CVS log for src/sys/dev/isa/sbdsp.c

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Keyword substitution: kv
Default branch: MAIN

Sync with HEAD.

Fix all remaining typos, mainly in comments but also in few definitions and log messages, reported by me in PR kern/54889.
Also fixed some additional typos in comments, found on review of same files or typos.

Sync with HEAD

Clean get_props().
- Make get_props() return AUDIO_PROP_{PLAYBACK,CAPTURE} properly.
  This eliminates need for audio.c to take care of such (old)
  drivers which don't return both of PLAYBACK and CAPTURE.
- All get_props() doesn't need to return AUDIO_PROP_MMAP.
  It is handled in the audio layer now.

Merge isaki-audio2 branch, the overhaul of audio subsystem.
- Interrupt-oriented system rather than thread-oriented.
- Improve stability, quality and performance.
- Split playback and record cleanly.  Improve halfduplex support.
- Many bugs are fixed including deadlocks, resource leaks, abuses, etc.
- Simplify audio filter mechanism.  The encoding/channels/frequency
  conversions are completely handled in the upper layer.  So the hard-
  ware driver only converts its hardware encoding (if necessary).
- audio_hw_if changes:
  - Obsoletes query_encoding and add query_format instead.
  - Obsoletes set_params and add set_format instead.
  - Remove drain, setfd, mappage.
  - The call sequences are changed.
- ioctl AUDIO_GETFD/SETFD, AUDIO_GETCHAN/SETCHAN are obsoleted.
- ioctl AUDIO_{QUERY,GET,SET}FORMAT are introduced.
- cleanup config attributes: au*conv and mulaw.
- All hardware drivers should follow it (I've done as much as possible).

Some file paths are changed:
- dev/audio.c         -> dev/audio/audio.c (rewritten)
- dev/audiovar.h      -> dev/audio/audiovar.h
- dev/audio_dai.h     -> dev/audio/audio_dai.h
- dev/audio_if.h      -> dev/audio/audio_if.h
- dev/audiobell.c     -> dev/audio/audiobell.c
- dev/audiobellvar.h  -> dev/audio/audiobellvar.h
- dev/mulaw.[ch]      -> dev/audio/mulaw.[ch] + dev/audio/alaw.c

Fix typos (made in this branch).

Move dev/audio_if.h -> dev/audio/audio_if.h

Adapt sbdsp to audio2.
- Use new query_format/set_format interfaces.
  The formats are created from sb[pr]modes tables.
- Drop INDEPENDENT property for models proir to SB_16.

Not yet fully adapted to audio2 but make it compilable.
- Drop FULLDUPLEX property.  It is done in a tricky way.

- add or adjust /* FALLTHROUGH */ where appropriate
- add __unreachable() after functions that can return but won't in
  this case, and thus can't be marked __dead easily

Sync with HEAD

Resolve a couple of conflicts (result of the uimin/uimax changes)

Rename min/max -> uimin/uimax for better honesty.

These functions are defined on unsigned int.  The generic name
min/max should not silently truncate to 32 bits on 64-bit systems.
This is purely a name change -- no functional change intended.

HOWEVER!  Some subsystems have

	#define min(a, b)	((a) < (b) ? (a) : (b))
	#define max(a, b)	((a) > (b) ? (a) : (b))

even though our standard name for that is MIN/MAX.  Although these
may invite multiple evaluation bugs, these do _not_ cause integer
truncation.

To avoid `fixing' these cases, I first changed the name in libkern,
and then compile-tested every file where min/max occurred in order to
confirm that it failed -- and thus confirm that nothing shadowed
min/max -- before changing it.

I have left a handful of bootloaders that are too annoying to
compile-test, and some dead code:

cobalt ews4800mips hp300 hppa ia64 luna68k vax
acorn32/if_ie.c (not included in any kernels)
macppc/if_gm.c (superseded by gem(4))

It should be easy to fix the fallout once identified -- this way of
doing things fails safe, and the goal here, after all, is to _avoid_
silent integer truncations, not introduce them.

Maybe one day we can reintroduce min/max as type-generic things that
never silently truncate.  But we should avoid doing that for a while,
so that existing code has a chance to be detected by the compiler for
conversion to uimin/uimax without changing the semantics until we can
properly audit it all.  (Who knows, maybe in some cases integer
truncation is actually intended!)

