wrong... */
+	printk("pcm0: chunkcount overrun\n");
+}
+
+
+static void
+fifo_recv(pcm_data *buf, int count)
+{
+    int i;
+
+    if (count > tmpbuf.size) {
+	for (i = 0; i < tmpbuf.size; i++)
+	    *buf++ = tmpbuf.buff[i];
+	count -= tmpbuf.size;
+	tmpbuf.size = 0;
+	if (pcm_s.stereo)
+	    fifo_recv_stereo(buf, count);
+	else
+	    fifo_recv_monoral(buf, count);
+    } else {
+	for (i = 0; i < count; i++)
+	    *buf++ = tmpbuf.buff[i];
+	for (i = 0; i < tmpbuf.size - count; i++)
+	    tmpbuf.buff[i] = tmpbuf.buff[i + count];
+	tmpbuf.size -= count;
+    }
+
+#ifdef PCM86_DEBUG
+    printk("fifo_recv(): %d bytes received\n",
+	   ((count / (pcm_s.bytes << pcm_s.stereo)) * pcm_s.chipspeed
+	    / pcm_s.speed) * pcm_s.bytes * 2);
+#endif
+}
+
+
+static void
+fifo_recv_stereo(pcm_data *buf, int count)
+{
+    /* Convert format and sampling speed. */
+    switch (pcm_s.format) {
+    case AFMT_MU_LAW:
+	fifo_recv_stereo_ulaw(buf, count);
+	break;
+    case AFMT_S8:
+	fifo_recv_stereo_8(buf, count, NO);
+	break;
+    case AFMT_U8:
+	fifo_recv_stereo_8(buf, count, YES);
+	break;
+    case AFMT_S16_LE:
+	fifo_recv_stereo_16le(buf, count, NO);
+	break;
+    case AFMT_U16_LE:
+	fifo_recv_stereo_16le(buf, count, YES);
+	break;
+    case AFMT_S16_BE:
+	fifo_recv_stereo_16be(buf, count, NO);
+	break;
+    case AFMT_U16_BE:
+	fifo_recv_stereo_16be(buf, count, YES);
+	break;
+    }
+}
+
+
+static void
+fifo_recv_monoral(pcm_data *buf, int count)
+{
+    /* Convert format and sampling speed. */
+    switch (pcm_s.format) {
+    case AFMT_MU_LAW:
+	fifo_recv_mono_ulaw(buf, count);
+	break;
+    case AFMT_S8:
+	fifo_recv_mono_8(buf, count, NO);
+	break;
+    case AFMT_U8:
+	fifo_recv_mono_8(buf, count, YES);
+	break;
+    case AFMT_S16_LE:
+	fifo_recv_mono_16le(buf, count, NO);
+	break;
+    case AFMT_U16_LE:
+	fifo_recv_mono_16le(buf, count, YES);
+	break;
+    case AFMT_S16_BE:
+	fifo_recv_mono_16be(buf, count, NO);
+	break;
+    case AFMT_U16_BE:
+	fifo_recv_mono_16be(buf, count, YES);
+	break;
+    }
+}
+
+
+static void
+fifo_recv_stereo_ulaw(pcm_data *buf, int count)
+{
+    int i, cnt;
+    signed char dl, dl0, dl1, dr, dr0, dr1;
+
+    cnt = count / 2;
+    if (pcm_s.speed == pcm_s.chipspeed) {
+	/* No reason to convert the pcm speed. */
+	for (i = 0; i < cnt; i++) {
+	    *buf++ = linear2ulaw[inb(pcm_s.iobase + 12)];
+	    *buf++ = linear2ulaw[inb(pcm_s.iobase + 12)];
+	}
+	if (count % 2) {
+	    *buf++ = linear2ulaw[inb(pcm_s.iobase + 12)];
+	    tmpbuf.buff[0] = linear2ulaw[inb(pcm_s.iobase + 12)];
+	    tmpbuf.size = 1;
+	}
+    } else {
+	/* Speed conversion with linear interpolation method. */
+	dl0 = pcm_s.last_l;
+	dr0 = pcm_s.last_r;
+	dl1 = inb(pcm_s.iobase + 12);
+	dr1 = inb(pcm_s.iobase + 12);
+	for (i = 0; i < cnt; i++) {
+	    while (pcm_s.acc >= pcm_s.speed) {
+		pcm_s.acc -= pcm_s.speed;
+		dl0 = dl1;
+		dr0 = dr1;
+		dl1 = inb(pcm_s.iobase + 12);
+		dr1 = inb(pcm_s.iobase + 12);
+	    }
+	    dl = ((dl0 * (pcm_s.speed - pcm_s.acc)) + (dl1 * pcm_s.acc))
+		/ pcm_s.speed;
+	    dr = ((dr0 * (pcm_s.speed - pcm_s.acc)) + (dr1 * pcm_s.acc))
+		/ pcm_s.speed;
+	    *buf++ = linear2ulaw[dl & 0xff];
+	    *buf++ = linear2ulaw[dr & 0xff];
+	    pcm_s.acc += pcm_s.chipspeed;
+	}
+	if (count % 2) {
+	    while (pcm_s.acc >= pcm_s.speed) {
+		pcm_s.acc -= pcm_s.speed;
+		dl0 = dl1;
+		dr0 = dr1;
+		dl1 = inb(pcm_s.iobase + 12);
+		dr1 = inb(pcm_s.iobase + 12);
+	    }
+	    dl = ((dl0 * (pcm_s.speed - pcm_s.acc)) + (dl1 * pcm_s.acc))
+		/ pcm_s.speed;
+	    dr = ((dr0 * (pcm_s.speed - pcm_s.acc)) + (dr1 * pcm_s.acc))
+		/ pcm_s.speed;
+	    *buf++ = linear2ulaw[dl & 0xff];
+	    tmpbuf.buff[0] = linear2ulaw[dr & 0xff];
+	    tmpbuf.size = 1;
+	}
+
+	pcm_s.last_l = dl0;
+	pcm_s.last_r = dr0;
+    }
+}
+
+
+static void
+fifo_recv_stereo_8(pcm_data *buf, int count, int uflag)
+{
+    int i, cnt;
+    signed char dl, dl0, dl1, dr, dr0, dr1, zlev;
+
+    zlev = uflag ? -128 : 0;
+
+    cnt = count / 2;
+    if (pcm_s.speed == pcm_s.chipspeed) {
+	/* No reason to convert the pcm speed. */
+	for (i = 0; i < cnt; i++) {
+	    *buf++ = inb(pcm_s.iobase + 12) + zlev;
+	    *buf++ = inb(pcm_s.iobase + 12) + zlev;
+	}
+	if (count % 2) {
+	    *buf++ = inb(pcm_s.iobase + 12) + zlev;
+	    tmpbuf.buff[0] = inb(pcm_s.iobase + 12) + zlev;
+	    tmpbuf.size = 1;
+	}
+    } else {
+	/* Speed conversion with linear interpolation method. */
+	dl0 = pcm_s.last_l;
+	dr0 = pcm_s.last_r;
+	dl1 = inb(pcm_s.iobase + 12);
+	dr1 = inb(pcm_s.iobase + 12);
+	for (i = 0; i < cnt; i++) {
+	    while (pcm_s.acc >= pcm_s.speed) {
+		pcm_s.acc -= pcm_s.speed;
+		dl0 = dl1;
+		dr0 = dr1;
+		dl1 = inb(pcm_s.iobase + 12);
+		dr1 = inb(pcm_s.iobase + 12);
+	    }
+	    dl = ((dl0 * (pcm_s.speed - pcm_s.acc)) + (dl1 * pcm_s.acc))
+		/ pcm_s.speed;
+	    dr = ((dr0 * (pcm_s.speed - pcm_s.acc)) + (dr1 * pcm_s.acc))
+		/ pcm_s.speed;
+	    *buf++ = dl + zlev;
+	    *buf++ = dr + zlev;
+	    pcm_s.acc += pcm_s.chipspeed;
+	}
+	if (count % 2) {
+	    while (pcm_s.acc >= pcm_s.speed) {
+		pcm_s.acc -= pcm_s.speed;
+		dl0 = dl1;
+		dr0 = dr1;
+		dl1 = inb(pcm_s.iobase + 12);
+		dr1 = inb(pcm_s.iobase + 12);
+	    }
+	    dl = ((dl0 * (pcm_s.speed - pcm_s.acc)) + (dl1 * pcm_s.acc))
+		/ pcm_s.speed;
+	    dr = ((dr0 * (pcm_s.speed - pcm_s.acc)) + (dr1 * pcm_s.acc))
+		/ pcm_s.speed;
+	    *buf++ = dl + zlev;
+	    tmpbuf.buff[0] = dr + zlev;
+	    tmpbuf.size = 1;
+	}
+
+	pcm_s.last_l = dl0;
+	pcm_s.last_r = dr0;
+    }
+}
+
+
+static void
+fifo_recv_stereo_16le(pcm_data *buf, int count, int uflag)
+{
+    int i, cnt;
+    short dl, dl0, dl1, dr, dr0, dr1, zlev;
+    pcm_data t[4];
+
+    zlev = uflag ? -128 : 0;
+
+    cnt = count / 4;
+    if (pcm_s.speed == pcm_s.chipspeed) {
+	/* No reason to convert the pcm speed. */
+	for (i = 0; i < cnt; i++) {
+	    *(buf + 1) = inb(pcm_s.iobase + 12) + zlev;
+	    *buf = inb(pcm_s.iobase + 12);
+	    *(buf + 3) = inb(pcm_s.iobase + 12) + zlev;
+	    *(buf + 2) = inb(pcm_s.iobase + 12);
+	    buf += 4;
+	}
+	if (count % 4) {
+	    t[1] = inb(pcm_s.iobase + 12) + zlev;
+	    t[0] = inb(pcm_s.iobase + 12);
+	    t[3] = inb(pcm_s.iobase + 12) + zlev;
+	    t[2] = inb(pcm_s.iobase + 12);
+	    tmpbuf.size = 0;
+	    for (i = 0; i < count % 4; i++)
+		*buf++ = t[i];
+	    for (i = count % 4; i < 4; i++)
+		tmpbuf.buff[tmpbuf.size++] = t[i];
+	}
+    } else {
+	/* Speed conversion with linear interpolation method. */
+	dl0 = pcm_s.last_l;
+	dr0 = pcm_s.last_r;
+	dl1 = inb(pcm_s.iobase + 12) << 8;
+	dl1 |= inb(pcm_s.iobase + 12);
+	dr1 = inb(pcm_s.iobase + 12) << 8;
+	dr1 |= inb(pcm_s.iobase + 12);
+	for (i = 0; i < cnt; i++) {
+	    while (pcm_s.acc >= pcm_s.speed) {
+		pcm_s.acc -= pcm_s.speed;
+		dl0 = dl1;
+		dr0 = dr1;
+		dl1 = inb(pcm_s.iobase + 12) << 8;
+		dl1 |= inb(pcm_s.iobase + 12);
+		dr1 = inb(pcm_s.iobase + 12) << 8;
+		dr1 |= inb(pcm_s.iobase + 12);
+	    }
+	    dl = ((dl0 * (pcm_s.speed - pcm_s.acc)) + (dl1 * pcm_s.acc))
+		/ pcm_s.speed;
+	    dr = ((dr0 * (pcm_s.speed - pcm_s.acc)) + (dr1 * pcm_s.acc))
+		/ pcm_s.speed;
+	    *buf++ = dl & 0xff;
+	    *buf++ = ((dl >> 8) & 0xff) + zlev;
+	    *buf++ = dr & 0xff;
+	    *buf++ = ((dr >> 8) & 0xff) + zlev;
+	    pcm_s.acc += pcm_s.chipspeed;
+	}
+	if (count % 4) {
+	    while (pcm_s.acc >= pcm_s.speed) {
+		pcm_s.acc -= pcm_s.speed;
+		dl0 = dl1;
+		dr0 = dr1;
+		dl1 = inb(pcm_s.iobase + 12) << 8;
+		dl1 |= inb(pcm_s.iobase + 12);
+		dr1 = inb(pcm_s.iobase + 12) << 8;
+		dr1 |= inb(pcm_s.iobase + 12);
+	    }
+	    dl = ((dl0 * (pcm_s.speed - pcm_s.acc)) + (dl1 * pcm_s.acc))
+		/ pcm_s.speed;
+	    dr = ((dr0 * (pcm_s.speed - pcm_s.acc)) + (dr1 * pcm_s.acc))
+		/ pcm_s.speed;
+	    t[0] = dl & 0xff;
+	    t[1] = ((dl >> 8) & 0xff) + zlev;
+	    t[2] = dr & 0xff;
+	    t[3] = ((dr >> 8) & 0xff) + zlev;
+	    tmpbuf.size = 0;
+	    for (i = 0; i < count % 4; i++)
+		*buf++ = t[i];
+	    for (i = count % 4; i < 4; i++)
+		tmpbuf.buff[tmpbuf.size++] = t[i];
+	}
+
+	pcm_s.last_l = dl0;
+	pcm_s.last_r = dr0;
+    }
+}
+
+
+static void
+fifo_recv_stereo_16be(pcm_data *buf, int count, int uflag)
+{
+    int i, cnt;
+    short dl, dl0, dl1, dr, dr0, dr1, zlev;
+    pcm_data t[4];
+
+    zlev = uflag ? -128 : 0;
+
+    cnt = count / 4;
+    if (pcm_s.speed == pcm_s.chipspeed) {
+	/* No reason to convert the pcm speed. */
+	for (i = 0; i < cnt; i++) {
+	    *buf++ = inb(pcm_s.iobase + 12) + zlev;
+	    *buf++ = inb(pcm_s.iobase + 12);
+	    *buf++ = inb(pcm_s.iobase + 12) + zlev;
+	    *buf++ = inb(pcm_s.iobase + 12);
+	}
+	if (count % 4) {
+	    t[0] = inb(pcm_s.iobase + 12) + zlev;
+	    t[1] = inb(pcm_s.iobase + 12);
+	    t[2] = inb(pcm_s.iobase + 12) + zlev;
+	    t[3] = inb(pcm_s.iobase + 12);
+	    tmpbuf.size = 0;
+	    for (i = 0; i < count % 4; i++)
+		*buf++ = t[i];
+	    for (i = count % 4; i < 4; i++)
+		tmpbuf.buff[tmpbuf.size++] = t[i];
+	}
+    } else {
+	/* Speed conversion with linear interpolation method. */
+	dl0 = pcm_s.last_l;
+	dr0 = pcm_s.last_r;
+	dl1 = inb(pcm_s.iobase + 12) << 8;
+	dl1 |= inb(pcm_s.iobase + 12);
+	dr1 = inb(pcm_s.iobase + 12) << 8;
+	dr1 |= inb(pcm_s.iobase + 12);
+	for (i = 0; i < cnt; i++) {
+	    while (pcm_s.acc >= pcm_s.speed) {
+		pcm_s.acc -= pcm_s.speed;
+		dl0 = dl1;
+		dr0 = dr1;
+		dl1 = inb(pcm_s.iobase + 12) << 8;
+		dl1 |= inb(pcm_s.iobase + 12);
+		dr1 = inb(pcm_s.iobase + 12) << 8;
+		dr1 |= inb(pcm_s.iobase + 12);
+	    }
+	    dl = ((dl0 * (pcm_s.speed - pcm_s.acc)) + (dl1 * pcm_s.acc))
+		/ pcm_s.speed;
+	    dr = ((dr0 * (pcm_s.speed - pcm_s.acc)) + (dr1 * pcm_s.acc))
+		/ pcm_s.speed;
+	    *buf++ = ((dl >> 8) & 0xff) + zlev;
+	    *buf++ = dl & 0xff;
+	    *buf++ = ((dr >> 8) & 0xff) + zlev;
+	    *buf++ = dr & 0xff;
+	    pcm_s.acc += pcm_s.chipspeed;
+	}
+	if (count % 4) {
+	    while (pcm_s.acc >= pcm_s.speed) {
+		pcm_s.acc -= pcm_s.speed;
+		dl0 = dl1;
+		dr0 = dr1;
+		dl1 = inb(pcm_s.iobase + 12) << 8;
+		dl1 |= inb(pcm_s.iobase + 12);
+		dr1 = inb(pcm_s.iobase + 12) << 8;
+		dr1 |= inb(pcm_s.iobase + 12);
+	    }
+	    dl = ((dl0 * (pcm_s.speed - pcm_s.acc)) + (dl1 * pcm_s.acc))
