Merge branch 'build':

Release v1.0.23
    9a3491c73400920fe6379fd685d0bb8c01c48311
diff --git a/.gitignore b/.gitignore
new file mode 100644
index 0000000..a8f8b6f
--- /dev/null
+++ b/.gitignore
@@ -0,0 +1,6 @@
+scripts/*.o
+scripts/docproc
+*~
+.*~
+*.orig
+*.rej
diff --git a/Documentation/DocBook/alsa-driver-api.tmpl b/Documentation/DocBook/alsa-driver-api.tmpl
new file mode 100644
index 0000000..0230a96
--- /dev/null
+++ b/Documentation/DocBook/alsa-driver-api.tmpl
@@ -0,0 +1,109 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!DOCTYPE book PUBLIC "-//OASIS//DTD DocBook XML V4.1.2//EN"
+	"http://www.oasis-open.org/docbook/xml/4.1.2/docbookx.dtd" []>
+
+<!-- ****************************************************** -->
+<!-- Header  -->
+<!-- ****************************************************** -->
+<book id="ALSA-Driver-API">
+  <bookinfo>
+    <title>The ALSA Driver API</title>
+
+    <legalnotice>
+    <para>
+    This document is free; 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. 
+    </para>
+
+    <para>
+    This document is distributed in the hope that it will be useful,
+    but <emphasis>WITHOUT ANY WARRANTY</emphasis>; without even the
+    implied warranty of <emphasis>MERCHANTABILITY or FITNESS FOR A
+    PARTICULAR PURPOSE</emphasis>. See the GNU General Public License
+    for more details.
+    </para>
+
+    <para>
+    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., 59 Temple Place, Suite 330, Boston,
+    MA 02111-1307 USA
+    </para>
+    </legalnotice>
+
+  </bookinfo>
+
+<toc></toc>
+
+  <chapter><title>Management of Cards and Devices</title>
+     <sect1><title>Card Management</title>
+!Esound/core/init.c
+     </sect1>
+     <sect1><title>Device Components</title>
+!Esound/core/device.c
+     </sect1>
+     <sect1><title>Module requests and Device File Entries</title>
+!Esound/core/sound.c
+     </sect1>
+     <sect1><title>Memory Management Helpers</title>
+!Esound/core/memory.c
+!Esound/core/memalloc.c
+     </sect1>
+  </chapter>
+  <chapter><title>PCM API</title>
+     <sect1><title>PCM Core</title>
+!Esound/core/pcm.c
+!Esound/core/pcm_lib.c
+!Esound/core/pcm_native.c
+     </sect1>
+     <sect1><title>PCM Format Helpers</title>
+!Esound/core/pcm_misc.c
+     </sect1>
+     <sect1><title>PCM Memory Management</title>
+!Esound/core/pcm_memory.c
+     </sect1>
+  </chapter>
+  <chapter><title>Control/Mixer API</title>
+     <sect1><title>General Control Interface</title>
+!Esound/core/control.c
+     </sect1>
+     <sect1><title>AC97 Codec API</title>
+!Esound/pci/ac97/ac97_codec.c
+!Esound/pci/ac97/ac97_pcm.c
+     </sect1>
+     <sect1><title>Virtual Master Control API</title>
+!Esound/core/vmaster.c
+!Iinclude/sound/control.h
+     </sect1>
+  </chapter>
+  <chapter><title>MIDI API</title>
+     <sect1><title>Raw MIDI API</title>
+!Esound/core/rawmidi.c
+     </sect1>
+     <sect1><title>MPU401-UART API</title>
+!Esound/drivers/mpu401/mpu401_uart.c
+     </sect1>
+  </chapter>
+  <chapter><title>Proc Info API</title>
+     <sect1><title>Proc Info Interface</title>
+!Esound/core/info.c
+     </sect1>
+  </chapter>
+  <chapter><title>Miscellaneous Functions</title>
+     <sect1><title>Hardware-Dependent Devices API</title>
+!Esound/core/hwdep.c
+     </sect1>
+     <sect1><title>Jack Abstraction Layer API</title>
+!Esound/core/jack.c
+     </sect1>
+     <sect1><title>ISA DMA Helpers</title>
+!Esound/core/isadma.c
+     </sect1>
+     <sect1><title>Other Helper Macros</title>
+!Iinclude/sound/core.h
+     </sect1>
+  </chapter>
+
+</book>
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
new file mode 100644
index 0000000..bfcbbf8
--- /dev/null
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -0,0 +1,2243 @@
+
+		Advanced Linux Sound Architecture - Driver
+		==========================================
+			    Configuration guide
+
+
+Kernel Configuration
+====================
+
+To enable ALSA support you need at least to build the kernel with
+primary sound card support (CONFIG_SOUND).  Since ALSA can emulate OSS,
+you don't have to choose any of the OSS modules.
+
+Enable "OSS API emulation" (CONFIG_SND_OSSEMUL) and both OSS mixer and
+PCM supports if you want to run OSS applications with ALSA.
+
+If you want to support the WaveTable functionality on cards such as
+SB Live! then you need to enable "Sequencer support"
+(CONFIG_SND_SEQUENCER).
+
+To make ALSA debug messages more verbose, enable the "Verbose printk"
+and "Debug" options.  To check for memory leaks, turn on "Debug memory"
+too.  "Debug detection" will add checks for the detection of cards.
+
+Please note that all the ALSA ISA drivers support the Linux isapnp API
+(if the card supports ISA PnP).  You don't need to configure the cards
+using isapnptools.
+
+
+Creating ALSA devices
+=====================
+
+This depends on your distribution, but normally you use the /dev/MAKEDEV
+script to create the necessary device nodes.  On some systems you use a
+script named 'snddevices'.
+
+
+Module parameters
+=================
+
+The user can load modules with options. If the module supports more than
+one card and you have more than one card of the same type then you can
+specify multiple values for the option separated by commas.
+
+Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
+
+  Module snd
+  ----------
+
+    The core ALSA module.  It is used by all ALSA card drivers.
+    It takes the following options which have global effects.
+
+    major	- major number for sound driver
+		- Default: 116
+    cards_limit
+		- limiting card index for auto-loading (1-8)
+		- Default: 1
+		- For auto-loading more than one card, specify this
+		  option together with snd-card-X aliases.
+    slots	- Reserve the slot index for the given driver.
+		  This option takes multiple strings.		
+		  See "Module Autoloading Support" section for details.
+    debug	- Specifies the debug message level
+		  (0 = disable debug prints, 1 = normal debug messages,
+		   2 = verbose debug messages)
+		  This option appears only when CONFIG_SND_DEBUG=y.
+		  This option can be dynamically changed via sysfs
+		  /sys/modules/snd/parameters/debug file.
+  
+  Module snd-pcm-oss
+  ------------------
+
+    The PCM OSS emulation module.
+    This module takes options which change the mapping of devices.
+
+    dsp_map	- PCM device number maps assigned to the 1st OSS device.
+		- Default: 0
+    adsp_map	- PCM device number maps assigned to the 2st OSS device.
+		- Default: 1
+    nonblock_open
+		- Don't block opening busy PCM devices.  Default: 1
+
+    For example, when dsp_map=2, /dev/dsp will be mapped to PCM #2 of
+    the card #0.  Similarly, when adsp_map=0, /dev/adsp will be mapped
+    to PCM #0 of the card #0.
+    For changing the second or later card, specify the option with
+    commas, such like "dsp_map=0,1".
+
+    nonblock_open option is used to change the behavior of the PCM
+    regarding opening the device.  When this option is non-zero,
+    opening a busy OSS PCM device won't be blocked but return
+    immediately with EAGAIN (just like O_NONBLOCK flag).
+    
+  Module snd-rawmidi
+  ------------------
+
+    This module takes options which change the mapping of devices.
+    similar to those of the snd-pcm-oss module.
+
+    midi_map	- MIDI device number maps assigned to the 1st OSS device.
+		- Default: 0
+    amidi_map	- MIDI device number maps assigned to the 2st OSS device.
+		- Default: 1
+
+  Common parameters for top sound card modules
+  --------------------------------------------
+
+    Each of top level sound card module takes the following options.
+
+    index	- index (slot #) of sound card
+		- Values: 0 through 31 or negative
+		- If nonnegative, assign that index number
+                - if negative, interpret as a bitmask of permissible
+		  indices; the first free permitted index is assigned
+		- Default: -1
+    id		- card ID (identifier or name)
+		- Can be up to 15 characters long
+		- Default: the card type
+		- A directory by this name is created under /proc/asound/
+		  containing information about the card
+		- This ID can be used instead of the index number in
+		  identifying the card
+    enable  	- enable card
+		- Default: enabled, for PCI and ISA PnP cards
+
+  Module snd-adlib
+  ----------------
+
+    Module for AdLib FM cards.
+
+    port	- port # for OPL chip
+
+    This module supports multiple cards. It does not support autoprobe, so
+    the port must be specified. For actual AdLib FM cards it will be 0x388.
+    Note that this card does not have PCM support and no mixer; only FM
+    synthesis.
+
+    Make sure you have "sbiload" from the alsa-tools package available and,
+    after loading the module, find out the assigned ALSA sequencer port
+    number through "sbiload -l". Example output:
+
+      Port     Client name                       Port name
+      64:0     OPL2 FM synth                     OPL2 FM Port
+
+    Load the std.sb and drums.sb patches also supplied by sbiload:
+
+      sbiload -p 64:0 std.sb drums.sb
+
+    If you use this driver to drive an OPL3, you can use std.o3 and drums.o3
+    instead. To have the card produce sound, use aplaymidi from alsa-utils:
+
+      aplaymidi -p 64:0 foo.mid
+
+  Module snd-ad1816a
+  ------------------
+
+    Module for sound cards based on Analog Devices AD1816A/AD1815 ISA chips.
+
+    clockfreq   - Clock frequency for AD1816A chip (default = 0, 33000Hz)
+    
+    This module supports multiple cards, autoprobe and PnP.
+    
+  Module snd-ad1848
+  -----------------
+
+    Module for sound cards based on AD1848/AD1847/CS4248 ISA chips.
+
+    port	- port # for AD1848 chip
+    irq		- IRQ # for AD1848  chip
+    dma1	- DMA # for AD1848 chip (0,1,3)
+    
+    This module supports multiple cards.  It does not support autoprobe
+    thus main port must be specified!!! Other ports are optional.
+    
+    The power-management is supported.
+
+  Module snd-ad1889
+  -----------------
+
+    Module for Analog Devices AD1889 chips.
+
+    ac97_quirk  - AC'97 workaround for strange hardware
+                  See the description of intel8x0 module for details.
+
+    This module supports multiple cards.
+
+  Module snd-ali5451
+  ------------------
+
+    Module for ALi M5451 PCI chip.
+
+    pcm_channels    - Number of hardware channels assigned for PCM
+    spdif           - Support SPDIF I/O
+    		    - Default: disabled
+
+    This module supports one chip and autoprobe.
+
+    The power-management is supported.
+
+  Module snd-als100
+  -----------------
+
+    Module for sound cards based on Avance Logic ALS100/ALS120 ISA chips.
+
+    This module supports multiple cards, autoprobe and PnP.
+
+    The power-management is supported.
+
+  Module snd-als300
+  -----------------
+
+    Module for  Avance Logic ALS300 and ALS300+
+
+    This module supports multiple cards.
+
+    The power-management is supported.
+
+  Module snd-als4000
+  ------------------
+
+    Module for sound cards based on Avance Logic ALS4000 PCI chip.
+
+    joystick_port - port # for legacy joystick support.
+                    0 = disabled (default), 1 = auto-detect
+    
+    This module supports multiple cards, autoprobe and PnP.
+
+    The power-management is supported.
+
+  Module snd-atiixp
+  -----------------
+
+    Module for ATI IXP 150/200/250/400 AC97 controllers.
+
+    ac97_clock		- AC'97 clock (default = 48000)
+    ac97_quirk		- AC'97 workaround for strange hardware
+			  See "AC97 Quirk Option" section below.
+    ac97_codec		- Workaround to specify which AC'97 codec 
+			  instead of probing.  If this works for you
+			  file a bug with your `lspci -vn` output.
+			  -2  -- Force probing.
+			  -1  -- Default behavior.
+			  0-2 -- Use the specified codec.
+    spdif_aclink	- S/PDIF transfer over AC-link (default = 1)
+
+    This module supports one card and autoprobe.
+
+    ATI IXP has two different methods to control SPDIF output.  One is
+    over AC-link and another is over the "direct" SPDIF output.  The
+    implementation depends on the motherboard, and you'll need to
+    choose the correct one via spdif_aclink module option.
+
+    The power-management is supported.
+
+  Module snd-atiixp-modem
+  -----------------------
+
+    Module for ATI IXP 150/200/250 AC97 modem controllers.
+
+    This module supports one card and autoprobe.
+
+    Note: The default index value of this module is -2, i.e. the first
+          slot is excluded.
+
+    The power-management is supported.
+
+  Module snd-au8810, snd-au8820, snd-au8830
+  -----------------------------------------
+
+    Module for Aureal Vortex, Vortex2 and Advantage device.
+
+    pcifix	- Control PCI workarounds
+		  0 = Disable all workarounds
+		  1 = Force the PCI latency of the Aureal card to 0xff
+		  2 = Force the Extend PCI#2 Internal Master for Efficient
+		      Handling of Dummy Requests on the VIA KT133 AGP Bridge
+		  3 = Force both settings
+		  255 = Autodetect what is required (default)
+
+    This module supports all ADB PCM channels, ac97 mixer, SPDIF, hardware
+    EQ, mpu401, gameport. A3D and wavetable support are still in development.
+    Development and reverse engineering work is being coordinated at
+    http://savannah.nongnu.org/projects/openvortex/
+    SPDIF output has a copy of the AC97 codec output, unless you use the
+    "spdif" pcm device, which allows raw data passthru.
+    The hardware EQ hardware and SPDIF is only present in the Vortex2 and 
+    Advantage.
+
+    Note: Some ALSA mixer applications don't handle the SPDIF sample rate 
+           control correctly. If you have problems regarding this, try
+           another ALSA compliant mixer (alsamixer works).
+
+  Module snd-aw2
+  --------------
+
+    Module for Audiowerk2 sound card
+
+    This module supports multiple cards.
+
+  Module snd-azt2320
+  ------------------
+
+    Module for sound cards based on Aztech System AZT2320 ISA chip (PnP only).
+
+    This module supports multiple cards, PnP and autoprobe.
+    
+    The power-management is supported.
+
+  Module snd-azt3328
+  ------------------
+
+    Module for sound cards based on Aztech AZF3328 PCI chip.
+
+    joystick	- Enable joystick (default off)
+
+    This module supports multiple cards.
+
+  Module snd-bt87x
+  ----------------
+
+    Module for video cards based on Bt87x chips.
+
+    digital_rate - Override the default digital rate (Hz)
+    load_all	- Load the driver even if the card model isn't known
+
+    This module supports multiple cards.
+
+    Note: The default index value of this module is -2, i.e. the first
+          slot is excluded.
+
+  Module snd-ca0106
+  -----------------
+
+    Module for Creative Audigy LS and SB Live 24bit
+
+    This module supports multiple cards.
+
+
+  Module snd-cmi8330
+  ------------------
+
+    Module for sound cards based on C-Media CMI8330 ISA chips.
+
+    isapnp	- ISA PnP detection - 0 = disable, 1 = enable (default)
+
+    with isapnp=0, the following options are available:
+
+    wssport	- port # for CMI8330 chip (WSS)
+    wssirq	- IRQ # for CMI8330 chip (WSS)
+    wssdma	- first DMA # for CMI8330 chip (WSS)
+    sbport	- port # for CMI8330 chip (SB16)
+    sbirq	- IRQ # for CMI8330 chip (SB16)
+    sbdma8	- 8bit DMA # for CMI8330 chip (SB16)
+    sbdma16	- 16bit DMA # for CMI8330 chip (SB16)
+    fmport	- (optional) OPL3 I/O port
+    mpuport	- (optional) MPU401 I/O port
+    mpuirq	- (optional) MPU401 irq #
+
+    This module supports multiple cards and autoprobe.
+
+    The power-management is supported.
+
+  Module snd-cmipci
+  -----------------
+
+    Module for C-Media CMI8338/8738/8768/8770 PCI sound cards.
+
+    mpu_port    - port address of MIDI interface (8338 only):
+		  0x300,0x310,0x320,0x330 = legacy port,
+		  0 = disable (default)
+    fm_port     - port address of OPL-3 FM synthesizer (8x38 only):
+		  0x388 = legacy port,
+		  1 = integrated PCI port (default on 8738),
+		  0 = disable
+    soft_ac3    - Software-conversion of raw SPDIF packets (model 033 only)
+                  (default = 1)
+    joystick_port - Joystick port address (0 = disable, 1 = auto-detect)
+
+    This module supports autoprobe and multiple cards.
+    
+    The power-management is supported.
+
+  Module snd-cs4231
+  -----------------
+
+    Module for sound cards based on CS4231 ISA chips.
+
+    port	- port # for CS4231 chip
+    mpu_port	- port # for MPU-401 UART (optional), -1 = disable
+    irq		- IRQ # for CS4231 chip
+    mpu_irq	- IRQ # for MPU-401 UART
+    dma1	- first DMA # for CS4231 chip
+    dma2	- second DMA # for CS4231 chip
+    
+    This module supports multiple cards. This module does not support autoprobe
+    thus main port must be specified!!! Other ports are optional.
+
+    The power-management is supported.
+    
+  Module snd-cs4236
+  -----------------
+
+    Module for sound cards based on CS4232/CS4232A,
+    	       	     	   	   CS4235/CS4236/CS4236B/CS4237B/
+                                   CS4238B/CS4239 ISA chips.
+
+    isapnp	- ISA PnP detection - 0 = disable, 1 = enable (default)
+
+    with isapnp=0, the following options are available:
+
+    port	- port # for CS4236 chip (PnP setup - 0x534)
+    cport	- control port # for CS4236 chip (PnP setup - 0x120,0x210,0xf00)
+    mpu_port	- port # for MPU-401 UART (PnP setup - 0x300), -1 = disable
+    fm_port	- FM port # for CS4236 chip (PnP setup - 0x388), -1 = disable
+    irq		- IRQ # for CS4236 chip (5,7,9,11,12,15)
+    mpu_irq	- IRQ # for MPU-401 UART (9,11,12,15)
+    dma1	- first DMA # for CS4236 chip (0,1,3)
+    dma2	- second DMA # for CS4236 chip (0,1,3), -1 = disable
+    
+    This module supports multiple cards. This module does not support autoprobe
+    (if ISA PnP is not used) thus main port and control port must be
+    specified!!! Other ports are optional.
+
+    The power-management is supported.
+
+    This module is aliased as snd-cs4232 since it provides the old
+    snd-cs4232 functionality, too.
+
+  Module snd-cs4281
+  -----------------
+
+    Module for Cirrus Logic CS4281 soundchip.
+
+    dual_codec	- Secondary codec ID (0 = disable, default)
+
+    This module supports multiple cards.
+
+    The power-management is supported.
+
+  Module snd-cs46xx
+  -----------------
+
+    Module for PCI sound cards based on CS4610/CS4612/CS4614/CS4615/CS4622/
+				       CS4624/CS4630/CS4280 PCI chips.
+
+    external_amp     - Force to enable external amplifier.
+    thinkpad         - Force to enable Thinkpad's CLKRUN control.
+    mmap_valid       - Support OSS mmap mode (default = 0).
+
+    This module supports multiple cards and autoprobe.
+    Usually external amp and CLKRUN controls are detected automatically
+    from PCI sub vendor/device ids.  If they don't work, give the options
+    above explicitly.
+
+    The power-management is supported.
+
+  Module snd-cs5530
+  _________________
+
+    Module for Cyrix/NatSemi Geode 5530 chip. 
+  
+  Module snd-cs5535audio
+  ----------------------
+
+    Module for multifunction CS5535 companion PCI device
+
+    The power-management is supported.
+
+  Module snd-ctxfi
+  ----------------
+
+    Module for Creative Sound Blaster X-Fi boards (20k1 / 20k2 chips)
+	* Creative Sound Blaster X-Fi Titanium Fatal1ty Champion Series
+	* Creative Sound Blaster X-Fi Titanium Fatal1ty Professional Series
+	* Creative Sound Blaster X-Fi Titanium Professional Audio
+	* Creative Sound Blaster X-Fi Titanium
+	* Creative Sound Blaster X-Fi Elite Pro
+	* Creative Sound Blaster X-Fi Platinum
+	* Creative Sound Blaster X-Fi Fatal1ty
+	* Creative Sound Blaster X-Fi XtremeGamer
+	* Creative Sound Blaster X-Fi XtremeMusic
+
+    reference_rate	- reference sample rate, 44100 or 48000 (default)
+    multiple		- multiple to ref. sample rate, 1 or 2 (default)
+    subsystem		- override the PCI SSID for probing; the value
+			  consists of SSVID << 16 | SSDID.  The default is
+			  zero, which means no override.
+
+    This module supports multiple cards.
+
+  Module snd-darla20
+  ------------------
+
+    Module for Echoaudio Darla20
+
+    This module supports multiple cards.
+    The driver requires the firmware loader support on kernel.
+
+  Module snd-darla24
+  ------------------
+
+    Module for Echoaudio Darla24
+
+    This module supports multiple cards.
+    The driver requires the firmware loader support on kernel.
+
+  Module snd-dt019x
+  -----------------
+
+    Module for Diamond Technologies DT-019X / Avance Logic ALS-007 (PnP
+    only)
+
+    This module supports multiple cards.  This module is enabled only with
+    ISA PnP support.
+
+    The power-management is supported.
+
+  Module snd-dummy
+  ----------------
+
+    Module for the dummy sound card. This "card" doesn't do any output
+    or input, but you may use this module for any application which
+    requires a sound card (like RealPlayer).
+
+    pcm_devs       - Number of PCM devices assigned to each card
+                     (default = 1, up to 4)
+    pcm_substreams - Number of PCM substreams assigned to each PCM
+                     (default = 8, up to 128)
+    hrtimer        - Use hrtimer (=1, default) or system timer (=0)
+    fake_buffer    - Fake buffer allocations (default = 1)
+
+    When multiple PCM devices are created, snd-dummy gives different
+    behavior to each PCM device:
+      0 = interleaved with mmap support
+      1 = non-interleaved with mmap support
+      2 = interleaved without mmap 
+      3 = non-interleaved without mmap
+
+    As default, snd-dummy drivers doesn't allocate the real buffers
+    but either ignores read/write or mmap a single dummy page to all
+    buffer pages, in order to save the resouces.  If your apps need
+    the read/ written buffer data to be consistent, pass fake_buffer=0
+    option.
+
+    The power-management is supported.
+
+  Module snd-echo3g
+  -----------------
+
+    Module for Echoaudio 3G cards (Gina3G/Layla3G)
+
+    This module supports multiple cards.
+    The driver requires the firmware loader support on kernel.
+
+  Module snd-emu10k1
+  ------------------
+
+    Module for EMU10K1/EMU10k2 based PCI sound cards.
+			* Sound Blaster Live!
+			* Sound Blaster PCI 512
+			* Emu APS (partially supported)
+			* Sound Blaster Audigy
+
+    extin   - bitmap of available external inputs for FX8010 (see bellow)
+    extout  - bitmap of available external outputs for FX8010 (see bellow)
+    seq_ports - allocated sequencer ports (4 by default)
+    max_synth_voices - limit of voices used for wavetable (64 by default)
+    max_buffer_size  - specifies the maximum size of wavetable/pcm buffers
+                       given in MB unit.  Default value is 128.
+    enable_ir - enable IR
+
+    This module supports multiple cards and autoprobe.
+
+    Input & Output configurations 			[extin/extout]
+	* Creative Card wo/Digital out			[0x0003/0x1f03]
+	* Creative Card w/Digital out			[0x0003/0x1f0f]
+	* Creative Card w/Digital CD in			[0x000f/0x1f0f]
+	* Creative Card wo/Digital out + LiveDrive	[0x3fc3/0x1fc3]
+	* Creative Card w/Digital out + LiveDrive	[0x3fc3/0x1fcf]
+	* Creative Card w/Digital CD in + LiveDrive	[0x3fcf/0x1fcf]
+	* Creative Card wo/Digital out + Digital I/O 2  [0x0fc3/0x1f0f]
+	* Creative Card w/Digital out + Digital I/O 2	[0x0fc3/0x1f0f]
+	* Creative Card w/Digital CD in + Digital I/O 2	[0x0fcf/0x1f0f]
+        * Creative Card 5.1/w Digital out + LiveDrive	[0x3fc3/0x1fff]
+	* Creative Card 5.1 (c) 2003			[0x3fc3/0x7cff]
+        * Creative Card all ins and outs		[0x3fff/0x7fff]
+    
+    The power-management is supported.
+
+  Module snd-emu10k1x
+  -------------------
+
+    Module for Creative Emu10k1X (SB Live Dell OEM version)
+
+    This module supports multiple cards.
