more scheduled OSS driver removal

This patch contains the scheduled removal of OSS drivers that:
- have ALSA drivers for the same hardware without known regressions and
- whose Kconfig options have been removed in 2.6.20.
Signed-off-by: Adrian Bunk <bunk@stusta.de>
Acked-by: Jeff Garzik <jeff@garzik.org>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>

Documentation/feature-removal-schedule.txt

@@ -119,13 +119,6 @@ Who:    Adrian Bunk <bunk@stusta.de>
 
 ---------------------------
 
-What:  drivers depending on OSS_OBSOLETE_DRIVER
-When:  options in 2.6.20, code in 2.6.22
-Why:   OSS drivers with ALSA replacements
-Who:   Adrian Bunk <bunk@stusta.de>
-
----------------------------
-
 What:	Unused EXPORT_SYMBOL/EXPORT_SYMBOL_GPL exports
 	(temporary transition config option provided until then)
 	The transition config option will also be removed at the same time.

Documentation/kernel-parameters.txt

@@ -238,16 +238,9 @@ and is between 256 and 4096 characters. It is defined in the file
 			Disable PIN 1 of APIC timer
 			Can be useful to work around chipset bugs.
 
-	ad1816=		[HW,OSS]
-			Format: <io>,<irq>,<dma>,<dma2>
-			See also Documentation/sound/oss/AD1816.
-
 	ad1848=		[HW,OSS]
 			Format: <io>,<irq>,<dma>,<dma2>,<type>
 
-	adlib=		[HW,OSS]
-			Format: <io>
-
 	advansys=	[HW,SCSI]
 			See header of drivers/scsi/advansys.c.
 
@@ -1206,9 +1199,6 @@ and is between 256 and 4096 characters. It is defined in the file
 	opl3=		[HW,OSS]
 			Format: <io>
 
-	opl3sa2=	[HW,OSS] Format:
-			<io>,<irq>,<dma>,<dma2>,<mss_io>,<mpu_io>,<ymode>,<loopback>[,<isapnp>,<multiple]
-
 	oprofile.timer=	[HW]
 			Use timer interrupt instead of performance counters
 

Documentation/sound/oss/AD1816

-Documentation for the AD1816(A) sound driver
-============================================
-
-Installation:
--------------
-
-To get your AD1816(A) based sound card work, you'll have to enable support for
-experimental code ("Prompt for development and/or incomplete code/drivers")
-and isapnp ("Plug and Play support", "ISA Plug and Play support"). Enable
-"Sound card support", "OSS modules support" and "Support for AD1816(A) based
-cards (EXPERIMENTAL)" in the sound configuration menu, too. Now build, install
-and reboot the new kernel as usual.
-
-Features:
----------
-
-List of features supported by this driver:
-- full-duplex support
-- supported audio formats: unsigned 8bit, signed 16bit little endian, 
-                           signed 16bit big endian, µ-law, A-law
-- supported channels: mono and stereo
-- supported recording sources: Master, CD, Line, Line1, Line2, Mic
-- supports phat 3d stereo circuit (Line 3)
-
-
-Supported cards:
-----------------
-
-The following cards are known to work with this driver:
-- Terratec Base 1 
-- Terratec Base 64 
-- HP Kayak 
-- Acer FX-3D
-- SY-1816 
-- Highscreen Sound-Boostar 32 Wave 3D
-- Highscreen Sound-Boostar 16 
-- AVM Apex Pro card 
-- (Aztech SC-16 3D) 
-- (Newcom SC-16 3D) 
-- (Terratec EWS64S) 
-
-Cards listed in brackets are not supported reliable. If you have such a card 
-you should add the extra parameter:
-  options=1
-when loading the ad1816 module via modprobe. 
-
-
-Troubleshooting:
-----------------
-
-First of all you should check, if the driver has been loaded
-properly.
-
-If loading of the driver succeeds, but playback/capture fails, check
-if you used the correct values for irq, dma and dma2 when loading the module.
-If one of them is wrong you usually get the following error message:
-
-Nov  6 17:06:13 tek01 kernel: Sound: DMA (output) timed out - IRQ/DRQ config error?
-
-If playback/capture is too fast or to slow, you should have a look at
-the clock chip of your sound card. The AD1816 was designed for a 33MHz
-oscillator, however most sound card manufacturer use slightly
-different oscillators as they are cheaper than 33MHz oscillators. If
-you have such a card you have to adjust the ad1816_clockfreq parameter
-above. For example: For a card using a 32.875MHz oscillator use
-ad1816_clockfreq=32875 instead of ad1816_clockfreq=33000.
-
-
-Updates, bugfixes and bugreports:
---------------------------------
-
-As the driver is still experimental and under development, you should
-watch out for updates.  Updates of the driver are available on the
-Internet from one of my home pages:
-  http://www.student.informatik.tu-darmstadt.de/~tek/projects/linux.html
-or:
-  http://www.tu-darmstadt.de/~tek01/projects/linux.html
-
-Bugreports, bugfixes and related questions should be sent via E-Mail to:
-  tek@rbg.informatik.tu-darmstadt.de
-
-Thorsten Knabe <tek@rbg.informatik.tu-darmstadt.de>
-Christoph Hellwig <hch@infradead.org>
-	Last modified: 2000/09/20

