[PATCH] The scheduled removal of some OSS drivers

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 2.6.17.

[michal.k.k.piotrowski@gmail.com: build fix]
Signed-off-by: Adrian Bunk <bunk@stusta.de>
Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski@gmail.com>
Cc: David Woodhouse <dwmw2@infradead.org>
Signed-off-by: Andrew Morton <akpm@osdl.org>
Signed-off-by: Linus Torvalds <torvalds@osdl.org>

Documentation/feature-removal-schedule.txt

@@ -29,14 +29,6 @@ Who:	Adrian Bunk <bunk@stusta.de>
 
 ---------------------------
 
-What:	drivers that were depending on OBSOLETE_OSS_DRIVER
-        (config options already removed)
-When:	before 2.6.19
-Why:	OSS drivers with ALSA replacements
-Who:	Adrian Bunk <bunk@stusta.de>
-
----------------------------
-
 What:	raw1394: requests of type RAW1394_REQ_ISO_SEND, RAW1394_REQ_ISO_LISTEN
 When:	November 2006
 Why:	Deprecated in favour of the new ioctl-based rawiso interface, which is

Documentation/kernel-parameters.txt

@@ -289,9 +289,6 @@ and is between 256 and 4096 characters. It is defined in the file
 
 	autotest	[IA64]
 
-	awe=		[HW,OSS] AWE32/SB32/AWE64 wave table synth
-			Format: <io>,<memsize>,<isapnp>
-
 	aztcd=		[HW,CD] Aztech CD268 CDROM driver
 			Format: <io>,0x79 (?)
 
@@ -536,10 +533,6 @@ and is between 256 and 4096 characters. It is defined in the file
 			Default value is 0.
 			Value can be changed at runtime via /selinux/enforce.
 
-	es1370=		[HW,OSS]
-			Format: <lineout>[,<micbias>]
-			See also header of sound/oss/es1370.c.
-
 	es1371=		[HW,OSS]
 			Format: <spdif>,[<nomix>,[<amplifier>]]
 			See also header of sound/oss/es1371.c.
@@ -580,9 +573,6 @@ and is between 256 and 4096 characters. It is defined in the file
 	gscd=		[HW,CD]
 			Format: <io>
 
-	gus=		[HW,OSS]
-			Format: <io>,<irq>,<dma>,<dma16>
-
 	gvp11=		[HW,SCSI]
 
 	hashdist=	[KNL,NUMA] Large hashes allocated during boot
@@ -841,12 +831,6 @@ and is between 256 and 4096 characters. It is defined in the file
 			(machvec) in a generic kernel.
 			Example: machvec=hpzx1_swiotlb
 
-	mad16=		[HW,OSS] Format:
-			<io>,<irq>,<dma>,<dma16>,<mpu_io>,<mpu_irq>,<joystick>
-
-	maui=		[HW,OSS]
-			Format: <io>,<irq>
-
 	max_loop=	[LOOP] Maximum number of loopback devices that can
 			be mounted
 			Format: <1-256>
@@ -1114,9 +1098,6 @@ and is between 256 and 4096 characters. It is defined in the file
 	opl3=		[HW,OSS]
 			Format: <io>
 
-	opl3sa=		[HW,OSS]
-			Format: <io>,<irq>,<dma>,<dma2>,<mpu_io>,<mpu_irq>
-
 	opl3sa2=	[HW,OSS] Format:
 			<io>,<irq>,<dma>,<dma2>,<mss_io>,<mpu_io>,<ymode>,<loopback>[,<isapnp>,<multiple]
 
@@ -1451,9 +1432,6 @@ and is between 256 and 4096 characters. It is defined in the file
 
 	sg_def_reserved_size=	[SCSI]
 
-	sgalaxy=	[HW,OSS]
-			Format: <io>,<irq>,<dma>,<dma2>,<sgbase>
-
 	shapers=	[NET]
 			Maximal number of shapers.
 
@@ -1594,9 +1572,6 @@ and is between 256 and 4096 characters. It is defined in the file
 
 	snd-ymfpci=	[HW,ALSA]
 
-	sonicvibes=	[HW,OSS]
-			Format: <reverb>
-
 	sonycd535=	[HW,CD]
 			Format: <io>[,<irq>]
 

Documentation/sound/oss/AWE32

-	Installing and using Creative AWE midi sound under Linux.
-
-This documentation is devoted to the Creative Sound Blaster AWE32, AWE64 and 
-SB32.
-
-1) Make sure you have an ORIGINAL Creative SB32, AWE32 or AWE64 card. This
-   is important, because the driver works only with real Creative cards.
-
-2) The first thing you need to do is re-compile your kernel with support for
-   your sound card. Run your favourite tool to configure the kernel and when
-   you get to the "Sound" menu you should enable support for the following:
-
-   Sound card support,
-   OSS sound modules,
-   100% Sound Blaster compatibles (SB16/32/64, ESS, Jazz16) support,
-   AWE32 synth
-
-   If your card is "Plug and Play" you will also need to enable these two
-   options, found under the "Plug and Play configuration" menu:
-
-   Plug and Play support
-   ISA Plug and Play support
-
-   Now compile and install the kernel in normal fashion. If you don't know
-   how to do this you can find instructions for this in the README file
-   located in the root directory of the kernel source.
-
-3) Before you can start playing midi files you will have to load a sound
-   bank file. The utility needed for doing this is called "sfxload", and it
-   is one of the utilities found in a package called "awesfx". If this
-   package is not available in your distribution you can download the AWE
-   snapshot from Creative Labs Open Source website:
-
-   http://www.opensource.creative.com/snapshot.html
-
-   Once you have unpacked the AWE snapshot you will see a "awesfx"
-   directory. Follow the instructions in awesfx/docs/INSTALL to install the
-   utilities in this package. After doing this, sfxload should be installed
-   as:
-
-   /usr/local/bin/sfxload
-
-   To enable AWE general midi synthesis you should also get the sound bank
-   file for general midi from:
-
-   http://members.xoom.com/yar/synthgm.sbk.gz
-
-   Copy it to a directory of your choice, and unpack it there.
-
-4) Edit /etc/modprobe.conf, and insert the following lines at the end of the
-   file:
-
-  alias sound-slot-0 sb
-  alias sound-service-0-1 awe_wave
-  install awe_wave /sbin/modprobe --first-time -i awe_wave && /usr/local/bin/sfxload PATH_TO_SOUND_BANK_FILE
-
-  You will of course have to change "PATH_TO_SOUND_BANK_FILE" to the full
-  path of the sound bank file. That will enable the Sound Blaster and AWE
-  wave synthesis. To play midi files you should get one of these programs if
-  you don't already have them:
-
-  Playmidi:			http://playmidi.openprojects.net
-
-  AWEMidi Player (drvmidi)  	Included in the previously mentioned AWE
-  				snapshot.
-
-  You will probably have to pass the "-e" switch to playmidi to have it use
-  your midi device. drvmidi should work without switches.
-
-  If something goes wrong please e-mail me. All comments and suggestions are
-  welcome.
-
-		    Yaroslav Rosomakho (alons55@dialup.ptt.ru)
-			    http://www.yar.opennet.ru
-
-Last Updated: Feb 3 2001

Documentation/sound/oss/CMI8338

-Audio driver for CM8338/CM8738 chips by Chen-Li Tien
-
-
-HARDWARE SUPPORTED
-================================================================================
-C-Media CMI8338
-C-Media CMI8738
-On-board C-Media chips
-
-
-STEPS TO BUILD DRIVER
-================================================================================
-
-  1. Backup the Config.in and Makefile in the sound driver directory
-     (/usr/src/linux/driver/sound).
-     The Configure.help provide help when you config driver in step
-     4, please backup the original one (/usr/src/linux/Document) and
-     copy this file.
-     The cmpci is document for the driver in detail, please copy it
-     to /usr/src/linux/Document/sound so you can refer it. Backup if
-     there is already one.
-
-  2. Extract the tar file by 'tar xvzf cmpci-xx.tar.gz' in the above
-     directory.
-
-  3. Change directory to /usr/src/linux
-
-  4. Config cm8338 driver by 'make menuconfig', 'make config' or
-     'make xconfig' command.
-
-  5. Please select Sound Card (CONFIG_SOUND=m) support and CMPCI
-     driver (CONFIG_SOUND_CMPCI=m) as modules. Resident mode not tested.
-     For driver option, please refer 'DRIVER PARAMETER'
-
-  6. Compile the kernel if necessary.
-
-  7. Compile the modules by 'make modules'.
-
-  8. Install the modules by 'make modules_install'
-
-
-INSTALL DRIVER
-================================================================================
-
-  1. Before first time to run the driver, create module dependency by
-     'depmod -a'
-
-  2. To install the driver manually, enter 'modprobe cmpci'.
-
-  3. Driver installation for various distributions:
-
-    a. Slackware 4.0
-       Add the 'modprobe cmpci' command in your /etc/rc.d/rc.modules
-       file.so you can start the driver automatically each time booting.
-
-    b. Caldera OpenLinux 2.2
-       Use LISA to load the cmpci module.
-
-    c. RedHat 6.0 and S.u.S.E. 6.1
-       Add following command in /etc/conf.modules:
-
-       alias sound cmpci
-
-	also visit http://www.cmedia.com.tw for installation instruction.
-
-DRIVER PARAMETER
-================================================================================
-
-  Some functions for the cm8738 can be configured in Kernel Configuration
-  or modules parameters. Set these parameters to 1 to enable.
-
-  mpuio:	I/O ports base for MPU-401, 0 if disabled.
-  fmio:		I/O ports base for OPL-3, 0 if disabled.
-  spdif_inverse:Inverse the S/PDIF-in signal, this depends on your
-		CD-ROM or DVD-ROM.
-  spdif_loop:   Enable S/PDIF loop, this route S/PDIF-in to S/PDIF-out
-                directly.
-  speakers:     Number of speakers used.
-  use_line_as_rear:Enable this if you want to use line-in as
-                rear-out.
-  use_line_as_bass:Enable this if you want to use line-in as
-                bass-out.
-  joystick:	Enable joystick. You will need to install Linux joystick
-		driver.
-

