45 files changed, 8275 insertions, 0 deletions

diff --git a/Documentation/sound/oss/AD1816 b/Documentation/sound/oss/AD1816
new file mode 100644
index 000000000000..14bd8f25d523
--- /dev/null
+++ b/Documentation/sound/oss/AD1816
@@ -0,0 +1,84 @@
+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
diff --git a/Documentation/sound/oss/ALS b/Documentation/sound/oss/ALS
new file mode 100644
index 000000000000..d01ffbfd5808
--- /dev/null
+++ b/Documentation/sound/oss/ALS
@@ -0,0 +1,66 @@
+ALS-007/ALS-100/ALS-200 based sound cards
+=========================================
+
+Support for sound cards based around the Avance Logic
+ALS-007/ALS-100/ALS-200 chip is included.  These chips are a single
+chip PnP sound solution which is mostly hardware compatible with the
+Sound Blaster 16 card, with most differences occurring in the use of
+the mixer registers.  For this reason the ALS code is integrated
+as part of the Sound Blaster 16 driver (adding only 800 bytes to the
+SB16 driver).
+
+To use an ALS sound card under Linux, enable the following options as
+modules in the sound configuration section of the kernel config:
+  - 100% Sound Blaster compatibles (SB16/32/64, ESS, Jazz16) support
+  - FM synthesizer (YM3812/OPL-3) support 
+  - standalone MPU401 support may be required for some cards; for the
+    ALS-007, when using isapnptools, it is required
+Since the ALS-007/100/200 are PnP cards, ISAPnP support should probably be
+compiled in.  If kernel level PnP support is not included, isapnptools will
+be required to configure the card before the sound modules are loaded.
+
+When using kernel level ISAPnP, the kernel should correctly identify and
+configure all resources required by the card when the "sb" module is
+inserted.  Note that the ALS-007 does not have a 16 bit DMA channel and that
+the MPU401 interface on this card uses a different interrupt to the audio
+section.  This should all be correctly configured by the kernel; if problems
+with the MPU401 interface surface, try using the standalone MPU401 module,
+passing "0" as the "sb" module's "mpu_io" module parameter to prevent the
+soundblaster driver attempting to register the MPU401 itself.  The onboard
+synth device can be accessed using the "opl3" module.
+
+If isapnptools is used to wake up the sound card (as in 2.2.x), the settings
+of the card's resources should be passed to the kernel modules ("sb", "opl3"
+and "mpu401") using the module parameters.  When configuring an ALS-007, be
+sure to specify different IRQs for the audio and MPU401 sections - this card
+requires they be different.  For "sb", "io", "irq" and "dma" should be set
+to the same values used to configure the audio section of the card with
+isapnp.  "dma16" should be explicitly set to "-1" for an ALS-007 since this
+card does not have a 16 bit dma channel; if not specified the kernel will
+default to using channel 5 anyway which will cause audio not to work. 
+"mpu_io" should be set to 0.  The "io" parameter of the "opl3" module should
+also agree with the setting used by isapnp.  To get the MPU401 interface
+working on an ALS-007 card, the "mpu401" module will be required since this
+card uses separate IRQs for the audio and MPU401 sections and there is no
+parameter available to pass a different IRQ to the "sb" driver (whose
+inbuilt MPU401 driver would otherwise be fine).  Insert the mpu401 module
+passing appropriate values using the "io" and "irq" parameters.
+
+The resulting sound driver will provide the following capabilities:
+  - 8 and 16 bit audio playback
+  - 8 and 16 bit audio recording
+  - Software selection of record source (line in, CD, FM, mic, master)
+  - Record and playback of midi data via the external MPU-401
+  - Playback of midi data using inbuilt FM synthesizer
+  - Control of the ALS-007 mixer via any OSS-compatible mixer programs. 
+    Controls available are Master (L&R), Line in (L&R), CD (L&R), 
+    DSP/PCM/audio out (L&R), FM (L&R) and Mic in (mono).
+
+Jonathan Woithe
+jwoithe@physics.adelaide.edu.au
+30 March 1998
+
+Modified 2000-02-26 by Dave Forrest, drf5n@virginia.edu to add ALS100/ALS200
+Modified 2000-04-10 by Paul Laufer, pelaufer@csupomona.edu to add ISAPnP info.
+Modified 2000-11-19 by Jonathan Woithe, jwoithe@physics.adelaide.edu.au
+ - updated information for kernel 2.4.x.
diff --git a/Documentation/sound/oss/AWE32 b/Documentation/sound/oss/AWE32
new file mode 100644
index 000000000000..cb179bfeb522
--- /dev/null
+++ b/Documentation/sound/oss/AWE32
@@ -0,0 +1,76 @@
+        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 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
diff --git a/Documentation/sound/oss/AudioExcelDSP16 b/Documentation/sound/oss/AudioExcelDSP16
new file mode 100644
index 000000000000..c0f08922993b
--- /dev/null
+++ b/Documentation/sound/oss/AudioExcelDSP16
@@ -0,0 +1,101 @@
+Driver
+------
+
+Informations about Audio Excel DSP 16 driver can be found in the source
+file aedsp16.c
+Please, read the head of the source before using it. It contain useful
+informations.
+
+Configuration
+-------------
+
+The Audio Excel configuration, is now done with the standard Linux setup.
+You have to configure the sound card (Sound Blaster or Microsoft Sound System)
+and, if you want it, the Roland MPU-401 (do not use the Sound Blaster MPU-401,
+SB-MPU401) in the main driver menu. Activate the lowlevel drivers then select
+the Audio Excel hardware that you want to initialize. Check the IRQ/DMA/MIRQ
+of the Audio Excel initialization: it must be the same as the SBPRO (or MSS)
+setup. If the parameters are different, correct it.
+I you own a Gallant's audio card based on SC-6600, activate the SC-6600 support.
+If you want to change the configuration of the sound board, be sure to
+check off all the configuration items before re-configure it.
+
+Module parameters
+-----------------
+To use this driver as a module, you must configure some module parameters, to
+set up I/O addresses, IRQ lines and DMA channels. Some parameters are
+mandatory while some others are optional. Here a list of parameters you can
+use with this module:
+
+Name            Description
+====            ===========
+MANDATORY
+io              I/O base address (0x220 or 0x240)
+irq             irq line (5, 7, 9, 10 or 11)
+dma             dma channel (0, 1 or 3)
+
+OPTIONAL
+mss_base        I/O base address for activate MSS mode (default SBPRO)
+                (0x530 or 0xE80)
+mpu_base        I/O base address for activate MPU-401 mode
+                (0x300, 0x310, 0x320 or 0x330)
+mpu_irq         MPU-401 irq line (5, 7, 9, 10 or 0)
+
+The /etc/modprobe.conf will have lines like this:
+
+options opl3 io=0x388
+options ad1848 io=0x530 irq=11 dma=3
+options aedsp16 io=0x220 irq=11 dma=3 mss_base=0x530
+
+Where the aedsp16 options are the options for this driver while opl3 and
+ad1848 are the corresponding options for the MSS and OPL3 modules.
+
+Loading MSS and OPL3 needs to pre load the aedsp16 module to set up correctly
+the sound card. Installation dependencies must be written in the modprobe.conf
+file:
+
+install ad1848 /sbin/modprobe aedsp16 && /sbin/modprobe -i ad1848
+install opl3 /sbin/modprobe aedsp16 && /sbin/modprobe -i opl3
+
+Then you must load the sound modules stack in this order:
+sound -> aedsp16 -> [ ad1848, opl3 ]
+
+With the above configuration, loading ad1848 or opl3 modules, will
+automatically load all the sound stack.
+
+Sound cards supported
+---------------------
+This driver supports the SC-6000 and SC-6600 based Gallant's sound card.
+It don't support the Audio Excel DSP 16 III (try the SC-6600 code).
+I'm working on the III version of the card: if someone have useful
+informations about it, please let me know.
+For all the non-supported audio cards, you have to boot MS-DOS (or WIN95)
+activating the audio card with the MS-DOS device driver, then you have to
+<ctrl>-<alt>-<del> and boot Linux.
+Follow these steps:
+
+1) Compile Linux kernel with standard sound driver, using the emulation
+   you want, with the parameters of your audio card,
+   e.g. Microsoft Sound System irq10 dma3
+2) Install your new kernel as the default boot kernel.
+3) Boot MS-DOS and configure the audio card with the boot time device
+   driver, for MSS irq10 dma3 in our example.
+4) <ctrl>-<alt>-<del> and boot Linux. This will maintain the DOS configuration
+   and will boot the new kernel with sound driver. The sound driver will find
+   the audio card and will recognize and attach it.
+
+Reports on User successes
+-------------------------
+
+> Date: Mon, 29 Jul 1996 08:35:40 +0100
+> From: Mr S J Greenaway <sjg95@unixfe.rl.ac.uk>
+> To: riccardo@cdc8g5.cdc.polimi.it (Riccardo Facchetti)
+> Subject: Re: Audio Excel DSP 16 initialization code
+>
+> Just to let you know got my Audio Excel (emulating a MSS) working
+> with my original SB16, thanks for the driver!
+
+
+Last revised: 20 August 1998
+Riccardo Facchetti
+fizban@tin.it
diff --git a/Documentation/sound/oss/CMI8330 b/Documentation/sound/oss/CMI8330
new file mode 100644
index 000000000000..9c439f1a6dba
--- /dev/null
+++ b/Documentation/sound/oss/CMI8330
@@ -0,0 +1,153 @@
+Documentation for CMI 8330 (SoundPRO) 
+-------------------------------------
+Alessandro Zummo <azummo@ita.flashnet.it>
+
+( Be sure to read Documentation/sound/oss/SoundPro too )
+
+
+This adapter is now directly supported by the sb driver.
+
+ The only thing you have to do is to compile the kernel sound
+support as a module and to enable kernel ISAPnP support,
+as shown below.
+
+
+CONFIG_SOUND=m
+CONFIG_SOUND_SB=m
+
+CONFIG_PNP=y
+CONFIG_ISAPNP=y
+
+
+and optionally:
+
+
+CONFIG_SOUND_MPU401=m
+
+ for MPU401 support.
+
+
+(I suggest you to use "make menuconfig" or "make xconfig" 
+ for a more comfortable configuration editing)
+
+
+
+Then you can do
+
+ modprobe sb
+
+and everything will be (hopefully) configured.
+
+You should get something similar in syslog:
+
+sb: CMI8330 detected.
+sb: CMI8330 sb base located at 0x220
+sb: CMI8330 mpu base located at 0x330
+sb: CMI8330 mail reports to Alessandro Zummo <azummo@ita.flashnet.it>
+sb: ISAPnP reports CMI 8330 SoundPRO at i/o 0x220, irq 7, dma 1,5
+
+
+
+
+The old documentation file follows for reference
+purposes.
+
+
+How to enable CMI 8330 (SOUNDPRO) soundchip on Linux
+------------------------------------------
+Stefan Laudat <Stefan.Laudat@asit.ro>
+
+[Note: The CMI 8338 is unrelated and is supported by cmpci.o]
+
+        
+        In order to use CMI8330 under Linux  you just have to use a proper isapnp.conf, a good isapnp and a little bit of patience.  I use isapnp 1.17, but
+you may get a better one I guess at http://www.roestock.demon.co.uk/isapnptools/.
+
+        Of course you will have to compile kernel sound support as module, as shown below:
+
+CONFIG_SOUND=m
+CONFIG_SOUND_OSS=m
+CONFIG_SOUND_SB=m
+CONFIG_SOUND_ADLIB=m
+CONFIG_SOUND_MPU401=m
+# Mikro$chaft sound system (kinda useful here ;))       
+CONFIG_SOUND_MSS=m
+
+        The /etc/isapnp.conf file will be:
+
+<snip below>
+
+
+(READPORT 0x0203)
+(ISOLATE PRESERVE)
+(IDENTIFY *)
+(VERBOSITY 2)
+(CONFLICT (IO FATAL)(IRQ FATAL)(DMA FATAL)(MEM FATAL)) # or WARNING
+(VERIFYLD N)
+
+
+# WSS 
+
+(CONFIGURE CMI0001/16777472 (LD 0
+(IO 0 (SIZE 8) (BASE 0x0530))
+(IO 1 (SIZE 8) (BASE 0x0388))
+(INT 0 (IRQ 7 (MODE +E)))
+(DMA 0 (CHANNEL 0))
+(NAME "CMI0001/16777472[0]{CMI8330/C3D Audio Adapter}")
+(ACT Y)
+))
+
+# MPU
+
+(CONFIGURE CMI0001/16777472 (LD 1
+(IO 0 (SIZE 2) (BASE 0x0330))
+(INT 0 (IRQ 11 (MODE +E)))
+(NAME "CMI0001/16777472[1]{CMI8330/C3D Audio Adapter}")
+(ACT Y)
+))
+
+# Joystick
+
+(CONFIGURE CMI0001/16777472 (LD 2
+(IO 0 (SIZE 8) (BASE 0x0200))
+(NAME "CMI0001/16777472[2]{CMI8330/C3D Audio Adapter}")
+(ACT Y)
+))
+
+# SoundBlaster 
+ 
+(CONFIGURE CMI0001/16777472 (LD 3
+(IO 0 (SIZE 16) (BASE 0x0220))
+(INT 0 (IRQ 5 (MODE +E)))
+(DMA 0 (CHANNEL 1))
+(DMA 1 (CHANNEL 5))
+(NAME "CMI0001/16777472[3]{CMI8330/C3D Audio Adapter}")
+(ACT Y)
+))
+
+
+(WAITFORKEY)
+
+<end of snip>
+
+        The module sequence is trivial:
+
+/sbin/insmod soundcore
+/sbin/insmod sound
+/sbin/insmod uart401
+# insert this first
+/sbin/insmod ad1848 io=0x530 irq=7 dma=0 soundpro=1
+# The sb module is an alternative to the ad1848 (Microsoft Sound System)
+# Anyhow, this is full duplex and has MIDI
+/sbin/insmod sb io=0x220 dma=1 dma16=5 irq=5 mpu_io=0x330
+
+
+
+Alma Chao <elysian@ethereal.torsion.org> suggests the following /etc/modprobe.conf:
+
+alias sound ad1848
+alias synth0 opl3
+options ad1848 io=0x530 irq=7 dma=0 soundpro=1
+options opl3 io=0x388
+
+        
diff --git a/Documentation/sound/oss/CMI8338 b/Documentation/sound/oss/CMI8338
new file mode 100644
index 000000000000..387d058c3f95
--- /dev/null
+++ b/Documentation/sound/oss/CMI8338
@@ -0,0 +1,85 @@
+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.
+
diff --git a/Documentation/sound/oss/CS4232 b/Documentation/sound/oss/CS4232
new file mode 100644
index 000000000000..7d6af7a5c1c2
--- /dev/null
+++ b/Documentation/sound/oss/CS4232
@@ -0,0 +1,23 @@
+To configure the Crystal CS423x sound chip and activate its DSP functions,
+modules may be loaded in this order:
+  
+        modprobe sound
+        insmod ad1848
+        insmod uart401
+        insmod cs4232 io=* irq=* dma=* dma2=*
+  
+This is the meaning of the parameters:
+  
+        io--I/O address of the Windows Sound System (normally 0x534)
+        irq--IRQ of this device
+        dma and dma2--DMA channels (DMA2 may be 0)
+  
+On some cards, the board attempts to do non-PnP setup, and fails.  If you
+have problems, use Linux' PnP facilities. 
+  
+To get MIDI facilities add
+  
+        insmod opl3 io=*
+  
+where "io" is the I/O address of the OPL3 synthesizer. This will be shown
+in /proc/sys/pnp and is normally 0x388.
diff --git a/Documentation/sound/oss/ESS b/Documentation/sound/oss/ESS
new file mode 100644
index 000000000000..bba93b4d2def
--- /dev/null
+++ b/Documentation/sound/oss/ESS
@@ -0,0 +1,34 @@
+Documentation for the ESS AudioDrive chips
+
+In 2.4 kernels the SoundBlaster driver not only tries to detect an ESS chip, it
+tries to detect the type of ESS chip too. The correct detection of the chip 
+doesn't always succeed however, so unless you use the kernel isapnp facilities
+(and you chip is pnp capable) the default behaviour is 2.0 behaviour which
+means: only detect ES688 and ES1688.
+
+All ESS chips now have a recording level setting. This is a need-to-have for
+people who want to use their ESS for recording sound.
+
+Every chip that's detected as a later-than-es1688 chip has a 6 bits logarithmic
+master volume control.
+
+Every chip that's detected as a ES1887 now has Full Duplex support. Made a 
+little testprogram that shows that is works, haven't seen a real program that
+needs this however.
+
+For ESS chips an additional parameter "esstype" can be specified. This controls
+the (auto) detection of the ESS chips. It can have 3 kinds of values:
+
+-1   Act like 2.0 kernels: only detect ES688 or ES1688.
+0        Try to auto-detect the chip (may fail for ES1688)
+688  The chip will be treated as  ES688
+1688  ,,  ,,   ,,  ,,    ,,   ,, ES1688
+1868  ,,  ,,   ,,  ,,    ,,   ,, ES1868
+1869  ,,  ,,   ,,  ,,    ,,   ,, ES1869
+1788  ,,  ,,   ,,  ,,    ,,   ,, ES1788
+1887  ,,  ,,   ,,  ,,    ,,   ,, ES1887
+1888  ,,  ,,   ,,  ,,    ,,   ,, ES1888
+
+Because Full Duplex is supported for ES1887 you can specify a second DMA
+channel by specifying module parameter dma16. It can be one of: 0, 1, 3 or 5.
+
diff --git a/Documentation/sound/oss/ESS1868 b/Documentation/sound/oss/ESS1868
new file mode 100644
index 000000000000..55e922f21bc0
--- /dev/null
+++ b/Documentation/sound/oss/ESS1868
@@ -0,0 +1,55 @@
+Documentation for the ESS1868F AudioDrive PnP sound card
+
+The ESS1868 sound card is a PnP ESS1688-compatible 16-bit sound card.
+
+It should be automatically detected by the Linux Kernel isapnp support when you
+load the sb.o module. Otherwise you should take care of:
+
+  *  The ESS1868 does not allow use of a 16-bit DMA, thus DMA 0, 1, 2, and 3
+     may only be used.
+
+  *  isapnptools version 1.14 does work with ESS1868.  Earlier versions might
+     not.
+
+  *  Sound support MUST be compiled as MODULES, not statically linked
+     into the kernel.
+
+
+NOTE: this is only needed when not using the kernel isapnp support!
+
+For configuring the sound card's I/O addresses, IRQ and DMA, here is a
+sample copy of the isapnp.conf directives regarding the ESS1868:
+
+(CONFIGURE ESS1868/-1 (LD 1
+(IO 0 (BASE 0x0220))
+(IO 1 (BASE 0x0388))
+(IO 2 (BASE 0x0330))
+(DMA 0 (CHANNEL 1))
+(INT 0 (IRQ 5 (MODE +E)))
+(ACT Y)
+))
+
+(for a full working isapnp.conf file, remember the
+(ISOLATE)
+(IDENTIFY *)
+at the beginning and the
+(WAITFORKEY)
+at the end.)
+
+In this setup, the main card I/O is 0x0220, FM synthesizer is 0x0388, and
+the MPU-401 MIDI port is located at 0x0330.  IRQ is IRQ 5, DMA is channel 1.
+
+After configuring the sound card via isapnp, to use the card you must load
+the sound modules with the proper I/O information.  Here is my setup:
+
+# ESS1868F AudioDrive initialization
+
+/sbin/modprobe sound
+/sbin/insmod uart401
+/sbin/insmod sb io=0x220 irq=5 dma=1 dma16=-1
+/sbin/insmod mpu401 io=0x330
+/sbin/insmod opl3 io=0x388
+/sbin/insmod v_midi
+
+opl3 is the FM synthesizer
+/sbin/insmod opl3 io=0x388
diff --git a/Documentation/sound/oss/INSTALL.awe b/Documentation/sound/oss/INSTALL.awe
new file mode 100644
index 000000000000..310f42ca1e83
--- /dev/null
+++ b/Documentation/sound/oss/INSTALL.awe
@@ -0,0 +1,134 @@
+================================================================
+        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!
diff --git a/Documentation/sound/oss/Introduction b/Documentation/sound/oss/Introduction
new file mode 100644
index 000000000000..15d4fb975ac0
--- /dev/null
+++ b/Documentation/sound/oss/Introduction
@@ -0,0 +1,459 @@
+Introduction    Notes on Modular Sound Drivers and Soundcore
+Wade Hampton 
+2/14/2001
+
+Purpose:  
+========
+This document provides some general notes on the modular 
+sound drivers and their configuration, along with the 
+support modules sound.o and soundcore.o.
+
+Note, some of this probably should be added to the Sound-HOWTO!
+
+Note, soundlow.o was present with 2.2 kernels but is not 
+required for 2.4.x kernels.  References have been removed
+to this.
+
+
+Copying:
+========
+none
+
+
+History:
+========
+0.1.0  11/20/1998  First version, draft
+1.0.0  11/1998     Alan Cox changes, incorporation in 2.2.0
+                   as Documentation/sound/oss/Introduction
+1.1.0  6/30/1999   Second version, added notes on making the drivers,
+                   added info on multiple sound cards of similar types,]
+                   added more diagnostics info, added info about esd.
+                   added info on OSS and ALSA.
+1.1.1  19991031    Added notes on sound-slot- and sound-service.
+                        (Alan Cox)
+1.1.2  20000920    Modified for Kernel 2.4 (Christoph Hellwig)
+1.1.3  20010214    Minor notes and corrections (Wade Hampton)
+                   Added examples of sound-slot-0, etc.
+
+
+Modular Sound Drivers:
+======================
+
+Thanks to the GREAT work by Alan Cox (alan@lxorguk.ukuu.org.uk),
+
+[And Oleg Drokin, Thomas Sailer, Andrew Veliath and more than a few 
+ others - not to mention Hannu's original code being designed well
+ enough to cope with that kind of chopping up](Alan)
+
+the standard Linux kernels support a modular sound driver.  From
+Alan's comments in linux/drivers/sound/README.FIRST:
+
+  The modular sound driver patches were funded by Red Hat Software 
+  (www.redhat.com). The sound driver here is thus a modified version of 
+  Hannu's code. Please bear that in mind when considering the appropriate
+  forums for bug reporting.
+
+The modular sound drivers may be loaded via insmod or modprobe.  
+To support all the various sound modules, there are two general 
+support modules that must be loaded first:
+ 
+   soundcore.o:   Top level handler for the sound system, provides
+                  a set of functions for registration of devices
+                  by type.
+
+   sound.o:       Common sound functions required by all modules.
+
+For the specific sound modules (e.g., sb.o for the Soundblaster), 
+read the documentation on that module to determine what options
+are available, for example IRQ, address, DMA.
+
+Warning, the options for different cards sometime use different names 
+for the same or a similar feature (dma1= versus dma16=).  As a last 
+resort, inspect the code (search for MODULE_PARM).
+
+Notes:
+
+1.  There is a new OpenSource sound driver called ALSA which is
+    currently under development:  http://www.alsa-project.org/
+    The ALSA drivers support some newer hardware that may not 
+    be supported by this sound driver and also provide some 
+    additional features.
+
+2.  The commercial OSS driver may be obtained from the site:
+    http://www/opensound.com.  This may be used for cards that
+    are unsupported by the kernel driver, or may be used
+    by other operating systems.  
+
+3.  The enlightenment sound daemon may be used for playing
+    multiple sounds at the same time via a single card, eliminating
+    some of the requirements for multiple sound card systems.  For
+    more information, see:  http://www.tux.org/~ricdude/EsounD.html  
+    The "esd" program may be used with the real-player and mpeg 
+    players like mpg123 and x11amp.  The newer real-player 
+    and some games even include built-in support for ESD!
+
+
+Building the Modules:
+=====================
+
+This document does not provide full details on building the 
+kernel, etc.  The notes below apply only to making the kernel
+sound modules.   If this conflicts with the kernel's README,
+the README takes precedence. 
+
+1.  To make the kernel sound modules, cd to your /usr/src/linux
+    directory (typically) and type make config, make menuconfig, 
+    or make xconfig (to start the command line, dialog, or x-based
+    configuration tool).  
+
+2.  Select the Sound option and a dialog will be displayed.  
+
+3.  Select M (module) for "Sound card support".
+
+4.  Select your sound driver(s) as a module.  For ProAudio, Sound
+    Blaster, etc., select M (module) for OSS sound modules.
+    [thanks to Marvin Stodolsky <stodolsk@erols.com>]A
+
+5.  Make the kernel (e.g., make bzImage), and install the kernel.
+
+6.  Make the modules and install them (make modules; make modules_install).
+
+Note, for 2.5.x kernels, make sure you have the newer module-init-tools 
+installed or modules will not be loaded properly.  2.5.x requires an
+updated module-init-tools.
+
+
+Plug and Play (PnP:
+===================
+
+If the sound card is an ISA PnP card, isapnp may be used
+to configure the card.  See the file isapnp.txt in the 
+directory one level up (e.g., /usr/src/linux/Documentation).
+
+Also the 2.4.x kernels provide PnP capabilities, see the 
+file NEWS in this directory.
+
+PCI sound cards are highly recommended, as they are far 
+easier to configure and from what I have read, they use
+less resources and are more CPU efficient.
+
+
+INSMOD:
+=======
+
+If loading via insmod, the common modules must be loaded in the 
+order below BEFORE loading the other sound modules.  The card-specific
+modules may then be loaded (most require parameters).  For example,
+I use the following via a shell script to load my SoundBlaster:
+
+SB_BASE=0x240
+SB_IRQ=9
+SB_DMA=3
+SB_DMA2=5
+SB_MPU=0x300
+#
+echo Starting sound
+/sbin/insmod soundcore
+/sbin/insmod sound  
+#
+echo Starting sound blaster....
+/sbin/insmod uart401
+/sbin/insmod sb io=$SB_BASE irq=$SB_IRQ dma=$SB_DMA dma16=$SB_DMA2 mpu_io=$SB_MP
+
+When using sound as a module, I typically put these commands
+in a file such as /root/soundon.sh.
+
+
+MODPROBE:
+=========
+
+If loading via modprobe, these common files are automatically loaded 
+when requested by modprobe.  For example, my /etc/modprobe.conf contains:
+
+alias sound sb 
+options sb io=0x240 irq=9 dma=3 dma16=5 mpu_io=0x300
+
+All you need to do to load the module is:
+
+        /sbin/modprobe sb
+
+
+Sound Status:
+=============
+
+The status of sound may be read/checked by:
+        cat (anyfile).au >/dev/audio
+
+[WWH:  This may not work properly for SoundBlaster PCI 128 cards
+such as the es1370/1 (see the es1370/1 files in this directory) 
+as they do not automatically support uLaw on /dev/audio.]
+
+The status of the modules and which modules depend on 
+which other modules may be checked by:
+        /sbin/lsmod
+
+/sbin/lsmod should show something like the following:
+        sb                     26280   0 
+        uart401                 5640   0  [sb]
+        sound                  57112   0  [sb uart401]
+        soundcore               1968   8  [sb sound]
+
+
+Removing Sound: 
+=============== 
+
+Sound may be removed by using /sbin/rmmod in the reverse order
+in which you load the modules.  Note, if a program has a sound device
+open (e.g., xmixer), that module (and the modules on which it 
+depends) may not be unloaded.
+
+For example, I use the following to remove my Soundblaster (rmmod
+in the reverse order in which I loaded the modules):
+
+/sbin/rmmod sb
+/sbin/rmmod uart401
+/sbin/rmmod sound
+/sbin/rmmod soundcore
+
+When using sound as a module, I typically put these commands
+in a script such as /root/soundoff.sh.
+
+
+Removing Sound for use with OSS: 
+================================ 
+
+If you get really stuck or have a card that the kernel modules
+will not support, you can get a commercial sound driver from
+http://www.opensound.com.  Before loading the commercial sound
+driver, you should do the following:
+
+1.  remove sound modules (detailed above)
+2.  remove the sound modules from /etc/modprobe.conf
+3.  move the sound modules from /lib/modules/<kernel>/misc
+    (for example, I make a /lib/modules/<kernel>/misc/tmp
+    directory and copy the sound module files to that 
+    directory).
+
+
+Multiple Sound Cards:
+=====================
+
+The sound drivers will support multiple sound cards and there
+are some great applications like multitrack that support them.  
+Typically, you need two sound cards of different types.  Note, this
+uses more precious interrupts and DMA channels and sometimes 
+can be a configuration nightmare.  I have heard reports of 3-4
+sound cards (typically I only use 2).  You can sometimes use
+multiple PCI sound cards of the same type.
+
+On my machine I have two sound cards (cs4232 and Soundblaster Vibra
+16).  By loading sound as modules, I can control which is the first
+sound device (/dev/dsp, /dev/audio, /dev/mixer) and which is 
+the second.  Normally, the cs4232 (Dell sound on the motherboard) 
+would be the first sound device, but I prefer the Soundblaster.  
+All you have to do is to load the one you want as /dev/dsp 
+first (in my case "sb") and then load the other one
+(in my case "cs4232").
+
+If you have two cards of the same type that are jumpered 
+cards or different PnP revisions, you may load the same 
+module twice.  For example, I have a SoundBlaster vibra 16
+and an older SoundBlaster 16 (jumpers).  To load the module
+twice, you need to do the following:
+
+1.  Copy the sound modules to a new name.  For example
+    sb.o could be copied (or symlinked) to sb1.o for the
+    second SoundBlaster.
+
+2.  Make a second entry in /etc/modprobe.conf, for example,
+    sound1 or sb1.  This second entry should refer to the
+    new module names for example sb1, and should include
+    the I/O, etc. for the second sound card.
+
+3.  Update your soundon.sh script, etc.
+
+Warning:  I have never been able to get two PnP sound cards of the
+same type to load at the same time.  I have tried this several times
+with the Soundblaster Vibra 16 cards.  OSS has indicated that this
+is a PnP problem....  If anyone has any luck doing this, please 
+send me an E-MAIL.  PCI sound cards should not have this problem.a
+Since this was originally release, I have received a couple of 
+mails from people who have accomplished this!
+
+NOTE: In Linux 2.4 the Sound Blaster driver (and only this one yet)
+supports multiple cards with one module by default.
+Read the file 'Soundblaster' in this directory for details.
+
+
+Sound Problems:
+===============
+
+First RTFM (including the troubleshooting section 
+in the Sound-HOWTO). 
+
+1)  If you are having problems loading the modules (for
+    example, if you get device conflict errors) try the
+    following:
+
+  A)  If you have Win95 or NT on the same computer,  
+      write down what addresses, IRQ, and DMA channels
+      those were using for the same hardware.  You probably 
+      can use these addresses, IRQs, and DMA channels.
+      You should really do this BEFORE attempting to get
+      sound working!
+  
+  B)  Check (cat) /proc/interrupts, /proc/ioports,
+      and /proc/dma.  Are you trying to use an address,
+      IRQ or DMA port that another device is using?
+  
+  C)  Check (cat) /proc/isapnp
+  
+  D)  Inspect your /var/log/messages file.  Often that will 
+      indicate what IRQ or IO port could not be obtained.
+  
+  E)  Try another port or IRQ.  Note this may involve 
+      using the PnP tools to move the sound card to 
+      another location.  Sometimes this is the only way 
+      and it is more or less trial and error.
+
+2)  If you get motor-boating (the same sound or part of a 
+    sound clip repeated), you probably have either an IRQ
+    or DMA conflict.  Move the card to another IRQ or DMA
+    port.  This has happened to me when playing long files 
+    when I had an IRQ conflict.
+
+3.  If you get dropouts or pauses when playing high sample
+    rate files such as using mpg123 or x11amp/xmms, you may 
+    have too slow of a CPU and may have to use the options to 
+    play the files at 1/2 speed.  For example, you may use
+    the -2 or -4 option on mpg123.  You may also get this
+    when trying to play mpeg files stored on a CD-ROM
+    (my Toshiba T8000 PII/366 sometimes has this problem).
+
+4.  If you get "cannot access device" errors, your /dev/dsp
+    files, etc. may be set to owner root, mode 600.  You 
+    may have to use the command:
+      chmod 666 /dev/dsp /dev/mixer /dev/audio
+
+5.  If you get "device busy" errors, another program has the
+    sound device open.  For example, if using the Enlightenment
+    sound daemon "esd", the "esd" program has the sound device.
+    If using "esd", please RTFM the docs on ESD.  For example,
+    esddsp <program> may be used to play files via a non-esd
+    aware program.
+
+6)  Ask for help on the sound list or send E-MAIL to the
+    sound driver author/maintainer.
+
+7)  Turn on debug in drivers/sound/sound_config.h (DEB, DDB, MDB).
+
+8)  If the system reports insufficient DMA memory then you may want to
+    load sound with the "dmabufs=1" option. Or in /etc/conf.modules add
+        
+        preinstall sound dmabufs=1
+
+    This makes the sound system allocate its buffers and hang onto them.
+
+    You may also set persistent DMA when building a 2.4.x kernel.
+
+
+Configuring Sound:
+==================
+
+There are several ways of configuring your sound:
+
+1)  On the kernel command line (when using the sound driver(s)
+    compiled in the kernel). Check the driver source and
+    documentation for details.
+
+2)  On the command line when using insmod or in a bash script
+    using command line calls to load sound.
+
+3)  In /etc/modprobe.conf when using modprobe.
+
+4)  Via Red Hat's GPL'd /usr/sbin/sndconfig program (text based).
+
+5)  Via the OSS soundconf program (with the commercial version
+    of the OSS driver.
+
+6)  By just loading the module and let isapnp do everything relevant
+    for you. This works only with a few drivers yet and - of course -
+    only with isapnp hardware.
+
+And I am sure, several other ways.  
+
+Anyone want to write a linuxconf module for configuring sound?
+
+
+Module Loading:
+===============
+
+When a sound card is first referenced and sound is modular, the sound system
+will ask for the sound devices to be loaded. Initially it requests that
+the driver for the sound system is loaded. It then will ask for 
+sound-slot-0, where 0 is the first sound card. (sound-slot-1 the second and
+so on). Thus you can do
+
+alias sound-slot-0 sb
+
+To load a soundblaster at this point. If the slot loading does not provide
+the desired device - for example a soundblaster does not directly provide
+a midi synth in all cases then it will request "sound-service-0-n" where n
+is
+
+  0     Mixer
+
+  2     MIDI
+
+  3, 4  DSP audio
+
+
+For example, I use the following to load my Soundblaster PCI 128
+(ES 1371) card first, followed by my SoundBlaster Vibra 16 card,
+then by my TV card:
+
+# Load the Soundblaster PCI 128 as /dev/dsp, /dev/dsp1, /dev/mixer
+alias sound-slot-0 es1371
+
+# Load the Soundblaster Vibra 16 as /dev/dsp2, /dev/mixer1
+alias sound-slot-1 sb
+options sb io=0x240 irq=5 dma=1 dma16=5 mpu_io=0x330
+
+# Load the BTTV (TV card) as /dev/mixer2
+alias sound-slot-2 bttv
+alias sound-service-2-0 tvmixer
+
+pre-install bttv  modprobe tuner ; modprobe tvmixer
+pre-install tvmixer modprobe msp3400; modprobe tvaudio 
+options tuner debug=0 type=8 
+options bttv  card=0 radio=0 pll=0
+
+
+For More Information (RTFM):
+============================
+1)  Information on kernel modules: manual pages for insmod and modprobe.