+		/ pcm_s.speed;
+	    dr = ((dr0 * (pcm_s.speed - pcm_s.acc)) + (dr1 * pcm_s.acc))
+		/ pcm_s.speed;
+	    t[0] = ((dl >> 8) & 0xff) + zlev;
+	    t[1] = dl & 0xff;
+	    t[2] = ((dr >> 8) & 0xff) + zlev;
+	    t[3] = dr & 0xff;
+	    tmpbuf.size = 0;
+	    for (i = 0; i < count % 4; i++)
+		*buf++ = t[i];
+	    for (i = count % 4; i < 4; i++)
+		tmpbuf.buff[tmpbuf.size++] = t[i];
+	}
+
+	pcm_s.last_l = dl0;
+	pcm_s.last_r = dr0;
+    }
+}
+
+
+static void
+fifo_recv_mono_ulaw(pcm_data *buf, int count)
+{
+    int i;
+    signed char d, d0, d1;
+
+    if (pcm_s.speed == pcm_s.chipspeed) {
+	/* No reason to convert the pcm speed. */
+	for (i = 0; i < count; i++) {
+	    d = ((signed char)inb(pcm_s.iobase + 12)
+		 + (signed char)inb(pcm_s.iobase + 12)) >> 1;
+	    *buf++ = linear2ulaw[d & 0xff];
+	}
+    } else {
+	/* Speed conversion with linear interpolation method. */
+	d0 = pcm_s.last_l;
+	d1 = ((signed char)inb(pcm_s.iobase + 12)
+	      + (signed char)inb(pcm_s.iobase + 12)) >> 1;
+	for (i = 0; i < count; i++) {
+	    while (pcm_s.acc >= pcm_s.speed) {
+		pcm_s.acc -= pcm_s.speed;
+		d0 = d1;
+		d1 = ((signed char)inb(pcm_s.iobase + 12)
+		      + (signed char)inb(pcm_s.iobase + 12)) >> 1;
+	    }
+	    d = ((d0 * (pcm_s.speed - pcm_s.acc)) + (d1 * pcm_s.acc))
+		/ pcm_s.speed;
+	    *buf++ = linear2ulaw[d & 0xff];
+	    pcm_s.acc += pcm_s.chipspeed;
+	}
+
+	pcm_s.last_l = d0;
+    }
+}
+
+
+static void
+fifo_recv_mono_8(pcm_data *buf, int count, int uflag)
+{
+    int i;
+    signed char d, d0, d1, zlev;
+
+    zlev = uflag ? -128 : 0;
+
+    if (pcm_s.speed == pcm_s.chipspeed) {
+	/* No reason to convert the pcm speed. */
+	for (i = 0; i < count; i++) {
+	    d = ((signed char)inb(pcm_s.iobase + 12)
+		 + (signed char)inb(pcm_s.iobase + 12)) >> 1;
+	    *buf++ = d + zlev;
+	}
+    } else {
+	/* Speed conversion with linear interpolation method. */
+	d0 = pcm_s.last_l;
+	d1 = ((signed char)inb(pcm_s.iobase + 12)
+	      + (signed char)inb(pcm_s.iobase + 12)) >> 1;
+	for (i = 0; i < count; i++) {
+	    while (pcm_s.acc >= pcm_s.speed) {
+		pcm_s.acc -= pcm_s.speed;
+		d0 = d1;
+		d1 = ((signed char)inb(pcm_s.iobase + 12)
+		      + (signed char)inb(pcm_s.iobase + 12)) >> 1;
+	    }
+	    d = ((d0 * (pcm_s.speed - pcm_s.acc)) + (d1 * pcm_s.acc))
+		/ pcm_s.speed;
+	    *buf++ = d + zlev;
+	    pcm_s.acc += pcm_s.chipspeed;
+	}
+
+	pcm_s.last_l = d0;
+    }
+}
+
+
+static void
+fifo_recv_mono_16le(pcm_data *buf, int count, int uflag)
+{
+    int i, cnt;
+    short d, d0, d1, el, er, zlev;
+
+    zlev = uflag ? -128 : 0;
+
+    cnt = count / 2;
+    if (pcm_s.speed == pcm_s.chipspeed) {
+	/* No reason to convert the pcm speed. */
+	for (i = 0; i < cnt; i++) {
+	    el = inb(pcm_s.iobase + 12) << 8;
+	    el |= inb(pcm_s.iobase + 12);
+	    er = inb(pcm_s.iobase + 12) << 8;
+	    er |= inb(pcm_s.iobase + 12);
+	    d = (el + er) >> 1;
+	    *buf++ = d & 0xff;
+	    *buf++ = ((d >> 8) & 0xff) + zlev;
+	}
+	if (count % 2) {
+	    el = inb(pcm_s.iobase + 12) << 8;
+	    el |= inb(pcm_s.iobase + 12);
+	    er = inb(pcm_s.iobase + 12) << 8;
+	    er |= inb(pcm_s.iobase + 12);
+	    d = (el + er) >> 1;
+	    *buf++ = d & 0xff;
+	    tmpbuf.buff[0] = ((d >> 8) & 0xff) + zlev;
+	    tmpbuf.size = 1;
+	}
+    } else {
+	/* Speed conversion with linear interpolation method. */
+	d0 = pcm_s.last_l;
+	el = inb(pcm_s.iobase + 12) << 8;
+	el |= inb(pcm_s.iobase + 12);
+	er = inb(pcm_s.iobase + 12) << 8;
+	er |= inb(pcm_s.iobase + 12);
+	d1 = (el + er) >> 1;
+	for (i = 0; i < cnt; i++) {
+	    while (pcm_s.acc >= pcm_s.speed) {
+		pcm_s.acc -= pcm_s.speed;
+		d0 = d1;
+		el = inb(pcm_s.iobase + 12) << 8;
+		el |= inb(pcm_s.iobase + 12);
+		er = inb(pcm_s.iobase + 12) << 8;
+		er |= inb(pcm_s.iobase + 12);
+		d1 = (el + er) >> 1;
+	    }
+	    d = ((d0 * (pcm_s.speed - pcm_s.acc)) + (d1 * pcm_s.acc))
+		/ pcm_s.speed;
+	    *buf++ = d & 0xff;
+	    *buf++ = ((d >> 8) & 0xff) + zlev;
+	    pcm_s.acc += pcm_s.chipspeed;
+	}
+	if (count % 2) {
+	    while (pcm_s.acc >= pcm_s.speed) {
+		pcm_s.acc -= pcm_s.speed;
+		d0 = d1;
+		el = inb(pcm_s.iobase + 12) << 8;
+		el |= inb(pcm_s.iobase + 12);
+		er = inb(pcm_s.iobase + 12) << 8;
+		er |= inb(pcm_s.iobase + 12);
+		d1 = (el + er) >> 1;
+	    }
+	    d = ((d0 * (pcm_s.speed - pcm_s.acc)) + (d1 * pcm_s.acc))
+		/ pcm_s.speed;
+	    *buf++ = d & 0xff;
+	    tmpbuf.buff[0] = ((d >> 8) & 0xff) + zlev;
+	    tmpbuf.size = 1;
+	}
+
+	pcm_s.last_l = d0;
+    }
+}
+
+
+static void
+fifo_recv_mono_16be(pcm_data *buf, int count, int uflag)
+{
+    int i, cnt;
+    short d, d0, d1, el, er, zlev;
+
+    zlev = uflag ? -128 : 0;
+
+    cnt = count / 2;
+    if (pcm_s.speed == pcm_s.chipspeed) {
+	/* No reason to convert the pcm speed. */
+	for (i = 0; i < cnt; i++) {
+	    el = inb(pcm_s.iobase + 12) << 8;
+	    el |= inb(pcm_s.iobase + 12);
+	    er = inb(pcm_s.iobase + 12) << 8;
+	    er |= inb(pcm_s.iobase + 12);
+	    d = (el + er) >> 1;
+	    *buf++ = ((d >> 8) & 0xff) + zlev;
+	    *buf++ = d & 0xff;
+	}
+	if (count % 2) {
+	    el = inb(pcm_s.iobase + 12) << 8;
+	    el |= inb(pcm_s.iobase + 12);
+	    er = inb(pcm_s.iobase + 12) << 8;
+	    er |= inb(pcm_s.iobase + 12);
+	    d = (el + er) >> 1;
+	    *buf++ = ((d >> 8) & 0xff) + zlev;
+	    tmpbuf.buff[0] = d & 0xff;
+	    tmpbuf.size = 1;
+	}
+    } else {
+	/* Speed conversion with linear interpolation method. */
+	d0 = pcm_s.last_l;
+	el = inb(pcm_s.iobase + 12) << 8;
+	el |= inb(pcm_s.iobase + 12);
+	er = inb(pcm_s.iobase + 12) << 8;
+	er |= inb(pcm_s.iobase + 12);
+	d1 = (el + er) >> 1;
+	for (i = 0; i < cnt; i++) {
+	    while (pcm_s.acc >= pcm_s.speed) {
+		pcm_s.acc -= pcm_s.speed;
+		d0 = d1;
+		el = inb(pcm_s.iobase + 12) << 8;
+		el |= inb(pcm_s.iobase + 12);
+		er = inb(pcm_s.iobase + 12) << 8;
+		er |= inb(pcm_s.iobase + 12);
+		d1 = (el + er) >> 1;
+	    }
+	    d = ((d0 * (pcm_s.speed - pcm_s.acc)) + (d1 * pcm_s.acc))
+		/ pcm_s.speed;
+	    *buf++ = ((d >> 8) & 0xff) + zlev;
+	    *buf++ = d & 0xff;
+	    pcm_s.acc += pcm_s.chipspeed;
+	}
+	if (count % 2) {
+	    while (pcm_s.acc >= pcm_s.speed) {
+		pcm_s.acc -= pcm_s.speed;
+		d0 = d1;
+		el = inb(pcm_s.iobase + 12) << 8;
+		el |= inb(pcm_s.iobase + 12);
+		er = inb(pcm_s.iobase + 12) << 8;
+		er |= inb(pcm_s.iobase + 12);
+		d1 = (el + er) >> 1;
+	    }
+	    d = ((d0 * (pcm_s.speed - pcm_s.acc)) + (d1 * pcm_s.acc))
+		/ pcm_s.speed;
+	    *buf++ = ((d >> 8) & 0xff) + zlev;
+	    tmpbuf.buff[0] = d & 0xff;
+	    tmpbuf.size = 1;
+	}
+
+	pcm_s.last_l = d0;
+    }
+}
+
+
+static void
+pcm_stop(void)
+{
+    fifo_stop();		/* stop FIFO */
+    fifo_reset();		/* reset FIFO buffer */
+
+    /* Reset driver's status. */
+    pcm_s.intr_busy = NO;
+    pcm_s.intr_last = NO;
+    pcm_s.intr_trailer = NO;
+    pcm_s.acc = 0;
+    pcm_s.last_l = 0;
+    pcm_s.last_r = 0;
+
+    DEB(printk("pcm_stop\n"));
+}
+
+
+static void
+pcm_init(void)
+{
+    /* Initialize registers on the board. */
+    pcm_stop();
+    if (pcm_s.board_type == PC980173_FAMILY)
+	dsp73_init();
+
+    /* Set default volume. */
+    set_volume(DEFAULT_VOLUME);
+
+    /* Initialize driver's status. */
+    pcm_s.opened = NO;
+    pcm_initialized = YES;
+}
+
+
+/*
+ * Codes for global use
+ */
+
+int
+probe_pcm86(struct address_info *hw_config)
+{
+    return pcm86_detect(hw_config);
+}
+
+
+long
+attach_pcm86(long mem_start, struct address_info *hw_config)
+{
+    if (pcm_s.board_type == NO_SUPPORTED_BOARD)
+	return mem_start;
+
+    /* Initialize the board. */
+    pcm_init();
+
+    printk("pcm0: <%s>", pcm86_operations.name);
+
+    if (num_audiodevs < MAX_AUDIO_DEV) {
+	my_dev = num_audiodevs++;
+	audio_devs[my_dev] = &pcm86_operations;
+	audio_devs[my_dev]->buffcount = DSP_BUFFCOUNT;
+	audio_devs[my_dev]->buffsize = DSP_BUFFSIZE;
+#ifdef PCM86_DEBUG
+	printk("\nbuffsize = %d", DSP_BUFFSIZE);
+#endif
+    } else
+	printk("pcm0: Too many PCM devices available");
+
+    return mem_start;
+}
+
+
+static int
+pcm86_detect(struct address_info *hw_config)
+{
+    int opna_iobase = 0x188, irq = 12, i;
+    unsigned char tmp;
+
+    if (hw_config->io_base == -1) {
+	printf("pcm0: iobase not specified. Assume default port(0x%x)\n",
+	       PCM86_IOBASE);
+	hw_config->io_base = PCM86_IOBASE;
+    }
+    pcm_s.iobase = hw_config->io_base;
+
+    /* auto configuration */
+    tmp = inb(pcm_s.iobase) & 0xfc;
+    switch ((tmp & 0xf0) >> 4) {
+    case 2:
+	opna_iobase = 0x188;
+	pcm_s.board_type = PC980173_FAMILY;
+	break;
+    case 3:
+	opna_iobase = 0x288;
+	pcm_s.board_type = PC980173_FAMILY;
+	break;
+    case 4:
+	opna_iobase = 0x188;
+	pcm_s.board_type = PC980186_FAMILY;
+	break;
+    case 5:
+	opna_iobase = 0x288;
+	pcm_s.board_type = PC980186_FAMILY;
+	break;
+    default:
+	pcm_s.board_type = NO_SUPPORTED_BOARD;
+	return NO;
+    }
+
+    /* Enable OPNA(YM2608) facilities. */
+    outb(pcm_s.iobase, tmp | 0x01);
+
+    /* Wait for OPNA to be ready. */
+    i = 100000;		/* Some large value */
+    while((inb(opna_iobase) & 0x80) && (i-- > 0));
+
+    /* Make IOA/IOB port ready (IOA:input, IOB:output) */
+    outb(opna_iobase, 0x07);
+    outb(0x5f, 0);	/* Because OPNA ports are comparatively slow(?), */
+    outb(0x5f, 0);	/* we'd better wait a moment. */
+    outb(0x5f, 0);
+    outb(0x5f, 0);
+    tmp = inb(opna_iobase + 2) & 0x3f;
+    outb(opna_iobase + 2, tmp | 0x80);
+
+    /* Wait for OPNA to be ready. */
+    i = 100000;		/* Some large value */
+    while((inb(opna_iobase) & 0x80) && (i-- > 0));
+
+    /* Get irq number from IOA port. */
+    outb(opna_iobase, 0x0e);
+    outb(0x5f, 0);
+    outb(0x5f, 0);
+    outb(0x5f, 0);
+    outb(0x5f, 0);
+    tmp = inb(opna_iobase + 2) & 0xc0;
+    switch (tmp >> 6) {
+    case 0:	/* INT0 (IRQ3)*/
+	irq = 3;
+	break;
+    case 1:	/* INT6 (IRQ13)*/
+	irq = 13;
+	break;
+    case 2:	/* INT4 (IRQ10)*/
+	irq = 10;
+	break;
+    case 3:	/* INT5 (IRQ12)*/
+	irq = 12;
+	break;
+    default:	/* error */
+	return NO;
+    }
+
+    /* Wait for OPNA to be ready. */
+    i = 100000;		/* Some large value */
+    while((inb(opna_iobase) & 0x80) && (i-- > 0));