+
+  Module snd-ens1370
+  ------------------
+
+    Module for Ensoniq AudioPCI ES1370 PCI sound cards.
+			* SoundBlaster PCI 64
+			* SoundBlaster PCI 128
+
+    joystick		- Enable joystick (default off)
+
+    This module supports multiple cards and autoprobe.
+    
+    The power-management is supported.
+
+  Module snd-ens1371
+  ------------------
+
+    Module for Ensoniq AudioPCI ES1371 PCI sound cards.
+			* SoundBlaster PCI 64
+			* SoundBlaster PCI 128
+			* SoundBlaster Vibra PCI
+
+    joystick_port	- port # for joystick (0x200,0x208,0x210,0x218),
+			  0 = disable (default), 1 = auto-detect
+
+    This module supports multiple cards and autoprobe.
+    
+    The power-management is supported.
+
+  Module snd-es968
+  ----------------
+
+    Module for sound cards based on ESS ES968 chip (PnP only).
+
+    This module supports multiple cards, PnP and autoprobe.
+    
+    The power-management is supported.
+
+  Module snd-es1688
+  -----------------
+
+    Module for ESS AudioDrive ES-1688 and ES-688 sound cards.
+
+    port	- port # for ES-1688 chip (0x220,0x240,0x260)
+    fm_port	- port # for OPL3 (option; share the same port as default)
+    mpu_port	- port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable (default)
+    irq		- IRQ # for ES-1688 chip (5,7,9,10)
+    mpu_irq	- IRQ # for MPU-401 port (5,7,9,10)
+    dma8	- DMA # for ES-1688 chip (0,1,3)
+
+    This module supports multiple cards and autoprobe (without MPU-401 port).
+
+  Module snd-es18xx
+  -----------------
+
+    Module for ESS AudioDrive ES-18xx sound cards.
+
+    isapnp	- ISA PnP detection - 0 = disable, 1 = enable (default)
+
+    with isapnp=0, the following options are available:
+
+    port	- port # for ES-18xx chip (0x220,0x240,0x260)
+    mpu_port	- port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable (default)
+    fm_port	- port # for FM (optional, not used)
+    irq		- IRQ # for ES-18xx chip (5,7,9,10)
+    dma1	- first DMA # for ES-18xx chip (0,1,3)
+    dma2	- first DMA # for ES-18xx chip (0,1,3)
+
+    This module supports multiple cards, ISA PnP and autoprobe (without MPU-401
+    port if native ISA PnP routines are not used).
+    When dma2 is equal with dma1, the driver works as half-duplex.
+
+    The power-management is supported.
+
+  Module snd-es1938
+  -----------------
+
+    Module for sound cards based on ESS Solo-1 (ES1938,ES1946) chips.
+
+    This module supports multiple cards and autoprobe.
+
+    The power-management is supported.
+
+  Module snd-es1968
+  -----------------
+
+    Module for sound cards based on ESS Maestro-1/2/2E (ES1968/ES1978) chips.
+
+    total_bufsize	- total buffer size in kB (1-4096kB)
+    pcm_substreams_p	- playback channels (1-8, default=2)
+    pcm_substreams_c	- capture channels (1-8, default=0)
+    clock		- clock (0 = auto-detection)
+    use_pm		- support the power-management (0 = off, 1 = on,
+			  2 = auto (default))
+    enable_mpu		- enable MPU401 (0 = off, 1 = on, 2 = auto (default))
+    joystick		- enable joystick (default off)       
+
+    This module supports multiple cards and autoprobe.
+
+    The power-management is supported.
+
+  Module snd-fm801
+  ----------------
+
+    Module for ForteMedia FM801 based PCI sound cards.
+
+    tea575x_tuner       - Enable TEA575x tuner
+                          - 1 = MediaForte 256-PCS
+                          - 2 = MediaForte 256-PCPR
+                          - 3 = MediaForte 64-PCR  
+                          - High 16-bits are video (radio) device number + 1
+                          - example: 0x10002 (MediaForte 256-PCPR, device 1)
+
+    This module supports multiple cards and autoprobe.
+    
+    The power-management is supported.
+
+  Module snd-gina20
+  -----------------
+
+    Module for Echoaudio Gina20
+
+    This module supports multiple cards.
+    The driver requires the firmware loader support on kernel.
+
+  Module snd-gina24
+  -----------------
+
+    Module for Echoaudio Gina24
+
+    This module supports multiple cards.
+    The driver requires the firmware loader support on kernel.
+
+  Module snd-gusclassic
+  ---------------------
+
+    Module for Gravis UltraSound Classic sound card.
+
+    port	- port # for GF1 chip (0x220,0x230,0x240,0x250,0x260)
+    irq		- IRQ # for GF1 chip (3,5,9,11,12,15)
+    dma1	- DMA # for GF1 chip (1,3,5,6,7)
+    dma2	- DMA # for GF1 chip (1,3,5,6,7,-1=disable)
+    joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V)
+    voices	- GF1 voices limit (14-32)
+    pcm_voices	- reserved PCM voices
+
+    This module supports multiple cards and autoprobe.
+
+  Module snd-gusextreme
+  ---------------------
+
+    Module for Gravis UltraSound Extreme (Synergy ViperMax) sound card.
+
+    port	- port # for ES-1688 chip (0x220,0x230,0x240,0x250,0x260)
+    gf1_port	- port # for GF1 chip (0x210,0x220,0x230,0x240,0x250,0x260,0x270)
+    mpu_port	- port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable
+    irq		- IRQ # for ES-1688 chip (5,7,9,10)
+    gf1_irq	- IRQ # for GF1 chip (3,5,9,11,12,15)
+    mpu_irq	- IRQ # for MPU-401 port (5,7,9,10)
+    dma8	- DMA # for ES-1688 chip (0,1,3)
+    dma1	- DMA # for GF1 chip (1,3,5,6,7)
+    joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V)
+    voices	- GF1 voices limit (14-32)
+    pcm_voices	- reserved PCM voices
+
+    This module supports multiple cards and autoprobe (without MPU-401 port).
+
+  Module snd-gusmax
+  -----------------
+
+    Module for Gravis UltraSound MAX sound card.
+
+    port	- port # for GF1 chip (0x220,0x230,0x240,0x250,0x260)
+    irq		- IRQ # for GF1 chip (3,5,9,11,12,15)
+    dma1	- DMA # for GF1 chip (1,3,5,6,7)
+    dma2	- DMA # for GF1 chip (1,3,5,6,7,-1=disable)
+    joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V)
+    voices	- GF1 voices limit (14-32)
+    pcm_voices	- reserved PCM voices
+
+    This module supports multiple cards and autoprobe.
+    
+  Module snd-hda-intel
+  --------------------
+
+    Module for Intel HD Audio (ICH6, ICH6M, ESB2, ICH7, ICH8, ICH9, ICH10,
+			PCH, SCH),
+		ATI SB450, SB600, R600, RS600, RS690, RS780, RV610, RV620,
+			RV630, RV635, RV670, RV770,
+		VIA VT8251/VT8237A,
+		SIS966, ULI M5461
+
+    [Multiple options for each card instance]
+    model	- force the model name
+    position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF)
+    probe_mask  - Bitmask to probe codecs (default = -1, meaning all slots)
+    		  When the bit 8 (0x100) is set, the lower 8 bits are used
+		  as the "fixed" codec slots; i.e. the driver probes the
+		  slots regardless what hardware reports back
+    probe_only	- Only probing and no codec initialization (default=off);
+		  Useful to check the initial codec status for debugging
+    bdl_pos_adj	- Specifies the DMA IRQ timing delay in samples.
+		Passing -1 will make the driver to choose the appropriate
+		value based on the controller chip.
+    patch	- Specifies the early "patch" files to modify the HD-audio
+    		setup before initializing the codecs.  This option is
+		available only when CONFIG_SND_HDA_PATCH_LOADER=y is set.
+		See HD-Audio.txt for details.
+    beep_mode	- Selects the beep registration mode (0=off, 1=on, 2=
+		dynamic registration via mute switch on/off); the default
+		value is set via CONFIG_SND_HDA_INPUT_BEEP_MODE kconfig.
+    
+    [Single (global) options]
+    single_cmd  - Use single immediate commands to communicate with
+		codecs (for debugging only)
+    enable_msi	- Enable Message Signaled Interrupt (MSI) (default = off)
+    power_save	- Automatic power-saving timeout (in second, 0 =
+		disable)
+    power_save_controller - Reset HD-audio controller in power-saving mode
+		(default = on)
+
+    This module supports multiple cards and autoprobe.
+    
+    See Documentation/sound/alsa/HD-Audio.txt for more details about
+    HD-audio driver.
+
+    Each codec may have a model table for different configurations.
+    If your machine isn't listed there, the default (usually minimal)
+    configuration is set up.  You can pass "model=<name>" option to
+    specify a certain model in such a case.  There are different
+    models depending on the codec chip.  The list of available models
+    is found in HD-Audio-Models.txt
+
+    The model name "genric" is treated as a special case.  When this
+    model is given, the driver uses the generic codec parser without
+    "codec-patch".  It's sometimes good for testing and debugging.
+
+    If the default configuration doesn't work and one of the above
+    matches with your device, report it together with alsa-info.sh
+    output (with --no-upload option) to kernel bugzilla or alsa-devel
+    ML (see the section "Links and Addresses").
+
+    power_save and power_save_controller options are for power-saving
+    mode.  See powersave.txt for details.
+
+    Note 2: If you get click noises on output, try the module option
+	    position_fix=1 or 2.  position_fix=1 will use the SD_LPIB
+	    register value without FIFO size correction as the current
+	    DMA pointer.  position_fix=2 will make the driver to use
+	    the position buffer instead of reading SD_LPIB register.
+	    (Usually SD_LPIB register is more accurate than the
+	    position buffer.)
+
+    NB: If you get many "azx_get_response timeout" messages at
+    loading, it's likely a problem of interrupts (e.g. ACPI irq
+    routing).  Try to boot with options like "pci=noacpi".  Also, you
+    can try "single_cmd=1" module option.  This will switch the
+    communication method between HDA controller and codecs to the
+    single immediate commands instead of CORB/RIRB.  Basically, the
+    single command mode is provided only for BIOS, and you won't get
+    unsolicited events, too.  But, at least, this works independently
+    from the irq.  Remember this is a last resort, and should be
+    avoided as much as possible...
+    
+    MORE NOTES ON "azx_get_response timeout" PROBLEMS:
+    On some hardwares, you may need to add a proper probe_mask option
+    to avoid the "azx_get_response timeout" problem above, instead.
+    This occurs when the access to non-existing or non-working codec slot
+    (likely a modem one) causes a stall of the communication via HD-audio
+    bus.  You can see which codec slots are probed by enabling
+    CONFIG_SND_DEBUG_VERBOSE, or simply from the file name of the codec
+    proc files.  Then limit the slots to probe by probe_mask option.
+    For example, probe_mask=1 means to probe only the first slot, and
+    probe_mask=4 means only the third slot.
+
+    The power-management is supported.
+
+  Module snd-hdsp
+  ---------------
+
+    Module for RME Hammerfall DSP audio interface(s)
+
+    This module supports multiple cards.
+
+    Note: The firmware data can be automatically loaded via hotplug
+          when CONFIG_FW_LOADER is set.  Otherwise, you need to load
+          the firmware via hdsploader utility included in alsa-tools
+          package.
+          The firmware data is found in alsa-firmware package.
+
+    Note: snd-page-alloc module does the job which snd-hammerfall-mem
+          module did formerly.  It will allocate the buffers in advance
+          when any HDSP cards are found.  To make the buffer
+          allocation sure, load snd-page-alloc module in the early
+          stage of boot sequence.  See "Early Buffer Allocation"
+	  section.
+
+  Module snd-hdspm
+  ----------------
+
+    Module for RME HDSP MADI board.
+
+    precise_ptr		- Enable precise pointer, or disable.
+    line_outs_monitor	- Send playback streams to analog outs by default.
+    enable_monitor	- Enable Analog Out on Channel 63/64 by default.
+
+    See hdspm.txt for details.
+
+  Module snd-hifier
+  -----------------
+
+    Module for the MediaTek/TempoTec HiFier Fantasia sound card.
+
+    This module supports autoprobe and multiple cards.
+
+  Module snd-ice1712
+  ------------------
+
+    Module for Envy24 (ICE1712) based PCI sound cards.
+			* MidiMan M Audio Delta 1010
+			* MidiMan M Audio Delta 1010LT
+			* MidiMan M Audio Delta DiO 2496
+			* MidiMan M Audio Delta 66
+			* MidiMan M Audio Delta 44
+			* MidiMan M Audio Delta 410
+			* MidiMan M Audio Audiophile 2496
+                        * TerraTec EWS 88MT
+                        * TerraTec EWS 88D
+                        * TerraTec EWX 24/96
+                        * TerraTec DMX 6Fire
+			* TerraTec Phase 88
+                        * Hoontech SoundTrack DSP 24
+                        * Hoontech SoundTrack DSP 24 Value
+                        * Hoontech SoundTrack DSP 24 Media 7.1
+			* Event Electronics, EZ8
+                        * Digigram VX442
+			* Lionstracs, Mediastaton
+			* Terrasoniq TS 88
+
+    model       - Use the given board model, one of the following:
+		  delta1010, dio2496, delta66, delta44, audiophile, delta410,
+		  delta1010lt, vx442, ewx2496, ews88mt, ews88mt_new, ews88d,
+		  dmx6fire, dsp24, dsp24_value, dsp24_71, ez8,
+		  phase88, mediastation
+    omni	- Omni I/O support for MidiMan M-Audio Delta44/66
+    cs8427_timeout - reset timeout for the CS8427 chip (S/PDIF transceiver)
+                     in msec resolution, default value is 500 (0.5 sec)
+
+    This module supports multiple cards and autoprobe. Note: The consumer part
+    is not used with all Envy24 based cards (for example in the MidiMan Delta
+    serie).
+
+    Note: The supported board is detected by reading EEPROM or PCI
+	  SSID (if EEPROM isn't available).  You can override the
+	  model by passing "model" module option in case that the
+	  driver isn't configured properly or you want to try another
+	  type for testing.
+
+  Module snd-ice1724
+  ------------------
+
+    Module for Envy24HT (VT/ICE1724), Envy24PT (VT1720) based PCI sound cards.
+			* MidiMan M Audio Revolution 5.1
+			* MidiMan M Audio Revolution 7.1
+			* MidiMan M Audio Audiophile 192
+			* AMP Ltd AUDIO2000
+			* TerraTec Aureon 5.1 Sky
+			* TerraTec Aureon 7.1 Space
+			* TerraTec Aureon 7.1 Universe
+			* TerraTec Phase 22
+			* TerraTec Phase 28
+			* AudioTrak Prodigy 7.1
+			* AudioTrak Prodigy 7.1 LT
+			* AudioTrak Prodigy 7.1 XT
+			* AudioTrak Prodigy 7.1 HIFI
+			* AudioTrak Prodigy 7.1 HD2
+			* AudioTrak Prodigy 192
+			* Pontis MS300
+			* Albatron K8X800 Pro II 
+			* Chaintech ZNF3-150
+			* Chaintech ZNF3-250
+			* Chaintech 9CJS
+			* Chaintech AV-710
+			* Shuttle SN25P
+			* Onkyo SE-90PCI
+			* Onkyo SE-200PCI
+			* ESI Juli@
+			* ESI Maya44
+			* Hercules Fortissimo IV
+			* EGO-SYS WaveTerminal 192M
+
+    model       - Use the given board model, one of the following:
+		  revo51, revo71, amp2000, prodigy71, prodigy71lt,
+		  prodigy71xt, prodigy71hifi, prodigyhd2, prodigy192,
+		  juli, aureon51, aureon71, universe, ap192, k8x800,
+		  phase22, phase28, ms300, av710, se200pci, se90pci,
+		  fortissimo4, sn25p, WT192M, maya44
+
+    This module supports multiple cards and autoprobe.
+
+    Note: The supported board is detected by reading EEPROM or PCI
+	  SSID (if EEPROM isn't available).  You can override the
+	  model by passing "model" module option in case that the
+	  driver isn't configured properly or you want to try another
+	  type for testing.
+
+  Module snd-indigo
+  -----------------
+
+    Module for Echoaudio Indigo
+
+    This module supports multiple cards.
+    The driver requires the firmware loader support on kernel.
+
+  Module snd-indigodj
+  -------------------
+
+    Module for Echoaudio Indigo DJ
+
+    This module supports multiple cards.
+    The driver requires the firmware loader support on kernel.
+
+  Module snd-indigoio
+  -------------------
+
+    Module for Echoaudio Indigo IO
+
+    This module supports multiple cards.
+    The driver requires the firmware loader support on kernel.
+
+  Module snd-intel8x0
+  -------------------
+
+    Module for AC'97 motherboards from Intel and compatibles.
+			* Intel i810/810E, i815, i820, i830, i84x, MX440
+				ICH5, ICH6, ICH7, 6300ESB, ESB2
+			* SiS 7012 (SiS 735)
+			* NVidia NForce, NForce2, NForce3, MCP04, CK804
+				 CK8, CK8S, MCP501
+			* AMD AMD768, AMD8111
+			* ALi m5455
+
+    ac97_clock	  - AC'97 codec clock base (0 = auto-detect)
+    ac97_quirk    - AC'97 workaround for strange hardware
+		    See "AC97 Quirk Option" section below.
+    buggy_irq     - Enable workaround for buggy interrupts on some
+                    motherboards (default yes on nForce chips,
+		    otherwise off)
+    buggy_semaphore - Enable workaround for hardwares with buggy
+		    semaphores (e.g. on some ASUS laptops)
+		    (default off)
+    spdif_aclink  - Use S/PDIF over AC-link instead of direct connection
+		    from the controller chip
+		    (0 = off, 1 = on, -1 = default)
+
+    This module supports one chip and autoprobe.
+
+    Note: the latest driver supports auto-detection of chip clock.
+    if you still encounter too fast playback, specify the clock
+    explicitly via the module option "ac97_clock=41194".
+
+    Joystick/MIDI ports are not supported by this driver.  If your
+    motherboard has these devices, use the ns558 or snd-mpu401
+    modules, respectively.
+
+    The power-management is supported.
+    
+  Module snd-intel8x0m
+  --------------------
+
+    Module for Intel ICH (i8x0) chipset MC97 modems.
+			* Intel i810/810E, i815, i820, i830, i84x, MX440
+				ICH5, ICH6, ICH7
+			* SiS 7013 (SiS 735)
+			* NVidia NForce, NForce2, NForce2s, NForce3
+			* AMD AMD8111
+			* ALi m5455
+
+    ac97_clock	  - AC'97 codec clock base (0 = auto-detect)
+
+    This module supports one card and autoprobe.
+
+    Note: The default index value of this module is -2, i.e. the first
+          slot is excluded.
+
+    The power-management is supported.
+
+  Module snd-interwave
+  --------------------
+
+    Module for Gravis UltraSound PnP, Dynasonic 3-D/Pro, STB Sound Rage 32
+    and other sound cards based on AMD InterWave (tm) chip.
+  
+    joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V)
+    midi	- 1 = MIDI UART enable, 0 = MIDI UART disable (default)
+    pcm_voices	- reserved PCM voices for the synthesizer (default 2)
+    effect	- 1 = InterWave effects enable (default 0);
+                  requires 8 voices
+    isapnp	- ISA PnP detection - 0 = disable, 1 = enable (default)
+
+    with isapnp=0, the following options are available:
+
+    port	- port # for InterWave chip (0x210,0x220,0x230,0x240,0x250,0x260)
+    irq		- IRQ # for InterWave chip (3,5,9,11,12,15)
+    dma1	- DMA # for InterWave chip (0,1,3,5,6,7)
+    dma2	- DMA # for InterWave chip (0,1,3,5,6,7,-1=disable)
+
+    This module supports multiple cards, autoprobe and ISA PnP.
+
+  Module snd-interwave-stb
+  ------------------------
+
+    Module for UltraSound 32-Pro (sound card from STB used by Compaq)
+    and other sound cards based on AMD InterWave (tm) chip with TEA6330T
+    circuit for extended control of bass, treble and master volume.
+  
+    joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V)
+    midi	- 1 = MIDI UART enable, 0 = MIDI UART disable (default)
+    pcm_voices	- reserved PCM voices for the synthesizer (default 2)
+    effect	- 1 = InterWave effects enable (default 0);
+                  requires 8 voices
+    isapnp	- ISA PnP detection - 0 = disable, 1 = enable (default)
+
+    with isapnp=0, the following options are available:
+
+    port	- port # for InterWave chip (0x210,0x220,0x230,0x240,0x250,0x260)
+    port_tc	- tone control (i2c bus) port # for TEA6330T chip (0x350,0x360,0x370,0x380)
+    irq		- IRQ # for InterWave chip (3,5,9,11,12,15)
+    dma1	- DMA # for InterWave chip (0,1,3,5,6,7)
+    dma2	- DMA # for InterWave chip (0,1,3,5,6,7,-1=disable)
+
+    This module supports multiple cards, autoprobe and ISA PnP.
+
+  Module snd-jazz16
+  -------------------
+
+    Module for Media Vision Jazz16 chipset. The chipset consists of 3 chips:
+    MVD1216 + MVA416 + MVA514.
+
+    port	- port # for SB DSP chip (0x210,0x220,0x230,0x240,0x250,0x260)
+    irq		- IRQ # for SB DSP chip (3,5,7,9,10,15)
+    dma8	- DMA # for SB DSP chip (1,3)
+    dma16	- DMA # for SB DSP chip (5,7)
+    mpu_port	- MPU-401 port # (0x300,0x310,0x320,0x330)
+    mpu_irq	- MPU-401 irq # (2,3,5,7)
+
+    This module supports multiple cards.
+
+  Module snd-korg1212
+  -------------------
+
+    Module for Korg 1212 IO PCI card
+
+    This module supports multiple cards.
+
+  Module snd-layla20
+  ------------------
+
+    Module for Echoaudio Layla20
+
+    This module supports multiple cards.
+    The driver requires the firmware loader support on kernel.
+
+  Module snd-layla24
+  ------------------
+
+    Module for Echoaudio Layla24
+
+    This module supports multiple cards.
+    The driver requires the firmware loader support on kernel.
+
+  Module snd-lx6464es
+  -------------------
+
+    Module for Digigram LX6464ES boards
+
+    This module supports multiple cards.
+
+  Module snd-maestro3
+  -------------------
+
+    Module for Allegro/Maestro3 chips
+
+    external_amp     - enable external amp (enabled by default)
+    amp_gpio         - GPIO pin number for external amp (0-15) or
+                       -1 for default pin (8 for allegro, 1 for
+                       others) 
+
+    This module supports autoprobe and multiple chips.
+
+    Note: the binding of amplifier is dependent on hardware.
+    If there is no sound even though all channels are unmuted, try to
+    specify other gpio connection via amp_gpio option. 
+    For example, a Panasonic notebook might need "amp_gpio=0x0d"
+    option.
+
+    The power-management is supported.
+
+  Module snd-mia
+  ---------------
+
+    Module for Echoaudio Mia
+
+    This module supports multiple cards.
+    The driver requires the firmware loader support on kernel.
+
+  Module snd-miro
+  ---------------
+
+    Module for Miro soundcards: miroSOUND PCM 1 pro, 
+				miroSOUND PCM 12,
+				miroSOUND PCM 20 Radio.
+
+    port	- Port # (0x530,0x604,0xe80,0xf40)
+    irq		- IRQ # (5,7,9,10,11)
+    dma1	- 1st dma # (0,1,3)
+    dma2	- 2nd dma # (0,1)
+    mpu_port	- MPU-401 port # (0x300,0x310,0x320,0x330)
+    mpu_irq	- MPU-401 irq # (5,7,9,10)
+    fm_port	- FM Port # (0x388)
+    wss		- enable WSS mode
+    ide		- enable onboard ide support
+
+  Module snd-mixart
+  -----------------
+
+    Module for Digigram miXart8 sound cards.
+
+    This module supports multiple cards.
+    Note: One miXart8 board will be represented as 4 alsa cards.
+          See MIXART.txt for details.
+
+    When the driver is compiled as a module and the hotplug firmware
+    is supported, the firmware data is loaded via hotplug automatically.
+    Install the necessary firmware files in alsa-firmware package.
+    When no hotplug fw loader is available, you need to load the
+    firmware via mixartloader utility in alsa-tools package.
+
+  Module snd-mona
+  ---------------
+
+    Module for Echoaudio Mona
+
+    This module supports multiple cards.
+    The driver requires the firmware loader support on kernel.
+
+  Module snd-mpu401
+  -----------------
+
+    Module for MPU-401 UART devices.
+
+    port	- port number or -1 (disable)
+    irq		- IRQ number or -1 (disable)
+    pnp		- PnP detection - 0 = disable, 1 = enable (default)
+
+    This module supports multiple devices and PnP.