+
+2)  Information on PnP, RTFM manual pages for isapnp.
+
+3)  Sound-HOWTO and Sound-Playing-HOWTO.
+
+4)  OSS's WWW site at http://www.opensound.com.
+
+5)  All the files in Documentation/sound.
+
+6)  The comments and code in linux/drivers/sound.
+
+7)  The sndconfig and rhsound documentation from Red Hat.
+
+8)  The Linux-sound mailing list:  sound-list@redhat.com.
+
+9)  Enlightenment documentation (for info on esd)
+    http://www.tux.org/~ricdude/EsounD.html.
+
+10) ALSA home page:  http://www.alsa-project.org/
+
+
+Contact Information:
+====================
+Wade Hampton:  (whampton@staffnet.com)
+
diff --git a/Documentation/sound/oss/MAD16 b/Documentation/sound/oss/MAD16
new file mode 100644
index 000000000000..865dbd848742
--- /dev/null
+++ b/Documentation/sound/oss/MAD16
@@ -0,0 +1,56 @@
+(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
diff --git a/Documentation/sound/oss/Maestro b/Documentation/sound/oss/Maestro
new file mode 100644
index 000000000000..4a80eb3f8e00
--- /dev/null
+++ b/Documentation/sound/oss/Maestro
@@ -0,0 +1,123 @@
+        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.
diff --git a/Documentation/sound/oss/Maestro3 b/Documentation/sound/oss/Maestro3
new file mode 100644
index 000000000000..a113718e8034
--- /dev/null
+++ b/Documentation/sound/oss/Maestro3
@@ -0,0 +1,92 @@
+        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.
diff --git a/Documentation/sound/oss/MultiSound b/Documentation/sound/oss/MultiSound
new file mode 100644
index 000000000000..e4a18bb7f73a
--- /dev/null
+++ b/Documentation/sound/oss/MultiSound
@@ -0,0 +1,1137 @@
+#! /bin/sh
+#
+#  Turtle Beach MultiSound Driver Notes
+#  -- Andrew Veliath <andrewtv@usa.net>
+#
+#  Last update:                      September 10, 1998
+#  Corresponding msnd driver:        0.8.3
+#
+# ** This file is a README (top part) and shell archive (bottom part).
+#    The corresponding archived utility sources can be unpacked by
+#    running `sh MultiSound' (the utilities are only needed for the
+#    Pinnacle and Fiji cards). **
+#
+#
+#  -=-=- Getting Firmware -=-=-
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#  
+#  See the section `Obtaining and Creating Firmware Files' in this
+#  document for instructions on obtaining the necessary firmware
+#  files.
+#  
+#  
+#  Supported Features
+#  ~~~~~~~~~~~~~~~~~~
+#  
+#  Currently, full-duplex digital audio (/dev/dsp only, /dev/audio is
+#  not currently available) and mixer functionality (/dev/mixer) are
+#  supported (memory mapped digital audio is not yet supported).
+#  Digital transfers and monitoring can be done as well if you have
+#  the digital daughterboard (see the section on using the S/PDIF port
+#  for more information).
+#
+#  Support for the Turtle Beach MultiSound Hurricane architecture is
+#  composed of the following modules (these can also operate compiled
+#  into the kernel):
+#  
+#  msnd               - MultiSound base (requires soundcore)
+#
+#  msnd_classic       - Base audio/mixer support for Classic, Monetery and
+#                       Tahiti cards
+#
+#  msnd_pinnacle      - Base audio/mixer support for Pinnacle and Fiji cards
+#  
+#  
+#  Important Notes - Read Before Using
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#  
+#  The firmware files are not included (may change in future).  You
+#  must obtain these images from Turtle Beach (they are included in
+#  the MultiSound Development Kits), and place them in /etc/sound for
+#  example, and give the full paths in the Linux configuration.  If
+#  you are compiling in support for the MultiSound driver rather than
+#  using it as a module, these firmware files must be accessible
+#  during kernel compilation.
+#
+#  Please note these files must be binary files, not assembler.  See
+#  the section later in this document for instructions to obtain these
+#  files.
+#  
+#  
+#  Configuring Card Resources
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+#  ** This section is very important, as your card may not work at all
+#     or your machine may crash if you do not do this correctly. **
+#
+#  * Classic/Monterey/Tahiti
+#  
+#  These cards are configured through the driver msnd_classic.  You must
+#  know the io port, then the driver will select the irq and memory resources
+#  on the card.  It is up to you to know if these are free locations or now,
+#  a conflict can lock the machine up.
+#
+#  * Pinnacle/Fiji
+#
+#  The Pinnacle and Fiji cards have an extra config port, either
+#  0x250, 0x260 or 0x270.  This port can be disabled to have the card
+#  configured strictly through PnP, however you lose the ability to
+#  access the IDE controller and joystick devices on this card when
+#  using PnP.  The included pinnaclecfg program in this shell archive
+#  can be used to configure the card in non-PnP mode, and in PnP mode
+#  you can use isapnptools.  These are described briefly here.
+#
+#  pinnaclecfg is not required; you can use the msnd_pinnacle module
+#  to fully configure the card as well.  However, pinnaclecfg can be
+#  used to change the resource values of a particular device after the
+#  msnd_pinnacle module has been loaded.  If you are compiling the
+#  driver into the kernel, you must set these values during compile
+#  time, however other peripheral resource values can be changed with
+#  the pinnaclecfg program after the kernel is loaded.
+#
+#
+#  *** PnP mode
+#  
+#  Use pnpdump to obtain a sample configuration if you can; I was able
+#  to obtain one with the command `pnpdump 1 0x203' -- this may vary
+#  for you (running pnpdump by itself did not work for me).  Then,
+#  edit this file and use isapnp to uncomment and set the card values.
+#  Use these values when inserting the msnd_pinnacle module.  Using
+#  this method, you can set the resources for the DSP and the Kurzweil
+#  synth (Pinnacle).  Since Linux does not directly support PnP
+#  devices, you may have difficulty when using the card in PnP mode
+#  when it the driver is compiled into the kernel.  Using non-PnP mode
+#  is preferable in this case.
+#
+#  Here is an example mypinnacle.conf for isapnp that sets the card to
+#  io base 0x210, irq 5 and mem 0xd8000, and also sets the Kurzweil
+#  synth to 0x330 and irq 9 (may need editing for your system):
+#
+#  (READPORT 0x0203)
+#  (CSN 2)
+#  (IDENTIFY *)
+#  
+#  # DSP
+#  (CONFIGURE BVJ0440/-1 (LD 0
+#          (INT 0 (IRQ 5 (MODE +E))) (IO 0 (BASE 0x0210)) (MEM 0 (BASE 0x0d8000))
+#          (ACT Y)))
+#  
+#  # Kurzweil Synth (Pinnacle Only)
+#  (CONFIGURE BVJ0440/-1 (LD 1
+#          (IO 0 (BASE 0x0330)) (INT 0 (IRQ 9 (MODE +E)))
+#          (ACT Y)))
+#  
+#  (WAITFORKEY)
+#
+#
+#  *** Non-PnP mode
+#  
+#  The second way is by running the card in non-PnP mode.  This
+#  actually has some advantages in that you can access some other
+#  devices on the card, such as the joystick and IDE controller.  To
+#  configure the card, unpack this shell archive and build the
+#  pinnaclecfg program.  Using this program, you can assign the
+#  resource values to the card's devices, or disable the devices.  As
+#  an alternative to using pinnaclecfg, you can specify many of the
+#  configuration values when loading the msnd_pinnacle module (or
+#  during kernel configuration when compiling the driver into the
+#  kernel).
+#
+#  If you specify cfg=0x250 for the msnd_pinnacle module, it
+#  automatically configure the card to the given io, irq and memory
+#  values using that config port (the config port is jumper selectable
+#  on the card to 0x250, 0x260 or 0x270).
+#
+#  See the `msnd_pinnacle Additional Options' section below for more
+#  information on these parameters (also, if you compile the driver
+#  directly into the kernel, these extra parameters can be useful
+#  here).
+#
+#
+# ** It is very easy to cause problems in your machine if you choose a
+#    resource value which is incorrect. **
+#  
+#
+#  Examples
+#  ~~~~~~~~
+#  
+#  * MultiSound Classic/Monterey/Tahiti:
+#  
+#  modprobe soundcore
+#  insmod msnd
+#  insmod msnd_classic io=0x290 irq=7 mem=0xd0000
+#  
+#  * MultiSound Pinnacle in PnP mode:
+#  
+#  modprobe soundcore
+#  insmod msnd
+#  isapnp mypinnacle.conf
+#  insmod msnd_pinnacle io=0x210 irq=5 mem=0xd8000 <-- match mypinnacle.conf values
+#  
+#  * MultiSound Pinnacle in non-PnP mode (replace 0x250 with your configuration port,
+#    one of 0x250, 0x260 or 0x270):
+#  
+#  insmod soundcore
+#  insmod msnd
+#  insmod msnd_pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000
+#  
+# * To use the MPU-compatible Kurzweil synth on the Pinnacle in PnP
+#   mode, add the following (assumes you did `isapnp mypinnacle.conf'):
+#  
+#  insmod sound
+#  insmod mpu401 io=0x330 irq=9                    <-- match mypinnacle.conf values
+#  
+# * To use the MPU-compatible Kurzweil synth on the Pinnacle in non-PnP
+#   mode, add the following.  Note how we first configure the peripheral's
+#   resources, _then_ install a Linux driver for it:
+#  
+#  insmod sound
+#  pinnaclecfg 0x250 mpu 0x330 9
+#  insmod mpu401 io=0x330 irq=9
+#
+#  -- OR you can use the following sequence without pinnaclecfg in non-PnP mode:
+#
+#  insmod soundcore
+#  insmod msnd
+#  insmod msnd_pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000 mpu_io=0x330 mpu_irq=9
+#  insmod sound
+#  insmod mpu401 io=0x330 irq=9
+#
+# * To setup the joystick port on the Pinnacle in non-PnP mode (though
+#   you have to find the actual Linux joystick driver elsewhere), you
+#   can use pinnaclecfg:
+#
+#   pinnaclecfg 0x250 joystick 0x200
+#
+#  -- OR you can configure this using msnd_pinnacle with the following:
+#
+#  insmod soundcore
+#  insmod msnd
+#  insmod msnd_pinnacle cfg=0x250 io=0x290 irq=5 mem=0xd0000 joystick_io=0x200
+#
+#  
+#  msnd_classic, msnd_pinnacle Required Options
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#  
+#  If the following options are not given, the module will not load.
+#  Examine the kernel message log for informative error messages.
+#  WARNING--probing isn't supported so try to make sure you have the
+#  correct shared memory area, otherwise you may experience problems.
+#  
+#  io                   I/O base of DSP, e.g. io=0x210
+#  irq                  IRQ number, e.g. irq=5
+#  mem                  Shared memory area, e.g. mem=0xd8000
+#  
+#  
+#  msnd_classic, msnd_pinnacle Additional Options
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#  
+#  fifosize             The digital audio FIFOs, in kilobytes.  If not
+#                       specified, the default will be used.  Increasing
+#                       this value will reduce the chance of a FIFO
+#                       underflow at the expense of increasing overall
+#                       latency.  For example, fifosize=512 will
+#                       allocate 512kB read and write FIFOs (1MB total).
+#                       While this may reduce dropouts, a heavy machine
+#                       load will undoubtedly starve the FIFO of data
+#                       and you will eventually get dropouts.  One
+#                       option is to alter the scheduling priority of
+#                       the playback process, using `nice' or some form
+#                       of POSIX soft real-time scheduling.
+#
+#  calibrate_signal     Setting this to one calibrates the ADCs to the
+#                       signal, zero calibrates to the card (defaults
+#                       to zero).
+#  
+#  
+#  msnd_pinnacle Additional Options
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+#  digital              Specify digital=1 to enable the S/PDIF input
+#                       if you have the digital daughterboard
+#                       adapter. This will enable access to the
+#                       DIGITAL1 input for the soundcard in the mixer.
+#                       Some mixer programs might have trouble setting
+#                       the DIGITAL1 source as an input.  If you have
+#                       trouble, you can try the setdigital.c program
+#                       at the bottom of this document.
+#
+#  cfg                  Non-PnP configuration port for the Pinnacle
+#                       and Fiji (typically 0x250, 0x260 or 0x270,
+#                       depending on the jumper configuration).  If
+#                       this option is omitted, then it is assumed
+#                       that the card is in PnP mode, and that the
+#                       specified DSP resource values are already
+#                       configured with PnP (i.e. it won't attempt to
+#                       do any sort of configuration).
+#
+#  When the Pinnacle is in non-PnP mode, you can use the following
+#  options to configure particular devices.  If a full specification
+#  for a device is not given, then the device is not configured.  Note
+#  that you still must use a Linux driver for any of these devices
+#  once their resources are setup (such as the Linux joystick driver,
+#  or the MPU401 driver from OSS for the Kurzweil synth).
+#
+#  mpu_io               I/O port of MPU (on-board Kurzweil synth)
+#  mpu_irq              IRQ of MPU (on-board Kurzweil synth)
+#  ide_io0              First I/O port of IDE controller
+#  ide_io1              Second I/O port of IDE controller
+#  ide_irq              IRQ IDE controller
+#  joystick_io          I/O port of joystick
+#  
+#  
+#  Obtaining and Creating Firmware Files
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#  
+#       For the Classic/Tahiti/Monterey
+#       ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ 
+#  
+#  Download to /tmp and unzip the following file from Turtle Beach:
+#  
+#       ftp://ftp.voyetra.com/pub/tbs/msndcl/msndvkit.zip
+#  
+#  When unzipped, unzip the file named MsndFiles.zip.  Then copy the
+#  following firmware files to /etc/sound (note the file renaming):
+#  
+#    cp DSPCODE/MSNDINIT.BIN /etc/sound/msndinit.bin
+#    cp DSPCODE/MSNDPERM.REB /etc/sound/msndperm.bin
+#  
+#  When configuring the Linux kernel, specify /etc/sound/msndinit.bin and
+#  /etc/sound/msndperm.bin for the two firmware files (Linux kernel
+#  versions older than 2.2 do not ask for firmware paths, and are
+#  hardcoded to /etc/sound).
+#
+#  If you are compiling the driver into the kernel, these files must
+#  be accessible during compilation, but will not be needed later.
+#  The files must remain, however, if the driver is used as a module.
+#  
+#  
+#       For the Pinnacle/Fiji
+#       ~~~~~~~~~~~~~~~~~~~~~
+#  
+#  Download to /tmp and unzip the following file from Turtle Beach (be
+#  sure to use the entire URL; some have had trouble navigating to the
+#  URL):
+#  
+#       ftp://ftp.voyetra.com/pub/tbs/pinn/pnddk100.zip
+#
+#  Unpack this shell archive, and run make in the created directory
+#  (you need a C compiler and flex to build the utilities).  This
+#  should give you the executables conv, pinnaclecfg and setdigital.
+#  conv is only used temporarily here to create the firmware files,
+#  while pinnaclecfg is used to configure the Pinnacle or Fiji card in
+#  non-PnP mode, and setdigital can be used to set the S/PDIF input on
+#  the mixer (pinnaclecfg and setdigital should be copied to a
+#  convenient place, possibly run during system initialization).
+#
+#  To generating the firmware files with the `conv' program, we create
+#  the binary firmware files by doing the following conversion
+#  (assuming the archive unpacked into a directory named PINNDDK):
+#  
+#    ./conv < PINNDDK/dspcode/pndspini.asm > /etc/sound/pndspini.bin
+#    ./conv < PINNDDK/dspcode/pndsperm.asm > /etc/sound/pndsperm.bin
+#  
+#  The conv (and conv.l) program is not needed after conversion and can
+#  be safely deleted.  Then, when configuring the Linux kernel, specify
+#  /etc/sound/pndspini.bin and /etc/sound/pndsperm.bin for the two
+#  firmware files (Linux kernel versions older than 2.2 do not ask for
+#  firmware paths, and are hardcoded to /etc/sound).
+#  
+#  If you are compiling the driver into the kernel, these files must
+#  be accessible during compilation, but will not be needed later.
+#  The files must remain, however, if the driver is used as a module.
+#
+#  
+#  Using Digital I/O with the S/PDIF Port
+#  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+#
+#  If you have a Pinnacle or Fiji with the digital daughterboard and
+#  want to set it as the input source, you can use this program if you
+#  have trouble trying to do it with a mixer program (be sure to
+#  insert the module with the digital=1 option, or say Y to the option
+#  during compiled-in kernel operation).  Upon selection of the S/PDIF
+#  port, you should be able monitor and record from it.
+#
+#  There is something to note about using the S/PDIF port.  Digital
+#  timing is taken from the digital signal, so if a signal is not
+#  connected to the port and it is selected as recording input, you
+#  will find PCM playback to be distorted in playback rate.  Also,
+#  attempting to record at a sampling rate other than the DAT rate may
+#  be problematic (i.e. trying to record at 8000Hz when the DAT signal
+#  is 44100Hz).  If you have a problem with this, set the recording
+#  input to analog if you need to record at a rate other than that of
+#  the DAT rate.
+#
+#
+#  -- Shell archive attached below, just run `sh MultiSound' to extract.
+#     Contains Pinnacle/Fiji utilities to convert firmware, configure
+#     in non-PnP mode, and select the DIGITAL1 input for the mixer.
+#
+#
+#!/bin/sh
+# This is a shell archive (produced by GNU sharutils 4.2).
+# To extract the files from this archive, save it to some FILE, remove
+# everything before the `!/bin/sh' line above, then type `sh FILE'.
+#
+# Made on 1998-12-04 10:07 EST by <andrewtv@ztransform.velsoft.com>.
+# Source directory was `/home/andrewtv/programming/pinnacle/pinnacle'.
+#
+# Existing files will *not* be overwritten unless `-c' is specified.
+#
+# This shar contains:
+# length mode       name
+# ------ ---------- ------------------------------------------
+#   2046 -rw-rw-r-- MultiSound.d/setdigital.c
+#  10235 -rw-rw-r-- MultiSound.d/pinnaclecfg.c
+#    106 -rw-rw-r-- MultiSound.d/Makefile
+#    141 -rw-rw-r-- MultiSound.d/conv.l
+#   1472 -rw-rw-r-- MultiSound.d/msndreset.c
+#
+save_IFS="${IFS}"
+IFS="${IFS}:"
+gettext_dir=FAILED
+locale_dir=FAILED
+first_param="$1"
+for dir in $PATH
+do
+  if test "$gettext_dir" = FAILED && test -f $dir/gettext \
+     && ($dir/gettext --version >/dev/null 2>&1)
+  then
+    set `$dir/gettext --version 2>&1`
+    if test "$3" = GNU
+    then
+      gettext_dir=$dir
+    fi
+  fi
+  if test "$locale_dir" = FAILED && test -f $dir/shar \
+     && ($dir/shar --print-text-domain-dir >/dev/null 2>&1)
+  then
+    locale_dir=`$dir/shar --print-text-domain-dir`
+  fi
+done
+IFS="$save_IFS"
+if test "$locale_dir" = FAILED || test "$gettext_dir" = FAILED
+then
+  echo=echo
+else
+  TEXTDOMAINDIR=$locale_dir
+  export TEXTDOMAINDIR
+  TEXTDOMAIN=sharutils
+  export TEXTDOMAIN
+  echo="$gettext_dir/gettext -s"
+fi
+touch -am 1231235999 $$.touch >/dev/null 2>&1
+if test ! -f 1231235999 && test -f $$.touch; then
+  shar_touch=touch
+else
+  shar_touch=:
+  echo
+  $echo 'WARNING: not restoring timestamps.  Consider getting and'
+  $echo "installing GNU \`touch', distributed in GNU File Utilities..."
+  echo
+fi
+rm -f 1231235999 $$.touch
+#
+if mkdir _sh01426; then
+  $echo 'x -' 'creating lock directory'
+else
+  $echo 'failed to create lock directory'
+  exit 1
+fi
+# ============= MultiSound.d/setdigital.c ==============
+if test ! -d 'MultiSound.d'; then
+  $echo 'x -' 'creating directory' 'MultiSound.d'
+  mkdir 'MultiSound.d'
+fi
+if test -f 'MultiSound.d/setdigital.c' && test "$first_param" != -c; then
+  $echo 'x -' SKIPPING 'MultiSound.d/setdigital.c' '(file already exists)'
+else
+  $echo 'x -' extracting 'MultiSound.d/setdigital.c' '(text)'
+  sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/setdigital.c' &&
+/*********************************************************************
+X *
+X * setdigital.c - sets the DIGITAL1 input for a mixer
+X *
+X * Copyright (C) 1998 Andrew Veliath
+X *
+X * This program is free software; you can redistribute it and/or modify
+X * it under the terms of the GNU General Public License as published by
+X * the Free Software Foundation; either version 2 of the License, or
+X * (at your option) any later version.
+X *
+X * This program is distributed in the hope that it will be useful,
+X * but WITHOUT ANY WARRANTY; without even the implied warranty of
+X * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+X * GNU General Public License for more details.
+X *
+X * You should have received a copy of the GNU General Public License
+X * along with this program; if not, write to the Free Software
+X * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+X *
+X ********************************************************************/
+X
+#include <stdio.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <sys/ioctl.h>
+#include <sys/soundcard.h>
+X
+int main(int argc, char *argv[])
+{
+X       int fd;
+X       unsigned long recmask, recsrc;
+X
+X       if (argc != 2) {
+X               fprintf(stderr, "usage: setdigital <mixer device>\n");
+X               exit(1);
+X       }
+X
+X       if ((fd = open(argv[1], O_RDWR)) < 0) {
+X               perror(argv[1]);
+X               exit(1);
+X       }
+X
+X       if (ioctl(fd, SOUND_MIXER_READ_RECMASK, &recmask) < 0) {
+X               fprintf(stderr, "error: ioctl read recording mask failed\n");
+X               perror("ioctl");
+X               close(fd);
+X               exit(1);
+X       }
+X
+X       if (!(recmask & SOUND_MASK_DIGITAL1)) {
+X               fprintf(stderr, "error: cannot find DIGITAL1 device in mixer\n");
+X               close(fd);
+X               exit(1);
+X       }
+X
+X       if (ioctl(fd, SOUND_MIXER_READ_RECSRC, &recsrc) < 0) {
+X               fprintf(stderr, "error: ioctl read recording source failed\n");
+X               perror("ioctl");
+X               close(fd);
+X               exit(1);
+X       }
+X
+X       recsrc |= SOUND_MASK_DIGITAL1;
+X       
+X       if (ioctl(fd, SOUND_MIXER_WRITE_RECSRC, &recsrc) < 0) {
+X               fprintf(stderr, "error: ioctl write recording source failed\n");
+X               perror("ioctl");
+X               close(fd);
+X               exit(1);
+X       }
+X
+X       close(fd);
+X       
+X       return 0;
+}
+SHAR_EOF
+  $shar_touch -am 1204092598 'MultiSound.d/setdigital.c' &&
+  chmod 0664 'MultiSound.d/setdigital.c' ||
+  $echo 'restore of' 'MultiSound.d/setdigital.c' 'failed'
+  if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \
+  && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then
+    md5sum -c << SHAR_EOF >/dev/null 2>&1 \
+    || $echo 'MultiSound.d/setdigital.c:' 'MD5 check failed'
+e87217fc3e71288102ba41fd81f71ec4  MultiSound.d/setdigital.c
+SHAR_EOF
+  else
+    shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/setdigital.c'`"
+    test 2046 -eq "$shar_count" ||
+    $echo 'MultiSound.d/setdigital.c:' 'original size' '2046,' 'current size' "$shar_count!"
+  fi
+fi
+# ============= MultiSound.d/pinnaclecfg.c ==============
+if test -f 'MultiSound.d/pinnaclecfg.c' && test "$first_param" != -c; then
+  $echo 'x -' SKIPPING 'MultiSound.d/pinnaclecfg.c' '(file already exists)'
+else
+  $echo 'x -' extracting 'MultiSound.d/pinnaclecfg.c' '(text)'
+  sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/pinnaclecfg.c' &&
+/*********************************************************************
+X *
+X * pinnaclecfg.c - Pinnacle/Fiji Device Configuration Program
+X *
+X * This is for NON-PnP mode only.  For PnP mode, use isapnptools.
+X *
+X * This is Linux-specific, and must be run with root permissions.
+X *
+X * Part of the Turtle Beach MultiSound Sound Card Driver for Linux
+X *
+X * Copyright (C) 1998 Andrew Veliath
+X *
+X * This program is free software; you can redistribute it and/or modify
+X * it under the terms of the GNU General Public License as published by
+X * the Free Software Foundation; either version 2 of the License, or
+X * (at your option) any later version.
+X *
+X * This program is distributed in the hope that it will be useful,
+X * but WITHOUT ANY WARRANTY; without even the implied warranty of
+X * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+X * GNU General Public License for more details.
+X *
+X * You should have received a copy of the GNU General Public License
+X * along with this program; if not, write to the Free Software
+X * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+X *
+X ********************************************************************/
+X
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <errno.h>
+#include <unistd.h>
+#include <asm/io.h>
+#include <asm/types.h>
+X
+#define IREG_LOGDEVICE          0x07
+#define IREG_ACTIVATE           0x30
+#define LD_ACTIVATE             0x01
+#define LD_DISACTIVATE          0x00
+#define IREG_EECONTROL          0x3F
+#define IREG_MEMBASEHI          0x40
+#define IREG_MEMBASELO          0x41
+#define IREG_MEMCONTROL         0x42
+#define IREG_MEMRANGEHI         0x43
+#define IREG_MEMRANGELO         0x44
+#define MEMTYPE_8BIT            0x00
+#define MEMTYPE_16BIT           0x02
+#define MEMTYPE_RANGE           0x00
+#define MEMTYPE_HIADDR          0x01
+#define IREG_IO0_BASEHI         0x60
+#define IREG_IO0_BASELO         0x61
+#define IREG_IO1_BASEHI         0x62
+#define IREG_IO1_BASELO         0x63
+#define IREG_IRQ_NUMBER         0x70
+#define IREG_IRQ_TYPE           0x71
+#define IRQTYPE_HIGH            0x02
+#define IRQTYPE_LOW             0x00
+#define IRQTYPE_LEVEL           0x01
+#define IRQTYPE_EDGE            0x00
+X
+#define HIBYTE(w)               ((BYTE)(((WORD)(w) >> 8) & 0xFF))
+#define LOBYTE(w)               ((BYTE)(w))
+#define MAKEWORD(low,hi)        ((WORD)(((BYTE)(low))|(((WORD)((BYTE)(hi)))<<8)))
+X
+typedef __u8                    BYTE;
+typedef __u16                   USHORT;
+typedef __u16                   WORD;
+X
+static int config_port = -1;
+X
+static int msnd_write_cfg(int cfg, int reg, int value)
+{
+X       outb(reg, cfg);
+X       outb(value, cfg + 1);
+X       if (value != inb(cfg + 1)) {
+X               fprintf(stderr, "error: msnd_write_cfg: I/O error\n");
+X               return -EIO;
+X       }
+X       return 0;
+}
+X
+static int msnd_read_cfg(int cfg, int reg)
+{
+X       outb(reg, cfg);
+X       return inb(cfg + 1);
+}
+X
+static int msnd_write_cfg_io0(int cfg, int num, WORD io)
+{
+X       if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X               return -EIO;
+X       if (msnd_write_cfg(cfg, IREG_IO0_BASEHI, HIBYTE(io)))
+X               return -EIO;
+X       if (msnd_write_cfg(cfg, IREG_IO0_BASELO, LOBYTE(io)))
+X               return -EIO;
+X       return 0;
+}
+X
+static int msnd_read_cfg_io0(int cfg, int num, WORD *io)
+{
+X       if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X               return -EIO;
+X       
+X       *io = MAKEWORD(msnd_read_cfg(cfg, IREG_IO0_BASELO),
+X                      msnd_read_cfg(cfg, IREG_IO0_BASEHI));
+X
+X       return 0;
+}
+X
+static int msnd_write_cfg_io1(int cfg, int num, WORD io)
+{
+X       if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X               return -EIO;
+X       if (msnd_write_cfg(cfg, IREG_IO1_BASEHI, HIBYTE(io)))
+X               return -EIO;
+X       if (msnd_write_cfg(cfg, IREG_IO1_BASELO, LOBYTE(io)))
+X               return -EIO;
+X       return 0;
+}
+X
+static int msnd_read_cfg_io1(int cfg, int num, WORD *io)
+{
+X       if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X               return -EIO;
+X       
+X       *io = MAKEWORD(msnd_read_cfg(cfg, IREG_IO1_BASELO),
+X                      msnd_read_cfg(cfg, IREG_IO1_BASEHI));
+X
+X       return 0;
+}
+X
+static int msnd_write_cfg_irq(int cfg, int num, WORD irq)
+{
+X       if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X               return -EIO;
+X       if (msnd_write_cfg(cfg, IREG_IRQ_NUMBER, LOBYTE(irq)))
+X               return -EIO;
+X       if (msnd_write_cfg(cfg, IREG_IRQ_TYPE, IRQTYPE_EDGE))
+X               return -EIO;
+X       return 0;
+}
+X
+static int msnd_read_cfg_irq(int cfg, int num, WORD *irq)
+{
+X       if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X               return -EIO;
+X       
+X       *irq = msnd_read_cfg(cfg, IREG_IRQ_NUMBER);
+X
+X       return 0;
+}
+X
+static int msnd_write_cfg_mem(int cfg, int num, int mem)
+{
+X       WORD wmem;
+X
+X       mem >>= 8;
+X       mem &= 0xfff;
+X       wmem = (WORD)mem;
+X       if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X               return -EIO;
+X       if (msnd_write_cfg(cfg, IREG_MEMBASEHI, HIBYTE(wmem)))
+X               return -EIO;
+X       if (msnd_write_cfg(cfg, IREG_MEMBASELO, LOBYTE(wmem)))
+X               return -EIO;
+X       if (wmem && msnd_write_cfg(cfg, IREG_MEMCONTROL, (MEMTYPE_HIADDR | MEMTYPE_16BIT)))
+X               return -EIO;
+X       return 0;
+}
+X
+static int msnd_read_cfg_mem(int cfg, int num, int *mem)
+{
+X       if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X               return -EIO;
+X       
+X       *mem = MAKEWORD(msnd_read_cfg(cfg, IREG_MEMBASELO),
+X                       msnd_read_cfg(cfg, IREG_MEMBASEHI));
+X       *mem <<= 8;
+X
+X       return 0;
+}
+X
+static int msnd_activate_logical(int cfg, int num)
+{
+X       if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X               return -EIO;
+X       if (msnd_write_cfg(cfg, IREG_ACTIVATE, LD_ACTIVATE))
+X               return -EIO;
+X       return 0;
+}
+X
+static int msnd_write_cfg_logical(int cfg, int num, WORD io0, WORD io1, WORD irq, int mem)
+{
+X       if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X               return -EIO;
+X       if (msnd_write_cfg_io0(cfg, num, io0))
+X               return -EIO;
+X       if (msnd_write_cfg_io1(cfg, num, io1))
+X               return -EIO;
+X       if (msnd_write_cfg_irq(cfg, num, irq))
+X               return -EIO;
+X       if (msnd_write_cfg_mem(cfg, num, mem))
+X               return -EIO;
+X       if (msnd_activate_logical(cfg, num))
+X               return -EIO;
+X       return 0;
+}
+X
+static int msnd_read_cfg_logical(int cfg, int num, WORD *io0, WORD *io1, WORD *irq, int *mem)
+{
+X       if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+X               return -EIO;
+X       if (msnd_read_cfg_io0(cfg, num, io0))
+X               return -EIO;
+X       if (msnd_read_cfg_io1(cfg, num, io1))
+X               return -EIO;
+X       if (msnd_read_cfg_irq(cfg, num, irq))
+X               return -EIO;
+X       if (msnd_read_cfg_mem(cfg, num, mem))
+X               return -EIO;
+X       return 0;
+}
+X
+static void usage(void)
+{
+X       fprintf(stderr,
+X               "\n"
+X               "pinnaclecfg 1.0\n"
+X               "\n"
+X               "usage: pinnaclecfg <config port> [device config]\n"
+X               "\n"
+X               "This is for use with the card in NON-PnP mode only.\n"
+X               "\n"
+X               "Available devices (not all available for Fiji):\n"
+X               "\n"
+X               "        Device                       Description\n"
+X               "        -------------------------------------------------------------------\n"
+X               "        reset                        Reset all devices (i.e. disable)\n"
+X               "        show                         Display current device configurations\n"
+X               "\n"
+X               "        dsp <io> <irq> <mem>         Audio device\n"
+X               "        mpu <io> <irq>               Internal Kurzweil synth\n"
+X               "        ide <io0> <io1> <irq>        On-board IDE controller\n"
+X               "        joystick <io>                Joystick port\n"
+X               "\n");
+X       exit(1);
+}
+X
+static int cfg_reset(void)
+{
+X       int i;
+X
+X       for (i = 0; i < 4; ++i)
+X               msnd_write_cfg_logical(config_port, i, 0, 0, 0, 0);
+X       
+X       return 0;
+}
+X
+static int cfg_show(void)
+{
+X       int i;
+X       int count = 0;
+X
+X       for (i = 0; i < 4; ++i) {
+X               WORD io0, io1, irq;
+X               int mem;
+X               msnd_read_cfg_logical(config_port, i, &io0, &io1, &irq, &mem);
+X               switch (i) {
+X               case 0:
+X                       if (io0 || irq || mem) {
+X                               printf("dsp 0x%x %d 0x%x\n", io0, irq, mem);
+X                               ++count;
+X                       }
+X                       break;
+X               case 1:
+X                       if (io0 || irq) {
+X                               printf("mpu 0x%x %d\n", io0, irq);
+X                               ++count;
+X                       }
+X                       break;
+X               case 2:
+X                       if (io0 || io1 || irq) {
+X                               printf("ide 0x%x 0x%x %d\n", io0, io1, irq);
+X                               ++count;
+X                       }
+X                       break;
+X               case 3:
+X                       if (io0) {
+X                               printf("joystick 0x%x\n", io0);
+X                               ++count;
+X                       }
+X                       break;
+X               }
+X       }
+X
+X       if (count == 0)
+X               fprintf(stderr, "no devices configured\n");
+X       
+X       return 0;
+}
+X
+static int cfg_dsp(int argc, char *argv[])
+{
+X       int io, irq, mem;
+X
+X       if (argc < 3 ||
+X           sscanf(argv[0], "0x%x", &io) != 1 ||
+X           sscanf(argv[1], "%d", &irq) != 1 ||
+X           sscanf(argv[2], "0x%x", &mem) != 1)
+X               usage();
+X
+X       if (!(io == 0x290 ||
+X             io == 0x260 ||
+X             io == 0x250 ||
+X             io == 0x240 ||
+X             io == 0x230 ||
+X             io == 0x220 ||
+X             io == 0x210 ||
+X             io == 0x3e0)) {
+X               fprintf(stderr, "error: io must be one of "
+X                       "210, 220, 230, 240, 250, 260, 290, or 3E0\n");
+X               usage();
+X       }
+X       
+X       if (!(irq == 5 ||
+X             irq == 7 ||
+X             irq == 9 ||
+X             irq == 10 ||
+X             irq == 11 ||
+X             irq == 12)) {
+X               fprintf(stderr, "error: irq must be one of "
+X                       "5, 7, 9, 10, 11 or 12\n");
+X               usage();
+X       }
+X
+X       if (!(mem == 0xb0000 ||
+X             mem == 0xc8000 ||
+X             mem == 0xd0000 ||
+X             mem == 0xd8000 ||
+X             mem == 0xe0000 ||
+X             mem == 0xe8000)) {
+X               fprintf(stderr, "error: mem must be one of "
+X                       "0xb0000, 0xc8000, 0xd0000, 0xd8000, 0xe0000 or 0xe8000\n");
+X               usage();
+X       }
+X
+X       return msnd_write_cfg_logical(config_port, 0, io, 0, irq, mem);
+}
+X
+static int cfg_mpu(int argc, char *argv[])
+{
+X       int io, irq;
+X
+X       if (argc < 2 ||
+X           sscanf(argv[0], "0x%x", &io) != 1 ||
+X           sscanf(argv[1], "%d", &irq) != 1)
+X               usage();
+X       
+X       return msnd_write_cfg_logical(config_port, 1, io, 0, irq, 0);
+}
+X
+static int cfg_ide(int argc, char *argv[])
+{
+X       int io0, io1, irq;
+X
+X       if (argc < 3 ||
+X           sscanf(argv[0], "0x%x", &io0) != 1 ||
+X           sscanf(argv[0], "0x%x", &io1) != 1 ||
+X           sscanf(argv[1], "%d", &irq) != 1)
+X               usage();
+X       
+X       return msnd_write_cfg_logical(config_port, 2, io0, io1, irq, 0);
+}
+X
+static int cfg_joystick(int argc, char *argv[])
+{
+X       int io;
+X
+X       if (argc < 1 ||
+X           sscanf(argv[0], "0x%x", &io) != 1)
+X               usage();
+X       
+X       return msnd_write_cfg_logical(config_port, 3, io, 0, 0, 0);
+}
+X
+int main(int argc, char *argv[])
+{
+X       char *device;
+X       int rv = 0;
+X
+X       --argc; ++argv;
+X
+X       if (argc < 2)
+X               usage();
+X
+X       sscanf(argv[0], "0x%x", &config_port);
+X       if (config_port != 0x250 && config_port != 0x260 && config_port != 0x270) {
+X               fprintf(stderr, "error: <config port> must be 0x250, 0x260 or 0x270\n");
+X               exit(1);
+X       }
+X       if (ioperm(config_port, 2, 1)) {
+X               perror("ioperm");
+X               fprintf(stderr, "note: pinnaclecfg must be run as root\n");
+X               exit(1);
+X       }
+X       device = argv[1];
+X
+X       argc -= 2; argv += 2;
+X
+X       if (strcmp(device, "reset") == 0)
+X               rv = cfg_reset();
+X       else if (strcmp(device, "show") == 0)
+X               rv = cfg_show();
+X       else if (strcmp(device, "dsp") == 0)
+X               rv = cfg_dsp(argc, argv);
+X       else if (strcmp(device, "mpu") == 0)
+X               rv = cfg_mpu(argc, argv);
+X       else if (strcmp(device, "ide") == 0)
+X               rv = cfg_ide(argc, argv);
+X       else if (strcmp(device, "joystick") == 0)
+X               rv = cfg_joystick(argc, argv);
+X       else {
+X               fprintf(stderr, "error: unknown device %s\n", device);
+X               usage();
+X       }
+X
+X       if (rv)
+X               fprintf(stderr, "error: device configuration failed\n");
+X       
+X       return 0;
+}
+SHAR_EOF
+  $shar_touch -am 1204092598 'MultiSound.d/pinnaclecfg.c' &&
+  chmod 0664 'MultiSound.d/pinnaclecfg.c' ||
+  $echo 'restore of' 'MultiSound.d/pinnaclecfg.c' 'failed'
+  if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \
+  && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then
+    md5sum -c << SHAR_EOF >/dev/null 2>&1 \
+    || $echo 'MultiSound.d/pinnaclecfg.c:' 'MD5 check failed'
+366bdf27f0db767a3c7921d0a6db20fe  MultiSound.d/pinnaclecfg.c
+SHAR_EOF
+  else
+    shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/pinnaclecfg.c'`"
+    test 10235 -eq "$shar_count" ||
+    $echo 'MultiSound.d/pinnaclecfg.c:' 'original size' '10235,' 'current size' "$shar_count!"