+
+    /* Reset OPNA timer register. */
+    outb(opna_iobase, 0x27);
+    outb(0x5f, 0);
+    outb(0x5f, 0);
+    outb(0x5f, 0);
+    outb(0x5f, 0);
+    outb(opna_iobase + 2, 0x30);
+
+    /* Ok.  Detection finished. */
+    sprintf(pcm86_operations.name, board_name[pcm_s.board_type]);
+    pcm_initialized = NO;
+    pcm_s.irq = irq;
+
+    if ((hw_config->irq > 0) && (hw_config->irq != irq))
+	printf("pcm0: change irq %d -> %d\n", hw_config->irq, irq);
+    hw_config->irq = irq;
+
+    return YES;
+}
+
+
+static int
+pcm86_open(int dev, int mode)
+{
+    int err;
+
+    if (!pcm_initialized)
+	return RET_ERROR(ENXIO);
+
+    if (pcm_s.intr_busy || pcm_s.opened)
+	return RET_ERROR(EBUSY);
+
+    if ((err = snd_set_irq_handler(pcm_s.irq, pcmintr, "PC-9801-73/86")) < 0)
+	return err;
+
+    pcm_stop();
+
+    tmpbuf.size = 0;
+    pcm_s.intr_mode = IMODE_NONE;
+    pcm_s.opened = YES;
+
+    return 0;
+}
+
+
+static void
+pcm86_close(int dev)
+{
+    snd_release_irq(pcm_s.irq);
+
+    pcm_s.opened = NO;
+}
+
+
+static void
+pcm86_output_block(int dev, unsigned long buf, int count, int intrflag,
+		   int dma_restart)
+{
+    unsigned long flags, cnt;
+    int maxchunksize;
+
+#ifdef PCM86_DEBUG
+    printk("pcm86_output_block():");
+    if (audio_devs[dev]->dmap->flags & DMA_BUSY)
+	printk(" DMA_BUSY");
+    if (audio_devs[dev]->dmap->flags & DMA_RESTART)
+	printk(" DMA_RESTART");
+    if (audio_devs[dev]->dmap->flags & DMA_ACTIVE)
+	printk(" DMA_ACTIVE");
+    if (audio_devs[dev]->dmap->flags & DMA_STARTED)
+	printk(" DMA_STARTED");
+    if (audio_devs[dev]->dmap->flags & DMA_ALLOC_DONE)
+	printk(" DMA_ALLOC_DONE");
+    printk("\n");
+#endif
+
+#if 0
+    DISABLE_INTR(flags);
+#endif
+
+#ifdef PCM86_DEBUG
+    printk("pcm86_output_block(): count = %d, intrsize= %d\n",
+	   count, pcm_s.intr_size);
+#endif
+
+    pcm_s.pdma_buf = (pcm_data *)buf;
+    pcm_s.pdma_count = count;
+    pcm_s.pdma_chunkcount = 1;
+    maxchunksize = (((PCM86_FIFOSIZE - pcm_s.intr_size * 2)
+		     / (pcm_s.bytes * 2)) * pcm_s.speed
+		    / pcm_s.chipspeed) * (pcm_s.bytes << pcm_s.stereo);
+    if (count > maxchunksize)
+	pcm_s.pdma_chunkcount = 2 * count / maxchunksize;
+    /*
+     * Let chunksize = (float)count / (float)pcm_s.pdma_chunkcount.
+     * Data of size chunksize is sent to the FIFO buffer on the 86-board
+     * on every occuring of interrupt.
+     * By assuming that pcm_s.intr_size < PCM86_FIFOSIZE / 2, we can conclude
+     * that the FIFO buffer never overflows from the following lemma.
+     *
+     * Lemma:
+     *	     maxchunksize / 2 <= chunksize <= maxchunksize.
+     *   (Though pcm_s.pdma_chunkcount is obtained through the flooring
+     *    function, this inequality holds.)
+     * Proof) Omitted.
+     */
+
+    fifo_output_block();
+
+    pcm_s.intr_last = NO;
+    pcm_s.intr_mode = IMODE_OUTPUT;
+    if (!pcm_s.intr_busy)
+	fifo_start(IMODE_OUTPUT);
+    pcm_s.intr_busy = YES;
+
+#if 0
+    RESTORE_INTR(flags);
+#endif
+}
+
+
+static void
+pcm86_start_input(int dev, unsigned long buf, int count, int intrflag,
+		  int dma_restart)
+{
+    unsigned long flags, cnt;
+    int maxchunksize;
+
+#ifdef PCM86_DEBUG
+    printk("pcm86_start_input():");
+    if (audio_devs[dev]->dmap->flags & DMA_BUSY)
+	printk(" DMA_BUSY");
+    if (audio_devs[dev]->dmap->flags & DMA_RESTART)
+	printk(" DMA_RESTART");
+    if (audio_devs[dev]->dmap->flags & DMA_ACTIVE)
+	printk(" DMA_ACTIVE");
+    if (audio_devs[dev]->dmap->flags & DMA_STARTED)
+	printk(" DMA_STARTED");
+    if (audio_devs[dev]->dmap->flags & DMA_ALLOC_DONE)
+	printk(" DMA_ALLOC_DONE");
+    printk("\n");
+#endif
+
+#if 0
+    DISABLE_INTR(flags);
+#endif
+
+    pcm_s.intr_size = PCM86_INTRSIZE_IN;
+
+#ifdef PCM86_DEBUG
+    printk("pcm86_start_input(): count = %d, intrsize= %d\n",
+	   count, pcm_s.intr_size);
+#endif
+
+    pcm_s.pdma_buf = (pcm_data *)buf;
+    pcm_s.pdma_count = count;
+    pcm_s.pdma_chunkcount = 1;
+    maxchunksize = ((pcm_s.intr_size / (pcm_s.bytes * 2)) * pcm_s.speed
+		    / pcm_s.chipspeed) * (pcm_s.bytes << pcm_s.stereo);
+    if (count > maxchunksize)
+	pcm_s.pdma_chunkcount = 2 * count / maxchunksize;
+
+    pcm_s.intr_mode = IMODE_INPUT;
+    if (!pcm_s.intr_busy)
+	fifo_start(IMODE_INPUT);
+    pcm_s.intr_busy = YES;
+
+#if 0
+    RESTORE_INTR(flags);
+#endif
+}
+
+
+static int
+pcm86_ioctl(int dev, unsigned int cmd, unsigned int arg, int local)
+{
+    switch (cmd) {
+    case SOUND_PCM_WRITE_RATE:
+	if (local)
+	    return set_speed(arg);
+	return IOCTL_OUT(arg, set_speed(IOCTL_IN(arg)));
+
+    case SOUND_PCM_READ_RATE:
+	if (local)
+	    return pcm_s.speed;
+	return IOCTL_OUT(arg, pcm_s.speed);
+
+    case SNDCTL_DSP_STEREO:
+	if (local)
+	    return set_stereo(arg);
+	return IOCTL_OUT(arg, set_stereo(IOCTL_IN(arg)));
+
+    case SOUND_PCM_WRITE_CHANNELS:
+	if (local)
+	    return set_stereo(arg - 1) + 1;
+	return IOCTL_OUT(arg, set_stereo(IOCTL_IN(arg) - 1) + 1);
+
+    case SOUND_PCM_READ_CHANNELS:
+	if (local)
+	    return pcm_s.stereo + 1;
+	return IOCTL_OUT(arg, pcm_s.stereo + 1);
+
+    case SNDCTL_DSP_SETFMT:
+	if (local)
+	    return set_format(arg);
+	return IOCTL_OUT(arg, set_format(IOCTL_IN(arg)));
+
+    case SOUND_PCM_READ_BITS:
+	if (local)
+	    return pcm_s.bytes * 8;
+	return IOCTL_OUT(arg, pcm_s.bytes * 8);
+    }
+
+    /* Invalid ioctl request */
+    return RET_ERROR(EINVAL);
+}
+
+
+static int
+pcm86_prepare_for_input(int dev, int bufsize, int nbufs)
+{
+    pcm_s.intr_size = PCM86_INTRSIZE_IN;
+    pcm_s.intr_mode = IMODE_NONE;
+    pcm_s.acc = 0;
+    pcm_s.last_l = 0;
+    pcm_s.last_r = 0;
+
+    DEB(printk("pcm86_prepare_for_input\n"));
+
+    return 0;
+}
+
+
+static int
+pcm86_prepare_for_output(int dev, int bufsize, int nbufs)
+{
+    pcm_s.intr_size = PCM86_INTRSIZE_OUT;
+    pcm_s.intr_mode = IMODE_NONE;
+    pcm_s.acc = 0;
+    pcm_s.last_l = 0;
+    pcm_s.last_r = 0;
+
+    DEB(printk("pcm86_prepare_for_output\n"));
+
+    return 0;
+}
+
+
+static void
+pcm86_reset(int dev)
+{
+    pcm_stop();
+}
+
+
+static void
+pcm86_halt_xfer(int dev)
+{
+    pcm_stop();
+
+    DEB(printk("pcm86_halt_xfer\n"));
+}
+
+
+void
+pcmintr(int unit)
+{
+    unsigned char tmp;
+
+    if ((inb(pcm_s.iobase + 8) & 0x10) == 0)
+	return;		/* not FIFO intr. */
+
+    switch(pcm_s.intr_mode) {
+    case IMODE_OUTPUT:
+	if (pcm_s.intr_trailer) {
+	    DEB(printk("pcmintr(): fifo_reset\n"));
+	    fifo_reset();
+	    pcm_s.intr_trailer = NO;
+	    pcm_s.intr_busy = NO;
+	}
+	if (pcm_s.pdma_count > 0)
+	    fifo_output_block();
+	else
+	    DMAbuf_outputintr(my_dev, 1);
+	/* Reset intr. flag. */
+	tmp = inb(pcm_s.iobase + 8);
+	outb(pcm_s.iobase + 8, tmp & ~0x10);
+	outb(pcm_s.iobase + 8, tmp | 0x10);
+	break;
+
+    case IMODE_INPUT:
+	fifo_input_block();
+	if (pcm_s.pdma_count == 0)
+	    DMAbuf_inputintr(my_dev);
+	/* Reset intr. flag. */
+	tmp = inb(pcm_s.iobase + 8);
+	outb(pcm_s.iobase + 8, tmp & ~0x10);
+	outb(pcm_s.iobase + 8, tmp | 0x10);
+	break;
+
+    default:
+	pcm_stop();
+	printk("pcm0: unexpected interrupt\n");
+    }
+}
+
+
+#endif	/* EXCLUDE_PCM86, EXCLUDE_AUDIO */
+
+#endif	/* CONFIGURE_SOUNDCARD */
diff --git a/sys/i386/isa/sound/sb_defs.h b/sys/i386/isa/sound/sb_defs.h
new file mode 100644
index 000000000000..e21c7c86e9ee
--- /dev/null
+++ b/sys/i386/isa/sound/sb_defs.h
@@ -0,0 +1,43 @@
+#ifdef PC98
+#define DSP_RESET	(sbc_base + 0x600)
+#define DSP_READ	(sbc_base + 0xA00)
+#define DSP_WRITE	(sbc_base + 0xC00)
+#define DSP_COMMAND	(sbc_base + 0xC00)
+#define DSP_STATUS	(sbc_base + 0xC00)
+#define DSP_DATA_AVAIL	(sbc_base + 0xE00)
+#define DSP_DATA_AVL16	(sbc_base + 0xF00)
+#define MIXER_ADDR	(sbc_base + 0x400)
+#define MIXER_DATA	(sbc_base + 0x500)
+#define OPL3_LEFT	(sbc_base + 0x000)
+#define OPL3_RIGHT	(sbc_base + 0x200)
+#define OPL3_BOTH	(sbc_base + 0x800)
+#else
+#define DSP_RESET	(sbc_base + 0x6)
+#define DSP_READ	(sbc_base + 0xA)
+#define DSP_WRITE	(sbc_base + 0xC)
+#define DSP_COMMAND	(sbc_base + 0xC)
+#define DSP_STATUS	(sbc_base + 0xC)
+#define DSP_DATA_AVAIL	(sbc_base + 0xE)
+#define DSP_DATA_AVL16	(sbc_base + 0xF)
+#define MIXER_ADDR	(sbc_base + 0x4)
+#define MIXER_DATA	(sbc_base + 0x5)
+#define OPL3_LEFT	(sbc_base + 0x0)
+#define OPL3_RIGHT	(sbc_base + 0x2)
+#define OPL3_BOTH	(sbc_base + 0x8)
+#endif
+/* DSP Commands */
+
+#define DSP_CMD_SPKON		0xD1
+#define DSP_CMD_SPKOFF		0xD3
+#define DSP_CMD_DMAON		0xD0
+#define DSP_CMD_DMAOFF		0xD4
+
+#define IMODE_NONE		0
+#define IMODE_OUTPUT		1
+#define IMODE_INPUT		2
+#define IMODE_INIT		3
+#define IMODE_MIDI		4
+
+#define NORMAL_MIDI	0
+#define UART_MIDI	1
+
diff --git a/sys/i386/isa/sound/sbcard.h b/sys/i386/isa/sound/sbcard.h
new file mode 100644
index 000000000000..464128ccf46c
--- /dev/null
+++ b/sys/i386/isa/sound/sbcard.h
@@ -0,0 +1,54 @@
+/*
+ * file: sbcard.h
+ */
+
+extern int sbc_major, sbc_minor ;
+/*
+ * sound blaster registers
+ */
+
+#define DSP_RESET	(sbc_base + 0x6)
+#define DSP_READ	(sbc_base + 0xA)
+#define DSP_WRITE	(sbc_base + 0xC)
+#define DSP_COMMAND	(sbc_base + 0xC)
+#define DSP_STATUS	(sbc_base + 0xC)
+#define DSP_DATA_AVAIL	(sbc_base + 0xE)
+#define DSP_DATA_AVL16	(sbc_base + 0xF)
+#define MIXER_ADDR	(sbc_base + 0x4)
+#define MIXER_DATA	(sbc_base + 0x5)
+#define OPL3_LEFT	(sbc_base + 0x0)
+#define OPL3_RIGHT	(sbc_base + 0x2)
+#define OPL3_BOTH	(sbc_base + 0x8)
+
+/*
+ * DSP Commands. There are many, and in many cases they are used explicitly
+ */
+
+#define DSP_CMD_SPKON		0xD1
+#define DSP_CMD_SPKOFF		0xD3
+#define DSP_CMD_DMAON		0xD0	/* ??? the comment says Halt DMA */
+#define DSP_CMD_DMAOFF		0xD4	/* ??? comment says continue dma */
+
+#define	DSP_CMD_DMAHALT		0xD0
+#define	DSP_CMD_TCONST		0x40	/* set time constant */
+#define	DSP_CMD_HSSIZE		0x48	/* high speed dma count */
+#define	DSP_CMD_HSDAC		0x91	/* high speed dac */
+#define	DSP_CMD_HSADC		0x99	/* high speed adc */
+#define	DSP_CMD_DAC8		0x14	/* 8-bit dac (dma count) */
+#define	DSP_CMD_ADC8		0x24	/* 8-bit adc (dma count) */
+
+#define	DSP_CMD_GETVER		0xE1
+#define	DSP_CMD_GETID		0xE7	/* return id bytes */
+
+#if 0 /*** unknown ***/