Pull up following revision(s) (requested by nat in ticket #761):

	sys/dev/isa/sbdsp.c: revision 1.137

Add AUDIO_ENCODING_SLINEAR to supported encodings.  This allows sb, sbpro
& sb16 to configure with 8-bit precision.

Tested in games/mame with a 486DX emulation.

update from HEAD

Add AUDIO_ENCODING_SLINEAR to supported encodings.  This allows sb, sbpro
& sb16 to configure with 8-bit precision.

Tested in games/mame with a 486DX emulation.

Sync with HEAD

KNF. No functional change.

sync with head

Merge jmcneill-audiomp3 branch, which is derived from ad-audiomp2. From
the original ad-audiomp branch notes:

  Add MP locking to the audio drivers.

  Making the audio drivers MP safe is necessary before efforts
  can be made to make the VM system MP safe.

  The are two locks per device instance, an ISR lock and
  a character device lock. The ISR lock replaces calls to
  splaudio()/splx(), and will be held across calls to device
  methods which were called at splaudio() before (e.g.
  trigger_output). The character device lock is held across
  calls to nearly all of the methods, excluding some only
  used for initialization, e.g. get_locks.

Welcome to 5.99.57.

improve the last change, after discussing it with rmind

don't kpause with intr_lock held

port ad-audiomp2 branch changes to 5.99.56
todo:
 - port ad-audiomp2 pcppi changes
 - these drivers need to be fixed:
   - emuxki
   - gcscaudio
   - hdafg
   - uaudio
   - umidi
   - MD audio drivers

Sync with HEAD.

sync with head

A pointer typedef entails trading too much flexibility to declare const
and non-const types, and the kernel uses both const and non-const
PMF qualifiers and device suspensors, so change the pmf_qual_t and
device_suspensor_t typedefs from "pointers to const" to non-pointer,
non-const types.

Expand PMF_FN_* macros.

sync with head

aprint-ify

Checkpoint work in progress.

Checkpoint work on MIDI.

Convert some more drivers.

Sync with HEAD.

sync with head.

sync with head.

Remove clause 3 and 4 from TNF licenses

convert to pmf(9)
ok jmcneill

Sync with HEAD.

Fix fallout from device_t/softc split to mpu(4):
always pass the mpu_softc to mpu_intr(). Found by Takeshi Nakayama.

sync with head.

sync with HEAD

sync with head.

Split device_t and softc for sb(4) and all its attachments.

sync with HEAD

sync with head.

Sync with HEAD.

Follow the merge of pmap.c on i386 and amd64 and move
pmap_init_tmp_pgtbl into arch/x86/x86/pmap.c. Modify the ACPI wakeup
code to restore CR4 before jumping back into kernel space as the large
page option might cover that.

Sync with HEAD.

Sync with head.

machine/{bus,cpu,intr}.h -> sys/{bus,cpu,intr}.h

Add MP locking for audio drivers.

sync with head.

sync with head.

Sync with head.

__unused removal on arguments; approved by core.

sync with head

- sprinkle __unused on function decls.
- fix a couple of unused bugs
- no more -Wno-unused for i386

Add "name" parameter to powerhook_establish, to aid debugging. No objections
on tech-kern@

sync with head

sync with head.

add missing initializers

sync with head.

Merge 2006-05-24 NetBSD-current into the "peter-altq" branch.

sync with head.

integrate kauth.

Sync with head.

sync with head.

Coverity CID 1320: Don't try to use drq16 as an index in an array if it is -1

sync with head.

Merge 2006-03-31 NetBSD-current into the "peter-altq" branch.

Use device_cfdata().

sync with head.

Use the SI capitalization for "Hz", "kHz", and "MHz" in comments and strings.
Add a space between numbers and Hz unit.

merge ktrace-lwp.

sync with -current

sync with head.  xen and whitespace.  xen part is not finished.

Sync with HEAD.

Hi Perry!

nuke trailing whitespace

sync with head.

Sync with HEAD.

Update printf format in debug printf to account for sample_rate now
being an unsigned int.