+    
+  Module snd-msnd-classic
+  -----------------------
+
+    Module for Turtle Beach MultiSound Classic, Tahiti or Monterey
+    soundcards.
+
+    io		- Port # for msnd-classic card
+    irq		- IRQ # for msnd-classic card
+    mem		- Memory address (0xb0000, 0xc8000, 0xd0000, 0xd8000,
+		  0xe0000 or 0xe8000)
+    write_ndelay - enable write ndelay (default = 1)
+    calibrate_signal - calibrate signal (default = 0)
+    isapnp	- ISA PnP detection - 0 = disable, 1 = enable (default)
+    digital	- Digital daughterboard present (default = 0)
+    cfg		- Config port (0x250, 0x260 or 0x270) default = PnP
+    reset	- Reset all devices
+    mpu_io	- MPU401 I/O port
+    mpu_irq	- MPU401 irq#
+    ide_io0	- IDE port #0
+    ide_io1	- IDE port #1
+    ide_irq	- IDE irq#
+    joystick_io	- Joystick I/O port
+
+    The driver requires firmware files "turtlebeach/msndinit.bin" and
+    "turtlebeach/msndperm.bin" in the proper firmware directory.
+
+    See Documentation/sound/oss/MultiSound for important information
+    about this driver.  Note that it has been discontinued, but the 
+    Voyetra Turtle Beach knowledge base entry for it is still available
+    at
+	http://www.turtlebeach.com/site/kb_ftp/790.asp
+
+  Module snd-msnd-pinnacle
+  ------------------------
+
+    Module for Turtle Beach MultiSound Pinnacle/Fiji soundcards.
+
+    io		- Port # for pinnacle/fiji card
+    irq		- IRQ # for pinnalce/fiji card
+    mem		- Memory address (0xb0000, 0xc8000, 0xd0000, 0xd8000,
+		  0xe0000 or 0xe8000)
+    write_ndelay - enable write ndelay (default = 1)
+    calibrate_signal - calibrate signal (default = 0)
+    isapnp	- ISA PnP detection - 0 = disable, 1 = enable (default)
+
+    The driver requires firmware files "turtlebeach/pndspini.bin" and
+    "turtlebeach/pndsperm.bin" in the proper firmware directory.
+
+  Module snd-mtpav
+  ----------------
+
+    Module for MOTU MidiTimePiece AV multiport MIDI (on the parallel
+    port).
+
+    port	- I/O port # for MTPAV (0x378,0x278, default=0x378)
+    irq		- IRQ # for MTPAV (7,5, default=7)
+    hwports	- number of supported hardware ports, default=8.
+    
+    Module supports only 1 card.  This module has no enable option.
+
+  Module snd-mts64
+  ----------------
+
+    Module for Ego Systems (ESI) Miditerminal 4140
+
+    This module supports multiple devices.
+    Requires parport (CONFIG_PARPORT).
+
+  Module snd-nm256
+  ----------------
+
+    Module for NeoMagic NM256AV/ZX chips
+
+    playback_bufsize - max playback frame size in kB (4-128kB)
+    capture_bufsize  - max capture frame size in kB (4-128kB)
+    force_ac97       - 0 or 1 (disabled by default)
+    buffer_top       - specify buffer top address
+    use_cache        - 0 or 1 (disabled by default)
+    vaio_hack        - alias buffer_top=0x25a800
+    reset_workaround - enable AC97 RESET workaround for some laptops
+    reset_workaround2 - enable extended AC97 RESET workaround for some
+		      other laptops
+
+    This module supports one chip and autoprobe.
+
+    The power-management is supported.
+
+    Note: on some notebooks the buffer address cannot be detected
+    automatically, or causes hang-up during initialization.
+    In such a case, specify the buffer top address explicitly via
+    the buffer_top option.
+    For example,
+      Sony F250: buffer_top=0x25a800
+      Sony F270: buffer_top=0x272800
+    The driver supports only ac97 codec.  It's possible to force
+    to initialize/use ac97 although it's not detected.  In such a
+    case, use force_ac97=1 option - but *NO* guarantee whether it
+    works!
+
+    Note: The NM256 chip can be linked internally with non-AC97
+    codecs.  This driver supports only the AC97 codec, and won't work
+    with machines with other (most likely CS423x or OPL3SAx) chips,
+    even though the device is detected in lspci.  In such a case, try
+    other drivers, e.g. snd-cs4232 or snd-opl3sa2.  Some has ISA-PnP
+    but some doesn't have ISA PnP.  You'll need to specify isapnp=0
+    and proper hardware parameters in the case without ISA PnP.
+
+    Note: some laptops need a workaround for AC97 RESET.  For the
+    known hardware like Dell Latitude LS and Sony PCG-F305, this
+    workaround is enabled automatically.  For other laptops with a
+    hard freeze, you can try reset_workaround=1 option.
+
+    Note: Dell Latitude CSx laptops have another problem regarding
+    AC97 RESET.  On these laptops, reset_workaround2 option is
+    turned on as default.  This option is worth to try if the
+    previous reset_workaround option doesn't help.
+
+    Note: This driver is really crappy.  It's a porting from the
+    OSS driver, which is a result of black-magic reverse engineering.
+    The detection of codec will fail if the driver is loaded *after*
+    X-server as described above.  You might be able to force to load
+    the module, but it may result in hang-up.   Hence, make sure that
+    you load this module *before* X if you encounter this kind of
+    problem.
+
+  Module snd-opl3sa2
+  ------------------
+
+    Module for Yamaha OPL3-SA2/SA3 sound cards.
+
+    isapnp	- ISA PnP detection - 0 = disable, 1 = enable (default)
+
+    with isapnp=0, the following options are available:
+
+    port	- control port # for OPL3-SA chip (0x370)
+    sb_port	- SB port # for OPL3-SA chip (0x220,0x240)
+    wss_port	- WSS port # for OPL3-SA chip (0x530,0xe80,0xf40,0x604)
+    midi_port	- port # for MPU-401 UART (0x300,0x330), -1 = disable
+    fm_port	- FM port # for OPL3-SA chip (0x388), -1 = disable
+    irq		- IRQ # for OPL3-SA chip (5,7,9,10)
+    dma1	- first DMA # for Yamaha OPL3-SA chip (0,1,3)
+    dma2	- second DMA # for Yamaha OPL3-SA chip (0,1,3), -1 = disable
+    
+    This module supports multiple cards and ISA PnP.  It does not support
+    autoprobe (if ISA PnP is not used) thus all ports must be specified!!!
+    
+    The power-management is supported.
+
+  Module snd-opti92x-ad1848
+  -------------------------
+
+    Module for sound cards based on OPTi 82c92x and Analog Devices AD1848 chips.
+    Module works with OAK Mozart cards as well.
+    
+    isapnp    - ISA PnP detection - 0 = disable, 1 = enable (default)
+
+    with isapnp=0, the following options are available:
+
+    port      - port # for WSS chip (0x530,0xe80,0xf40,0x604)
+    mpu_port  - port # for MPU-401 UART (0x300,0x310,0x320,0x330)
+    fm_port   - port # for OPL3 device (0x388)
+    irq       - IRQ # for WSS chip (5,7,9,10,11)
+    mpu_irq   - IRQ # for MPU-401 UART (5,7,9,10)
+    dma1      - first DMA # for WSS chip (0,1,3)
+
+    This module supports only one card, autoprobe and PnP.
+
+  Module snd-opti92x-cs4231
+  -------------------------
+
+    Module for sound cards based on OPTi 82c92x and Crystal CS4231 chips.
+    
+    isapnp    - ISA PnP detection - 0 = disable, 1 = enable (default)
+
+    with isapnp=0, the following options are available:
+
+    port      - port # for WSS chip (0x530,0xe80,0xf40,0x604)
+    mpu_port  - port # for MPU-401 UART (0x300,0x310,0x320,0x330)
+    fm_port   - port # for OPL3 device (0x388)
+    irq       - IRQ # for WSS chip (5,7,9,10,11)
+    mpu_irq   - IRQ # for MPU-401 UART (5,7,9,10)
+    dma1      - first DMA # for WSS chip (0,1,3)
+    dma2      - second DMA # for WSS chip (0,1,3)
+
+    This module supports only one card, autoprobe and PnP.
+
+  Module snd-opti93x
+  ------------------
+
+    Module for sound cards based on OPTi 82c93x chips.
+    
+    isapnp    - ISA PnP detection - 0 = disable, 1 = enable (default)
+
+    with isapnp=0, the following options are available:
+
+    port      - port # for WSS chip (0x530,0xe80,0xf40,0x604)
+    mpu_port  - port # for MPU-401 UART (0x300,0x310,0x320,0x330)
+    fm_port   - port # for OPL3 device (0x388)
+    irq       - IRQ # for WSS chip (5,7,9,10,11)
+    mpu_irq   - IRQ # for MPU-401 UART (5,7,9,10)
+    dma1      - first DMA # for WSS chip (0,1,3)
+    dma2      - second DMA # for WSS chip (0,1,3)
+
+    This module supports only one card, autoprobe and PnP.
+
+  Module snd-oxygen
+  -----------------
+
+    Module for sound cards based on the C-Media CMI8788 chip:
+    * Asound A-8788
+    * AuzenTech X-Meridian
+    * Bgears b-Enspirer
+    * Club3D Theatron DTS
+    * HT-Omega Claro (plus)
+    * HT-Omega Claro halo (XT)
+    * Razer Barracuda AC-1
+    * Sondigo Inferno
+
+    This module supports autoprobe and multiple cards.
+
+  Module snd-pcsp
+  -----------------
+
+    Module for internal PC-Speaker.
+
+    nopcm	- Disable PC-Speaker PCM sound. Only beeps remain.
+    nforce_wa	- enable NForce chipset workaround. Expect bad sound.
+
+    This module supports system beeps, some kind of PCM playback and
+    even a few mixer controls.
+
+  Module snd-pcxhr
+  ----------------
+
+    Module for Digigram PCXHR boards
+
+    This module supports multiple cards.
+
+  Module snd-portman2x4
+  ---------------------
+
+    Module for Midiman Portman 2x4 parallel port MIDI interface
+
+    This module supports multiple cards.
+
+  Module snd-powermac (on ppc only)
+  ---------------------------------
+
+    Module for PowerMac, iMac and iBook on-board soundchips
+
+    enable_beep     - enable beep using PCM (enabled as default)
+
+    Module supports autoprobe a chip.
+
+    Note: the driver may have problems regarding endianess.
+
+    The power-management is supported.
+
+  Module snd-pxa2xx-ac97 (on arm only)
+  ------------------------------------
+
+    Module for AC97 driver for the Intel PXA2xx chip
+
+    For ARM architecture only.
+
+    The power-management is supported.
+
+  Module snd-riptide
+  ------------------
+
+    Module for Conexant Riptide chip
+
+      joystick_port	- Joystick port # (default: 0x200)
+      mpu_port		- MPU401 port # (default: 0x330)
+      opl3_port		- OPL3 port # (default: 0x388)
+
+    This module supports multiple cards.
+    The driver requires the firmware loader support on kernel.
+    You need to install the firmware file "riptide.hex" to the standard
+    firmware path (e.g. /lib/firmware).
+
+  Module snd-rme32
+  ----------------
+
+    Module for RME Digi32, Digi32 Pro and Digi32/8 (Sek'd Prodif32, 
+    Prodif96 and Prodif Gold) sound cards.
+
+    This module supports multiple cards.
+
+  Module snd-rme96
+  ----------------
+
+    Module for RME Digi96, Digi96/8 and Digi96/8 PRO/PAD/PST sound cards.
+
+    This module supports multiple cards.
+
+  Module snd-rme9652
+  ------------------
+
+    Module for RME Digi9652 (Hammerfall, Hammerfall-Light) sound cards.
+
+    precise_ptr	- Enable precise pointer (doesn't work reliably).
+		  (default = 0)
+
+    This module supports multiple cards.
+
+    Note: snd-page-alloc module does the job which snd-hammerfall-mem
+          module did formerly.  It will allocate the buffers in advance
+          when any RME9652 cards are found.  To make the buffer
+          allocation sure, load snd-page-alloc module in the early
+          stage of boot sequence.  See "Early Buffer Allocation"
+	  section.
+
+  Module snd-sa11xx-uda1341 (on arm only)
+  ---------------------------------------
+
+    Module for Philips UDA1341TS on Compaq iPAQ H3600 sound card.
+
+    Module supports only one card.
+    Module has no enable and index options.
+
+    The power-management is supported.
+
+  Module snd-sb8
+  --------------
+
+    Module for 8-bit SoundBlaster cards: SoundBlaster 1.0,
+					 SoundBlaster 2.0,
+					 SoundBlaster Pro
+
+    port	- port # for SB DSP chip (0x220,0x240,0x260)
+    irq		- IRQ # for SB DSP chip (5,7,9,10)
+    dma8	- DMA # for SB DSP chip (1,3)
+
+    This module supports multiple cards and autoprobe.
+    
+    The power-management is supported.
+
+  Module snd-sb16 and snd-sbawe
+  -----------------------------
+
+    Module for 16-bit SoundBlaster cards: SoundBlaster 16 (PnP),
+					  SoundBlaster AWE 32 (PnP),
+					  SoundBlaster AWE 64 PnP
+
+    mic_agc	- Mic Auto-Gain-Control - 0 = disable, 1 = enable (default)
+    csp		- ASP/CSP chip support - 0 = disable (default), 1 = enable
+    isapnp	- ISA PnP detection - 0 = disable, 1 = enable (default)
+
+    with isapnp=0, the following options are available:
+
+    port	- port # for SB DSP 4.x chip (0x220,0x240,0x260)
+    mpu_port	- port # for MPU-401 UART (0x300,0x330), -1 = disable
+    awe_port	- base port # for EMU8000 synthesizer (0x620,0x640,0x660)
+                   (snd-sbawe module only)
+    irq		- IRQ # for SB DSP 4.x chip (5,7,9,10)
+    dma8	- 8-bit DMA # for SB DSP 4.x chip (0,1,3)
+    dma16	- 16-bit DMA # for SB DSP 4.x chip (5,6,7)
+    
+    This module supports multiple cards, autoprobe and ISA PnP.
+
+    Note: To use Vibra16X cards in 16-bit half duplex mode, you must
+          disable 16bit DMA with dma16 = -1 module parameter.
+          Also, all Sound Blaster 16 type cards can operate in 16-bit
+          half duplex mode through 8-bit DMA channel by disabling their
+          16-bit DMA channel.
+    
+    The power-management is supported.
+
+  Module snd-sc6000
+  -----------------
+
+    Module for Gallant SC-6000 soundcard and later models: SC-6600
+    and SC-7000.
+
+    port	- Port # (0x220 or 0x240)
+    mss_port	- MSS Port # (0x530 or 0xe80)
+    irq		- IRQ # (5,7,9,10,11)
+    mpu_irq	- MPU-401 IRQ # (5,7,9,10) ,0 - no MPU-401 irq
+    dma		- DMA # (1,3,0)
+    joystick	- Enable gameport - 0 = disable (default), 1 = enable
+
+    This module supports multiple cards.
+
+    This card is also known as Audio Excel DSP 16 or Zoltrix AV302.
+
+  Module snd-sgalaxy
+  ------------------
+
+    Module for Aztech Sound Galaxy sound card.
+
+    sbport	- Port # for SB16 interface (0x220,0x240)
+    wssport	- Port # for WSS interface (0x530,0xe80,0xf40,0x604)
+    irq		- IRQ # (7,9,10,11)
+    dma1	- DMA #
+
+    This module supports multiple cards.
+
+    The power-management is supported.
+
+  Module snd-sscape
+  -----------------
+
+    Module for ENSONIQ SoundScape cards.
+
+    port	- Port # (PnP setup)
+    wss_port	- WSS Port # (PnP setup)
+    irq		- IRQ # (PnP setup)
+    mpu_irq	- MPU-401 IRQ # (PnP setup)
+    dma		- DMA # (PnP setup)
+    dma2	- 2nd DMA # (PnP setup, -1 to disable)
+    joystick	- Enable gameport - 0 = disable (default), 1 = enable
+
+    This module supports multiple cards.
+
+    The driver requires the firmware loader support on kernel.
+
+  Module snd-sun-amd7930 (on sparc only)
+  --------------------------------------
+
+    Module for AMD7930 sound chips found on Sparcs.
+
+    This module supports multiple cards.
+
+  Module snd-sun-cs4231 (on sparc only)
+  -------------------------------------
+
+    Module for CS4231 sound chips found on Sparcs.
+
+    This module supports multiple cards.
+
+  Module snd-sun-dbri (on sparc only)
+  -----------------------------------
+
+    Module for DBRI sound chips found on Sparcs.
+
+    This module supports multiple cards.
+
+  Module snd-wavefront
+  --------------------
+
+    Module for Turtle Beach Maui, Tropez and Tropez+ sound cards.
+
+    use_cs4232_midi - Use CS4232 MPU-401 interface
+                      (inaccessibly located inside your computer)
+    isapnp          - ISA PnP detection - 0 = disable, 1 = enable (default)
+
+    with isapnp=0, the following options are available:
+
+    cs4232_pcm_port - Port # for CS4232 PCM interface.
+    cs4232_pcm_irq  - IRQ # for CS4232 PCM interface (5,7,9,11,12,15).
+    cs4232_mpu_port - Port # for CS4232 MPU-401 interface.
+    cs4232_mpu_irq  - IRQ # for CS4232 MPU-401 interface (9,11,12,15).
+    ics2115_port    - Port # for ICS2115
+    ics2115_irq     - IRQ # for ICS2115
+    fm_port         - FM OPL-3 Port #
+    dma1            - DMA1 # for CS4232 PCM interface.
+    dma2            - DMA2 # for CS4232 PCM interface.
+
+    The below are options for wavefront_synth features:
+    wf_raw	    - Assume that we need to boot the OS (default:no)
+	If yes, then during driver loading, the state of the board is
+	ignored, and we reset the board and load the firmware anyway.
+    fx_raw	    - Assume that the FX process needs help (default:yes)
+	If false, we'll leave the FX processor in whatever state it is
+	when the driver is loaded.  The default is to download the
+	microprogram and associated coefficients to set it up for
+	"default" operation, whatever that means.
+    debug_default   - Debug parameters for card initialization
+    wait_usecs	    - How long to wait without sleeping, usecs
+		      (default:150)
+	This magic number seems to give pretty optimal throughput
+	based on my limited experimentation. 
+	If you want to play around with it and find a better value, be
+	my guest. Remember, the idea is to get a number that causes us
+	to just busy wait for as many WaveFront commands as possible,
+	without coming up with a number so large that we hog the whole
+	CPU. 
+	Specifically, with this number, out of about 134,000 status
+	waits, only about 250 result in a sleep. 
+    sleep_interval  - How long to sleep when waiting for reply
+		      (default: 100)
+    sleep_tries	    - How many times to try sleeping during a wait
+		      (default: 50)
+    ospath	    - Pathname to processed ICS2115 OS firmware
+		      (default:wavefront.os)
+	The path name of the ISC2115 OS firmware.  In the recent
+	version, it's handled via firmware loader framework, so it
+	must be installed in the proper path, typically,
+	/lib/firmware.
+    reset_time	    - How long to wait for a reset to take effect
+		      (default:2)
+    ramcheck_time   - How many seconds to wait for the RAM test
+		      (default:20)
+    osrun_time	    - How many seconds to wait for the ICS2115 OS
+		      (default:10)
+
+    This module supports multiple cards and ISA PnP.
+
+    Note: the firmware file "wavefront.os" was located in the earlier
+          version in /etc.  Now it's loaded via firmware loader, and
+	  must be in the proper firmware path, such as /lib/firmware.
+	  Copy (or symlink) the file appropriately if you get an error
+	  regarding firmware downloading after upgrading the kernel.
+
+  Module snd-sonicvibes
+  ---------------------
+
+    Module for S3 SonicVibes PCI sound cards.
+			* PINE Schubert 32 PCI
+
+    reverb    - Reverb Enable - 1 = enable, 0 = disable (default)
+                  - SoundCard must have onboard SRAM for this.
+    mge       - Mic Gain Enable - 1 = enable, 0 = disable (default)
+    
+    This module supports multiple cards and autoprobe.
+
+  Module snd-serial-u16550
+  ------------------------
+
+    Module for UART16550A serial MIDI ports.
+
+    port	- port # for UART16550A chip
+    irq		- IRQ # for UART16550A chip, -1 = poll mode
+    speed	- speed in bauds (9600,19200,38400,57600,115200)
+		  38400 = default
+    base	- base for divisor in bauds (57600,115200,230400,460800)
+		  115200 = default
+    outs	- number of MIDI ports in a serial port (1-4)
+		  1 = default
+    adaptor	- Type of adaptor.
+                  0 = Soundcanvas, 1 = MS-124T, 2 = MS-124W S/A,
+		  3 = MS-124W M/B, 4 = Generic
+    
+    This module supports multiple cards. This module does not support autoprobe
+    thus the main port must be specified!!! Other options are optional.
+
+  Module snd-trident
+  ------------------
+
+    Module for Trident 4DWave DX/NX sound cards.
+			* Best Union  Miss Melody 4DWave PCI
+			* HIS  4DWave PCI
+			* Warpspeed  ONSpeed 4DWave PCI
+			* AzTech  PCI 64-Q3D
+			* Addonics  SV 750
+			* CHIC  True Sound 4Dwave
+			* Shark  Predator4D-PCI
+			* Jaton  SonicWave 4D
+			* SiS SI7018 PCI Audio
+			* Hoontech SoundTrack Digital 4DWave NX
+
+    pcm_channels   - max channels (voices) reserved for PCM
+    wavetable_size - max wavetable size in kB (4-?kb)
+
+    This module supports multiple cards and autoprobe.
+
+    The power-management is supported.
+
+  Module snd-ua101
+  ----------------
+
+    Module for the Edirol UA-101/UA-1000 audio/MIDI interfaces.
+
+    This module supports multiple devices, autoprobe and hotplugging.
+
+  Module snd-usb-audio
+  --------------------
+
+    Module for USB audio and USB MIDI devices.
+
+    vid             - Vendor ID for the device (optional)
+    pid             - Product ID for the device (optional)
+    nrpacks	    - Max. number of packets per URB (default: 8)
+    async_unlink    - Use async unlink mode (default: yes)
+    device_setup    - Device specific magic number (optional)
+                    - Influence depends on the device
+                    - Default: 0x0000 
+    ignore_ctl_error - Ignore any USB-controller regarding mixer
+    		       interface (default: no)
+
+    This module supports multiple devices, autoprobe and hotplugging.
+
+    NB: nrpacks parameter can be modified dynamically via sysfs.
+        Don't put the value over 20.  Changing via sysfs has no sanity
+	check.
+    NB: async_unlink=0 would cause Oops.  It remains just for
+        debugging purpose (if any).
+    NB: ignore_ctl_error=1 may help when you get an error at accessing
+        the mixer element such as URB error -22.  This happens on some
+        buggy USB device or the controller.
+
+  Module snd-usb-caiaq
+  --------------------
+
+    Module for caiaq UB audio interfaces,
+	    * Native Instruments RigKontrol2
+	    * Native Instruments Kore Controller
+	    * Native Instruments Audio Kontrol 1
+	    * Native Instruments Audio 8 DJ
+
+    This module supports multiple devices, autoprobe and hotplugging.
+
+  Module snd-usb-usx2y
+  --------------------
+
+    Module for Tascam USB US-122, US-224 and US-428 devices.
+
+    This module supports multiple devices, autoprobe and hotplugging.
+
+    Note: you need to load the firmware via usx2yloader utility included
+          in alsa-tools and alsa-firmware packages.
+
+  Module snd-via82xx
+  ------------------
+
+    Module for AC'97 motherboards based on VIA 82C686A/686B, 8233,
+    8233A, 8233C, 8235, 8237 (south) bridge.
+
+    mpu_port	- 0x300,0x310,0x320,0x330, otherwise obtain BIOS setup
+		  [VIA686A/686B only]
+    joystick	- Enable joystick (default off) [VIA686A/686B only]
+    ac97_clock	- AC'97 codec clock base (default 48000Hz)
+    dxs_support	- support DXS channels,
+		  0 = auto (default), 1 = enable, 2 = disable,
+		  3 = 48k only, 4 = no VRA, 5 = enable any sample
+		  rate and different sample rates on different
+		  channels
+		  [VIA8233/C, 8235, 8237 only]
+    ac97_quirk  - AC'97 workaround for strange hardware
+		  See "AC97 Quirk Option" section below.
+
+    This module supports one chip and autoprobe.
+
+    Note: on some SMP motherboards like MSI 694D the interrupts might
+          not be generated properly.  In such a case, please try to
+          set the SMP (or MPS) version on BIOS to 1.1 instead of
+          default value 1.4.  Then the interrupt number will be
+          assigned under 15. You might also upgrade your BIOS.
+    
+    Note: VIA8233/5/7 (not VIA8233A) can support DXS (direct sound)
+	  channels as the first PCM.  On these channels, up to 4
+	  streams can be played at the same time, and the controller
+	  can perform sample rate conversion with separate rates for
+	  each channel.