+
+#endif
+
+#define IMODE_NONE		0
+#define IMODE_OUTPUT		PCM_ENABLE_OUTPUT
+#define IMODE_INPUT		PCM_ENABLE_INPUT
+#define IMODE_INIT		3
+#define IMODE_MIDI		4
+
+#define NORMAL_MIDI	0
+#define UART_MIDI	1
diff --git a/sys/i386/isa/sound/sound.doc b/sys/i386/isa/sound/sound.doc
new file mode 100644
index 000000000000..f6db243c407b
--- /dev/null
+++ b/sys/i386/isa/sound/sound.doc
@@ -0,0 +1,87 @@
+Instructions on using audio on a FreeBSD 2.1 (or 2.0-current) system.
+See also /sys/i386/conf/LINT.
+
+You may add one or more of the following depending on what you do and don't
+want compiled into your kernel.  Note:  Excluding things with EXCLUDE_... 
+is NOT recommended unless you really know what you're doing.
+
+      options         "AUDIO_MPU401"          # INCLUDE MPU401 support
+      options         AUDIO_GUS               # INCLUDE GUS support
+      options         AUDIO_SBPRO             # INCLUDE SB Pro support
+      options         "AUDIO_SB16"            # INCLUDE SB 16 support
+      options         "AUDIO_YM3812"          # INCLUDE AdLib support
+      options         AUDIO_PAS               # INCLUDE Pro Audio Studio support
+      options         AUDIO_SB                # INCLUDE SB support
+      options         AUDIO_MSS               # INCLUDE MSS support
+      options         EXCLUDE_AUDIO           # NO digital audio support
+      options         EXCLUDE_SEQUENCER       # NO sequencer support
+      options         EXCLUDE_GUS_IODETECT    # NO GUS io detection
+      options         EXCLUDE_SB_EMULATION    # NO PAS SB emulation support
+      options         "EXCLUDE_OPL3"          # NO OPL3 chip support
+      options         EXCLUDE_PRO_MIDI        # NO PAS MIDI support
+      options         EXCLUDE_CHIP_MIDI       # NO MIDI chip support
+      options	      EXCLUDE_MIDI	      # NO MIDI support whatsoever
+
+To enable sound card support, you need to uncomment and add one or more of 
+the following lines to your kernel configuration file according to the
+directions below:
+
+#device snd5 at isa? port 0x330 irq 6 vector mpuintr
+#device snd4 at isa? port 0x220 irq 15 drq 6 vector gusintr
+#device snd3 at isa? port 0x388 irq 10 drq 6 vector pasintr
+#device snd2 at isa? port 0x220 irq 7 drq 1 vector sbintr
+#device snd6 at isa? port 0x220 irq 7 drq 5 vector sbintr
+#device snd10 at isa? port 0x530 irq 10 drq 1 vector adintr
+#device snd7 at isa? port 0x300
+#device snd1 at isa? port 0x388
+
+Note for PAS user: you should change snd1 line to
+#device snd1 at isa? port 0x38a
+(next stereo port) to avoid conflict with snd3
+
+       Unit numbers are:
+                 1 for Yamaha FM synth
+                 2 for SB/SB Pro DSP
+                 3 for PAS PCM and Midi
+                 4 for GUS
+                 5 for MPU-401 (there is separate driver for the SB16)
+                 6 for SB16 (DSP)
+                 7 for SB16 Midi (MPU-401 emulation)
+                 10 for Microsoft Windows Sound System (AD1846)
+
+       If you have ProAudioSpectrum, uncomment units 3, 2 and 1
+       If you have SoundBlaster 1.0 to 2.0 or SB Pro, uncomment 2 and 1.
+       If you have SoundBlaster 16, uncomment 2, 1, 6 and 7.
+       (use the same IRQ for the cards 2, 6 and 7. The DMA of the
+       card 2 is the 8 bit one and the DMA of the card 6 is the 16 bit one.
+       the port address of the card 7 is the Midi I/O address of the SB16.
+       If you have GravisUltrasound, uncomment 4
+       If you have MPU-401, uncomment 5
+
+NOTE:  The MPU-401 driver may or may not work, and is unfortunately
+unverifiable since no one I know has one.  If you can test this,
+please let me know!  Also note that you will have to change these
+settings if your soundcard is set for a non-standard address or IRQ.
+Please check your documentation (or verify with any provided DOS utilities
+that may have come with your card) and set the IRQ or address fields
+accordingly.
+
+Also:  Some systems with the OPTI chipset will require you to #define
+BROKEN_BUS_CLOCK in /sys/i386/sound/pas2_card.c.  Symptoms are that
+you will hear a lot of clicking and popping sounds, like a geiger counter,
+coming out of the PAS even when is not playing anything.
+
+Also:  You can configure more then one card on a single DMA using 
+ALLOW_CONFLICT_DMA.
+
+Probing problems:  Since the SB16 uses the same IRQ and addresses for
+the different drivers, some of the snd dirvers will not be probed because
+the kernel thinks there is a conflict.  This can be worked-around by
+setting the ALLOW_CONFLICT_IOADDR or ALLOW_CONFLICT_IRQ options.
+
+Warning: Setting the ALLOW_* options will will bypass checks for ALL drivers,
+so be careful when you use them!
+
+		- Jordan Hubbard (jkh@freefall.cdrom.com)
+		- Steven Wallace (swallace@freefall.cdrom.com)
+                - Sujal Patel (smpatel@wam.umd.edu)
diff --git a/sys/i386/isa/sound/sound_pnp.c b/sys/i386/isa/sound/sound_pnp.c
new file mode 100644
index 000000000000..d3d278b9cc10
--- /dev/null
+++ b/sys/i386/isa/sound/sound_pnp.c
@@ -0,0 +1,186 @@
+/*
+ * sound/sound_pnp.c
+ * 
+ * PnP soundcard support (Linux spesific)
+ * 
+ * Copyright by Hannu Savolainen 1995
+ * 
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are
+ * met: 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer. 2.
+ * Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED.  IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE FOR
+ * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+ * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
+ * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ * 
+ */
+#include <i386/isa/sound/sound_config.h>
+
+/*
+ * XXX check what to use in place of CONFIG_PNP
+ */
+#if (NSND > 0) && defined(CONFIG_PNP)
+
+#include <linux/pnp.h>
+
+static struct pnp_sounddev *pnp_devs[20] = { NULL };
+
+static int      max_pnpdevs = 20;
+static int      nr_pnpdevs = 0;
+static int      pnp_sig = 0;
+
+void
+install_pnp_sounddrv(struct pnp_sounddev * drv)
+{
+    if (nr_pnpdevs < max_pnpdevs) {
+	pnp_devs[nr_pnpdevs++] = drv;
+    } else
+	printf("Sound: More than 20 PnP drivers defined\n");
+}
+
+void
+cs4232_pnp(void *parm)
+{
+    struct pnp_dev *dev = (struct pnp_dev *) parm;
+    char           *name;
+
+    int             portmask = 0x00, irqmask = 0x01, dmamask = 0x03;
+    int             opl3_driver, wss_driver;
+
+    printf("CS4232 driver waking up\n");
+
+    if (dev->card && dev->card->name)
+	name = dev->card->name;
+    else
+	name = "PnP WSS";
+
+    if ((wss_driver = sndtable_identify_card("AD1848")))
+	portmask |= 0x01;	/* MSS */
+    else
+	printf("Sound: MSS/WSS device detected but no driver enabled\n");
+
+    if ((opl3_driver = sndtable_identify_card("OPL3")))
+	portmask |= 0x02;	/* OPL3 */
+    else
+	printf("Sound: OPL3/4 device detected but no driver enabled\n");
+
+    printf("WSS driver %d, OPL3 driver %d\n", wss_driver, opl3_driver);
+
+    if (!portmask)		/* No drivers available */
+	return;
+
+    if (!pnp_allocate_device(pnp_sig, dev, portmask, irqmask, dmamask, 0x00))
+	printf("Device activation failed\n");
+    else {
+	struct address_info hw_config;
+	int             wss_base, opl3_base;
+	int             irq;
+	int             dma1, dma2;
+
+	printf("Device activation OK\n");
+	wss_base = pnp_get_port(dev, 0);
+	opl3_base = pnp_get_port(dev, 1);
+	irq = pnp_get_irq(dev, 0);
+	dma1 = pnp_get_dma(dev, 0);
+	dma2 = pnp_get_dma(dev, 1);
+
+	printf("I/O0 %03x\n", wss_base);
+	printf("I/O1 %03x\n", opl3_base);
+	printf("IRQ %d\n", irq);
+	printf("DMA0 %d\n", dma1);
+	printf("DMA1 %d\n", dma2);
+
+	if (opl3_base && opl3_driver) {
+	    hw_config.io_base = opl3_base;
+	    hw_config.irq = 0;
+	    hw_config.dma = -1;
+	    hw_config.dma2 = -1;
+	    hw_config.always_detect = 0;
+	    hw_config.name = "";
+	    hw_config.osp = NULL;
+	    hw_config.card_subtype = 0;
+
+	    if (sndtable_probe(opl3_driver, &hw_config))
+		sndtable_init_card(opl3_driver, &hw_config);
+
+	}
+	if (wss_base && wss_driver) {
+	    hw_config.io_base = wss_base;
+	    hw_config.irq = irq;
+	    hw_config.dma = dma1;
+	    hw_config.dma2 = (dma2 == NO_DMA) ? dma1 : dma2;
+	    hw_config.always_detect = 0;
+	    hw_config.name = name;
+	    hw_config.osp = NULL;
+	    hw_config.card_subtype = 0;
+
+	    if (sndtable_probe(wss_driver, &hw_config))
+		sndtable_init_card(wss_driver, &hw_config);
+
+	}
+    }
+}
+
+static int
+pnp_activate(int id, struct pnp_dev * dev)
+{
+    int             i;
+
+    for (i = 0; i < nr_pnpdevs; i++)
+	if (pnp_devs[i]->id == id) {
+
+	    printf("PnP dev: %08x, %s\n", id, pnp_devid2asc(id));
+
+	    pnp_devs[i]->setup((void *) dev);
+	    return 1;
+	}
+    return 0;
+}
+
+void
+sound_pnp_init(void)
+{
+    static struct pnp_sounddev cs4232_dev =
+	{PNP_DEVID('C', 'S', 'C', 0x0000), cs4232_pnp, "CS4232"};
+
+    struct pnp_dev *dev;
+
+    install_pnp_sounddrv(&cs4232_dev);
+
+    dev = NULL;
+
+    if ((pnp_sig = pnp_connect("sound")) == -1) {
+	printf("Sound: Can't connect to kernel PnP services.\n");
+	return;
+    }
+    while ((dev = pnp_get_next_device(pnp_sig, dev)) != NULL) {
+	if (!pnp_activate(dev->key, dev)) {
+	    /* Scan all compatible devices */
+
+	    int             i;
+
+	    for (i = 0; i < dev->ncompat; i++)
+		if (pnp_activate(dev->compat_keys[i], dev))
+		    break;
+	}
+    }
+}
+
+void
+sound_pnp_disconnect(void)
+{
+    pnp_disconnect(pnp_sig);
+}
+#endif
diff --git a/sys/i386/isa/sound/soundvers.h b/sys/i386/isa/sound/soundvers.h
new file mode 100644
index 000000000000..ca892e8259da
--- /dev/null
+++ b/sys/i386/isa/sound/soundvers.h
@@ -0,0 +1 @@
+#define SOUND_VERSION_STRING "3.0-beta-950506"
diff --git a/sys/i386/isa/sound/sscape.c b/sys/i386/isa/sound/sscape.c
new file mode 100644
index 000000000000..9204b188aab8
--- /dev/null
+++ b/sys/i386/isa/sound/sscape.c
@@ -0,0 +1,1120 @@
+/*
+ * sound/sscape.c
+ *
+ * Low level driver for Ensoniq Soundscape
+ *
+ * Copyright by Hannu Savolainen 1994
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are
+ * met: 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer. 2.
+ * Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED.  IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE FOR
+ * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+ * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
+ * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ *
+ */
+
+#include "sound_config.h"
+
+#if defined(CONFIGURE_SOUNDCARD) && !defined(EXCLUDE_SSCAPE)
+
+#include "coproc.h"
+
+/*
+ *    I/O ports
+ */
+#define MIDI_DATA        0
+#define MIDI_CTRL        1
+#define HOST_CTRL        2
+#define TX_READY		0x02
+#define RX_READY		0x01
+#define HOST_DATA        3
+#define ODIE_ADDR        4
+#define ODIE_DATA        5
+
+/*
+ *    Indirect registers
+ */
+#define GA_INTSTAT_REG   0
+#define GA_INTENA_REG    1
+#define GA_DMAA_REG      2
+#define GA_DMAB_REG      3
+#define GA_INTCFG_REG    4
+#define GA_DMACFG_REG    5
+#define GA_CDCFG_REG     6
+#define GA_SMCFGA_REG    7
+#define GA_SMCFGB_REG    8
+#define GA_HMCTL_REG     9
+
+/*
+ * DMA channel identifiers (A and B)
+ */
+#define SSCAPE_DMA_A 		0
+#define SSCAPE_DMA_B		1
+
+#define PORT(name)	(devc->base+name)
+
+/*
+ * Host commands recognized by the OBP microcode
+ */
+#define CMD_GEN_HOST_ACK        0x80
+#define CMD_GEN_MPU_ACK         0x81
+#define CMD_GET_BOARD_TYPE      0x82
+#define CMD_SET_CONTROL         0x88
+#define CMD_GET_CONTROL         0x89
+#define CMD_SET_MT32            0x96
+#define CMD_GET_MT32            0x97
+#define CMD_SET_EXTMIDI         0x9b
+#define CMD_GET_EXTMIDI         0x9c
+
+#define CMD_ACK			0x80
+
+typedef struct sscape_info
+  {
+    int             base, irq, dma;
+    int             ok;		/* Properly detected */
+    int             dma_allocated;
+    int             my_audiodev;
+    int             opened;
+  }
+
+sscape_info;
+static struct sscape_info dev_info =
+{0};
+static struct sscape_info *devc = &dev_info;
+
+DEFINE_WAIT_QUEUE (sscape_sleeper, sscape_sleep_flag);
+
+#ifdef REVEAL_SPEA
+/* Spea and Reveal have assigned interrupt bits differently than Ensoniq */
+static char     valid_interrupts[] =
+{9, 7, 5, 15};
+
+#else
+static char     valid_interrupts[] =
+{9, 5, 7, 10};
+
+#endif
+
+static unsigned char
+sscape_read (struct sscape_info *devc, int reg)
+{
+  unsigned long   flags;
+  unsigned char   val;
+
+  DISABLE_INTR (flags);
+  OUTB (reg, PORT (ODIE_ADDR));
+  val = INB (PORT (ODIE_DATA));
+  RESTORE_INTR (flags);
+  return val;
+}
+
+static void
+sscape_write (struct sscape_info *devc, int reg, int data)
+{
+  unsigned long   flags;
+
+  DISABLE_INTR (flags);
+  OUTB (reg, PORT (ODIE_ADDR));
+  OUTB (data, PORT (ODIE_DATA));
+  RESTORE_INTR (flags);
+}
+
+static void
+host_open (struct sscape_info *devc)
+{
+  OUTB (0x00, PORT (HOST_CTRL));	/* Put the board to the host mode */
+}
+
+static void
+host_close (struct sscape_info *devc)
+{
+  OUTB (0x03, PORT (HOST_CTRL));	/* Put the board to the MIDI mode */
+}
+
+static int
+host_write (struct sscape_info *devc, unsigned char *data, int count)
+{
+  unsigned long   flags;
+  int             i, timeout;
+
+  DISABLE_INTR (flags);
+
+  /*
+     * Send the command and data bytes
+   */
+
+  for (i = 0; i < count; i++)
+    {
+      for (timeout = 10000; timeout > 0; timeout--)
+	if (INB (PORT (HOST_CTRL)) & TX_READY)
+	  break;
+
+      if (timeout <= 0)
+	{
+	  RESTORE_INTR (flags);
+	  return 0;
+	}
+
+      OUTB (data[i], PORT (HOST_DATA));
+    }
+
+
+  RESTORE_INTR (flags);
+
+  return 1;
+}
+
+static int
+host_read (struct sscape_info *devc)
+{
+  unsigned long   flags;
+  int             timeout;
+  unsigned char   data;
+
+  DISABLE_INTR (flags);
+
+  /*
+     * Read a byte
+   */
+
+  for (timeout = 10000; timeout > 0; timeout--)
+    if (INB (PORT (HOST_CTRL)) & RX_READY)
+      break;
+
+  if (timeout <= 0)
+    {
+      RESTORE_INTR (flags);
+      return -1;
+    }
+
+  data = INB (PORT (HOST_DATA));
+
+  RESTORE_INTR (flags);
+
+  return data;
+}
+
+static int
+host_command1 (struct sscape_info *devc, int cmd)
+{
+  unsigned char   buf[10];
+
+  buf[0] = (unsigned char) (cmd & 0xff);
+
+  return host_write (devc, buf, 1);
+}
+
+static int
+host_command2 (struct sscape_info *devc, int cmd, int parm1)
+{
+  unsigned char   buf[10];
+
+  buf[0] = (unsigned char) (cmd & 0xff);
+  buf[1] = (unsigned char) (parm1 & 0xff);
+
+  return host_write (devc, buf, 2);
+}
+
+static int
+host_command3 (struct sscape_info *devc, int cmd, int parm1, int parm2)
+{
+  unsigned char   buf[10];
+
+  buf[0] = (unsigned char) (cmd & 0xff);
+  buf[1] = (unsigned char) (parm1 & 0xff);
+  buf[2] = (unsigned char) (parm2 & 0xff);
+
+  return host_write (devc, buf, 3);
+}
+
+static void
+set_mt32 (struct sscape_info *devc, int value)
+{
+  host_open (devc);
+  host_command2 (devc, CMD_SET_MT32,
+		 value ? 1 : 0);
+  if (host_read (devc) != CMD_ACK)
+    {
+      printk ("SNDSCAPE: Setting MT32 mode failed\n");
+    }
+  host_close (devc);
+}
+
+static int
+get_board_type (struct sscape_info *devc)
+{
+  int             tmp;
+
+  host_open (devc);
+  if (!host_command1 (devc, CMD_GET_BOARD_TYPE))
+    tmp = -1;
+  else
+    tmp = host_read (devc);
+  host_close (devc);
+  return tmp;
+}
+
+void
+sscapeintr (INT_HANDLER_PARMS (irq, dummy))
+{
+  unsigned char   bits, tmp;
+  static int      debug = 0;
+
+  printk ("sscapeintr(0x%02x)\n", (bits = sscape_read (devc, GA_INTSTAT_REG)));
+  if (SOMEONE_WAITING (sscape_sleeper, sscape_sleep_flag))
+    {
+      WAKE_UP (sscape_sleeper, sscape_sleep_flag);
+    }
+
+  if (bits & 0x02)		/* Host interface interrupt */
+    {
+      printk ("SSCAPE: Host interrupt, data=%02x\n", host_read (devc));
+    }
+
+#if (!defined(EXCLUDE_MPU401) || !defined(EXCLUDE_MPU_EMU)) && !defined(EXCLUDE_MIDI)
+  if (bits & 0x01)
+    {
+      mpuintr (INT_HANDLER_CALL (irq));
+      if (debug++ > 10)		/* Temporary debugging hack */
+	{
+	  sscape_write (devc, GA_INTENA_REG, 0x00);	/* Disable all interrupts */
+	}
+    }
+#endif
+
+  /*
+     * Acknowledge interrupts (toggle the interrupt bits)
+   */
+
+  tmp = sscape_read (devc, GA_INTENA_REG);
+  sscape_write (devc, GA_INTENA_REG, (~bits & 0x0e) | (tmp & 0xf1));
+
+}
+
+static void
+sscape_enable_intr (struct sscape_info *devc, unsigned intr_bits)
+{
+  unsigned char   temp, orig;
+
+  temp = orig = sscape_read (devc, GA_INTENA_REG);
+  temp |= intr_bits;
+  temp |= 0x80;			/* Master IRQ enable */
+
+  if (temp == orig)
+    return;			/* No change */
+
+  sscape_write (devc, GA_INTENA_REG, temp);
+}
+
+static void
+sscape_disable_intr (struct sscape_info *devc, unsigned intr_bits)
+{
+  unsigned char   temp, orig;
+
+  temp = orig = sscape_read (devc, GA_INTENA_REG);
+  temp &= ~intr_bits;
+  if ((temp & ~0x80) == 0x00)
+    temp = 0x00;		/* Master IRQ disable */
+  if (temp == orig)
+    return;			/* No change */
+
+  sscape_write (devc, GA_INTENA_REG, temp);
+}
+
+static void
+do_dma (struct sscape_info *devc, int dma_chan, unsigned long buf, int blk_size, int mode)
+{
+  unsigned char   temp;
+
+  if (dma_chan != SSCAPE_DMA_A)
+    {
+      printk ("SSCAPE: Tried to use DMA channel  != A. Why?\n");
+      return;
+    }
+
+  DMAbuf_start_dma (devc->my_audiodev,
+		    buf,
+		    blk_size, mode);
+
+  temp = devc->dma << 4;	/* Setup DMA channel select bits */
+  if (devc->dma <= 3)
+    temp |= 0x80;		/* 8 bit DMA channel */
+
+  temp |= 1;			/* Trigger DMA */
+  sscape_write (devc, GA_DMAA_REG, temp);
+  temp &= 0xfe;			/* Clear DMA trigger */
+  sscape_write (devc, GA_DMAA_REG, temp);
+}
+
+static int
+verify_mpu (struct sscape_info *devc)
+{
+  /*
+     * The SoundScape board could be in three modes (MPU, 8250 and host).
+     * If the card is not in the MPU mode, enabling the MPU driver will
+     * cause infinite loop (the driver believes that there is always some
+     * received data in the buffer.
+     *
+     * Detect this by looking if there are more than 10 received MIDI bytes
+     * (0x00) in the buffer.
+   */
+
+  int             i;
+
+  for (i = 0; i < 10; i++)
+    {
+      if (INB (devc->base + HOST_CTRL) & 0x80)
+	return 1;
+
+      if (INB (devc->base) != 0x00)
+	return 1;
+    }
+
+  printk ("SoundScape: The device is not in the MPU-401 mode\n");
+  return 0;
+}
+
+static int
+sscape_coproc_open (void *dev_info, int sub_device)
+{
+  if (sub_device == COPR_MIDI)
+    {
+      set_mt32 (devc, 0);
+      if (!verify_mpu (devc))
+	return RET_ERROR (EIO);
+    }
+
+  return 0;
+}
+
+static void
+sscape_coproc_close (void *dev_info, int sub_device)
+{
+  struct sscape_info *devc = dev_info;
+  unsigned long   flags;
+
+  DISABLE_INTR (flags);
+  if (devc->dma_allocated)
+    {
+      sscape_write (devc, GA_DMAA_REG, 0x20);	/* DMA channel disabled */
+#ifndef EXCLUDE_NATIVE_PCM
+      DMAbuf_close_dma (devc->my_audiodev);
+#endif
+      devc->dma_allocated = 0;
+    }
+  RESET_WAIT_QUEUE (sscape_sleeper, sscape_sleep_flag);
+  RESTORE_INTR (flags);
+
+  return;
+}
+
+static void
+sscape_coproc_reset (void *dev_info)
+{
+}
+
+static int
+sscape_download_boot (struct sscape_info *devc, unsigned char *block, int size, int flag)
+{
+  unsigned long   flags;
+  unsigned char   temp;
+  int             done, timeout;
+
+  if (flag & CPF_FIRST)
+    {
+      /*
+         * First block. Have to allocate DMA and to reset the board
+         * before continuing.