Sync with HEAD.

ansify and KNF

merge kent-audio1 branch, which introduces audio filter pipeline to the MI
audio framework


Summary of changes:

* struct audio_params
  - remove sw_code, factor, factor_denom, hw_sample_rate,
hw_encoding ,hw_precision, and hw_channels.  Conversion information
is conveyed by stream_filter_list_t.
  - change the type of sample_rate: u_long -> u_int
  - add `validbits,' which represents the valid data size in
precision bits.  It is required in order to distinguish 24/32bit
from 24/24bit or 32/32bit.

* audio_hw_if
 - add two parameters to set_params()
	stream_filter_list_t *pfil, stream_filter_list *rfil
   A HW driver should set filter recipes for requested formats
 - constify audio_params parameters of trigger_output() and
trigger_input().  They represent audio formats for the hardware.
 - make open() and close() optional
 - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters
 to round_blocksize()

* sw_code is replaced with stream_filter_t.
  stream_filer_t converts audio data in an input buffer and writes
into another output buffer unlike sw_code, which converts data in
single buffer.
  converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c,
dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are
reimplemented as stream_filter_t

* MI audio
 - audiosetinfo() builds filter pipelines from stream_filter_list_t
filled by audio_hw_if::set_params()
 - audiosetinfo() returns with EINVAL if mmapped and set_params()
requests filters
 - audio_write(), audio_pint(), and audio_rint() invoke a filter
pipeline.
 - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS,
AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for
AUDIO_GETINFO handle values for a buffer nearest to userland.

* add `struct device *' parameter to ac97_attach()

* all of audio HW drivers follow audio_hw_if and ac97 changes

audio_hw_if change
 * make open() and close() optional
 * add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters
 to round_blocksize()

* remove stream_filter_list_append and stream_filter_list_prepend
* add stream_filter_list_t::{append(), prepend(), set()}

* adopt the filter pipeline framework

* ic/ad1848.c, isa/ess.c, isa/sbdsp.c
  remove direct call of set_params() for the default encoding.
  It should be called by the MI audio framework.

Fix the sync with head I botched.

Sync with HEAD.

Sync with HEAD

Remove junk code.

A few ISA sound drivers like to share dma channels, and hence deferred
isa_dmamap_create() calls to their open/close entrypoints. This worked
with some luck, but broke on i386 when _bus_dmamap_create started
to allocate bounce buffers upfront, since memory below 16M may well
not be available when the sound devices is opened for the Nth time.

To fix this, create a new simple interface, isa_drq_alloc/isa_drq_free,
wrappers around already existing bitmask macros. These are expected
to be used before an isa_dmamap_create call, and after an
isa_dmamap_destroy call, respectively. For the sb and ad1848 drivers,
they're deferred until open/close.

All isa_dmamap_create calls can now use BUS_DMA_ALLOCNOW and be done
at attach time.

DMA, not dma nor Dma.

Add extensible malloc types, adapted from FreeBSD.  This turns
malloc types into a structure, a pointer to which is passed around,
instead of an int constant.  Allow the limit to be adjusted when the
malloc type is defined, or with a function call, as suggested by
Jonathan Stone.

Catch up to -current.

More catchup.

Sync kqueue branch with -current.

Increase max sampling rate on SB16.  It seems almost all models supports it.

Catch up to -current.

add RCSID

Sync with HEAD.

Reset the mixer when resume from suspend.
This is required on some laptops with a buggy APM BIOS.
Code from Inoue Yoshinari <pf5y-inue@asahi-net.or.jp>.

Update thorpej_scsipi to -current as of a month ago
A i386 GENERIC kernel compiles without the siop, ahc and bha drivers
(will be updated later). i386 IDE/ATAPI and ncr work, as well as
sparc/esp_sbus. alpha should work as well (untested yet).
siop, ahc and bha will be updated once I've updated the branch to current
-current, as well as machine-dependant code.

Pull up mmap paddr_t/off_t changes from trunk.

remove include of <vm/vm.h>. <vm/vm.h> -> <uvm/uvm_extern.h>

Change the kernel mmap interface so that the offset to map is an
"off_t" and the return value is a "paddr_t" to allow mappings
at offsets past 2^31 bytes.  Somewhat inspired by FreeBSD, which
only changed the offset to a "vm_offset_t".