Documentation/sound/oss/OPL3-SA2

-Documentation for the OPL3-SA2, SA3, and SAx driver (opl3sa2.o)
----------------------------------------------------------------
-
-Scott Murray, scott@spiteful.org
-January 7, 2001
-
-NOTE: All trade-marked terms mentioned below are properties of their
-      respective owners.
-
-
-Supported Devices
------------------
-
-This driver is for PnP soundcards based on the following Yamaha audio
-controller chipsets:
-
-YMF711 aka OPL3-SA2
-YMF715 and YMF719 aka OPL3-SA3
-
-Up until recently (December 2000), I'd thought the 719 to be a
-different chipset, the OPL3-SAx.  After an email exhange with
-Yamaha, however, it turns out that the 719 is just a re-badged
-715, and the chipsets are identical.  The chipset detection code
-has been updated to reflect this.
-
-Anyways, all of these chipsets implement the following devices:
-
-OPL3 FM synthesizer
-Soundblaster Pro
-Microsoft/Windows Sound System
-MPU401 MIDI interface
-
-Note that this driver uses the MSS device, and to my knowledge these
-chipsets enforce an either/or situation with the Soundblaster Pro
-device and the MSS device.  Since the MSS device has better
-capabilities, I have implemented the driver to use it.
-
-
-Mixer Channels
---------------
-
-Older versions of this driver (pre-December 2000) had two mixers,
-an OPL3-SA2 or SA3 mixer and a MSS mixer.  The OPL3-SA[23] mixer
-device contained a superset of mixer channels consisting of its own
-channels and all of the MSS mixer channels.  To simplify the driver
-considerably, and to partition functionality better, the OPL3-SA[23]
-mixer device now contains has its own specific mixer channels.  They
-are:
-
-Volume     - Hardware master volume control
-Bass       - SA3 only, now supports left and right channels
-Treble     - SA3 only, now supports left and right channels
-Microphone - Hardware microphone input volume control
-Digital1   - Yamaha 3D enhancement "Wide" mixer
-
-All other mixer channels (e.g. "PCM", "CD", etc.) now have to be
-controlled via the "MS Sound System (CS4231)" mixer.  To facilitate
-this, the mixer device creation order has been switched so that
-the MSS mixer is created first.  This allows accessing the majority
-of the useful mixer channels even via single mixer-aware tools
-such as "aumix".
-
-
-Plug 'n Play
-------------
-
-In previous kernels (2.2.x), some configuration was required to
-get the driver to talk to the card.  Being the new millennium and
-all, the 2.4.x kernels now support auto-configuration if ISA PnP
-support is configured in.  Theoretically, the driver even supports
-having more than one card in this case.
-
-With the addition of PnP support to the driver, two new parameters
-have been added to control it:
-
-isapnp   - set to 0 to disable ISA PnP card detection
-
-multiple - set to 0 to disable multiple PnP card detection
-
-
-Optional Parameters
--------------------
-
-Recent (December 2000) additions to the driver (based on a patch
-provided by Peter Englmaier) are two new parameters:
-
-ymode -    Set Yamaha 3D enhancement mode:
-           0 = Desktop/Normal  5-12 cm speakers
-           1 = Notebook PC (1) 3 cm speakers
-           2 = Notebook PC (2) 1.5 cm speakers
-           3 = Hi-Fi           16-38 cm speakers
-
-loopback - Set A/D input source. Useful for echo cancellation:
-           0 = Mic Right channel (default)
-           1 = Mono output loopback
-
-The ymode parameter has been tested and does work.  The loopback
-parameter, however, is untested.  Any feedback on its usefulness
-would be appreciated.
-
-
-Manual Configuration
---------------------
-
-If for some reason you decide not to compile ISA PnP support into
-your kernel, or disabled the driver's usage of it by setting the
-isapnp parameter as discussed above, then you will need to do some
-manual configuration.  There are two ways of doing this.  The most
-common is to use the isapnptools package to initialize the card, and
-use the kernel module form of the sound subsystem and sound drivers.
-Alternatively, some BIOS's allow manual configuration of installed
-PnP devices in a BIOS menu, which should allow using the non-modular
-sound drivers, i.e. built into the kernel.
-
-I personally use isapnp and modules, and do not have access to a PnP
-BIOS machine to test.  If you have such a beast, configuring the
-driver to be built into the kernel should just work (thanks to work
-done by David Luyer <luyer@ucs.uwa.edu.au>).  You will still need
-to specify settings, which can be done by adding:
-
-opl3sa2=<io>,<irq>,<dma>,<dma2>,<mssio>,<mpuio>
-
-to the kernel command line.  For example:
-
-opl3sa2=0x370,5,0,1,0x530,0x330
-
-If you are instead using the isapnp tools (as most people have been
-before Linux 2.4.x), follow the directions in their documentation to
-produce a configuration file.  Here is the relevant excerpt I used to
-use for my SA3 card from my isapnp.conf:
-
-(CONFIGURE YMH0800/-1 (LD 0
-
-# NOTE: IO 0 is for the unused SoundBlaster part of the chipset.
-(IO 0 (BASE 0x0220))
-(IO 1 (BASE 0x0530))
-(IO 2 (BASE 0x0388))
-(IO 3 (BASE 0x0330))
-(IO 4 (BASE 0x0370))
-(INT 0 (IRQ 5 (MODE +E)))
-(DMA 0 (CHANNEL 0))
-(DMA 1 (CHANNEL 1))
-
-Here, note that:
-
-Port  Acceptable Range  Purpose
-----  ----------------  -------
-IO 0  0x0220 - 0x0280   SB base address, unused.
-IO 1  0x0530 - 0x0F48   MSS base address
-IO 2  0x0388 - 0x03F8   OPL3 base address
-IO 3  0x0300 - 0x0334   MPU base address
-IO 4  0x0100 - 0x0FFE   card's own base address for its control I/O ports
-
-The IRQ and DMA values can be any that are considered acceptable for a
-MSS.  Assuming you've got isapnp all happy, then you should be able to
-do something like the following (which matches up with the isapnp
-configuration above):
-
-modprobe mpu401
-modprobe ad1848
-modprobe opl3sa2 io=0x370 mss_io=0x530 mpu_io=0x330 irq=5 dma=0 dma2=1
-modprobe opl3 io=0x388
-
-See the section "Automatic Module Loading" below for how to set up
-/etc/modprobe.conf to automate this.
-
-An important thing to remember that the opl3sa2 module's io argument is
-for it's own control port, which handles the card's master mixer for
-volume (on all cards), and bass and treble (on SA3 cards).
-
-
-Troubleshooting
----------------
-
-If all goes well and you see no error messages, you should be able to
-start using the sound capabilities of your system.  If you get an
-error message while trying to insert the opl3sa2 module, then make
-sure that the values of the various arguments match what you specified
-in your isapnp configuration file, and that there is no conflict with
-another device for an I/O port or interrupt.  Checking the contents of
-/proc/ioports and /proc/interrupts can be useful to see if you're
-butting heads with another device.
-
-If you still cannot get the module to load, look at the contents of
-your system log file, usually /var/log/messages.  If you see the
-message "opl3sa2: Unknown Yamaha audio controller version", then you
-have a different chipset version than I've encountered so far.  Look
-for all messages in the log file that start with "opl3sa2: " and see
-if they provide any clues.  If you do not see the chipset version
-message, and none of the other messages present in the system log are
-helpful, email me some details and I'll try my best to help.
-
-
-Automatic Module Loading
-------------------------
-
-Lastly, if you're using modules and want to set up automatic module
-loading with kmod, the kernel module loader, here is the section I
-currently use in my modprobe.conf file:
-
-# Sound
-alias sound-slot-0 opl3sa2
-options opl3sa2 io=0x370 mss_io=0x530 mpu_io=0x330 irq=7 dma=0 dma2=3
-options opl3 io=0x388
-
-That's all it currently takes to get an OPL3-SA3 card working on my
-system.  Once again, if you have any other problems, email me at the
-address listed above.
-
-Scott