+  fi
+fi
+# ============= MultiSound.d/Makefile ==============
+if test -f 'MultiSound.d/Makefile' && test "$first_param" != -c; then
+  $echo 'x -' SKIPPING 'MultiSound.d/Makefile' '(file already exists)'
+else
+  $echo 'x -' extracting 'MultiSound.d/Makefile' '(text)'
+  sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/Makefile' &&
+CC      = gcc
+CFLAGS  = -O
+PROGS   = setdigital msndreset pinnaclecfg conv
+X
+all: $(PROGS)
+X
+clean:
+X       rm -f $(PROGS)
+SHAR_EOF
+  $shar_touch -am 1204092398 'MultiSound.d/Makefile' &&
+  chmod 0664 'MultiSound.d/Makefile' ||
+  $echo 'restore of' 'MultiSound.d/Makefile' 'failed'
+  if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \
+  && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then
+    md5sum -c << SHAR_EOF >/dev/null 2>&1 \
+    || $echo 'MultiSound.d/Makefile:' 'MD5 check failed'
+76ca8bb44e3882edcf79c97df6c81845  MultiSound.d/Makefile
+SHAR_EOF
+  else
+    shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/Makefile'`"
+    test 106 -eq "$shar_count" ||
+    $echo 'MultiSound.d/Makefile:' 'original size' '106,' 'current size' "$shar_count!"
+  fi
+fi
+# ============= MultiSound.d/conv.l ==============
+if test -f 'MultiSound.d/conv.l' && test "$first_param" != -c; then
+  $echo 'x -' SKIPPING 'MultiSound.d/conv.l' '(file already exists)'
+else
+  $echo 'x -' extracting 'MultiSound.d/conv.l' '(text)'
+  sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/conv.l' &&
+%%
+[ \n\t,\r]
+\;.*
+DB
+[0-9A-Fa-f]+H   { int n; sscanf(yytext, "%xH", &n); printf("%c", n); }
+%%
+int yywrap() { return 1; }
+main() { yylex(); }
+SHAR_EOF
+  $shar_touch -am 0828231798 'MultiSound.d/conv.l' &&
+  chmod 0664 'MultiSound.d/conv.l' ||
+  $echo 'restore of' 'MultiSound.d/conv.l' 'failed'
+  if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \
+  && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then
+    md5sum -c << SHAR_EOF >/dev/null 2>&1 \
+    || $echo 'MultiSound.d/conv.l:' 'MD5 check failed'
+d2411fc32cd71a00dcdc1f009e858dd2  MultiSound.d/conv.l
+SHAR_EOF
+  else
+    shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/conv.l'`"
+    test 141 -eq "$shar_count" ||
+    $echo 'MultiSound.d/conv.l:' 'original size' '141,' 'current size' "$shar_count!"
+  fi
+fi
+# ============= MultiSound.d/msndreset.c ==============
+if test -f 'MultiSound.d/msndreset.c' && test "$first_param" != -c; then
+  $echo 'x -' SKIPPING 'MultiSound.d/msndreset.c' '(file already exists)'
+else
+  $echo 'x -' extracting 'MultiSound.d/msndreset.c' '(text)'
+  sed 's/^X//' << 'SHAR_EOF' > 'MultiSound.d/msndreset.c' &&
+/*********************************************************************
+X *
+X * msndreset.c - resets the MultiSound card
+X *
+X * Copyright (C) 1998 Andrew Veliath
+X *
+X * This program is free software; you can redistribute it and/or modify
+X * it under the terms of the GNU General Public License as published by
+X * the Free Software Foundation; either version 2 of the License, or
+X * (at your option) any later version.
+X *
+X * This program is distributed in the hope that it will be useful,
+X * but WITHOUT ANY WARRANTY; without even the implied warranty of
+X * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+X * GNU General Public License for more details.
+X *
+X * You should have received a copy of the GNU General Public License
+X * along with this program; if not, write to the Free Software
+X * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+X *
+X ********************************************************************/
+X
+#include <stdio.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <sys/ioctl.h>
+#include <sys/soundcard.h>
+X
+int main(int argc, char *argv[])
+{
+X       int fd;
+X
+X       if (argc != 2) {
+X               fprintf(stderr, "usage: msndreset <mixer device>\n");
+X               exit(1);
+X       }
+X
+X       if ((fd = open(argv[1], O_RDWR)) < 0) {
+X               perror(argv[1]);
+X               exit(1);
+X       }
+X
+X       if (ioctl(fd, SOUND_MIXER_PRIVATE1, 0) < 0) {
+X               fprintf(stderr, "error: msnd ioctl reset failed\n");
+X               perror("ioctl");
+X               close(fd);
+X               exit(1);
+X       }
+X
+X       close(fd);
+X       
+X       return 0;
+}
+SHAR_EOF
+  $shar_touch -am 1204100698 'MultiSound.d/msndreset.c' &&
+  chmod 0664 'MultiSound.d/msndreset.c' ||
+  $echo 'restore of' 'MultiSound.d/msndreset.c' 'failed'
+  if ( md5sum --help 2>&1 | grep 'sage: md5sum \[' ) >/dev/null 2>&1 \
+  && ( md5sum --version 2>&1 | grep -v 'textutils 1.12' ) >/dev/null; then
+    md5sum -c << SHAR_EOF >/dev/null 2>&1 \
+    || $echo 'MultiSound.d/msndreset.c:' 'MD5 check failed'
+c52f876521084e8eb25e12e01dcccb8a  MultiSound.d/msndreset.c
+SHAR_EOF
+  else
+    shar_count="`LC_ALL= LC_CTYPE= LANG= wc -c < 'MultiSound.d/msndreset.c'`"
+    test 1472 -eq "$shar_count" ||
+    $echo 'MultiSound.d/msndreset.c:' 'original size' '1472,' 'current size' "$shar_count!"
+  fi
+fi
+rm -fr _sh01426
+exit 0
diff --git a/Documentation/sound/oss/NEWS b/Documentation/sound/oss/NEWS
new file mode 100644
index 000000000000..a81e0ef72ae9
--- /dev/null
+++ b/Documentation/sound/oss/NEWS
@@ -0,0 +1,42 @@
+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.
+
+
diff --git a/Documentation/sound/oss/NM256 b/Documentation/sound/oss/NM256
new file mode 100644
index 000000000000..b503217488b3
--- /dev/null
+++ b/Documentation/sound/oss/NM256
@@ -0,0 +1,280 @@
+=======================================================
+Documentation for the NeoMagic 256AV/256ZX sound driver
+=======================================================
+
+You're looking at version 1.1 of the driver.  (Woohoo!) It has been
+successfully tested against the following laptop models:
+
+        Sony Z505S/Z505SX/Z505DX/Z505RX
+        Sony F150, F160, F180, F250, F270, F280, PCG-F26
+        Dell Latitude CPi, CPt (various submodels)
+
+There are a few caveats, which is why you should read the entirety of
+this document first.
+
+This driver was developed without any support or assistance from
+NeoMagic.  There is no warranty, expressed, implied, or otherwise.  It
+is free software in the public domain; feel free to use it, sell it,
+give it to your best friends, even claim that you wrote it (but why?!)
+but don't go whining to me, NeoMagic, Sony, Dell, or anyone else
+when it blows up your computer.
+
+Version 1.1 contains a change to try and detect non-AC97 versions of
+the hardware, and not install itself appropriately.  It should also
+reinitialize the hardware on an APM resume event, assuming that APM
+was configured into your kernel.
+
+============
+Installation
+============
+
+Enable the sound drivers, the OSS sound drivers, and then the NM256
+driver.  The NM256 driver *must* be configured as a module (it won't
+give you any other choice).
+
+Next, do the usual "make modules" and "make modules_install".
+Finally, insmod the soundcore, sound and nm256 modules.
+
+When the nm256 driver module is loaded, you should see a couple of
+confirmation messages in the kernel logfile indicating that it found
+the device (the device does *not* use any I/O ports or DMA channels).
+Now try playing a wav file, futz with the CD-ROM if you have one, etc.
+
+The NM256 is entirely a PCI-based device, and all the necessary
+information is automatically obtained from the card.  It can only be
+configured as a module in a vain attempt to prevent people from
+hurting themselves.  It works correctly if it shares an IRQ with
+another device (it normally shares IRQ 9 with the builtin eepro100
+ethernet on the Sony Z505 laptops). 
+
+It does not run the card in any sort of compatibility mode. It will
+not work on laptops that have the SB16-compatible, AD1848-compatible
+or CS4232-compatible codec/mixer; you will want to use the appropriate
+compatible OSS driver with these chipsets.  I cannot provide any
+assistance with machines using the SB16, AD1848 or CS4232 compatible
+versions.  (The driver now attempts to detect the mixer version, and
+will refuse to load if it believes the hardware is not
+AC97-compatible.)
+
+The sound support is very basic, but it does include simultaneous
+playback and record capability.  The mixer support is also quite
+simple, although this is in keeping with the rather limited
+functionality of the chipset.
+
+There is no hardware synthesizer available, as the Losedows OPL-3 and
+MIDI support is done via hardware emulation.
+
+Only three recording devices are available on the Sony: the
+microphone, the CD-ROM input, and the volume device (which corresponds
+to the stereo output).  (Other devices may be available on other
+models of laptops.)  The Z505 series does not have a builtin CD-ROM,
+so of course the CD-ROM input doesn't work.  It does work on laptops
+with a builtin CD-ROM drive.
+
+The mixer device does not appear to have any tone controls, at least
+on the Z505 series.  The mixer module checks for tone controls in the
+AC97 mixer, and will enable them if they are available.
+
+==============
+Known problems
+==============
+
+  * There are known problems with PCMCIA cards and the eepro100 ethernet 
+    driver on the Z505S/Z505SX/Z505DX.  Keep reading.
+
+  * There are also potential problems with using a virtual X display, and
+    also problems loading the module after the X server has been started. 
+    Keep reading.
+
+  * The volume control isn't anywhere near linear.  Sorry.  This will be
+    fixed eventually, when I get sufficiently annoyed with it.  (I doubt
+    it will ever be fixed now, since I've never gotten sufficiently
+    annoyed with it and nobody else seems to care.)
+
+  * There are reports that the CD-ROM volume is very low.  Since I do not
+    have a CD-ROM equipped laptop, I cannot test this (it's kinda hard to
+    do remotely).
+
+  * Only 8 fixed-rate speeds are supported.  This is mainly a chipset
+    limitation.  It may be possible to support other speeds in the future.
+
+  * There is no support for the telephone mixer/codec.  There is support
+    for a phonein/phoneout device in the mixer driver;  whether or not 
+    it does anything is anyone's guess.  (Reports on this would be
+    appreciated.  You'll have to figure out how to get the phone to
+    go off-hook before it'll work, tho.)
+
+  * This driver was not written with any cooperation or support from
+    NeoMagic.  If you have any questions about this, see their website
+    for their official stance on supporting open source drivers.
+
+============
+Video memory
+============
+
+The NeoMagic sound engine uses a portion of the display memory to hold
+the sound buffer.  (Crazy, eh?)  The NeoMagic video BIOS sets up a
+special pointer at the top of video RAM to indicate where the top of
+the audio buffer should be placed.
+
+At the present time XFree86 is apparently not aware of this.  It will
+thus write over either the pointer or the sound buffer with abandon.
+(Accelerated-X seems to do a better job here.)
+
+This implies a few things:
+
+  * Sometimes the NM256 driver has to guess at where the buffer 
+    should be placed, especially if the module is loaded after the
+    X server is started.  It's usually correct, but it will consistently
+    fail on the Sony F250.
+
+  * Virtual screens greater than 1024x768x16 under XFree86 are
+    problematic on laptops with only 2.5MB of screen RAM. This
+    includes all of the 256AV-equipped laptops.  (Virtual displays
+    may or may not work on the 256ZX, which has at least 4MB of
+    video RAM.)
+
+If you start having problems with random noise being output either
+constantly (this is the usual symptom on the F250), or when windows
+are moved around (this is the usual symptom when using a virtual
+screen), the best fix is to
+
+  * Don't use a virtual frame buffer.
+  * Make sure you load the NM256 module before the X server is
+    started.
+
+On the F250, it is possible to force the driver to load properly even
+after the XFree86 server is started by doing:
+
+        insmod nm256 buffertop=0x25a800
+
+This forces the audio buffers to the correct offset in screen RAM.
+
+One user has reported a similar problem on the Sony F270, although
+others apparently aren't seeing any problems.  His suggested command
+is
+
+        insmod nm256 buffertop=0x272800
+
+=================
+Official WWW site
+=================
+
+The official site for the NM256 driver is:
+
+        http://www.uglx.org/sony.html
+
+You should always be able to get the latest version of the driver there,
+and the driver will be supported for the foreseeable future.
+
+==============
+Z505RX and IDE
+==============
+
+There appears to be a problem with the IDE chipset on the Z505RX; one
+of the symptoms is that sound playback periodically hangs (when the
+disk is accessed).  The user reporting the problem also reported that
+enabling all of the IDE chipset workarounds in the kernel solved the
+problem, tho obviously only one of them should be needed--if someone
+can give me more details I would appreciate it.
+
+==============================
+Z505S/Z505SX on-board Ethernet
+==============================
+
+If you're using the on-board Ethernet Pro/100 ethernet support on the Z505
+series, I strongly encourage you to download the latest eepro100 driver from
+Donald Becker's site:
+
+        ftp://cesdis.gsfc.nasa.gov/pub/linux/drivers/test/eepro100.c
+
+There was a reported problem on the Z505SX that if the ethernet
+interface is disabled and reenabled while the sound driver is loaded,
+the machine would lock up.  I have included a workaround that is
+working satisfactorily.  However, you may occasionally see a message
+about "Releasing interrupts, over 1000 bad interrupts" which indicates
+that the workaround is doing its job.
+
+==================================
+PCMCIA and the Z505S/Z505SX/Z505DX
+==================================
+
+There is also a known problem with the Sony Z505S and Z505SX hanging
+if a PCMCIA card is inserted while the ethernet driver is loaded, or
+in some cases if the laptop is suspended.  This is caused by tons of
+spurious IRQ 9s, probably generated from the PCMCIA or ACPI bridges.
+
+There is currently no fix for the problem that works in every case.
+The only known workarounds are to disable the ethernet interface
+before inserting or removing a PCMCIA card, or with some cards
+disabling the PCMCIA card before ejecting it will also help the
+problem with the laptop hanging when the card is ejected.
+
+One user has reported that setting the tcic's cs_irq to some value
+other than 9 (like 11) fixed the problem.  This doesn't work on my
+Z505S, however--changing the value causes the cardmgr to stop seeing
+card insertions and removals, cards don't seem to work correctly, and
+I still get hangs if a card is inserted when the kernel is booted.
+
+Using the latest ethernet driver and pcmcia package allows me to
+insert an Adaptec 1480A SlimScsi card without the laptop hanging,
+although I still have to shut down the card before ejecting or
+powering down the laptop.  However, similar experiments with a DE-660
+ethernet card still result in hangs when the card is inserted.  I am
+beginning to think that the interrupts are CardBus-related, since the
+Adaptec card is a CardBus card, and the DE-660 is not; however, I
+don't have any other CardBus cards to test with.
+
+======
+Thanks
+======
+
+First, I want to thank everyone (except NeoMagic of course) for their
+generous support and encouragement.  I'd like to list everyone's name
+here that replied during the development phase, but the list is
+amazingly long.
+
+I will be rather unfair and single out a few people, however:
+
+        Justin Maurer, for being the first random net.person to try it,
+        and for letting me login to his Z505SX to get it working there
+
+        Edi Weitz for trying out several different versions, and giving
+        me a lot of useful feedback
+
+        Greg Rumple for letting me login remotely to get the driver 
+        functional on the 256ZX, for his assistance on tracking
+        down all sorts of random stuff, and for trying out Accel-X
+
+        Zach Brown, for the initial AC97 mixer interface design
+
+        Jeff Garzik, for various helpful suggestions on the AC97
+        interface
+
+        "Mr. Bumpy" for feedback on the Z505RX
+
+        Bill Nottingham, for generous assistance in getting the mixer ID
+        code working
+
+=================
+Previous versions
+=================
+
+Versions prior to 0.3 (aka `noname') had problems with weird artifacts
+in the output and failed to set the recording rate properly.  These
+problems have long since been fixed.
+
+Versions prior to 0.5 had problems with clicks in the output when
+anything other than 16-bit stereo sound was being played, and also had
+periodic clicks when recording.
+
+Version 0.7 first incorporated support for the NM256ZX chipset, which
+is found on some Dell Latitude laptops (the CPt, and apparently
+some CPi models as well).  It also included the generic AC97
+mixer module.
+
+Version 0.75 renamed all the functions and files with slightly more
+generic names.
+
+Note that previous versions of this document claimed that recording was
+8-bit only; it actually has been working for 16-bits all along.
diff --git a/Documentation/sound/oss/OPL3 b/Documentation/sound/oss/OPL3
new file mode 100644
index 000000000000..2468ff827688
--- /dev/null
+++ b/Documentation/sound/oss/OPL3
@@ -0,0 +1,6 @@
+A pure OPL3 card is nice and easy to configure. Simply do
+
+insmod opl3 io=0x388
+
+Change the I/O address in the very unlikely case this card is differently
+configured
diff --git a/Documentation/sound/oss/OPL3-SA b/Documentation/sound/oss/OPL3-SA
new file mode 100644
index 000000000000..66a91835d918
--- /dev/null
+++ b/Documentation/sound/oss/OPL3-SA
@@ -0,0 +1,52 @@
+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>
diff --git a/Documentation/sound/oss/OPL3-SA2 b/Documentation/sound/oss/OPL3-SA2
new file mode 100644
index 000000000000..d8b6d2bbada6
--- /dev/null
+++ b/Documentation/sound/oss/OPL3-SA2
@@ -0,0 +1,210 @@
+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
diff --git a/Documentation/sound/oss/Opti b/Documentation/sound/oss/Opti
new file mode 100644
index 000000000000..c15af3c07d46
--- /dev/null
+++ b/Documentation/sound/oss/Opti
@@ -0,0 +1,222 @@
+Support for the OPTi 82C931 chip
+--------------------------------
+Note: parts of this README file apply also to other
+cards that use the mad16 driver.
+
+Some items in this README file are based on features
+added to the sound driver after Linux-2.1.91 was out.
+By the time of writing this I do not know which official
+kernel release will include these features.
+Please do not report inconsistencies on older Linux
+kernels.
+
+The OPTi 82C931 is supported in its non-PnP mode.
+Usually you do not need to set jumpers, etc. The sound driver
+will check the card status and if it is required it will
+force the card into a mode in which it can be programmed.
+
+If you have another OS installed on your computer it is recommended
+that Linux and the other OS use the same resources.
+
+Also, it is recommended that resources specified in /etc/modprobe.conf
+and resources specified in /etc/isapnp.conf agree.
+
+Compiling the sound driver
+--------------------------
+I highly recommend that you build a modularized sound driver.
+This document does not cover a sound-driver which is built in
+the kernel.
+
+Sound card support should be enabled as a module (chose m).
+Answer 'm' for  these items:
+  Generic OPL2/OPL3 FM synthesizer support              (CONFIG_SOUND_ADLIB)
+  Microsoft Sound System support                        (CONFIG_SOUND_MSS)
+  Support for OPTi MAD16 and/or Mozart based cards      (CONFIG_SOUND_MAD16)
+  FM synthesizer (YM3812/OPL-3) support                 (CONFIG_SOUND_YM3812)
+
+The configuration menu may ask for addresses, IRQ lines or DMA
+channels. If the card is used as a module the module loading
+options will override these values.
+
+For the OPTi 931 you can answer 'n' to:
+  Support MIDI in older MAD16 based cards (requires SB) (CONFIG_SOUND_MAD16_OLDCARD)
+If you do need MIDI support in a Mozart or C928 based card you
+need to answer 'm' to the above question. In that case you will
+also need to answer 'm' to:
+  '100% Sound Blaster compatibles (SB16/32/64, ESS, Jazz16) support' (CONFIG_SOUND_SB)
+
+Go on and compile your kernel and modules. Install the modules. Run depmod -a.
+
+Using isapnptools
+-----------------
+In most systems with a PnP BIOS you do not need to use isapnp. The
+initialization provided by the BIOS is sufficient for the driver
+to pick up the card and continue initialization.
+
+If that fails, or if you have other PnP cards, you need to use isapnp
+to initialize the card.
+This was tested with isapnptools-1.11 but I recommend that you use
+isapnptools-1.13 (or newer). Run pnpdump to dump the information
+about your PnP cards. Then edit the resulting file and select
+the options of your choice. This file is normally installed as
+/etc/isapnp.conf.
+
+The driver has one limitation with respect to I/O port resources:
+IO3 base must be 0x0E0C.  Although isapnp allows other ports, this
+address is hard-coded into the driver.
+
+Using kmod and autoloading the sound driver
+-------------------------------------------
+Comment: as of linux-2.1.90 kmod is replacing kerneld.
+The config file '/etc/modprobe.conf' is used as before.
+
+This is the sound part of my /etc/modprobe.conf file.
+Following that I will explain each line.
+
+alias mixer0 mad16
+alias audio0 mad16
+alias midi0  mad16
+alias synth0 opl3
+options sb mad16=1
+options mad16 irq=10 dma=0 dma16=1 io=0x530 joystick=1 cdtype=0
+options opl3 io=0x388
+install mad16 /sbin/modprobe -i mad16 && /sbin/ad1848_mixer_reroute 14 8 15 3 16 6
+
+If you have an MPU daughtercard or onboard MPU you will want to add to the
+"options mad16" line - eg 
+
+options mad16 irq=5 dma=0 dma16=3 io=0x530 mpu_io=0x330 mpu_irq=9
+
+To set the I/O and IRQ of the MPU.
+
+
+Explain:
+
+alias mixer0 mad16
+alias audio0 mad16
+alias midi0  mad16
+alias synth0 opl3
+
+When any sound device is opened the kernel requests auto-loading
+of char-major-14. There is a built-in alias that translates this
+request to loading the main sound module.
+
+The sound module in its turn will request loading of a sub-driver
+for mixer, audio, midi or synthesizer device. The first 3 are
+supported by the mad16 driver. The synth device is supported
+by the opl3 driver.
+
+There is currently no way to autoload the sound device driver
+if more than one card is installed.
+
+options sb mad16=1
+
+This is left for historical reasons. If you enable the
+config option 'Support MIDI in older MAD16 based cards (requires SB)'
+or if you use an older mad16 driver it will force loading of the
+SoundBlaster driver. This option tells the SB driver not to look
+for a SB card but to wait for the mad16 driver.
+
+options mad16 irq=10 dma=0 dma16=1 io=0x530 joystick=1 cdtype=0
+options opl3 io=0x388
+
+post-install mad16 /sbin/ad1848_mixer_reroute 14 8 15 3 16 6
+
+This sets resources and options for the mad16 and opl3 drivers.
+I use two DMA channels (only one is required) to enable full duplex.
+joystick=1 enables the joystick port. cdtype=0 disables the cd port.
+You can also set mpu_io and mpu_irq in the mad16 options for the
+uart401 driver.
+
+This tells modprobe to run /sbin/ad1848_mixer_reroute after
+mad16 is successfully loaded and initialized. The source
+for ad1848_mixer_reroute is appended to the end of this readme
+file. It is impossible for the sound driver to know the actual
+connections to the mixer. The 3 inputs intended for cd, synth
+and line-in are mapped to the generic inputs line1, line2 and
+line3. This program reroutes these mixer channels to their
+right names (note the right mapping depends on the actual sound
+card that you use).
+The numeric parameters mean:
+ 14=line1 8=cd    - reroute line1 to the CD input.
+ 15=line2 3=synth - reroute line2 to the synthesizer input.
+ 16=line3 6=line  - reroute line3 to the line input.
+For reference on other input names look at the file
+/usr/include/linux/soundcard.h.
+
+Using a joystick
+-----------------
+You must enable a joystick in the mad16 options. (also
+in /etc/isapnp.conf if you use it).
+Tested with regular analog joysticks.
+
+A CDROM drive connected to the sound card
+-----------------------------------------
+The 82C931 chip has support only for secondary ATAPI cdrom.
+(cdtype=8). Loading the mad16 driver resets the C931 chip
+and if a cdrom was already mounted it may cause a complete
+system hang. Do not use the sound card if you have an alternative.
+If you do use the sound card it is important that you load
+the mad16 driver (use "modprobe mad16" to prevent auto-unloading)
+before the cdrom is accessed the first time.
+
+Using the sound driver built-in to the kernel may help here, but...
+Most new systems have a PnP BIOS and also two IDE controllers.
+The IDE controller on the sound card may be needed only on older
+systems (which have only one IDE controller) but these systems
+also do not have a PnP BIOS - requiring isapnptools and a modularized
+driver.
+
+Known problems
+--------------
+1. See the section on "A CDROM drive connected to the sound card".
+
+2. On my system the codec cannot capture companded sound samples.
+   (eg., recording from /dev/audio). When any companded capture is
+   requested I get stereo-16 bit samples instead. Playback of
+   companded samples works well. Apparently this problem is not common
+   to all C931 based cards. I do not know how to identify cards that
+   have this problem.
+
+Source for ad1848_mixer_reroute.c
+---------------------------------
+#include <stdio.h>
+#include <fcntl.h>
+#include <linux/soundcard.h>
+
+static char *mixer_names[SOUND_MIXER_NRDEVICES] =
+        SOUND_DEVICE_LABELS;
+
+int
+main(int argc, char **argv) {
+        int val, from, to;
+        int i, fd;
+
+        fd = open("/dev/mixer", O_RDWR);
+        if(fd < 0) {
+                perror("/dev/mixer");
+                return 1;
+        }
+
+        for(i = 2; i < argc; i += 2) {
+                from = atoi(argv[i-1]);
+                to = atoi(argv[i]);
+
+                if(to == SOUND_MIXER_NONE)
+                        fprintf(stderr, "%s: turning off mixer %s\n",
+                                argv[0], mixer_names[to]);
+                else
+                        fprintf(stderr, "%s: rerouting mixer %s to %s\n",
+                                argv[0], mixer_names[from], mixer_names[to]);
+
+                val = from << 8 | to;
+
+                if(ioctl(fd, SOUND_MIXER_PRIVATE2, &val)) {
+                        perror("AD1848 mixer reroute");
+                        return 1;
+                }
+        }
+
+        return 0;
+}
+
diff --git a/Documentation/sound/oss/PAS16 b/Documentation/sound/oss/PAS16
new file mode 100644
index 000000000000..951b3dce51b4
--- /dev/null
+++ b/Documentation/sound/oss/PAS16
@@ -0,0 +1,163 @@
+Pro Audio Spectrum 16 for 2.3.99 and later
+=========================================
+by Thomas Molina (tmolina@home.com)
+last modified 3 Mar 2001
+Acknowledgement to Axel Boldt (boldt@math.ucsb.edu) for stuff taken
+from Configure.help, Riccardo Facchetti for stuff from README.OSS,
+and others whose names I could not find.
+
+This documentation is relevant for the PAS16 driver (pas2_card.c and
+friends) under kernel version 2.3.99 and later.  If you are
+unfamiliar with configuring sound under Linux, please read the
+Sound-HOWTO, Documentation/sound/oss/Introduction and other
+relevant docs first.
+
+The following information is relevant information from README.OSS
+and legacy docs for the Pro Audio Spectrum 16 (PAS16):
+==================================================================
+
+The pas2_card.c driver supports the following cards --
+Pro Audio Spectrum 16 (PAS16) and compatibles:
+        Pro Audio Spectrum 16
+        Pro Audio Studio 16
+        Logitech Sound Man 16
+        NOTE! The original Pro Audio Spectrum as well as the PAS+ are not
+              and will not be supported by the driver.
+
+The sound driver configuration dialog
+-------------------------------------
+
+Sound configuration starts by making some yes/no questions. Be careful
+when answering to these questions since answering y to a question may
+prevent some later ones from being asked. For example don't answer y to
+the question about (PAS16) if you don't really have a PAS16.  Sound
+configuration may also be made modular by answering m to configuration
+options presented. 
+
+Note also that all questions may not be asked. The configuration program
+may disable some questions depending on the earlier choices. It may also
+select some options automatically as well.
+
+  "ProAudioSpectrum 16 support",
+        - Answer 'y'_ONLY_ if you have a Pro Audio Spectrum _16_,
+          Pro Audio Studio 16 or Logitech SoundMan 16 (be sure that
+          you read the above list correctly). Don't answer 'y' if you
+          have some other card made by Media Vision or Logitech since they
+          are not PAS16 compatible.
+          NOTE! Since 3.5-beta10 you need to enable SB support (next question)
+          if you want to use the SB emulation of PAS16. It's also possible to
+          the emulation if you want to use a true SB card together with PAS16
+          (there is another question about this that is asked later).
+
+  "Generic OPL2/OPL3 FM synthesizer support",
+        - Answer 'y' if your card has a FM chip made by Yamaha (OPL2/OPL3/OPL4).
+          The PAS16 has an OPL3-compatible FM chip.
+
+With PAS16 you can use two audio device files at the same time. /dev/dsp (and
+/dev/audio) is connected to the 8/16 bit native codec and the /dev/dsp1 (and
+/dev/audio1) is connected to the SB emulation (8 bit mono only).
+
+
+The new stuff for 2.3.99 and later
+============================================================================
+The following configuration options from Documentation/Configure.help
+are relevant to configuring the PAS16:
+
+Sound card support
+CONFIG_SOUND
+  If you have a sound card in your computer, i.e. if it can say more
+  than an occasional beep, say Y. Be sure to have all the information
+  about your sound card and its configuration down (I/O port,
+  interrupt and DMA channel), because you will be asked for it.