Documentation/sound/oss/INSTALL.awe

-================================================================
-	INSTALLATION OF AWE32 SOUND DRIVER FOR LINUX
-	Takashi Iwai	<iwai@ww.uni-erlangen.de>
-================================================================
-
-----------------------------------------------------------------
-* Attention to SB-PnP Card Users
-
-If you're using PnP cards, the initialization of PnP is required
-before loading this driver.  You have now three options:
-  1. Use isapnptools.
-  2. Use in-kernel isapnp support.
-  3. Initialize PnP on DOS/Windows, then boot linux by loadlin.
-In this document, only the case 1 case is treated.
-
-----------------------------------------------------------------
-* Installation on Red Hat 5.0 Sound Driver
-
-Please use install-rh.sh under RedHat5.0 directory.
-DO NOT USE install.sh below.
-See INSTALL.RH for more details.
-
-----------------------------------------------------------------
-* Installation/Update by Shell Script
-
-  1. Become root
-
-	% su
-
-  2. If you have never configured the kernel tree yet, run make config
-    once (to make dependencies and symlinks).
-
-	# cd /usr/src/linux
-	# make xconfig
-    
-  3. Run install.sh script
-
-	# sh ./install.sh
-
-  4. Configure your kernel
-
-	(for Linux 2.[01].x user)
-	# cd /usr/src/linux
-	# make xconfig (or make menuconfig)
-
-	(for Linux 1.2.x user)
-	# cd /usr/src/linux
-	# make config
-
-    Answer YES to both "lowlevel drivers" and "AWE32 wave synth" items 
-    in Sound menu.  ("lowlevel drivers" will appear only in 2.x
-    kernel.)
-
-  5. Make your kernel (and modules), and install them as usual.
-
-	5a. make kernel image
-		# make zImage
-
-	5b. make modules and install them
-		# make modules && make modules_install
-
-	5c. If you're using lilo, copy the kernel image and run lilo.
-	    Otherwise, copy the kernel image to suitable directory or
-	    media for your system.
-
-  6. Reboot the kernel if necessary.
-	- If you updated only the modules, you don't have to reboot
-	  the system.  Just remove the old sound modules here.
-		in 
-		# rmmod sound.o		(linux-2.0 or OSS/Free)
-		# rmmod awe_wave.o	(linux-2.1)
-
-  7. If your AWE card is a PnP and not initialized yet, you'll have to
-    do it by isapnp tools.  Otherwise, skip to 8.
-
-	This section described only a brief explanation.  For more
-	details, please see the AWE64-Mini-HOWTO or isapnp tools FAQ.
-
-	7a. If you have no isapnp.conf file, generate it by pnpdump.
-	    Otherwise, skip to 7d.
-		# pnpdump > /etc/isapnp.conf
-
-	7b. Edit isapnp.conf file.  Comment out the appropriate
-	    lines containing desirable I/O ports, DMA and IRQs.
-	    Don't forget to enable (ACT Y) line.
-
-	7c. Add two i/o ports (0xA20 and 0xE20) in WaveTable part.
-	    ex)
-		(CONFIGURE CTL0048/58128 (LD 2
-		#     ANSI string -->WaveTable<--
-		  (IO 0 (BASE 0x0620))
-		  (IO 1 (BASE 0x0A20))
-		  (IO 2 (BASE 0x0E20))
-		  (ACT Y)
-		))
-
-	7d. Load the config file.
-	    CAUTION: This will reset all PnP cards!
-
-		# isapnp /etc/isapnp.conf
-
-  8. Load the sound module (if you configured it as a module):
-
-	for 2.0 kernel or OSS/Free monolithic module:
-
-		# modprobe sound.o
-
-	for 2.1 kernel:
-
-		# modprobe sound
-		# insmod uart401
-		# insmod sb io=0x220 irq=5 dma=1 dma16=5 mpu_io=0x330
-		(These values depend on your settings.)
-		# insmod awe_wave
-		(Be sure to load awe_wave after sb!)
-
-		See Documentation/sound/oss/AWE32 for
-		more details.
-
-  9. (only for obsolete systems) If you don't have /dev/sequencer
-     device file, make it according to Readme.linux file on
-     /usr/src/linux/drivers/sound. (Run a shell script included in
-     that file). <-- This file no longer exists in the recent kernels!
-
-  10. OK, load your own soundfont file, and enjoy MIDI!
-
-	% sfxload synthgm.sbk
-	% drvmidi foo.mid
-
-  11. For more advanced use (eg. dynamic loading, virtual bank and
-      etc.), please read the awedrv FAQ or the instructions in awesfx
-      and awemidi packages.
-
-Good luck!

Documentation/sound/oss/MAD16

-(This recipe has been edited to update the configuration symbols,
- and change over to modprobe.conf for 2.6)
-
-From: Shaw Carruthers <shaw@shawc.demon.co.uk>
-
-I have been using mad16 sound for some time now with no problems, current
-kernel 2.1.89
-
-lsmod shows:
-
-mad16                   5176   0 
-sb                     22044   0  [mad16]
-uart401                 5576   0  [mad16 sb]
-ad1848                 14176   1  [mad16]
-sound                  61928   0  [mad16 sb uart401 ad1848]
-
-.config has:
-
-CONFIG_SOUND=m
-CONFIG_SOUND_ADLIB=m
-CONFIG_SOUND_MAD16=m
-CONFIG_SOUND_YM3812=m
-
-modprobe.conf has:
-
-alias char-major-14-* mad16
-options sb mad16=1
-options mad16 io=0x530 irq=7 dma=0 dma16=1  && /usr/local/bin/aumix -w 15 -p 20 -m 0 -1 0 -2 0 -3 0 -i 0
-
-
-To get the built in mixer to work this needs to be:
-
-options adlib_card io=0x388     # FM synthesizer
-options sb mad16=1
-options mad16 io=0x530 irq=7 dma=0 dma16=1 mpu_io=816 mpu_irq=5 && /usr/local/bin/aumix -w 15 -p 20 -m 0 -1 0 -2 0 -3 0 -i 0
-
-The addition of the "mpu_io=816 mpu_irq=5" to the mad16 options line is
-
-------------------------------------------------------------------------
-The mad16 module in addition supports the following options:
-
-option:			meaning:			default:
-joystick=0,1 		disabled, enabled 		disabled
-cdtype=0x00,0x02,0x04,	disabled, Sony CDU31A,		disabled
-       0x06,0x08,0x0a   Mitsumi, Panasonic,
-			Secondary IDE, Primary IDE 
-cdport=0x340,0x320,					0x340
-       0x330,0x360
-cdirq=0,3,5,7,9,10,11 	disabled, IRQ3, ... 		disabled
-cddma=0,5,6,7 		disabled, DMA5, ... 		DMA5 for Mitsumi or IDE
-cddma=0,1,2,3 		disabled, DMA1, ... 		DMA3 for Sony or Panasonic
-opl4=0,1 		OPL3, OPL4 			OPL3	
-
-for more details see linux/drivers/sound/mad16.c
-
-Rui Sousa