Documentation/sound/oss/VIA-chipset

-Running sound cards on VIA chipsets
-
-o	There are problems with VIA chipsets and sound cards that appear to
-	lock the hardware solidly. Test programs under DOS have verified the
-	problem exists on at least some (but apparently not all) VIA boards
-
-o	VIA have so far failed to bother to answer support mail on the subject
-	so if you are a VIA engineer feeling aggrieved as you read this 
-	document go chase your own people. If there is a workaround please
-	let us know so we can implement it.
-
-
-Certain patterns of ISA DMA access used for most PC sound cards cause the
-VIA chipsets to lock up. From the collected reports this appears to cover a
-wide range of boards. Some also lock up with sound cards under Win* as well.
-
-Linux implements a workaround providing your chipset is PCI and you compiled
-with PCI Quirks enabled. If so you will see a message
-	"Activating ISA DMA bug workarounds"
-
-during booting. If you have a VIA PCI chipset that hangs when you use the
-sound and is not generating this message even with PCI quirks enabled
-please report the information to the linux-kernel list (see REPORTING-BUGS).
-
-If you are one of the tiny number of unfortunates with a 486 ISA/VLB VIA
-chipset board you need to do the following to build a special kernel for
-your board
-
-	edit linux/include/asm-i386/dma.h
-
-change
-
-#define isa_dma_bridge_buggy 	(0)
-
-to
-
-#define isa_dma_bridge_buggy 	(1)
-
-and rebuild a kernel without PCI quirk support.
-
-
-Other than this particular glitch the VIA [M]VP* chipsets appear to work
-perfectly with Linux.

Documentation/sound/oss/cs46xx

-
-Documentation for the Cirrus Logic/Crystal SoundFusion cs46xx/cs4280 audio 
-controller chips (2001/05/11)
-
-The cs46xx audio driver supports the DSP line of Cirrus controllers. 
-Specifically, the cs4610, cs4612, cs4614, cs4622, cs4624, cs4630 and the cs4280
-products.  This driver uses the generic ac97_codec driver for AC97 codec
-support.  
-
-
-Features:
-
-Full Duplex Playback/Capture supported from 8k-48k.
-16Bit Signed LE & 8Bit Unsigned, with Mono or Stereo supported.
-
-APM/PM - 2.2.x PM is enabled and functional. APM can also
-be enabled for 2.4.x by modifying the CS46XX_ACPI_SUPPORT macro
-definition.
-
-DMA playback buffer size is configurable from 16k (defaultorder=2) up to 2Meg 
-(defaultorder=11).  DMA capture buffer size is fixed at a single 4k page as
-two 2k fragments.
-
-MMAP seems to work well with QuakeIII, and test XMMS plugin.
-
-Myth2 works, but the polling logic is not fully correct, but is functional.
-
-The 2.4.4-ac6 gameport code in the cs461x joystick driver has been tested 
-with a Microsoft Sidewinder joystick (cs461x.o and sidewinder.o).  This 
-audio driver must be loaded prior to the joystick driver to enable the
-DSP task image supporting the joystick device.
-
-
-Limitations:
-
-SPDIF is currently not supported.
-
-Primary codec support only.  No secondary codec support is implemented.
-
-
-
-NOTES:
-
-Hercules Game Theatre XP - the EGPIO2 pin controls the external Amp,
-and has been tested.
-Module parameter hercules_egpio_disable set to 1, will force a 0 to EGPIODR
-to disable the external amplifier.
-
-VTB Santa Cruz - the GPIO7/GPIO8 on the Secondary Codec control
-the external amplifier for the "back" speakers, since we do not
-support the secondary codec then this external amp is not
-turned on.  The primary codec external amplifier is supported but
-note that the AC97 EAPD bit is inverted logic (amp_voyetra()).
-
-DMA buffer size - there are issues with many of the Linux applications
-concerning the optimal buffer size.  Several applications request a 
-certain fragment size and number and then do not verify that the driver
-has the ability to support the requested configuration.  
-SNDCTL_DSP_SETFRAGMENT ioctl is used to request a fragment size and
-number of fragments.  Some applications exit if an error is returned
-on this particular ioctl. Therefore, in alignment with the other OSS audio 
-drivers, no error is returned when a SETFRAGs IOCTL is received, but the 
-values passed from the app are not used in any buffer calculation 
-(ossfragshift/ossmaxfrags are not used).
-Use the "defaultorder=N" module parameter to change the buffer size if
-you have an application that requires a specific number of fragments
-or a specific buffer size (see below).
-
-Debug Interface
----------------
-There is an ioctl debug interface to allow runtime modification of the 
-debug print levels.  This debug interface code can be disabled from the 
-compilation process with commenting the following define:
-#define CSDEBUG_INTERFACE 1
-There is also a debug print methodolgy to select printf statements from
-different areas of the driver.  A debug print level is also used to allow
-additional printfs to be active.  Comment out the following line in the
-driver to disable compilation of the CS_DBGOUT print statements:
-#define CSDEBUG 1
- 
-Please see the definitions for cs_debuglevel and cs_debugmask for additional
-information on the debug levels and sections.
-
-There is also a csdbg executable to allow runtime manipulation of these 
-parameters.  for a copy email: twoller@crystal.cirrus.com
-
-
-
-MODULE_PARMS definitions
-------------------------
-module_param(defaultorder, ulong, 0);
-defaultorder=N
-where N is a value from 1 to 12
-The buffer order determines the size of the dma buffer for the driver.
-under Linux, a smaller buffer allows more responsiveness from many of the 
-applications (e.g. games).  A larger buffer allows some of the apps (esound) 
-to not underrun the dma buffer as easily.  As default, use 32k (order=3)
-rather than 64k as some of the games work more responsively.
-(2^N) * PAGE_SIZE = allocated buffer size
-
-module_param(cs_debuglevel, ulong, 0644);
-module_param(cs_debugmask, ulong, 0644);
-cs_debuglevel=N
-cs_debugmask=0xMMMMMMMM
-where N is a value from 0 (no debug printfs), to 9 (maximum)
-0xMMMMMMMM is a debug mask corresponding to the CS_xxx bits (see driver source).
-
-module_param(hercules_egpio_disable, ulong, 0);
-hercules_egpio_disable=N
-where N is a 0 (enable egpio), or a 1 (disable egpio support)
-
-module_param(initdelay, ulong, 0);
-initdelay=N
-This value is used to determine the millescond delay during the initialization
-code prior to powering up the PLL.  On laptops this value can be used to
-assist with errors on resume, mostly with IBM laptops.  Basically, if the 
-system is booted under battery power then the mdelay()/udelay() functions fail to 
-properly delay the required time.  Also, if the system is booted under AC power
-and then the power removed, the mdelay()/udelay() functions will not delay properly.
- 
-module_param(powerdown, ulong, 0);
-powerdown=N
-where N is 0 (disable any powerdown of the internal blocks) or 1 (enable powerdown)
-
-
-module_param(external_amp, bool, 0);
-external_amp=1
-if N is set to 1, then force enabling the EAPD support in the primary AC97 codec.
-override the detection logic and force the external amp bit in the AC97 0x26 register
-to be reset (0).  EAPD should be 0 for powerup, and 1 for powerdown.  The VTB Santa Cruz
-card has inverted logic, so there is a special function for these cards.
-
-module_param(thinkpad, bool, 0);
-thinkpad=1
-if N is set to 1, then force enabling the clkrun functionality.
-Currently, when the part is being used, then clkrun is disabled for the entire system,
-but re-enabled when the driver is released or there is no outstanding open count.
-