+       */
+
+      DISABLE_INTR (flags);
+      if (devc->dma_allocated == 0)
+	{
+#ifndef EXCLUDE_NATIVE_PCM
+	  if (DMAbuf_open_dma (devc->my_audiodev) < 0)
+	    {
+	      RESTORE_INTR (flags);
+	      return 0;
+	    }
+#endif
+
+	  devc->dma_allocated = 1;
+	}
+      RESTORE_INTR (flags);
+
+      sscape_write (devc, GA_HMCTL_REG,
+		    (temp = sscape_read (devc, GA_HMCTL_REG)) & 0x3f);	/*Reset */
+
+      for (timeout = 10000; timeout > 0; timeout--)
+	sscape_read (devc, GA_HMCTL_REG);	/* Delay */
+
+      /* Take board out of reset */
+      sscape_write (devc, GA_HMCTL_REG,
+		    (temp = sscape_read (devc, GA_HMCTL_REG)) | 0x80);
+    }
+
+  /*
+     * Transfer one code block using DMA
+   */
+  memcpy (audio_devs[devc->my_audiodev]->dmap->raw_buf[0], block, size);
+
+  DISABLE_INTR (flags);
+/******** INTERRUPTS DISABLED NOW ********/
+  do_dma (devc, SSCAPE_DMA_A,
+	  audio_devs[devc->my_audiodev]->dmap->raw_buf_phys[0],
+	  size, DMA_MODE_WRITE);
+
+  /*
+   * Wait until transfer completes.
+   */
+  RESET_WAIT_QUEUE (sscape_sleeper, sscape_sleep_flag);
+  done = 0;
+  timeout = 100;
+  while (!done && timeout-- > 0)
+    {
+      int             resid;
+
+      DO_SLEEP (sscape_sleeper, sscape_sleep_flag, 1);
+      clear_dma_ff (devc->dma);
+      if ((resid = get_dma_residue (devc->dma)) == 0)
+	done = 1;
+    }
+
+  RESTORE_INTR (flags);
+  if (!done)
+    return 0;
+
+  if (flag & CPF_LAST)
+    {
+      /*
+         * Take the board out of reset
+       */
+      OUTB (0x00, PORT (HOST_CTRL));
+      OUTB (0x00, PORT (MIDI_CTRL));
+
+      temp = sscape_read (devc, GA_HMCTL_REG);
+      temp |= 0x40;
+      sscape_write (devc, GA_HMCTL_REG, temp);	/* Kickstart the board */
+
+      /*
+         * Wait until the ODB wakes up
+       */
+
+      DISABLE_INTR (flags);
+      done = 0;
+      timeout = 5 * HZ;
+      while (!done && timeout-- > 0)
+	{
+	  DO_SLEEP (sscape_sleeper, sscape_sleep_flag, 1);
+	  if (INB (PORT (HOST_DATA)) == 0xff)	/* OBP startup acknowledge */
+	    done = 1;
+	}
+      RESTORE_INTR (flags);
+      if (!done)
+	{
+	  printk ("SoundScape: The OBP didn't respond after code download\n");
+	  return 0;
+	}
+
+      DISABLE_INTR (flags);
+      done = 0;
+      timeout = 5 * HZ;
+      while (!done && timeout-- > 0)
+	{
+	  DO_SLEEP (sscape_sleeper, sscape_sleep_flag, 1);
+	  if (INB (PORT (HOST_DATA)) == 0xfe)	/* Host startup acknowledge */
+	    done = 1;
+	}
+      RESTORE_INTR (flags);
+      if (!done)
+	{
+	  printk ("SoundScape: OBP Initialization failed.\n");
+	  return 0;
+	}
+
+      printk ("SoundScape board of type %d initialized OK\n",
+	      get_board_type (devc));
+
+#ifdef SSCAPE_DEBUG3
+      /*
+         * Temporary debugging aid. Print contents of the registers after
+         * downloading the code.
+       */
+      {
+	int             i;
+
+	for (i = 0; i < 13; i++)
+	  printk ("I%d = %02x (new value)\n", i, sscape_read (devc, i));
+      }
+#endif
+
+    }
+
+  return 1;
+}
+
+static int
+download_boot_block (void *dev_info, copr_buffer * buf)
+{
+  if (buf->len <= 0 || buf->len > sizeof (buf->data))
+    return RET_ERROR (EINVAL);
+
+  if (!sscape_download_boot (devc, buf->data, buf->len, buf->flags))
+    {
+      printk ("SSCAPE: Unable to load microcode block to the OBP.\n");
+      return RET_ERROR (EIO);
+    }
+
+  return 0;
+}
+
+static int
+sscape_coproc_ioctl (void *dev_info, unsigned int cmd, unsigned int arg, int local)
+{
+
+  switch (cmd)
+    {
+    case SNDCTL_COPR_RESET:
+      sscape_coproc_reset (dev_info);
+      return 0;
+      break;
+
+    case SNDCTL_COPR_LOAD:
+      {
+	copr_buffer    *buf;
+	int             err;
+
+	buf = (copr_buffer *) KERNEL_MALLOC (sizeof (copr_buffer));
+	IOCTL_FROM_USER ((char *) buf, (char *) arg, 0, sizeof (*buf));
+	err = download_boot_block (dev_info, buf);
+	KERNEL_FREE (buf);
+	return err;
+      }
+      break;
+
+    default:
+      return RET_ERROR (EINVAL);
+    }
+
+  return RET_ERROR (EINVAL);
+}
+
+static coproc_operations sscape_coproc_operations =
+{
+  "SoundScape M68K",
+  sscape_coproc_open,
+  sscape_coproc_close,
+  sscape_coproc_ioctl,
+  sscape_coproc_reset,
+  &dev_info
+};
+
+static int
+sscape_audio_open (int dev, int mode)
+{
+  unsigned long   flags;
+  sscape_info    *devc = (sscape_info *) audio_devs[dev]->devc;
+
+  DISABLE_INTR (flags);
+  if (devc->opened)
+    {
+      RESTORE_INTR (flags);
+      return RET_ERROR (EBUSY);
+    }
+
+  if (devc->dma_allocated == 0)
+    {
+      int             err;
+
+      if ((err = DMAbuf_open_dma (devc->my_audiodev)) < 0)
+	{
+	  RESTORE_INTR (flags);
+	  return err;
+	}
+
+      devc->dma_allocated = 1;
+    }
+
+  devc->opened = 1;
+  RESTORE_INTR (flags);
+#ifdef SSCAPE_DEBUG4
+  /*
+     * Temporary debugging aid. Print contents of the registers
+     * when the device is opened.
+   */
+  {
+    int             i;
+
+    for (i = 0; i < 13; i++)
+      printk ("I%d = %02x\n", i, sscape_read (devc, i));
+  }
+#endif
+
+  return 0;
+}
+
+static void
+sscape_audio_close (int dev)
+{
+  unsigned long   flags;
+  sscape_info    *devc = (sscape_info *) audio_devs[dev]->devc;
+
+  DEB (printk ("sscape_audio_close(void)\n"));
+
+  DISABLE_INTR (flags);
+
+  if (devc->dma_allocated)
+    {
+      sscape_write (devc, GA_DMAA_REG, 0x20);	/* DMA channel disabled */
+      DMAbuf_close_dma (dev);
+      devc->dma_allocated = 0;
+    }
+  devc->opened = 0;
+
+  RESTORE_INTR (flags);
+}
+
+static int
+set_speed (sscape_info * devc, int arg)
+{
+  return 8000;
+}
+
+static int
+set_channels (sscape_info * devc, int arg)
+{
+  return 1;
+}
+
+static int
+set_format (sscape_info * devc, int arg)
+{
+  return AFMT_U8;
+}
+
+static int
+sscape_audio_ioctl (int dev, unsigned int cmd, unsigned int arg, int local)
+{
+  sscape_info    *devc = (sscape_info *) audio_devs[dev]->devc;
+
+  switch (cmd)
+    {
+    case SOUND_PCM_WRITE_RATE:
+      if (local)
+	return set_speed (devc, arg);
+      return IOCTL_OUT (arg, set_speed (devc, IOCTL_IN (arg)));
+
+    case SOUND_PCM_READ_RATE:
+      if (local)
+	return 8000;
+      return IOCTL_OUT (arg, 8000);
+
+    case SNDCTL_DSP_STEREO:
+      if (local)
+	return set_channels (devc, arg + 1) - 1;
+      return IOCTL_OUT (arg, set_channels (devc, IOCTL_IN (arg) + 1) - 1);
+
+    case SOUND_PCM_WRITE_CHANNELS:
+      if (local)
+	return set_channels (devc, arg);
+      return IOCTL_OUT (arg, set_channels (devc, IOCTL_IN (arg)));
+
+    case SOUND_PCM_READ_CHANNELS:
+      if (local)
+	return 1;
+      return IOCTL_OUT (arg, 1);
+
+    case SNDCTL_DSP_SAMPLESIZE:
+      if (local)
+	return set_format (devc, arg);
+      return IOCTL_OUT (arg, set_format (devc, IOCTL_IN (arg)));
+
+    case SOUND_PCM_READ_BITS:
+      if (local)
+	return 8;
+      return IOCTL_OUT (arg, 8);
+
+    default:;
+    }
+  return RET_ERROR (EINVAL);
+}
+
+static void
+sscape_audio_output_block (int dev, unsigned long buf, int count, int intrflag, int dma_restart)
+{
+}
+
+static void
+sscape_audio_start_input (int dev, unsigned long buf, int count, int intrflag, int dma_restart)
+{
+}
+
+static int
+sscape_audio_prepare_for_input (int dev, int bsize, int bcount)
+{
+  return 0;
+}
+
+static int
+sscape_audio_prepare_for_output (int dev, int bsize, int bcount)
+{
+  return 0;
+}
+
+static void
+sscape_audio_halt (int dev)
+{
+}
+
+static void
+sscape_audio_reset (int dev)
+{
+  sscape_audio_halt (dev);
+}
+
+static struct audio_operations sscape_audio_operations =
+{
+  "Ensoniq SoundScape channel A",
+  0,
+  AFMT_U8 | AFMT_S16_LE,
+  NULL,
+  sscape_audio_open,
+  sscape_audio_close,
+  sscape_audio_output_block,
+  sscape_audio_start_input,
+  sscape_audio_ioctl,
+  sscape_audio_prepare_for_input,
+  sscape_audio_prepare_for_output,
+  sscape_audio_reset,
+  sscape_audio_halt,
+  NULL,
+  NULL
+};
+
+long
+attach_sscape (long mem_start, struct address_info *hw_config)
+{
+  int             my_dev;
+
+#ifndef SSCAPE_REGS
+  /*
+     * Config register values for Spea/V7 Media FX and Ensoniq S-2000.
+     * These values are card
+     * dependent. If you have another SoundScape based card, you have to
+     * find the correct values. Do the following:
+     *  - Compile this driver with SSCAPE_DEBUG1 defined.
+     *  - Shut down and power off your machine.
+     *  - Boot with DOS so that the SSINIT.EXE program is run.
+     *  - Warm boot to {Linux|SYSV|BSD} and write down the lines displayed
+     *    when detecting the SoundScape.
+     *  - Modify the following list to use the values printed during boot.
+     *    Undefine the SSCAPE_DEBUG1
+   */
+#define SSCAPE_REGS { \
+/* I0 */	0x00, \
+		0xf0, /* Note! Ignored. Set always to 0xf0 */ \
+		0x20, /* Note! Ignored. Set always to 0x20 */ \
+		0x20, /* Note! Ignored. Set always to 0x20 */ \
+		0xf5, /* Ignored */ \
+		0x10, \
+		0x00, \
+		0x2e, /* I7 MEM config A. Likely to vary between models */ \
+		0x00, /* I8 MEM config A. Likely to vary between models */ \
+/* I9 */	0x40 /* Ignored */ \
+	}
+#endif
+
+  unsigned long   flags;
+  static unsigned char regs[10] = SSCAPE_REGS;
+
+  int             i, irq_bits = 0xff;
+
+  if (!probe_sscape (hw_config))
+    return mem_start;
+
+  printk (" <Ensoniq Soundscape>");
+
+  for (i = 0; i < sizeof (valid_interrupts); i++)
+    if (hw_config->irq == valid_interrupts[i])
+      {
+	irq_bits = i;
+	break;
+      }
+
+  if (hw_config->irq > 15 || (regs[4] = irq_bits == 0xff))
+    {
+      printk ("Invalid IRQ%d\n", hw_config->irq);
+      return mem_start;
+    }
+
+  DISABLE_INTR (flags);
+
+  for (i = 1; i < 10; i++)
+    switch (i)
+      {
+      case 1:			/* Host interrupt enable */
+	sscape_write (devc, i, 0xf0);	/* All interrupts enabled */
+	break;
+
+      case 2:			/* DMA A status/trigger register */
+      case 3:			/* DMA B status/trigger register */
+	sscape_write (devc, i, 0x20);	/* DMA channel disabled */
+	break;
+
+      case 4:			/* Host interrupt config reg */
+	sscape_write (devc, i, 0xf0 | (irq_bits << 2) | irq_bits);
+	break;
+
+      case 5:			/* Don't destroy CD-ROM DMA config bits (0xc0) */
+	sscape_write (devc, i, (regs[i] & 0x3f) |
+		      (sscape_read (devc, i) & 0x0c));
+	break;
+
+      case 6:			/* CD-ROM config. Don't touch. */
+	break;
+
+      case 9:			/* Master control reg. Don't modify CR-ROM bits. Disable SB emul */
+	sscape_write (devc, i,
+		      (sscape_read (devc, i) & 0xf0) | 0x00);
+	break;
+
+      default:
+	sscape_write (devc, i, regs[i]);
+      }
+
+  RESTORE_INTR (flags);
+
+#ifdef SSCAPE_DEBUG2
+  /*
+     * Temporary debugging aid. Print contents of the registers after
+     * changing them.