Documentation/sound/oss/Maestro

-	An OSS/Lite Driver for the ESS Maestro family of sound cards
-
-			Zach Brown, December 1999
-
-Driver Status and Availability
-------------------------------
-
-The most recent version of this driver will hopefully always be available at
-	http://www.zabbo.net/maestro/
-
-I will try and maintain the most recent stable version of the driver
-in both the stable and development kernel lines.
-
-ESS Maestro Chip Family
------------------------
-
-There are 3 main variants of the ESS Maestro PCI sound chip.  The first
-is the Maestro 1.  It was originally produced by Platform Tech as the
-'AGOGO'.  It can be recognized by Platform Tech's PCI ID 0x1285 with
-0x0100 as the device ID.  It was put on some sound boards and a few laptops.  
-ESS bought the design and cleaned it up as the Maestro 2.  This starts
-their marking with the ESS vendor ID 0x125D and the 'year' device IDs.
-The Maestro 2 claims 0x1968 while the Maestro 2e has 0x1978.
-
-The various families of Maestro are mostly identical as far as this 
-driver is concerned.  It doesn't touch the DSP parts that differ (though
-it could for FM synthesis).
-
-Driver OSS Behavior
---------------------
-
-This OSS driver exports /dev/mixer and /dev/dsp to applications, which
-mostly adhere to the OSS spec.   This driver doesn't register itself
-with /dev/sndstat, so don't expect information to appear there.
-
-The /dev/dsp device exported behaves almost as expected.  Playback is
-supported in all the various lovely formats.  8/16bit stereo/mono from
-8khz to 48khz, and mmap()ing for playback behaves.  Capture/recording
-is limited due to oddities with the Maestro hardware.  One can only
-record in 16bit stereo.  For recording the maestro uses non interleaved
-stereo buffers so that mmap()ing the incoming data does not result in
-a ring buffer of LRLR data.  mmap()ing of the read buffers is therefore
-disallowed until this can be cleaned up.
-
-/dev/mixer is an interface to the AC'97 codec on the Maestro.  It is
-worth noting that there are a variety of AC'97s that can be wired to
-the Maestro.  Which is used is entirely up to the hardware implementor.
-This should only be visible to the user by the presence, or lack, of
-'Bass' and 'Treble' sliders in the mixer.  Not all AC'97s have them.
-
-The driver doesn't support MIDI or FM playback at the moment.  Typically
-the Maestro is wired to an MPU MIDI chip, but some hardware implementations
-don't.  We need to assemble a white list of hardware implementations that
-have MIDI wired properly before we can claim to support it safely.
-
-Compiling and Installing
-------------------------
-
-With the drivers inclusion into the kernel, compiling and installing
-is the same as most OSS/Lite modular sound drivers.  Compilation
-of the driver is enabled through the CONFIG_SOUND_MAESTRO variable
-in the config system.  
-
-It may be modular or statically linked.  If it is modular it should be
-installed with the rest of the modules for the kernel on the system.
-Typically this will be in /lib/modules/ somewhere.  'alias sound maestro'
-should also be added to your module configs (typically /etc/conf.modules)
-if you're using modular OSS/Lite sound and want to default to using a
-maestro chip.
-
-As this is a PCI device, the module does not need to be informed of
-any IO or IRQ resources it should use, it devines these from the
-system.  Sometimes, on sucky PCs, the BIOS fails to allocated resources
-for the maestro.  This will result in a message like:
-	maestro: PCI subsystem reports IRQ 0, this might not be correct.
-from the kernel.  Should this happen the sound chip most likely will
-not operate correctly.  To solve this one has to dig through their BIOS
-(typically entered by hitting a hot key at boot time) and figure out
-what magic needs to happen so that the BIOS will reward the maestro with
-an IRQ.  This operation is incredibly system specific, so you're on your
-own.  Sometimes the magic lies in 'PNP Capable Operating System' settings.
-
-There are very few options to the driver.  One is 'debug' which will 
-tell the driver to print minimal debugging information as it runs.  This
-can be collected with 'dmesg' or through the klogd daemon.
-
-The other, more interesting option, is 'dsps_order'.  Typically at
-install time the driver will only register one available /dev/dsp device
-for its use.  The 'dsps_order' module parameter allows for more devices
-to be allocated, as a power of two.  Up to 4 devices can be registered
-( dsps_order=2 ).  These devices act as fully distinct units and use
-separate channels in the maestro.
-
-Power Management
-----------------
-
-As of version 0.14, this driver has a minimal understanding of PCI
-Power Management.  If it finds a valid power management capability
-on the PCI device it will attempt to use the power management
-functions of the maestro.  It will only do this on Maestro 2Es and
-only on machines that are known to function well.  You can
-force the use of power management by setting the 'use_pm' module
-option to 1, or can disable it entirely by setting it to 0.
-
-When using power management, the driver does a few things
-differently.  It will keep the chip in a lower power mode
-when the module is inserted but /dev/dsp is not open.  This
-allows the mixer to function but turns off the clocks
-on other parts of the chip.  When /dev/dsp is opened the chip
-is brought into full power mode, and brought back down
-when it is closed.  It also powers down the chip entirely
-when the module is removed or the machine is shutdown.  This
-can have nonobvious consequences.  CD audio may not work
-after a power managing driver is removed.  Also, software that
-doesn't understand power management may not be able to talk
-to the powered down chip until the machine goes through a hard
-reboot to bring it back.
-
-.. more details ..
-------------------
-
-drivers/sound/maestro.c contains comments that hopefully explain
-the maestro implementation.

Documentation/sound/oss/Maestro3

-	An OSS/Lite Driver for the ESS Maestro3 family of sound chips
-
-			Zach Brown, January 2001
-
-Driver Status and Availability
-------------------------------
-
-The most recent version of this driver will hopefully always be available at
-	http://www.zabbo.net/maestro3/
-
-I will try and maintain the most recent stable version of the driver
-in both the stable and development kernel lines.
-
-Historically I've sucked pretty hard at actually doing that, however.
-
-ESS Maestro3 Chip Family
------------------------
-
-The 'Maestro3' is much like the Maestro2 chip.  The noted improvement
-is the removal of the silicon in the '2' that did PCM mixing.  All that
-work is now done through a custom DSP called the ASSP, the Asynchronus
-Specific Signal Processor.
-
-The 'Allegro' is a baby version of the Maestro3.  I'm not entirely clear
-on the extent of the differences, but the driver supports them both :)
-
-The 'Allegro' shows up as PCI ID 0x1988 and the Maestro3 as 0x1998,
-both under ESS's vendor ID of 0x125D.  The Maestro3 can also show up as
-0x199a when hardware strapping is used.
-
-The chip can also act as a multi function device.  The modem IDs follow
-the audio multimedia device IDs.  (so the modem part of an Allegro shows
-up as 0x1989)
-
-Driver OSS Behavior
---------------------
-
-This OSS driver exports /dev/mixer and /dev/dsp to applications, which
-mostly adhere to the OSS spec.   This driver doesn't register itself
-with /dev/sndstat, so don't expect information to appear there.
-
-The /dev/dsp device exported behaves as expected.  Playback is
-supported in all the various lovely formats.  8/16bit stereo/mono from
-8khz to 48khz, with both read()/write(), and mmap().
-
-/dev/mixer is an interface to the AC'97 codec on the Maestro3.  It is
-worth noting that there are a variety of AC'97s that can be wired to
-the Maestro3.  Which is used is entirely up to the hardware implementor.
-This should only be visible to the user by the presence, or lack, of
-'Bass' and 'Treble' sliders in the mixer.  Not all AC'97s have them.
-The Allegro has an onchip AC'97.
-
-The driver doesn't support MIDI or FM playback at the moment.
-
-Compiling and Installing
-------------------------
-
-With the drivers inclusion into the kernel, compiling and installing
-is the same as most OSS/Lite modular sound drivers.  Compilation
-of the driver is enabled through the CONFIG_SOUND_MAESTRO3 variable
-in the config system.  
-
-It may be modular or statically linked.  If it is modular it should be
-installed with the rest of the modules for the kernel on the system.
-Typically this will be in /lib/modules/ somewhere.  'alias sound-slot-0
-maestro3' should also be added to your module configs (typically
-/etc/modprobe.conf) if you're using modular OSS/Lite sound and want to
-default to using a maestro3 chip.
-
-There are very few options to the driver.  One is 'debug' which will 
-tell the driver to print minimal debugging information as it runs.  This
-can be collected with 'dmesg' or through the klogd daemon.
-
-One is 'external_amp', which tells the driver to attempt to enable
-an external amplifier.  This defaults to '1', you can tell the driver
-not to bother enabling such an amplifier by setting it to '0'.
-
-And the last is 'gpio_pin', which tells the driver which GPIO pin number
-the external amp uses (0-15), The Allegro uses 8 by default, all others 1.
-If everything loads correctly and seems to be working but you get no sound, 
-try tweaking this value. 
-
-Systems known to need a different value
-        Panasonic ToughBook CF-72: gpio_pin=13 
-
-Power Management
-----------------
-
-This driver has a minimal understanding of PCI Power Management.  It will
-try and power down the chip when the system is suspended, and power
-it up with it is resumed.  It will also try and power down the chip
-when the machine is shut down.

Documentation/sound/oss/NEWS

-Linux 2.4 Sound Changes
-2000-September-25
-Christoph Hellwig, <hch@infradead.org>
-
-
-
-=== isapnp support
-
-The Linux 2.4 Kernel does have reliable in-kernel isapnp support.
-Some drivers (sb.o, ad1816.o awe_wave.o) do now support automatically
-detecting and configuring isapnp devices.
-If you have a not yet supported isapnp soundcard, mail me the content
-of '/proc/isapnp' on your system and some information about your card
-and its driver(s) so I can try to get isapnp working for it.
-
-
-
-=== soundcard resources on kernel commandline
-
-Before Linux 2.4 you had to specify the resources for sounddrivers
-statically linked into the kernel at compile time
-(in make config/menuconfig/xconfig). In Linux 2.4 the resources are
-now specified at the boot-time kernel commandline (e.g. the lilo
-'append=' line or everything that's after the kernel name in grub).
-Read the Configure.help entry for your card for the parameters.
-
-
-=== softoss is gone
-
-In Linux 2.4 the softoss in-kernel software synthesizer is no more aviable.
-Use a user space software synthesizer like timidity instead.
-
-
-
-=== /dev/sndstat and /proc/sound are gone
-
-In older Linux versions those files exported some information about the
-OSS/Free configuration to userspace. In Linux 2.3 they were removed because
-they did not support the growing number of pci soundcards and there were
-some general problems with this interface.
-
-

Documentation/sound/oss/OPL3-SA

-OPL3-SA1 sound driver (opl3sa.o)
-
----
-Note: This howto only describes how to setup the OPL3-SA1 chip; this info
-does not apply to the SA2, SA3, or SA4. 
----
-
-The Yamaha OPL3-SA1 sound chip is usually found built into motherboards, and
-it's a decent little chip offering a WSS mode, a SB Pro emulation mode, MPU401
-and OPL3 FM Synth capabilities.
-
-You can enable inclusion of the driver via CONFIG_SOUND_OPL3SA1=m, or
-CONFIG_SOUND_OPL3SA1=y through 'make config/xconfig/menuconfig'.
-
-You'll need to know all of the relevant info (irq, dma, and io port) for the
-chip's WSS mode, since that is the mode the kernel sound driver uses, and of
-course you'll also need to know about where the MPU401 and OPL3 ports and
-IRQs are if you want to use those.
-
-Here's the skinny on how to load it as a module:
-
-	modprobe opl3sa io=0x530 irq=11 dma=0 dma2=1 mpu_io=0x330 mpu_irq=5
-
-Module options in detail:
-
-	io:	This is the WSS's port base.
-	irq:	This is the WSS's IRQ.
-	dma:	This is the WSS's DMA line. In my BIOS setup screen this was
-		listed as "WSS Play DMA"
-	dma2:	This is the WSS's secondary DMA line. My BIOS calls it the
-		"WSS capture DMA"
-	
-	mpu_io:	This is the MPU401's port base.
-	mpu_irq: This is the MPU401's IRQ.
-
-If you'd like to use the OPL3 FM Synthesizer, make sure you enable
-CONFIG_SOUND_YM3812 (in 'make config'). That'll build the opl3.o module.
-
-Then a simple 'insmod opl3 io=0x388', and you now have FM Synth.
-
-You can also use the SoftOSS software synthesizer instead of the builtin OPL3.
-Here's how:
-
-Say 'y' or 'm' to "SoftOSS software wave table engine" in make config.
-
-If you said yes, the software synth is available once you boot your new
-kernel.
-
-If you chose to build it as a module, just insmod the resulting softoss2.o
-
-Questions? Comments?
-<stiker@northlink.com>