MAINTAINERS

@@ -194,13 +194,6 @@ M:	jes@trained-monkey.org
 L:	linux-acenic@sunsite.dk
 S:	Maintained
 
-ACI MIXER DRIVER
-P:	Robert Siemer
-M:	Robert.Siemer@gmx.de
-L:	linux-sound@vger.kernel.org
-W:	http://www.stud.uni-karlsruhe.de/~uh1b/
-S:	Maintained
-
 IPS SCSI RAID DRIVER
 P:	Adaptec OEM Raid Solutions
 M:	aacraid@adaptec.com
@@ -272,21 +265,6 @@ L:	linux-acpi@vger.kernel.org
 W:	http://acpi.sourceforge.net/
 S:	Supported
 
-AD1816 SOUND DRIVER
-P:	Thorsten Knabe
-M:	Thorsten Knabe <linux@thorsten-knabe.de>
-W:	http://linux.thorsten-knabe.de
-S:	Maintained
-
-AD1889 SOUND DRIVER
-P:	Kyle McMartin
-M:	kyle@parisc-linux.org
-P:	Thibaut Varene
-M:	T-Bone@parisc-linux.org
-W:	http://wiki.parisc-linux.org/AD1889
-L:	parisc-linux@lists.parisc-linux.org
-S:	Maintained
-
 ADM1025 HARDWARE MONITOR DRIVER
 P:	Jean Delvare
 M:	khali@linux-fr.org
@@ -1404,13 +1382,6 @@ M:	raisch@de.ibm.com
 L:	general@lists.openfabrics.org
 S:	Supported
 
-EMU10K1 SOUND DRIVER
-P:	James Courtier-Dutton
-M:	James@superbug.demon.co.uk
-L:	emu10k1-devel@lists.sourceforge.net
-W:	http://sourceforge.net/projects/emu10k1/
-S:	Maintained
-
 EMULEX LPFC FC SCSI DRIVER
 P:	James Smart
 M:	james.smart@emulex.com
@@ -2730,12 +2701,6 @@ L:	osst-users@lists.sourceforge.net
 L:	linux-scsi@vger.kernel.org
 S:	Maintained
 
-OPL3-SA2, SA3, and SAx DRIVER
-P:	Zwane Mwaikambo
-M:	zwane@arm.linux.org.uk
-L:	linux-sound@vger.kernel.org
-S:	Maintained
-
 OPROFILE
 P:	Philippe Elie
 M:	phil.el@wanadoo.fr