+	  As default (dxs_support = 0), 48k fixed rate is chosen
+	  except for the known devices since the output is often
+	  noisy except for 48k on some mother boards due to the
+	  bug of BIOS.
+	  Please try once dxs_support=5 and if it works on other
+	  sample rates (e.g. 44.1kHz of mp3 playback), please let us
+	  know the PCI subsystem vendor/device id's (output of
+	  "lspci -nv").
+	  If dxs_support=5 does not work, try dxs_support=4; if it
+	  doesn't work too, try dxs_support=1.  (dxs_support=1 is
+	  usually for old motherboards.  The correct implemented
+	  board should work with 4 or 5.)  If it still doesn't
+	  work and the default setting is ok, dxs_support=3 is the
+	  right choice.  If the default setting doesn't work at all,
+	  try dxs_support=2 to disable the DXS channels.
+	  In any cases, please let us know the result and the
+	  subsystem vendor/device ids.  See "Links and Addresses"
+	  below.
+
+    Note: for the MPU401 on VIA823x, use snd-mpu401 driver
+	  additionally.  The mpu_port option is for VIA686 chips only.
+
+    The power-management is supported.
+
+  Module snd-via82xx-modem
+  ------------------------
+
+    Module for VIA82xx AC97 modem
+
+    ac97_clock	- AC'97 codec clock base (default 48000Hz)
+
+    This module supports one card and autoprobe.
+
+    Note: The default index value of this module is -2, i.e. the first
+          slot is excluded.
+
+    The power-management is supported.
+
+  Module snd-virmidi
+  ------------------
+
+    Module for virtual rawmidi devices.
+    This module creates virtual rawmidi devices which communicate
+    to the corresponding ALSA sequencer ports.
+
+    midi_devs	- MIDI devices # (1-4, default=4)
+    
+    This module supports multiple cards.
+
+  Module snd-virtuoso
+  -------------------
+
+    Module for sound cards based on the Asus AV100/AV200 chips,
+    i.e., Xonar D1, DX, D2, D2X, DS, HDAV1.3 (Deluxe), Essence ST
+    (Deluxe) and Essence STX.
+
+    This module supports autoprobe and multiple cards.
+
+  Module snd-vx222
+  ----------------
+
+    Module for Digigram VX-Pocket VX222, V222 v2 and Mic cards.
+
+    mic		- Enable Microphone on V222 Mic (NYI)
+    ibl		- Capture IBL size. (default = 0, minimum size)
+
+    This module supports multiple cards.
+
+    When the driver is compiled as a module and the hotplug firmware
+    is supported, the firmware data is loaded via hotplug automatically.
+    Install the necessary firmware files in alsa-firmware package.
+    When no hotplug fw loader is available, you need to load the
+    firmware via vxloader utility in alsa-tools package.  To invoke
+    vxloader automatically, add the following to /etc/modprobe.conf
+
+	install snd-vx222 /sbin/modprobe --first-time -i snd-vx222 && /usr/bin/vxloader
+
+    (for 2.2/2.4 kernels, add "post-install /usr/bin/vxloader" to
+     /etc/modules.conf, instead.)
+    IBL size defines the interrupts period for PCM.  The smaller size
+    gives smaller latency but leads to more CPU consumption, too.
+    The size is usually aligned to 126.  As default (=0), the smallest
+    size is chosen.  The possible IBL values can be found in
+    /proc/asound/cardX/vx-status proc file.
+
+    The power-management is supported.
+
+  Module snd-vxpocket
+  -------------------
+
+    Module for Digigram VX-Pocket VX2 and 440 PCMCIA cards.
+
+    ibl      - Capture IBL size. (default = 0, minimum size)
+
+    This module supports multiple cards.  The module is compiled only when
+    PCMCIA is supported on kernel.
+
+    With the older 2.6.x kernel, to activate the driver via the card
+    manager, you'll need to set up /etc/pcmcia/vxpocket.conf.  See the
+    sound/pcmcia/vx/vxpocket.c.  2.6.13 or later kernel requires no
+    longer require a config file.
+
+    When the driver is compiled as a module and the hotplug firmware
+    is supported, the firmware data is loaded via hotplug automatically.
+    Install the necessary firmware files in alsa-firmware package.
+    When no hotplug fw loader is available, you need to load the
+    firmware via vxloader utility in alsa-tools package.
+
+    About capture IBL, see the description of snd-vx222 module.
+
+    Note: snd-vxp440 driver is merged to snd-vxpocket driver since
+           ALSA 1.0.10.
+
+    The power-management is supported.
+
+  Module snd-ymfpci
+  -----------------
+
+    Module for Yamaha PCI chips (YMF72x, YMF74x & YMF75x).
+
+    mpu_port      - 0x300,0x330,0x332,0x334, 0 (disable) by default,
+                    1 (auto-detect for YMF744/754 only)
+    fm_port       - 0x388,0x398,0x3a0,0x3a8, 0 (disable) by default
+                    1 (auto-detect for YMF744/754 only)
+    joystick_port - 0x201,0x202,0x204,0x205, 0 (disable) by default,
+                    1 (auto-detect)
+    rear_switch   - enable shared rear/line-in switch (bool)
+
+    This module supports autoprobe and multiple chips.
+
+    The power-management is supported.
+
+  Module snd-pdaudiocf
+  --------------------
+
+    Module for Sound Core PDAudioCF sound card.
+
+    The power-management is supported.
+
+
+AC97 Quirk Option
+=================
+
+The ac97_quirk option is used to enable/override the workaround for
+specific devices on drivers for on-board AC'97 controllers like
+snd-intel8x0.  Some hardware have swapped output pins between Master
+and Headphone, or Surround (thanks to confusion of AC'97
+specifications from version to version :-)
+
+The driver provides the auto-detection of known problematic devices,
+but some might be unknown or wrongly detected.  In such a case, pass
+the proper value with this option.
+
+The following strings are accepted:
+    - default	Don't override the default setting
+    - none	Disable the quirk
+    - hp_only	Bind Master and Headphone controls as a single control
+    - swap_hp	Swap headphone and master controls
+    - swap_surround  Swap master and surround controls
+    - ad_sharing  For AD1985, turn on OMS bit and use headphone
+    - alc_jack	For ALC65x, turn on the jack sense mode
+    - inv_eapd	Inverted EAPD implementation
+    - mute_led	Bind EAPD bit for turning on/off mute LED
+
+For backward compatibility, the corresponding integer value -1, 0,
+... are  accepted, too.
+
+For example, if "Master" volume control has no effect on your device
+but only "Headphone" does, pass ac97_quirk=hp_only module option.
+
+
+Configuring Non-ISAPNP Cards
+============================
+
+When the kernel is configured with ISA-PnP support, the modules
+supporting the isapnp cards will have module options "isapnp".
+If this option is set, *only* the ISA-PnP devices will be probed.
+For probing the non ISA-PnP cards, you have to pass "isapnp=0" option
+together with the proper i/o and irq configuration.
+
+When the kernel is configured without ISA-PnP support, isapnp option
+will be not built in.
+
+
+Module Autoloading Support
+==========================
+
+The ALSA drivers can be loaded automatically on demand by defining
+module aliases.  The string 'snd-card-%1' is requested for ALSA native
+devices where %i is sound card number from zero to seven.
+
+To auto-load an ALSA driver for OSS services, define the string
+'sound-slot-%i' where %i means the slot number for OSS, which
+corresponds to the card index of ALSA.  Usually, define this
+as the same card module.
+
+An example configuration for a single emu10k1 card is like below:
+----- /etc/modprobe.conf
+alias snd-card-0 snd-emu10k1
+alias sound-slot-0 snd-emu10k1
+----- /etc/modprobe.conf
+
+The available number of auto-loaded sound cards depends on the module
+option "cards_limit" of snd module.  As default it's set to 1.
+To enable the auto-loading of multiple cards, specify the number of
+sound cards in that option.
+
+When multiple cards are available, it'd better to specify the index
+number for each card via module option, too, so that the order of
+cards is kept consistent.
+
+An example configuration for two sound cards is like below:
+
+----- /etc/modprobe.conf
+# ALSA portion
+options snd cards_limit=2
+alias snd-card-0 snd-interwave
+alias snd-card-1 snd-ens1371
+options snd-interwave index=0
+options snd-ens1371 index=1
+# OSS/Free portion
+alias sound-slot-0 snd-interwave
+alias sound-slot-1 snd-ens1371
+----- /etc/modprobe.conf
+
+In this example, the interwave card is always loaded as the first card
+(index 0) and ens1371 as the second (index 1).
+
+Alternative (and new) way to fixate the slot assignment is to use
+"slots" option of snd module.  In the case above, specify like the
+following: 
+
+options snd slots=snd-interwave,snd-ens1371
+
+Then, the first slot (#0) is reserved for snd-interwave driver, and
+the second (#1) for snd-ens1371.  You can omit index option in each
+driver if slots option is used (although you can still have them at
+the same time as long as they don't conflict).
+
+The slots option is especially useful for avoiding the possible
+hot-plugging and the resultant slot conflict.  For example, in the
+case above again, the first two slots are already reserved.  If any
+other driver (e.g. snd-usb-audio) is loaded before snd-interwave or
+snd-ens1371, it will be assigned to the third or later slot.
+
+When a module name is given with '!', the slot will be given for any
+modules but that name.  For example, "slots=!snd-pcsp" will reserve
+the first slot for any modules but snd-pcsp. 
+
+
+ALSA PCM devices to OSS devices mapping
+=======================================
+
+/dev/snd/pcmC0D0[c|p]  -> /dev/audio0 (/dev/audio) -> minor 4
+/dev/snd/pcmC0D0[c|p]  -> /dev/dsp0 (/dev/dsp)     -> minor 3
+/dev/snd/pcmC0D1[c|p]  -> /dev/adsp0 (/dev/adsp)   -> minor 12
+/dev/snd/pcmC1D0[c|p]  -> /dev/audio1              -> minor 4+16 = 20
+/dev/snd/pcmC1D0[c|p]  -> /dev/dsp1                -> minor 3+16 = 19
+/dev/snd/pcmC1D1[c|p]  -> /dev/adsp1               -> minor 12+16 = 28
+/dev/snd/pcmC2D0[c|p]  -> /dev/audio2              -> minor 4+32 = 36
+/dev/snd/pcmC2D0[c|p]  -> /dev/dsp2                -> minor 3+32 = 39
+/dev/snd/pcmC2D1[c|p]  -> /dev/adsp2               -> minor 12+32 = 44
+
+The first number from /dev/snd/pcmC{X}D{Y}[c|p] expression means
+sound card number and second means device number.  The ALSA devices
+have either 'c' or 'p' suffix indicating the direction, capture and
+playback, respectively.
+
+Please note that the device mapping above may be varied via the module
+options of snd-pcm-oss module.
+
+
+Proc interfaces (/proc/asound)
+==============================
+
+/proc/asound/card#/pcm#[cp]/oss
+-------------------------------
+  String "erase" - erase all additional informations about OSS applications
+  String "<app_name> <fragments> <fragment_size> [<options>]"
+
+   <app_name> - name of application with (higher priority) or without path
+   <fragments> - number of fragments or zero if auto
+   <fragment_size> - size of fragment in bytes or zero if auto
+   <options> - optional parameters
+	  - disable   the application tries to open a pcm device for
+		      this channel but does not want to use it.
+		      (Cause a bug or mmap needs)
+		      It's good for Quake etc...
+	  - direct    don't use plugins
+	  - block     force block mode (rvplayer)
+	  - non-block force non-block mode
+	  - whole-frag  write only whole fragments (optimization affecting
+			playback only)
+	  - no-silence  do not fill silence ahead to avoid clicks
+	  - buggy-ptr	Returns the whitespace blocks in GETOPTR ioctl
+			instead of filled blocks
+
+  Example: echo "x11amp 128 16384" > /proc/asound/card0/pcm0p/oss
+           echo "squake 0 0 disable" > /proc/asound/card0/pcm0c/oss
+	   echo "rvplayer 0 0 block" > /proc/asound/card0/pcm0p/oss
+
+
+Early Buffer Allocation
+=======================
+
+Some drivers (e.g. hdsp) require the large contiguous buffers, and
+sometimes it's too late to find such spaces when the driver module is
+actually loaded due to memory fragmentation.  You can pre-allocate the
+PCM buffers by loading snd-page-alloc module and write commands to its
+proc file in prior, for example, in the early boot stage like
+/etc/init.d/*.local scripts.
+
+Reading the proc file /proc/drivers/snd-page-alloc shows the current
+usage of page allocation.  In writing, you can send the following
+commands to the snd-page-alloc driver:
+
+  - add VENDOR DEVICE MASK SIZE BUFFERS
+
+    VENDOR and DEVICE are PCI vendor and device IDs.  They take
+    integer numbers (0x prefix is needed for the hex).
+    MASK is the PCI DMA mask.  Pass 0 if not restricted.
+    SIZE is the size of each buffer to allocate.  You can pass
+    k and m suffix for KB and MB.  The max number is 16MB.
+    BUFFERS is the number of buffers to allocate.  It must be greater
+    than 0.  The max number is 4.
+
+  - erase
+
+    This will erase the all pre-allocated buffers which are not in
+    use.
+
+
+Links and Addresses
+===================
+
+  ALSA project homepage
+       http://www.alsa-project.org
+
+  Kernel Bugzilla
+       http://bugzilla.kernel.org/
+
+  ALSA Developers ML
+       mailto:alsa-devel@alsa-project.org
+
+  alsa-info.sh script
+       http://www.alsa-project.org/alsa-info.sh
diff --git a/Documentation/sound/alsa/Audigy-mixer.txt b/Documentation/sound/alsa/Audigy-mixer.txt
new file mode 100644
index 0000000..7f10dc6
--- /dev/null
+++ b/Documentation/sound/alsa/Audigy-mixer.txt
@@ -0,0 +1,345 @@
+
+		Sound Blaster Audigy mixer / default DSP code
+		===========================================
+
+This is based on SB-Live-mixer.txt.
+
+The EMU10K2 chips have a DSP part which can be programmed to support 
+various ways of sample processing, which is described here.
+(This article does not deal with the overall functionality of the 
+EMU10K2 chips. See the manuals section for further details.)
+
+The ALSA driver programs this portion of chip by default code
+(can be altered later) which offers the following functionality:
+
+
+1) Digital mixer controls
+-------------------------
+
+These controls are built using the DSP instructions. They offer extended
+functionality. Only the default build-in code in the ALSA driver is described
+here. Note that the controls work as attenuators: the maximum value is the 
+neutral position leaving the signal unchanged. Note that if the  same destination 
+is mentioned in multiple controls, the signal is accumulated and can be wrapped 
+(set to maximal or minimal value without checking of overflow).
+
+
+Explanation of used abbreviations:
+
+DAC    - digital to analog converter
+ADC    - analog to digital converter
+I2S    - one-way three wire serial bus for digital sound by Philips Semiconductors
+         (this standard is used for connecting standalone DAC and ADC converters)
+LFE    - low frequency effects (subwoofer signal)
+AC97   - a chip containing an analog mixer, DAC and ADC converters
+IEC958 - S/PDIF
+FX-bus - the EMU10K2 chip has an effect bus containing 64 accumulators.
+         Each of the synthesizer voices can feed its output to these accumulators
+         and the DSP microcontroller can operate with the resulting sum.
+
+name='PCM Front Playback Volume',index=0
+
+This control is used to attenuate samples for left and right front PCM FX-bus
+accumulators. ALSA uses accumulators 8 and 9 for left and right front PCM 
+samples for 5.1 playback. The result samples are forwarded to the front DAC PCM 
+slots of the Philips DAC.
+
+name='PCM Surround Playback Volume',index=0
+
+This control is used to attenuate samples for left and right surround PCM FX-bus
+accumulators. ALSA uses accumulators 2 and 3 for left and right surround PCM 
+samples for 5.1 playback. The result samples are forwarded to the surround DAC PCM 
+slots of the Philips DAC.
+
+name='PCM Center Playback Volume',index=0
+
+This control is used to attenuate samples for center PCM FX-bus accumulator.
+ALSA uses accumulator 6 for center PCM sample for 5.1 playback. The result sample
+is forwarded to the center DAC PCM slot of the Philips DAC.
+
+name='PCM LFE Playback Volume',index=0
+
+This control is used to attenuate sample for LFE PCM FX-bus accumulator. 
+ALSA uses accumulator 7 for LFE PCM sample for 5.1 playback. The result sample 
+is forwarded to the LFE DAC PCM slot of the Philips DAC.
+
+name='PCM Playback Volume',index=0
+
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples for
+stereo playback. The result samples are forwarded to the front DAC PCM slots 
+of the Philips DAC.
+
+name='PCM Capture Volume',index=0
+
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulator. ALSA uses accumulators 0 and 1 for left and right PCM.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Music Playback Volume',index=0
+
+This control is used to attenuate samples for left and right MIDI FX-bus
+accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples.
+The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
+
+name='Music Capture Volume',index=0
+
+These controls are used to attenuate samples for left and right MIDI FX-bus
+accumulator. ALSA uses accumulators 4 and 5 for left and right PCM.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Mic Playback Volume',index=0
+
+This control is used to attenuate samples for left and right Mic input.
+For Mic input is used AC97 codec. The result samples are forwarded to 
+the front DAC PCM slots of the Philips DAC. Samples are forwarded to Mic
+capture FIFO (device 1 - 16bit/8KHz mono) too without volume control.
+
+name='Mic Capture Volume',index=0
+
+This control is used to attenuate samples for left and right Mic input.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Audigy CD Playback Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 TTL
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the front DAC PCM slots of the Philips DAC.
+
+name='Audigy CD Capture Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 TTL
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the ADC capture FIFO (thus to the standard capture PCM device).
+
+name='IEC958 Optical Playback Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 optical
+digital input. The result samples are forwarded to the front DAC PCM slots
+of the Philips DAC.
+
+name='IEC958 Optical Capture Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 optical
+digital inputs. The result samples are forwarded to the ADC capture FIFO
+(thus to the standard capture PCM device).
+
+name='Line2 Playback Volume',index=0
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the AudigyDrive). The result samples are forwarded to the front
+DAC PCM slots of the Philips DAC.
+
+name='Line2 Capture Volume',index=1
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the AudigyDrive). The result samples are forwarded to the ADC
+capture FIFO (thus to the standard capture PCM device).
+
+name='Analog Mix Playback Volume',index=0
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs from Philips ADC. The result samples are forwarded to the front
+DAC PCM slots of the Philips DAC. This contains mix from analog sources
+like CD, Line In, Aux, ....
+
+name='Analog Mix Capture Volume',index=1
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs Philips ADC. The result samples are forwarded to the ADC
+capture FIFO (thus to the standard capture PCM device).
+
+name='Aux2 Playback Volume',index=0
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the AudigyDrive). The result samples are forwarded to the front
+DAC PCM slots of the Philips DAC.
+
+name='Aux2 Capture Volume',index=1
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the AudigyDrive). The result samples are forwarded to the ADC
+capture FIFO (thus to the standard capture PCM device).
+
+name='Front Playback Volume',index=0
+
+All stereo signals are mixed together and mirrored to surround, center and LFE.
+This control is used to attenuate samples for left and right front speakers of
+this mix.
+
+name='Surround Playback Volume',index=0
+
+All stereo signals are mixed together and mirrored to surround, center and LFE.
+This control is used to attenuate samples for left and right surround speakers of
+this mix.
+
+name='Center Playback Volume',index=0
+
+All stereo signals are mixed together and mirrored to surround, center and LFE.
+This control is used to attenuate sample for center speaker of this mix.
+
+name='LFE Playback Volume',index=0
+
+All stereo signals are mixed together and mirrored to surround, center and LFE.
+This control is used to attenuate sample for LFE speaker of this mix.
+
+name='Tone Control - Switch',index=0
+
+This control turns the tone control on or off. The samples for front, rear
+and center / LFE outputs are affected.
+
+name='Tone Control - Bass',index=0
+
+This control sets the bass intensity. There is no neutral value!!
+When the tone control code is activated, the samples are always modified.
+The closest value to pure signal is 20.
+
+name='Tone Control - Treble',index=0
+
+This control sets the treble intensity. There is no neutral value!!
+When the tone control code is activated, the samples are always modified.
+The closest value to pure signal is 20.
+
+name='Master Playback Volume',index=0
+
+This control is used to attenuate samples for front, surround, center and 
+LFE outputs.
+
+name='IEC958 Optical Raw Playback Switch',index=0
+
+If this switch is on, then the samples for the IEC958 (S/PDIF) digital
+output are taken only from the raw FX8010 PCM, otherwise standard front
+PCM samples are taken.
+
+
+2) PCM stream related controls
+------------------------------
+
+name='EMU10K1 PCM Volume',index 0-31
+
+Channel volume attenuation in range 0-0xffff. The maximum value (no
+attenuation) is default. The channel mapping for three values is
+as follows:
+
+	0 - mono, default 0xffff (no attenuation)
+	1 - left, default 0xffff (no attenuation)
+	2 - right, default 0xffff (no attenuation)
+
+name='EMU10K1 PCM Send Routing',index 0-31
+
+This control specifies the destination - FX-bus accumulators. There 24
+values with this mapping:
+
+	 0 -  mono, A destination (FX-bus 0-63), default 0
+	 1 -  mono, B destination (FX-bus 0-63), default 1
+	 2 -  mono, C destination (FX-bus 0-63), default 2
+	 3 -  mono, D destination (FX-bus 0-63), default 3
+	 4 -  mono, E destination (FX-bus 0-63), default 0
+	 5 -  mono, F destination (FX-bus 0-63), default 0
+	 6 -  mono, G destination (FX-bus 0-63), default 0
+	 7 -  mono, H destination (FX-bus 0-63), default 0
+	 8 -  left, A destination (FX-bus 0-63), default 0
+	 9 -  left, B destination (FX-bus 0-63), default 1
+	10 -  left, C destination (FX-bus 0-63), default 2
+	11 -  left, D destination (FX-bus 0-63), default 3
+	12 -  left, E destination (FX-bus 0-63), default 0
+	13 -  left, F destination (FX-bus 0-63), default 0
+	14 -  left, G destination (FX-bus 0-63), default 0
+	15 -  left, H destination (FX-bus 0-63), default 0
+	16 - right, A destination (FX-bus 0-63), default 0
+	17 - right, B destination (FX-bus 0-63), default 1
+	18 - right, C destination (FX-bus 0-63), default 2
+	19 - right, D destination (FX-bus 0-63), default 3
+	20 - right, E destination (FX-bus 0-63), default 0
+	21 - right, F destination (FX-bus 0-63), default 0
+	22 - right, G destination (FX-bus 0-63), default 0
+	23 - right, H destination (FX-bus 0-63), default 0
+
+Don't forget that it's illegal to assign a channel to the same FX-bus accumulator 
+more than once (it means 0=0 && 1=0 is an invalid combination).
+ 
+name='EMU10K1 PCM Send Volume',index 0-31
+
+It specifies the attenuation (amount) for given destination in range 0-255.
+The channel mapping is following:
+
+	 0 -  mono, A destination attn, default 255 (no attenuation)
+	 1 -  mono, B destination attn, default 255 (no attenuation)
+	 2 -  mono, C destination attn, default 0 (mute)
+	 3 -  mono, D destination attn, default 0 (mute)
+	 4 -  mono, E destination attn, default 0 (mute)
+	 5 -  mono, F destination attn, default 0 (mute)
+	 6 -  mono, G destination attn, default 0 (mute)
+	 7 -  mono, H destination attn, default 0 (mute)
+	 8 -  left, A destination attn, default 255 (no attenuation)
+	 9 -  left, B destination attn, default 0 (mute)
+	10 -  left, C destination attn, default 0 (mute)
+	11 -  left, D destination attn, default 0 (mute)
+	12 -  left, E destination attn, default 0 (mute)
+	13 -  left, F destination attn, default 0 (mute)
+	14 -  left, G destination attn, default 0 (mute)
+	15 -  left, H destination attn, default 0 (mute)
+	16 - right, A destination attn, default 0 (mute)
+	17 - right, B destination attn, default 255 (no attenuation)
+	18 - right, C destination attn, default 0 (mute)
+	19 - right, D destination attn, default 0 (mute)
+	20 - right, E destination attn, default 0 (mute)
+	21 - right, F destination attn, default 0 (mute)
+	22 - right, G destination attn, default 0 (mute)
+	23 - right, H destination attn, default 0 (mute)
+
+
+
+4) MANUALS/PATENTS:
+-------------------
+
+ftp://opensource.creative.com/pub/doc
+-------------------------------------
+
+        Files:
+        LM4545.pdf      AC97 Codec
+
+        m2049.pdf       The EMU10K1 Digital Audio Processor
+
+        hog63.ps        FX8010 - A DSP Chip Architecture for Audio Effects
+
+
+WIPO Patents
+------------
+        Patent numbers:
+        WO 9901813 (A1) Audio Effects Processor with multiple asynchronous (Jan. 14, 1999)
+                        streams
+
+        WO 9901814 (A1) Processor with Instruction Set for Audio Effects (Jan. 14, 1999)
+
+        WO 9901953 (A1) Audio Effects Processor having Decoupled Instruction
+                        Execution and Audio Data Sequencing (Jan. 14, 1999)
+
+
+US Patents (http://www.uspto.gov/)
+----------------------------------
+
+        US 5925841      Digital Sampling Instrument employing cache memory (Jul. 20, 1999)
+
+        US 5928342      Audio Effects Processor integrated on a single chip (Jul. 27, 1999)
+                        with a multiport memory onto which multiple asynchronous
+                        digital sound samples can be concurrently loaded
+
+        US 5930158      Processor with Instruction Set for Audio Effects (Jul. 27, 1999)
+
+        US 6032235      Memory initialization circuit (Tram) (Feb. 29, 2000)
+
+        US 6138207      Interpolation looping of audio samples in cache connected to    (Oct. 24, 2000)
+                        system bus with prioritization and modification of bus transfers
+                        in accordance with loop ends and minimum block sizes
+
+        US 6151670      Method for conserving memory storage using a (Nov. 21, 2000)
+                        pool of  short term memory registers
+
+        US 6195715      Interrupt control for multiple programs communicating with      (Feb. 27, 2001)
+                        a common interrupt by associating programs to GP registers,
+                        defining interrupt register, polling GP registers, and invoking
+                        callback routine associated with defined interrupt register
diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt
new file mode 100644
index 0000000..a4c53d8
--- /dev/null
+++ b/Documentation/sound/alsa/Audiophile-Usb.txt
@@ -0,0 +1,442 @@
+	Guide to using M-Audio Audiophile USB with ALSA and Jack	v1.5
+	========================================================
+
+	    Thibault Le Meur <Thibault.LeMeur@supelec.fr>
+
+This document is a guide to using the M-Audio Audiophile USB (tm) device with 
+ALSA and JACK.