Documentation/sound/oss/README.awe

-================================================================
-	AWE32 Sound Driver for Linux / FreeBSD
-		version 0.4.3; Nov. 1, 1998
-
-	Takashi Iwai <iwai@ww.uni-erlangen.de>
-================================================================
-
-* GENERAL NOTES
-
-This is a sound driver extension for SoundBlaster AWE32 and other
-compatible cards (AWE32-PnP, SB32, SB32-PnP, AWE64 & etc) to enable
-the wave synth operations.  The driver is provided for Linux 1.2.x
-and 2.[012].x kernels, as well as FreeBSD, on Intel x86 and DEC
-Alpha systems.
-
-This driver was written by Takashi Iwai <iwai@ww.uni-erlangen.de>,
-and provided "as is".  The original source (awedrv-0.4.3.tar.gz) and
-binary packages are available on the following URL:
-	http://bahamut.mm.t.u-tokyo.ac.jp/~iwai/awedrv/
-Note that since the author is apart from this web site, the update is
-not frequent now.
-
-
-* NOTE TO LINUX USERS
-
-To enable this driver on linux-2.[01].x kernels, you need turn on 
-"AWE32 synth" options in sound menu when configure your linux kernel
-and modules.  The precise installation procedure is described in the
-AWE64-Mini-HOWTO and linux-kernel/Documetation/sound/AWE32.
-
-If you're using PnP cards, the card must be initialized before loading
-the sound driver.  There're several options to do this:
-    - Initialize the card via ISA PnP tools, and load the sound module.
-    - Initialize the card on DOS, and load linux by loadlin.exe
-    - Use PnP kernel driver (for Linux-2.x.x)
-The detailed instruction for the solution using isapnp tools is found
-in many documents like above.  A brief instruction is also included in
-the installation document of this package.
-For PnP driver project, please refer to the following URL:
-	http://www-jcr.lmh.ox.ac.uk/~pnp/
-
-
-* USING THE DRIVER
-
-The awedrv has several different playing modes to realize easy channel 
-allocation for MIDI songs.  To hear the exact sound quality, you need
-to obtain the extended sequencer program, drvmidi or playmidi-2.5.
-
-For playing MIDI files, you *MUST* load the soundfont file on the
-driver previously by sfxload utility.  Otherwise you'll here no sounds 
-at all!  All the utilities and driver source packages are found in the
-above URL.  The sfxload program is included in the package
-awesfx-0.4.3.tgz.  Binary packages are available there, too.  See the
-instruction in each package for installation.
-
-Loading a soundfont file is very simple.  Just execute the command
-
-	% sfxload synthgm.sbk
-
-Then, sfxload transfers the file "synthgm.sbk" to the driver.
-Both SF1 and SF2 formats are accepted.
-
-Now you can hear midi musics by a midi player.
-
-	% drvmidi foo.mid
-
-If you run MIDI player after MOD player, you need to load soundfont
-files again, since MOD player programs clear the previous loaded
-samples by their own data.
-
-If you have only 512kb on the sound card, I recommend to use dynamic
-sample loading via -L option of drvmidi.  2MB GM/GS soundfont file is
-available in most midi files.
-
-	% sfxload synthgm
-	% drvmidi -L 2mbgmgs foo.mid
-
-This makes a big difference (believe me)!  For more details, please
-refer to the FAQ list which is available on the URL above.
-
-The current chorus, reverb and equalizer status can be changed by
-aweset utility program (included in awesfx package).  Note that
-some awedrv-native programs (like drvmidi and xmp) will change the
-current settings by themselves.  The aweset program is effective
-only for other programs like playmidi.
-
-Enjoy.
-
-
-* COMPILE FLAGS
-
-Compile conditions are defined in awe_config.h.
-
-[Compatibility Conditions]
-The following flags are defined automatically when using installation
-shell script.
-
-- AWE_MODULE_SUPPORT
-    indicates your Linux kernel supports module for each sound card
-    (in recent 2.1 or 2.2 kernels and unofficial patched 2.0 kernels
-    as distributed in the RH5.0 package).
-    This flag is automatically set when you're using 2.1.x kernels.
-    You can pass the base address and memory size via the following
-    module options,
-	io = base I/O port address (eg. 0x620)
-	memsize = DRAM size in kilobytes (eg. 512)
-    As default, AWE driver probes these values automatically.
-
-
-[Hardware Conditions]
-You DON'T have to define the following two values.
-Define them only when the driver couldn't detect the card properly.
-
-- AWE_DEFAULT_BASE_ADDR		(default: not defined)
-    specifies the base port address of your AWE32 card.
-    0 means to autodetect the address.
-
-- AWE_DEFAULT_MEM_SIZE		(default: not defined)
-    specifies the memory size of your AWE32 card in kilobytes.
-    -1 means to autodetect its size.
-    
-
-[Sample Table Size]
-From ver.0.4.0, sample tables are allocated dynamically (except
-Linux-1.2.x system), so you need NOT to touch these parameters.
-Linux-1.2.x users may need to increase these values to appropriate size 
-if the sound card is equipped with more DRAM.
-
-- AWE_MAX_SF_LISTS, AWE_MAX_SAMPLES, AWE_MAX_INFOS
-
-
-[Other Conditions]
-
-- AWE_ALWAYS_INIT_FM		(default: not defined)
-    indicates the AWE driver always initialize FM passthrough even
-    without DRAM on board.  Emu8000 chip has a restriction for playing
-    samples on DRAM that at least two channels must be occupied as
-    passthrough channels. 
-
-- AWE_DEBUG_ON			(default: defined)
-    turns on debugging messages if defined.
-
-- AWE_HAS_GUS_COMPATIBILITY	(default: defined)
-    Enables GUS compatibility mode if defined, reading GUS patches and 
-    GUS control commands.  Define this option to use GMOD or other
-    GUS module players.
-
-- CONFIG_AWE32_MIDIEMU		(default: defined)
-    Adds a MIDI emulation device by Emu8000 wavetable.  The emulation
-    device can be accessed as an external MIDI, and sends the MIDI
-    control codes directly.  XG and GS sysex/NRPN are accepted.
-    No MIDI input is supported.
-
-- CONFIG_AWE32_MIXER		(default: not defined)
-    Adds a mixer device for AWE32 bass/treble equalizer control.
-    You can access this device using /dev/mixer?? (usually mixer01).
-
-- AWE_USE_NEW_VOLUME_CALC	(default: defined)
-    Use the new method to calculate the volume change as compatible
-    with DOS/Win drivers.  This option can be toggled via aweset
-    program, or drvmidi player.
-
-- AWE_CHECK_VTARGET		(default: defined)
-    Check the current volume target value when searching for an
-    empty channel to allocate a new voice.  This is experimentally
-    implemented in this version.  (probably, this option doesn't
-    affect the sound quality severely...)
-
-- AWE_ALLOW_SAMPLE_SHARING	(default: defined)
-   Allow sample sharing for differently loaded patches.
-   This function is available only together with awesfx-0.4.3p3.
-   Note that this is still an experimental option.
-
-- DEF_FM_CHORUS_DEPTH		(default: 0x10)
-    The default strength to be sent to the chorus effect engine.
-    From 0 to 0xff.  Larger numbers may often cause weird sounds.
-
-- DEF_FM_REVERB_DEPTH		(default: 0x10)
-    The default strength to be sent to the reverb effect engine.
-    From 0 to 0xff.  Larger numbers may often cause weird sounds.
-
-
-* ACKNOWLEDGMENTS
-
-Thanks to Witold Jachimczyk (witek@xfactor.wpi.edu) for much advice
-on programming of AWE32.  Much code is brought from his AWE32-native 
-MOD player, ALMP.
-The port of awedrv to FreeBSD is done by Randall Hopper
-(rhh@ct.picker.com).
-The new volume calculation routine was derived from Mark Weaver's
-ADIP compatible routines.
-I also thank linux-awe-ml members for their efforts
-to reboot their system many times :-)
-
-
-* TODO'S
-
-- Complete DOS/Win compatibility
-- DSP-like output
-
-
-* COPYRIGHT
-
-Copyright (C) 1996-1998 Takashi Iwai
-
-This program is free software; you can redistribute it and/or modify
-it under the terms of the GNU General Public License as published by
-the Free Software Foundation; either version 2 of the License, or
-(at your option) any later version.
-
-This program is distributed in the hope that it will be useful,
-but WITHOUT ANY WARRANTY; without even the implied warranty of
-MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-GNU General Public License for more details.
-
-You should have received a copy of the GNU General Public License
-along with this program; if not, write to the Free Software
-Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.