+
+  You want to read the Sound-HOWTO, available from
+  http://www.tldp.org/docs.html#howto . General information
+  about the modular sound system is contained in the files
+  Documentation/sound/oss/Introduction. The file
+  Documentation/sound/oss/README.OSS contains some slightly outdated but
+  still useful information as well.
+
+OSS sound modules
+CONFIG_SOUND_OSS
+  OSS is the Open Sound System suite of sound card drivers. They make
+  sound programming easier since they provide a common API. Say Y or M
+  here (the module will be called sound.o) if you haven't found a
+  driver for your sound card above, then pick your driver from the
+  list below.
+
+Persistent DMA buffers
+CONFIG_SOUND_DMAP
+  Linux can often have problems allocating DMA buffers for ISA sound
+  cards on machines with more than 16MB of RAM. This is because ISA
+  DMA buffers must exist below the 16MB boundary and it is quite
+  possible that a large enough free block in this region cannot be
+  found after the machine has been running for a while. If you say Y
+  here the DMA buffers (64Kb) will be allocated at boot time and kept
+  until the shutdown. This option is only useful if you said Y to
+  "OSS sound modules", above. If you said M to "OSS sound modules"
+  then you can get the persistent DMA buffer functionality by passing
+  the command-line argument "dmabuf=1" to the sound.o module.
+
+  Say y here for PAS16.
+
+ProAudioSpectrum 16 support
+CONFIG_SOUND_PAS
+  Answer Y only if you have a Pro Audio Spectrum 16, ProAudio Studio
+  16 or Logitech SoundMan 16 sound card. Don't answer Y if you have
+  some other card made by Media Vision or Logitech since they are not
+  PAS16 compatible.  It is not necessary to enable the  separate
+  Sound Blaster support; it is included in the PAS driver.
+
+  If you compile the driver into the kernel, you have to add
+  "pas2=<io>,<irq>,<dma>,<dma2>,<sbio>,<sbirq>,<sbdma>,<sbdma2>
+  to the kernel command line.
+
+FM Synthesizer (YM3812/OPL-3) support
+CONFIG_SOUND_YM3812
+  Answer Y if your card has a FM chip made by Yamaha (OPL2/OPL3/OPL4).
+  Answering Y is usually a safe and recommended choice, however some
+  cards may have software (TSR) FM emulation. Enabling FM support with
+  these cards may cause trouble (I don't currently know of any such
+  cards, however).
+  Please read the file Documentation/sound/oss/OPL3 if your card has an
+  OPL3 chip.
+  If you compile the driver into the kernel, you have to add
+  "opl3=<io>" to the kernel command line.
+
+  If you compile your drivers into the kernel, you MUST configure 
+  OPL3 support as a module for PAS16 support to work properly.
+  You can then get OPL3 functionality by issuing the command:
+  insmod opl3
+  In addition, you must either add the following line to 
+  /etc/modprobe.conf:
+  options opl3 io=0x388
+  or else add the following line to /etc/lilo.conf:
+  opl3=0x388
+
+
+EXAMPLES
+===================================================================
+To use the PAS16 in my computer I have enabled the following sound
+configuration options:
+
+CONFIG_SOUND=y
+CONFIG_SOUND_OSS=y
+CONFIG_SOUND_TRACEINIT=y
+CONFIG_SOUND_DMAP=y
+CONFIG_SOUND_PAS=y
+CONFIG_SOUND_SB=n
+CONFIG_SOUND_YM3812=m
+
+I have also included the following append line in /etc/lilo.conf:
+append="pas2=0x388,10,3,-1,0x220,5,1,-1 sb=0x220,5,1,-1 opl3=0x388"
+
+The io address of 0x388 is default configuration on the PAS16.  The
+irq of 10 and dma of 3 may not match your installation.  The above 
+configuration enables PAS16, 8-bit Soundblaster and OPL3
+functionality.  If Soundblaster functionality is not desired, the
+following line would be appropriate:
+append="pas2=0x388,10,3,-1,0,-1,-1,-1 opl3=0x388"
+
+If sound is built totally modular, the above options may be 
+specified in /etc/modprobe.conf for pas2, sb and opl3
+respectively. 
diff --git a/Documentation/sound/oss/PSS b/Documentation/sound/oss/PSS
new file mode 100644
index 000000000000..187b9525e1f6
--- /dev/null
+++ b/Documentation/sound/oss/PSS
@@ -0,0 +1,41 @@
+The PSS cards and other ECHO based cards provide an onboard DSP with 
+downloadable programs and also has an AD1848 "Microsoft Sound System" 
+device. The PSS driver enables MSS and MPU401 modes of the card. SB 
+is not enabled since it doesn't work concurrently with MSS. 
+
+If you build this driver as a module then the driver takes the following
+parameters
+
+pss_io.         The I/O base the PSS card is configured at (normally 0x220
+                or 0x240)
+
+mss_io          The base address of the Microsoft Sound System interface.
+                This is normally 0x530, but may be 0x604 or other addresses.
+
+mss_irq         The interrupt assigned to the Microsoft Sound System
+                emulation. IRQ's 3,5,7,9,10,11 and 12 are available. If you
+                get IRQ errors be sure to check the interrupt is set to 
+                "ISA/Legacy" in the BIOS on modern machines.
+
+mss_dma         The DMA channel used by the Microsoft Sound System.
+                This can be 0, 1, or 3. DMA 0 is not available on older 
+                machines and will cause a crash on them.
+
+mpu_io          The MPU emulation base address. This sets the base of the
+                synthesizer. It is typically 0x330 but can be altered.
+
+mpu_irq         The interrupt to use for the synthesizer. It must differ
+                from the IRQ used by the Microsoft Sound System port.
+
+
+The mpu_io/mpu_irq fields are optional. If they are not specified the 
+synthesizer parts are not configured.
+
+When the module is loaded it looks for a file called 
+/etc/sound/pss_synth. This is the firmware file from the DOS install disks.
+This fil holds a general MIDI emulation. The file expected is called
+genmidi.ld on newer DOS driver install disks and synth.ld on older ones.
+
+You can also load alternative DSP algorithms into the card if you wish. One
+alternative driver can be found at http://www.mpg123.de/
+
diff --git a/Documentation/sound/oss/PSS-updates b/Documentation/sound/oss/PSS-updates
new file mode 100644
index 000000000000..c84dd7597e64
--- /dev/null
+++ b/Documentation/sound/oss/PSS-updates
@@ -0,0 +1,88 @@
+        This file contains notes for users of PSS sound cards who wish to use the 
+newly added features of the newest version of this driver.
+
+        The major enhancements present in this new revision of this driver is the 
+addition of two new module parameters that allow you to take full advantage of 
+all the features present on your PSS sound card.  These features include the 
+ability to enable both the builtin CDROM and joystick ports.
+
+pss_enable_joystick
+
+        This parameter is basically a flag.  A 0 will leave the joystick port 
+disabled, while a non-zero value would enable the joystick port.  The default 
+setting is pss_enable_joystick=0 as this keeps this driver fully compatible 
+with systems that were using previous versions of this driver.  If you wish to 
+enable the joystick port you will have to add pss_enable_joystick=1 as an 
+argument to the driver.  To actually use the joystick port you will then have 
+to load the joystick driver itself.  Just remember to load the joystick driver 
+AFTER the pss sound driver.
+
+pss_cdrom_port
+
+        This parameter takes a port address as its parameter.  Any available port 
+address can be specified to enable the CDROM port, except for 0x0 and -1 as 
+these values would leave the port disabled.  Like the joystick port, the cdrom 
+port will require that an appropriate CDROM driver be loaded before you can make 
+use of the newly enabled CDROM port.  Like the joystick port option above, 
+remember to load the CDROM driver AFTER the pss sound driver.  While it may 
+differ on some PSS sound cards, all the PSS sound cards that I have seen have a 
+builtin Wearnes CDROM port.  If this is the case with your PSS sound card you 
+should load aztcd with the appropriate port option that matches the port you 
+assigned to the CDROM port when you loaded your pss sound driver.  (ex. 
+modprobe pss pss_cdrom_port=0x340 && modprobe aztcd aztcd=0x340)  The default 
+setting of this parameter leaves the CDROM port disabled to maintain full 
+compatibility with systems using previous versions of this driver.
+
+        Other options have also been added for the added convenience and utility 
+of the user.  These options are only available if this driver is loaded as a 
+module.
+
+pss_no_sound
+
+        This module parameter is a flag that can be used to tell the driver to 
+just configure non-sound components.  0 configures all components, a non-0 
+value will only attept to configure the CDROM and joystick ports.  This 
+parameter can be used by a user who only wished to use the builtin joystick 
+and/or CDROM port(s) of his PSS sound card.  If this driver is loaded with this 
+parameter and with the parameter below set to true then a user can safely unload 
+this driver with the following command "rmmod pss && rmmod ad1848 && rmmod 
+mpu401 && rmmod sound && rmmod soundcore" and retain the full functionality of 
+his CDROM and/or joystick port(s) while gaining back the memory previously used 
+by the sound drivers.  This default setting of this parameter is 0 to retain 
+full behavioral compatibility with previous versions of this driver.
+
+pss_keep_settings
+
+        This parameter can be used to specify whether you want the driver to reset 
+all emulations whenever its unloaded.  This can be useful for those who are 
+sharing resources (io ports, IRQ's, DMA's) between different ISA cards.  This 
+flag can also be useful in that future versions of this driver may reset all 
+emulations by default on the driver's unloading (as it probably should), so 
+specifying it now will ensure that all future versions of this driver will 
+continue to work as expected.  The default value of this parameter is 1 to 
+retain full behavioral compatibility with previous versions of this driver.
+
+pss_firmware
+
+        This parameter can be used to specify the file containing the firmware 
+code so that a user could tell the driver where that file is located instead 
+of having to put it in a predefined location with a predefined name.  The 
+default setting of this parameter is "/etc/sound/pss_synth" as this was the 
+path and filename the hardcoded value in the previous versions of this driver.
+
+Examples:
+
+# Normal PSS sound card system, loading of drivers.
+# Should be specified in an rc file (ex. Slackware uses /etc/rc.d/rc.modules).
+
+/sbin/modprobe pss pss_io=0x220 mpu_io=0x338 mpu_irq=9 mss_io=0x530 mss_irq=10 mss_dma=1 pss_cdrom_port=0x340 pss_enable_joystick=1
+/sbin/modprobe aztcd aztcd=0x340
+/sbin/modprobe joystick
+
+# System using the PSS sound card just for its CDROM and joystick ports.
+# Should be specified in an rc file (ex. Slackware uses /etc/rc.d/rc.modules).
+
+/sbin/modprobe pss pss_io=0x220 pss_cdrom_port=0x340 pss_enable_joystick=1 pss_no_sound=1
+/sbin/rmmod pss && /sbin/rmmod ad1848 && /sbin/rmmod mpu401 && /sbin/rmmod sound && /sbin/rmmod soundcore  # This line not needed, but saves memory.
+/sbin/modprobe aztcd aztcd=0x340
+/sbin/modprobe joystick
diff --git a/Documentation/sound/oss/README.OSS b/Documentation/sound/oss/README.OSS
new file mode 100644
index 000000000000..fd42b05b2f55
--- /dev/null
+++ b/Documentation/sound/oss/README.OSS
@@ -0,0 +1,1456 @@
+Introduction
+------------
+
+This file is a collection of all the old Readme files distributed with
+OSS/Lite by Hannu Savolainen. Since the new Linux sound driver is founded
+on it I think these information may still be interesting for users that
+have to configure their sound system.
+
+Be warned: Alan Cox is the current maintainer of the Linux sound driver so if
+you have problems with it, please contact him or the current device-specific
+driver maintainer (e.g. for aedsp16 specific problems contact me). If you have
+patches, contributions or suggestions send them to Alan: I'm sure they are
+welcome.
+
+In this document you will find a lot of references about OSS/Lite or ossfree:
+they are gone forever. Keeping this in mind and with a grain of salt this
+document can be still interesting and very helpful.
+
+[ File edited 17.01.1999 - Riccardo Facchetti ]
+[ Edited miroSOUND section 19.04.2001 - Robert Siemer ]
+
+OSS/Free version 3.8 release notes
+----------------------------------
+
+Please read the SOUND-HOWTO (available from sunsite.unc.edu and other Linux FTP 
+sites). It gives instructions about using sound with Linux. It's bit out of
+date but still very useful. Information about bug fixes and such things
+is available from the web page (see above).
+
+Please check http://www.opensound.com/pguide for more info about programming
+with OSS API.
+
+   ====================================================
+-  THIS VERSION ____REQUIRES____ Linux 2.1.57 OR LATER.
+   ====================================================
+
+Packages "snd-util-3.8.tar.gz" and "snd-data-0.1.tar.Z"
+contain useful utilities to be used with this driver.
+See http://www.opensound.com/ossfree/getting.html for
+download instructions.
+
+If you are looking for the installation instructions, please
+look forward into this document.
+
+Supported sound cards
+---------------------
+
+See below.
+
+Contributors
+------------
+
+This driver contains code by several contributors. In addition several other
+persons have given useful suggestions. The following is a list of major
+contributors. (I could have forgotten some names.)
+
+        Craig Metz      1/2 of the PAS16 Mixer and PCM support
+        Rob Hooft       Volume computation algorithm for the FM synth.
+        Mika Liljeberg  uLaw encoding and decoding routines
+        Jeff Tranter    Linux SOUND HOWTO document
+        Greg Lee        Volume computation algorithm for the GUS and
+                        lots of valuable suggestions.
+        Andy Warner     ISC port
+        Jim Lowe,
+        Amancio Hasty Jr        FreeBSD/NetBSD port
+        Anders Baekgaard        Bug hunting and valuable suggestions.
+        Joerg Schubert  SB16 DSP support (initial version).
+        Andrew Robinson Improvements to the GUS driver
+        Megens SA       MIDI recording for SB and SB Pro (initial version).
+        Mikael Nordqvist  Linear volume support for GUS and
+                          nonblocking /dev/sequencer.
+        Ian Hartas              SVR4.2 port
+        Markus Aroharju and
+        Risto Kankkunen         Major contributions to the mixer support
+                                of GUS v3.7.
+        Hunyue Yau      Mixer support for SG NX Pro.
+        Marc Hoffman    PSS support (initial version).
+        Rainer Vranken  Initialization for Jazz16 (initial version).
+        Peter Trattler  Initial version of loadable module support for Linux.
+        JRA Gibson      16 bit mode for Jazz16 (initial version)
+        Davor Jadrijevic MAD16 support (initial version)
+        Gregor Hoffleit Mozart support (initial version)
+        Riccardo Facchetti Audio Excel DSP 16 (aedsp16) support
+        James Hightower Spotting a tiny but important bug in CS423x support.
+        Denis Sablic    OPTi 82C924 specific enhancements (non PnP mode)
+        Tim MacKenzie   Full duplex support for OPTi 82C930.
+        
+        Please look at lowlevel/README for more contributors.
+
+There are probably many other names missing. If you have sent me some
+patches and your name is not in the above list, please inform me.
+
+Sending your contributions or patches
+-------------------------------------
+
+First of all it's highly recommended to contact me before sending anything
+or before even starting to do any work. Tell me what you suggest to be
+changed or what you have planned to do. Also ensure you are using the
+very latest (development) version of OSS/Free since the change may already be
+implemented there. In general it's a major waste of time to try to improve a
+several months old version. Information about the latest version can be found
+from http://www.opensound.com/ossfree. In general there is no point in
+sending me patches relative to production kernels.
+
+Sponsors etc.
+-------------
+
+The following companies have greatly helped development of this driver 
+in form of a free copy of their product:
+
+Novell, Inc.            UnixWare personal edition + SDK
+The Santa Cruz Operation, Inc.  A SCO OpenServer + SDK
+Ensoniq Corp,           a SoundScape card and extensive amount of assistance
+MediaTrix Peripherals Inc, a AudioTrix Pro card + SDK
+Acer, Inc.              a pair of AcerMagic S23 cards.
+
+In addition the following companies have provided me sufficient amount
+of technical information at least some of their products (free or $$$):
+
+Advanced Gravis Computer Technology Ltd.
+Media Vision Inc.
+Analog Devices Inc.
+Logitech Inc.
+Aztech Labs Inc.
+Crystal Semiconductor Corporation,
+Integrated Circuit Systems Inc.
+OAK Technology
+OPTi
+Turtle Beach
+miro
+Ad Lib Inc. ($$)
+Music Quest Inc. ($$)
+Creative Labs ($$$)
+
+If you have some problems
+=========================
+
+Read the sound HOWTO (sunsite.unc.edu:/pub/Linux/docs/...?).
+Also look at the home page (http://www.opensound.com/ossfree). It may
+contain info about some recent bug fixes.
+
+It's likely that you have some problems when trying to use the sound driver
+first time. Sound cards don't have standard configuration so there are no
+good default configuration to use. Please try to use same I/O, DMA and IRQ
+values for the sound card than with DOS.
+
+If you get an error message when trying to use the driver, please look
+at /var/adm/messages for more verbose error message.
+
+
+The following errors are likely with /dev/dsp and /dev/audio.
+
+        - "No such device or address".
+        This error indicates that there are no suitable hardware for the
+        device file or the sound driver has been compiled without support for
+        this particular device. For example /dev/audio and /dev/dsp will not
+        work if "digitized voice support" was not enabled during "make config".
+        
+        - "Device or resource busy". Probably the IRQ (or DMA) channel 
+        required by the sound card is in use by some other device/driver.
+
+        - "I/O error". Almost certainly (99%) it's an IRQ or DMA conflict.
+        Look at the kernel messages in /var/adm/notice for more info.
+
+        - "Invalid argument". The application is calling ioctl()
+        with impossible parameters. Check that the application is
+        for sound driver version 2.X or later.
+
+Linux installation
+==================
+
+IMPORTANT!      Read this if you are installing a separately
+                distributed version of this driver.
+
+                Check that your kernel version works with this
+                release of the driver (see Readme). Also verify
+                that your current kernel version doesn't have more
+                recent sound driver version than this one. IT'S HIGHLY
+                RECOMMENDED THAT YOU USE THE SOUND DRIVER VERSION THAT
+                IS DISTRIBUTED WITH KERNEL SOURCES.
+
+- When installing separately distributed sound driver you should first
+  read the above notice. Then try to find proper directory where and how
+  to install the driver sources. You should not try to install a separately
+  distributed driver version if you are not able to find the proper way
+  yourself (in this case use the version that is distributed with kernel
+  sources). Remove old version of linux/drivers/sound directory before
+  installing new files.
+
+- To build the device files you need to run the enclosed shell script 
+  (see below). You need to do this only when installing sound driver
+  first time or when upgrading to much recent version than the earlier
+  one.
+
+- Configure and compile Linux as normally (remember to include the
+  sound support during "make config"). Please refer to kernel documentation
+  for instructions about configuring and compiling kernel. File Readme.cards
+  contains card specific instructions for configuring this driver for
+  use with various sound cards.
+
+Boot time configuration (using lilo and insmod) 
+-----------------------------------------------
+
+This information has been removed. Too many users didn't believe
+that it's really not necessary to use this method. Please look at
+Readme of sound driver version 3.0.1 if you still want to use this method.
+
+Problems
+--------
+
+Common error messages:
+
+- /dev/???????: No such file or directory.
+Run the script at the end of this file.
+
+- /dev/???????: No such device.
+You are not running kernel which contains the sound driver. When using
+modularized sound driver this error means that the sound driver is not
+loaded.
+
+- /dev/????: No such device or address.
+Sound driver didn't detect suitable card when initializing. Please look at
+Readme.cards for info about configuring the driver with your card. Also
+check for possible boot (insmod) time error messages in /var/adm/messages.
+
+- Other messages or problems
+Please check http://www.opensound.com/ossfree for more info.
+
+Configuring version 3.8 (for Linux) with some common sound cards
+================================================================
+
+This document describes configuring sound cards with the freeware version of
+Open Sound Systems (OSS/Free). Information about the commercial version
+(OSS/Linux) and its configuration is available from 
+http://www.opensound.com/linux.html. Information presented here is
+not valid for OSS/Linux. 
+
+If you are unsure about how to configure OSS/Free
+you can download the free evaluation version of OSS/Linux from the above
+address. There is a chance that it can autodetect your sound card. In this case
+you can use the information included in soundon.log when configuring OSS/Free.
+
+
+IMPORTANT!      This document covers only cards that were "known" when
+                this driver version was released. Please look at
+                http://www.opensound.com/ossfree for info about
+                cards introduced recently.
+
+                When configuring the sound driver, you should carefully
+                check each sound configuration option (particularly
+                "Support for /dev/dsp and /dev/audio"). The default values
+                offered by these programs are not necessarily valid.
+
+
+THE BIGGEST MISTAKES YOU CAN MAKE
+=================================
+
+1. Assuming that the card is Sound Blaster compatible when it's not.
+--------------------------------------------------------------------
+
+The number one mistake is to assume that your card is compatible with
+Sound Blaster. Only the cards made by Creative Technology or which have
+one or more chips labeled by Creative are SB compatible. In addition there
+are few sound chipsets which are SB compatible in Linux such as ESS1688 or
+Jazz16. Note that SB compatibility in DOS/Windows does _NOT_ mean anything
+in Linux. 
+
+IF YOU REALLY ARE 150% SURE YOU HAVE A SOUND BLASTER YOU CAN SKIP THE REST OF
+THIS CHAPTER.
+
+For most other "supposed to be SB compatible" cards you have to use other
+than SB drivers (see below).  It is possible to get most sound cards to work
+in SB mode but in general it's a complete waste of time. There are several
+problems which you will encounter by using SB mode with cards that are not
+truly SB compatible:
+
+- The SB emulation is at most SB Pro (DSP version 3.x) which means that 
+you get only 8 bit audio (there is always an another ("native") mode which
+gives the 16 bit capability). The 8 bit only operation is the reason why
+many users claim that sound quality in Linux is much worse than in DOS.
+In addition some applications require 16 bit mode and they produce just
+noise with a 8 bit only device.
+- The card may work only in some cases but refuse to work most of the
+time. The SB compatible mode always requires special initialization which is 
+done by the DOS/Windows drivers. This kind of cards work in Linux after
+you have warm booted it after DOS but they don't work after cold boot
+(power on or reset).
+- You get the famous "DMA timed out" messages. Usually all SB clones have
+software selectable IRQ and DMA settings. If the (power on default) values
+currently used by the card don't match configuration of the driver you will
+get the above error message whenever you try to record or play. There are
+few other reasons to the DMA timeout message but using the SB mode seems
+to be the most common cause.
+
+2. Trying to use a PnP (Plug & Play) card just like an ordinary sound card
+--------------------------------------------------------------------------
+
+Plug & Play is a protocol defined by Intel and Microsoft. It lets operating
+systems to easily identify and reconfigure I/O ports, IRQs and DMAs of ISA
+cards. The problem with PnP cards is that the standard Linux doesn't currently
+(versions 2.1.x and earlier) don't support PnP. This means that you will have
+to use some special tricks (see later) to get a PnP card alive. Many PnP cards
+work after they have been initialized but this is not always the case.
+
+There are sometimes both PnP and non-PnP versions of the same sound card.
+The non-PnP version is the original model which usually has been discontinued
+more than an year ago. The PnP version has the same name but with "PnP"
+appended to it (sometimes not). This causes major confusion since the non-PnP
+model works with Linux but the PnP one doesn't.
+
+You should carefully check if "Plug & Play" or "PnP" is mentioned in the name
+of the card or in the documentation or package that came with the card. 
+Everything described in the rest of this document is not necessarily valid for
+PnP models of sound cards even you have managed to wake up the card properly.
+Many PnP cards are simply too different from their non-PnP ancestors which are
+covered by this document.
+
+
+Cards that are not (fully) supported by this driver
+===================================================
+
+See http://www.opensound.com/ossfree for information about sound cards 
+to be supported in future. 
+
+
+How to use sound without recompiling kernel and/or sound driver
+===============================================================
+
+There is a commercial sound driver which comes in precompiled form and doesn't
+require recompiling of the kernel. See http://www.4Front-tech.com/oss.html for
+more info.
+
+
+Configuring PnP cards
+=====================
+
+New versions of most sound cards use the so-called ISA PnP protocol for
+soft configuring their I/O, IRQ, DMA and shared memory resources.
+Currently at least cards made by Creative Technology (SB32 and SB32AWE
+PnP), Gravis (GUS PnP and GUS PnP Pro), Ensoniq (Soundscape PnP) and
+Aztech (some Sound Galaxy models) use PnP technology. The CS4232/4236 audio
+chip by Crystal Semiconductor (Intel Atlantis, HP Pavilion and many other
+motherboards) is also based on PnP technology but there is a "native" driver
+available for it (see information about CS4232 later in this document).
+
+PnP sound cards (as well as most other PnP ISA cards) are not supported
+by this version of the driver . Proper
+support for them should be released during 97 once the kernel level
+PnP support is available.
+
+There is a method to get most of the PnP cards to work. The basic method
+is the following:
+
+1) Boot DOS so the card's DOS drivers have a chance to initialize it.
+2) _Cold_ boot to Linux by using "loadlin.exe".  Hitting ctrl-alt-del
+works with older machines but causes a hard reset of all cards on recent
+(Pentium) machines.
+3) If you have the sound driver in Linux configured properly, the card should
+work now. "Proper" means that I/O, IRQ and DMA settings are the same as in
+DOS. The hard part is to find which settings were used. See the documentation of
+your card for more info.
+
+Windows 95 could work as well as DOS but running loadlin may be difficult.
+Probably you should "shut down" your machine to MS-DOS mode before running it.
+
+Some machines have a BIOS utility for setting PnP resources. This is a good
+way to configure some cards. In this case you don't need to boot DOS/Win95
+before starting Linux.
+
+Another way to initialize PnP cards without DOS/Win95 is a Linux based
+PnP isolation tool. When writing this there is a pre alpha test version
+of such a tool available from ftp://ftp.demon.co.uk/pub/unix/linux/utils. The
+file is called isapnptools-*. Please note that this tool is just a temporary
+solution which may be incompatible with future kernel versions having proper
+support for PnP cards. There are bugs in setting DMA channels in earlier
+versions of isapnptools so at least version 1.6 is required with sound cards.
+
+Yet another way to use PnP cards is to use (commercial) OSS/Linux drivers.  See
+http://www.opensound.com/linux.html for more info. This is probably the way you
+should do it if you don't want to spend time recompiling the kernel and 
+required tools.
+
+
+Read this before trying to configure the driver
+===============================================
+
+There are currently many cards that work with this driver. Some of the cards
+have native support while others work since they emulate some other
+card (usually SB, MSS/WSS and/or MPU401). The following cards have native
+support in the driver. Detailed instructions for configuring these cards
+will be given later in this document.
+
+Pro Audio Spectrum 16 (PAS16) and compatibles:
+        Pro Audio Spectrum 16
+        Pro Audio Studio 16
+        Logitech Sound Man 16
+        NOTE! The original Pro Audio Spectrum as well as the PAS+ are not
+              and will not be supported by the driver.
+
+Media Vision Jazz16 based cards
+        Pro Sonic 16
+        Logitech SoundMan Wave
+        (Other Jazz based cards should work but I don't have any reports
+        about them).
+
+Sound Blasters
+        SB 1.0 to 2.0
+        SB Pro
+        SB 16
+        SB32/64/AWE
+                Configure SB32/64/AWE just like SB16. See lowlevel/README.awe
+                for information about using the wave table synth.
+                NOTE! AWE63/Gold and 16/32/AWE "PnP" cards need to be activated
+                      using isapnptools before they work with OSS/Free.
+        SB16 compatible cards by other manufacturers than Creative.
+                You have been fooled since there are _no_ SB16 compatible
+                cards on the market (as of May 1997). It's likely that your card
+                is compatible just with SB Pro but there is also a non-SB-
+                compatible 16 bit mode. Usually it's MSS/WSS but it could also
+                be a proprietary one like MV Jazz16 or ESS ES688. OPTi
+                MAD16 chips are very common in so called "SB 16 bit cards"
+                (try with the MAD16 driver).
+
+        ======================================================================
+        "Supposed to be SB compatible" cards.
+                Forget the SB compatibility and check for other alternatives
+                first. The only cards that work with the SB driver in
+                Linux have been made by Creative Technology (there is at least
+                one chip on the card with "CREATIVE" printed on it). The
+                only other SB compatible chips are ESS and Jazz16 chips
+                (maybe ALSxxx chips too but they probably don't work).
+                Most other "16 bit SB compatible" cards such as "OPTi/MAD16" or
+                "Crystal" are _NOT_ SB compatible in Linux.
+
+                Practically all sound cards have some kind of SB emulation mode
+                in addition to their native (16 bit) mode. In most cases this
+                (8 bit only) SB compatible mode doesn't work with Linux. If
+                you get it working it may cause problems with games and
+                applications which require 16 bit audio. Some 16 bit only
+                applications don't check if the card actually supports 16 bits.
+                They just dump 16 bit data to a 8 bit card which produces just
+                noise.
+
+                In most cases the 16 bit native mode is supported by Linux.
+                Use the SB mode with "clones" only if you don't find anything
+                better from the rest of this doc.
+        ======================================================================
+
+Gravis Ultrasound (GUS)
+        GUS
+        GUS + the 16 bit option
+        GUS MAX
+        GUS ACE (No MIDI port and audio recording)
+        GUS PnP (with RAM)
+
+MPU-401 and compatibles
+        The driver works both with the full (intelligent mode) MPU-401
+        cards (such as MPU IPC-T and MQX-32M) and with the UART only
+        dumb MIDI ports. MPU-401 is currently the most common MIDI
+        interface. Most sound cards are compatible with it. However,
+        don't enable MPU401 mode blindly. Many cards with native support
+        in the driver have their own MPU401 driver. Enabling the standard one
+        will cause a conflict with these cards. So check if your card is
+        in the list of supported cards before enabling MPU401.
+
+Windows Sound System (MSS/WSS)
+        Even when Microsoft has discontinued their own Sound System card 
+        they managed to make it a standard. MSS compatible cards are based on 
+        a codec chip which is easily available from at least two manufacturers
+        (AD1848 by Analog Devices and CS4231/CS4248 by Crystal Semiconductor).
+        Currently most sound cards are based on one of the MSS compatible codec
+        chips. The CS4231 is used in the high quality cards such as GUS MAX,
+        MediaTrix AudioTrix Pro and TB Tropez (GUS MAX is not MSS compatible). 
+
+        Having a AD1848, CS4248 or CS4231 codec chip on the card is a good
+        sign. Even if the card is not MSS compatible, it could be easy to write
+        support for it. Note also that most MSS compatible cards
+        require special boot time initialization which may not be present
+        in the driver. Also, some MSS compatible cards have native support.
+        Enabling the MSS support with these cards is likely to
+        cause a conflict. So check if your card is listed in this file before
+        enabling the MSS support.
+
+Yamaha FM synthesizers (OPL2, OPL3 (not OPL3-SA) and OPL4)
+        Most sound cards have a FM synthesizer chip. The OPL2 is a 2
+        operator chip used in the original AdLib card. Currently it's used
+        only in the cheapest (8 bit mono) cards. The OPL3 is a 4 operator 
+        FM chip which provides better sound quality and/or more available 
+        voices than the OPL2. The OPL4 is a new chip that has an OPL3 and
+        a wave table synthesizer packed onto the same chip. The driver supports
+        just the OPL3 mode directly. Most cards with an OPL4 (like
+        SM Wave and AudioTrix Pro) support the OPL4 mode using MPU401
+        emulation. Writing a native OPL4 support is difficult
+        since Yamaha doesn't give information about their sample ROM chip.
+
+        Enable the generic OPL2/OPL3 FM synthesizer support if your
+        card has a FM chip made by Yamaha. Don't enable it if your card
+        has a software (TRS) based FM emulator.
+
+        ----------------------------------------------------------------
+        NOTE! OPL3-SA is different chip than the ordinary OPL3. In addition
+        to the FM synth this chip has also digital audio (WSS) and
+        MIDI (MPU401) capabilities. Support for OPL3-SA is described below.
+        ----------------------------------------------------------------
+
+Yamaha OPL3-SA1
+
+        Yamaha OPL3-SA1 (YMF701) is an audio controller chip used on some
+        (Intel) motherboards and on cheap sound cards. It should not be
+        confused with the original OPL3 chip (YMF278) which is entirely
+        different chip. OPL3-SA1 has support for MSS, MPU401 and SB Pro
+        (not used in OSS/Free) in addition to the OPL3 FM synth.
+
+        There are also chips called OPL3-SA2, OPL3-SA3, ..., OPL3SA-N. They
+        are PnP chips and will not work with the OPL3-SA1 driver. You should 
+        use the standard MSS, MPU401 and OPL3 options with these chips and to
+        activate the card using isapnptools.
+
+4Front Technologies SoftOSS
+
+        SoftOSS is a software based wave table emulation which works with
+        any 16 bit stereo sound card. Due to its nature a fast CPU is
+        required (P133 is minimum). Although SoftOSS does _not_ use MMX
+        instructions it has proven out that recent processors (which appear
+        to have MMX) perform significantly better with SoftOSS than earlier
+        ones. For example a P166MMX beats a PPro200. SoftOSS should not be used
+        on 486 or 386 machines.
+
+        The amount of CPU load caused by SoftOSS can be controlled by
+        selecting the CONFIG_SOFTOSS_RATE and CONFIG_SOFTOSS_VOICES
+        parameters properly (they will be prompted by make config). It's
+        recommended to set CONFIG_SOFTOSS_VOICES to 32. If you have a
+        P166MMX or faster (PPro200 is not faster) you can set
+        CONFIG_SOFTOSS_RATE to 44100 (kHz). However with slower systems it
+        recommended to use sampling rates around 22050 or even 16000 kHz.
+        Selecting too high values for these parameters may hang your
+        system when playing MIDI files with hight degree of polyphony
+        (number of concurrently playing notes). It's also possible to
+        decrease CONFIG_SOFTOSS_VOICES. This makes it possible to use
+        higher sampling rates. However using fewer voices decreases
+        playback quality more than decreasing the sampling rate.
+
+        SoftOSS keeps the samples loaded on the system's RAM so much RAM is
+        required. SoftOSS should never be used on machines with less than 16 MB
+        of RAM since this is potentially dangerous (you may accidentally run out
+        of memory which probably crashes the machine). 
+
+        SoftOSS implements the wave table API originally designed for GUS. For
+        this reason all applications designed for GUS should work (at least
+        after minor modifications). For example gmod/xgmod and playmidi -g are
+        known to work.
+
+        To work SoftOSS will require GUS compatible
+        patch files to be installed on the system (in /dos/ultrasnd/midi). You
+        can use the public domain MIDIA patchset available from several ftp
+        sites.