+
+History
+=======
+* v1.4 - Thibault Le Meur (2007-07-11)
+ - Added Low Endianness nature of 16bits-modes
+   found by Hakan Lennestal <Hakan.Lennestal@brfsodrahamn.se>
+ - Modifying document structure
+* v1.5 - Thibault Le Meur (2007-07-12)
+ - Added AC3/DTS passthru info
+
+
+1 - Audiophile USB Specs and correct usage
+==========================================
+
+This part is a reminder of important facts about the functions and limitations 
+of the device.
+
+The device has 4 audio interfaces, and 2 MIDI ports:
+ * Analog Stereo Input (Ai)
+   - This port supports 2 pairs of line-level audio inputs (1/4" TS and RCA) 
+   - When the 1/4" TS (jack) connectors are connected, the RCA connectors
+     are disabled
+ * Analog Stereo Output (Ao)
+ * Digital Stereo Input (Di)
+ * Digital Stereo Output (Do)
+ * Midi In (Mi)
+ * Midi Out (Mo)
+
+The internal DAC/ADC has the following characteristics:
+* sample depth of 16 or 24 bits
+* sample rate from 8kHz to 96kHz
+* Two interfaces can't use different sample depths at the same time.
+Moreover, the Audiophile USB documentation gives the following Warning:
+"Please exit any audio application running before switching between bit depths"
+
+Due to the USB 1.1 bandwidth limitation, a limited number of interfaces can be 
+activated at the same time depending on the audio mode selected:
+ * 16-bit/48kHz ==> 4 channels in + 4 channels out
+   - Ai+Ao+Di+Do
+ * 24-bit/48kHz ==> 4 channels in + 2 channels out, 
+                    or 2 channels in + 4 channels out
+   - Ai+Ao+Do or Ai+Di+Ao or Ai+Di+Do or Di+Ao+Do
+ * 24-bit/96kHz ==> 2 channels in _or_ 2 channels out (half duplex only)
+   - Ai or Ao or Di or Do
+
+Important facts about the Digital interface:
+--------------------------------------------
+ * The Do port additionally supports surround-encoded AC-3 and DTS passthrough, 
+though I haven't tested it under Linux
+   - Note that in this setup only the Do interface can be enabled
+ * Apart from recording an audio digital stream, enabling the Di port is a way 
+to synchronize the device to an external sample clock
+   - As a consequence, the Di port must be enable only if an active Digital 
+source is connected
+   - Enabling Di when no digital source is connected can result in a 
+synchronization error (for instance sound played at an odd sample rate)
+
+
+2 - Audiophile USB MIDI support in ALSA
+=======================================
+
+The Audiophile USB MIDI ports will be automatically supported once the
+following modules have been loaded:
+ * snd-usb-audio
+ * snd-seq-midi
+
+No additional setting is required.
+
+
+3 - Audiophile USB Audio support in ALSA
+========================================
+
+Audio functions of the Audiophile USB device are handled by the snd-usb-audio 
+module. This module can work in a default mode (without any device-specific 
+parameter), or in an "advanced" mode with the device-specific parameter called 
+"device_setup".
+
+3.1 - Default Alsa driver mode
+------------------------------
+
+The default behavior of the snd-usb-audio driver is to list the device 
+capabilities at startup and activate the required mode when required 
+by the applications: for instance if the user is recording in a 
+24bit-depth-mode and immediately after wants to switch to a 16bit-depth mode,
+the snd-usb-audio module will reconfigure the device on the fly.
+
+This approach has the advantage to let the driver automatically switch from sample 
+rates/depths automatically according to the user's needs. However, those who 
+are using the device under windows know that this is not how the device is meant to
+work: under windows applications must be closed before using the m-audio control
+panel to switch the device working mode. Thus as we'll see in next section, this 
+Default Alsa driver mode can lead to device misconfigurations.
+
+Let's get back to the Default Alsa driver mode for now.  In this case the 
+Audiophile interfaces are mapped to alsa pcm devices in the following 
+way (I suppose the device's index is 1):
+ * hw:1,0 is Ao in playback and Di in capture
+ * hw:1,1 is Do in playback and Ai in capture
+ * hw:1,2 is Do in AC3/DTS passthrough mode
+
+In this mode, the device uses Big Endian byte-encoding so that 
+supported audio format are S16_BE for 16-bit depth modes and S24_3BE for 
+24-bits depth mode.
+
+One exception is the hw:1,2 port which was reported to be Little Endian 
+compliant (supposedly supporting S16_LE) but processes in fact only S16_BE streams.
+This has been fixed in kernel 2.6.23 and above and now the hw:1,2 interface 
+is reported to be big endian in this default driver mode.
+
+Examples:
+ * playing a S24_3BE encoded raw file to the Ao port
+   % aplay -D hw:1,0 -c2 -t raw -r48000 -fS24_3BE test.raw
+ * recording a  S24_3BE encoded raw file from the Ai port
+   % arecord -D hw:1,1 -c2  -t raw -r48000 -fS24_3BE test.raw
+ * playing a S16_BE encoded raw file to the Do port
+   % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test.raw
+ * playing an ac3 sample file to the Do port
+   % aplay -D hw:1,2 --channels=6 ac3_S16_BE_encoded_file.raw
+
+If you're happy with the default Alsa driver mode and don't experience any 
+issue with this mode, then you can skip the following chapter.
+
+3.2 - Advanced module setup
+---------------------------
+
+Due to the hardware constraints described above, the device initialization made 
+by the Alsa driver in default mode may result in a corrupted state of the 
+device. For instance, a particularly annoying issue is that the sound captured 
+from the Ai interface sounds distorted (as if boosted with an excessive high
+volume gain).
+
+For people having this problem, the snd-usb-audio module has a new module 
+parameter called "device_setup" (this parameter was introduced in kernel
+release 2.6.17)
+
+3.2.1 - Initializing the working mode of the Audiophile USB
+
+As far as the Audiophile USB device is concerned, this value let the user 
+specify:
+ * the sample depth
+ * the sample rate
+ * whether the Di port is used or not 
+
+When initialized with "device_setup=0x00", the snd-usb-audio module has
+the same behaviour as when the parameter is omitted (see paragraph "Default 
+Alsa driver mode" above)
+
+Others modes are described in the following subsections.
+
+3.2.1.1 - 16-bit modes
+
+The two supported modes are:
+
+ * device_setup=0x01
+   - 16bits 48kHz mode with Di disabled
+   - Ai,Ao,Do can be used at the same time
+   - hw:1,0 is not available in capture mode
+   - hw:1,2 is not available
+
+ * device_setup=0x11
+   - 16bits 48kHz mode with Di enabled
+   - Ai,Ao,Di,Do can be used at the same time
+   - hw:1,0 is available in capture mode
+   - hw:1,2 is not available
+
+In this modes the device operates only at 16bits-modes. Before kernel 2.6.23,
+the devices where reported to be Big-Endian when in fact they were Little-Endian
+so that playing a file was a matter of using:
+   % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test_S16_LE.raw
+where "test_S16_LE.raw" was in fact a little-endian sample file.
+
+Thanks to Hakan Lennestal (who discovered the Little-Endiannes of the device in
+these modes) a fix has been committed (expected in kernel 2.6.23) and
+Alsa now reports Little-Endian interfaces. Thus playing a file now is as simple as
+using:
+   % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_LE test_S16_LE.raw
+
+3.2.1.2 - 24-bit modes
+
+The three supported modes are:
+
+ * device_setup=0x09
+   - 24bits 48kHz mode with Di disabled
+   - Ai,Ao,Do can be used at the same time
+   - hw:1,0 is not available in capture mode
+   - hw:1,2 is not available
+
+ * device_setup=0x19
+   - 24bits 48kHz mode with Di enabled
+   - 3 ports from {Ai,Ao,Di,Do} can be used at the same time
+   - hw:1,0 is available in capture mode and an active digital source must be 
+     connected to Di
+   - hw:1,2 is not available
+
+ * device_setup=0x0D or 0x10
+   - 24bits 96kHz mode
+   - Di is enabled by default for this mode but does not need to be connected 
+     to an active source
+   - Only 1 port from {Ai,Ao,Di,Do} can be used at the same time
+   - hw:1,0 is available in captured mode
+   - hw:1,2 is not available
+
+In these modes the device is only Big-Endian compliant (see "Default Alsa driver 
+mode" above for an aplay command example)
+
+3.2.1.3 - AC3 w/ DTS passthru mode
+
+Thanks to Hakan Lennestal, I now have a report saying that this mode works.
+
+ * device_setup=0x03
+   - 16bits 48kHz mode with only the Do port enabled 
+   - AC3 with DTS passthru
+   - Caution with this setup the Do port is mapped to the pcm device hw:1,0
+
+The command line used to playback the AC3/DTS encoded .wav-files in this mode:
+   % aplay -D hw:1,0 --channels=6 ac3_S16_LE_encoded_file.raw
+
+3.2.2 - How to use the device_setup parameter
+----------------------------------------------
+
+The parameter can be given:
+
+ * By manually probing the device (as root):
+   # modprobe -r snd-usb-audio
+   # modprobe snd-usb-audio index=1 device_setup=0x09
+
+ * Or while configuring the modules options in your modules configuration file
+   - For Fedora distributions, edit the /etc/modprobe.conf file:
+       alias snd-card-1 snd-usb-audio
+       options snd-usb-audio index=1 device_setup=0x09
+
+CAUTION when initializing the device
+-------------------------------------
+
+ * Correct initialization on the device requires that device_setup is given to
+   the module BEFORE the device is turned on. So, if you use the "manual probing"
+   method described above, take care to power-on the device AFTER this initialization.
+
+ * Failing to respect this will lead to a misconfiguration of the device. In this case
+   turn off the device, unprobe the snd-usb-audio module, then probe it again with
+   correct device_setup parameter and then (and only then) turn on the device again.
+
+ * If you've correctly initialized the device in a valid mode and then want to switch
+   to  another mode (possibly with another sample-depth), please use also the following 
+   procedure:
+   - first turn off the device
+   - de-register the snd-usb-audio module (modprobe -r)
+   - change the device_setup parameter by changing the device_setup
+     option in /etc/modprobe.conf 
+   - turn on the device
+ * A workaround for this last issue has been applied to kernel 2.6.23, but it may not
+   be enough to ensure the 'stability' of the device initialization.
+
+3.2.3 - Technical details for hackers
+-------------------------------------
+This section is for hackers, wanting to understand details about the device
+internals and how Alsa supports it.
+
+3.2.3.1 - Audiophile USB's device_setup structure
+
+If you want to understand the device_setup magic numbers for the Audiophile 
+USB, you need some very basic understanding of binary computation. However, 
+this is not required to use the parameter and you may skip this section.
+
+The device_setup is one byte long and its structure is the following:
+
+       +---+---+---+---+---+---+---+---+
+       | b7| b6| b5| b4| b3| b2| b1| b0|
+       +---+---+---+---+---+---+---+---+
+       | 0 | 0 | 0 | Di|24B|96K|DTS|SET|
+       +---+---+---+---+---+---+---+---+
+
+Where:
+ * b0 is the "SET" bit
+   - it MUST be set if device_setup is initialized 
+ * b1 is the "DTS" bit
+   - it is set only for Digital output with DTS/AC3
+   - this setup is not tested
+ * b2 is the Rate selection flag
+   - When set to "1" the rate range is 48.1-96kHz
+   - Otherwise the sample rate range is 8-48kHz
+ * b3 is the bit depth selection flag
+   - When set to "1" samples are 24bits long
+   - Otherwise they are 16bits long
+   - Note that b2 implies b3 as the 96kHz mode is only supported for 24 bits 
+     samples
+ * b4 is the Digital input flag
+   - When set to "1" the device assumes that an active digital source is 
+     connected 
+   - You shouldn't enable Di if no source is seen on the port (this leads to 
+     synchronization issues)
+   - b4 is implied by b2 (since only one port is enabled at a time no synch 
+     error can occur) 
+ * b5 to b7 are reserved for future uses, and must be set to "0"
+   - might become Ao, Do, Ai, for b7, b6, b4 respectively
+
+Caution:
+ * there is no check on the value you will give to device_setup
+   - for instance choosing 0x05 (16bits 96kHz) will fail back to 0x09 since 
+     b2 implies b3. But _there_will_be_no_warning_ in /var/log/messages
+ * Hardware constraints due to the USB bus limitation aren't checked
+   - choosing b2 will prepare all interfaces for 24bits/96kHz but you'll
+     only be able to use one at the same time
+
+3.2.3.2 -  USB implementation details for this device
+
+You may safely skip this section if you're not interested in driver 
+hacking.
+
+This section describes some internal aspects of the device and summarizes the 
+data I got by usb-snooping the windows and Linux drivers.
+
+The M-Audio Audiophile USB has 7 USB Interfaces:
+a "USB interface":
+ * USB Interface nb.0
+ * USB Interface nb.1
+   - Audio Control function
+ * USB Interface nb.2
+   - Analog Output
+ * USB Interface nb.3
+   - Digital Output
+ * USB Interface nb.4
+   - Analog Input
+ * USB Interface nb.5
+   - Digital Input
+ * USB Interface nb.6
+   - MIDI interface compliant with the MIDIMAN quirk 
+
+Each interface has 5 altsettings (AltSet 1,2,3,4,5) except:
+ * Interface 3 (Digital Out) has an extra Alset nb.6 
+ * Interface 5 (Digital In) does not have Alset nb.3 and 5 
+
+Here is a short description of the AltSettings capabilities:
+ * AltSettings 1 corresponds to
+  - 24-bit depth, 48.1-96kHz sample mode
+  - Adaptive playback (Ao and Do), Synch capture (Ai), or Asynch capture (Di)
+ * AltSettings 2 corresponds to
+  - 24-bit depth, 8-48kHz sample mode
+  - Asynch capture and playback  (Ao,Ai,Do,Di)
+ * AltSettings 3 corresponds to
+  - 24-bit depth, 8-48kHz sample mode
+  - Synch capture (Ai) and Adaptive playback (Ao,Do)
+ * AltSettings 4 corresponds to
+  - 16-bit depth, 8-48kHz sample mode
+  - Asynch capture and playback  (Ao,Ai,Do,Di)
+ * AltSettings 5 corresponds to
+  - 16-bit depth, 8-48kHz sample mode
+  - Synch capture (Ai) and Adaptive playback (Ao,Do)
+ * AltSettings 6 corresponds to
+  - 16-bit depth, 8-48kHz sample mode
+  - Synch playback (Do), audio format type III IEC1937_AC-3
+
+In order to ensure a correct initialization of the device, the driver 
+_must_know_ how the device will be used:
+ * if DTS is chosen, only Interface 2 with AltSet nb.6 must be
+   registered
+ * if 96KHz only AltSets nb.1 of each interface must be selected
+ * if samples are using 24bits/48KHz then AltSet 2 must me used if
+   Digital input is connected, and only AltSet nb.3 if Digital input
+   is not connected
+ * if samples are using 16bits/48KHz then AltSet 4 must me used if
+   Digital input is connected, and only AltSet nb.5 if Digital input
+   is not connected
+
+When device_setup is given as a parameter to the snd-usb-audio module, the 
+parse_audio_endpoints function uses a quirk called 
+"audiophile_skip_setting_quirk" in order to prevent AltSettings not 
+corresponding to device_setup from being registered in the driver.
+
+4 - Audiophile USB and Jack support
+===================================
+
+This section deals with support of the Audiophile USB device in Jack.
+
+There are 2 main potential issues when using Jackd with the device:
+* support for Big-Endian devices in 24-bit modes
+* support for 4-in / 4-out channels
+
+4.1 - Direct support in Jackd
+-----------------------------
+
+Jack supports big endian devices only in recent versions (thanks to
+Andreas Steinmetz for his first big-endian patch). I can't remember 
+exactly when this support was released into jackd, let's just say that
+with jackd version 0.103.0 it's almost ok (just a small bug is affecting 
+16bits Big-Endian devices, but since you've read carefully the above
+paragraphs, you're now using kernel >= 2.6.23 and your 16bits devices 
+are now Little Endians ;-) ).
+
+You can run jackd with the following command for playback with Ao and
+record with Ai:
+  % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1
+
+4.2 - Using Alsa plughw
+-----------------------
+If you don't have a recent Jackd installed, you can downgrade to using
+the Alsa "plug" converter.
+
+For instance here is one way to run Jack with 2 playback channels on Ao and 2 
+capture channels from Ai:
+  % jackd -R -dalsa -dplughw:1 -r48000 -p256 -n2 -D -Cplughw:1,1
+
+However you may see the following warning message:
+"You appear to be using the ALSA software "plug" layer, probably a result of 
+using the "default" ALSA device. This is less efficient than it could be. 
+Consider using a hardware device instead rather than using the plug layer."
+
+4.3 - Getting 2 input and/or output interfaces in Jack
+------------------------------------------------------
+
+As you can see, starting the Jack server this way will only enable 1 stereo
+input (Di or Ai) and 1 stereo output (Ao or Do).
+
+This is due to the following restrictions:
+* Jack can only open one capture device and one playback device at a time
+* The Audiophile USB is seen as 2 (or three) Alsa devices: hw:1,0, hw:1,1
+  (and optionally hw:1,2)
+
+If you want to get Ai+Di and/or Ao+Do support with Jack, you would need to
+combine the Alsa devices into one logical "complex" device.
+
+If you want to give it a try, I recommend reading the information from
+this page: http://www.sound-man.co.uk/linuxaudio/ice1712multi.html
+It is related to another device (ice1712) but can be adapted to suit
+the Audiophile USB.
+
+Enabling multiple Audiophile USB interfaces for Jackd will certainly require:
+* Making sure your Jackd version has the MMAP_COMPLEX patch (see the ice1712 page)
+* (maybe) patching the alsa-lib/src/pcm/pcm_multi.c file (see the ice1712 page)
+* define a multi device (combination of hw:1,0 and hw:1,1) in your .asoundrc
+  file 
+* start jackd with this device
+
+I had no success in testing this for now, if you have any success with this kind 
+of setup, please drop me an email.
diff --git a/Documentation/sound/alsa/Bt87x.txt b/Documentation/sound/alsa/Bt87x.txt
new file mode 100644
index 0000000..f158cde
--- /dev/null
+++ b/Documentation/sound/alsa/Bt87x.txt
@@ -0,0 +1,78 @@
+Intro
+=====
+
+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.  snd-bt87x 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...
+
+Some cards do not bother to connect anything to the audio input pins of
+the chip, and some other cards use the audio function to transport MPEG
+video data, so it's quite possible that audio recording may not work
+with your card.
+
+
+Driver Status
+=============
+
+The driver is now stable.  However, it doesn't know about many TV cards,
+and it refuses to load for cards it doesn't know.
+
+If the driver complains ("Unknown TV card found, the audio driver will
+not load"), you can specify the load_all=1 option to force the driver to
+try to use the audio capture function of your card.  If the frequency of
+recorded data is not right, try to specify the digital_rate option with
+other values than the default 32000 (often it's 44100 or 64000).
+
+If you have an unknown card, please mail the ID and board name to
+<alsa-devel@alsa-project.org>, regardless of whether audio capture works
+or not, so that future versions of this driver know about your card.
+
+
+Audio modes
+===========
+
+The chip knows two different modes (digital/analog).  snd-bt87x
+registers two PCM devices, one for each mode.  They cannot be used at
+the same time.
+
+
+Digital audio mode
+==================
+
+The first device (hw:X,0) gives you 16 bit stereo sound.  The sample
+rate depends on the external source which feeds the Bt87x with digital
+sound via I2S interface.
+
+
+Analog audio mode (A/D)
+=======================
+
+The second device (hw:X,1) gives you 8 or 16 bit mono sound.  Supported
+sample rates are between 119466 and 448000 Hz (yes, these numbers are
+that high).  If you've set the CONFIG_SND_BT87X_OVERCLOCK option, the
+maximum sample rate is 1792000 Hz, but audio data becomes unusable
+beyond 896000 Hz on my card.
+
+The chip has three analog inputs.  Consequently you'll get a mixer
+device to control these.
+
+
+Have fun,
+
+  Clemens
+
+
+Written by Clemens Ladisch <clemens@ladisch.de>
+big parts copied from btaudio.txt by Gerd Knorr <kraxel@bytesex.org>
diff --git a/Documentation/sound/alsa/CMIPCI.txt b/Documentation/sound/alsa/CMIPCI.txt
new file mode 100644
index 0000000..16935c8
--- /dev/null
+++ b/Documentation/sound/alsa/CMIPCI.txt
@@ -0,0 +1,254 @@
+         Brief Notes on C-Media 8338/8738/8768/8770 Driver
+         =================================================
+
+                   Takashi Iwai <tiwai@suse.de>
+
+
+Front/Rear Multi-channel Playback
+---------------------------------
+
+CM8x38 chip can use ADC as the second DAC so that two different stereo
+channels can be used for front/rear playbacks.  Since there are two
+DACs, both streams are handled independently unlike the 4/6ch multi-
+channel playbacks in the section below.
+
+As default, ALSA driver assigns the first PCM device (i.e. hw:0,0 for
+card#0) for front and 4/6ch playbacks, while the second PCM device
+(hw:0,1) is assigned to the second DAC for rear playback.
+
+There are slight differences between the two DACs:
+
+- The first DAC supports U8 and S16LE formats, while the second DAC
+  supports only S16LE.
+- The second DAC supports only two channel stereo.
+
+Please note that the CM8x38 DAC doesn't support continuous playback
+rate but only fixed rates: 5512, 8000, 11025, 16000, 22050, 32000,
+44100 and 48000 Hz.
+
+The rear output can be heard only when "Four Channel Mode" switch is
+disabled.  Otherwise no signal will be routed to the rear speakers.
+As default it's turned on.
+
+*** WARNING ***
+When "Four Channel Mode" switch is off, the output from rear speakers
+will be FULL VOLUME regardless of Master and PCM volumes.
+This might damage your audio equipment.  Please disconnect speakers
+before your turn off this switch.
+*** WARNING ***
+
+[ Well.. I once got the output with correct volume (i.e. same with the
+  front one) and was so excited.  It was even with "Four Channel" bit
+  on and "double DAC" mode.  Actually I could hear separate 4 channels
+  from front and rear speakers!  But.. after reboot, all was gone.
+  It's a very pity that I didn't save the register dump at that
+  time..  Maybe there is an unknown register to achieve this... ]
+
+If your card has an extra output jack for the rear output, the rear
+playback should be routed there as default.  If not, there is a
+control switch in the driver "Line-In As Rear", which you can change
+via alsamixer or somewhat else.  When this switch is on, line-in jack
+is used as rear output.
+
+There are two more controls regarding to the rear output.
+The "Exchange DAC" switch is used to exchange front and rear playback
+routes, i.e. the 2nd DAC is output from front output.
+
+
+4/6 Multi-Channel Playback
+--------------------------
+
+The recent CM8738 chips support for the 4/6 multi-channel playback
+function.  This is useful especially for AC3 decoding.
+
+When the multi-channel is supported, the driver name has a suffix
+"-MC" such like "CMI8738-MC6".  You can check this name from
+/proc/asound/cards.