Includes updates for the i386, pc532 and sh3 mmmmap from Jason Thorpe.

Add a new function call to the ISA DMA API: isa_dmamaxsize(), which returns
the maximum transfer size for the specified DMA channel.  Make all clients
of ISA DMA use this call to determine their maximum transfer size.

Simplify sbdsp_pause().

Pull up to last week's -current.

Sync with -current

Make little endian and big endian version of all functions
that deal with 16 bit samples.  Let the drivers decide which ones
to use.

Fix one more #ifdef, and now it actually compiles again.  *sigh*

Use NMPU not NMIDI around call to mpu_intr.

Update from trunk.

Move the mpu device declaration to conf/files.
Let the mpu device attach at the sb device, and then midi at the mpu.
Update the mpu at eso attachment.

Make sure we clean up any DMA maps if the open fails.

Several things:
* Rearrange the speed mapping table and adjust the code so that the highest
  rate can actually be used.  Previously we ended up rounding up slightly
  lower speeds and then losing because set_params couldn't set the mode
  back to the current one.
* Allow 260 as a valid I/O address, since the SB1 can be jumpered to this.
* Change the MPU-401 code so it can be attached as a separate device.
  (XXX Really, the SB code ought to just attach a subdevice itself.)
* Do not attach an OPL on the SB1.  Writing to the OPL registers at
  SB_base+0 on this card wedges my machine.
  (XXX Should we access it at 388 instead?  The Creative web site claims
  that this board *does* have an OPL2, but I haven't played with this
  extensively.)
* Allocate the SB DMA channels at open time, rather than attach time, so
  that a single DRQ can be used for multiple cards (if only one is in use
  at a given time).
  (XXX Let me tell you why this is a horrible hack.  If the ISA DMA code
  tries to allocate a bounce buffer after boot time, it will generally fail,
  because there is no contiguous memory below 16MB and the code to allocate
  contiguous pages doesn't know how to move things around.  Now, we
  shouldn't ever be using bounce buffers here, because we use
  isa_dmamem_alloc().  So we just turn off BUS_DMA_ALLOCNOW and we don't
  actually try to.  That's cool, and it even works, but isa_dmamem_alloc()
  has the same problem.  It just happens that we allocate the ring buffers
  at boot time, and whenever we reallocate them (due to the buffer size
  changing), we just deallocated the previous (contiguous) buffer, so we get
  lucky.  This is absolutely disgusting and needs to be fixed.)

Use DMAMODE_LOOPDEMAND.  Tested on a ViBRA16.

spaces -> tabs

Remove some superfluous softc elements.

Add separate halt_output and halt_input routines.

Fix slight ommission in previous.

Pass the direction to the allocm and round_buffersize methods.
Some drivers need this to properly allocate DMAable memory.

If we have an 8-bit DRQ, always pass that to isa_malloc().

Avoid arithmetic on `void *' since that's not ANSI C.

* Redo the way the way the MIDI driver attaches to audio devices.
* Improve the midisyn layer a little.
* Add a driver for the Yamaha OPL[23] FM synths.
  The opl driver is not finished yet; it sounds pretty awful.

For some strange reason I cannot get any FM sound from my SB64 cards,
but a regular SB16 works fine.

Fix typo.

Minor change.

Slow down, Wilbur.

Fix a typo I introduced in the MIDI code.

For models prior to the SB16, there is only one clock divider, so couple the
sample rates.  However, only do this if both play and record are being used
simultaneously; otherwise we can't play 22KHz on the SB1.

Use the new trigger_{in,out}put interface.
Also, even on the SB1, we can leave the DMA controller in auto-initalize
mode and just send a command to the board for each block.  This may help
prevent FIFO underruns.

Add MIDI support.  The MIDI devices can be accessed as ``raw'' through
the /dev/rmidiN devices, or with a sequencer interface via /dev/music.
So far the only supported MIDI device is the MPU401 port on SoundBlaster
(and only on SB on isapnp, since we do not have locators with multiple
values yet).

Adjust for changes to the ISA DMA API.

Use lower case "agc" for consistency.

Pull up 1.78 (manuel)

Fix for the vibra16x from Lennart Augusts: saying that the dsp4.16 was
a SB64 was just a guess. The vibra16x is really a sb16.