+
+        *********************************************************************
+        IMPORTANT NOTICE! The original patch set distributed with the Gravis 
+        Ultrasound card is not in public domain (even though it's available from
+        some FTP sites). You should contact Voice Crystal (www.voicecrystal.com)
+        if you like to use these patches with SoftOSS included in OSS/Free.
+        *********************************************************************
+
+PSS based cards (AD1848 + ADSP-2115 + Echo ESC614 ASIC)
+        Analog Devices and Echo Speech have together defined a sound card
+        architecture based on the above chips. The DSP chip is used
+        for emulation of SB Pro, FM and General MIDI/MT32.
+
+        There are several cards based on this architecture. The most known
+        ones are Orchid SW32 and Cardinal DSP16. 
+
+        The driver supports downloading DSP algorithms to these cards.
+
+        NOTE! You will have to use the "old" config script when configuring
+        PSS cards.
+
+MediaTrix AudioTrix Pro
+        The ATP card is built around a CS4231 codec and an OPL4 synthesizer
+        chips. The OPL4 mode is supported by a microcontroller running a
+        General MIDI emulator. There is also a SB 1.5 compatible playback mode.
+
+Ensoniq SoundScape and compatibles
+        Ensoniq has designed a sound card architecture based on the
+        OTTO synthesizer chip used in their professional MIDI synthesizers.
+        Several companies (including Ensoniq, Reveal and Spea) are selling
+        cards based on this architecture.
+
+        NOTE! The SoundScape PnP is not supported by OSS/Free. Ensoniq VIVO and
+        VIVO90 cards are not compatible with Soundscapes so the Soundscape
+        driver will not work with them. You may want to use OSS/Linux with these
+        cards.
+
+OPTi MAD16 and Mozart based cards
+        The Mozart (OAK OTI-601), MAD16 (OPTi 82C928), MAD16 Pro (OPTi 82C929),
+        OPTi 82C924/82C925 (in _non_ PnP mode) and OPTi 82C930 interface
+        chips are used in many different sound cards, including some
+        cards by Reveal miro and Turtle Beach (Tropez). The purpose of these
+        chips is to connect other audio components to the PC bus. The
+        interface chip performs address decoding for the other chips.
+        NOTE! Tropez Plus is not MAD16 but CS4232 based.
+        NOTE! MAD16 PnP cards (82C924, 82C925, 82C931) are not MAD16 compatible
+        in the PnP mode. You will have to use them in MSS mode after having
+        initialized them using isapnptools or DOS. 82C931 probably requires
+        initialization using DOS/Windows (running isapnptools is not enough).
+        It's possible to use 82C931 with OSS/Free by jumpering it to non-PnP
+        mode (provided that the card has a jumper for this). In non-PnP mode
+        82C931 is compatible with 82C930 and should work with the MAD16 driver
+        (without need to use isapnptools or DOS to initialize it). All OPTi
+        chips are supported by OSS/Linux (both in PnP and non-PnP modes).
+
+Audio Excel DSP16 
+        Support for this card was written by Riccardo Faccetti
+        (riccardo@cdc8g5.cdc.polimi.it). The AEDSP16 driver included in
+        the lowlevel/ directory. To use it you should enable the
+        "Additional low level drivers" option.
+
+Crystal CS4232 and CS4236 based cards such as AcerMagic S23, TB Tropez _Plus_ and 
+        many PC motherboards (Compaq, HP, Intel, ...)
+        CS4232 is a PnP multimedia chip which contains a CS3231A codec,
+        SB and MPU401 emulations. There is support for OPL3 too.
+        Unfortunately the MPU401 mode doesn't work (I don't know how to
+        initialize it). CS4236 is an enhanced (compatible) version of CS4232.
+        NOTE! Don't ever try to use isapnptools with CS4232 since this will just
+        freeze your machine (due to chip bugs). If you have problems in getting
+        CS4232 working you could try initializing it with DOS (CS4232C.EXE) and
+        then booting Linux using loadlin. CS4232C.EXE loads a secret firmware
+        patch which is not documented by Crystal.
+
+Turtle Beach Maui and Tropez "classic"
+        This driver version supports sample, patch and program loading commands
+        described in the Maui/Tropez User's manual. 
+        There is now full initialization support too. The audio side of
+        the Tropez is based on the MAD16 chip (see above).
+        NOTE! Tropez Plus is different card than Tropez "classic" and will not
+        work fully in Linux. You can get audio features working by configuring
+        the card as a CS4232 based card (above).
+
+
+Jumpers and software configuration
+==================================
+
+Some of the earliest sound cards were jumper configurable. You have to
+configure the driver use I/O, IRQ and DMA settings
+that match the jumpers. Just few 8 bit cards are fully jumper 
+configurable (SB 1.x/2.x, SB Pro and clones).
+Some cards made by Aztech have an EEPROM which contains the 
+config info. These cards behave much like hardware jumpered cards.
+
+Most cards have jumper for the base I/O address but other parameters
+are software configurable. Sometimes there are few other jumpers too.
+
+Latest cards are fully software configurable or they are PnP ISA
+compatible. There are no jumpers on the board.
+
+The driver handles software configurable cards automatically. Just configure
+the driver to use I/O, IRQ and DMA settings which are known to work.
+You could usually use the same values than with DOS and/or Windows.
+Using different settings is possible but not recommended since it may cause
+some trouble (for example when warm booting from an OS to another or
+when installing new hardware to the machine).
+
+Sound driver sets the soft configurable parameters of the card automatically
+during boot. Usually you don't need to run any extra initialization
+programs when booting Linux but there are some exceptions. See the
+card-specific instructions below for more info.
+
+The drawback of software configuration is that the driver needs to know
+how the card must be initialized. It cannot initialize unknown cards
+even if they are otherwise compatible with some other cards (like SB,
+MPU401 or Windows Sound System).
+
+
+What if your card was not listed above?
+=======================================
+
+The first thing to do is to look at the major IC chips on the card.
+Many of the latest sound cards are based on some standard chips. If you
+are lucky, all of them could be supported by the driver. The most common ones
+are the OPTi MAD16, Mozart, SoundScape (Ensoniq) and the PSS architectures
+listed above. Also look at the end of this file for list of unsupported
+cards and the ones which could be supported later.
+
+The last resort is to send _exact_ name and model information of the card
+to me together with a list of the major IC chips (manufactured, model) to 
+me. I could then try to check if your card looks like something familiar.
+
+There are many more cards in the world than listed above. The first thing to
+do with these cards is to check if they emulate some other card or interface
+such as SB, MSS and/or MPU401. In this case there is a chance to get the
+card to work by booting DOS before starting Linux (boot DOS, hit ctrl-alt-del
+and boot Linux without hard resetting the machine). In this method the
+DOS based driver initializes the hardware to use known I/O, IRQ and DMA
+settings. If sound driver is configured to use the same settings, everything
+should work OK.
+
+
+Configuring sound driver (with Linux)
+=====================================
+
+The sound driver is currently distributed as part of the Linux kernel. The 
+files are in /usr/src/linux/drivers/sound/. 
+
+****************************************************************************
+*       ALWAYS USE THE SOUND DRIVER VERSION WHICH IS DISTRIBUTED WITH      *
+*       THE KERNEL SOURCE PACKAGE YOU ARE USING. SOME ALPHA AND BETA TEST  *
+*       VERSIONS CAN BE INSTALLED FROM A SEPARATELY DISTRIBUTED PACKAGE    *
+*       BUT CHECK THAT THE PACKAGE IS NOT MUCH OLDER (OR NEWER) THAN THE   *
+*       KERNEL YOU ARE USING. IT'S POSSIBLE THAT THE KERNEL/DRIVER         *
+*       INTERFACE CHANGES BETWEEN KERNEL RELEASES WHICH MAY CAUSE SOME     *
+*       INCOMPATIBILITY PROBLEMS.                                          *
+*                                                                          *
+*       IN CASE YOU INSTALL A SEPARATELY DISTRIBUTED SOUND DRIVER VERSION, *
+*       BE SURE TO REMOVE OR RENAME THE OLD SOUND DRIVER DIRECTORY BEFORE  *
+*       INSTALLING THE NEW ONE. LEAVING OLD FILES TO THE SOUND DRIVER      *
+*       DIRECTORY _WILL_ CAUSE PROBLEMS WHEN THE DRIVER IS USED OR         *
+*       COMPILED.                                                          *
+****************************************************************************
+
+To configure the driver, run "make config" in the kernel source directory
+(/usr/src/linux). Answer "y" or "m" to the question about Sound card support
+(after the questions about mouse, CD-ROM, ftape, etc. support).  Questions
+about options for sound will then be asked.
+
+After configuring the kernel and sound driver and compile the kernel 
+following instructions in the kernel README.
+
+The sound driver configuration dialog
+-------------------------------------
+
+Sound configuration starts by making some yes/no questions. Be careful
+when answering to these questions since answering y to a question may
+prevent some later ones from being asked. For example don't answer y to
+the first question (PAS16) if you don't really have a PAS16. Don't enable
+more cards than you really need since they just consume memory. Also
+some drivers (like MPU401) may conflict with your SCSI controller and
+prevent kernel from booting. If you card was in the list of supported
+cards (above), please look at the card specific config instructions
+(later in this file) before starting to configure. Some cards must be
+configured in way which is not obvious.
+
+So here is the beginning of the config dialog. Answer 'y' or 'n' to these
+questions. The default answer is shown so that (y/n) means 'y' by default and
+(n/y) means 'n'. To use the default value, just hit ENTER. But be careful
+since using the default _doesn't_ guarantee anything.
+
+Note also that all questions may not be asked. The configuration program
+may disable some questions depending on the earlier choices. It may also
+select some options automatically as well.
+
+  "ProAudioSpectrum 16 support",
+        - Answer 'y'_ONLY_ if you have a Pro Audio Spectrum _16_,
+          Pro Audio Studio 16 or Logitech SoundMan 16 (be sure that
+          you read the above list correctly). Don't answer 'y' if you
+          have some other card made by Media Vision or Logitech since they
+          are not PAS16 compatible.
+          NOTE! Since 3.5-beta10 you need to enable SB support (next question)
+          if you want to use the SB emulation of PAS16. It's also possible to
+          the emulation if you want to use a true SB card together with PAS16
+          (there is another question about this that is asked later).
+  "Sound Blaster support",
+        - Answer 'y' if you have an original SB card made by Creative Labs
+          or a full 100% hardware compatible clone (like Thunderboard or
+          SM Games). If your card was in the list of supported cards (above),
+          please look at the card specific instructions later in this file
+          before answering this question. For an unknown card you may answer 
+          'y' if the card claims to be SB compatible.
+         Enable this option also with PAS16 (changed since v3.5-beta9).
+
+         Don't enable SB if you have a MAD16 or Mozart compatible card.
+
+  "Generic OPL2/OPL3 FM synthesizer support",
+        - Answer 'y' if your card has a FM chip made by Yamaha (OPL2/OPL3/OPL4).
+          Answering 'y' is usually a safe and recommended choice. However some
+          cards may have software (TSR) FM emulation. Enabling FM support
+          with these cards may cause trouble. However I don't currently know
+          such cards.
+  "Gravis Ultrasound support",
+        - Answer 'y' if you have GUS or GUS MAX. Answer 'n' if you don't
+          have GUS since the GUS driver consumes much memory.
+          Currently I don't have experiences with the GUS ACE so I don't
+          know what to answer with it.
+  "MPU-401 support (NOT for SB16)",
+        - Be careful with this question. The MPU401 interface is supported
+          by almost any sound card today. However some natively supported cards
+          have their own driver for MPU401. Enabling the MPU401 option with
+          these cards will cause a conflict. Also enabling MPU401 on a system
+          that doesn't really have a MPU401 could cause some trouble. If your
+          card was in the list of supported cards (above), please look at
+          the card specific instructions later in this file.
+
+          In MOST cases this MPU401 driver should only be used with "true"
+          MIDI-only MPU401 professional cards. In most other cases there
+          is another way to get the MPU401 compatible interface of a
+          sound card to work.
+          Support for the MPU401 compatible MIDI port of SB16, ESS1688
+          and MV Jazz16 cards is included in the SB driver. Use it instead
+          of this separate MPU401 driver with these cards. As well 
+          Soundscape, PSS and Maui drivers include their own MPU401
+          options.
+
+          It's safe to answer 'y' if you have a true MPU401 MIDI interface
+          card. 
+  "6850 UART Midi support",
+        - It's safe to answer 'n' to this question in all cases. The 6850
+          UART interface is so rarely used.
+  "PSS (ECHO-ADI2111) support",
+        - Answer 'y' only if you have Orchid SW32, Cardinal DSP16 or some
+          other card based on the PSS chipset (AD1848 codec + ADSP-2115
+          DSP chip + Echo ESC614 ASIC CHIP).
+  "16 bit sampling option of GUS (_NOT_ GUS MAX)",
+        - Answer 'y' if you have installed the 16 bit sampling daughtercard
+          to your GUS. Answer 'n' if you have GUS MAX. Enabling this option
+          disables GUS MAX support.
+  "GUS MAX support",
+        - Answer 'y' only if you have a GUS MAX.
+  "Microsoft Sound System support",
+        - Again think carefully before answering 'y' to this question. It's
+          safe to answer 'y' in case you have the original Windows Sound 
+          System card made by Microsoft or Aztech SG 16 Pro (or NX16 Pro).
+          Also you may answer 'y' in case your card was not listed earlier
+          in this file. For cards having native support in the driver, consult
+          the card specific instructions later in this file. Some drivers
+          have their own MSS support and enabling this option will cause a
+          conflict. 
+          Note! The MSS driver permits configuring two DMA channels. This is a
+          "nonstandard" feature and works only with very few cards (if any).
+          In most cases the second DMA channel should be disabled or set to
+          the same channel than the first one. Trying to configure two separate
+          channels with cards that don't support this feature will prevent
+          audio (at least recording) from working.
+  "Ensoniq Soundscape support",
+        - Answer 'y' if you have a sound card based on the Ensoniq SoundScape
+          chipset. Such cards are being manufactured at least by Ensoniq,
+          Spea and Reveal (note that Reveal makes other cards also).  The oldest
+          cards made by Spea don't work properly with Linux. 
+          Soundscape PnP as well as Ensoniq VIVO work only with the commercial
+          OSS/Linux version.
+  "MediaTrix AudioTrix Pro support",
+        - Answer 'y' if you have the AudioTrix Pro.
+  "Support for MAD16 and/or Mozart based cards",
+        - Answer y if your card has a Mozart (OAK OTI-601) or MAD16
+          (OPTi 82C928, 82C929, 82C924/82C925 or 82C930) audio interface chip. 
+          These chips are
+          currently quite common so it's possible that many no-name cards
+          have one of them. In addition the MAD16 chip is used in some
+          cards made by known manufacturers such as Turtle Beach (Tropez),
+          Reveal (some models) and Diamond (some recent models).
+          Note OPTi 82C924 and 82C925 are MAD16 compatible only in non PnP
+          mode (jumper selectable on many cards).
+  "Support for TB Maui"
+        - This enables TB Maui specific initialization. Works with TB Maui
+        and TB Tropez (may not work with Tropez Plus).
+
+
+Then the configuration program asks some y/n questions about the higher
+level services. It's recommended to answer 'y' to each of these questions.
+Answer 'n' only if you know you will not need the option.
+
+  "MIDI interface support",
+        - Answering 'n' disables /dev/midi## devices and access to any
+          MIDI ports using /dev/sequencer and /dev/music. This option
+          also affects any MPU401 and/or General MIDI compatible devices.
+  "FM synthesizer (YM3812/OPL-3) support",
+        - Answer 'y' here.
+  "/dev/sequencer support",
+        - Answering 'n' disables /dev/sequencer and /dev/music.
+
+Entering the I/O, IRQ and DMA config parameters
+-----------------------------------------------
+
+After the above questions the configuration program prompts for the
+card specific configuration information. Usually just a set of
+I/O address, IRQ and DMA numbers are asked. With some cards the program
+asks for some files to be used during initialization of the card. For example
+many cards have a DSP chip or microprocessor which must be initialized by
+downloading a program (microcode) file to the card.
+
+Instructions for answering these questions are given in the next section.
+
+
+Card specific information
+=========================
+
+This section gives additional instructions about configuring some cards.
+Please refer manual of your card for valid I/O, IRQ and DMA numbers. Using
+the same settings with DOS/Windows and Linux is recommended. Using
+different values could cause some problems when switching between
+different operating systems.
+
+Sound Blasters (the original ones by Creative)
+---------------------------------------------
+
+NOTE! Check if you have a PnP Sound Blaster (cards sold after summer 1995
+      are almost certainly PnP ones). With PnP cards you should use isapnptools
+      to activate them (see above).
+
+It's possible to configure these cards to use different I/O, IRQ and
+DMA settings. Since the possible/default settings have changed between various
+models, you have to consult manual of your card for the proper ones. It's
+a good idea to use the same values than with DOS/Windows. With SB and SB Pro
+it's the only choice. SB16 has software selectable IRQ and DMA channels but
+using different values with DOS and Linux is likely to cause troubles. The
+DOS driver is not able to reset the card properly after warm boot from Linux
+if Linux has used different IRQ or DMA values.
+
+The original (steam) Sound Blaster (versions 1.x and 2.x) use always
+DMA1. There is no way to change it.
+
+The SB16 needs two DMA channels. A 8 bit one (1 or 3) is required for
+8 bit operation and a 16 bit one (5, 6 or 7) for the 16 bit mode. In theory
+it's possible to use just one (8 bit) DMA channel by answering the 8 bit
+one when the configuration program asks for the 16 bit one. This may work
+in some systems but is likely to cause terrible noise on some other systems.
+
+It's possible to use two SB16/32/64 at the same time. To do this you should
+first configure OSS/Free for one card. Then edit local.h manually and define
+SB2_BASE, SB2_IRQ, SB2_DMA and SB2_DMA2 for the second one. You can't get
+the OPL3, MIDI and EMU8000 devices of the second card to work. If you are
+going to use two PnP Sound Blasters, ensure that they are of different model
+and have different PnP IDs. There is no way to get two cards with the same
+card ID and serial number to work. The easiest way to check this is trying 
+if isapnptools can see both cards or just one.
+
+NOTE!   Don't enable the SM Games option (asked by the configuration program)
+        if you are not 101% sure that your card is a Logitech Soundman Games
+        (not a SM Wave or SM16).
+
+SB Clones
+---------
+
+First of all: There are no SB16 clones. There are SB Pro clones with a
+16 bit mode which is not SB16 compatible. The most likely alternative is that
+the 16 bit mode means MSS/WSS.
+
+There are just a few fully 100% hardware SB or SB Pro compatible cards.
+I know just Thunderboard and SM Games. Other cards require some kind of
+hardware initialization before they become SB compatible. Check if your card
+was listed in the beginning of this file. In this case you should follow
+instructions for your card later in this file.
+
+For other not fully SB clones you may try initialization using DOS in
+the following way:
+
+        - Boot DOS so that the card specific driver gets run.
+        - Hit ctrl-alt-del (or use loadlin) to boot Linux. Don't
+          switch off power or press the reset button.
+        - If you use the same I/O, IRQ and DMA settings in Linux, the
+          card should work.
+
+If your card is both SB and MSS compatible, I recommend using the MSS mode.
+Most cards of this kind are not able to work in the SB and the MSS mode 
+simultaneously. Using the MSS mode provides 16 bit recording and playback.
+
+ProAudioSpectrum 16 and compatibles
+-----------------------------------
+
+PAS16 has a SB emulation chip which can be used together with the native
+(16 bit) mode of the card. To enable this emulation you should configure 
+the driver to have SB support too (this has been changed since version
+3.5-beta9 of this driver).
+
+With current driver versions it's also possible to use PAS16 together with 
+another SB compatible card. In this case you should configure SB support
+for the other card and to disable the SB emulation of PAS16 (there is a
+separate questions about this).
+
+With PAS16 you can use two audio device files at the same time. /dev/dsp (and
+/dev/audio) is connected to the 8/16 bit native codec and the /dev/dsp1 (and
+/dev/audio1) is connected to the SB emulation (8 bit mono only).
+
+Gravis Ultrasound
+-----------------
+
+There are many different revisions of the Ultrasound card (GUS). The
+earliest ones (pre 3.7) don't have a hardware mixer. With these cards
+the driver uses a software emulation for synth and pcm playbacks. It's
+also possible to switch some of the inputs (line in, mic) off by setting
+mixer volume of the channel level below 10%. For recording you have
+to select the channel as a recording source and to use volume above 10%.
+
+GUS 3.7 has a hardware mixer.
+
+GUS MAX and the 16 bit sampling daughtercard have a CS4231 codec chip which 
+also contains a mixer.
+
+Configuring GUS is simple. Just enable the GUS support and GUS MAX or
+the 16 bit daughtercard if you have them. Note that enabling the daughter
+card disables GUS MAX driver.
+
+NOTE for owners of the 16 bit daughtercard: By default the daughtercard
+uses /dev/dsp (and /dev/audio). Command "ln -sf /dev/dsp1 /dev/dsp"
+selects the daughter card as the default device.
+
+With just the standard GUS enabled the configuration program prompts
+for the I/O, IRQ and DMA numbers for the card. Use the same values than
+with DOS.
+
+With the daughter card option enabled you will be prompted for the I/O,
+IRQ and DMA numbers for the daughter card. You have to use different I/O
+and DMA values than for the standard GUS. The daughter card permits
+simultaneous recording and playback. Use /dev/dsp (the daughtercard) for
+recording and /dev/dsp1 (GUS GF1) for playback.
+
+GUS MAX uses the same I/O address and IRQ settings than the original GUS
+(GUS MAX = GUS + a CS4231 codec). In addition an extra DMA channel may be used.
+Using two DMA channels permits simultaneous playback using two devices
+(dev/dsp0 and /dev/dsp1). The second DMA channel is required for
+full duplex audio.
+To enable the second DMA channels, give a valid DMA channel when the config
+program asks for the GUS MAX DMA (entering -1 disables the second DMA).
+Using 16 bit DMA channels (5,6 or 7) is recommended.
+
+If you have problems in recording with GUS MAX, you could try to use
+just one 8 bit DMA channel. Recording will not work with one DMA
+channel if it's a 16 bit one.
+
+Microphone input of GUS MAX is connected to mixer in little bit nonstandard
+way. There is actually two microphone volume controls. Normal "mic" controls
+only recording level. Mixer control "speaker" is used to control volume of
+microphone signal connected directly to line/speaker out. So just decrease
+volume of "speaker" if you have problems with microphone feedback.
+
+GUS ACE works too but any attempt to record or to use the MIDI port
+will fail.
+
+GUS PnP (with RAM) is partially supported but it needs to be initialized using
+DOS or isapnptools before starting the driver.
+
+MPU401 and Windows Sound System
+-------------------------------
+
+Again. Don't enable these options in case your card is listed
+somewhere else in this file.
+
+Configuring these cards is obvious (or it should be). With MSS
+you should probably enable the OPL3 synth also since
+most MSS compatible cards have it. However check that this is true
+before enabling OPL3.
+
+Sound driver supports more than one MPU401 compatible cards at the same time
+but the config program asks config info for just the first of them.
+Adding the second or third MPU interfaces must be done manually by
+editing sound/local.h (after running the config program). Add defines for
+MPU2_BASE & MPU2_IRQ (and MPU3_BASE & MPU3_IRQ) to the file.
+
+CAUTION!
+
+The default I/O base of Adaptec AHA-1542 SCSI controller is 0x330 which
+is also the default of the MPU401 driver. Don't configure the sound driver to
+use 0x330 as the MPU401 base if you have a AHA1542. The kernel will not boot
+if you make this mistake.
+
+PSS
+---
+
+Even the PSS cards are compatible with SB, MSS and MPU401, you must not
+enable these options when configuring the driver. The configuration
+program handles these options itself. (You may use the SB, MPU and MSS options
+together with PSS if you have another card on the system).
+
+The PSS driver enables MSS and MPU401 modes of the card. SB is not enabled 
+since it doesn't work concurrently with MSS. The driver loads also a
+DSP algorithm which is used to for the general MIDI emulation. The
+algorithm file (.ld) is read by the config program and written to a
+file included when the pss.c is compiled. For this reason the config
+program asks if you want to download the file. Use the genmidi.ld file
+distributed with the DOS/Windows drivers of the card (don't use the mt32.ld).
+With some cards the file is called 'synth.ld'. You must have access to
+the file when configuring the driver. The easiest way is to mount the DOS
+partition containing the file with Linux. 
+
+It's possible to load your own DSP algorithms and run them with the card.
+Look at the directory pss_test of snd-util-3.0.tar.gz for more info.
+
+AudioTrix Pro
+-------------
+
+You have to enable the OPL3 and SB (not SB Pro or SB16) drivers in addition
+to the native AudioTrix driver. Don't enable MSS or MPU drivers.
+
+Configuring ATP is little bit tricky since it uses so many I/O, IRQ and
+DMA numbers. Using the same values than with DOS/Win is a good idea. Don't
+attempt to use the same IRQ or DMA channels twice.
+
+The SB mode of ATP is implemented so the ATP driver just enables SB
+in the proper address. The SB driver handles the rest. You have to configure
+both the SB driver and the SB mode of ATP to use the same IRQ, DMA and I/O
+settings.
+
+Also the ATP has a microcontroller for the General MIDI emulation (OPL4).
+For this reason the driver asks for the name of a file containing the
+microcode (TRXPRO.HEX). This file is usually located in the directory
+where the DOS drivers were installed. You must have access to this file
+when configuring the driver. 
+
+If you have the effects daughtercard, it must be initialized by running
+the setfx program of snd-util-3.0.tar.gz package. This step is not required
+when using the (future) binary distribution version of the driver.
+
+Ensoniq SoundScape
+------------------
+
+NOTE!   The new PnP SoundScape is not supported yet. Soundscape compatible
+        cards made by Reveal don't work with Linux. They use older revision
+        of the Soundscape chipset which is not fully compatible with
+        newer cards made by Ensoniq.
+
+The SoundScape driver handles initialization of MSS and MPU supports
+itself so you don't need to enable other drivers than SoundScape
+(enable also the /dev/dsp, /dev/sequencer and MIDI supports).
+
+!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
+!!!!!                                                                   !!!!
+!!!!! NOTE! Before version 3.5-beta6 there WERE two sets of audio       !!!!
+!!!!!       device files (/dev/dsp0 and /dev/dsp1). The first one WAS   !!!!
+!!!!!       used only for card initialization and the second for audio  !!!!
+!!!!!       purposes. It WAS required to change /dev/dsp (a symlink) to !!!!
+!!!!!       point to /dev/dsp1.                                         !!!!
+!!!!!                                                                   !!!!
+!!!!!       This is not required with OSS versions 3.5-beta6 and later  !!!!
+!!!!!       since there is now just one audio device file. Please       !!!!
+!!!!!       change /dev/dsp to point back to /dev/dsp0 if you are       !!!!
+!!!!!       upgrading from an earlier driver version using              !!!!
+!!!!!       (cd /dev;rm dsp;ln -s dsp0 dsp).                            !!!!
+!!!!!                                                                   !!!!
+!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
+
+The configuration program asks one DMA channel and two interrupts. One IRQ
+and one DMA is used by the MSS codec. The second IRQ is required for the
+MPU401 mode (you have to use different IRQs for both purposes).
+There were earlier two DMA channels for SoundScape but the current driver
+version requires just one.
+
+The SoundScape card has a Motorola microcontroller which must initialized
+_after_ boot (the driver doesn't initialize it during boot).
+The initialization is done by running the 'ssinit' program which is
+distributed in the snd-util-3.0.tar.gz package. You have to edit two
+defines in the ssinit.c and then compile the program. You may run ssinit 
+manually (after each boot) or add it to /etc/rc.d/rc.local.
+
+The ssinit program needs the microcode file that comes with the DOS/Windows
+driver of the card. You will need to use version 1.30.00 or later
+of the microcode file (sndscape.co0 or sndscape.co1 depending on
+your card model). THE OLD sndscape.cod WILL NOT WORK. IT WILL HANG YOUR
+MACHINE. The only way to get the new microcode file is to download
+and install the DOS/Windows driver from ftp://ftp.ensoniq.com/pub.
+
+Then you have to select the proper microcode file to use: soundscape.co0
+is the right one for most cards and sndscape.co1 is for few (older) cards
+made by Reveal and/or Spea. The driver has capability to detect the card
+version during boot. Look at the boot log messages in /var/adm/messages
+and locate the sound driver initialization message for the SoundScape
+card. If the driver displays string <Ensoniq Soundscape (old)>, you have
+an old card and you will need to use sndscape.co1. For other cards use
+soundscape.co0. New Soundscape revisions such as Elite and PnP use
+code files with higher numbers (.co2, .co3, etc.).
+
+NOTE!   Ensoniq Soundscape VIVO is not compatible with other Soundscape cards.
+        Currently it's possible to use it in Linux only with OSS/Linux
+        drivers.
+
+Check /var/adm/messages after running ssinit. The driver prints
+the board version after downloading the microcode file. That version
+number must match the number in the name of the microcode file (extension).
+
+Running ssinit with a wrong version of the sndscape.co? file is not
+dangerous as long as you don't try to use a file called sndscape.cod.
+If you have initialized the card using a wrong microcode file (sounds
+are terrible), just modify ssinit.c to use another microcode file and try
+again. It's possible to use an earlier version of sndscape.co[01] but it
+may sound weird.
+
+MAD16 (Pro) and Mozart
+----------------------
+
+You need to enable just the MAD16 /Mozart support when configuring
+the driver. _Don't_ enable SB, MPU401 or MSS. However you will need the
+/dev/audio, /dev/sequencer and MIDI supports.
+
+Mozart and OPTi 82C928 (the original MAD16) chips don't support
+MPU401 mode so enter just 0 when the configuration program asks the
+MPU/MIDI I/O base. The MAD16 Pro (OPTi 82C929) and 82C930 chips have MPU401
+mode.
+
+TB Tropez is based on the 82C929 chip. It has two MIDI ports.
+The one connected to the MAD16 chip is the second one (there is a second
+MIDI connector/pins somewhere??). If you have not connected the second MIDI
+port, just disable the MIDI port of MAD16. The 'Maui' compatible synth of
+Tropez is jumper configurable and not connected to the MAD16 chip (the
+Maui driver can be used with it).
+
+Some MAD16 based cards may cause feedback, whistle or terrible noise if the
+line3 mixer channel is turned too high. This happens at least with Shuttle
+Sound System. Current driver versions set volume of line3 low enough so
+this should not be a problem.
+
+If you have a MAD16 card which have an OPL4 (FM + Wave table) synthesizer
+chip (_not_ an OPL3), you have to append a line containing #define MAD16_OPL4
+to the file linux/drivers/sound/local.h (after running make config).
+
+MAD16 cards having a CS4231 codec support full duplex mode. This mode
+can be enabled by configuring the card to use two DMA channels. Possible
+DMA channel pairs are: 0&1, 1&0 and 3&0.
+
+NOTE! Cards having an OPTi 82C924/82C925 chip work with OSS/Free only in
+non-PnP mode (usually jumper selectable). The PnP mode is supported only
+by OSS/Linux.
+
+MV Jazz (ProSonic)
+------------------
+
+The Jazz16 driver is just a hack made to the SB Pro driver. However it works
+fairly well. You have to enable SB, SB Pro (_not_ SB16) and MPU401 supports
+when configuring the driver. The configuration program asks later if you
+want support for MV Jazz16 based cards (after asking SB base address). Answer
+'y' here and the driver asks the second (16 bit) DMA channel.
+
+The Jazz16 driver uses the MPU401 driver in a way which will cause
+problems if you have another MPU401 compatible card. In this case you must
+give address of the Jazz16 based MPU401 interface when the config
+program prompts for the MPU401 information. Then look at the MPU401
+specific section for instructions about configuring more than one MPU401 cards.
+
+Logitech Soundman Wave
+----------------------
+
+Read the above MV Jazz specific instructions first.
+
+The Logitech SoundMan Wave (don't confuse this with the SM16 or SM Games) is
+a MV Jazz based card which has an additional OPL4 based wave table
+synthesizer. The OPL4 chip is handled by an on board microcontroller
+which must be initialized during boot. The config program asks if
+you have a SM Wave immediately after asking the second DMA channel of jazz16.
+If you answer 'y', the config program will ask name of the file containing
+code to be loaded to the microcontroller. The file is usually called
+MIDI0001.BIN and it's located in the DOS/Windows driver directory. The file
+may also be called as TSUNAMI.BIN or something else (older cards?).
+
+The OPL4 synth will be inaccessible without loading the microcontroller code.
+
+Also remember to enable SB MPU401 support if you want to use the OPL4 mode.
+(Don't enable the 'normal' MPU401 device as with some earlier driver
+versions (pre 3.5-alpha8)).
+
+NOTE!   Don't answer 'y' when the driver asks about SM Games support
+        (the next question after the MIDI0001.BIN name). However
+        answering 'y' doesn't cause damage your computer so don't panic. 
+
+Sound Galaxies
+--------------
+
+There are many different Sound Galaxy cards made by Aztech. The 8 bit
+ones are fully SB or SB Pro compatible and there should be no problems
+with them. 
+
+The older 16 bit cards (SG Pro16, SG NX Pro16, Nova and Lyra) have
+an EEPROM chip for storing the configuration data. There is a microcontroller
+which initializes the card to match the EEPROM settings when the machine
+is powered on. These cards actually behave just like they have jumpers
+for all of the settings. Configure driver for MSS, MPU, SB/SB Pro  and OPL3 
+supports with these cards. 
+
+There are some new Sound Galaxies in the market. I have no experience with
+them so read the card's manual carefully.
+
+ESS ES1688 and ES688 'AudioDrive' based cards
+---------------------------------------------
+
+Support for these two ESS chips is embedded in the SB driver.
+Configure these cards just like SB. Enable the 'SB MPU401 MIDI port'
+if you want to use MIDI features of ES1688. ES688 doesn't have MPU mode
+so you don't need to enable it (the driver uses normal SB MIDI automatically
+with ES688).
+
+NOTE! ESS cards are not compatible with MSS/WSS so don't worry if MSS support
+of OSS doesn't work with it.
+
+There are some ES1688/688 based sound cards and (particularly) motherboards
+which use software configurable I/O port relocation feature of the chip.
+This ESS proprietary feature is supported only by OSS/Linux.
+
+There are ES1688 based cards which use different interrupt pin assignment than
+recommended by ESS (5, 7, 9/2 and 10). In this case all IRQs don't work.
+At least a card called (Pearl?) Hypersound 16 supports IRQ 15 but it doesn't
+work.
+
+ES1868 is a PnP chip which is (supposed to be) compatible with ESS1688
+probably works with OSS/Free after initialization using isapnptools.