+
+When the 4/6-ch output is enabled, the second DAC accepts up to 6 (or
+4) channels.  While the dual DAC supports two different rates or
+formats, the 4/6-ch playback supports only the same condition for all
+channels.  Since the multi-channel playback mode uses both DACs, you
+cannot operate with full-duplex.
+
+The 4.0 and 5.1 modes are defined as the pcm "surround40" and "surround51"
+in alsa-lib.  For example, you can play a WAV file with 6 channels like
+
+	% aplay -Dsurround51 sixchannels.wav
+
+For programming the 4/6 channel playback, you need to specify the PCM
+channels as you like and set the format S16LE.  For example, for playback
+with 4 channels,
+
+	snd_pcm_hw_params_set_access(pcm, hw, SND_PCM_ACCESS_RW_INTERLEAVED);
+	    // or mmap if you like
+	snd_pcm_hw_params_set_format(pcm, hw, SND_PCM_FORMAT_S16_LE);
+	snd_pcm_hw_params_set_channels(pcm, hw, 4);
+
+and use the interleaved 4 channel data.
+
+There are some control switchs affecting to the speaker connections:
+
+"Line-In Mode"	- an enum control to change the behavior of line-in
+	jack.  Either "Line-In", "Rear Output" or "Bass Output" can
+	be selected.  The last item is available only with model 039
+	or newer. 
+	When "Rear Output" is chosen, the surround channels 3 and 4
+	are output to line-in jack.
+"Mic-In Mode"	- an enum control to change the behavior of mic-in
+	jack.  Either "Mic-In" or "Center/LFE Output" can be
+	selected. 
+	When "Center/LFE Output" is chosen, the center and bass
+	channels (channels 5 and 6) are output to mic-in jack. 
+
+Digital I/O
+-----------
+
+The CM8x38 provides the excellent SPDIF capability with very cheap
+price (yes, that's the reason I bought the card :)
+
+The SPDIF playback and capture are done via the third PCM device
+(hw:0,2).  Usually this is assigned to the PCM device "spdif".
+The available rates are 44100 and 48000 Hz.
+For playback with aplay, you can run like below:
+
+	% aplay -Dhw:0,2 foo.wav
+
+or
+
+	% aplay -Dspdif foo.wav
+
+24bit format is also supported experimentally.
+
+The playback and capture over SPDIF use normal DAC and ADC,
+respectively, so you cannot playback both analog and digital streams
+simultaneously.
+
+To enable SPDIF output, you need to turn on "IEC958 Output Switch"
+control via mixer or alsactl ("IEC958" is the official name of
+so-called S/PDIF).  Then you'll see the red light on from the card so
+you know that's working obviously :)
+The SPDIF input is always enabled, so you can hear SPDIF input data
+from line-out with "IEC958 In Monitor" switch at any time (see
+below).
+
+You can play via SPDIF even with the first device (hw:0,0),
+but SPDIF is enabled only when the proper format (S16LE), sample rate
+(441100 or 48000) and channels (2) are used.  Otherwise it's turned
+off.  (Also don't forget to turn on "IEC958 Output Switch", too.)
+
+
+Additionally there are relevant control switches:
+
+"IEC958 Mix Analog" - Mix analog PCM playback and FM-OPL/3 streams and
+	output through SPDIF.  This switch appears only on old chip
+	models (CM8738 033 and 037).
+	Note: without this control you can output PCM to SPDIF.
+	This is "mixing" of streams, so e.g. it's not for AC3 output
+	(see the next section).
+
+"IEC958 In Select"  - Select SPDIF input, the internal CD-in (false)
+	and the external input (true).
+
+"IEC958 Loop"       - SPDIF input data is loop back into SPDIF
+	output (aka bypass)
+
+"IEC958 Copyright"  - Set the copyright bit.
+
+"IEC958 5V"         - Select 0.5V (coax) or 5V (optical) interface.
+	On some cards this doesn't work and you need to change the
+	configuration with hardware dip-switch.
+
+"IEC958 In Monitor" - SPDIF input is routed to DAC.
+
+"IEC958 In Phase Inverse" - Set SPDIF input format as inverse.
+	[FIXME: this doesn't work on all chips..]
+
+"IEC958 In Valid"   - Set input validity flag detection.
+
+Note: When "PCM Playback Switch" is on, you'll hear the digital output
+stream through analog line-out.
+
+
+The AC3 (RAW DIGITAL) OUTPUT
+----------------------------
+
+The driver supports raw digital (typically AC3) i/o over SPDIF.  This
+can be toggled via IEC958 playback control, but usually you need to
+access it via alsa-lib.  See alsa-lib documents for more details.
+
+On the raw digital mode, the "PCM Playback Switch" is automatically
+turned off so that non-audio data is heard from the analog line-out.
+Similarly the following switches are off: "IEC958 Mix Analog" and
+"IEC958 Loop".  The switches are resumed after closing the SPDIF PCM
+device automatically to the previous state.
+
+On the model 033, AC3 is implemented by the software conversion in
+the alsa-lib.  If you need to bypass the software conversion of IEC958
+subframes, pass the "soft_ac3=0" module option.  This doesn't matter
+on the newer models.
+
+
+ANALOG MIXER INTERFACE
+----------------------
+
+The mixer interface on CM8x38 is similar to SB16.
+There are Master, PCM, Synth, CD, Line, Mic and PC Speaker playback
+volumes.  Synth, CD, Line and Mic have playback and capture switches,
+too, as well as SB16.
+
+In addition to the standard SB mixer, CM8x38 provides more functions.
+- PCM playback switch
+- PCM capture switch (to capture the data sent to DAC)
+- Mic Boost switch
+- Mic capture volume
+- Aux playback volume/switch and capture switch
+- 3D control switch
+
+
+MIDI CONTROLLER
+---------------
+
+With CMI8338 chips, the MPU401-UART interface is disabled as default.
+You need to set the module option "mpu_port" to a valid I/O port address
+to enable MIDI support.  Valid I/O ports are 0x300, 0x310, 0x320 and
+0x330.  Choose a value that doesn't conflict with other cards.
+
+With CMI8738 and newer chips, the MIDI interface is enabled by default
+and the driver automatically chooses a port address.
+
+There is _no_ hardware wavetable function on this chip (except for
+OPL3 synth below).
+What's said as MIDI synth on Windows is a software synthesizer
+emulation.  On Linux use TiMidity or other softsynth program for
+playing MIDI music.
+
+
+FM OPL/3 Synth
+--------------
+
+The FM OPL/3 is also enabled as default only for the first card.
+Set "fm_port" module option for more cards.
+
+The output quality of FM OPL/3 is, however, very weird.
+I don't know why..
+
+CMI8768 and newer chips do not have the FM synth.
+
+
+Joystick and Modem
+------------------
+
+The legacy joystick is supported.  To enable the joystick support, pass
+joystick_port=1 module option.  The value 1 means the auto-detection.
+If the auto-detection fails, try to pass the exact I/O address.
+
+The modem is enabled dynamically via a card control switch "Modem".
+
+
+Debugging Information
+---------------------
+
+The registers are shown in /proc/asound/cardX/cmipci.  If you have any
+problem (especially unexpected behavior of mixer), please attach the
+output of this proc file together with the bug report.
diff --git a/Documentation/sound/alsa/ControlNames.txt b/Documentation/sound/alsa/ControlNames.txt
new file mode 100644
index 0000000..fea65bb
--- /dev/null
+++ b/Documentation/sound/alsa/ControlNames.txt
@@ -0,0 +1,85 @@
+This document describes standard names of mixer controls.
+
+Syntax: SOURCE [DIRECTION] FUNCTION
+
+DIRECTION:
+  <nothing>	(both directions)
+  Playback
+  Capture
+  Bypass Playback
+  Bypass Capture
+
+FUNCTION:
+  Switch	(on/off switch)
+  Volume
+  Route		(route control, hardware specific)
+
+SOURCE:
+  Master
+  Master Mono
+  Hardware Master
+  Speaker	(internal speaker)
+  Headphone
+  Beep		(beep generator)
+  Phone
+  Phone Input
+  Phone Output
+  Synth
+  FM
+  Mic
+  Line
+  CD
+  Video
+  Zoom Video
+  Aux
+  PCM
+  PCM Front
+  PCM Rear
+  PCM Pan
+  Loopback
+  Analog Loopback	(D/A -> A/D loopback)
+  Digital Loopback	(playback -> capture loopback - without analog path)
+  Mono
+  Mono Output
+  Multi
+  ADC
+  Wave
+  Music
+  I2S
+  IEC958
+
+Exceptions:
+  [Digital] Capture Source
+  [Digital] Capture Switch	(aka input gain switch)
+  [Digital] Capture Volume	(aka input gain volume)
+  [Digital] Playback Switch	(aka output gain switch)
+  [Digital] Playback Volume	(aka output gain volume)
+  Tone Control - Switch
+  Tone Control - Bass
+  Tone Control - Treble
+  3D Control - Switch
+  3D Control - Center
+  3D Control - Depth
+  3D Control - Wide
+  3D Control - Space
+  3D Control - Level
+  Mic Boost [(?dB)]
+
+PCM interface:
+
+  Sample Clock Source	{ "Word", "Internal", "AutoSync" }
+  Clock Sync Status	{ "Lock", "Sync", "No Lock" }
+  External Rate		/* external capture rate */
+  Capture Rate		/* capture rate taken from external source */
+
+IEC958 (S/PDIF) interface:
+
+  IEC958 [...] [Playback|Capture] Switch	/* turn on/off the IEC958 interface */
+  IEC958 [...] [Playback|Capture] Volume	/* digital volume control */
+  IEC958 [...] [Playback|Capture] Default	/* default or global value - read/write */
+  IEC958 [...] [Playback|Capture] Mask		/* consumer and professional mask */
+  IEC958 [...] [Playback|Capture] Con Mask	/* consumer mask */
+  IEC958 [...] [Playback|Capture] Pro Mask	/* professional mask */
+  IEC958 [...] [Playback|Capture] PCM Stream	/* the settings assigned to a PCM stream */
+  IEC958 Q-subcode [Playback|Capture] Default	/* Q-subcode bits */
+  IEC958 Preamble [Playback|Capture] Default	/* burst preamble words (4*16bits) */
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
new file mode 100644
index 0000000..1d38b0d
--- /dev/null
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -0,0 +1,412 @@
+  Model name	Description
+  ----------    -----------
+ALC880
+======
+  3stack	3-jack in back and a headphone out
+  3stack-digout	3-jack in back, a HP out and a SPDIF out
+  5stack	5-jack in back, 2-jack in front
+  5stack-digout	5-jack in back, 2-jack in front, a SPDIF out
+  6stack	6-jack in back, 2-jack in front
+  6stack-digout	6-jack with a SPDIF out
+  w810		3-jack
+  z71v		3-jack (HP shared SPDIF)
+  asus		3-jack (ASUS Mobo)
+  asus-w1v	ASUS W1V
+  asus-dig	ASUS with SPDIF out
+  asus-dig2	ASUS with SPDIF out (using GPIO2)
+  uniwill	3-jack
+  fujitsu	Fujitsu Laptops (Pi1536)
+  F1734		2-jack
+  lg		LG laptop (m1 express dual)
+  lg-lw		LG LW20/LW25 laptop
+  tcl		TCL S700
+  clevo		Clevo laptops (m520G, m665n)
+  medion	Medion Rim 2150
+  test		for testing/debugging purpose, almost all controls can be
+		adjusted.  Appearing only when compiled with
+		$CONFIG_SND_DEBUG=y
+  auto		auto-config reading BIOS (default)
+
+ALC260
+======
+  hp		HP machines
+  hp-3013	HP machines (3013-variant)
+  hp-dc7600	HP DC7600
+  fujitsu	Fujitsu S7020
+  acer		Acer TravelMate
+  will		Will laptops (PB V7900)
+  replacer	Replacer 672V
+  favorit100	Maxdata Favorit 100XS
+  basic		fixed pin assignment (old default model)
+  test		for testing/debugging purpose, almost all controls can
+		adjusted.  Appearing only when compiled with
+		$CONFIG_SND_DEBUG=y
+  auto		auto-config reading BIOS (default)
+
+ALC262
+======
+  fujitsu	Fujitsu Laptop
+  hp-bpc	HP xw4400/6400/8400/9400 laptops
+  hp-bpc-d7000	HP BPC D7000
+  hp-tc-t5735	HP Thin Client T5735
+  hp-rp5700	HP RP5700
+  benq		Benq ED8
+  benq-t31	Benq T31
+  hippo		Hippo (ATI) with jack detection, Sony UX-90s
+  hippo_1	Hippo (Benq) with jack detection
+  sony-assamd	Sony ASSAMD
+  toshiba-s06	Toshiba S06
+  toshiba-rx1	Toshiba RX1
+  tyan		Tyan Thunder n6650W (S2915-E)
+  ultra		Samsung Q1 Ultra Vista model
+  lenovo-3000	Lenovo 3000 y410
+  nec		NEC Versa S9100
+  basic		fixed pin assignment w/o SPDIF
+  auto		auto-config reading BIOS (default)
+
+ALC267/268
+==========
+  quanta-il1	Quanta IL1 mini-notebook
+  3stack	3-stack model
+  toshiba	Toshiba A205
+  acer		Acer laptops
+  acer-dmic	Acer laptops with digital-mic
+  acer-aspire	Acer Aspire One
+  dell		Dell OEM laptops (Vostro 1200)
+  zepto		Zepto laptops
+  test		for testing/debugging purpose, almost all controls can
+		adjusted.  Appearing only when compiled with
+		$CONFIG_SND_DEBUG=y
+  auto		auto-config reading BIOS (default)
+
+ALC269
+======
+  basic		Basic preset
+  quanta	Quanta FL1
+  eeepc-p703	ASUS Eeepc P703 P900A
+  eeepc-p901	ASUS Eeepc P901 S101
+  fujitsu	FSC Amilo
+  lifebook	Fujitsu Lifebook S6420
+  auto		auto-config reading BIOS (default)
+
+ALC662/663/272
+==============
+  3stack-dig	3-stack (2-channel) with SPDIF
+  3stack-6ch	 3-stack (6-channel)
+  3stack-6ch-dig 3-stack (6-channel) with SPDIF
+  6stack-dig	 6-stack with SPDIF
+  lenovo-101e	 Lenovo laptop
+  eeepc-p701	ASUS Eeepc P701
+  eeepc-ep20	ASUS Eeepc EP20
+  ecs		ECS/Foxconn mobo
+  m51va		ASUS M51VA
+  g71v		ASUS G71V
+  h13		ASUS H13
+  g50v		ASUS G50V
+  asus-mode1	ASUS
+  asus-mode2	ASUS
+  asus-mode3	ASUS
+  asus-mode4	ASUS
+  asus-mode5	ASUS
+  asus-mode6	ASUS
+  dell		Dell with ALC272
+  dell-zm1	Dell ZM1 with ALC272
+  samsung-nc10	Samsung NC10 mini notebook
+  auto		auto-config reading BIOS (default)
+
+ALC882/883/885/888/889
+======================
+  3stack-dig	3-jack with SPDIF I/O
+  6stack-dig	6-jack digital with SPDIF I/O
+  arima		Arima W820Di1
+  targa		Targa T8, MSI-1049 T8
+  asus-a7j	ASUS A7J
+  asus-a7m	ASUS A7M
+  macpro	MacPro support
+  mb5		Macbook 5,1
+  macmini3	Macmini 3,1
+  mba21		Macbook Air 2,1
+  mbp3		Macbook Pro rev3
+  imac24	iMac 24'' with jack detection
+  imac91	iMac 9,1
+  w2jc		ASUS W2JC
+  3stack-2ch-dig	3-jack with SPDIF I/O (ALC883)
+  alc883-6stack-dig	6-jack digital with SPDIF I/O (ALC883)
+  3stack-6ch    3-jack 6-channel
+  3stack-6ch-dig 3-jack 6-channel with SPDIF I/O
+  6stack-dig-demo  6-jack digital for Intel demo board
+  acer		Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc)
+  acer-aspire	Acer Aspire 9810
+  acer-aspire-4930g Acer Aspire 4930G
+  acer-aspire-6530g Acer Aspire 6530G
+  acer-aspire-7730g Acer Aspire 7730G
+  acer-aspire-8930g Acer Aspire 8930G
+  medion	Medion Laptops
+  medion-md2	Medion MD2
+  targa-dig	Targa/MSI
+  targa-2ch-dig	Targa/MSI with 2-channel
+  targa-8ch-dig Targa/MSI with 8-channel (MSI GX620)
+  laptop-eapd   3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE)
+  lenovo-101e	Lenovo 101E
+  lenovo-nb0763	Lenovo NB0763
+  lenovo-ms7195-dig Lenovo MS7195
+  lenovo-sky	Lenovo Sky
+  haier-w66	Haier W66
+  3stack-hp	HP machines with 3stack (Lucknow, Samba boards)
+  6stack-dell	Dell machines with 6stack (Inspiron 530)
+  mitac		Mitac 8252D
+  clevo-m540r	Clevo M540R (6ch + digital)
+  clevo-m720	Clevo M720 laptop series
+  fujitsu-pi2515 Fujitsu AMILO Pi2515
+  fujitsu-xa3530 Fujitsu AMILO XA3530
+  3stack-6ch-intel Intel DG33* boards
+  intel-alc889a	Intel IbexPeak with ALC889A
+  intel-x58	Intel DX58 with ALC889
+  asus-p5q	ASUS P5Q-EM boards
+  mb31		MacBook 3,1
+  sony-vaio-tt  Sony VAIO TT
+  auto		auto-config reading BIOS (default)
+
+ALC861/660
+==========
+  3stack	3-jack
+  3stack-dig	3-jack with SPDIF I/O
+  6stack-dig	6-jack with SPDIF I/O
+  3stack-660	3-jack (for ALC660)
+  uniwill-m31	Uniwill M31 laptop
+  toshiba	Toshiba laptop support
+  asus		Asus laptop support
+  asus-laptop	ASUS F2/F3 laptops
+  auto		auto-config reading BIOS (default)
+
+ALC861VD/660VD
+==============
+  3stack	3-jack
+  3stack-dig	3-jack with SPDIF OUT
+  6stack-dig	6-jack with SPDIF OUT
+  3stack-660	3-jack (for ALC660VD)
+  3stack-660-digout 3-jack with SPDIF OUT (for ALC660VD)
+  lenovo	Lenovo 3000 C200
+  dallas	Dallas laptops
+  hp		HP TX1000
+  asus-v1s	ASUS V1Sn
+  auto		auto-config reading BIOS (default)
+
+CMI9880
+=======
+  minimal	3-jack in back
+  min_fp	3-jack in back, 2-jack in front
+  full		6-jack in back, 2-jack in front
+  full_dig	6-jack in back, 2-jack in front, SPDIF I/O
+  allout	5-jack in back, 2-jack in front, SPDIF out
+  auto		auto-config reading BIOS (default)
+
+AD1882 / AD1882A
+================
+  3stack	3-stack mode (default)
+  6stack	6-stack mode
+
+AD1884A / AD1883 / AD1984A / AD1984B
+====================================
+  desktop	3-stack desktop (default)
+  laptop	laptop with HP jack sensing
+  mobile	mobile devices with HP jack sensing
+  thinkpad	Lenovo Thinkpad X300
+  touchsmart	HP Touchsmart
+
+AD1884
+======
+  N/A
+
+AD1981
+======
+  basic		3-jack (default)
+  hp		HP nx6320
+  thinkpad	Lenovo Thinkpad T60/X60/Z60
+  toshiba	Toshiba U205
+
+AD1983
+======
+  N/A
+
+AD1984
+======
+  basic		default configuration
+  thinkpad	Lenovo Thinkpad T61/X61
+  dell_desktop	Dell T3400
+
+AD1986A
+=======
+  6stack	6-jack, separate surrounds (default)
+  3stack	3-stack, shared surrounds
+  laptop	2-channel only (FSC V2060, Samsung M50)
+  laptop-eapd	2-channel with EAPD (ASUS A6J)
+  laptop-automute 2-channel with EAPD and HP-automute (Lenovo N100)
+  ultra		2-channel with EAPD (Samsung Ultra tablet PC)
+  samsung	2-channel with EAPD (Samsung R65)
+  samsung-p50	2-channel with HP-automute (Samsung P50)
+
+AD1988/AD1988B/AD1989A/AD1989B
+==============================
+  6stack	6-jack
+  6stack-dig	ditto with SPDIF
+  3stack	3-jack
+  3stack-dig	ditto with SPDIF
+  laptop	3-jack with hp-jack automute
+  laptop-dig	ditto with SPDIF
+  auto		auto-config reading BIOS (default)
+
+Conexant 5045
+=============
+  laptop-hpsense    Laptop with HP sense (old model laptop)
+  laptop-micsense   Laptop with Mic sense (old model fujitsu)
+  laptop-hpmicsense Laptop with HP and Mic senses
+  benq		Benq R55E
+  laptop-hp530	HP 530 laptop
+  test		for testing/debugging purpose, almost all controls
+		can be adjusted.  Appearing only when compiled with
+		$CONFIG_SND_DEBUG=y
+
+Conexant 5047
+=============
+  laptop	Basic Laptop config 
+  laptop-hp	Laptop config for some HP models (subdevice 30A5)
+  laptop-eapd	Laptop config with EAPD support
+  test		for testing/debugging purpose, almost all controls
+		can be adjusted.  Appearing only when compiled with
+		$CONFIG_SND_DEBUG=y
+
+Conexant 5051
+=============
+  laptop	Basic Laptop config (default)
+  hp		HP Spartan laptop
+  hp-dv6736	HP dv6736
+  hp-f700	HP Compaq Presario F700
+  lenovo-x200	Lenovo X200 laptop
+  toshiba	Toshiba Satellite M300
+
+Conexant 5066
+=============
+  laptop	Basic Laptop config (default)
+  dell-laptop	Dell laptops
+  olpc-xo-1_5	OLPC XO 1.5
+  ideapad       Lenovo IdeaPad U150
+
+STAC9200
+========
+  ref		Reference board
+  oqo		OQO Model 2
+  dell-d21	Dell (unknown)
+  dell-d22	Dell (unknown)
+  dell-d23	Dell (unknown)
+  dell-m21	Dell Inspiron 630m, Dell Inspiron 640m
+  dell-m22	Dell Latitude D620, Dell Latitude D820
+  dell-m23	Dell XPS M1710, Dell Precision M90
+  dell-m24	Dell Latitude 120L
+  dell-m25	Dell Inspiron E1505n
+  dell-m26	Dell Inspiron 1501
+  dell-m27	Dell Inspiron E1705/9400
+  gateway-m4	Gateway laptops with EAPD control
+  gateway-m4-2	Gateway laptops with EAPD control
+  panasonic	Panasonic CF-74
+  auto		BIOS setup (default)
+
+STAC9205/9254
+=============
+  ref		Reference board
+  dell-m42	Dell (unknown)
+  dell-m43	Dell Precision
+  dell-m44	Dell Inspiron
+  eapd		Keep EAPD on (e.g. Gateway T1616)
+  auto		BIOS setup (default)
+
+STAC9220/9221
+=============
+  ref		Reference board
+  3stack	D945 3stack
+  5stack	D945 5stack + SPDIF
+  intel-mac-v1	Intel Mac Type 1
+  intel-mac-v2	Intel Mac Type 2
+  intel-mac-v3	Intel Mac Type 3
+  intel-mac-v4	Intel Mac Type 4
+  intel-mac-v5	Intel Mac Type 5
+  intel-mac-auto Intel Mac (detect type according to subsystem id)
+  macmini	Intel Mac Mini (equivalent with type 3)
+  macbook	Intel Mac Book (eq. type 5)
+  macbook-pro-v1 Intel Mac Book Pro 1st generation (eq. type 3)
+  macbook-pro	Intel Mac Book Pro 2nd generation (eq. type 3)
+  imac-intel	Intel iMac (eq. type 2)
+  imac-intel-20	Intel iMac (newer version) (eq. type 3)
+  ecs202	ECS/PC chips
+  dell-d81	Dell (unknown)
+  dell-d82	Dell (unknown)
+  dell-m81	Dell (unknown)
+  dell-m82	Dell XPS M1210
+  auto		BIOS setup (default)
+
+STAC9202/9250/9251
+==================
+  ref		Reference board, base config
+  m1		Some Gateway MX series laptops (NX560XL)
+  m1-2		Some Gateway MX series laptops (MX6453)
+  m2		Some Gateway MX series laptops (M255)
+  m2-2		Some Gateway MX series laptops
+  m3		Some Gateway MX series laptops
+  m5		Some Gateway MX series laptops (MP6954)
+  m6		Some Gateway NX series laptops
+  auto		BIOS setup (default)
+
+STAC9227/9228/9229/927x
+=======================
+  ref		Reference board
+  ref-no-jd	Reference board without HP/Mic jack detection
+  3stack	D965 3stack
+  5stack	D965 5stack + SPDIF
+  5stack-no-fp	D965 5stack without front panel
+  dell-3stack	Dell Dimension E520
+  dell-bios	Fixes with Dell BIOS setup
+  volknob	Fixes with volume-knob widget 0x24
+  auto		BIOS setup (default)
+
+STAC92HD71B*
+============
+  ref		Reference board
+  dell-m4-1	Dell desktops
+  dell-m4-2	Dell desktops
+  dell-m4-3	Dell desktops
+  hp-m4		HP mini 1000
+  hp-dv5	HP dv series
+  hp-hdx	HP HDX series
+  hp-dv4-1222nr	HP dv4-1222nr (with LED support)
+  auto		BIOS setup (default)
+
+STAC92HD73*
+===========
+  ref		Reference board
+  no-jd		BIOS setup but without jack-detection
+  intel		Intel DG45* mobos
+  dell-m6-amic	Dell desktops/laptops with analog mics
+  dell-m6-dmic	Dell desktops/laptops with digital mics
+  dell-m6	Dell desktops/laptops with both type of mics
+  dell-eq	Dell desktops/laptops
+  alienware	Alienware M17x
+  auto		BIOS setup (default)
+
+STAC92HD83*
+===========
+  ref		Reference board
+  mic-ref	Reference board with power management for ports
+  dell-s14	Dell laptop
+  hp		HP laptops with (inverted) mute-LED
+  auto		BIOS setup (default)
+
+STAC9872
+========
+  vaio		VAIO laptop without SPDIF
+  auto		BIOS setup (default)
+
+Cirrus Logic CS4206/4207
+========================
+  mbp55		MacBook Pro 5,5
+  imac27	IMac 27 Inch
+  auto		BIOS setup (default)
diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt
new file mode 100644
index 0000000..bdafdbd
--- /dev/null
+++ b/Documentation/sound/alsa/HD-Audio.txt
@@ -0,0 +1,721 @@
+MORE NOTES ON HD-AUDIO DRIVER
+=============================
+					Takashi Iwai <tiwai@suse.de>
+
+
+GENERAL
+-------
+
+HD-audio is the new standard on-board audio component on modern PCs
+after AC97.  Although Linux has been supporting HD-audio since long
+time ago, there are often problems with new machines.  A part of the
+problem is broken BIOS, and the rest is the driver implementation.