sound/oss/Makefile

@@ -18,20 +18,15 @@ obj-$(CONFIG_SOUND_TRIX)	+= trix.o ad1848.o sb_lib.o uart401.o
 obj-$(CONFIG_SOUND_SSCAPE)	+= sscape.o ad1848.o mpu401.o
 obj-$(CONFIG_SOUND_CS4232)	+= cs4232.o uart401.o
 obj-$(CONFIG_SOUND_MSS)		+= ad1848.o
-obj-$(CONFIG_SOUND_OPL3SA2)	+= opl3sa2.o ad1848.o mpu401.o
 obj-$(CONFIG_SOUND_PAS)		+= pas2.o sb.o sb_lib.o uart401.o
 obj-$(CONFIG_SOUND_SB)		+= sb.o sb_lib.o uart401.o
 obj-$(CONFIG_SOUND_KAHLUA)	+= kahlua.o
 obj-$(CONFIG_SOUND_MPU401)	+= mpu401.o
 obj-$(CONFIG_SOUND_UART6850)	+= uart6850.o
-obj-$(CONFIG_SOUND_ADLIB)	+= adlib_card.o opl3.o
 obj-$(CONFIG_SOUND_YM3812)	+= opl3.o
 obj-$(CONFIG_SOUND_VMIDI)	+= v_midi.o
 obj-$(CONFIG_SOUND_VIDC)	+= vidc_mod.o
 obj-$(CONFIG_SOUND_WAVEARTIST)	+= waveartist.o
-obj-$(CONFIG_SOUND_AD1816)	+= ad1816.o
-obj-$(CONFIG_SOUND_AD1889)	+= ad1889.o ac97_codec.o
-obj-$(CONFIG_SOUND_ACI_MIXER)	+= aci.o
 
 obj-$(CONFIG_SOUND_VIA82CXXX)	+= via82cxxx_audio.o ac97_codec.o
 ifeq ($(CONFIG_MIDI_VIA82CXXX),y)
@@ -40,24 +35,16 @@ endif
 obj-$(CONFIG_SOUND_MSNDCLAS)	+= msnd.o msnd_classic.o
 obj-$(CONFIG_SOUND_MSNDPIN)	+= msnd.o msnd_pinnacle.o
 obj-$(CONFIG_SOUND_VWSND)	+= vwsnd.o
-obj-$(CONFIG_SOUND_NM256)	+= nm256_audio.o ac97.o
 obj-$(CONFIG_SOUND_ICH)		+= i810_audio.o ac97_codec.o
 obj-$(CONFIG_SOUND_ES1371)	+= es1371.o ac97_codec.o
 obj-$(CONFIG_SOUND_VRC5477)	+= nec_vrc5477.o ac97_codec.o
 obj-$(CONFIG_SOUND_AU1550_AC97)	+= au1550_ac97.o ac97_codec.o
-obj-$(CONFIG_SOUND_FUSION)	+= cs46xx.o ac97_codec.o
 obj-$(CONFIG_SOUND_TRIDENT)	+= trident.o ac97_codec.o
-obj-$(CONFIG_SOUND_EMU10K1)	+= ac97_codec.o
 obj-$(CONFIG_SOUND_BCM_CS4297A)	+= swarm_cs4297a.o
 obj-$(CONFIG_SOUND_BT878)	+= btaudio.o
 
 obj-$(CONFIG_SOUND_WM97XX)	+= ac97_plugin_wm97xx.o
 
-ifeq ($(CONFIG_MIDI_EMU10K1),y)
-  obj-$(CONFIG_SOUND_EMU10K1)	+= sound.o
-endif
-
-obj-$(CONFIG_SOUND_EMU10K1)	+= emu10k1/
 obj-$(CONFIG_DMASOUND)		+= dmasound/
 