+
+Reveal cards
+------------
+
+There are several different cards made/marketed by Reveal. Some of them
+are compatible with SoundScape and some use the MAD16 chip. You may have
+to look at the card and try to identify its origin.
+
+Diamond
+-------
+
+The oldest (Sierra Aria based) sound cards made by Diamond are not supported
+(they may work if the card is initialized using DOS). The recent (LX?)
+models are based on the MAD16 chip which is supported by the driver.
+
+Audio Excel DSP16
+-----------------
+
+Support for this card is currently not functional. A new driver for it
+should be available later this year.
+
+PCMCIA cards
+------------
+
+Sorry, can't help. Some cards may work and some don't.
+
+TI TM4000M notebooks
+--------------------
+
+These computers have a built in sound support based on the Jazz chipset.
+Look at the instructions for MV Jazz (above). It's also important to note
+that there is something wrong with the mouse port and sound at least on
+some TM models. Don't enable the "C&T 82C710 mouse port support" when
+configuring Linux. Having it enabled is likely to cause mysterious problems
+and kernel failures when sound is used.
+
+miroSOUND
+---------
+
+The miroSOUND PCM1-pro, PCM12 and PCM20 radio has been used
+successfully. These cards are based on the MAD16, OPL4, and CS4231A chips
+and everything said in the section about MAD16 cards applies here,
+too. The only major difference between the PCMxx and other MAD16 cards
+is that instead of the mixer in the CS4231 codec a separate mixer
+controlled by an on-board 80C32 microcontroller is used. Control of
+the mixer takes place via the ACI (miro's audio control interface)
+protocol that is implemented in a separate lowlevel driver. Make sure
+you compile this ACI driver together with the normal MAD16 support
+when you use a miroSOUND PCMxx card. The ACI mixer is controlled by
+/dev/mixer and the CS4231 mixer by /dev/mixer1 (depends on load
+time). Only in special cases you want to change something regularly on
+the CS4231 mixer.
+
+The miroSOUND PCM12 and PCM20 radio is capable of full duplex
+operation (simultaneous PCM replay and recording), which allows you to
+implement nice real-time signal processing audio effect software and
+network telephones. The ACI mixer has to be switched into the "solo"
+mode for duplex operation in order to avoid feedback caused by the
+mixer (input hears output signal). You can de-/activate this mode
+through toggleing the record button for the wave controller with an
+OSS-mixer.
+
+The PCM20 contains a radio tuner, which is also controlled by
+ACI. This radio tuner is supported by the ACI driver together with the
+miropcm20.o module. Also the 7-band equalizer is integrated
+(limited by the OSS-design). Developement has started and maybe
+finished for the RDS decoder on this card, too. You will be able to
+read RadioText, the Programme Service name, Programme TYpe and
+others. Even the v4l radio module benefits from it with a refined
+strength value. See aci.[ch] and miropcm20*.[ch] for more details.
+
+The following configuration parameters have worked fine for the PCM12
+in Markus Kuhn's system, many other configurations might work, too:
+CONFIG_MAD16_BASE=0x530, CONFIG_MAD16_IRQ=11, CONFIG_MAD16_DMA=3, 
+CONFIG_MAD16_DMA2=0, CONFIG_MAD16_MPU_BASE=0x330, CONFIG_MAD16_MPU_IRQ=10,
+DSP_BUFFSIZE=65536, SELECTED_SOUND_OPTIONS=0x00281000.
+
+Bas van der Linden is using his PCM1-pro with a configuration that
+differs in: CONFIG_MAD16_IRQ=7, CONFIG_MAD16_DMA=1, CONFIG_MAD16_MPU_IRQ=9
+
+Compaq Deskpro XL
+-----------------
+
+The builtin sound hardware of Compaq Deskpro XL is now supported. 
+You need to configure the driver with MSS and OPL3 supports enabled.
+In addition you need to manually edit linux/drivers/sound/local.h and
+to add a line containing "#define DESKPROXL" if you used 
+make menuconfig/xconfig.
+
+Others?
+-------
+
+Since there are so many different sound cards, it's likely that I have 
+forgotten to mention many of them. Please inform me if you know yet another
+card which works with Linux, please inform me (or is anybody else
+willing to maintain a database of supported cards (just like in XF86)?).
+
+Cards not supported yet
+=======================
+
+Please check the version of sound driver you are using before 
+complaining that your card is not supported. It's possible you are 
+using a driver version which was released months before your card was
+introduced.
+
+First of all, there is an easy way to make most sound cards work with Linux.
+Just use the DOS based driver to initialize the card to a known state, then use
+loadlin.exe to boot Linux. If Linux is configured to use the same I/O, IRQ and
+DMA numbers as DOS, the card could work.
+(ctrl-alt-del can be used in place of loadlin.exe but it doesn't work with
+new motherboards). This method works also with all/most PnP sound cards.
+
+Don't get fooled with SB compatibility. Most cards are compatible with
+SB but that may require a TSR which is not possible with Linux. If
+the card is compatible with MSS, it's a better choice. Some cards
+don't work in the SB and MSS modes at the same time.
+
+Then there are cards which are no longer manufactured and/or which
+are relatively rarely used (such as the 8 bit ProAudioSpectrum
+models). It's extremely unlikely that such cards ever get supported.
+Adding support for a new card requires much work and increases time
+required in maintaining the driver (some changes need to be done
+to all low level drivers and be tested too, maybe with multiple
+operating systems). For this reason I have made a decision to not support
+obsolete cards. It's possible that someone else makes a separately 
+distributed driver (diffs) for the card. 
+
+Writing a driver for a new card is not possible if there are no 
+programming information available about the card. If you don't
+find your new card from this file, look from the home page 
+(http://www.opensound.com/ossfree). Then please contact
+manufacturer of the card and ask if they have (or are willing to)
+released technical details of the card. Do this before contacting me. I
+can only answer 'no' if there are no programming information available.
+
+I have made decision to not accept code based on reverse engineering
+to the driver. There are three main reasons: First I don't want to break
+relationships to sound card manufacturers. The second reason is that
+maintaining and supporting a driver without any specs will be a pain.
+The third reason is that companies have freedom to refuse selling their
+products to other than Windows users.
+
+Some companies don't give low level technical information about their
+products to public or at least their require signing a NDA. It's not
+possible to implement a freeware driver for them. However it's possible
+that support for such cards become available in the commercial version
+of this driver (see http://www.4Front-tech.com/oss.html for more info).
+
+There are some common audio chipsets that are not supported yet. For example
+Sierra Aria and IBM Mwave. It's possible that these architectures
+get some support in future but I can't make any promises. Just look
+at the home page (http://www.opensound.com/ossfree/new_cards.html)
+for latest info.
+
+Information about unsupported sound cards and chipsets is welcome as well
+as free copies of sound cards, SDKs and operating systems.
+
+If you have any corrections and/or comments, please contact me.
+
+Hannu Savolainen
+hannu@opensound.com
+
+Personal home page:        http://www.compusonic.fi/~hannu
+home page of OSS/Free: http://www.opensound.com/ossfree
+
+home page of commercial OSS
+(Open Sound System) drivers: http://www.opensound.com/oss.html
diff --git a/Documentation/sound/oss/README.awe b/Documentation/sound/oss/README.awe
new file mode 100644
index 000000000000..80054cd8fcde
--- /dev/null
+++ b/Documentation/sound/oss/README.awe
@@ -0,0 +1,218 @@
+================================================================
+        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.
diff --git a/Documentation/sound/oss/README.modules b/Documentation/sound/oss/README.modules
new file mode 100644
index 000000000000..e691d74e1e5e
--- /dev/null
+++ b/Documentation/sound/oss/README.modules
@@ -0,0 +1,106 @@
+Building a modular sound driver
+================================
+
+  The following information is current as of linux-2.1.85. Check the other
+readme files, especially README.OSS, for information not specific to
+making sound modular.
+
+  First, configure your kernel. This is an idea of what you should be
+setting in the sound section:
+
+<M> Sound card support 
+
+<M> 100% Sound Blaster compatibles (SB16/32/64, ESS, Jazz16) support 
+
+  I have SoundBlaster. Select your card from the list.
+
+<M> Generic OPL2/OPL3 FM synthesizer support 
+<M> FM synthesizer (YM3812/OPL-3) support 
+
+  If you don't set these, you will probably find you can play .wav files
+but not .midi. As the help for them says, set them unless you know your
+card does not use one of these chips for FM support.
+
+  Once you are configured, make zlilo, modules, modules_install; reboot.
+Note that it is no longer necessary or possible to configure sound in the
+drivers/sound dir. Now one simply configures and makes one's kernel and
+modules in the usual way.
+
+ Then, add to your /etc/modprobe.conf something like:
+
+alias char-major-14-* sb
+install sb /sbin/modprobe -i sb && /sbin/modprobe adlib_card
+options sb io=0x220 irq=7 dma=1 dma16=5 mpu_io=0x330
+options adlib_card io=0x388     # FM synthesizer
+
+ Alternatively, if you have compiled in kernel level ISAPnP support:
+
+alias char-major-14 sb
+post-install sb /sbin/modprobe "-k" "adlib_card"
+options adlib_card io=0x388
+
+  The effect of this is that the sound driver and all necessary bits and
+pieces autoload on demand, assuming you use kerneld (a sound choice) and
+autoclean when not in use. Also, options for the device drivers are
+set. They will not work without them. Change as appropriate for your card.
+If you are not yet using the very cool kerneld, you will have to "modprobe
+-k sb" yourself to get things going. Eventually things may be fixed so
+that this kludgery is not necessary; for the time being, it seems to work
+well.
+
+  Replace 'sb' with the driver for your card, and give it the right
+options. To find the filename of the driver, look in
+/lib/modules/<kernel-version>/misc. Mine looks like:
+
+adlib_card.o # This is the generic OPLx driver
+opl3.o # The OPL3 driver
+sb.o # <<The SoundBlaster driver. Yours may differ.>>
+sound.o # The sound driver
+uart401.o # Used by sb, maybe other cards
+
+ Whichever card you have, try feeding it the options that would be the
+default if you were making the driver wired, not as modules. You can
+look at function referred to by module_init() for the card to see what
+args are expected.
+
+ Note that at present there is no way to configure the io, irq and other
+parameters for the modular drivers as one does for the wired drivers.. One
+needs to pass the modules the necessary parameters as arguments, either
+with /etc/modprobe.conf or with command-line args to modprobe, e.g.
+
+modprobe sb io=0x220 irq=7 dma=1 dma16=5 mpu_io=0x330
+modprobe adlib_card io=0x388
+
+ recommend using /etc/modprobe.conf.
+
+Persistent DMA Buffers:
+
+The sound modules normally allocate DMA buffers during open() and
+deallocate them during close(). Linux can often have problems allocating
+DMA buffers for ISA cards on machines with more than 16MB RAM. This is
+because ISA DMA buffers must exist below the 16MB boundary and it is quite
+possible that we can't find a large enough free block in this region after
+the machine has been running for any amount of time. The way to avoid this
+problem is to allocate the DMA buffers during module load and deallocate
+them when the module is unloaded. For this to be effective we need to load
+the sound modules right after the kernel boots, either manually or by an
+init script, and keep them around until we shut down. This is a little
+wasteful of RAM, but it guarantees that sound always works.
+
+To make the sound driver use persistent DMA buffers we need to pass the
+sound.o module a "dmabuf=1" command-line argument. This is normally done
+in /etc/modprobe.conf like so:
+
+options sound           dmabuf=1
+
+If you have 16MB or less RAM or a PCI sound card, this is wasteful and
+unnecessary. It is possible that machine with 16MB or less RAM will find
+this option useful, but if your machine is so memory-starved that it
+cannot find a 64K block free, you will be wasting even more RAM by keeping
+the sound modules loaded and the DMA buffers allocated when they are not
+needed. The proper solution is to upgrade your RAM. But you do also have
+this improper solution as well. Use it wisely.
+
+  I'm afraid I know nothing about anything but my setup, being more of a
+text-mode guy anyway. If you have options for other cards or other helpful
+hints, send them to me, Jim Bray, jb@as220.org, http://as220.org/jb.
diff --git a/Documentation/sound/oss/README.ymfsb b/Documentation/sound/oss/README.ymfsb
new file mode 100644
index 000000000000..af8a7d3a4e8e
--- /dev/null
+++ b/Documentation/sound/oss/README.ymfsb
@@ -0,0 +1,107 @@
+Legacy audio driver for YMF7xx PCI cards.
+
+
+FIRST OF ALL
+============
+
+  This code references YAMAHA's sample codes and data sheets.
+  I respect and thank for all people they made open the informations
+  about YMF7xx cards.
+
+  And this codes heavily based on Jeff Garzik <jgarzik@pobox.com>'s
+  old VIA 82Cxxx driver (via82cxxx.c). I also respect him.
+
+
+DISCLIMER
+=========
+
+ This driver is currently at early ALPHA stage. It may cause serious
+ damage to your computer when used.
+ PLEASE USE IT AT YOUR OWN RISK.
+
+
+ABOUT THIS DRIVER
+=================
+
+  This code enables you to use your YMF724[A-F], YMF740[A-C], YMF744, YMF754
+ cards. When enabled, your card acts as "SoundBlaster Pro" compatible card.
+ It can only play 22.05kHz / 8bit / Stereo samples, control external MIDI
+ port.
+ If you want to use your card as recent "16-bit" card, you should use
+ Alsa or OSS/Linux driver. Of course you can write native PCI driver for
+ your cards :)
+
+
+USAGE
+=====
+
+ # modprobe ymfsb (options)
+
+
+OPTIONS FOR MODULE
+==================
+
+  io           : SB base address     (0x220, 0x240, 0x260, 0x280)
+  synth_io     : OPL3 base address   (0x388, 0x398, 0x3a0, 0x3a8)
+  dma          : DMA number          (0,1,3)
+  master_volume: AC'97 PCM out Vol   (0-100)
+  spdif_out    : SPDIF-out flag      (0:disable 1:enable)
+
+ These options will change in future...
+
+
+FREQUENCY
+=========
+
+  When playing sounds via this driver, you will hear its pitch is slightly
+ lower than original sounds. Since this driver recognizes your card acts
+ with 21.739kHz sample rates rather than 22.050kHz (I think it must be
+ hardware restriction). So many players become tone deafness.
+ To prevent this, you should express some options to your sound player
+ that specify correct sample frequency. For example, to play your MP3 file
+ correctly with mpg123, specify the frequency like following:
+
+  %  mpg123 -r 21739 foo.mp3
+
+
+SPDIF OUT
+=========
+
+  With installing modules with option 'spdif_out=1', you can enjoy your
+ sounds from SPDIF-out of your card (if it had).
+ Its Fs is fixed to 48kHz (It never means the sample frequency become 
+ up to 48kHz. All sounds via SPDIF-out also 22kHz samples). So your
+ digital-in capable components has to be able to handle 48kHz Fs.
+
+
+COPYING
+=======
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of the GNU General Public License as published by
+ the Free Software Foundation; either version 2, or (at your option)
+ any later version.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+
+
+TODO
+====
+ * support for multiple cards
+   (set the different SB_IO,MPU_IO,OPL_IO for each cards)
+
+ * support for OPL (dmfm) : There will be no requirements... :-<
+
+
+AUTHOR
+======
+
+ Daisuke Nagano <breeze.nagano@nifty.ne.jp>
+
diff --git a/Documentation/sound/oss/SoundPro b/Documentation/sound/oss/SoundPro
new file mode 100644
index 000000000000..9d4db1f29d3c
--- /dev/null
+++ b/Documentation/sound/oss/SoundPro
@@ -0,0 +1,105 @@
+Documentation for the SoundPro CMI8330 extensions in the WSS driver (ad1848.o)
+------------------------------------------------------------------------------
+
+( Be sure to read Documentation/sound/oss/CMI8330 too )
+
+Ion Badulescu, ionut@cs.columbia.edu
+February 24, 1999
+
+(derived from the OPL3-SA2 documentation by Scott Murray)
+
+The SoundPro CMI8330 (ISA) is a chip usually found on some Taiwanese
+motherboards.  The official name in the documentation is CMI8330, SoundPro
+is the nickname and the big inscription on the chip itself.
+
+The chip emulates a WSS as well as a SB16, but it has certain differences
+in the mixer section which require separate support. It also emulates an
+MPU401 and an OPL3 synthesizer, so you probably want to enable support
+for these, too.
+
+The chip identifies itself as an AD1848, but its mixer is significantly
+more advanced than the original AD1848 one. If your system works with
+either WSS or SB16 and you are having problems with some mixer controls
+(no CD audio, no line-in, etc), you might want to give this driver a try.
+Detection should work, but it hasn't been widely tested, so it might still
+mis-identify the chip. You can still force soundpro=1 in the modprobe
+parameters for ad1848. Please let me know if it happens to you, so I can
+adjust the detection routine.
+
+The chip is capable of doing full-duplex, but since the driver sees it as an
+AD1848, it cannot take advantage of this.  Moreover, the full-duplex mode is
+not achievable through the WSS interface, b/c it needs a dma16 line which is
+assigned only to the SB16 subdevice (with isapnp). Windows documentation
+says the user must use WSS Playback and SB16 Recording for full-duplex, so
+it might be possible to do the same thing under Linux. You can try loading
+up both ad1848 and sb then use one for playback and the other for
+recording. I don't know if this works, b/c I haven't tested it. Anyway, if
+you try it, be very careful: the SB16 mixer *mostly* works, but certain
+settings can have unexpected effects. Use the WSS mixer for best results.
+
+There is also a PCI SoundPro chip. I have not seen this chip, so I have
+no idea if the driver will work with it. I suspect it won't.
+
+As with PnP cards, some configuration is required.  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.
+Since in this latter case you cannot use module parameters, you will
+have to enable support for the SoundPro at compile time.
+
+The IRQ and DMA values can be any that are considered acceptable for a
+WSS.  Assuming you've got isapnp all happy, then you should be able to
+do something like the following (which *must* match the isapnp/BIOS
+configuration):
+
+modprobe ad1848 io=0x530 irq=11 dma=0 soundpro=1
+-and maybe-
+modprobe sb io=0x220 irq=5 dma=1 dma16=5
+
+-then-
+modprobe mpu401 io=0x330 irq=9
+modprobe opl3 io=0x388
+
+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 module(s), 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 do not see the chipset version message, and none of the other
+messages present in the system log are helpful, try adding 'debug=1'
+to the ad1848 parameters, email me the syslog results and I'll do
+my best to help. 
+
+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 conf.modules file:
+
+# Sound
+post-install sound modprobe -k ad1848; modprobe -k mpu401; modprobe -k opl3
+options ad1848 io=0x530 irq=11 dma=0
+options sb io=0x220 irq=5 dma=1 dma16=5
+options mpu401 io=0x330 irq=9
+options opl3 io=0x388
+
+The above ensures that ad1848 will be loaded whenever the sound system
+is being used.
+
+Good luck.
+
+Ion
+
+NOT REALLY TESTED:
+- recording
+- recording device selection
+- full-duplex
+
+TODO:
+- implement mixer support for surround, loud, digital CD switches.
+- come up with a scheme which allows recording volumes for each subdevice.
+This is a major OSS API change.
diff --git a/Documentation/sound/oss/Soundblaster b/Documentation/sound/oss/Soundblaster
new file mode 100644
index 000000000000..b288d464ba8b
--- /dev/null
+++ b/Documentation/sound/oss/Soundblaster
@@ -0,0 +1,53 @@
+modprobe sound
+insmod uart401
+insmod sb ...
+
+This loads the driver for the Sound Blaster and assorted clones. Cards that
+are covered by other drivers should not be using this driver.
+
+The Sound Blaster module takes the following arguments
+
+io              I/O address of the Sound Blaster chip (0x220,0x240,0x260,0x280)
+irq             IRQ of the Sound Blaster chip (5,7,9,10)
+dma             8-bit DMA channel for the Sound Blaster (0,1,3)
+dma16           16-bit DMA channel for SB16 and equivalent cards (5,6,7)
+mpu_io          I/O for MPU chip if present (0x300,0x330)
+
+sm_games=1      Set if you have a Logitech soundman games
+acer=1          Set this to detect cards in some ACER notebooks
+mwave_bug=1     Set if you are trying to use this driver with mwave (see on)
+type            Use this to specify a specific card type
+
+The following arguments are taken if ISAPnP support is compiled in
+
+isapnp=0        Set this to disable ISAPnP detection (use io=0xXXX etc. above)
+multiple=0      Set to disable detection of multiple Soundblaster cards.
+                Consider it a bug if this option is needed, and send in a
+                report.
+pnplegacy=1     Set this to be able to use a PnP card(s) along with a single
+                non-PnP (legacy) card.  Above options for io, irq, etc. are
+                needed, and will apply only to the legacy card.
+reverse=1       Reverses the order of the search in the PnP table.
+uart401=1       Set to enable detection of mpu devices on some clones.
+isapnpjump=n    Jumps to slot n in the driver's PnP table. Use the source,
+                Luke.
+
+You may well want to load the opl3 driver for synth music on most SB and
+clone SB devices
+
+insmod opl3 io=0x388
+
+Using Mwave
+
+To make this driver work with Mwave you must set mwave_bug. You also need
+to warm boot from DOS/Windows with the required firmware loaded under this
+OS. IBM are being difficult about documenting how to load this firmware.
+
+Avance Logic ALS007
+
+This card is supported; see the separate file ALS007 for full details.
+
+Avance Logic ALS100
+
+This card is supported; setup should be as for a standard Sound Blaster 16.
+The driver will identify the audio device as a "Sound Blaster 16 (ALS-100)".
diff --git a/Documentation/sound/oss/Tropez+ b/Documentation/sound/oss/Tropez+
new file mode 100644
index 000000000000..b93a6b734fc0
--- /dev/null
+++ b/Documentation/sound/oss/Tropez+
@@ -0,0 +1,26 @@
+From: Paul Barton-Davis <pbd@op.net>
+
+Here is the configuration I use with a Tropez+ and my modular
+driver:
+
+  alias char-major-14 wavefront
+  alias synth0 wavefront
+  alias mixer0 cs4232
+  alias audio0 cs4232
+  pre-install wavefront modprobe "-k" "cs4232"
+  post-install wavefront modprobe "-k" "opl3"
+  options wavefront io=0x200 irq=9
+  options cs4232 synthirq=9 synthio=0x200 io=0x530 irq=5 dma=1 dma2=0
+  options opl3 io=0x388
+
+Things to note: 
+
+       the wavefront options "io" and "irq" ***MUST*** match the "synthio"
+       and "synthirq" cs4232 options.
+
+       you can do without the opl3 module if you don't
+       want to use the OPL/[34] synth on the soundcard
+
+       the opl3 io parameter is conventionally not adjustable.
+
+Please see drivers/sound/README.wavefront for more details.
diff --git a/Documentation/sound/oss/VIA-chipset b/Documentation/sound/oss/VIA-chipset
new file mode 100644
index 000000000000..37865234e54d
--- /dev/null
+++ b/Documentation/sound/oss/VIA-chipset
@@ -0,0 +1,43 @@
+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.
diff --git a/Documentation/sound/oss/VIBRA16 b/Documentation/sound/oss/VIBRA16
new file mode 100644
index 000000000000..68a5a46beb88
--- /dev/null
+++ b/Documentation/sound/oss/VIBRA16
@@ -0,0 +1,80 @@
+Sound Blaster 16X Vibra addendum
+--------------------------------
+by Marius Ilioaea <mariusi@protv.ro>
+   Stefan Laudat  <stefan@asit.ro>
+
+Sat Mar 6 23:55:27 EET 1999
+
+                        Hello again,
+        
+        Playing with a SB Vibra 16x soundcard we found it very difficult
+to setup because the kernel reported a lot of DMA errors and wouldn't
+simply play any sound.
+        A good starting point is that the vibra16x chip full-duplex facility
+is neither still exploited by the sb driver found in the linux kernel 
+(tried it with a 2.2.2-ac7), nor in the commercial OSS package (it reports
+it as half-duplex soundcard). Oh, I almost forgot, the RedHat sndconfig
+failed detecting it ;)
+        So, the big problem still remains, because the sb module wants a
+8-bit and a 16-bit dma, which we could not allocate for vibra... it supports
+only two 8-bit dma channels, the second one will be passed to the module
+as a 16 bit channel, the kernel will yield about that but everything will
+be okay, trust us. 
+        The only inconvenient you may find is that you will have
+some sound playing jitters if you have HDD dma support enabled - but this
+will happen with almost all soundcards...
+
+        A fully working isapnp.conf is just here:
+
+<snip here>
+
+(READPORT 0x0203)
+(ISOLATE PRESERVE)
+(IDENTIFY *)
+(VERBOSITY 2)
+(CONFLICT (IO FATAL)(IRQ FATAL)(DMA FATAL)(MEM FATAL)) # or WARNING
+# SB 16 and OPL3 devices
+(CONFIGURE CTL00f0/-1 (LD 0
+(INT 0 (IRQ 5 (MODE +E)))
+(DMA 0 (CHANNEL 1))
+(DMA 1 (CHANNEL 3))
+(IO 0 (SIZE 16) (BASE 0x0220))
+(IO 2 (SIZE 4) (BASE 0x0388))
+(NAME "CTL00f0/-1[0]{Audio               }")
+(ACT Y)
+))
+
+# Joystick device - only if you need it :-/
+
+(CONFIGURE CTL00f0/-1 (LD 1
+(IO 0 (SIZE 1) (BASE 0x0200))
+(NAME "CTL00f0/-1[1]{Game                }")
+(ACT Y)
+))
+(WAITFORKEY)
+
+<end of snipping>
+
+        So, after a good kernel modules compilation and a 'depmod -a kernel_ver'
+you may want to:
+
+modprobe sb io=0x220 irq=5 dma=1 dma16=3
+
+        Or, take the hard way:
+
+modprobe soundcore
+modprobe sound
+modprobe uart401
+modprobe sb io=0x220 irq=5 dma=1 dma16=3
+# do you need MIDI?
+modprobe opl3=0x388
+
+        Just in case, the kernel sound support should be:
+
+CONFIG_SOUND=m
+CONFIG_SOUND_OSS=m
+CONFIG_SOUND_SB=m
+        
+        Enjoy your new noisy Linux box! ;)
+        
+
diff --git a/Documentation/sound/oss/WaveArtist b/Documentation/sound/oss/WaveArtist
new file mode 100644
index 000000000000..f4f3407cd818
--- /dev/null
+++ b/Documentation/sound/oss/WaveArtist
@@ -0,0 +1,170 @@
+
+ (the following is from the armlinux CVS)
+
+ WaveArtist mixer and volume levels can be accessed via these commands:
+
+  nn30  read registers nn, where nn = 00 - 09 for mixer settings
+                                              0a - 13 for channel volumes
+  mm31  write the volume setting in pairs, where mm = (nn - 10) / 2
+  rr32  write the mixer settings in pairs, where rr = nn/2
+  xx33  reset all settings to default
+  0y34  select mono source, y=0 = left, y=1 = right
+
+                                           bits
+ nn  15  14 13 12 11   10    9     8     7    6     5     4     3     2     1     0
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 00 | 0 | 0  0  1  1 | left line mixer gain       | left aux1 mixer gain        |lmute|
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 01 | 0 | 0  1  0  1 | left aux2 mixer gain       | right 2 left mic gain       |mmute|
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 02 | 0 | 0  1  1  1 | left mic mixer gain        | left mic  | left mixer gain |dith |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 03 | 0 | 1  0  0  1 | left mixer input select                |lrfg | left ADC gain   |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 04 | 0 | 1  0  1  1 | right line mixer gain      | right aux1 mixer gain       |rmute|
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 05 | 0 | 1  1  0  1 | right aux2 mixer gain      | left 2 right mic gain       |test |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 06 | 0 | 1  1  1  1 | right mic mixer gain       | right mic |right mixer gain |rbyps|
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 07 | 1 | 0  0  0  1 | right mixer select                     |rrfg | right ADC gain  |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 08 | 1 | 0  0  1  1 | mono mixer gain            |right ADC mux sel|left ADC mux sel |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 09 | 1 | 0  1  0  1 |loopb|left linout|loop|ADCch|TxFch|OffCD|test |loopb|loopb|osamp|
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 0a | 0 | left PCM channel volume                                                     |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 0b | 0 | right PCM channel volume                                                    |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 0c | 0 | left FM channel volume                                                      |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 0d | 0 | right FM channel volume                                                     |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 0e | 0 | left wavetable channel volume                                               |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 0f | 0 | right wavetable channel volume                                              |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 10 | 0 | left PCM expansion channel volume                                           |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 11 | 0 | right PCM expansion channel volume                                          |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 12 | 0 | left FM expansion channel volume                                            |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+ 13 | 0 | right FM expansion channel volume                                           |
+----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
+
+ lmute: left mute
+ mmute: mono mute
+ dith: dithds
+ lrfg:
+ rmute: right mute
+ rbyps: right bypass
+ rrfg:
+ ADCch:
+ TxFch:
+ OffCD:
+ osamp:
+
+ And the following diagram is derived from the description in the CVS archive:
+
+ MIC L (mouthpiece)
+   +------+
+ -->PreAmp>-\
+   +--^---+ |
+      |     |
+    r2b4-5  |                                +--------+
+       /----*-------------------------------->5       |
+       |                                     |        |
+       | /----------------------------------->4       |
+       | |                                   |        |
+       | | /--------------------------------->3 1of5  |  +---+
+       | | |                                 |  mux   >-->AMP>--> ADC L
+       | | | /------------------------------->2       |  +-^-+
+       | | | |                               |        |    |
+ Line  | | | | +----+  +------+  +---+  /---->1       |  r3b3-0
+ ------------*->mute>--> Gain >-->   |  |    |        |
+  L    | | |   +----+  +------+  |   |  |  *->0       |
+       | | |                     |   |  |    +---^----+
+ Aux2  | | |   +----+  +------+  |   |  |        |
+ ----------*--->mute>--> Gain >--> M |  |     r8b0-2
+  L    | |     +----+  +------+  |   |  |
+       | |                       |   |  \------\
+ Aux1  | |     +----+  +------+  |   |         |
+ --------*----->mute>--> Gain >--> I |         |
+  L    |       +----+  +------+  |   |         |
+       |                         |   |         |
+       |       +----+  +------+  |   |  +---+  |
+       *------->mute>--> Gain >--> X >-->AMP>--*
+       |       +----+  +------+  |   |  +-^-+  |
+       |                         |   |    |    |
+       |       +----+  +------+  |   |  r2b1-3 |  
+       | /----->mute>--> Gain >--> E |         |
+       | |     +----+  +------+  |   |         |
+       | |                       |   |         |
+       | |     +----+  +------+  |   |         |
+       | | /--->mute>--> Gain >--> R |         |
+       | | |   +----+  +------+  |   |         |
+       | | |                     |   |         |   r9b8-9
+       | | |   +----+  +------+  |   |         |     |
+       | | | /->mute>--> Gain >-->   |         | +---v---+
+       | | | | +----+  +------+  +---+       /-*->0      |
+ DAC   | | | |                               |   |       |
+ ------------*----------------------------------->?      |  +----+
+  L    | | |                                 |   |  Mux  >-->mute>--> L output
+       | | |                                 | /->?      |  +--^-+
+       | | |                                 | | |       |     |
+       | | |                           /--------->?      |   r0b0
+       | | |                           |     | | +-------+
+       | | |                           |     | |
+ Mono  | | |                           |     | | +-------+
+ ----------*                           |     \--->       |  +----+
+       | | |                           |       | |  Mix  >-->mute>--> Mono output
+       | | |                           |       *->       |  +--^-+
+       | | |                           |       | +-------+     |
+       | | |                           |       |             r1b0
+ DAC   | | |                           |       | +-------+
+ ------------*-------------------------*--------->1      |  +----+
+  R    | | | |                                 | |  Mux  >-->mute>--> R output
+       | | | | +----+  +------+  +---+         *->0      |  +--^-+
+       | | | \->mute>--> Gain >-->   |         | +---^---+     |
+       | | |   +----+  +------+  |   |         |     |       r5b0
+       | | |                     |   |         |   r6b0
+       | | |   +----+  +------+  |   |         |
+       | | \--->mute>--> Gain >--> M |         |
+       | |     +----+  +------+  |   |         |
+       | |                       |   |         |
+       | |     +----+  +------+  |   |         |
+       | *----->mute>--> Gain >--> I |         |
+       | |     +----+  +------+  |   |         |
+       | |                       |   |         |
+       | |     +----+  +------+  |   |  +---+  |
+       \------->mute>--> Gain >--> X >-->AMP>--*
+         |     +----+  +------+  |   |  +-^-+  |
+      /--/                       |   |    |    |
+ Aux1 |        +----+  +------+  |   |  r6b1-3 |
+ -------*------>mute>--> Gain >--> E |         |
+  R   | |      +----+  +------+  |   |         |
+      | |                        |   |         |
+ Aux2 | |      +----+  +------+  |   |  /------/
+ ---------*---->mute>--> Gain >--> R |  |
+  R   | | |    +----+  +------+  |   |  |
+      | | |                      |   |  |    +--------+
+ Line | | |    +----+  +------+  |   |  |  *->0       |
+ -----------*-->mute>--> Gain >-->   |  |    |        |
+  R   | | | |  +----+  +------+  +---+  \---->1       |
+      | | | |                                |        |
+      | | | \-------------------------------->2       |  +---+
+      | | |                                  |  Mux   >-->AMP>--> ADC R
+      | | \---------------------------------->3       |  +-^-+
+      | |                                    |        |    |
+      | \------------------------------------>4       |  r7b3-0
+      |                                      |        |
+      \-----*-------------------------------->5       |
+            |                                +---^----+
+    r6b4-5  |                                    |
+      |     |                                  r8b3-5
+   +--v---+ |
+ -->PreAmp>-/
+   +------+
+ MIC R (electret mic)
diff --git a/Documentation/sound/oss/Wavefront b/Documentation/sound/oss/Wavefront
new file mode 100644
index 000000000000..16f57ea43052
--- /dev/null
+++ b/Documentation/sound/oss/Wavefront
@@ -0,0 +1,339 @@
+             An OSS/Free Driver for WaveFront soundcards
+               (Turtle Beach Maui, Tropez, Tropez Plus)
+
+                     Paul Barton-Davis, July 1998
+
+                          VERSION 0.2.5
+
+Driver Status
+-------------
+
+Requires: Kernel 2.1.106 or later (the driver is included with kernels
+2.1.109 and above)
+          
+As of 7/22/1998, this driver is currently in *BETA* state. This means
+that it compiles and runs, and that I use it on my system (Linux
+2.1.106) with some reasonably demanding applications and uses.  I
+believe the code is approaching an initial "finished" state that
+provides bug-free support for the Tropez Plus.
+
+Please note that to date, the driver has ONLY been tested on a Tropez
+Plus. I would very much like to hear (and help out) people with Tropez
+and Maui cards, since I think the driver can support those cards as
+well. 