Sync w/ trunk (mycroft)

Set the gain to 1/2 by default; 3/4 is too loud and can damage headphones and
speakers.

Make the audio API (almost) SunOS compatible.
The changes is to allow some limited mixer manipulation through
the audio device (instead of the mixer device).
This rendered 4 methods in audio_hw_if unused so garbage collect these.

Make sure the 'input' class is used even for the 1345 mixer.

Update marc-pcmcia branch from trunk.

AudioElinear -> AudioEslinear

AudioC[A-Z] -> AudioC[a-z]

Keep filter when setting source on SB Pro.
Close PR 2593.

Update marc-pcmcia branch from trunk.

Change the way we determine what interrupt routine to use to make
it work when drq8 == drq16.

Update marc-pcmcia branch from trunk.

Substantial rewrite of the SB driver to make it do full duplex on an SB16.
Because of the braindead design of the SB the input and output have to
use different precisions (8 and 16 bits).  It is possible to set the driver
to use 8 bits on both and it will the emulate 8 bits on the output by
expanding it to 16 bits.

Update marc-pcmcia branch from trunk.

Add mute and channels swap to the mixer capabilites.

- Change audio_hw_if a little: set_param now sets the play and record modes
  at the same time instead by using two different calls.  This enables
  it to check more easily if the combined mode is all right.
- Improve the error checking in audio.c.
- Add a new audio property, AUDIO_PROP_INDEPENDENT, show if the
  play and record settings are independent.
- Fix some buglets in audio.c.

Update marc-pcmcia branch from trunk.

Avoid munging audio_defaults.

Round mixer values correctly when setting them.
This is important for the OSS emulation since OSS
uses a different range of values for the mixer.

Audio:
* Make it possible to use software assisted encodings that expand the
  sample size.
* Use 16 bits per sample when emulating mulaw coding on the SB.
* Prepare for SB16 without CT1745 mixer.

Audio changes:
	- Change the way attach and open works to allow multiple audio
	  devices.
	- Split the mulaw.c file into two to avoid dragging in mulaw
	  convertsion when they are not needed.  Add 16 bit alaw/mulaw tables.
	- Change the way audio properties are gotten.
	- Recognize more versions os SoundBlaster.

Audio: Remove the machine dependant code I put in audio.c by mistake.
This adds another method to audio_hw_if.  Also remove a field from
audio_hw_if that was not read-only.

audio: Minor cleanup and enable looping DMA for PSS and PAS.

audio: Simplify handling of AUDIO_SETFD and committing of encoding mode.

Changes to the sudio system:
  - It is now possible to handle devices that want "looping" DMA,
    e.g. the SoundBlaster correctly.  The WSS and SB drivers use this.
    To do this several new methods were introduced in audio_hw_if.
  - Different silence handling (forced by previous change).
  - The audio driver can now be mmap()-ed, but due to problems in
    the VM system only for writing for now.
  - The OSS (Linux) audio emulation takes advantage of some of the
    new features.

In the name of backwards compatibility AUDIO_ENCODING_LINEAR has been
renamed AUDIO_ENCODING_SLINEAR and AUDIO_ENCODING_LINEAR reverts to the
NetBSD 1.2 sematics.  A kernel with COMPAT_12 defined will accept
AUDIO_ENCODING_LINEAR and treat it as before, without COMPAT_12 it
will be rejected.

Fix a braino in record source selection and a pasto in detecting
16 bit mode recording.

Pull thorpej-bus-dma branch into mainline.

Update from trunk.

make Jazz16 16-bit output work properly again.  Still hangs somewhere
(not yet diagnosed) on occasion in high-speed mode, but a ^C will exit
immediately--no 30-second drainage timeout.

Add Alaw encoding emulation.

A cosmetic name change.

Minor corrections to previous.  Notably, make recording work on SBPro again.

Update from trunk.

Redo the way selecting the mode for SB cards is done completely.
It is now table driven since there are so many different variations
of SB cards out there.
Also fix a bug that stopped SB2 and SBPro from working.

Make the audio default parameters into a globally accessible constant.

Redo the SB mixer code.  The driver can now distinguish between the
four different mixer options: none, CT1335, CT1345, and CT1745.
It should now be possible to access most of the SB16 functionality.