+   */
+  {
+    int             i;
+
+    for (i = 0; i < 13; i++)
+      printk ("I%d = %02x (new value)\n", i, sscape_read (devc, i));
+  }
+#endif
+
+#if !defined(EXCLUDE_MIDI) && !defined(EXCLUDE_MPU_EMU)
+  hw_config->always_detect = 1;
+  if (probe_mpu401 (hw_config))
+    {
+      int             prev_devs;
+
+      prev_devs = num_midis;
+      mem_start = attach_mpu401 (mem_start, hw_config);
+
+      if (num_midis == (prev_devs + 1))		/* The MPU driver installed itself */
+	midi_devs[prev_devs]->coproc = &sscape_coproc_operations;
+    }
+#endif
+
+#ifndef EXCLUDE_NATIVE_PCM
+  /* Not supported yet */
+
+#ifndef EXCLUDE_AUDIO
+  if (num_audiodevs < MAX_AUDIO_DEV)
+    {
+      audio_devs[my_dev = num_audiodevs++] = &sscape_audio_operations;
+      audio_devs[my_dev]->dmachan = hw_config->dma;
+      audio_devs[my_dev]->buffcount = 1;
+      audio_devs[my_dev]->buffsize = DSP_BUFFSIZE;
+      audio_devs[my_dev]->devc = devc;
+      devc->my_audiodev = my_dev;
+      devc->opened = 0;
+      audio_devs[my_dev]->coproc = &sscape_coproc_operations;
+      if (snd_set_irq_handler (hw_config->irq, sscapeintr, "SoundScape") < 0)
+	printk ("Error: Can't allocate IRQ for SoundScape\n");
+
+      sscape_write (devc, GA_INTENA_REG, 0x80);		/* Master IRQ enable */
+    }
+  else
+    printk ("SoundScape: More than enough audio devices detected\n");
+#endif
+#endif
+  devc->ok = 1;
+  return mem_start;
+}
+
+int
+probe_sscape (struct address_info *hw_config)
+{
+  unsigned char   save;
+
+  devc->base = hw_config->io_base;
+  devc->irq = hw_config->irq;
+  devc->dma = hw_config->dma;
+
+  /*
+     * First check that the address register of "ODIE" is
+     * there and that it has exactly 4 writeable bits.
+     * First 4 bits
+   */
+  if ((save = INB (PORT (ODIE_ADDR))) & 0xf0)
+    return 0;
+
+  OUTB (0x00, PORT (ODIE_ADDR));
+  if (INB (PORT (ODIE_ADDR)) != 0x00)
+    return 0;
+
+  OUTB (0xff, PORT (ODIE_ADDR));
+  if (INB (PORT (ODIE_ADDR)) != 0x0f)
+    return 0;
+
+  OUTB (save, PORT (ODIE_ADDR));
+
+  /*
+     * Now verify that some indirect registers return zero on some bits.
+     * This may break the driver with some future revisions of "ODIE" but...
+   */
+
+  if (sscape_read (devc, 0) & 0x0c)
+    return 0;
+
+  if (sscape_read (devc, 1) & 0x0f)
+    return 0;
+
+  if (sscape_read (devc, 5) & 0x0f)
+    return 0;
+
+#ifdef SSCAPE_DEBUG1
+  /*
+     * Temporary debugging aid. Print contents of the registers before
+     * changing them.
+   */
+  {
+    int             i;
+
+    for (i = 0; i < 13; i++)
+      printk ("I%d = %02x (old value)\n", i, sscape_read (devc, i));
+  }
+#endif
+
+  return 1;
+}
+
+int
+probe_ss_ms_sound (struct address_info *hw_config)
+{
+  int             i, irq_bits = 0xff;
+
+  if (devc->ok == 0)
+    {
+      printk ("SoundScape: Invalid initialization order.\n");
+      return 0;
+    }
+
+  for (i = 0; i < sizeof (valid_interrupts); i++)
+    if (hw_config->irq == valid_interrupts[i])
+      {
+	irq_bits = i;
+	break;
+      }
+#ifdef REVEAL_SPEA
+  {
+    int             tmp, status = 0;
+    int             cc;
+
+    if (!((tmp = sscape_read (devc, GA_HMCTL_REG)) & 0xc0))
+      {
+	sscape_write (devc, GA_HMCTL_REG, tmp | 0x80);
+	for (cc = 0; cc < 200000; ++cc)
+	  INB (devc->base + ODIE_ADDR);
+      }
+  }
+#endif
+
+  if (hw_config->irq > 15 || irq_bits == 0xff)
+    {
+      printk ("SoundScape: Invalid MSS IRQ%d\n", hw_config->irq);
+      return 0;
+    }
+
+  return ad1848_detect (hw_config->io_base);
+}
+
+long
+attach_ss_ms_sound (long mem_start, struct address_info *hw_config)
+{
+  /*
+     * This routine configures the SoundScape card for use with the
+     * Win Sound System driver. The AD1848 codec interface uses the CD-ROM
+     * config registers of the "ODIE".
+   */
+
+  int             i, irq_bits = 0xff;
+
+#ifdef EXCLUDE_NATIVE_PCM
+  int             prev_devs = num_audiodevs;
+
+#endif
+
+  /*
+     * Setup the DMA polarity.
+   */
+  sscape_write (devc, GA_DMACFG_REG, 0x50);
+
+  /*
+     * Take the gate-arry off of the DMA channel.
+   */
+  sscape_write (devc, GA_DMAB_REG, 0x20);
+
+  /*
+     * Init the AD1848 (CD-ROM) config reg.
+   */
+
+  for (i = 0; i < sizeof (valid_interrupts); i++)
+    if (hw_config->irq == valid_interrupts[i])
+      {
+	irq_bits = i;
+	break;
+      }
+
+  sscape_write (devc, GA_CDCFG_REG, 0x89 | (hw_config->dma << 4) |
+		(irq_bits << 1));
+
+  if (hw_config->irq == devc->irq)
+    printk ("SoundScape: Warning! The WSS mode can't share IRQ with MIDI\n");
+
+  ad1848_init ("SoundScape", hw_config->io_base,
+	       hw_config->irq,
+	       hw_config->dma,
+	       hw_config->dma);
+
+#ifdef EXCLUDE_NATIVE_PCM
+  if (num_audiodevs == (prev_devs + 1))		/* The AD1848 driver installed itself */
+    audio_devs[prev_devs]->coproc = &sscape_coproc_operations;
+#endif
+#ifdef SSCAPE_DEBUG5
+  /*
+     * Temporary debugging aid. Print contents of the registers
+     * after the AD1848 device has been initialized.
+   */
+  {
+    int             i;
+
+    for (i = 0; i < 13; i++)
+      printk ("I%d = %02x\n", i, sscape_read (devc, i));
+  }
+#endif
+
+  return mem_start;
+}
+
+#endif
diff --git a/sys/i386/isa/sound/trix.c b/sys/i386/isa/sound/trix.c
new file mode 100644
index 000000000000..6e3db0fa7c90
--- /dev/null
+++ b/sys/i386/isa/sound/trix.c
@@ -0,0 +1,323 @@
+/*
+ * sound/trix.c
+ *
+ * Low level driver for the MediaTriX AudioTriX Pro
+ * (MT-0002-PC Control Chip)
+ *
+ * Copyright by Hannu Savolainen 1995
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are
+ * met: 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer. 2.
+ * Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED.  IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE FOR
+ * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+ * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
+ * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ *
+ */
+
+#include "sound_config.h"
+
+#if defined(CONFIGURE_SOUNDCARD) && !defined(EXCLUDE_TRIX)
+
+#ifdef INCLUDE_TRIX_BOOT
+#include "trix_boot.h"
+#endif
+
+static int      kilroy_was_here = 0;	/* Don't detect twice */
+static int      sb_initialized = 0;
+static int      mpu_initialized = 0;
+
+static unsigned char
+trix_read (int addr)
+{
+  OUTB ((unsigned char) addr, 0x390);	/* MT-0002-PC ASIC address */
+  return INB (0x391);		/* MT-0002-PC ASIC data */
+}
+
+static void
+trix_write (int addr, int data)
+{
+  OUTB ((unsigned char) addr, 0x390);	/* MT-0002-PC ASIC address */
+  OUTB ((unsigned char) data, 0x391);	/* MT-0002-PC ASIC data */
+}
+
+static void
+download_boot (int base)
+{
+  int             i = 0, n = sizeof (trix_boot);
+
+  trix_write (0xf8, 0x00);	/* ??????? */
+  OUTB (0x01, base + 6);	/* Clear the internal data pointer */
+  OUTB (0x00, base + 6);	/* Restart */
+
+  /*
+     *  Write the boot code to the RAM upload/download register.
+     *  Each write increments the internal data pointer.
+   */
+  OUTB (0x01, base + 6);	/* Clear the internal data pointer */
+  OUTB (0x1A, 0x390);		/* Select RAM download/upload port */
+
+  for (i = 0; i < n; i++)
+    OUTB (trix_boot[i], 0x391);
+  for (i = n; i < 10016; i++)	/* Clear up to first 16 bytes of data RAM */
+    OUTB (0x00, 0x391);
+  OUTB (0x00, base + 6);	/* Reset */
+  OUTB (0x50, 0x390);		/* ?????? */
+}
+
+static int
+trix_set_wss_port (struct address_info *hw_config)
+{
+  unsigned char   addr_bits;
+
+  if (kilroy_was_here)		/* Already initialized */
+    return 0;
+
+  kilroy_was_here = 1;
+
+  if (trix_read (0x15) != 0x71)	/* No asic signature */
+    return 0;
+
+  /*
+     * Disable separate wave playback and recording DMA channels since
+     * the driver doesn't support duplex mode yet.
+   */
+
+  trix_write (0x13, trix_read (0x13) & ~0x80);
+  trix_write (0x14, trix_read (0x14) & ~0x80);
+
+  /*
+     * Configure the ASIC to place the codec to the proper I/O location
+   */
+
+  switch (hw_config->io_base)
+    {
+    case 0x530:
+      addr_bits = 0;
+      break;
+    case 0x604:
+      addr_bits = 1;
+      break;
+    case 0xE80:
+      addr_bits = 2;
+      break;
+    case 0xF40:
+      addr_bits = 3;
+      break;
+    default:
+      return 0;
+    }
+
+  trix_write (0x19, (trix_read (0x19) & 0x03) | addr_bits);
+  return 1;
+}
+
+/*
+ *    Probe and attach routines for the Windows Sound System mode of
+ *      AudioTriX Pro
+ */
+
+int
+probe_trix_wss (struct address_info *hw_config)
+{
+  /*
+     * Check if the IO port returns valid signature. The original MS Sound
+     * system returns 0x04 while some cards (AudioTriX Pro for example)
+     * return 0x00.
+   */
+  if (!trix_set_wss_port (hw_config))
+    return 0;
+
+  if ((INB (hw_config->io_base + 3) & 0x3f) != 0x00)
+    {
+      DDB (printk ("No MSS signature detected on port 0x%x\n", hw_config->io_base));
+      return 0;
+    }
+
+  if (hw_config->irq > 11)
+    {
+      printk ("AudioTriX: Bad WSS IRQ %d\n", hw_config->irq);
+      return 0;
+    }
+
+  if (hw_config->dma != 0 && hw_config->dma != 1 && hw_config->dma != 3)
+    {
+      printk ("AudioTriX: Bad WSS DMA %d\n", hw_config->dma);
+      return 0;
+    }
+
+  /*
+     * Check that DMA0 is not in use with a 8 bit board.
+   */
+
+  if (hw_config->dma == 0 && INB (hw_config->io_base + 3) & 0x80)
+    {
+      printk ("AudioTriX: Can't use DMA0 with a 8 bit card\n");
+      return 0;
+    }
+
+  if (hw_config->irq > 7 && hw_config->irq != 9 && INB (hw_config->io_base + 3) & 0x80)
+    {
+      printk ("AudioTriX: Can't use IRQ%d with a 8 bit card\n", hw_config->irq);
+      return 0;
+    }
+
+  return ad1848_detect (hw_config->io_base + 4);
+}
+
+long
+attach_trix_wss (long mem_start, struct address_info *hw_config)
+{
+  static unsigned char interrupt_bits[12] =
+  {-1, -1, -1, -1, -1, -1, -1, 0x08, -1, 0x10, 0x18, 0x20};
+  char            bits;
+
+  static unsigned char dma_bits[4] =
+  {1, 2, 0, 3};
+
+  int             config_port = hw_config->io_base + 0, version_port = hw_config->io_base + 3;
+
+  if (!kilroy_was_here)
+    return mem_start;
+
+  /*
+     * Set the IRQ and DMA addresses.