+This document explains the brief trouble-shooting and debugging
+methods for the	HD-audio hardware.
+
+The HD-audio component consists of two parts: the controller chip and 
+the codec chips on the HD-audio bus.  Linux provides a single driver
+for all controllers, snd-hda-intel.  Although the driver name contains
+a word of a well-known hardware vendor, it's not specific to it but for
+all controller chips by other companies.  Since the HD-audio
+controllers are supposed to be compatible, the single snd-hda-driver
+should work in most cases.  But, not surprisingly, there are known
+bugs and issues specific to each controller type.  The snd-hda-intel
+driver has a bunch of workarounds for these as described below.
+
+A controller may have multiple codecs.  Usually you have one audio
+codec and optionally one modem codec.  In theory, there might be
+multiple audio codecs, e.g. for analog and digital outputs, and the
+driver might not work properly because of conflict of mixer elements.
+This should be fixed in future if such hardware really exists.
+
+The snd-hda-intel driver has several different codec parsers depending
+on the codec.  It has a generic parser as a fallback, but this
+functionality is fairly limited until now.  Instead of the generic
+parser, usually the codec-specific parser (coded in patch_*.c) is used
+for the codec-specific implementations.  The details about the
+codec-specific problems are explained in the later sections.
+
+If you are interested in the deep debugging of HD-audio, read the
+HD-audio specification at first.  The specification is found on
+Intel's web page, for example:
+
+- http://www.intel.com/standards/hdaudio/
+
+
+HD-AUDIO CONTROLLER
+-------------------
+
+DMA-Position Problem
+~~~~~~~~~~~~~~~~~~~~
+The most common problem of the controller is the inaccurate DMA
+pointer reporting.  The DMA pointer for playback and capture can be
+read in two ways, either via a LPIB register or via a position-buffer
+map.  As default the driver tries to read from the io-mapped
+position-buffer, and falls back to LPIB if the position-buffer appears
+dead.  However, this detection isn't perfect on some devices.  In such
+a case, you can change the default method via `position_fix` option.
+
+`position_fix=1` means to use LPIB method explicitly.
+`position_fix=2` means to use the position-buffer.  0 is the default
+value, the automatic check and fallback to LPIB as described in the
+above.  If you get a problem of repeated sounds, this option might
+help.
+
+In addition to that, every controller is known to be broken regarding
+the wake-up timing.  It wakes up a few samples before actually
+processing the data on the buffer.  This caused a lot of problems, for
+example, with ALSA dmix or JACK.  Since 2.6.27 kernel, the driver puts
+an artificial delay to the wake up timing.  This delay is controlled
+via `bdl_pos_adj` option. 
+
+When `bdl_pos_adj` is a negative value (as default), it's assigned to
+an appropriate value depending on the controller chip.  For Intel
+chips, it'd be 1 while it'd be 32 for others.  Usually this works.
+Only in case it doesn't work and you get warning messages, you should
+change this parameter to other values.
+
+
+Codec-Probing Problem
+~~~~~~~~~~~~~~~~~~~~~
+A less often but a more severe problem is the codec probing.  When
+BIOS reports the available codec slots wrongly, the driver gets
+confused and tries to access the non-existing codec slot.  This often
+results in the total screw-up, and destructs the further communication
+with the codec chips.  The symptom appears usually as error messages
+like:
+------------------------------------------------------------------------
+  hda_intel: azx_get_response timeout, switching to polling mode:
+        last cmd=0x12345678
+  hda_intel: azx_get_response timeout, switching to single_cmd mode:
+        last cmd=0x12345678
+------------------------------------------------------------------------
+
+The first line is a warning, and this is usually relatively harmless.
+It means that the codec response isn't notified via an IRQ.  The
+driver uses explicit polling method to read the response.  It gives
+very slight CPU overhead, but you'd unlikely notice it.
+
+The second line is, however, a fatal error.  If this happens, usually
+it means that something is really wrong.  Most likely you are
+accessing a non-existing codec slot.
+
+Thus, if the second error message appears, try to narrow the probed
+codec slots via `probe_mask` option.  It's a bitmask, and each bit
+corresponds to the codec slot.  For example, to probe only the first
+slot, pass `probe_mask=1`.  For the first and the third slots, pass
+`probe_mask=5` (where 5 = 1 | 4), and so on.
+
+Since 2.6.29 kernel, the driver has a more robust probing method, so
+this error might happen rarely, though.
+
+On a machine with a broken BIOS, sometimes you need to force the
+driver to probe the codec slots the hardware doesn't report for use.
+In such a case, turn the bit 8 (0x100) of `probe_mask` option on.
+Then the rest 8 bits are passed as the codec slots to probe
+unconditionally.  For example, `probe_mask=0x103` will force to probe
+the codec slots 0 and 1 no matter what the hardware reports.
+
+
+Interrupt Handling
+~~~~~~~~~~~~~~~~~~
+HD-audio driver uses MSI as default (if available) since 2.6.33
+kernel as MSI works better on some machines, and in general, it's
+better for performance.  However, Nvidia controllers showed bad
+regressions with MSI (especially in a combination with AMD chipset),
+thus we disabled MSI for them.
+
+There seem also still other devices that don't work with MSI.  If you
+see a regression wrt the sound quality (stuttering, etc) or a lock-up
+in the recent kernel, try to pass `enable_msi=0` option to disable
+MSI.  If it works, you can add the known bad device to the blacklist
+defined in hda_intel.c.  In such a case, please report and give the
+patch back to the upstream developer. 
+
+
+HD-AUDIO CODEC
+--------------
+
+Model Option
+~~~~~~~~~~~~
+The most common problem regarding the HD-audio driver is the
+unsupported codec features or the mismatched device configuration.
+Most of codec-specific code has several preset models, either to
+override the BIOS setup or to provide more comprehensive features.
+
+The driver checks PCI SSID and looks through the static configuration
+table until any matching entry is found.  If you have a new machine,
+you may see a message like below:
+------------------------------------------------------------------------
+    hda_codec: ALC880: BIOS auto-probing.
+------------------------------------------------------------------------
+Meanwhile, in the earlier versions, you would see a message like:
+------------------------------------------------------------------------
+    hda_codec: Unknown model for ALC880, trying auto-probe from BIOS...
+------------------------------------------------------------------------
+Even if you see such a message, DON'T PANIC.  Take a deep breath and
+keep your towel.  First of all, it's an informational message, no
+warning, no error.  This means that the PCI SSID of your device isn't
+listed in the known preset model (white-)list.  But, this doesn't mean
+that the driver is broken.  Many codec-drivers provide the automatic
+configuration mechanism based on the BIOS setup.
+
+The HD-audio codec has usually "pin" widgets, and BIOS sets the default
+configuration of each pin, which indicates the location, the
+connection type, the jack color, etc.  The HD-audio driver can guess
+the right connection judging from these default configuration values.
+However -- some codec-support codes, such as patch_analog.c, don't
+support the automatic probing (yet as of 2.6.28).  And, BIOS is often,
+yes, pretty often broken.  It sets up wrong values and screws up the
+driver.
+
+The preset model is provided basically to overcome such a situation.
+When the matching preset model is found in the white-list, the driver
+assumes the static configuration of that preset and builds the mixer
+elements and PCM streams based on the static information.  Thus, if
+you have a newer machine with a slightly different PCI SSID from the
+existing one, you may have a good chance to re-use the same model.
+You can pass the `model` option to specify the preset model instead of
+PCI SSID look-up.
+
+What `model` option values are available depends on the codec chip.
+Check your codec chip from the codec proc file (see "Codec Proc-File"
+section below).  It will show the vendor/product name of your codec
+chip.  Then, see Documentation/sound/alsa/HD-Audio-Models.txt file,
+the section of HD-audio driver.  You can find a list of codecs
+and `model` options belonging to each codec.  For example, for Realtek
+ALC262 codec chip, pass `model=ultra` for devices that are compatible
+with Samsung Q1 Ultra.
+
+Thus, the first thing you can do for any brand-new, unsupported and
+non-working HD-audio hardware is to check HD-audio codec and several
+different `model` option values.  If you have any luck, some of them
+might suit with your device well.
+
+Some codecs such as ALC880 have a special model option `model=test`.
+This configures the driver to provide as many mixer controls as
+possible for every single pin feature except for the unsolicited
+events (and maybe some other specials).  Adjust each mixer element and
+try the I/O in the way of trial-and-error until figuring out the whole
+I/O pin mappings.
+
+Note that `model=generic` has a special meaning.  It means to use the
+generic parser regardless of the codec.  Usually the codec-specific
+parser is much better than the generic parser (as now).  Thus this
+option is more about the debugging purpose.
+
+Speaker and Headphone Output
+~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+One of the most frequent (and obvious) bugs with HD-audio is the
+silent output from either or both of a built-in speaker and a
+headphone jack.  In general, you should try a headphone output at
+first.  A speaker output often requires more additional controls like
+the external amplifier bits.  Thus a headphone output has a slightly
+better chance.
+
+Before making a bug report, double-check whether the mixer is set up
+correctly.  The recent version of snd-hda-intel driver provides mostly
+"Master" volume control as well as "Front" volume (where Front
+indicates the front-channels).  In addition, there can be individual
+"Headphone" and "Speaker" controls.
+
+Ditto for the speaker output.  There can be "External Amplifier"
+switch on some codecs.  Turn on this if present.
+
+Another related problem is the automatic mute of speaker output by
+headphone plugging.  This feature is implemented in most cases, but
+not on every preset model or codec-support code.
+
+In anyway, try a different model option if you have such a problem.
+Some other models may match better and give you more matching
+functionality.  If none of the available models works, send a bug
+report.  See the bug report section for details.
+
+If you are masochistic enough to debug the driver problem, note the
+following:
+
+- The speaker (and the headphone, too) output often requires the
+  external amplifier.  This can be set usually via EAPD verb or a
+  certain GPIO.  If the codec pin supports EAPD, you have a better
+  chance via SET_EAPD_BTL verb (0x70c).  On others, GPIO pin (mostly
+  it's either GPIO0 or GPIO1) may turn on/off EAPD.
+- Some Realtek codecs require special vendor-specific coefficients to
+  turn on the amplifier.  See patch_realtek.c.
+- IDT codecs may have extra power-enable/disable controls on each
+  analog pin.  See patch_sigmatel.c.
+- Very rare but some devices don't accept the pin-detection verb until
+  triggered.  Issuing GET_PIN_SENSE verb (0xf09) may result in the
+  codec-communication stall.  Some examples are found in
+  patch_realtek.c.
+
+
+Capture Problems
+~~~~~~~~~~~~~~~~
+The capture problems are often because of missing setups of mixers.
+Thus, before submitting a bug report, make sure that you set up the
+mixer correctly.  For example, both "Capture Volume" and "Capture
+Switch" have to be set properly in addition to the right "Capture
+Source" or "Input Source" selection.  Some devices have "Mic Boost"
+volume or switch.
+
+When the PCM device is opened via "default" PCM (without pulse-audio
+plugin), you'll likely have "Digital Capture Volume" control as well.
+This is provided for the extra gain/attenuation of the signal in
+software, especially for the inputs without the hardware volume
+control such as digital microphones.  Unless really needed, this
+should be set to exactly 50%, corresponding to 0dB -- neither extra
+gain nor attenuation.  When you use "hw" PCM, i.e., a raw access PCM,
+this control will have no influence, though.
+
+It's known that some codecs / devices have fairly bad analog circuits,
+and the recorded sound contains a certain DC-offset.  This is no bug
+of the driver.
+
+Most of modern laptops have no analog CD-input connection.  Thus, the
+recording from CD input won't work in many cases although the driver
+provides it as the capture source.  Use CDDA instead.
+
+The automatic switching of the built-in and external mic per plugging
+is implemented on some codec models but not on every model.  Partly
+because of my laziness but mostly lack of testers.  Feel free to
+submit the improvement patch to the author.
+
+
+Direct Debugging
+~~~~~~~~~~~~~~~~
+If no model option gives you a better result, and you are a tough guy
+to fight against evil, try debugging via hitting the raw HD-audio
+codec verbs to the device.  Some tools are available: hda-emu and
+hda-analyzer.  The detailed description is found in the sections
+below.  You'd need to enable hwdep for using these tools.  See "Kernel
+Configuration" section.
+
+
+OTHER ISSUES
+------------
+
+Kernel Configuration
+~~~~~~~~~~~~~~~~~~~~
+In general, I recommend you to enable the sound debug option,
+`CONFIG_SND_DEBUG=y`, no matter whether you are debugging or not.
+This enables snd_printd() macro and others, and you'll get additional
+kernel messages at probing.
+
+In addition, you can enable `CONFIG_SND_DEBUG_VERBOSE=y`.  But this
+will give you far more messages.  Thus turn this on only when you are
+sure to want it.
+
+Don't forget to turn on the appropriate `CONFIG_SND_HDA_CODEC_*`
+options.  Note that each of them corresponds to the codec chip, not
+the controller chip.  Thus, even if lspci shows the Nvidia controller,
+you may need to choose the option for other vendors.  If you are
+unsure, just select all yes.
+
+`CONFIG_SND_HDA_HWDEP` is a useful option for debugging the driver.
+When this is enabled, the driver creates hardware-dependent devices
+(one per each codec), and you have a raw access to the device via
+these device files.  For example, `hwC0D2` will be created for the
+codec slot #2 of the first card (#0).  For debug-tools such as
+hda-verb and hda-analyzer, the hwdep device has to be enabled.
+Thus, it'd be better to turn this on always.
+
+`CONFIG_SND_HDA_RECONFIG` is a new option, and this depends on the
+hwdep option above.  When enabled, you'll have some sysfs files under
+the corresponding hwdep directory.  See "HD-audio reconfiguration"
+section below.
+
+`CONFIG_SND_HDA_POWER_SAVE` option enables the power-saving feature.
+See "Power-saving" section below.
+
+
+Codec Proc-File
+~~~~~~~~~~~~~~~
+The codec proc-file is a treasure-chest for debugging HD-audio.
+It shows most of useful information of each codec widget.
+
+The proc file is located in /proc/asound/card*/codec#*, one file per
+each codec slot.  You can know the codec vendor, product id and
+names, the type of each widget, capabilities and so on.
+This file, however, doesn't show the jack sensing state, so far.  This
+is because the jack-sensing might be depending on the trigger state.
+
+This file will be picked up by the debug tools, and also it can be fed
+to the emulator as the primary codec information.  See the debug tools
+section below.
+
+This proc file can be also used to check whether the generic parser is
+used.  When the generic parser is used, the vendor/product ID name
+will appear as "Realtek ID 0262", instead of "Realtek ALC262".
+
+
+HD-Audio Reconfiguration
+~~~~~~~~~~~~~~~~~~~~~~~~
+This is an experimental feature to allow you re-configure the HD-audio
+codec dynamically without reloading the driver.  The following sysfs
+files are available under each codec-hwdep device directory (e.g. 
+/sys/class/sound/hwC0D0):
+
+vendor_id::
+  Shows the 32bit codec vendor-id hex number.  You can change the
+  vendor-id value by writing to this file.
+subsystem_id::
+  Shows the 32bit codec subsystem-id hex number.  You can change the
+  subsystem-id value by writing to this file.
+revision_id::
+  Shows the 32bit codec revision-id hex number.  You can change the
+  revision-id value by writing to this file.
+afg::
+  Shows the AFG ID.  This is read-only.
+mfg::
+  Shows the MFG ID.  This is read-only.
+name::
+  Shows the codec name string.  Can be changed by writing to this
+  file.
+modelname::
+  Shows the currently set `model` option.  Can be changed by writing
+  to this file.
+init_verbs::
+  The extra verbs to execute at initialization.  You can add a verb by
+  writing to this file.  Pass three numbers: nid, verb and parameter
+  (separated with a space).
+hints::
+  Shows / stores hint strings for codec parsers for any use.
+  Its format is `key = value`.  For example, passing `hp_detect = yes`
+  to IDT/STAC codec parser will result in the disablement of the
+  headphone detection.
+init_pin_configs::
+  Shows the initial pin default config values set by BIOS.
+driver_pin_configs::
+  Shows the pin default values set by the codec parser explicitly.
+  This doesn't show all pin values but only the changed values by
+  the parser.  That is, if the parser doesn't change the pin default
+  config values by itself, this will contain nothing.
+user_pin_configs::
+  Shows the pin default config values to override the BIOS setup.
+  Writing this (with two numbers, NID and value) appends the new
+  value.  The given will be used instead of the initial BIOS value at
+  the next reconfiguration time.  Note that this config will override
+  even the driver pin configs, too.
+reconfig::
+  Triggers the codec re-configuration.  When any value is written to
+  this file, the driver re-initialize and parses the codec tree
+  again.  All the changes done by the sysfs entries above are taken
+  into account.
+clear::
+  Resets the codec, removes the mixer elements and PCM stuff of the
+  specified codec, and clear all init verbs and hints.
+
+For example, when you want to change the pin default configuration
+value of the pin widget 0x14 to 0x9993013f, and let the driver
+re-configure based on that state, run like below:
+------------------------------------------------------------------------
+  # echo 0x14 0x9993013f > /sys/class/sound/hwC0D0/user_pin_configs
+  # echo 1 > /sys/class/sound/hwC0D0/reconfig  
+------------------------------------------------------------------------
+
+
+Early Patching
+~~~~~~~~~~~~~~
+When CONFIG_SND_HDA_PATCH_LOADER=y is set, you can pass a "patch" as a
+firmware file for modifying the HD-audio setup before initializing the
+codec.  This can work basically like the reconfiguration via sysfs in
+the above, but it does it before the first codec configuration.
+
+A patch file is a plain text file which looks like below:
+
+------------------------------------------------------------------------
+  [codec]
+  0x12345678 0xabcd1234 2
+
+  [model]
+  auto
+
+  [pincfg]
+  0x12 0x411111f0
+
+  [verb]
+  0x20 0x500 0x03
+  0x20 0x400 0xff
+
+  [hint]
+  hp_detect = yes
+------------------------------------------------------------------------
+
+The file needs to have a line `[codec]`.  The next line should contain
+three numbers indicating the codec vendor-id (0x12345678 in the
+example), the codec subsystem-id (0xabcd1234) and the address (2) of
+the codec.  The rest patch entries are applied to this specified codec
+until another codec entry is given.
+
+The `[model]` line allows to change the model name of the each codec.
+In the example above, it will be changed to model=auto.
+Note that this overrides the module option.
+
+After the `[pincfg]` line, the contents are parsed as the initial
+default pin-configurations just like `user_pin_configs` sysfs above.
+The values can be shown in user_pin_configs sysfs file, too.
+
+Similarly, the lines after `[verb]` are parsed as `init_verbs`
+sysfs entries, and the lines after `[hint]` are parsed as `hints`
+sysfs entries, respectively.
+
+Another example to override the codec vendor id from 0x12345678 to
+0xdeadbeef is like below:
+------------------------------------------------------------------------
+  [codec]
+  0x12345678 0xabcd1234 2
+
+  [vendor_id]
+  0xdeadbeef
+------------------------------------------------------------------------
+
+In the similar way, you can override the codec subsystem_id via
+`[subsystem_id]`, the revision id via `[revision_id]` line.
+Also, the codec chip name can be rewritten via `[chip_name]` line.
+------------------------------------------------------------------------
+  [codec]
+  0x12345678 0xabcd1234 2
+
+  [subsystem_id]
+  0xffff1111
+
+  [revision_id]
+  0x10
+
+  [chip_name]
+  My-own NEWS-0002
+------------------------------------------------------------------------
+
+The hd-audio driver reads the file via request_firmware().  Thus,
+a patch file has to be located on the appropriate firmware path,
+typically, /lib/firmware.  For example, when you pass the option
+`patch=hda-init.fw`, the file /lib/firmware/hda-init-fw must be
+present.
+
+The patch module option is specific to each card instance, and you
+need to give one file name for each instance, separated by commas.
+For example, if you have two cards, one for an on-board analog and one 
+for an HDMI video board, you may pass patch option like below:
+------------------------------------------------------------------------
+    options snd-hda-intel patch=on-board-patch,hdmi-patch
+------------------------------------------------------------------------
+
+
+Power-Saving
+~~~~~~~~~~~~
+The power-saving is a kind of auto-suspend of the device.  When the
+device is inactive for a certain time, the device is automatically
+turned off to save the power.  The time to go down is specified via
+`power_save` module option, and this option can be changed dynamically
+via sysfs.
+
+The power-saving won't work when the analog loopback is enabled on
+some codecs.  Make sure that you mute all unneeded signal routes when
+you want the power-saving.
+
+The power-saving feature might cause audible click noises at each
+power-down/up depending on the device.  Some of them might be
+solvable, but some are hard, I'm afraid.  Some distros such as
+openSUSE enables the power-saving feature automatically when the power
+cable is unplugged.  Thus, if you hear noises, suspect first the
+power-saving.  See /sys/module/snd_hda_intel/parameters/power_save to
+check the current value.  If it's non-zero, the feature is turned on.
+
+
+Development Tree
+~~~~~~~~~~~~~~~~
+The latest development codes for HD-audio are found on sound git tree:
+
+- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git
+
+The master branch or for-next branches can be used as the main
+development branches in general while the HD-audio specific patches
+are committed in topic/hda branch.
+
+If you are using the latest Linus tree, it'd be better to pull the
+above GIT tree onto it.  If you are using the older kernels, an easy
+way to try the latest ALSA code is to build from the snapshot
+tarball.  There are daily tarballs and the latest snapshot tarball.
+All can be built just like normal alsa-driver release packages, that
+is, installed via the usual spells: configure, make and make
+install(-modules).  See INSTALL in the package.  The snapshot tarballs
+are found at:
+
+- ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/
+
+
+Sending a Bug Report
+~~~~~~~~~~~~~~~~~~~~
+If any model or module options don't work for your device, it's time
+to send a bug report to the developers.  Give the following in your
+bug report:
+
+- Hardware vendor, product and model names
+- Kernel version (and ALSA-driver version if you built externally)
+- `alsa-info.sh` output; run with `--no-upload` option.  See the
+  section below about alsa-info
+
+If it's a regression, at best, send alsa-info outputs of both working
+and non-working kernels.  This is really helpful because we can
+compare the codec registers directly.
+
+Send a bug report either the followings:
+
+kernel-bugzilla::
+  http://bugme.linux-foundation.org/
+alsa-devel ML::
+  alsa-devel@alsa-project.org
+
+
+DEBUG TOOLS
+-----------
+
+This section describes some tools available for debugging HD-audio
+problems.
+
+alsa-info
+~~~~~~~~~
+The script `alsa-info.sh` is a very useful tool to gather the audio
+device information.  You can fetch the latest version from:
+
+- http://www.alsa-project.org/alsa-info.sh
+
+Run this script as root, and it will gather the important information
+such as the module lists, module parameters, proc file contents
+including the codec proc files, mixer outputs and the control
+elements.  As default, it will store the information onto a web server
+on alsa-project.org.  But, if you send a bug report, it'd be better to
+run with `--no-upload` option, and attach the generated file.