 # Declare multi-part drivers.

sound/oss/ac97.h

-/*
- * ac97.h 
- * 
- * definitions for the AC97, Intel's Audio Codec 97 Spec
- * also includes support for a generic AC97 interface
- */
-
-#ifndef _AC97_H_
-#define _AC97_H_
-#include "sound_config.h"
-#include "sound_calls.h"
-
-#define  AC97_RESET              0x0000      //
-#define  AC97_MASTER_VOL_STEREO  0x0002      // Line Out
-#define  AC97_HEADPHONE_VOL      0x0004      // 
-#define  AC97_MASTER_VOL_MONO    0x0006      // TAD Output
-#define  AC97_MASTER_TONE        0x0008      //
-#define  AC97_PCBEEP_VOL         0x000a      // none
-#define  AC97_PHONE_VOL          0x000c      // TAD Input (mono)
-#define  AC97_MIC_VOL            0x000e      // MIC Input (mono)
-#define  AC97_LINEIN_VOL         0x0010      // Line Input (stereo)
-#define  AC97_CD_VOL             0x0012      // CD Input (stereo)
-#define  AC97_VIDEO_VOL          0x0014      // none
-#define  AC97_AUX_VOL            0x0016      // Aux Input (stereo)
-#define  AC97_PCMOUT_VOL         0x0018      // Wave Output (stereo)
-#define  AC97_RECORD_SELECT      0x001a      //
-#define  AC97_RECORD_GAIN        0x001c
-#define  AC97_RECORD_GAIN_MIC    0x001e
-#define  AC97_GENERAL_PURPOSE    0x0020
-#define  AC97_3D_CONTROL         0x0022
-#define  AC97_MODEM_RATE         0x0024
-#define  AC97_POWER_CONTROL      0x0026
-
-/* registers 0x0028 - 0x0058 are reserved */
-
-/* AC'97 2.0 */
-#define AC97_EXTENDED_ID	0x0028	/* Extended Audio ID */
-#define AC97_EXTENDED_STATUS	0x002A	/* Extended Audio Status */
-#define AC97_PCM_FRONT_DAC_RATE 0x002C  /* PCM Front DAC Rate */
-#define AC97_PCM_SURR_DAC_RATE  0x002E  /* PCM Surround DAC Rate */
-#define AC97_PCM_LFE_DAC_RATE   0x0030  /* PCM LFE DAC Rate */
-#define AC97_PCM_LR_ADC_RATE	0x0032	/* PCM LR DAC Rate */
-#define AC97_PCM_MIC_ADC_RATE   0x0034  /* PCM MIC ADC Rate */
-#define AC97_CENTER_LFE_MASTER  0x0036  /* Center + LFE Master Volume */
-#define AC97_SURROUND_MASTER    0x0038  /* Surround (Rear) Master Volume */
-#define AC97_RESERVED_3A	0x003A	/* Reserved */
-/* range 0x3c-0x58 - MODEM */
-
-/* registers 0x005a - 0x007a are vendor reserved */
-
-#define  AC97_VENDOR_ID1         0x007c
-#define  AC97_VENDOR_ID2         0x007e
-
-/* volume control bit defines */
-
-#define AC97_MUTE                0x8000
-#define AC97_MICBOOST            0x0040
-#define AC97_LEFTVOL             0x3f00
-#define AC97_RIGHTVOL            0x003f
-
-/* record mux defines */
-
-#define AC97_RECMUX_MIC          0x0000
-#define AC97_RECMUX_CD           0x0101
-#define AC97_RECMUX_VIDEO        0x0202      /* not used */
-#define AC97_RECMUX_AUX          0x0303      
-#define AC97_RECMUX_LINE         0x0404      
-#define AC97_RECMUX_STEREO_MIX   0x0505
-#define AC97_RECMUX_MONO_MIX     0x0606
-#define AC97_RECMUX_PHONE        0x0707
-
-
-/* general purpose register bit defines */
-
-#define AC97_GP_LPBK             0x0080      /* Loopback mode */
-#define AC97_GP_MS               0x0100      /* Mic Select 0=Mic1, 1=Mic2 */
-#define AC97_GP_MIX              0x0200      /* Mono output select 0=Mix, 1=Mic */
-#define AC97_GP_RLBK             0x0400      /* Remote Loopback - Modem line codec */
-#define AC97_GP_LLBK             0x0800      /* Local Loopback - Modem Line codec */
-#define AC97_GP_LD               0x1000      /* Loudness 1=on */
-#define AC97_GP_3D               0x2000      /* 3D Enhancement 1=on */
-#define AC97_GP_ST               0x4000      /* Stereo Enhancement 1=on */
-#define AC97_GP_POP              0x8000      /* Pcm Out Path, 0=pre 3D, 1=post 3D */
-
-
-/* powerdown control and status bit defines */
-
-/* status */
-#define AC97_PWR_MDM             0x0010      /* Modem section ready */
-#define AC97_PWR_REF             0x0008      /* Vref nominal */
-#define AC97_PWR_ANL             0x0004      /* Analog section ready */
-#define AC97_PWR_DAC             0x0002      /* DAC section ready */
-#define AC97_PWR_ADC             0x0001      /* ADC section ready */
-
-/* control */
-#define AC97_PWR_PR0             0x0100      /* ADC and Mux powerdown */
-#define AC97_PWR_PR1             0x0200      /* DAC powerdown */
-#define AC97_PWR_PR2             0x0400      /* Output mixer powerdown (Vref on) */
-#define AC97_PWR_PR3             0x0800      /* Output mixer powerdown (Vref off) */
-#define AC97_PWR_PR4             0x1000      /* AC-link powerdown */
-#define AC97_PWR_PR5             0x2000      /* Internal Clk disable */
-#define AC97_PWR_PR6             0x4000      /* HP amp powerdown */
-#define AC97_PWR_PR7             0x8000      /* Modem off - if supported */
-
-/* useful power states */
-#define AC97_PWR_D0              0x0000      /* everything on */
-#define AC97_PWR_D1              AC97_PWR_PR0|AC97_PWR_PR1|AC97_PWR_PR4
-#define AC97_PWR_D2              AC97_PWR_PR0|AC97_PWR_PR1|AC97_PWR_PR2|AC97_PWR_PR3|AC97_PWR_PR4
-#define AC97_PWR_D3              AC97_PWR_PR0|AC97_PWR_PR1|AC97_PWR_PR2|AC97_PWR_PR3|AC97_PWR_PR4
-#define AC97_PWR_ANLOFF          AC97_PWR_PR2|AC97_PWR_PR3  /* analog section off */
-
-/* Total number of defined registers.  */
-#define AC97_REG_CNT 64
-
-/* Generic AC97 mixer interface. */
-
-/* Structure describing access to the hardware. */
-struct ac97_hwint
-{
-    /* Perform any hardware-specific reset and initialization.  Returns
-     0 on success, or a negative error code.  */
-    int (*reset_device) (struct ac97_hwint *dev);
-
-    /* Returns the contents of the specified register REG.  The caller
-       should check to see if the desired contents are available in
-       the cache first, if applicable. Returns a positive unsigned value
-       representing the contents of the register, or a negative error
-       code.  */
-    int (*read_reg) (struct ac97_hwint *dev, u8 reg);
-
-    /* Writes VALUE to register REG.  Returns 0 on success, or a
-       negative error code.  */
-    int (*write_reg) (struct ac97_hwint *dev, u8 reg, u16 value);
-
-    /* Hardware-specific information. */
-    void *driver_private;
-
-    /* Three OSS masks. */
-    int mixer_devmask;
-    int mixer_stereomask;
-    int mixer_recmask;
-
-    /* The mixer cache. The indices correspond to the AC97 hardware register
-       number / 2, since the register numbers are always an even number.
-
-       Unknown values are set to -1; unsupported registers contain a
-       -2.  */
-    int last_written_mixer_values[AC97_REG_CNT];
-
-    /* A cache of values written via OSS; we need these so we can return
-       the values originally written by the user.
-
-       Why the original user values?  Because the real-world hardware
-       has less precision, and some existing applications assume that
-       they will get back the exact value that they wrote (aumix).
-
-       A -1 value indicates that no value has been written to this mixer
-       channel via OSS.  */
-    int last_written_OSS_values[SOUND_MIXER_NRDEVICES];
-};
-
-/* Values stored in the register cache.  */
-#define AC97_REGVAL_UNKNOWN -1
-#define AC97_REG_UNSUPPORTED -2
-
-struct ac97_mixer_value_list
-{
-    /* Mixer channel to set.  List is terminated by a value of -1.  */
-    int oss_channel;
-    /* The scaled value to set it to; values generally range from 0-100. */
-    union {
-	struct {
-	    u8 left, right;
-	} stereo;
-	u8 mono;
-    } value;
-};
-
-/* Initialize the ac97 mixer by resetting it.  */
-extern int ac97_init (struct ac97_hwint *dev);
-
-/* Sets the mixer DEV to the values in VALUE_LIST.  Returns 0 on success,
-   or a negative error code.  */
-extern int ac97_set_values (struct ac97_hwint *dev,
-			    struct ac97_mixer_value_list *value_list);
-
-/* Writes the specified VALUE to the AC97 register REG in the mixer.
-   Takes care of setting the last-written cache as well.  */
-extern int ac97_put_register (struct ac97_hwint *dev, u8 reg, u16 value);
-
-/* Default ioctl. */
-extern int ac97_mixer_ioctl (struct ac97_hwint *dev, unsigned int cmd,
-			     void __user * arg);
-
-#endif
-
-/*
- * Local variables:
- *  c-basic-offset: 4
- * End:
- */