Documentation/sound/oss/es1370

-/proc/sound, /dev/sndstat
--------------------------
-
-/proc/sound and /dev/sndstat is not supported by the
-driver. To find out whether the driver succeeded loading,
-check the kernel log (dmesg).
-
-
-ALaw/uLaw sample formats
-------------------------
-
-This driver does not support the ALaw/uLaw sample formats.
-ALaw is the default mode when opening a sound device
-using OSS/Free. The reason for the lack of support is
-that the hardware does not support these formats, and adding
-conversion routines to the kernel would lead to very ugly
-code in the presence of the mmap interface to the driver.
-And since xquake uses mmap, mmap is considered important :-)
-and no sane application uses ALaw/uLaw these days anyway.
-In short, playing a Sun .au file as follows:
-
-cat my_file.au > /dev/dsp
-
-does not work. Instead, you may use the play script from
-Chris Bagwell's sox-12.14 package (available from the URL
-below) to play many different audio file formats.
-The script automatically determines the audio format
-and does do audio conversions if necessary.
-http://home.sprynet.com/sprynet/cbagwell/projects.html
-
-
-Blocking vs. nonblocking IO
----------------------------
-
-Unlike OSS/Free this driver honours the O_NONBLOCK file flag
-not only during open, but also during read and write.
-This is an effort to make the sound driver interface more
-regular. Timidity has problems with this; a patch
-is available from http://www.ife.ee.ethz.ch/~sailer/linux/pciaudio.html.
-(Timidity patched will also run on OSS/Free).
-
-
-MIDI UART
----------
-
-The driver supports a simple MIDI UART interface, with
-no ioctl's supported.
-
-
-MIDI synthesizer
-----------------
-
-This soundcard does not have any hardware MIDI synthesizer;
-MIDI synthesis has to be done in software. To allow this
-the driver/soundcard supports two PCM (/dev/dsp) interfaces.
-The second one goes to the mixer "synth" setting and supports
-only a limited set of sampling rates (44100, 22050, 11025, 5512).
-By setting lineout to 1 on the driver command line
-(eg. insmod es1370 lineout=1) it is even possible on some
-cards to convert the LINEIN jack into a second LINEOUT jack, thus
-making it possible to output four independent audio channels!
-
-There is a freely available software package that allows
-MIDI file playback on this soundcard called Timidity.
-See http://www.cgs.fi/~tt/timidity/.
-
-
-
-Thomas Sailer
-t.sailer@alumni.ethz.ch

Documentation/sound/oss/solo1

-Recording
----------
-
-Recording does not work on the author's card, but there
-is at least one report of it working on later silicon.
-The chip behaves differently than described in the data sheet,
-likely due to a chip bug. Working around this would require
-the help of ESS (for example by publishing an errata sheet),
-but ESS has not done so far.
-
-Also, the chip only supports 24 bit addresses for recording,
-which means it cannot work on some Alpha mainboards.
-
-
-/proc/sound, /dev/sndstat
--------------------------
-
-/proc/sound and /dev/sndstat is not supported by the
-driver. To find out whether the driver succeeded loading,
-check the kernel log (dmesg).
-
-
-ALaw/uLaw sample formats
-------------------------
-
-This driver does not support the ALaw/uLaw sample formats.
-ALaw is the default mode when opening a sound device
-using OSS/Free. The reason for the lack of support is
-that the hardware does not support these formats, and adding
-conversion routines to the kernel would lead to very ugly
-code in the presence of the mmap interface to the driver.
-And since xquake uses mmap, mmap is considered important :-)
-and no sane application uses ALaw/uLaw these days anyway.
-In short, playing a Sun .au file as follows:
-
-cat my_file.au > /dev/dsp
-
-does not work. Instead, you may use the play script from
-Chris Bagwell's sox-12.14 package (or later, available from the URL
-below) to play many different audio file formats.
-The script automatically determines the audio format
-and does do audio conversions if necessary.
-http://home.sprynet.com/sprynet/cbagwell/projects.html
-
-
-Blocking vs. nonblocking IO
----------------------------
-
-Unlike OSS/Free this driver honours the O_NONBLOCK file flag
-not only during open, but also during read and write.
-This is an effort to make the sound driver interface more
-regular. Timidity has problems with this; a patch
-is available from http://www.ife.ee.ethz.ch/~sailer/linux/pciaudio.html.
-(Timidity patched will also run on OSS/Free).
-
-
-MIDI UART
----------
-
-The driver supports a simple MIDI UART interface, with
-no ioctl's supported.
-
-
-MIDI synthesizer
-----------------
-
-The card has an OPL compatible FM synthesizer.
-
-Thomas Sailer
-t.sailer@alumni.ethz.ch

Documentation/sound/oss/sonicvibes

-/proc/sound, /dev/sndstat
--------------------------
-
-/proc/sound and /dev/sndstat is not supported by the
-driver. To find out whether the driver succeeded loading,
-check the kernel log (dmesg).
-
-
-ALaw/uLaw sample formats
-------------------------
-
-This driver does not support the ALaw/uLaw sample formats.
-ALaw is the default mode when opening a sound device
-using OSS/Free. The reason for the lack of support is
-that the hardware does not support these formats, and adding
-conversion routines to the kernel would lead to very ugly
-code in the presence of the mmap interface to the driver.
-And since xquake uses mmap, mmap is considered important :-)
-and no sane application uses ALaw/uLaw these days anyway.
-In short, playing a Sun .au file as follows:
-
-cat my_file.au > /dev/dsp
-
-does not work. Instead, you may use the play script from
-Chris Bagwell's sox-12.14 package (available from the URL
-below) to play many different audio file formats.
-The script automatically determines the audio format
-and does do audio conversions if necessary.
-http://home.sprynet.com/sprynet/cbagwell/projects.html
-
-
-Blocking vs. nonblocking IO
----------------------------
-
-Unlike OSS/Free this driver honours the O_NONBLOCK file flag
-not only during open, but also during read and write.
-This is an effort to make the sound driver interface more
-regular. Timidity has problems with this; a patch
-is available from http://www.ife.ee.ethz.ch/~sailer/linux/pciaudio.html.
-(Timidity patched will also run on OSS/Free).
-
-
-MIDI UART
----------
-
-The driver supports a simple MIDI UART interface, with
-no ioctl's supported.
-
-
-MIDI synthesizer
-----------------
-
-The card both has an OPL compatible FM synthesizer as well as
-a wavetable synthesizer.
-
-I haven't managed so far to get the OPL synth running.
-
-Using the wavetable synthesizer requires allocating
-1-4MB of physically contiguous memory, which isn't possible
-currently on Linux without ugly hacks like the bigphysarea
-patch. Therefore, the driver doesn't support wavetable
-synthesis.
-
-
-No support from S3
-------------------
-
-I do not get any support from S3. Therefore, the driver
-still has many problems. For example, although the manual
-states that the chip should be able to access the sample
-buffer anywhere in 32bit address space, I haven't managed to
-get it working with buffers above 16M. Therefore, the card
-has the same disadvantages as ISA soundcards.
-
-Given that the card is also very noisy, and if you haven't
-already bought it, you should strongly opt for one of the
-comparatively priced Ensoniq products.
-
-
-Thomas Sailer
-t.sailer@alumni.ethz.ch

MAINTAINERS

@@ -1900,11 +1900,6 @@ M: 	rroesler@syskonnect.de
 W: 	http://www.syskonnect.com
 S: 	Supported
 
-MAESTRO PCI SOUND DRIVERS
-P:	Zach Brown
-M:	zab@zabbo.net
-S:	Odd Fixes
-
 MAN-PAGES: MANUAL PAGES FOR LINUX -- Sections 2, 3, 4, 5, and 7
 P: Michael Kerrisk
 M: mtk-manpages@gmx.net
@@ -3402,12 +3397,6 @@ M:	Henk.Vergonet@gmail.com
 L:	usbb2k-api-dev@nongnu.org
 S:	Maintained
 
-YMFPCI YAMAHA PCI SOUND (Use ALSA instead)
-P:	Pete Zaitcev
-M:	zaitcev@yahoo.com
-L:	linux-kernel@vger.kernel.org
-S:	Obsolete
-
 Z8530 DRIVER FOR AX.25
 P:	Joerg Reuter
 M:	jreuter@yaina.de

include/linux/Kbuild

@@ -160,7 +160,6 @@ header-y += video_decoder.h
 header-y += video_encoder.h
 header-y += videotext.h
 header-y += vt.h
-header-y += wavefront.h
 header-y += wireless.h
 header-y += xattr.h
 header-y += x25.h

include/linux/ac97_codec.h

@@ -331,8 +331,6 @@ extern int ac97_read_proc (char *page_out, char **start, off_t off,
 extern int ac97_probe_codec(struct ac97_codec *);
 extern unsigned int ac97_set_adc_rate(struct ac97_codec *codec, unsigned int rate);
 extern unsigned int ac97_set_dac_rate(struct ac97_codec *codec, unsigned int rate);
-extern int ac97_save_state(struct ac97_codec *codec);
-extern int ac97_restore_state(struct ac97_codec *codec);
 
 extern struct ac97_codec *ac97_alloc_codec(void);
 extern void ac97_release_codec(struct ac97_codec *codec);
@@ -346,9 +344,6 @@ struct ac97_driver {
 	void (*remove) (struct ac97_codec *codec, struct ac97_driver *driver);
 };
 