+
+Finally, the driver has not been tested (or even compiled) as a static
+(non-modular) part of the kernel. Alan Cox's good work in modularizing
+OSS/Free for Linux makes this rather unnecessary.
+
+Some Questions
+--------------
+
+**********************************************************************
+0) What does this driver do that the maui driver did not ?
+**********************************************************************
+
+* can fully initialize a WaveFront card from cold boot - no DOS 
+          utilities needed
+* working patch/sample/program loading and unloading (the maui
+      driver didn't document how to make this work, and assumed
+      user-level preparation of the patch data for writing
+      to the board. ick.)
+* full user-level access to all WaveFront commands
+* for the Tropez Plus, (primitive) control of the YSS225 FX processor
+* Virtual MIDI mode supported - 2 MIDI devices accessible via the
+          WaveFront's MPU401/UART emulation. One
+          accesses the WaveFront synth, the other accesses the
+          external MIDI connector. Full MIDI read/write semantics
+          for both devices.
+* OSS-compliant /dev/sequencer interface for the WaveFront synth,
+          including native and GUS-format patch downloading.
+* semi-intelligent patch management (prototypical at this point)
+
+**********************************************************************
+1) What to do about MIDI interfaces ?
+**********************************************************************
+
+The Tropez Plus (and perhaps other WF cards) can in theory support up
+to 2 physical MIDI interfaces. One of these is connected to the
+ICS2115 chip (the WaveFront synth itself) and is controlled by
+MPU/UART-401 emulation code running as part of the WaveFront OS.  The
+other is controlled by the CS4232 chip present on the board. However,
+physical access to the CS4232 connector is difficult, and it is
+unlikely (though not impossible) that you will want to use it.
+
+An older version of this driver introduced an additional kernel config
+variable which controlled whether or not the CS4232 MIDI interface was
+configured. Because of Alan Cox's work on modularizing the sound
+drivers, and now backporting them to 2.0.34 kernels, there seems to be
+little reason to support "static" configuration variables, and so this
+has been abandoned in favor of *only* module parameters. Specifying
+"mpuio" and "mpuirq" for the cs4232 parameter will result in the
+CS4232 MIDI interface being configured; leaving them unspecified will
+leave it unconfigured (and thus unusable).
+
+BTW, I have heard from one Tropez+ user that the CS4232 interface is
+more reliable than the ICS2115 one. I have had no problems with the
+latter, and I don't have the right cable to test the former one
+out. Reports welcome.
+
+**********************************************************************
+2) Why does line XXX of the code look like this .... ?
+**********************************************************************
+
+Either because it's not finished yet, or because you're a better coder
+than I am, or because you don't understand some aspect of how the card
+or the code works. 
+
+I absolutely welcome comments, criticisms and suggestions about the
+design and implementation of the driver. 
+
+**********************************************************************
+3) What files are included ?
+**********************************************************************
+
+   drivers/sound/README.wavefront       -- this file
+
+   drivers/sound/wavefront.patch        -- patches for the 2.1.106 sound drivers
+                                           needed to make the rest of this work
+                                           DO NOT USE IF YOU'VE APPLIED THEM 
+                                           BEFORE, OR HAVE 2.1.109 OR ABOVE
+
+   drivers/sound/wavfront.c             -- the driver
+   drivers/sound/ys225.h                -- data declarations for FX config
+   drivers/sound/ys225.c                -- data definitions for FX config
+   drivers/sound/wf_midi.c              -- the "uart401" driver 
+                                              to support virtual MIDI mode.
+   include/wavefront.h                  -- the header file
+   Documentation/sound/oss/Tropez+          -- short docs on configuration
+
+**********************************************************************
+4) How do I compile/install/use it ?
+**********************************************************************
+
+PART ONE: install the source code into your sound driver directory
+
+  cd <top-of-your-2.1.106-code-base-e.g.-/usr/src/linux>
+  tar -zxvf <where-you-put/wavefront.tar.gz>
+
+PART TWO: apply the patches
+
+     DO THIS ONLY IF YOU HAVE A KERNEL VERSION BELOW 2.1.109
+     AND HAVE NOT ALREADY INSTALLED THE PATCH(ES).
+
+  cd drivers/sound
+  patch < wavefront.patch
+
+PART THREE: configure your kernel
+
+  cd <top of your kernel tree>
+  make xconfig (or whichever config option you use)
+
+         - choose YES for Sound Support       
+         - choose MODULE (M) for OSS Sound Modules
+         - choose MODULE(M) to YM3812/OPL3 support
+         - choose MODULE(M) for WaveFront support
+         - choose MODULE(M) for CS4232 support
+
+         - choose "N" for everything else (unless you have other
+              soundcards you want support for)
+
+
+   make boot
+   .
+   .
+   .
+   <whatever you normally do for a kernel install>
+   make modules
+   .
+   .
+   .
+   make modules_install
+
+Here's my autoconf.h SOUND section:
+
+/*
+ * Sound
+ */
+#define CONFIG_SOUND 1
+#undef  CONFIG_SOUND_OSS
+#define CONFIG_SOUND_OSS_MODULE 1
+#undef  CONFIG_SOUND_PAS
+#undef  CONFIG_SOUND_SB
+#undef  CONFIG_SOUND_ADLIB
+#undef  CONFIG_SOUND_GUS
+#undef  CONFIG_SOUND_MPU401
+#undef  CONFIG_SOUND_PSS
+#undef  CONFIG_SOUND_MSS
+#undef  CONFIG_SOUND_SSCAPE
+#undef  CONFIG_SOUND_TRIX
+#undef  CONFIG_SOUND_MAD16
+#undef  CONFIG_SOUND_WAVEFRONT
+#define CONFIG_SOUND_WAVEFRONT_MODULE 1
+#undef  CONFIG_SOUND_CS4232
+#define CONFIG_SOUND_CS4232_MODULE 1
+#undef  CONFIG_SOUND_MAUI
+#undef  CONFIG_SOUND_SGALAXY
+#undef  CONFIG_SOUND_OPL3SA1
+#undef  CONFIG_SOUND_SOFTOSS
+#undef  CONFIG_SOUND_YM3812
+#define CONFIG_SOUND_YM3812_MODULE 1
+#undef  CONFIG_SOUND_VMIDI
+#undef  CONFIG_SOUND_UART6850
+/*
+ * Additional low level sound drivers
+ */
+#undef  CONFIG_LOWLEVEL_SOUND
+
+************************************************************
+6) How do I configure my card ?
+************************************************************
+
+You need to edit /etc/modprobe.conf. Here's mine (edited to show the
+relevant details):
+
+  # Sound system
+  alias char-major-14-* wavefront
+  alias synth0 wavefront
+  alias mixer0 cs4232
+  alias audio0 cs4232
+  install wavefront /sbin/modprobe cs4232 && /sbin/modprobe -i wavefront && /sbin/modprobe opl3
+  options wavefront io=0x200 irq=9
+  options cs4232 synthirq=9 synthio=0x200 io=0x530 irq=5 dma=1 dma2=0
+  options opl3 io=0x388
+
+Things to note: 
+
+       the wavefront options "io" and "irq" ***MUST*** match the "synthio"
+       and "synthirq" cs4232 options.
+
+       you can do without the opl3 module if you don't
+       want to use the OPL/[34] FM synth on the soundcard
+
+       the opl3 io parameter is conventionally not adjustable.
+       In theory, any not-in-use IO port address would work, but
+       just use 0x388 and stick with the crowd.
+
+**********************************************************************
+7) What about firmware ?
+**********************************************************************
+
+Turtle Beach have not given me permission to distribute their firmware
+for the ICS2115. However, if you have a WaveFront card, then you
+almost certainly have the firmware, and if not, its freely available
+on their website, at:
+
+   http://www.tbeach.com/tbs/downloads/scardsdown.htm#tropezplus 
+
+The file is called WFOS2001.MOT (for the Tropez+).
+
+This driver, however, doesn't use the pure firmware as distributed,
+but instead relies on a somewhat processed form of it. You can
+generate this very easily. Following an idea from Andrew Veliath's
+Pinnacle driver, the following flex program will generate the
+processed version:
+
+---- cut here -------------------------
+%option main
+%%
+^S[28].*\r$ printf ("%c%.*s", yyleng-1,yyleng-1,yytext);
+<<EOF>> { fputc ('\0', stdout); return; }
+\n {} 
+.  {}
+---- cut here -------------------------
+
+To use it, put the above in file (say, ws.l) compile it like this:
+
+      shell> flex -ows.c ws.l
+      shell> cc -o ws ws.c
+      
+and then use it like this:
+
+    ws < my-copy-of-the-oswf.mot-file > /etc/sound/wavefront.os
+
+If you put it somewhere else, you'll always have to use the wf_ospath
+module parameter (see below) or alter the source code.
+
+**********************************************************************
+7) How do I get it working ?
+**********************************************************************
+
+Optionally, you can reboot with the "new" kernel (even though the only
+changes have really been made to a module).
+
+Then, as root do:
+
+     modprobe wavefront
+
+You should get something like this in /var/log/messages:
+
+    WaveFront: firmware 1.20 already loaded.
+
+or 
+
+    WaveFront: no response to firmware probe, assume raw.
+
+then:
+
+    WaveFront: waiting for memory configuration ...
+    WaveFront: hardware version 1.64
+    WaveFront: available DRAM 8191k
+    WaveFront: 332 samples used (266 real, 13 aliases, 53 multi), 180 empty
+    WaveFront: 128 programs slots in use
+    WaveFront: 256 patch slots filled, 142 in use
+
+The whole process takes about 16 seconds, the longest waits being
+after reporting the hardware version (during the firmware download),
+and after reporting program status (during patch status inquiry).  Its
+shorter (about 10 secs) if the firmware is already loaded (i.e. only
+warm reboots since the last firmware load).
+
+The "available DRAM" line will vary depending on how much added RAM
+your card has. Mine has 8MB.
+
+To check basically functionality, use play(1) or splay(1) to send a
+.WAV or other audio file through the audio portion. Then use playmidi
+to play a General MIDI file. Try the "-D 0" to hear the
+difference between sending MIDI to the WaveFront and using the OPL/3,
+which is the default (I think ...). If you have an external synth(s)
+hooked to the soundcard, you can use "-e" to route to the
+external synth(s) (in theory, -D 1 should work as well, but I think
+there is a bug in playmidi which prevents this from doing what it
+should). 
+
+**********************************************************************
+8) What are the module parameters ?
+**********************************************************************
+
+Its best to read wavefront.c for this, but here is a summary:
+
+integers: 
+          wf_raw  - if set, ignore apparent presence of firmware
+                    loaded onto the ICS2115, reset the whole
+                    board, and initialize it from scratch. (default = 0)
+
+          fx_raw  - if set, always initialize the YSS225 processor
+                    on the Tropez plus. (default = 1)
+
+          < The next 4 are basically for kernel hackers to allow
+            tweaking the driver for testing purposes. >             
+
+          wait_usecs        -  loop timer used when waiting for
+                               status conditions on the board. 
+                               The default is 150.
+
+          debug_default    - debugging flags. See sound/wavefront.h
+                             for WF_DEBUG_* values. Default is zero.
+                             Setting this allows you to debug the
+                             driver during module installation.
+strings:
+          ospath - path to get to the pre-processed OS firmware.
+                    (default: /etc/sound/wavefront.os)
+
+**********************************************************************
+9) Who should I contact if I have problems?
+**********************************************************************
+
+Just me: Paul Barton-Davis <pbd@op.net>
+
+
diff --git a/Documentation/sound/oss/btaudio b/Documentation/sound/oss/btaudio
new file mode 100644
index 000000000000..1a693e69d44b
--- /dev/null
+++ b/Documentation/sound/oss/btaudio
@@ -0,0 +1,92 @@
+
+Intro
+=====
+
+people start bugging me about this with questions, looks like I
+should write up some documentation for this beast.  That way I
+don't have to answer that much mails I hope.  Yes, I'm lazy...
+
+
+You might have noticed that the bt878 grabber cards have actually
+_two_ PCI functions:
+
+$ lspci
+[ ... ]
+00:0a.0 Multimedia video controller: Brooktree Corporation Bt878 (rev 02)
+00:0a.1 Multimedia controller: Brooktree Corporation Bt878 (rev 02)
+[ ... ]
+
+The first does video, it is backward compatible to the bt848.  The second
+does audio.  btaudio is a driver for the second function.  It's a sound
+driver which can be used for recording sound (and _only_ recording, no
+playback).  As most TV cards come with a short cable which can be plugged
+into your sound card's line-in you probably don't need this driver if all
+you want to do is just watching TV...
+
+
+Driver Status
+=============
+
+Still somewhat experimental.  The driver should work stable, i.e. it
+should'nt crash your box.  It might not work as expected, have bugs,
+not being fully OSS API compilant, ...
+
+Latest versions are available from http://bytesex.org/bttv/, the
+driver is in the bttv tarball.  Kernel patches might be available too,
+have a look at http://bytesex.org/bttv/listing.html.
+
+The chip knows two different modes.  btaudio registers two dsp
+devices, one for each mode.  They can not be used at the same time.
+
+
+Digital audio mode
+==================
+
+The chip gives you 16 bit stereo sound.  The sample rate depends on
+the external source which feeds the bt878 with digital sound via I2S
+interface.  There is a insmod option (rate) to tell the driver which
+sample rate the hardware uses (32000 is the default).
+
+One possible source for digital sound is the msp34xx audio processor
+chip which provides digital sound via I2S with 32 kHz sample rate.  My
+Hauppauge board works this way.
+
+The Osprey-200 reportly gives you digital sound with 44100 Hz sample
+rate.  It is also possible that you get no sound at all.
+
+
+analog mode (A/D)
+=================
+
+You can tell the driver to use this mode with the insmod option "analog=1".
+The chip has three analog inputs.  Consequently you'll get a mixer device
+to control these.
+
+The analog mode supports mono only.  Both 8 + 16 bit.  Both are _signed_
+int, which is uncommon for the 8 bit case.  Sample rate range is 119 kHz
+to 448 kHz.  Yes, the number of digits is correct.  The driver supports
+downsampling by powers of two, so you can ask for more usual sample rates
+like 44 kHz too.
+
+With my Hauppauge I get noisy sound on the second input (mapped to line2
+by the mixer device).  Others get a useable signal on line1.
+
+
+some examples
+=============
+
+* read audio data from btaudio (dsp2), send to es1730 (dsp,dsp1):
+  $ sox -w -r 32000 -t ossdsp /dev/dsp2 -t ossdsp /dev/dsp
+
+* read audio data from btaudio, send to esound daemon (which might be
+  running on another host):
+  $ sox -c 2 -w -r 32000 -t ossdsp /dev/dsp2 -t sw - | esdcat -r 32000
+  $ sox -c 1 -w -r 32000 -t ossdsp /dev/dsp2 -t sw - | esdcat -m -r 32000
+
+
+Have fun,
+
+  Gerd
+
+-- 
+Gerd Knorr <kraxel@bytesex.org>
diff --git a/Documentation/sound/oss/cs46xx b/Documentation/sound/oss/cs46xx
new file mode 100644
index 000000000000..88d6cf8b39f3
--- /dev/null
+++ b/Documentation/sound/oss/cs46xx
@@ -0,0 +1,138 @@
+
+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_PARM(defaultorder, "i");
+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_PARM(cs_debuglevel, "i");
+MODULE_PARM(cs_debugmask, "i");
+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_PARM(hercules_egpio_disable, "i");
+hercules_egpio_disable=N
+where N is a 0 (enable egpio), or a 1 (disable egpio support)
+
+MODULE_PARM(initdelay, "i");
+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_PARM(powerdown, "i");
+powerdown=N
+where N is 0 (disable any powerdown of the internal blocks) or 1 (enable powerdown)
+
+
+MODULE_PARM(external_amp, "i");
+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_PARM(thinkpad, "i");
+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.
+
diff --git a/Documentation/sound/oss/es1370 b/Documentation/sound/oss/es1370
new file mode 100644
index 000000000000..7b38b1a096a3
--- /dev/null
+++ b/Documentation/sound/oss/es1370
@@ -0,0 +1,70 @@
+/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
diff --git a/Documentation/sound/oss/es1371 b/Documentation/sound/oss/es1371
new file mode 100644
index 000000000000..c3151266771c
--- /dev/null
+++ b/Documentation/sound/oss/es1371
@@ -0,0 +1,64 @@
+/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.
+
+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
diff --git a/Documentation/sound/oss/mwave b/Documentation/sound/oss/mwave
new file mode 100644
index 000000000000..858334bb46b0
--- /dev/null
+++ b/Documentation/sound/oss/mwave
@@ -0,0 +1,185 @@
+        How to try to survive an IBM Mwave under Linux SB drivers
+
+
++       IBM have now released documentation of sorts and Torsten is busy
+        trying to make the Mwave work. This is not however a trivial task.
+
+----------------------------------------------------------------------------
+
+OK, first thing - the IRQ problem IS a problem, whether the test is bypassed or
+not.  It is NOT a Linux problem, but an MWAVE problem that is fixed with the
+latest MWAVE patches. So, in other words, don't bypass the test for MWAVES!
+
+I have Windows 95 on /dev/hda1, swap on /dev/hda2, and Red Hat 5 on /dev/hda3.
+
+The steps, then:
+
+    Boot to Linux. 
+    Mount Windows 95 file system (assume mount point = /dos95).
+    mkdir /dos95/linux
+    mkdir /dos95/linux/boot
+    mkdir /dos95/linux/boot/parms
+
+    Copy the kernel, any initrd image, and loadlin to /dos95/linux/boot/.
+
+    Reboot to Windows 95.
+
+    Edit C:/msdos.sys and add or change the following:
+
+        Logo=0
+        BootGUI=0
+
+    Note that msdos.sys is a text file but it needs to be made 'unhidden',
+    readable and writable before it can be edited.  This can be done with
+    DOS' "attrib" command.
+
+    Edit config.sys to have multiple config menus. I have one for windows 95 and
+    five for Linux, like this:
+------------
+[menu]
+menuitem=W95, Windows 95
+menuitem=LINTP, Linux - ThinkPad
+menuitem=LINTP3, Linux - ThinkPad Console
+menuitem=LINDOC, Linux - Docked
+menuitem=LINDOC3, Linux - Docked Console
+menuitem=LIN1, Linux - Single User Mode
+REM menudefault=W95,10
+
+[W95]
+
+[LINTP]
+
+[LINDOC]
+
+[LINTP3]
+
+[LINDOC3]
+
+[LIN1]
+
+[COMMON]
+FILES=30
+REM Please read README.TXT in C:\MWW subdirectory before changing the DOS= statement.
+DOS=HIGH,UMB
+DEVICE=C:\MWW\MANAGER\MWD50430.EXE
+SHELL=c:\command.com /e:2048
+-------------------
+
+The important things are the SHELL and DEVICE statements.
+
+    Then change autoexec.bat. Basically everything in there originally should be
+    done ONLY when Windows 95 is booted. Then you add new things specifically
+    for Linux.  Mine is as follows
+
+---------------
+@ECHO OFF
+if "%CONFIG%" == "W95" goto W95
+
+REM
+REM Linux stuff
+REM
+SET MWPATH=C:\MWW\DLL;C:\MWW\MWGAMES;C:\MWW\DSP
+SET BLASTER=A220 I5 D1
+SET MWROOT=C:\MWW
+SET LIBPATH=C:\MWW\DLL
+SET PATH=C:\WINDOWS;C:\MWW\DLL;
+CALL MWAVE START NOSHOW
+c:\linux\boot\loadlin.exe @c:\linux\boot\parms\%CONFIG%.par
+
+:W95
+REM
+REM Windows 95 stuff
+REM
+c:\toolkit\guard
+SET MSINPUT=C:\MSINPUT
+SET MWPATH=C:\MWW\DLL;C:\MWW\MWGAMES;C:\MWW\DSP
+REM The following is used by DOS games to recognize Sound Blaster hardware.
+REM If hardware settings are changed, please change this line as well.
+REM See the Mwave README file for instructions.
+SET BLASTER=A220 I5 D1
+SET MWROOT=C:\MWW
+SET LIBPATH=C:\MWW\DLL
+SET PATH=C:\WINDOWS;C:\WINDOWS\COMMAND;E:\ORAWIN95\BIN;f:\msdev\bin;e:\v30\bin.dbg;v:\devt\v30\bin;c:\JavaSDK\Bin;C:\MWW\DLL;
+SET INCLUDE=f:\MSDEV\INCLUDE;F:\MSDEV\MFC\INCLUDE
+SET LIB=F:\MSDEV\LIB;F:\MSDEV\MFC\LIB
+win
+
+------------------------
+
+Now build a file in c:\linux\boot\parms for each Linux config that you have.
+
+For example, my LINDOC3 config is for a docked Thinkpad at runlevel 3 with no
+initrd image, and has a parameter file named LINDOC3.PAR in c:\linux\boot\parms:
+
+-----------------------
+#   LOADLIN @param_file image=other_image root=/dev/other
+#
+#       Linux Console in docking station
+#
+c:\linux\boot\zImage.krn        # First value must be filename of Linux kernel.
+root=/dev/hda3                  # device which gets mounted as root FS
+ro                              # Other kernel arguments go here.
+apm=off
+doc=yes
+3
+-----------------------
+
+The doc=yes parameter is an environment variable used by my init scripts, not
+a kernel argument.
+
+However, the apm=off parameter IS a kernel argument!  APM, at least in my setup,
+causes the kernel to crash when loaded via loadlin (but NOT when loaded via
+LILO). The APM stuff COULD be forced out of the kernel via the kernel compile
+options. Instead, I got an unofficial patch to the APM drivers that allows them
+to be dynamically deactivated via kernel arguments. Whatever you chose to
+document, APM, it seems, MUST be off for setups like mine.
+
+Now make sure C:\MWW\MWCONFIG.REF looks like this:
+
+----------------------
+[NativeDOS]
+Default=SB1.5
+SBInputSource=CD
+SYNTH=FM
+QSound=OFF
+Reverb=OFF
+Chorus=OFF
+ReverbDepth=5
+ChorusDepth=5
+SBInputVolume=5
+SBMainVolume=10
+SBWaveVolume=10
+SBSynthVolume=10
+WaveTableVolume=10
+AudioPowerDriver=ON
+
+[FastCFG]
+Show=No
+HideOption=Off
+-----------------------------
+
+OR the Default= line COULD be
+
+Default=SBPRO
+
+Reboot to Windows 95 and choose Linux. When booted, use sndconfig to configure
+the sound modules and voilà - ThinkPad sound with Linux.
+
+Now the gotchas - you can either have CD sound OR Mixers but not both. That's a
+problem with the SB1.5 (CD sound) or SBPRO (Mixers) settings. No one knows why
+this is!
+
+For some reason MPEG3 files, when played through mpg123, sound like they
+are playing at 1/8th speed - not very useful!  If you have ANY insight
+on why this second thing might be happening, I would be grateful.
+
+===========================================================
+    _/      _/_/_/_/
+   _/_/  _/_/     _/
+  _/ _/_/ _/_/_/_/      Martin John Bartlett
+ _/  _/  _/     _/      (martin@nitram.demon.co.uk)
+_/      _/_/_/_/
+       _/
+_/    _/
+ _/_/
+===========================================================
diff --git a/Documentation/sound/oss/rme96xx b/Documentation/sound/oss/rme96xx
new file mode 100644
index 000000000000..87d7b7b65fa1
--- /dev/null
+++ b/Documentation/sound/oss/rme96xx
@@ -0,0 +1,767 @@
+Beta release of the rme96xx (driver for RME 96XX cards like the 
+"Hammerfall" and the "Hammerfall light") 
+
+Important: The driver module has to be installed on a freshly rebooted system, 
+otherwise the driver might not be able to acquire its buffers.
+
+features:
+
+ - OSS programming interface (i.e. runs with standard OSS soundsoftware) 
+ - OSS/Multichannel interface (OSS multichannel is done by just aquiring
+   more than 2 channels). The driver does not use more than one device 
+   ( yet .. this feature may be implemented later ) 
+ - more than one RME card supported
+
+The driver uses a specific multichannel interface, which I will document
+when the driver gets stable. (take a look at the defines in rme96xx.h,
+which adds blocked multichannel formats i.e instead of 
+lrlrlrlr --> llllrrrr  etc.
+
+Use the "rmectrl" programm to look at the status of the card .. 
+or use xrmectrl, a GUI interface for the ctrl program.
+
+What you can do with the rmectrl program is to set the stereo device for
+OSS emulation (e.g. if you use SPDIF out).
+
+You do:
+
+./ctrl offset 24 24
+
+which makes the stereo device use channels 25 and 26.