Update from trunk.

Mask the level registers as the documentation suggests.

Change lowater calculation a little.
Fix mixer setting bug and add some missing splaudio().
Set SB drq16 to 5 and don't pretend to be SB v1.0.

Back to pretending to be a SB v1.0 again, the mixer code for higher
versions is buggy.

Clean up emulation code, SB, and GUS drivers and add some more encoding
emulation to SB.

Adapt to new software interface to the on-board ISA DMA controller.

Change the interface between high and lowlevel audio drivers again:
Set the encoding parameters slightly differently.
Remove the SW encoding/decodinf functions from this interface
and move them to the audio_parameter struct; this is both more efficient
and flexible.

Convert to new orthogonal audio encoding scheme and implement
some of the new encodings.  The change to ioctl AUDIO_GETENC is
NOT backwards compatible.

update sb_printsc() to match current reality.

XXX Pretend that all SB cards are of model 1.
XXX This way you can at least use SB cards until the real fixes go in.

Change the interface between the high and low level audio drivers
so that all audio encoding parameters are set at once.  This simplifies
the interface and make error checking easier.

partial fix for PR 3385:  correctly identify JAZZ-16 when AUDIO_DEBUG is
enabled

Use autoinitialize DMA mode on the SB2 and later.  Partly from Torsten Duwe.

Make sure we round the block size to a multiple of the sample size.

Remove bogus block rounding code.  If the application sets a
block size explicitly, it had better know what it's doing.  From
PR 2587, by Robert Baron.

Turn off a debugging printf().

Add 16-bit DRQ support on PnP, and enable 16-bit output.

Combine set_encoding and set_precision into a single set_format interface.

Don't share the silence block between devices.  Make silence filling work for
more encodings, and make it device-independent.  From Lennart Augustsson, in
PR kern/3305.

Merge in changes from The Trunk

Fix bug introduced in the conversion to bus_*(). Don't always say that
we are a JASS16. From enami tsugutomo.

Update from trunk.

- convert to bus_space_*
- split attachment code to sb_isa.c

Move 'not finished' printf within AUDIO_DEBUG ifdefs, it should normally
be silent about this and caused confusion with stray interrupts.

fix PR 2851: make it compile under -Wformat

backout kprintf changes

printf -> kprintf, sprintf -> ksprintf

Use intr.h.

- prototype fixes

add MediaVision Jazz16 chipset support.  It must be configured with
'flags 1' on the sb? kernel configuration file line (because it frobs a
noncontiguous IO port to configure the Jazz16 extensions).

Also, remove static sb_device structure and fill in user's buffer on
each request.

Add support to the ISA DMA framework for auto-initialize mode.
Add experimental SB16 code, disabled for now.

Do not allow the gain to be changed on an original SoundBlaster.

Fix mono volume adjustment.

XXX This is a kluge.
Check to see if the DMA transfer is finished; if not, don't do interrupt
processing.

Declare a variable...

Clean this up a bit.

Add bass/treble input filter adjustment controls to SB (Pro) driver

Various:
* Snap the sample rate when setting it, and remember only the time constant.
* Set the time constant when changing between play/record.
* Always return the actual sample rate with AUDIO_GETINFO.

Various, including:
* Add more delays while writing registers.
* Replace sc_dma{in,out}_inprogress with sc_dmadir.
* Eliminate the need for sc_locked.
* Add more DPRINTF()s.

Convert IRQ, DRQ, and port numbers to int.

GUS audio driver from John Kohl.

Cleanup

Remove  strings

Fix holes in device index space. PR#1028

Fix nits from Mike Long <mike.long@analog.com> PR#996

Fix obvious bug in reading from SB_MIC_PORT.

update for new locations

clean up several ISA device interfaces: autoconfiguration, header
inclusion, and interrupt configuration.  more work still needs to be done,
but it's getting better...

Use void* rather than caddr_t in many places.

Fixes defect port-i386/860 from John Woods (jfw@jfwhome.funhouse.com) with
supplied patch. Kernel will not link with only ProAudio Spectrum driver
specified.

Fix SB8 playback. From David Gluss <david@pure.com>

Cleanup for SB16 (From Jan Sparud)

SoundBlaster driver for SB, SB16 and SBPRO