-extern int ac97_register_driver(struct ac97_driver *driver);
-extern void ac97_unregister_driver(struct ac97_driver *driver);
-
 /* quirk types */
 enum {
 	AC97_TUNE_DEFAULT = -1, /* use default from quirk list (not valid in list) */

include/linux/sound.h

@@ -35,10 +35,8 @@ extern int register_sound_special_device(const struct file_operations *fops, int
 extern int register_sound_mixer(const struct file_operations *fops, int dev);
 extern int register_sound_midi(const struct file_operations *fops, int dev);
 extern int register_sound_dsp(const struct file_operations *fops, int dev);
-extern int register_sound_synth(const struct file_operations *fops, int dev);
 
 extern void unregister_sound_special(int unit);
 extern void unregister_sound_mixer(int unit);
 extern void unregister_sound_midi(int unit);
 extern void unregister_sound_dsp(int unit);
-extern void unregister_sound_synth(int unit);

sound/oss/Makefile

@@ -15,71 +15,42 @@ obj-$(CONFIG_SOUND_HAL2)	+= hal2.o
 obj-$(CONFIG_SOUND_AEDSP16)	+= aedsp16.o
 obj-$(CONFIG_SOUND_PSS)		+= pss.o ad1848.o mpu401.o
 obj-$(CONFIG_SOUND_TRIX)	+= trix.o ad1848.o sb_lib.o uart401.o
-obj-$(CONFIG_SOUND_OPL3SA1)	+= opl3sa.o ad1848.o uart401.o
 obj-$(CONFIG_SOUND_SSCAPE)	+= sscape.o ad1848.o mpu401.o
-obj-$(CONFIG_SOUND_MAD16)	+= mad16.o ad1848.o sb_lib.o uart401.o
 obj-$(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_WAVEFRONT)	+= wavefront.o
-obj-$(CONFIG_SOUND_MAUI)	+= maui.o mpu401.o
 obj-$(CONFIG_SOUND_MPU401)	+= mpu401.o
 obj-$(CONFIG_SOUND_UART6850)	+= uart6850.o
-obj-$(CONFIG_SOUND_GUS)		+= gus.o ad1848.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_SGALAXY)	+= sgalaxy.o ad1848.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_AWE32_SYNTH)	+= awe_wave.o
 
 obj-$(CONFIG_SOUND_VIA82CXXX)	+= via82cxxx_audio.o ac97_codec.o
 ifeq ($(CONFIG_MIDI_VIA82CXXX),y)
   obj-$(CONFIG_SOUND_VIA82CXXX) += sound.o uart401.o
 endif
-obj-$(CONFIG_SOUND_YMFPCI)	+= ymfpci.o ac97_codec.o
-ifeq ($(CONFIG_SOUND_YMFPCI_LEGACY),y)
-  obj-$(CONFIG_SOUND_YMFPCI)    += opl3.o uart401.o
-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_SONICVIBES)	+= sonicvibes.o
-obj-$(CONFIG_SOUND_CMPCI)	+= cmpci.o
-ifeq ($(CONFIG_SOUND_CMPCI_FM),y)
-  obj-$(CONFIG_SOUND_CMPCI)     += sound.o opl3.o
-endif
-ifeq ($(CONFIG_SOUND_CMPCI_MIDI),y)
-  obj-$(CONFIG_SOUND_CMPCI)     += sound.o mpu401.o
-endif
-obj-$(CONFIG_SOUND_ES1370)	+= es1370.o
 obj-$(CONFIG_SOUND_ES1371)	+= es1371.o ac97_codec.o
 obj-$(CONFIG_SOUND_VRC5477)	+= nec_vrc5477.o ac97_codec.o
-obj-$(CONFIG_SOUND_AU1000)	+= au1000.o ac97_codec.o
 obj-$(CONFIG_SOUND_AU1550_AC97)	+= au1550_ac97.o ac97_codec.o
-obj-$(CONFIG_SOUND_ESSSOLO1)	+= esssolo1.o
 obj-$(CONFIG_SOUND_FUSION)	+= cs46xx.o ac97_codec.o
-obj-$(CONFIG_SOUND_MAESTRO)	+= maestro.o
-obj-$(CONFIG_SOUND_MAESTRO3)	+= maestro3.o ac97_codec.o
 obj-$(CONFIG_SOUND_TRIDENT)	+= trident.o ac97_codec.o
-obj-$(CONFIG_SOUND_HARMONY)	+= harmony.o
 obj-$(CONFIG_SOUND_EMU10K1)	+= ac97_codec.o
 obj-$(CONFIG_SOUND_BCM_CS4297A)	+= swarm_cs4297a.o
-obj-$(CONFIG_SOUND_RME96XX)     += rme96xx.o
 obj-$(CONFIG_SOUND_BT878)	+= btaudio.o
-obj-$(CONFIG_SOUND_ALI5455)	+= ali5455.o ac97_codec.o
-obj-$(CONFIG_SOUND_FORTE)	+= forte.o ac97_codec.o
 
-obj-$(CONFIG_SOUND_AD1980)	+= ac97_plugin_ad1980.o ac97_codec.o
 obj-$(CONFIG_SOUND_WM97XX)	+= ac97_plugin_wm97xx.o
 
 ifeq ($(CONFIG_MIDI_EMU10K1),y)
@@ -87,7 +58,6 @@ ifeq ($(CONFIG_MIDI_EMU10K1),y)
 endif
 
 obj-$(CONFIG_SOUND_EMU10K1)	+= emu10k1/
-obj-$(CONFIG_SOUND_CS4281)	+= cs4281/
 obj-$(CONFIG_DMASOUND)		+= dmasound/
 
 # Declare multi-part drivers.
@@ -98,17 +68,15 @@ sound-objs	:= 							\
     midi_syms.o midi_synth.o midibuf.o					\
     sequencer.o sequencer_syms.o sound_timer.o sys_timer.o
 
-gus-objs	:= gus_card.o gus_midi.o gus_vol.o gus_wave.o ics2101.o
 pas2-objs	:= pas2_card.o pas2_midi.o pas2_mixer.o pas2_pcm.o
 sb-objs		:= sb_card.o
 sb_lib-objs	:= sb_common.o sb_audio.o sb_midi.o sb_mixer.o sb_ess.o
 vidc_mod-objs	:= vidc.o vidc_fill.o
-wavefront-objs  := wavfront.o wf_midi.o yss225.o
 
 hostprogs-y	:= bin2hex hex2hex
 
 # Files generated that shall be removed upon make clean
-clean-files := maui_boot.h msndperm.c msndinit.c pndsperm.c pndspini.c \
+clean-files := msndperm.c msndinit.c pndsperm.c pndspini.c \
                pss_boot.h trix_boot.h
 
 # Firmware files that need translation
@@ -118,21 +86,6 @@ clean-files := maui_boot.h msndperm.c msndinit.c pndsperm.c pndspini.c \
 # will be forced to be remade.
 #
 
-# Turtle Beach Maui / Tropez
-
-$(obj)/maui.o: $(obj)/maui_boot.h
-
-ifeq ($(CONFIG_MAUI_HAVE_BOOT),y)
-    $(obj)/maui_boot.h: $(patsubst "%", %, $(CONFIG_MAUI_BOOT_FILE)) $(obj)/bin2hex
-	$(obj)/bin2hex -i maui_os < $< > $@
-else
-    $(obj)/maui_boot.h:
-	(							\
-	    echo 'static unsigned char * maui_os = NULL;';	\
-	    echo 'static int maui_osLen = 0;';			\
-	) > $@
-endif
-
 # Turtle Beach MultiSound
 
 ifeq ($(CONFIG_MSNDCLAS_HAVE_BOOT),y)

sound/oss/ac97.c

@@ -112,25 +112,6 @@ ac97_init (struct ac97_hwint *dev)
     return 0;
 }
 
-/* Reset the mixer to the currently saved settings.  */
-int
-ac97_reset (struct ac97_hwint *dev)
-{
-    int x;
-
-    if (dev->reset_device (dev))
-	return -1;
-
-    /* Now set the registers back to their last-written values. */
-    for (x = 0; mixerRegs[x].ac97_regnum != -1; x++) {
-	int regnum = mixerRegs[x].ac97_regnum;
-	int value = dev->last_written_mixer_values [regnum / 2];
-	if (value >= 0)
-	    ac97_put_register (dev, regnum, value);
-    }
-    return 0;
-}
-
 /* Return the contents of register REG; use the cache if the value in it
    is valid.  Returns a negative error code on failure. */
 static int
@@ -441,7 +422,6 @@ EXPORT_SYMBOL(ac97_init);
 EXPORT_SYMBOL(ac97_set_values);
 EXPORT_SYMBOL(ac97_put_register);
 EXPORT_SYMBOL(ac97_mixer_ioctl);
-EXPORT_SYMBOL(ac97_reset);
 MODULE_LICENSE("GPL");
 
 

sound/oss/ac97.h

@@ -192,9 +192,6 @@ extern int ac97_put_register (struct ac97_hwint *dev, u8 reg, u16 value);
 extern int ac97_mixer_ioctl (struct ac97_hwint *dev, unsigned int cmd,
 			     void __user * arg);
 
-/* Do a complete reset on the AC97 mixer, restoring all mixer registers to
-   the current values.  Normally used after an APM resume event.  */
-extern int ac97_reset (struct ac97_hwint *dev);
 #endif
 
 /*

sound/oss/ac97_codec.c

@@ -1399,95 +1399,6 @@ unsigned int ac97_set_adc_rate(struct ac97_codec *codec, unsigned int rate)
 
 EXPORT_SYMBOL(ac97_set_adc_rate);
 