+
+Guenter Geiger <geiger@epy.co.at>
+
+copy the first part of the attached source code into rmectrl.c
+and the  second part into xrmectrl (or get the program from
+http://gige.xdv.org/pages/soft/pages/rme)
+
+to compile: gcc -o rmectrl rmectrl.c
+------------------------------ snip ------------------------------------
+
+#include <stdio.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <sys/ioctl.h>
+#include <fcntl.h>
+#include <linux/soundcard.h>
+#include <math.h>
+#include <unistd.h>
+#include <stdlib.h>
+#include "rme96xx.h"
+
+/*
+  remctrl.c
+  (C) 2000 Guenter Geiger <geiger@debian.org>
+  HP20020201 - Heiko Purnhagen <purnhage@tnt.uni-hannover.de>
+*/
+
+/* # define DEVICE_NAME "/dev/mixer" */
+# define DEVICE_NAME "/dev/mixer1"
+
+
+void usage(void)
+{
+     fprintf(stderr,"usage: rmectrl [/dev/mixer<n>] [command [options]]\n\n");
+     fprintf(stderr,"where command is one of:\n");
+     fprintf(stderr,"  help                       show this help\n");
+     fprintf(stderr,"  status                     show status bits\n");
+     fprintf(stderr,"  control                    show control bits\n");
+     fprintf(stderr,"  mix                        show mixer/offset status\n");
+     fprintf(stderr,"  master <n>                 set sync master\n");
+     fprintf(stderr,"  pro <n>                    set spdif out pro\n");
+     fprintf(stderr,"  emphasis <n>               set spdif out emphasis\n");
+     fprintf(stderr,"  dolby <n>                  set spdif out no audio\n");
+     fprintf(stderr,"  optout <n>                 set spdif out optical\n");
+     fprintf(stderr,"  wordclock <n>              set sync wordclock\n");
+     fprintf(stderr,"  spdifin <n>                set spdif in (0=optical,1=coax,2=intern)\n");
+     fprintf(stderr,"  syncref <n>                set sync source (0=ADAT1,1=ADAT2,2=ADAT3,3=SPDIF)\n");
+     fprintf(stderr,"  adat1cd <n>                set ADAT1 on internal CD\n");
+     fprintf(stderr,"  offset <devnr> <in> <out>  set dev (0..3) offset (0..25)\n");
+     exit(-1);
+}
+
+
+int main(int argc, char* argv[])
+{
+     int cards;
+     int ret;
+     int i;
+     double ft;
+     int fd, fdwr;
+     int param,orig;
+     rme_status_t stat;
+     rme_ctrl_t ctrl;
+     char *device;
+     int argidx;
+
+     if (argc < 2)
+          usage();
+
+     if (*argv[1]=='/') {
+          device = argv[1];
+          argidx = 2;
+     }
+     else {
+          device = DEVICE_NAME;
+          argidx = 1;
+     }
+
+     fprintf(stdout,"mixer device %s\n",device);
+     if ((fd = open(device,O_RDONLY)) < 0) {
+          fprintf(stdout,"opening device failed\n");
+          exit(-1);
+     }
+
+     if ((fdwr = open(device,O_WRONLY)) < 0) {
+          fprintf(stdout,"opening device failed\n");
+          exit(-1);
+     }
+
+     if (argc < argidx+1)
+          usage();
+
+     if (!strcmp(argv[argidx],"help"))
+        usage();
+     if (!strcmp(argv[argidx],"-h"))
+        usage();
+     if (!strcmp(argv[argidx],"--help"))
+        usage();
+
+     if (!strcmp(argv[argidx],"status")) {
+          ioctl(fd,SOUND_MIXER_PRIVATE2,&stat);
+          fprintf(stdout,"stat.irq %d\n",stat.irq);
+          fprintf(stdout,"stat.lockmask %d\n",stat.lockmask);
+          fprintf(stdout,"stat.sr48 %d\n",stat.sr48);
+          fprintf(stdout,"stat.wclock %d\n",stat.wclock);
+          fprintf(stdout,"stat.bufpoint %d\n",stat.bufpoint);
+          fprintf(stdout,"stat.syncmask %d\n",stat.syncmask);
+          fprintf(stdout,"stat.doublespeed %d\n",stat.doublespeed);
+          fprintf(stdout,"stat.tc_busy %d\n",stat.tc_busy);
+          fprintf(stdout,"stat.tc_out %d\n",stat.tc_out);
+          fprintf(stdout,"stat.crystalrate %d (0=64k 3=96k 4=88.2k 5=48k 6=44.1k 7=32k)\n",stat.crystalrate);
+          fprintf(stdout,"stat.spdif_error %d\n",stat.spdif_error);
+          fprintf(stdout,"stat.bufid %d\n",stat.bufid);
+          fprintf(stdout,"stat.tc_valid %d\n",stat.tc_valid);
+          exit (0);
+     }
+
+     if (!strcmp(argv[argidx],"control")) {
+          ioctl(fd,SOUND_MIXER_PRIVATE3,&ctrl);
+          fprintf(stdout,"ctrl.start %d\n",ctrl.start);
+          fprintf(stdout,"ctrl.latency %d (0=64 .. 7=8192)\n",ctrl.latency);
+          fprintf(stdout,"ctrl.master %d\n",ctrl.master);
+          fprintf(stdout,"ctrl.ie %d\n",ctrl.ie);
+          fprintf(stdout,"ctrl.sr48 %d\n",ctrl.sr48);
+          fprintf(stdout,"ctrl.spare %d\n",ctrl.spare);
+          fprintf(stdout,"ctrl.doublespeed %d\n",ctrl.doublespeed);
+          fprintf(stdout,"ctrl.pro %d\n",ctrl.pro);
+          fprintf(stdout,"ctrl.emphasis %d\n",ctrl.emphasis);
+          fprintf(stdout,"ctrl.dolby %d\n",ctrl.dolby);
+          fprintf(stdout,"ctrl.opt_out %d\n",ctrl.opt_out);
+          fprintf(stdout,"ctrl.wordclock %d\n",ctrl.wordclock);
+          fprintf(stdout,"ctrl.spdif_in %d (0=optical,1=coax,2=intern)\n",ctrl.spdif_in);
+          fprintf(stdout,"ctrl.sync_ref %d (0=ADAT1,1=ADAT2,2=ADAT3,3=SPDIF)\n",ctrl.sync_ref);
+          fprintf(stdout,"ctrl.spdif_reset %d\n",ctrl.spdif_reset);
+          fprintf(stdout,"ctrl.spdif_select %d\n",ctrl.spdif_select);
+          fprintf(stdout,"ctrl.spdif_clock %d\n",ctrl.spdif_clock);
+          fprintf(stdout,"ctrl.spdif_write %d\n",ctrl.spdif_write);
+          fprintf(stdout,"ctrl.adat1_cd %d\n",ctrl.adat1_cd);
+          exit (0);
+     }
+
+     if (!strcmp(argv[argidx],"mix")) {
+          rme_mixer mix;
+          int i;
+
+          for (i=0; i<4; i++) {
+               mix.devnr = i;
+               ioctl(fd,SOUND_MIXER_PRIVATE1,&mix);
+               if (mix.devnr == i) {
+                    fprintf(stdout,"devnr %d\n",mix.devnr);
+                    fprintf(stdout,"mix.i_offset %2d (0-25)\n",mix.i_offset);
+                    fprintf(stdout,"mix.o_offset %2d (0-25)\n",mix.o_offset);
+               }
+          }
+          exit (0);
+     }
+
+/* the control flags */
+
+     if (argc < argidx+2)
+          usage();
+
+     if (!strcmp(argv[argidx],"master")) {
+          int val = atoi(argv[argidx+1]);
+          ioctl(fd,SOUND_MIXER_PRIVATE3,&ctrl);
+          printf("master = %d\n",val);
+          ctrl.master = val;
+          ioctl(fdwr,SOUND_MIXER_PRIVATE3,&ctrl);
+          exit (0);
+     }
+
+     if (!strcmp(argv[argidx],"pro")) {
+          int val = atoi(argv[argidx+1]);
+          ioctl(fd,SOUND_MIXER_PRIVATE3,&ctrl);
+          printf("pro = %d\n",val);
+          ctrl.pro = val;
+          ioctl(fdwr,SOUND_MIXER_PRIVATE3,&ctrl);
+          exit (0);
+     }
+
+     if (!strcmp(argv[argidx],"emphasis")) {
+          int val = atoi(argv[argidx+1]);
+          ioctl(fd,SOUND_MIXER_PRIVATE3,&ctrl);
+          printf("emphasis = %d\n",val);
+          ctrl.emphasis = val;
+          ioctl(fdwr,SOUND_MIXER_PRIVATE3,&ctrl);
+          exit (0);
+     }
+
+     if (!strcmp(argv[argidx],"dolby")) {
+          int val = atoi(argv[argidx+1]);
+          ioctl(fd,SOUND_MIXER_PRIVATE3,&ctrl);
+          printf("dolby = %d\n",val);
+          ctrl.dolby = val;
+          ioctl(fdwr,SOUND_MIXER_PRIVATE3,&ctrl);
+          exit (0);
+     }
+
+     if (!strcmp(argv[argidx],"optout")) {
+          int val = atoi(argv[argidx+1]);
+          ioctl(fd,SOUND_MIXER_PRIVATE3,&ctrl);
+          printf("optout = %d\n",val);
+          ctrl.opt_out = val;
+          ioctl(fdwr,SOUND_MIXER_PRIVATE3,&ctrl);
+          exit (0);
+     }
+
+     if (!strcmp(argv[argidx],"wordclock")) {
+          int val = atoi(argv[argidx+1]);
+          ioctl(fd,SOUND_MIXER_PRIVATE3,&ctrl);
+          printf("wordclock = %d\n",val);
+          ctrl.wordclock = val;
+          ioctl(fdwr,SOUND_MIXER_PRIVATE3,&ctrl);
+          exit (0);
+     }
+
+     if (!strcmp(argv[argidx],"spdifin")) {
+          int val = atoi(argv[argidx+1]);
+          ioctl(fd,SOUND_MIXER_PRIVATE3,&ctrl);
+          printf("spdifin = %d\n",val);
+          ctrl.spdif_in = val;
+          ioctl(fdwr,SOUND_MIXER_PRIVATE3,&ctrl);
+          exit (0);
+     }
+
+     if (!strcmp(argv[argidx],"syncref")) {
+          int val = atoi(argv[argidx+1]);
+          ioctl(fd,SOUND_MIXER_PRIVATE3,&ctrl);
+          printf("syncref = %d\n",val);
+          ctrl.sync_ref = val;
+          ioctl(fdwr,SOUND_MIXER_PRIVATE3,&ctrl);
+          exit (0);
+     }
+
+     if (!strcmp(argv[argidx],"adat1cd")) {
+          int val = atoi(argv[argidx+1]);
+          ioctl(fd,SOUND_MIXER_PRIVATE3,&ctrl);
+          printf("adat1cd = %d\n",val);
+          ctrl.adat1_cd = val;
+          ioctl(fdwr,SOUND_MIXER_PRIVATE3,&ctrl);
+          exit (0);
+     }
+
+/* setting offset */
+
+     if (argc < argidx+4)
+          usage();
+
+     if (!strcmp(argv[argidx],"offset")) {
+          rme_mixer mix;
+
+          mix.devnr = atoi(argv[argidx+1]);
+
+          mix.i_offset = atoi(argv[argidx+2]);
+          mix.o_offset = atoi(argv[argidx+3]);
+          ioctl(fdwr,SOUND_MIXER_PRIVATE1,&mix);
+          fprintf(stdout,"devnr %d\n",mix.devnr);
+          fprintf(stdout,"mix.i_offset to %d\n",mix.i_offset);
+          fprintf(stdout,"mix.o_offset to %d\n",mix.o_offset);
+          exit (0);
+     }
+
+     usage();
+     exit (0); /* to avoid warning */
+}
+
+
+---------------------------- <snip> --------------------------------
+#!/usr/bin/wish
+
+# xrmectrl
+# (C) 2000 Guenter Geiger <geiger@debian.org>
+# HP20020201 - Heiko Purnhagen <purnhage@tnt.uni-hannover.de>
+
+#set defaults "-relief ridged"
+set CTRLPROG "./rmectrl"
+if {$argc} {
+    set CTRLPROG "$CTRLPROG $argv"
+}
+puts "CTRLPROG $CTRLPROG"
+
+frame .butts
+button .butts.exit -text "Exit" -command "exit" -relief ridge
+#button .butts.state -text "State" -command "get_all"
+
+pack .butts.exit -side left
+pack .butts -side bottom
+
+
+#
+# STATUS
+#
+
+frame .status
+
+# Sampling Rate
+
+frame .status.sr
+label .status.sr.text -text "Sampling Rate" -justify left
+radiobutton .status.sr.441 -selectcolor red -text "44.1 kHz" -width 10 -anchor nw -variable srate -value 44100 -font times
+radiobutton .status.sr.480 -selectcolor red -text "48 kHz" -width 10 -anchor nw -variable srate -value 48000 -font times
+radiobutton .status.sr.882 -selectcolor red -text "88.2 kHz" -width 10 -anchor nw -variable srate -value 88200 -font times
+radiobutton .status.sr.960 -selectcolor red -text "96 kHz" -width 10 -anchor nw  -variable srate -value 96000 -font times
+
+pack .status.sr.text .status.sr.441 .status.sr.480 .status.sr.882 .status.sr.960 -side top -padx 3
+
+# Lock
+
+frame .status.lock
+label .status.lock.text -text "Lock" -justify left
+checkbutton .status.lock.adat1 -selectcolor red -text "ADAT1" -anchor nw -width 10 -variable adatlock1 -font times
+checkbutton .status.lock.adat2 -selectcolor red -text "ADAT2" -anchor nw -width 10 -variable adatlock2 -font times
+checkbutton .status.lock.adat3 -selectcolor red -text "ADAT3" -anchor nw -width 10 -variable adatlock3 -font times
+
+pack .status.lock.text .status.lock.adat1 .status.lock.adat2 .status.lock.adat3 -side top -padx 3 
+
+# Sync
+
+frame .status.sync
+label .status.sync.text -text "Sync" -justify left
+checkbutton .status.sync.adat1 -selectcolor red -text "ADAT1" -anchor nw -width 10 -variable adatsync1 -font times
+checkbutton .status.sync.adat2 -selectcolor red -text "ADAT2" -anchor nw -width 10 -variable adatsync2 -font times
+checkbutton .status.sync.adat3 -selectcolor red -text "ADAT3" -anchor nw -width 10 -variable adatsync3 -font times
+
+pack .status.sync.text .status.sync.adat1 .status.sync.adat2 .status.sync.adat3 -side top -padx 3 
+
+# Timecode
+
+frame .status.tc
+label .status.tc.text -text "Timecode" -justify left
+checkbutton .status.tc.busy -selectcolor red -text "busy" -anchor nw -width 10 -variable tcbusy -font times
+checkbutton .status.tc.out -selectcolor red -text "out" -anchor nw -width 10 -variable tcout -font times
+checkbutton .status.tc.valid -selectcolor red -text "valid" -anchor nw -width 10 -variable tcvalid -font times
+
+pack .status.tc.text .status.tc.busy .status.tc.out .status.tc.valid -side top -padx 3 
+
+# SPDIF In
+
+frame .status.spdif
+label .status.spdif.text -text "SPDIF In" -justify left
+label .status.spdif.sr -text "--.- kHz" -anchor n -width 10 -font times
+checkbutton .status.spdif.error -selectcolor red -text "Input Lock" -anchor nw -width 10 -variable spdiferr -font times
+
+pack .status.spdif.text .status.spdif.sr .status.spdif.error -side top -padx 3 
+
+pack .status.sr .status.lock .status.sync .status.tc .status.spdif -side left -fill x -anchor n -expand 1
+
+
+#
+# CONTROL 
+#
+
+proc setprof {} {
+    global CTRLPROG
+    global spprof
+    exec $CTRLPROG pro $spprof
+}
+
+proc setemph {} {
+    global CTRLPROG
+    global spemph
+    exec $CTRLPROG emphasis $spemph
+}
+
+proc setnoaud {} {
+    global CTRLPROG
+    global spnoaud
+    exec $CTRLPROG dolby $spnoaud
+}
+
+proc setoptical {} {
+    global CTRLPROG
+    global spoptical
+    exec $CTRLPROG optout $spoptical
+}
+
+proc setspdifin {} {
+    global CTRLPROG
+    global spdifin
+    exec $CTRLPROG spdifin [expr $spdifin - 1]
+}
+
+proc setsyncsource {} {
+    global CTRLPROG
+    global syncsource
+    exec $CTRLPROG syncref [expr $syncsource -1]
+}
+
+
+proc setmaster {} {
+    global CTRLPROG
+    global master
+    exec $CTRLPROG master $master
+}
+
+proc setwordclock {} {
+    global CTRLPROG
+    global wordclock
+    exec $CTRLPROG wordclock $wordclock
+}
+
+proc setadat1cd {} {
+    global CTRLPROG
+    global adat1cd
+    exec $CTRLPROG adat1cd $adat1cd
+}
+
+
+frame .control
+
+# SPDIF In & SPDIF Out
+
+
+frame .control.spdif
+
+frame .control.spdif.in
+label .control.spdif.in.text -text "SPDIF In" -justify left
+radiobutton .control.spdif.in.input1 -text "Optical" -anchor nw -width 13 -variable spdifin -value 1 -command setspdifin -selectcolor blue -font times
+radiobutton .control.spdif.in.input2 -text "Coaxial" -anchor nw -width 13 -variable spdifin -value 2 -command setspdifin -selectcolor blue -font times
+radiobutton .control.spdif.in.input3 -text "Intern " -anchor nw -width 13 -variable spdifin -command setspdifin -value 3 -selectcolor blue -font times
+
+checkbutton .control.spdif.in.adat1cd -text "ADAT1 Intern" -anchor nw -width 13 -variable adat1cd -command setadat1cd -selectcolor blue -font times
+
+pack .control.spdif.in.text .control.spdif.in.input1 .control.spdif.in.input2 .control.spdif.in.input3 .control.spdif.in.adat1cd
+
+label .control.spdif.space
+
+frame .control.spdif.out
+label .control.spdif.out.text -text "SPDIF Out" -justify left
+checkbutton .control.spdif.out.pro -text "Professional" -anchor nw -width 13 -variable spprof -command setprof -selectcolor blue -font times
+checkbutton .control.spdif.out.emphasis -text "Emphasis" -anchor nw -width 13 -variable spemph -command setemph -selectcolor blue -font times
+checkbutton .control.spdif.out.dolby -text "NoAudio" -anchor nw -width 13 -variable spnoaud -command setnoaud -selectcolor blue -font times
+checkbutton .control.spdif.out.optout -text "Optical Out" -anchor nw -width 13 -variable spoptical -command setoptical -selectcolor blue -font times
+
+pack .control.spdif.out.optout .control.spdif.out.dolby .control.spdif.out.emphasis .control.spdif.out.pro .control.spdif.out.text -side bottom
+
+pack .control.spdif.in .control.spdif.space .control.spdif.out -side top -fill y -padx 3 -expand 1
+
+# Sync Mode & Sync Source
+
+frame .control.sync
+frame .control.sync.mode
+label .control.sync.mode.text -text "Sync Mode" -justify left
+checkbutton .control.sync.mode.master -text "Master" -anchor nw -width 13 -variable master -command setmaster -selectcolor blue -font times
+checkbutton .control.sync.mode.wc -text "Wordclock" -anchor nw -width 13 -variable wordclock -command setwordclock -selectcolor blue -font times
+
+pack .control.sync.mode.text .control.sync.mode.master .control.sync.mode.wc
+
+label .control.sync.space
+
+frame .control.sync.src
+label .control.sync.src.text -text "Sync Source" -justify left
+radiobutton .control.sync.src.input1 -text "ADAT1" -anchor nw -width 13 -variable syncsource -value 1 -command setsyncsource -selectcolor blue -font times
+radiobutton .control.sync.src.input2 -text "ADAT2" -anchor nw -width 13 -variable syncsource -value 2 -command setsyncsource -selectcolor blue -font times
+radiobutton .control.sync.src.input3 -text "ADAT3" -anchor nw -width 13 -variable syncsource -command setsyncsource -value 3 -selectcolor blue -font times
+radiobutton .control.sync.src.input4 -text "SPDIF" -anchor nw -width 13 -variable syncsource -command setsyncsource -value 4 -selectcolor blue -font times
+
+pack .control.sync.src.input4 .control.sync.src.input3 .control.sync.src.input2 .control.sync.src.input1 .control.sync.src.text -side bottom
+
+pack .control.sync.mode .control.sync.space .control.sync.src -side top -fill y -padx 3 -expand 1
+
+label .control.space -text "" -width 10
+
+# Buffer Size
+
+frame .control.buf
+label .control.buf.text -text "Buffer Size (Latency)" -justify left
+radiobutton .control.buf.b1 -selectcolor red -text "64 (1.5 ms)" -width 13 -anchor nw -variable ssrate -value 1 -font times
+radiobutton .control.buf.b2 -selectcolor red -text "128 (3 ms)" -width 13 -anchor nw -variable ssrate -value 2 -font times
+radiobutton .control.buf.b3 -selectcolor red -text "256 (6 ms)" -width 13 -anchor nw -variable ssrate -value 3 -font times
+radiobutton .control.buf.b4 -selectcolor red -text "512 (12 ms)" -width 13 -anchor nw -variable ssrate -value 4 -font times
+radiobutton .control.buf.b5 -selectcolor red -text "1024 (23 ms)" -width 13 -anchor nw -variable ssrate -value 5 -font times
+radiobutton .control.buf.b6 -selectcolor red -text "2048 (46 ms)" -width 13 -anchor nw -variable ssrate -value 6 -font times
+radiobutton .control.buf.b7 -selectcolor red -text "4096 (93 ms)" -width 13 -anchor nw -variable ssrate -value 7 -font times
+radiobutton .control.buf.b8 -selectcolor red -text "8192 (186 ms)" -width 13 -anchor nw -variable ssrate -value 8 -font times
+
+pack .control.buf.text .control.buf.b1 .control.buf.b2 .control.buf.b3 .control.buf.b4 .control.buf.b5 .control.buf.b6 .control.buf.b7 .control.buf.b8 -side top -padx 3 
+
+# Offset
+
+frame .control.offset
+
+frame .control.offset.in
+label .control.offset.in.text -text "Offset In" -justify left
+label .control.offset.in.off0 -text "dev\#0: -" -anchor nw -width 10 -font times
+label .control.offset.in.off1 -text "dev\#1: -" -anchor nw -width 10 -font times
+label .control.offset.in.off2 -text "dev\#2: -" -anchor nw -width 10 -font times
+label .control.offset.in.off3 -text "dev\#3: -" -anchor nw -width 10 -font times
+
+pack .control.offset.in.text .control.offset.in.off0 .control.offset.in.off1 .control.offset.in.off2 .control.offset.in.off3
+
+label .control.offset.space
+
+frame .control.offset.out
+label .control.offset.out.text -text "Offset Out" -justify left
+label .control.offset.out.off0 -text "dev\#0: -" -anchor nw -width 10 -font times
+label .control.offset.out.off1 -text "dev\#1: -" -anchor nw -width 10 -font times
+label .control.offset.out.off2 -text "dev\#2: -" -anchor nw -width 10 -font times
+label .control.offset.out.off3 -text "dev\#3: -" -anchor nw -width 10 -font times
+
+pack .control.offset.out.off3 .control.offset.out.off2 .control.offset.out.off1 .control.offset.out.off0 .control.offset.out.text -side bottom
+
+pack .control.offset.in .control.offset.space .control.offset.out -side top -fill y -padx 3 -expand 1
+
+
+pack .control.spdif .control.sync .control.space .control.buf .control.offset -side left -fill both -anchor n -expand 1
+
+
+label .statustext -text Status -justify center -relief ridge
+label .controltext -text Control -justify center -relief ridge
+
+label .statusspace
+label .controlspace
+
+pack .statustext .status .statusspace .controltext .control .controlspace -side top -anchor nw -fill both -expand 1
+
+
+proc get_bit {output sstr} {
+    set idx1 [string last [concat $sstr 1] $output]
+    set idx1 [expr $idx1 != -1]
+    return $idx1
+}
+
+proc get_val {output sstr} {
+    set val [string wordend $output [string last $sstr $output]] 
+    set val [string range $output $val [expr $val+1]]
+    return $val
+}
+
+proc get_val2 {output sstr} {
+    set val [string wordend $output [string first $sstr $output]] 
+    set val [string range $output $val [expr $val+2]]
+    return $val
+}
+
+proc get_control {} {
+    global spprof
+    global spemph
+    global spnoaud
+    global spoptical
+    global spdifin
+    global ssrate
+    global master
+    global wordclock
+    global syncsource
+    global CTRLPROG
+
+    set f [open "| $CTRLPROG control" r+]
+    set ooo [read $f 1000]
+    close $f
+#    puts $ooo
+
+    set spprof [ get_bit $ooo "pro"]
+    set spemph [ get_bit $ooo "emphasis"]
+    set spnoaud [ get_bit $ooo "dolby"]
+    set spoptical [ get_bit $ooo "opt_out"]
+    set spdifin [ expr [ get_val $ooo "spdif_in"] + 1]
+    set ssrate [ expr [ get_val $ooo "latency"] + 1]
+    set master [ expr [ get_val $ooo "master"]]
+    set wordclock [ expr [ get_val $ooo "wordclock"]]
+    set syncsource [ expr [ get_val $ooo "sync_ref"] + 1]
+}
+
+proc get_status {} {
+    global srate
+    global ctrlcom
+
+    global adatlock1
+    global adatlock2
+    global adatlock3
+
+    global adatsync1
+    global adatsync2
+    global adatsync3
+
+    global tcbusy
+    global tcout
+    global tcvalid
+
+    global spdiferr
+    global crystal
+    global .status.spdif.text
+    global CTRLPROG
+
+
+    set f [open "| $CTRLPROG status" r+]
+    set ooo [read $f 1000]
+    close $f
+#    puts $ooo
+
+# samplerate
+
+    set idx1 [string last "sr48 1" $ooo]
+    set idx2 [string last "doublespeed 1" $ooo]
+    if {$idx1 >= 0} {
+        set fact1 48000
+    } else {
+        set fact1 44100
+    }
+
+    if {$idx2 >= 0} {
+        set fact2 2
+    } else {
+        set fact2 1
+    }
+    set srate [expr $fact1 * $fact2]
+#   ADAT lock
+
+    set val [get_val $ooo lockmask]
+    set adatlock1 0
+    set adatlock2 0
+    set adatlock3 0
+    if {[expr $val & 1]} {
+       set adatlock3 1
+    }
+    if {[expr $val & 2]} {
+       set adatlock2 1
+    }
+    if {[expr $val & 4]} {
+       set adatlock1 1
+    }
+
+#  ADAT sync
+    set val [get_val $ooo syncmask]
+    set adatsync1 0
+    set adatsync2 0
+    set adatsync3 0
+
+    if {[expr $val & 1]} {
+       set adatsync3 1
+    }
+    if {[expr $val & 2]} {
+       set adatsync2 1
+    }
+    if {[expr $val & 4]} {
+       set adatsync1 1
+    }
+
+# TC busy
+
+    set tcbusy [get_bit $ooo "busy"]
+    set tcout [get_bit $ooo "out"]
+    set tcvalid [get_bit $ooo "valid"]
+    set spdiferr [expr [get_bit $ooo "spdif_error"] == 0]
+
+#  000=64kHz, 100=88.2kHz, 011=96kHz
+#  111=32kHz, 110=44.1kHz, 101=48kHz
+
+    set val [get_val $ooo crystalrate]
+
+    set crystal "--.- kHz"
+    if {$val == 0} {
+        set crystal "64 kHz"
+    }
+    if {$val == 4} {
+        set crystal "88.2 kHz"
+    }
+    if {$val == 3} {
+        set crystal "96 kHz"
+    }
+    if {$val == 7} {
+        set crystal "32 kHz"
+    }
+    if {$val == 6} {
+        set crystal "44.1 kHz"
+    }
+    if {$val == 5} {
+        set crystal "48 kHz"
+    }
+    .status.spdif.sr configure -text $crystal
+}
+
+proc get_offset {} {
+    global inoffset
+    global outoffset
+    global CTRLPROG
+
+    set f [open "| $CTRLPROG mix" r+]
+    set ooo [read $f 1000]
+    close $f
+#    puts $ooo
+
+    if { [string match "*devnr*" $ooo] } {
+        set ooo [string range $ooo [string wordend $ooo [string first devnr $ooo]] end]
+        set val [get_val2 $ooo i_offset]
+        .control.offset.in.off0 configure -text "dev\#0: $val"
+        set val [get_val2 $ooo o_offset]
+        .control.offset.out.off0 configure -text "dev\#0: $val"
+    } else {
+        .control.offset.in.off0 configure -text "dev\#0: -"
+        .control.offset.out.off0 configure -text "dev\#0: -"
+    }
+    if { [string match "*devnr*" $ooo] } {
+        set ooo [string range $ooo [string wordend $ooo [string first devnr $ooo]] end]
+        set val [get_val2 $ooo i_offset]
+        .control.offset.in.off1 configure -text "dev\#1: $val"
+        set val [get_val2 $ooo o_offset]
+        .control.offset.out.off1 configure -text "dev\#1: $val"
+    } else {
+        .control.offset.in.off1 configure -text "dev\#1: -"
+        .control.offset.out.off1 configure -text "dev\#1: -"
+    }
+    if { [string match "*devnr*" $ooo] } {
+        set ooo [string range $ooo [string wordend $ooo [string first devnr $ooo]] end]
+        set val [get_val2 $ooo i_offset]
+        .control.offset.in.off2 configure -text "dev\#2: $val"
+        set val [get_val2 $ooo o_offset]
+        .control.offset.out.off2 configure -text "dev\#2: $val"
+    } else {
+        .control.offset.in.off2 configure -text "dev\#2: -"
+        .control.offset.out.off2 configure -text "dev\#2: -"
+    }
+    if { [string match "*devnr*" $ooo] } {
+        set ooo [string range $ooo [string wordend $ooo [string first devnr $ooo]] end]
+        set val [get_val2 $ooo i_offset]
+        .control.offset.in.off3 configure -text "dev\#3: $val"
+        set val [get_val2 $ooo o_offset]
+        .control.offset.out.off3 configure -text "dev\#3: $val"
+    } else {
+        .control.offset.in.off3 configure -text "dev\#3: -"
+        .control.offset.out.off3 configure -text "dev\#3: -"
+    }
+}
+
+
+proc get_all {} {
+get_status
+get_control
+get_offset
+}
+
+# main
+while {1} {
+  after 200
+  get_all
+  update
+}
diff --git a/Documentation/sound/oss/solo1 b/Documentation/sound/oss/solo1
new file mode 100644
index 000000000000..6f53d407d027
--- /dev/null
+++ b/Documentation/sound/oss/solo1
@@ -0,0 +1,70 @@
+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 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
diff --git a/Documentation/sound/oss/sonicvibes b/Documentation/sound/oss/sonicvibes
new file mode 100644
index 000000000000..84dee2e0b37d
--- /dev/null
+++ b/Documentation/sound/oss/sonicvibes
@@ -0,0 +1,81 @@
+/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
diff --git a/Documentation/sound/oss/ultrasound b/Documentation/sound/oss/ultrasound
new file mode 100644
index 000000000000..32cd50478b36
--- /dev/null
+++ b/Documentation/sound/oss/ultrasound
@@ -0,0 +1,30 @@
+modprobe sound
+insmod ad1848
+insmod gus io=* irq=* dma=* ...
+
+This loads the driver for the Gravis Ultrasound family of sound cards.
+
+The gus module takes the following arguments
+
+io              I/O address of the Ultrasound card (eg. io=0x220)
+irq             IRQ of the Sound Blaster card 
+dma             DMA channel for the Sound Blaster
+dma16           2nd DMA channel, only needed for full duplex operation
+type            1 for PnP card
+gus16           1 for using 16 bit sampling daughter board
+no_wave_dma     Set to disable DMA usage for wavetable (see note)
+db16            ???
+
+
+no_wave_dma option
+
+This option defaults to a value of 0, which allows the Ultrasound wavetable
+DSP to use DMA for for playback and downloading samples. This is the same
+as the old behaviour. If set to 1, no DMA is needed for downloading samples,
+and allows owners of a GUS MAX to make use of simultaneous digital audio
+(/dev/dsp), MIDI, and wavetable playback.
+
+
+If you have problems in recording with GUS MAX, you could try to use
+just one 8 bit DMA channel. Recording will not work with one DMA
+channel if it's a 16 bit one.
diff --git a/Documentation/sound/oss/vwsnd b/Documentation/sound/oss/vwsnd
new file mode 100644
index 000000000000..a6ea0a1df9e4
--- /dev/null
+++ b/Documentation/sound/oss/vwsnd
@@ -0,0 +1,293 @@
+vwsnd - Sound driver for the Silicon Graphics 320 and 540 Visual
+Workstations' onboard audio.
+
+Copyright 1999 Silicon Graphics, Inc.  All rights reserved.
+
+
+At the time of this writing, March 1999, there are two models of
+Visual Workstation, the 320 and the 540.  This document only describes
+those models.  Future Visual Workstation models may have different
+sound capabilities, and this driver will probably not work on those
+boxes.
+
+The Visual Workstation has an Analog Devices AD1843 "SoundComm" audio
+codec chip.  The AD1843 is accessed through the Cobalt I/O ASIC, also
+known as Lithium.  This driver programs both both chips.
+
+==============================================================================
+QUICK CONFIGURATION
+
+        # insmod soundcore
+        # insmod vwsnd
+
+==============================================================================
+I/O CONNECTIONS
+
+On the Visual Workstation, only three of the AD1843 inputs are hooked
+up.  The analog line in jacks are connected to the AD1843's AUX1
+input.  The CD audio lines are connected to the AD1843's AUX2 input.
+The microphone jack is connected to the AD1843's MIC input.  The mic
+jack is mono, but the signal is delivered to both the left and right
+MIC inputs.  You can record in stereo from the mic input, but you will
+get the same signal on both channels (within the limits of A/D
+accuracy).  Full scale on the Line input is +/- 2.0 V.  Full scale on
+the MIC input is 20 dB less, or +/- 0.2 V.
+
+The AD1843's LOUT1 outputs are connected to the Line Out jacks.  The
+AD1843's HPOUT outputs are connected to the speaker/headphone jack.
+LOUT2 is not connected.  Line out's maximum level is +/- 2.0 V peak to
+peak.  The speaker/headphone out's maximum is +/- 4.0 V peak to peak.
+
+The AD1843's PCM input channel and one of its output channels (DAC1)
+are connected to Lithium.  The other output channel (DAC2) is not
+connected.
+
+==============================================================================
+CAPABILITIES
+
+The AD1843 has PCM input and output (Pulse Code Modulation, also known
+as wavetable).  PCM input and output can be mono or stereo in any of
+four formats.  The formats are 16 bit signed and 8 bit unsigned,
+u-Law, and A-Law format.  Any sample rate from 4 KHz to 49 KHz is
+available, in 1 Hz increments.
+
+The AD1843 includes an analog mixer that can mix all three input
+signals (line, mic and CD) into the analog outputs.  The mixer has a
+separate gain control and mute switch for each input.
+
+There are two outputs, line out and speaker/headphone out.  They
+always produce the same signal, and the speaker always has 3 dB more
+gain than the line out.  The speaker/headphone output can be muted,
+but this driver does not export that function.
+
+The hardware can sync audio to the video clock, but this driver does
+not have a way to specify syncing to video.
+
+==============================================================================
+PROGRAMMING
+
+This section explains the API supported by the driver.  Also see the
+Open Sound Programming Guide at http://www.opensound.com/pguide/ .
+This section assumes familiarity with that document.
+
+The driver has two interfaces, an I/O interface and a mixer interface.
+There is no MIDI or sequencer capability.
+
+==============================================================================
+PROGRAMMING PCM I/O
+
+The I/O interface is usually accessed as /dev/audio or /dev/dsp.
+Using the standard Open Sound System (OSS) ioctl calls, the sample
+rate, number of channels, and sample format may be set within the
+limitations described above.  The driver supports triggering.  It also
+supports getting the input and output pointers with one-sample
+accuracy.
+
+The SNDCTL_DSP_GETCAP ioctl returns these capabilities.
+
+        DSP_CAP_DUPLEX - driver supports full duplex.
+
+        DSP_CAP_TRIGGER - driver supports triggering.
+
+        DSP_CAP_REALTIME - values returned by SNDCTL_DSP_GETIPTR
+        and SNDCTL_DSP_GETOPTR are accurate to a few samples.
+
+Memory mapping (mmap) is not implemented.
+
+The driver permits subdivided fragment sizes from 64 to 4096 bytes.
+The number of fragments can be anything from 3 fragments to however
+many fragments fit into 124 kilobytes.  It is up to the user to
+determine how few/small fragments can be used without introducing
+glitches with a given workload.  Linux is not realtime, so we can't
+promise anything.  (sigh...)
+
+When this driver is switched into or out of mu-Law or A-Law mode on
+output, it may produce an audible click.  This is unavoidable.  To
+prevent clicking, use signed 16-bit mode instead, and convert from
+mu-Law or A-Law format in software.
+
+==============================================================================
+PROGRAMMING THE MIXER INTERFACE
+
+The mixer interface is usually accessed as /dev/mixer.  It is accessed
+through ioctls.  The mixer allows the application to control gain or
+mute several audio signal paths, and also allows selection of the
+recording source.
+
+Each of the constants described here can be read using the
+MIXER_READ(SOUND_MIXER_xxx) ioctl.  Those that are not read-only can
+also be written using the MIXER_WRITE(SOUND_MIXER_xxx) ioctl.  In most
+cases, <sys/soundcard.h> defines constants SOUND_MIXER_READ_xxx and
+SOUND_MIXER_WRITE_xxx which work just as well.
+
+SOUND_MIXER_CAPS        Read-only
+
+This is a mask of optional driver capabilities that are implemented.
+This driver's only capability is SOUND_CAP_EXCL_INPUT, which means
+that only one recording source can be active at a time.
+
+SOUND_MIXER_DEVMASK     Read-only
+
+This is a mask of the sound channels.  This driver's channels are PCM,
+LINE, MIC, CD, and RECLEV.
+
+SOUND_MIXER_STEREODEVS  Read-only
+
+This is a mask of which sound channels are capable of stereo.  All
+channels are capable of stereo.  (But see caveat on MIC input in I/O
+CONNECTIONS section above).
+
+SOUND_MIXER_OUTMASK     Read-only
+
+This is a mask of channels that route inputs through to outputs.
+Those are LINE, MIC, and CD.
+
+SOUND_MIXER_RECMASK     Read-only
+
+This is a mask of channels that can be recording sources.  Those are
+PCM, LINE, MIC, CD.
+
+SOUND_MIXER_PCM         Default: 0x5757 (0 dB)
+
+This is the gain control for PCM output.  The left and right channel
+gain are controlled independently.  This gain control has 64 levels,
+which range from -82.5 dB to +12.0 dB in 1.5 dB steps.  Those 64
+levels are mapped onto 100 levels at the ioctl, see below.
+
+SOUND_MIXER_LINE        Default: 0x4a4a (0 dB)
+
+This is the gain control for mixing the Line In source into the
+outputs.  The left and right channel gain are controlled
+independently.  This gain control has 32 levels, which range from
+-34.5 dB to +12.0 dB in 1.5 dB steps.  Those 32 levels are mapped onto
+100 levels at the ioctl, see below.
+
+SOUND_MIXER_MIC         Default: 0x4a4a (0 dB)
+
+This is the gain control for mixing the MIC source into the outputs.
+The left and right channel gain are controlled independently.  This
+gain control has 32 levels, which range from -34.5 dB to +12.0 dB in
+1.5 dB steps.  Those 32 levels are mapped onto 100 levels at the
+ioctl, see below.
+
+SOUND_MIXER_CD          Default: 0x4a4a (0 dB)
+
+This is the gain control for mixing the CD audio source into the
+outputs.  The left and right channel gain are controlled
+independently.  This gain control has 32 levels, which range from
+-34.5 dB to +12.0 dB in 1.5 dB steps.  Those 32 levels are mapped onto
+100 levels at the ioctl, see below.
+
+SOUND_MIXER_RECLEV       Default: 0 (0 dB)
+
+This is the gain control for PCM input (RECording LEVel).  The left
+and right channel gain are controlled independently.  This gain
+control has 16 levels, which range from 0 dB to +22.5 dB in 1.5 dB
+steps.  Those 16 levels are mapped onto 100 levels at the ioctl, see
+below.
+
+SOUND_MIXER_RECSRC       Default: SOUND_MASK_LINE
+
+This is a mask of currently selected PCM input sources (RECording
+SouRCes).  Because the AD1843 can only have a single recording source
+at a time, only one bit at a time can be set in this mask.  The
+allowable values are SOUND_MASK_PCM, SOUND_MASK_LINE, SOUND_MASK_MIC,
+or SOUND_MASK_CD.  Selecting SOUND_MASK_PCM sets up internal
+resampling which is useful for loopback testing and for hardware
+sample rate conversion.  But software sample rate conversion is
+probably faster, so I don't know how useful that is.
+
+SOUND_MIXER_OUTSRC      DEFAULT: SOUND_MASK_LINE|SOUND_MASK_MIC|SOUND_MASK_CD
+
+This is a mask of sources that are currently passed through to the
+outputs.  Those sources whose bits are not set are muted.
+
+==============================================================================
+GAIN CONTROL
+
+There are five gain controls listed above.  Each has 16, 32, or 64
+steps.  Each control has 1.5 dB of gain per step.  Each control is
+stereo.
+
+The OSS defines the argument to a channel gain ioctl as having two
+components, left and right, each of which ranges from 0 to 100.  The
+two components are packed into the same word, with the left side gain
+in the least significant byte, and the right side gain in the second
+least significant byte.  In C, we would say this.
+
+        #include <assert.h>
+
+        ...
+
+                assert(leftgain >= 0 && leftgain <= 100);
+                assert(rightgain >= 0 && rightgain <= 100);
+                arg = leftgain | rightgain << 8;
+
+So each OSS gain control has 101 steps.  But the hardware has 16, 32,
+or 64 steps.  The hardware steps are spread across the 101 OSS steps
+nearly evenly.  The conversion formulas are like this, given N equals
+16, 32, or 64.
+
+        int round = N/2 - 1;
+        OSS_gain_steps = (hw_gain_steps * 100 + round) / (N - 1);
+        hw_gain_steps = (OSS_gain_steps * (N - 1) + round) / 100;
+
+Here is a snippet of C code that will return the left and right gain
+of any channel in dB.  Pass it one of the predefined gain_desc_t
+structures to access any of the five channels' gains.
+
+        typedef struct gain_desc {
+                float min_gain;
+                float gain_step;
+                int nbits;
+                int chan;
+        } gain_desc_t;
+
+        const gain_desc_t gain_pcm    = { -82.5, 1.5, 6, SOUND_MIXER_PCM    };
+        const gain_desc_t gain_line   = { -34.5, 1.5, 5, SOUND_MIXER_LINE   };
+        const gain_desc_t gain_mic    = { -34.5, 1.5, 5, SOUND_MIXER_MIC    };
+        const gain_desc_t gain_cd     = { -34.5, 1.5, 5, SOUND_MIXER_CD     };
+        const gain_desc_t gain_reclev = {   0.0, 1.5, 4, SOUND_MIXER_RECLEV };
+
+        int get_gain_dB(int fd, const gain_desc_t *gp,
+                        float *left, float *right)
+        {
+                int word;
+                int lg, rg;
+                int mask = (1 << gp->nbits) - 1;
+
+                if (ioctl(fd, MIXER_READ(gp->chan), &word) != 0)
+                        return -1;      /* fail */
+                lg = word & 0xFF;
+                rg = word >> 8 & 0xFF;
+                lg = (lg * mask + mask / 2) / 100;
+                rg = (rg * mask + mask / 2) / 100;
+                *left = gp->min_gain + gp->gain_step * lg;
+                *right = gp->min_gain + gp->gain_step * rg;
+                return 0;
+        }       
+
+And here is the corresponding routine to set a channel's gain in dB.
+
+        int set_gain_dB(int fd, const gain_desc_t *gp, float left, float right)
+        {
+                float max_gain =
+                        gp->min_gain + (1 << gp->nbits) * gp->gain_step;
+                float round = gp->gain_step / 2;
+                int mask = (1 << gp->nbits) - 1;
+                int word;
+                int lg, rg;
+
+                if (left < gp->min_gain || right < gp->min_gain)
+                        return EINVAL;
+                lg = (left - gp->min_gain + round) / gp->gain_step;
+                rg = (right - gp->min_gain + round) / gp->gain_step;
+                if (lg >= (1 << gp->nbits) || rg >= (1 << gp->nbits))
+                        return EINVAL;
+                lg = (100 * lg + mask / 2) / mask;
+                rg = (100 * rg + mask / 2) / mask;
+                word = lg | rg << 8;
+
+                return ioctl(fd, MIXER_WRITE(gp->chan), &word);
+        }
+