+   */
+
+  bits = interrupt_bits[hw_config->irq];
+  if (bits == -1)
+    return mem_start;
+
+  OUTB (bits | 0x40, config_port);
+  if ((INB (version_port) & 0x40) == 0)
+    printk ("[IRQ Conflict?]");
+
+  OUTB (bits | dma_bits[hw_config->dma], config_port);	/* Write IRQ+DMA setup */
+
+  ad1848_init ("AudioTriX Pro", hw_config->io_base + 4,
+	       hw_config->irq,
+	       hw_config->dma,
+	       hw_config->dma);
+  return mem_start;
+}
+
+int
+probe_trix_sb (struct address_info *hw_config)
+{
+
+  int             tmp;
+  unsigned char   conf;
+  static char     irq_translate[] =
+  {-1, -1, -1, 0, 1, 2, -1, 3};
+
+#ifndef INCLUDE_TRIX_BOOT
+  return 0;			/* No boot code -> no fun */
+#endif
+  if (!kilroy_was_here)
+    return 0;			/* AudioTriX Pro has not been detected earlier */
+
+  if (sb_initialized)
+    return 0;
+
+  if (hw_config->io_base & 0xffffff8f != 0x200)
+    return 0;
+
+  tmp = hw_config->irq;
+  if (tmp > 7)
+    return 0;
+  if (irq_translate[tmp] == -1)
+    return 0;
+
+  tmp = hw_config->dma;
+  if (tmp != 1 && tmp != 3)
+    return 0;
+
+  conf = 0x84;			/* DMA and IRQ enable */
+  conf |= hw_config->io_base & 0x70;	/* I/O address bits */
+  conf |= irq_translate[hw_config->irq];
+  if (hw_config->dma == 3)
+    conf |= 0x08;
+  trix_write (0x1b, conf);
+
+  download_boot (hw_config->io_base);
+  sb_initialized = 1;
+
+  return 1;
+}
+
+long
+attach_trix_sb (long mem_start, struct address_info *hw_config)
+{
+  printk (" <AudioTriX>");
+  return mem_start;
+}
+
+long
+attach_trix_mpu (long mem_start, struct address_info *hw_config)
+{
+  return attach_mpu401 (mem_start, hw_config);
+}
+
+int
+probe_trix_mpu (struct address_info *hw_config)
+{
+  unsigned char   conf;
+  static char     irq_bits[] =
+  {-1, -1, -1, 1, 2, 3, -1, 4, -1, 5};
+
+  if (!kilroy_was_here)
+    return 0;			/* AudioTriX Pro has not been detected earlier */
+
+  if (!sb_initialized)
+    return 0;
+
+  if (mpu_initialized)
+    return 0;
+
+  if (hw_config->irq > 9)
+    return 0;
+
+  if (irq_bits[hw_config->irq] == -1)
+    return 0;
+
+  switch (hw_config->io_base)
+    {
+    case 0x330:
+      conf = 0x00;
+      break;
+    case 0x370:
+      conf = 0x04;
+      break;
+    case 0x3b0:
+      conf = 0x08;
+      break;
+    case 0x3f0:
+      conf = 0x0c;
+      break;
+    default:
+      return 0;			/* Invalid port */
+    }
+
+  conf |= irq_bits[hw_config->irq] << 4;
+
+  trix_write (0x19, (trix_read (0x19) & 0x83) | conf);
+
+  mpu_initialized = 1;
+
+  return probe_mpu401 (hw_config);
+}
+
+#endif
diff --git a/sys/i386/isa/sound/trix_boot.h b/sys/i386/isa/sound/trix_boot.h
new file mode 100644
index 000000000000..d6bd16aff382
--- /dev/null
+++ b/sys/i386/isa/sound/trix_boot.h
@@ -0,0 +1,2488 @@
+/*
+ *	 Computer generated file. Do not edit.
+ */
+static unsigned char trix_boot[] = {
+0x80,0x2e,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x02,0x40,0xe1,0x00,0x00,
+0x00,0x00,0x00,0x02,0x44,0x3f,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,
+0x00,0x00,0x00,0x02,0x44,0xcb,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,
+0x74,0x00,0xa2,0xe3,0x92,0xd3,0xa2,0xe4,0x92,0xd4,0x75,0x81,0x45,0x12,0x3d,0x2e,
+0xd2,0xaa,0xd2,0xaf,0x12,0x04,0xb9,0x7e,0x97,0x7f,0x54,0x90,0x00,0x00,0x02,0x0e,
+0x14,0x80,0xfe,0x12,0x00,0x5c,0x12,0x03,0xc7,0x02,0x00,0x74,0xd0,0x00,0xd0,0x01,
+0xc0,0x02,0xc0,0x03,0xc0,0x01,0xc0,0x00,0x8b,0x82,0x8a,0x83,0xe0,0xcc,0xfa,0xa3,
+0xe0,0xcd,0xfb,0x22,0xd0,0x82,0xd0,0x83,0xf5,0xf0,0xea,0xf0,0xa3,0xeb,0xf0,0xe5,
+0xf0,0x22,0xeb,0x33,0xe4,0x95,0xe0,0xfa,0xed,0x33,0xe4,0x95,0xe0,0xfc,0xec,0x33,
+0xea,0x6c,0x13,0xc0,0xe0,0x12,0x03,0xcf,0xd0,0xf0,0x60,0x0c,0x30,0xf6,0x02,0xaa,
+0xf0,0xea,0x33,0xb3,0xe4,0xfa,0x33,0xfb,0x22,0x7a,0x00,0x7c,0x00,0x12,0x03,0xcf,
+0x60,0x05,0xb3,0xe4,0xfa,0x33,0xfb,0x22,0xeb,0x33,0xe4,0x95,0xe0,0xfa,0xed,0x33,
+0xe4,0x95,0xe0,0xfc,0xec,0x33,0xea,0x6c,0x13,0xc0,0xe0,0x12,0x03,0xcf,0xd0,0xe0,
+0x30,0xe6,0x01,0xfa,0xe4,0xca,0x23,0x54,0x01,0xfb,0x22,0x7a,0x00,0x7c,0x00,0x12,
+0x03,0xcf,0xe4,0xfa,0x33,0xfb,0x22,0xeb,0x33,0xe4,0x95,0xe0,0xfa,0xed,0x33,0xe4,
+0x95,0xe0,0xfc,0xec,0x33,0xea,0x6c,0x13,0xc0,0xe0,0x12,0x03,0xcf,0xd0,0xe0,0x30,
+0xe6,0x01,0xfa,0xe4,0xca,0xf4,0x23,0x54,0x01,0xfb,0x22,0x7a,0x00,0x7c,0x00,0x12,
+0x03,0xcf,0xe4,0xfa,0xb3,0x33,0xfb,0x22,0x7a,0x00,0x7c,0x00,0x12,0x03,0xcf,0x0b,
+0x60,0x04,0xe4,0xfa,0x33,0xfb,0xeb,0x22,0x7a,0x00,0x7c,0x00,0x12,0x03,0xcf,0x0b,
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+0x39,0x3c,0x3c,0x3a,0x3d,0x3e,0x0b,0x62,0x2e,0x64,0x09,0x0e,0x0d,0x0c,0x6b,0x56,
+0x52,0x4e,0x4e,0x4b,0x5e,0x2f,0x75,0x76,0x76,0x76,0x74,0x73,0x77,0x73,0x09,0x37,
+0x7c,0x7b,0x7d,0x70,0x52
+};
+
diff --git a/sys/i386/isa/sound/ultrasound.h b/sys/i386/isa/sound/ultrasound.h
new file mode 100644
index 000000000000..40e2443e6f65
--- /dev/null
+++ b/sys/i386/isa/sound/ultrasound.h
@@ -0,0 +1,121 @@
+#ifndef _ULTRASOUND_H_
+#define _ULTRASOUND_H_
+/*
+ * Copyright by Hannu Savolainen 1993
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ *    notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ *    notice, this list of conditions and the following disclaimer in the
+ *    documentation and/or other materials provided with the distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND
+ * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ * ARE DISCLAIMED.  IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE
+ * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+ * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+ * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ *
+ */
+
+/*
+ *	ultrasound.h - Macros for programming the Gravis Ultrasound
+ *			These macros are extremely device dependent
+ *			and not portable.
+ */
+
+/*
+ *	Private events for Gravis Ultrasound (GUS)
+ *
+ *	Format:
+ *		byte 0 		- SEQ_PRIVATE (0xfe)
+ *		byte 1 		- Synthesizer device number (0-N)
+ *		byte 2 		- Command (see below)
+ *		byte 3 		- Voice number (0-31)
+ *		bytes 4 and 5	- parameter P1 (unsigned short)
+ *		bytes 6 and 7	- parameter P2 (unsigned short)
+ *
+ *	Commands:
+ *		Each command affects one voice defined in byte 3.
+ *		Unused parameters (P1 and/or P2 *MUST* be initialized to zero).
+ *		_GUS_NUMVOICES	- Sets max. number of concurrent voices (P1=14-31, default 16)
+ *		_GUS_VOICESAMPLE- ************ OBSOLETE *************
+ *		_GUS_VOICEON	- Starts voice (P1=voice mode)
+ *		_GUS_VOICEOFF	- Stops voice (no parameters)
+ *		_GUS_VOICEFADE	- Stops the voice smoothly.
+ *		_GUS_VOICEMODE	- Alters the voice mode, don't start or stop voice (P1=voice mode)
+ *		_GUS_VOICEBALA	- Sets voice balence (P1, 0=left, 7=middle and 15=right, default 7)
+ *		_GUS_VOICEFREQ	- Sets voice (sample) playback frequency (P1=Hz)
+ *		_GUS_VOICEVOL	- Sets voice volume (P1=volume, 0xfff=max, 0xeff=half, 0x000=off)
+ *		_GUS_VOICEVOL2	- Sets voice volume (P1=volume, 0xfff=max, 0xeff=half, 0x000=off)
+ *				  (Like GUS_VOICEVOL but doesn't change the hw
+ *				  volume. It just updates volume in the voice table).
+ *
+ *		_GUS_RAMPRANGE	- Sets limits for volume ramping (P1=low volume, P2=high volume)
+ *		_GUS_RAMPRATE	- Sets the speed for volume ramping (P1=scale, P2=rate)
+ *		_GUS_RAMPMODE	- Sets the volume ramping mode (P1=ramping mode)
+ *		_GUS_RAMPON	- Starts volume ramping (no parameters)
+ *		_GUS_RAMPOFF	- Stops volume ramping (no parameters)
+ *		_GUS_VOLUME_SCALE - Changes the volume calculation constants
+ *				  for all voices.
+ */
+
+#define _GUS_NUMVOICES		0x00
+#define _GUS_VOICESAMPLE	0x01	/* OBSOLETE */
+#define _GUS_VOICEON		0x02
+#define _GUS_VOICEOFF		0x03
+#define _GUS_VOICEMODE		0x04
+#define _GUS_VOICEBALA		0x05
+#define _GUS_VOICEFREQ		0x06
+#define _GUS_VOICEVOL		0x07
+#define _GUS_RAMPRANGE		0x08
+#define _GUS_RAMPRATE		0x09
+#define _GUS_RAMPMODE		0x0a
+#define _GUS_RAMPON		0x0b
+#define _GUS_RAMPOFF		0x0c
+#define _GUS_VOICEFADE		0x0d
+#define _GUS_VOLUME_SCALE	0x0e
+#define _GUS_VOICEVOL2		0x0f
+#define _GUS_VOICE_POS		0x10
+
+/*
+ *	GUS API macros
+ */
+
+#define _GUS_CMD(chn, voice, cmd, p1, p2) \
+					{_SEQ_NEEDBUF(8); _seqbuf[_seqbufptr] = SEQ_PRIVATE;\
+					_seqbuf[_seqbufptr+1] = (chn); _seqbuf[_seqbufptr+2] = cmd;\
+					_seqbuf[_seqbufptr+3] = voice;\
+					*(unsigned short*)&_seqbuf[_seqbufptr+4] = p1;\
+					*(unsigned short*)&_seqbuf[_seqbufptr+6] = p2;\
+					_SEQ_ADVBUF(8);}
+
+#define GUS_NUMVOICES(chn, p1)			_GUS_CMD(chn, 0, _GUS_NUMVOICES, (p1), 0)
+#define GUS_VOICESAMPLE(chn, voice, p1)		_GUS_CMD(chn, voice, _GUS_VOICESAMPLE, (p1), 0)	/* OBSOLETE */
+#define GUS_VOICEON(chn, voice, p1)		_GUS_CMD(chn, voice, _GUS_VOICEON, (p1), 0)
+#define GUS_VOICEOFF(chn, voice)		_GUS_CMD(chn, voice, _GUS_VOICEOFF, 0, 0)
+#define GUS_VOICEFADE(chn, voice)		_GUS_CMD(chn, voice, _GUS_VOICEFADE, 0, 0)
+#define GUS_VOICEMODE(chn, voice, p1)		_GUS_CMD(chn, voice, _GUS_VOICEMODE, (p1), 0)
+#define GUS_VOICEBALA(chn, voice, p1)		_GUS_CMD(chn, voice, _GUS_VOICEBALA, (p1), 0)
+#define GUS_VOICEFREQ(chn, voice, p)		_GUS_CMD(chn, voice, _GUS_VOICEFREQ, \
+							(p) & 0xffff, ((p) >> 16) & 0xffff)
+#define GUS_VOICEVOL(chn, voice, p1)		_GUS_CMD(chn, voice, _GUS_VOICEVOL, (p1), 0)
+#define GUS_VOICEVOL2(chn, voice, p1)		_GUS_CMD(chn, voice, _GUS_VOICEVOL2, (p1), 0)
+#define GUS_RAMPRANGE(chn, voice, low, high)	_GUS_CMD(chn, voice, _GUS_RAMPRANGE, (low), (high))
+#define GUS_RAMPRATE(chn, voice, p1, p2)	_GUS_CMD(chn, voice, _GUS_RAMPRATE, (p1), (p2))
+#define GUS_RAMPMODE(chn, voice, p1)		_GUS_CMD(chn, voice, _GUS_RAMPMODE, (p1), 0)
+#define GUS_RAMPON(chn, voice, p1)		_GUS_CMD(chn, voice, _GUS_RAMPON, (p1), 0)
+#define GUS_RAMPOFF(chn, voice)			_GUS_CMD(chn, voice, _GUS_RAMPOFF, 0, 0)
+#define GUS_VOLUME_SCALE(chn, voice, p1, p2)	_GUS_CMD(chn, voice, _GUS_VOLUME_SCALE, (p1), (p2))
+#define GUS_VOICE_POS(chn, voice, p)		_GUS_CMD(chn, voice, _GUS_VOICE_POS, \
+							(p) & 0xffff, ((p) >> 16) & 0xffff)
+
+#endif