-int ac97_save_state(struct ac97_codec *codec)
-{
-	return 0;	
-}
-
-EXPORT_SYMBOL(ac97_save_state);
-
-int ac97_restore_state(struct ac97_codec *codec)
-{
-	int i;
-	unsigned int left, right, val;
-
-	for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) {
-		if (!supported_mixer(codec, i)) 
-			continue;
-
-		val = codec->mixer_state[i];
-		right = val >> 8;
-		left = val  & 0xff;
-		codec->write_mixer(codec, i, left, right);
-	}
-	return 0;
-}
-
-EXPORT_SYMBOL(ac97_restore_state);
-
-/**
- *	ac97_register_driver	-	register a codec helper
- *	@driver: Driver handler
- *
- *	Register a handler for codecs matching the codec id. The handler
- *	attach function is called for all present codecs and will be 
- *	called when new codecs are discovered.
- */
- 
-int ac97_register_driver(struct ac97_driver *driver)
-{
-	struct list_head *l;
-	struct ac97_codec *c;
-	
-	mutex_lock(&codec_mutex);
-	INIT_LIST_HEAD(&driver->list);
-	list_add(&driver->list, &codec_drivers);
-	
-	list_for_each(l, &codecs)
-	{
-		c = list_entry(l, struct ac97_codec, list);
-		if(c->driver != NULL || ((c->model ^ driver->codec_id) & driver->codec_mask))
-			continue;
-		if(driver->probe(c, driver))
-			continue;
-		c->driver = driver;
-	}
-	mutex_unlock(&codec_mutex);
-	return 0;
-}
-
-EXPORT_SYMBOL_GPL(ac97_register_driver);
-
-/**
- *	ac97_unregister_driver	-	unregister a codec helper
- *	@driver: Driver handler
- *
- *	Unregister a handler for codecs matching the codec id. The handler
- *	remove function is called for all matching codecs.
- */
- 
-void ac97_unregister_driver(struct ac97_driver *driver)
-{
-	struct list_head *l;
-	struct ac97_codec *c;
-	
-	mutex_lock(&codec_mutex);
-	list_del_init(&driver->list);
-
-	list_for_each(l, &codecs)
-	{
-		c = list_entry(l, struct ac97_codec, list);
-		if (c->driver == driver) {
-			driver->remove(c, driver);
-			c->driver = NULL;
-		}
-	}
-	
-	mutex_unlock(&codec_mutex);
-}
-
-EXPORT_SYMBOL_GPL(ac97_unregister_driver);
-
 static int swap_headphone(int remove_master)
 {
 	struct list_head *l;

sound/oss/ac97_plugin_ad1980.c

-/*
-    ac97_plugin_ad1980.c  Copyright (C) 2003 Red Hat, Inc. All rights reserved.
-
-   The contents of this file are subject to the Open Software License version 1.1
-   that can be found at http://www.opensource.org/licenses/osl-1.1.txt and is 
-   included herein by reference. 
-   
-   Alternatively, the contents of this file may be used under the
-   terms of the GNU General Public License version 2 (the "GPL") as 
-   distributed in the kernel source COPYING file, in which
-   case the provisions of the GPL are applicable instead of the
-   above.  If you wish to allow the use of your version of this file
-   only under the terms of the GPL and not to allow others to use
-   your version of this file under the OSL, indicate your decision
-   by deleting the provisions above and replace them with the notice
-   and other provisions required by the GPL.  If you do not delete
-   the provisions above, a recipient may use your version of this
-   file under either the OSL or the GPL.
-   
-   Authors: 	Alan Cox <alan@redhat.com>
-
-   This is an example codec plugin. This one switches the connections
-   around to match the setups some vendors use with audio switched to
-   non standard front connectors not the normal rear ones
-
-   This code primarily exists to demonstrate how to use the codec
-   interface
-
-*/
-
-#include <linux/config.h>
-#include <linux/module.h>
-#include <linux/init.h>
-#include <linux/kernel.h>
-#include <linux/ac97_codec.h>
-
-/**
- *	ad1980_remove		-	codec remove callback
- *	@codec: The codec that is being removed
- *
- *	This callback occurs when an AC97 codec is being removed. A
- *	codec remove call will not occur for a codec during that codec
- *	probe callback.
- *
- *	Most drivers will need to lock their remove versus their 
- *	use of the codec after the probe function.
- */
- 
-static void __devexit ad1980_remove(struct ac97_codec *codec, struct ac97_driver *driver)
-{
-	/* Nothing to do in the simple example */
-}
-
-
-/**
- *	ad1980_probe		-	codec found callback
- *	@codec: ac97 codec matching the idents
- *	@driver: ac97_driver it matched
- *
- *	This entry point is called when a codec is found which matches
- *	the driver. At the point it is called the codec is basically
- *	operational, mixer operations have been initialised and can
- *	be overriden. Called in process context. The field driver_private
- *	is available for the driver to use to store stuff.
- *
- *	The caller can claim the device by returning zero, or return
- *	a negative error code. 
- */
- 
-static int ad1980_probe(struct ac97_codec *codec, struct ac97_driver *driver)
-{
-	u16 control;
-
-#define AC97_AD_MISC	0x76
-
-	/* Switch the inputs/outputs over (from Dell code) */
-	control = codec->codec_read(codec, AC97_AD_MISC);
-	codec->codec_write(codec, AC97_AD_MISC, control | 0x4420);
-	
-	/* We could refuse the device since we dont need to hang around,
-	   but we will claim it */
-	return 0;
-}
-	
- 
-static struct ac97_driver ad1980_driver = {
-	.codec_id	= 0x41445370,
-	.codec_mask	= 0xFFFFFFFF,
-	.name		= "AD1980 example",
-	.probe		= ad1980_probe,
-	.remove		= __devexit_p(ad1980_remove),
-};
-
-/**
- *	ad1980_exit		-	module exit path
- *
- *	Our module is being unloaded. At this point unregister_driver
- *	will call back our remove handler for any existing codecs. You
- *	may not unregister_driver from interrupt context or from a 
- *	probe/remove callback.
- */
-
-static void ad1980_exit(void)
-{
-	ac97_unregister_driver(&ad1980_driver);
-}
-
-/**
- *	ad1980_init		-	set up ad1980 handlers
- *
- *	After we call the register function it will call our probe
- *	function for each existing matching device before returning to us.
- *	Any devices appearing afterwards whose id's match the codec_id
- *	will also cause the probe function to be called.
- *	You may not register_driver from interrupt context or from a 
- *	probe/remove callback.
- */
- 
-static int ad1980_init(void)
-{
-	return ac97_register_driver(&ad1980_driver);
-}
-
-module_init(ad1980_init);
-module_exit(ad1980_exit);
-MODULE_LICENSE("GPL");

sound/oss/ad1848.c

@@ -195,6 +195,7 @@ static void     ad1848_halt(int dev);
 static void     ad1848_halt_input(int dev);
 static void     ad1848_halt_output(int dev);
 static void     ad1848_trigger(int dev, int bits);
+static irqreturn_t adintr(int irq, void *dev_id, struct pt_regs *dummy);
 
 #ifndef EXCLUDE_TIMERS
 static int ad1848_tmr_install(int dev);
@@ -2195,7 +2196,7 @@ void ad1848_unload(int io_base, int irq, int dma_playback, int dma_capture, int
 		printk(KERN_ERR "ad1848: Can't find device to be unloaded. Base=%x\n", io_base);
 }
 
-irqreturn_t adintr(int irq, void *dev_id, struct pt_regs *dummy)
+static irqreturn_t adintr(int irq, void *dev_id, struct pt_regs *dummy)
 {
 	unsigned char status;
 	ad1848_info *devc;
@@ -2802,7 +2803,6 @@ EXPORT_SYMBOL(ad1848_detect);
 EXPORT_SYMBOL(ad1848_init);
 EXPORT_SYMBOL(ad1848_unload);
 EXPORT_SYMBOL(ad1848_control);
-EXPORT_SYMBOL(adintr);
 EXPORT_SYMBOL(probe_ms_sound);
 EXPORT_SYMBOL(attach_ms_sound);
 EXPORT_SYMBOL(unload_ms_sound);

sound/oss/ad1848.h

@@ -18,7 +18,6 @@ void ad1848_unload (int io_base, int irq, int dma_playback, int dma_capture, int
 int ad1848_detect (struct resource *ports, int *flags, int *osp);
 int ad1848_control(int cmd, int arg);
 
-irqreturn_t adintr(int irq, void *dev_id, struct pt_regs * dummy);
 void attach_ms_sound(struct address_info * hw_config, struct resource *ports, struct module * owner);
 
 int probe_ms_sound(struct address_info *hw_config, struct resource *ports);

sound/oss/audio_syms.c

@@ -10,7 +10,5 @@ char audio_syms_symbol;
 #include "sound_calls.h"
 