sound/oss/aci.h

-#ifndef _ACI_H_
-#define _ACI_H_
-
-extern int aci_port;
-extern int aci_version;		/* ACI firmware version	*/
-extern int aci_rw_cmd(int write1, int write2, int write3);
-
-#define aci_indexed_cmd(a, b) aci_rw_cmd(a, b, -1)
-#define aci_write_cmd(a, b)   aci_rw_cmd(a, b, -1)
-#define aci_read_cmd(a)       aci_rw_cmd(a,-1, -1)
-
-#define COMMAND_REGISTER    (aci_port)      /* write register */
-#define STATUS_REGISTER     (aci_port + 1)  /* read register */
-#define BUSY_REGISTER       (aci_port + 2)  /* also used for rds */
-
-#define RDS_REGISTER        BUSY_REGISTER
-
-#define ACI_SET_MUTE          0x0d
-#define ACI_SET_POWERAMP      0x0f
-#define ACI_SET_TUNERMUTE     0xa3
-#define ACI_SET_TUNERMONO     0xa4
-#define ACI_SET_IDE           0xd0
-#define ACI_SET_WSS           0xd1
-#define ACI_SET_SOLOMODE      0xd2
-#define ACI_WRITE_IGAIN       0x03
-#define ACI_WRITE_TUNE        0xa7
-#define ACI_READ_TUNERSTEREO  0xa8
-#define ACI_READ_TUNERSTATION 0xa9
-#define ACI_READ_VERSION      0xf1
-#define ACI_READ_IDCODE       0xf2
-#define ACI_INIT              0xff
-#define ACI_STATUS            0xf0
-#define     ACI_S_GENERAL     0x00
-#define     ACI_S_READ_IGAIN  0x21
-#define ACI_ERROR_OP          0xdf
-
-/*
- * The following macro SCALE can be used to scale one integer volume
- * value into another one using only integer arithmetic. If the input
- * value x is in the range 0 <= x <= xmax, then the result will be in
- * the range 0 <= SCALE(xmax,ymax,x) <= ymax.
- *
- * This macro has for all xmax, ymax > 0 and all 0 <= x <= xmax the
- * following nice properties:
- *
- * - SCALE(xmax,ymax,xmax) = ymax
- * - SCALE(xmax,ymax,0) = 0
- * - SCALE(xmax,ymax,SCALE(ymax,xmax,SCALE(xmax,ymax,x))) = SCALE(xmax,ymax,x)
- *
- * In addition, the rounding error is minimal and nicely distributed.
- * The proofs are left as an exercise to the reader.
- */
-
-#define SCALE(xmax,ymax,x) (((x)*(ymax)+(xmax)/2)/(xmax))
-
-
-#endif  /* _ACI_H_ */