 EXPORT_SYMBOL(DMAbuf_start_dma);
-EXPORT_SYMBOL(DMAbuf_open_dma);
-EXPORT_SYMBOL(DMAbuf_close_dma);
 EXPORT_SYMBOL(DMAbuf_inputintr);
 EXPORT_SYMBOL(DMAbuf_outputintr);

sound/oss/awe_hw.h

-/*
- * sound/oss/awe_hw.h
- *
- * Access routines and definitions for the low level driver for the 
- * Creative AWE32/SB32/AWE64 wave table synth.
- *   version 0.4.4; Jan. 4, 2000
- *
- * Copyright (C) 1996-2000 Takashi Iwai
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
-
-#ifndef AWE_HW_H_DEF
-#define AWE_HW_H_DEF
-
-/*
- * Emu-8000 control registers
- * name(channel)	reg, port
- */
-
-#define awe_cmd_idx(reg,ch)	(((reg)<< 5) | (ch))
-
-#define Data0    0		/* 0x620: doubleword r/w */
-#define Data1    1		/* 0xA20: doubleword r/w */
-#define Data2    2		/* 0xA22: word r/w */
-#define Data3    3		/* 0xE20: word r/w */
-#define Pointer  4		/* 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_0080(ch)	awe_cmd_idx(4,ch), Data0	/* DW: ?? */
-#define AWE_00A0(ch)	awe_cmd_idx(5,ch), Data0	/* DW: ?? */
-#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_HWCF7	awe_cmd_idx(1,14), Data1	/* DW: config dw 7? (not documented) */
-#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?; not documented in ADIP) */
-#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_MAX_CHANNELS	32	/* max midi channels (must >= voices) */
-#define AWE_MAX_LAYERS	AWE_MAX_VOICES	/* maximum number of multiple layers */
-
-#define AWE_DRAM_OFFSET		0x200000
-#define AWE_MAX_DRAM_SIZE	(28 * 1024)	/* 28 MB is max onboard memory */
-
-#endif

sound/oss/awe_wave.h

-/*
- * sound/oss/awe_wave.h
- *
- * Configuration of AWE32/SB32/AWE64 wave table synth driver.
- *   version 0.4.4; Jan. 4, 2000
- *
- * Copyright (C) 1996-1998 Takashi Iwai
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
-
-/*
- * chorus & reverb effects send for FM chip: from 0 to 0xff
- * larger numbers often cause weird sounds.
- */
-
-#define DEF_FM_CHORUS_DEPTH	0x10
-#define DEF_FM_REVERB_DEPTH	0x10
-
-
-/*
- * other compile conditions
- */
-
-/* initialize FM passthrough even without extended RAM */
-#undef AWE_ALWAYS_INIT_FM
-
-/* debug on */
-#define AWE_DEBUG_ON
-
-/* GUS compatible mode */
-#define AWE_HAS_GUS_COMPATIBILITY
-
-/* add MIDI emulation by wavetable */
-#define CONFIG_AWE32_MIDIEMU
-
-/* add mixer control of emu8000 equalizer */
-#undef CONFIG_AWE32_MIXER
-
-/* use new volume calculation method as default */
-#define AWE_USE_NEW_VOLUME_CALC
-
-/* check current volume target for searching empty voices */
-#define AWE_CHECK_VTARGET
-
-/* allow sample sharing */
-#define AWE_ALLOW_SAMPLE_SHARING
-
-/*
- * AWE32 card configuration:
- * uncomment the following lines *ONLY* when auto detection doesn't
- * work properly on your machine.
- */
-
-/*#define AWE_DEFAULT_BASE_ADDR	0x620*/	/* base port address */
-/*#define AWE_DEFAULT_MEM_SIZE	512*/	/* kbytes */
-
-/*
- * AWE driver version number
- */
-#define AWE_MAJOR_VERSION	0
-#define AWE_MINOR_VERSION	4
-#define AWE_TINY_VERSION	4
-#define AWE_VERSION_NUMBER	((AWE_MAJOR_VERSION<<16)|(AWE_MINOR_VERSION<<8)|AWE_TINY_VERSION)
-#define AWEDRV_VERSION		"0.4.4"

sound/oss/cs4281/Makefile

-# Makefile for Cirrus Logic-Crystal CS4281 
-#
-
-obj-$(CONFIG_SOUND_CS4281) += cs4281.o
-
-cs4281-objs += cs4281m.o

sound/oss/cs4281/cs4281_wrapper-24.c

-/*******************************************************************************
-*
-*      "cs4281_wrapper.c" --  Cirrus Logic-Crystal CS4281 linux audio driver.
-*
-*      Copyright (C) 2000,2001  Cirrus Logic Corp.  
-*            -- tom woller (twoller@crystal.cirrus.com) or
-*               (audio@crystal.cirrus.com).
-*
-*      This program is free software; you can redistribute it and/or modify
-*      it under the terms of the GNU General Public License as published by
-*      the Free Software Foundation; either version 2 of the License, or
-*      (at your option) any later version.
-*
-*      This program is distributed in the hope that it will be useful,
-*      but WITHOUT ANY WARRANTY; without even the implied warranty of
-*      MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-*      GNU General Public License for more details.
-*
-*      You should have received a copy of the GNU General Public License
-*      along with this program; if not, write to the Free Software
-*      Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
-*
-* 12/20/00 trw - new file. 
-*
-*******************************************************************************/
-
-#include <linux/spinlock.h>
-
-static int cs4281_resume_null(struct pci_dev *pcidev) { return 0; }
-static int cs4281_suspend_null(struct pci_dev *pcidev, pm_message_t state) { return 0; }
-
-#define free_dmabuf(state, dmabuf) \
-	pci_free_consistent(state->pcidev, \
-			    PAGE_SIZE << (dmabuf)->buforder, \
-			    (dmabuf)->rawbuf, (dmabuf)->dmaaddr);
-#define free_dmabuf2(state, dmabuf) \
-	pci_free_consistent((state)->pcidev, \
-				    PAGE_SIZE << (state)->buforder_tmpbuff, \
-				    (state)->tmpbuff, (state)->dmaaddr_tmpbuff);
-#define cs4x_pgoff(vma) ((vma)->vm_pgoff)
-

sound/oss/cs4281/cs4281pm-24.c

-/*******************************************************************************
-*
-*      "cs4281pm.c" --  Cirrus Logic-Crystal CS4281 linux audio driver.
-*
-*      Copyright (C) 2000,2001  Cirrus Logic Corp.  
-*            -- tom woller (twoller@crystal.cirrus.com) or
-*               (audio@crystal.cirrus.com).
-*
-*      This program is free software; you can redistribute it and/or modify
-*      it under the terms of the GNU General Public License as published by
-*      the Free Software Foundation; either version 2 of the License, or
-*      (at your option) any later version.
-*
-*      This program is distributed in the hope that it will be useful,
-*      but WITHOUT ANY WARRANTY; without even the implied warranty of
-*      MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-*      GNU General Public License for more details.
-*
-*      You should have received a copy of the GNU General Public License
-*      along with this program; if not, write to the Free Software
-*      Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
-*
-* 12/22/00 trw - new file. 
-*
-*******************************************************************************/
-
-#ifndef NOT_CS4281_PM
-#include <linux/pm.h>
-
-static int cs4281_suspend(struct cs4281_state *s);
-static int cs4281_resume(struct cs4281_state *s);
-/* 
-* for now (12/22/00) only enable the pm_register PM support.
-* allow these table entries to be null.
-#define CS4281_SUSPEND_TBL cs4281_suspend_tbl
-#define CS4281_RESUME_TBL cs4281_resume_tbl
-*/
-#define CS4281_SUSPEND_TBL cs4281_suspend_null
-#define CS4281_RESUME_TBL cs4281_resume_null
-
-#else /* CS4281_PM */
-#define CS4281_SUSPEND_TBL cs4281_suspend_null
-#define CS4281_RESUME_TBL cs4281_resume_null
-#endif /* CS4281_PM */
-

sound/oss/dmabuf.c

@@ -1155,36 +1155,6 @@ void DMAbuf_inputintr(int dev)
 	spin_unlock_irqrestore(&dmap->lock,flags);
 }
 
-int DMAbuf_open_dma(int dev)
-{
-	/*
-	 *    NOTE!  This routine opens only the primary DMA channel (output).
-	 */
-	struct audio_operations *adev = audio_devs[dev];
-	int err;
-
-	if ((err = open_dmap(adev, OPEN_READWRITE, adev->dmap_out)) < 0)
-		return -EBUSY;
-	dma_init_buffers(adev->dmap_out);
-	adev->dmap_out->flags |= DMA_ALLOC_DONE;
-	adev->dmap_out->fragment_size = adev->dmap_out->buffsize;
-
-	if (adev->dmap_out->dma >= 0) {
-		unsigned long flags;
-
-		flags=claim_dma_lock();
-		clear_dma_ff(adev->dmap_out->dma);
-		disable_dma(adev->dmap_out->dma);
-		release_dma_lock(flags);
-	}
-	return 0;
-}
-
-void DMAbuf_close_dma(int dev)
-{
-	close_dmap(audio_devs[dev], audio_devs[dev]->dmap_out);
-}
-
 void DMAbuf_init(int dev, int dma1, int dma2)
 {
 	struct audio_operations *adev = audio_devs[dev];

sound/oss/gus.h

-
-#include "ad1848.h"
-
-/*	From gus_card.c */
-int gus_set_midi_irq(int num);
-irqreturn_t gusintr(int irq, void *dev_id, struct pt_regs * dummy);
-
-/*	From gus_wave.c */
-int gus_wave_detect(int baseaddr);
-void gus_wave_init(struct address_info *hw_config);
-void gus_wave_unload (struct address_info *hw_config);
-void gus_voice_irq(void);
-void gus_write8(int reg, unsigned int data);
-void guswave_dma_irq(void);
-void gus_delay(void);
-int gus_default_mixer_ioctl (int dev, unsigned int cmd, void __user *arg);
-void gus_timer_command (unsigned int addr, unsigned int val);
-
-/*	From gus_midi.c */
-void gus_midi_init(struct address_info *hw_config);
-void gus_midi_interrupt(int dummy);
-
-/*	From ics2101.c */
-int ics2101_mixer_init(void);

sound/oss/mpu401.h

@@ -10,5 +10,3 @@ int probe_mpu401(struct address_info *hw_config, struct resource *ports);
 int attach_mpu401(struct address_info * hw_config, struct module *owner);
 void unload_mpu401(struct address_info *hw_info);
 
-int intchk_mpu401(void *dev_id);
-irqreturn_t mpuintr(int irq, void *dev_id, struct pt_regs * dummy);