sound/oss/ad1889.h

-#ifndef _AD1889_H_
-#define _AD1889_H_
-
-#define AD_DS_WSMC	0x00	/* DMA input wave/syn mixer control */
-#define AD_DS_RAMC	0x02	/* DMA output resamp/ADC mixer control */
-#define AD_DS_WADA	0x04	/* DMA input wave attenuation */
-#define AD_DS_SYDA	0x06	/* DMA input syn attentuation */
-#define AD_DS_WAS	0x08	/* wave input sample rate */
-#define AD_DS_RES	0x0a	/* resampler output sample rate */
-#define AD_DS_CCS	0x0c	/* chip control/status */
-
-#define AD_DMA_RESBA	0x40	/* RES base addr */
-#define AD_DMA_RESCA	0x44	/* RES current addr */
-#define AD_DMA_RESBC	0x48	/* RES base cnt */
-#define AD_DMA_RESCC	0x4c	/* RES current count */
-#define AD_DMA_ADCBA	0x50	/* ADC */
-#define AD_DMA_ADCCA	0x54
-#define AD_DMA_ADCBC	0x58
-#define AD_DMA_ADCCC	0x5c
-#define AD_DMA_SYNBA	0x60	/* SYN */
-#define AD_DMA_SYNCA	0x64
-#define AD_DMA_SYNBC	0x68
-#define AD_DMA_SYNCC	0x6c
-#define AD_DMA_WAVBA	0x70	/* WAV */
-#define AD_DMA_WAVCA	0x74
-#define AD_DMA_WAVBC	0x78
-#define AD_DMA_WAVCC	0x7c
-#define AD_DMA_RESICC	0x80	/* RES interrupt current count */
-#define AD_DMA_RESIBC	0x84	/* RES interrupt base count */
-#define AD_DMA_ADCICC	0x88	/* ADC interrupt current count */
-#define AD_DMA_ADCIBC	0x8c	/* ADC interrupt base count */
-#define AD_DMA_SYNICC	0x90	/* SYN interrupt current count */
-#define AD_DMA_SYNIBC	0x94	/* SYN interrupt base count */
-#define AD_DMA_WAVICC	0x98	/* WAV interrupt current count */
-#define AD_DMA_WAVIBC	0x9c	/* WAV interrupt base count */
-#define AD_DMA_RESCTRL	0xa0	/* RES PCI control/status */
-#define AD_DMA_ADCCTRL	0xa8	/* ADC PCI control/status */
-#define AD_DMA_SYNCTRL	0xb0	/* SYN PCI control/status */
-#define AD_DMA_WAVCTRL	0xb8	/* WAV PCI control/status */
-#define AD_DMA_DISR	0xc0	/* PCI DMA intr status */
-#define AD_DMA_CHSS	0xc4	/* PCI DMA channel stop status */
-
-#define AD_GPIO_IPC	0xc8	/* IO port ctrl */
-#define AD_GPIO_OP	0xca	/* IO output status */
-#define AD_GPIO_IP	0xcc	/* IO input status */
-
-/* AC97 registers, 0x100 - 0x17f; see ac97.h */
-#define AD_AC97_BASE    0x100   /* ac97 base register */
-#define AD_AC97_ACIC	0x180	/* AC Link interface ctrl */
-
-/* OPL3; BAR1 */
-#define AD_OPL_M0AS	0x00	/* Music0 address/status */
-#define AD_OPL_M0DATA	0x01	/* Music0 data */
-#define AD_OPL_M1A	0x02	/* Music1 address */
-#define AD_OPL_M1DATA	0x03	/* Music1 data */
-/* 0x04-0x0f reserved */
-
-/* MIDI; BAR2 */
-#define AD_MIDA		0x00	/* MIDI data */
-#define AD_MISC		0x01	/* MIDI status/cmd */
-/* 0x02-0xff reserved */
-
-#define AD_DS_IOMEMSIZE	512
-#define AD_OPL_MEMSIZE	16
-#define AD_MIDI_MEMSIZE	16
-
-#define AD_WAV_STATE	0
-#define AD_ADC_STATE	1
-#define AD_MAX_STATES	2
-
-#define DMA_SIZE	(128*1024)
-
-#define DMA_FLAG_MAPPED	1
-
-struct ad1889_dev;
-
-typedef struct ad1889_state {
-	struct ad1889_dev *card;
-
-	mode_t open_mode;
-	struct dmabuf {
-		unsigned int rate;
-		unsigned char fmt, enable;
-
-		/* buf management */
-		size_t rawbuf_size;
-		void *rawbuf;
-		dma_addr_t dma_handle;	/* mapped address */
-		unsigned long dma_len;	/* number of bytes mapped */
-
-		/* indexes into rawbuf for setting up DMA engine */
-		volatile unsigned long rd_ptr, wr_ptr;
-
-		wait_queue_head_t wait; /* to wait for buf servicing */
-
-		/* OSS bits */
-		unsigned int mapped:1;
-		unsigned int ready:1;
-		unsigned int ossfragshift;
-		int ossmaxfrags;
-		unsigned int subdivision;
-	} dmabuf;
-
-	struct mutex mutex;
-} ad1889_state_t;
-
-typedef struct ad1889_dev {
-	void __iomem *regbase;
-	struct pci_dev *pci;
-	
-	spinlock_t lock;
-
-	int dev_audio;
-
-	/* states; one per channel; right now only WAV and ADC */
-	struct ad1889_state state[AD_MAX_STATES];
-
-	/* AC97 codec */
-	struct ac97_codec *ac97_codec;
-	u16 ac97_features;
-
-	/* debugging stuff */
-	struct stats {
-		unsigned int wav_intrs, adc_intrs;
-		unsigned int blocks, underrun, error;
-	} stats;
-} ad1889_dev_t;
-
-typedef struct ad1889_reg {
-	const char *name;
-	int offset;
-	int width;
-} ad1889_reg_t;
-
-#endif

sound/oss/adlib_card.c

-/*
- * sound/oss/adlib_card.c
- *
- * Detection routine for the AdLib card.
- *
- * Copyright (C) by Hannu Savolainen 1993-1997
- *
- * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
- * Version 2 (June 1991). See the "COPYING" file distributed with this software
- * for more info.
- */
-
-#include <linux/module.h>
-#include <linux/init.h>
-
-#include "sound_config.h"
-
-#include "opl3.h"
-
-static void __init attach_adlib_card(struct address_info *hw_config)
-{
-	hw_config->slots[0] = opl3_init(hw_config->io_base, hw_config->osp, THIS_MODULE);
-}
-
-static int __init probe_adlib(struct address_info *hw_config)
-{
-	return opl3_detect(hw_config->io_base, hw_config->osp);
-}
-
-static struct address_info cfg;
-
-static int __initdata io = -1;
-
-module_param(io, int, 0);
-
-static int __init init_adlib(void)
-{
-	cfg.io_base = io;
-
-	if (cfg.io_base == -1) {
-		printk(KERN_ERR "adlib: must specify I/O address.\n");
-		return -EINVAL;
-	}
-	if (probe_adlib(&cfg) == 0)
-		return -ENODEV;
-	attach_adlib_card(&cfg);
-
-	return 0;
-}
-
-static void __exit cleanup_adlib(void)
-{
-	sound_unload_synthdev(cfg.slots[0]);
-	
-}
-
-module_init(init_adlib);
-module_exit(cleanup_adlib);
-
-#ifndef MODULE
-static int __init setup_adlib(char *str)
-{
-        /* io */
-	int ints[2];
-	str = get_options(str, ARRAY_SIZE(ints), ints);
-	
-	io = ints[1];
-
-	return 1;
-}
-__setup("adlib=", setup_adlib);
-#endif
-MODULE_LICENSE("GPL");

sound/oss/emu10k1/Makefile

-# Makefile for Creative Labs EMU10K1 
-#
-# 12 Apr 2000 Rui Sousa
-
-obj-$(CONFIG_SOUND_EMU10K1) += emu10k1.o
-
-emu10k1-objs := audio.o cardmi.o cardmo.o cardwi.o cardwo.o ecard.o \
-                efxmgr.o emuadxmg.o hwaccess.o irqmgr.o main.o midi.o \
-                mixer.o passthrough.o recmgr.o timer.o voicemgr.o
-
-ifdef DEBUG
-    EXTRA_CFLAGS += -DEMU10K1_DEBUG
-endif
-
-ifdef CONFIG_MIDI_EMU10K1
-    EXTRA_CFLAGS += -DEMU10K1_SEQUENCER
-endif

sound/oss/opl3.h

-
-int opl3_detect (int ioaddr, int *osp);
-int opl3_init(int ioaddr, int *osp, struct module *owner);
-
-void enable_opl3_mode(int left, int right, int both);