+
+There are some other useful options.  See `--help` option output for
+details.
+
+When a probe error occurs or when the driver obviously assigns a
+mismatched model, it'd be helpful to load the driver with
+`probe_only=1` option (at best after the cold reboot) and run
+alsa-info at this state.  With this option, the driver won't configure
+the mixer and PCM but just tries to probe the codec slot.  After
+probing, the proc file is available, so you can get the raw codec
+information before modified by the driver.  Of course, the driver
+isn't usable with `probe_only=1`.  But you can continue the
+configuration via hwdep sysfs file if hda-reconfig option is enabled.
+Using `probe_only` mask 2 skips the reset of HDA codecs (use
+`probe_only=3` as module option). The hwdep interface can be used
+to determine the BIOS codec initialization.
+
+
+hda-verb
+~~~~~~~~
+hda-verb is a tiny program that allows you to access the HD-audio
+codec directly.  You can execute a raw HD-audio codec verb with this.
+This program accesses the hwdep device, thus you need to enable the
+kernel config `CONFIG_SND_HDA_HWDEP=y` beforehand.
+
+The hda-verb program takes four arguments: the hwdep device file, the
+widget NID, the verb and the parameter.  When you access to the codec
+on the slot 2 of the card 0, pass /dev/snd/hwC0D2 to the first
+argument, typically.  (However, the real path name depends on the
+system.)
+
+The second parameter is the widget number-id to access.  The third
+parameter can be either a hex/digit number or a string corresponding
+to a verb.  Similarly, the last parameter is the value to write, or
+can be a string for the parameter type.
+
+------------------------------------------------------------------------
+  % hda-verb /dev/snd/hwC0D0 0x12 0x701 2
+  nid = 0x12, verb = 0x701, param = 0x2
+  value = 0x0
+
+  % hda-verb /dev/snd/hwC0D0 0x0 PARAMETERS VENDOR_ID
+  nid = 0x0, verb = 0xf00, param = 0x0
+  value = 0x10ec0262
+
+  % hda-verb /dev/snd/hwC0D0 2 set_a 0xb080
+  nid = 0x2, verb = 0x300, param = 0xb080
+  value = 0x0
+------------------------------------------------------------------------
+
+Although you can issue any verbs with this program, the driver state
+won't be always updated.  For example, the volume values are usually
+cached in the driver, and thus changing the widget amp value directly
+via hda-verb won't change the mixer value.
+
+The hda-verb program is found in the ftp directory:
+
+- ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/misc/
+
+Also a git repository is available:
+
+- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-verb.git
+
+See README file in the tarball for more details about hda-verb
+program.
+
+
+hda-analyzer
+~~~~~~~~~~~~
+hda-analyzer provides a graphical interface to access the raw HD-audio
+control, based on pyGTK2 binding.  It's a more powerful version of
+hda-verb.  The program gives you an easy-to-use GUI stuff for showing
+the widget information and adjusting the amp values, as well as the
+proc-compatible output.
+
+The hda-analyzer:
+
+- http://git.alsa-project.org/?p=alsa.git;a=tree;f=hda-analyzer
+
+is a part of alsa.git repository in alsa-project.org:
+
+- git://git.alsa-project.org/alsa.git
+
+Codecgraph
+~~~~~~~~~~
+Codecgraph is a utility program to generate a graph and visualizes the
+codec-node connection of a codec chip.  It's especially useful when
+you analyze or debug a codec without a proper datasheet.  The program
+parses the given codec proc file and converts to SVG via graphiz
+program.
+
+The tarball and GIT trees are found in the web page at:
+
+- http://helllabs.org/codecgraph/
+
+
+hda-emu
+~~~~~~~
+hda-emu is an HD-audio emulator.  The main purpose of this program is
+to debug an HD-audio codec without the real hardware.  Thus, it
+doesn't emulate the behavior with the real audio I/O, but it just
+dumps the codec register changes and the ALSA-driver internal changes
+at probing and operating the HD-audio driver.
+
+The program requires a codec proc-file to simulate.  Get a proc file
+for the target codec beforehand, or pick up an example codec from the
+codec proc collections in the tarball.  Then, run the program with the
+proc file, and the hda-emu program will start parsing the codec file
+and simulates the HD-audio driver:
+
+------------------------------------------------------------------------
+  % hda-emu codecs/stac9200-dell-d820-laptop
+  # Parsing..
+  hda_codec: Unknown model for STAC9200, using BIOS defaults
+  hda_codec: pin nid 08 bios pin config 40c003fa
+  ....
+------------------------------------------------------------------------
+
+The program gives you only a very dumb command-line interface.  You
+can get a proc-file dump at the current state, get a list of control
+(mixer) elements, set/get the control element value, simulate the PCM
+operation, the jack plugging simulation, etc.
+
+The package is found in:
+
+- ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/misc/
+
+A git repository is available:
+
+- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-emu.git
+
+See README file in the tarball for more details about hda-emu
+program.
diff --git a/Documentation/sound/alsa/Joystick.txt b/Documentation/sound/alsa/Joystick.txt
new file mode 100644
index 0000000..ccda41b
--- /dev/null
+++ b/Documentation/sound/alsa/Joystick.txt
@@ -0,0 +1,86 @@
+Analog Joystick Support on ALSA Drivers
+=======================================
+                          Oct. 14, 2003
+           Takashi Iwai <tiwai@suse.de>
+
+General
+-------
+
+First of all, you need to enable GAMEPORT support on Linux kernel for
+using a joystick with the ALSA driver.  For the details of gameport
+support, refer to Documentation/input/joystick.txt.
+
+The joystick support of ALSA drivers is different between ISA and PCI
+cards.  In the case of ISA (PnP) cards, it's usually handled by the
+independent module (ns558).  Meanwhile, the ALSA PCI drivers have the
+built-in gameport support.  Hence, when the ALSA PCI driver is built
+in the kernel, CONFIG_GAMEPORT must be 'y', too.  Otherwise, the
+gameport support on that card will be (silently) disabled.
+
+Some adapter modules probe the physical connection of the device at
+the load time.  It'd be safer to plug in the joystick device before
+loading the module.
+
+
+PCI Cards
+---------
+
+For PCI cards, the joystick is enabled when the appropriate module
+option is specified.  Some drivers don't need options, and the
+joystick support is always enabled.  In the former ALSA version, there
+was a dynamic control API for the joystick activation.  It was
+changed, however, to the static module options because of the system
+stability and the resource management.
+
+The following PCI drivers support the joystick natively.
+
+    Driver	Module Option	Available Values
+    ---------------------------------------------------------------------------
+    als4000	joystick_port	0 = disable (default), 1 = auto-detect,
+                                manual: any address (e.g. 0x200)
+    au88x0	N/A		N/A
+    azf3328	joystick	0 = disable, 1 = enable, -1 = auto (default)
+    ens1370	joystick	0 = disable (default), 1 = enable
+    ens1371	joystick_port	0 = disable (default), 1 = auto-detect,
+                                manual: 0x200, 0x208, 0x210, 0x218
+    cmipci	joystick_port	0 = disable (default), 1 = auto-detect,
+                                manual: any address (e.g. 0x200)
+    cs4281	N/A		N/A
+    cs46xx	N/A		N/A
+    es1938	N/A		N/A
+    es1968	joystick	0 = disable (default), 1 = enable
+    sonicvibes	N/A		N/A
+    trident	N/A		N/A
+    via82xx(*1)	joystick	0 = disable (default), 1 = enable
+    ymfpci	joystick_port	0 = disable (default), 1 = auto-detect,
+                                manual: 0x201, 0x202, 0x204, 0x205(*2)
+    ---------------------------------------------------------------------------
+
+    *1)  VIA686A/B only
+    *2)  With YMF744/754 chips, the port address can be chosen arbitrarily
+
+The following drivers don't support gameport natively, but there are
+additional modules.  Load the corresponding module to add the gameport
+support.
+
+    Driver	Additional Module
+    -----------------------------
+    emu10k1	emu10k1-gp
+    fm801	fm801-gp
+    -----------------------------
+
+Note: the "pcigame" and "cs461x" modules are for the OSS drivers only.
+      These ALSA drivers (cs46xx, trident and au88x0) have the
+      built-in gameport support.
+
+As mentioned above, ALSA PCI drivers have the built-in gameport
+support, so you don't have to load ns558 module.  Just load "joydev"
+and the appropriate adapter module (e.g. "analog").
+
+
+ISA Cards
+---------
+
+ALSA ISA drivers don't have the built-in gameport support.
+Instead, you need to load "ns558" module in addition to "joydev" and
+the adapter module (e.g. "analog").
diff --git a/Documentation/sound/alsa/MIXART.txt b/Documentation/sound/alsa/MIXART.txt
new file mode 100644
index 0000000..ef42c44
--- /dev/null
+++ b/Documentation/sound/alsa/MIXART.txt
@@ -0,0 +1,100 @@
+    Alsa driver for Digigram miXart8 and miXart8AES/EBU soundcards
+	    Digigram <alsa@digigram.com>
+
+
+GENERAL
+=======
+
+The miXart8 is a multichannel audio processing and mixing soundcard
+that has 4 stereo audio inputs and 4 stereo audio outputs.
+The miXart8AES/EBU is the same with a add-on card that offers further
+4 digital stereo audio inputs and outputs.
+Furthermore the add-on card offers external clock synchronisation
+(AES/EBU, Word Clock, Time Code and Video Synchro)
+
+The mainboard has a PowerPC that offers onboard mpeg encoding and
+decoding, samplerate conversions and various effects.
+
+The driver don't work properly at all until the certain firmwares
+are loaded, i.e. no PCM nor mixer devices will appear.
+Use the mixartloader that can be found in the alsa-tools package.
+
+
+VERSION 0.1.0
+=============
+
+One miXart8 board will be represented as 4 alsa cards, each with 1
+stereo analog capture 'pcm0c' and 1 stereo analog playback 'pcm0p' device.
+With a miXart8AES/EBU there is in addition 1 stereo digital input
+'pcm1c' and 1 stereo digital output 'pcm1p' per card.
+
+Formats
+-------
+U8, S16_LE, S16_BE, S24_3LE, S24_3BE, FLOAT_LE, FLOAT_BE
+Sample rates : 8000 - 48000 Hz continuously
+
+Playback
+--------
+For instance the playback devices are configured to have max. 4
+substreams performing hardware mixing. This could be changed to a
+maximum of 24 substreams if wished.
+Mono files will be played on the left and right channel. Each channel
+can be muted for each stream to use 8 analog/digital outputs separately.
+
+Capture
+-------
+There is one substream per capture device. For instance only stereo
+formats are supported.
+
+Mixer
+-----
+<Master> and <Master Capture> : analog volume control of playback and capture PCM.
+<PCM 0-3> and <PCM Capture> : digital volume control of each analog substream.
+<AES 0-3> and <AES Capture> : digital volume control of each AES/EBU substream.
+<Monitoring> : Loopback from 'pcm0c' to 'pcm0p' with digital volume
+and mute control.
+
+Rem : for best audio quality try to keep a 0 attenuation on the PCM
+and AES volume controls which is set by 219 in the range from 0 to 255
+(about 86% with alsamixer)
+
+
+NOT YET IMPLEMENTED
+===================
+
+- external clock support (AES/EBU, Word Clock, Time Code, Video Sync)
+- MPEG audio formats
+- mono record
+- on-board effects and samplerate conversions
+- linked streams
+
+
+FIRMWARE
+========
+
+[As of 2.6.11, the firmware can be loaded automatically with hotplug
+ when CONFIG_FW_LOADER is set.  The mixartloader is necessary only
+ for older versions or when you build the driver into kernel.]
+ 
+For loading the firmware automatically after the module is loaded, use
+the post-install command.  For example, add the following entry to
+/etc/modprobe.conf for miXart driver:
+
+	install snd-mixart /sbin/modprobe --first-time -i snd-mixart && \
+			   /usr/bin/mixartloader
+(for 2.2/2.4 kernels, add "post-install snd-mixart /usr/bin/vxloader" to
+ /etc/modules.conf, instead.)
+
+The firmware binaries are installed on /usr/share/alsa/firmware
+(or /usr/local/share/alsa/firmware, depending to the prefix option of
+configure).  There will be a miXart.conf file, which define the dsp image
+files.
+
+The firmware files are copyright by Digigram SA
+
+
+COPYRIGHT
+=========
+
+Copyright (c) 2003 Digigram SA <alsa@digigram.com>
+Distributable under GPL.
diff --git a/Documentation/sound/alsa/OSS-Emulation.txt b/Documentation/sound/alsa/OSS-Emulation.txt
new file mode 100644
index 0000000..022aaeb
--- /dev/null
+++ b/Documentation/sound/alsa/OSS-Emulation.txt
@@ -0,0 +1,305 @@
+		NOTES ON KERNEL OSS-EMULATION
+		=============================
+
+		Jan. 22, 2004  Takashi Iwai <tiwai@suse.de>
+
+
+Modules
+=======
+
+ALSA provides a powerful OSS emulation on the kernel.
+The OSS emulation for PCM, mixer and sequencer devices is implemented
+as add-on kernel modules, snd-pcm-oss, snd-mixer-oss and snd-seq-oss.
+When you need to access the OSS PCM, mixer or sequencer devices, the
+corresponding module has to be loaded.
+
+These modules are loaded automatically when the corresponding service
+is called.  The alias is defined sound-service-x-y, where x and y are
+the card number and the minor unit number.  Usually you don't have to
+define these aliases by yourself.
+
+Only necessary step for auto-loading of OSS modules is to define the
+card alias in /etc/modprobe.conf, such as
+
+	alias sound-slot-0 snd-emu10k1
+
+As the second card, define sound-slot-1 as well.
+Note that you can't use the aliased name as the target name (i.e.
+"alias sound-slot-0 snd-card-0" doesn't work any more like the old
+modutils).
+
+The currently available OSS configuration is shown in
+/proc/asound/oss/sndstat.  This shows in the same syntax of
+/dev/sndstat, which is available on the commercial OSS driver.
+On ALSA, you can symlink /dev/sndstat to this proc file.
+
+Please note that the devices listed in this proc file appear only
+after the corresponding OSS-emulation module is loaded.  Don't worry
+even if "NOT ENABLED IN CONFIG" is shown in it.
+
+
+Device Mapping
+==============
+
+ALSA supports the following OSS device files:
+
+	PCM:
+		/dev/dspX
+		/dev/adspX
+
+	Mixer:
+		/dev/mixerX
+
+	MIDI:
+		/dev/midi0X
+		/dev/amidi0X
+
+	Sequencer:
+		/dev/sequencer
+		/dev/sequencer2 (aka /dev/music)
+
+where X is the card number from 0 to 7.
+
+(NOTE: Some distributions have the device files like /dev/midi0 and
+       /dev/midi1.  They are NOT for OSS but for tclmidi, which is
+       a totally different thing.)
+
+Unlike the real OSS, ALSA cannot use the device files more than the
+assigned ones.  For example, the first card cannot use /dev/dsp1 or
+/dev/dsp2, but only /dev/dsp0 and /dev/adsp0.
+
+As seen above, PCM and MIDI may have two devices.  Usually, the first
+PCM device (hw:0,0 in ALSA) is mapped to /dev/dsp and the secondary
+device (hw:0,1) to /dev/adsp (if available).  For MIDI, /dev/midi and
+/dev/amidi, respectively.
+
+You can change this device mapping via the module options of
+snd-pcm-oss and snd-rawmidi.  In the case of PCM, the following
+options are available for snd-pcm-oss:
+
+	dsp_map		PCM device number assigned to /dev/dspX
+			(default = 0)
+	adsp_map	PCM device number assigned to /dev/adspX
+			(default = 1)
+
+For example, to map the third PCM device (hw:0,2) to /dev/adsp0,
+define like this:
+
+	options snd-pcm-oss adsp_map=2
+
+The options take arrays.  For configuring the second card, specify
+two entries separated by comma.  For example, to map the third PCM
+device on the second card to /dev/adsp1, define like below:
+
+	options snd-pcm-oss adsp_map=0,2
+
+To change the mapping of MIDI devices, the following options are
+available for snd-rawmidi:
+
+	midi_map	MIDI device number assigned to /dev/midi0X
+			(default = 0)
+	amidi_map	MIDI device number assigned to /dev/amidi0X
+			(default = 1)
+
+For example, to assign the third MIDI device on the first card to
+/dev/midi00, define as follows:
+
+	options snd-rawmidi midi_map=2
+
+
+PCM Mode
+========
+
+As default, ALSA emulates the OSS PCM with so-called plugin layer,
+i.e. tries to convert the sample format, rate or channels
+automatically when the card doesn't support it natively.
+This will lead to some problems for some applications like quake or
+wine, especially if they use the card only in the MMAP mode.
+
+In such a case, you can change the behavior of PCM per application by
+writing a command to the proc file.  There is a proc file for each PCM
+stream, /proc/asound/cardX/pcmY[cp]/oss, where X is the card number
+(zero-based), Y the PCM device number (zero-based), and 'p' is for
+playback and 'c' for capture, respectively.  Note that this proc file
+exists only after snd-pcm-oss module is loaded.
+
+The command sequence has the following syntax:
+
+	app_name fragments fragment_size [options]
+
+app_name is the name of application with (higher priority) or without
+path.
+fragments specifies the number of fragments or zero if no specific
+number is given.
+fragment_size is the size of fragment in bytes or zero if not given.
+options is the optional parameters.  The following options are
+available:
+
+	disable		the application tries to open a pcm device for
+			this channel but does not want to use it.
+	direct		don't use plugins
+	block		force block open mode
+	non-block	force non-block open mode
+	partial-frag	write also partial fragments (affects playback only)
+	no-silence	do not fill silence ahead to avoid clicks
+
+The disable option is useful when one stream direction (playback or
+capture) is not handled correctly by the application although the
+hardware itself does support both directions.
+The direct option is used, as mentioned above, to bypass the automatic
+conversion and useful for MMAP-applications.
+For example, to playback the first PCM device without plugins for
+quake, send a command via echo like the following:
+
+	% echo "quake 0 0 direct" > /proc/asound/card0/pcm0p/oss
+
+While quake wants only playback, you may append the second command
+to notify driver that only this direction is about to be allocated:
+
+	% echo "quake 0 0 disable" > /proc/asound/card0/pcm0c/oss
+
+The permission of proc files depend on the module options of snd.
+As default it's set as root, so you'll likely need to be superuser for
+sending the command above.
+
+The block and non-block options are used to change the behavior of
+opening the device file.
+
+As default, ALSA behaves as original OSS drivers, i.e. does not block
+the file when it's busy. The -EBUSY error is returned in this case.
+
+This blocking behavior can be changed globally via nonblock_open
+module option of snd-pcm-oss.  For using the blocking mode as default
+for OSS devices, define like the following:
+
+	options snd-pcm-oss nonblock_open=0
+
+The partial-frag and no-silence commands have been added recently.
+Both commands are for optimization use only.  The former command
+specifies to invoke the write transfer only when the whole fragment is
+filled.  The latter stops writing the silence data ahead
+automatically.  Both are disabled as default.
+
+You can check the currently defined configuration by reading the proc
+file.  The read image can be sent to the proc file again, hence you
+can save the current configuration
+
+	% cat /proc/asound/card0/pcm0p/oss > /somewhere/oss-cfg
+
+and restore it like
+
+	% cat /somewhere/oss-cfg > /proc/asound/card0/pcm0p/oss
+
+Also, for clearing all the current configuration, send "erase" command
+as below:
+
+	% echo "erase" > /proc/asound/card0/pcm0p/oss
+
+
+Mixer Elements
+==============
+
+Since ALSA has completely different mixer interface, the emulation of
+OSS mixer is relatively complicated.  ALSA builds up a mixer element
+from several different ALSA (mixer) controls based on the name
+string.  For example, the volume element SOUND_MIXER_PCM is composed
+from "PCM Playback Volume" and "PCM Playback Switch" controls for the
+playback direction and from "PCM Capture Volume" and "PCM Capture
+Switch" for the capture directory (if exists).  When the PCM volume of
+OSS is changed, all the volume and switch controls above are adjusted
+automatically.
+
+As default, ALSA uses the following control for OSS volumes:
+
+	OSS volume		ALSA control		Index
+	-----------------------------------------------------
+	SOUND_MIXER_VOLUME 	Master			0
+	SOUND_MIXER_BASS	Tone Control - Bass	0
+	SOUND_MIXER_TREBLE	Tone Control - Treble	0
+	SOUND_MIXER_SYNTH	Synth			0
+	SOUND_MIXER_PCM		PCM			0
+	SOUND_MIXER_SPEAKER	PC Speaker 		0
+	SOUND_MIXER_LINE	Line			0
+	SOUND_MIXER_MIC		Mic 			0
+	SOUND_MIXER_CD		CD 			0
+	SOUND_MIXER_IMIX	Monitor Mix 		0
+	SOUND_MIXER_ALTPCM	PCM			1
+	SOUND_MIXER_RECLEV	(not assigned)
+	SOUND_MIXER_IGAIN	Capture			0
+	SOUND_MIXER_OGAIN	Playback		0
+	SOUND_MIXER_LINE1	Aux			0
+	SOUND_MIXER_LINE2	Aux			1
+	SOUND_MIXER_LINE3	Aux			2
+	SOUND_MIXER_DIGITAL1	Digital			0
+	SOUND_MIXER_DIGITAL2	Digital			1
+	SOUND_MIXER_DIGITAL3	Digital			2
+	SOUND_MIXER_PHONEIN	Phone			0
+	SOUND_MIXER_PHONEOUT	Phone			1
+	SOUND_MIXER_VIDEO	Video			0
+	SOUND_MIXER_RADIO	Radio			0
+	SOUND_MIXER_MONITOR	Monitor			0
+
+The second column is the base-string of the corresponding ALSA
+control.  In fact, the controls with "XXX [Playback|Capture]
+[Volume|Switch]" will be checked in addition.
+
+The current assignment of these mixer elements is listed in the proc
+file, /proc/asound/cardX/oss_mixer, which will be like the following
+
+	VOLUME "Master" 0
+	BASS "" 0
+	TREBLE "" 0
+	SYNTH "" 0
+	PCM "PCM" 0
+	...
+
+where the first column is the OSS volume element, the second column
+the base-string of the corresponding ALSA control, and the third the
+control index.  When the string is empty, it means that the
+corresponding OSS control is not available.
+
+For changing the assignment, you can write the configuration to this
+proc file.  For example, to map "Wave Playback" to the PCM volume,
+send the command like the following:
+
+	% echo 'VOLUME "Wave Playback" 0' > /proc/asound/card0/oss_mixer
+
+The command is exactly as same as listed in the proc file.  You can
+change one or more elements, one volume per line.  In the last
+example, both "Wave Playback Volume" and "Wave Playback Switch" will
+be affected when PCM volume is changed.
+
+Like the case of PCM proc file, the permission of proc files depend on
+the module options of snd.  you'll likely need to be superuser for
+sending the command above.
+
+As well as in the case of PCM proc file, you can save and restore the
+current mixer configuration by reading and writing the whole file
+image.
+
+
+Duplex Streams
+==============
+
+Note that when attempting to use a single device file for playback and
+capture, the OSS API provides no way to set the format, sample rate or
+number of channels different in each direction.  Thus
+	io_handle = open("device", O_RDWR)
+will only function correctly if the values are the same in each direction.
+
+To use different values in the two directions, use both
+	input_handle = open("device", O_RDONLY)
+	output_handle = open("device", O_WRONLY)
+and set the values for the corresponding handle.
+
+
+Unsupported Features
+====================
+
+MMAP on ICE1712 driver
+----------------------
+ICE1712 supports only the unconventional format, interleaved
+10-channels 24bit (packed in 32bit) format.  Therefore you cannot mmap
+the buffer as the conventional (mono or 2-channels, 8 or 16bit) format
+on OSS.
+
diff --git a/Documentation/sound/alsa/Procfile.txt b/Documentation/sound/alsa/Procfile.txt
new file mode 100644
index 0000000..07301de
--- /dev/null
+++ b/Documentation/sound/alsa/Procfile.txt
@@ -0,0 +1,235 @@
+		Proc Files of ALSA Drivers
+		==========================
+		Takashi Iwai <tiwai@suse.de>
+
+General
+-------
+
+ALSA has its own proc tree, /proc/asound.  Many useful information are
+found in this tree.  When you encounter a problem and need debugging,
+check the files listed in the following sections.
+
+Each card has its subtree cardX, where X is from 0 to 7. The
+card-specific files are stored in the card* subdirectories.
+
+
+Global Information
+------------------
+
+cards
+	Shows the list of currently configured ALSA drivers,
+	index, the id string, short and long descriptions.
+
+version
+	Shows the version string and compile date.
+
+modules
+	Lists the module of each card
+
+devices
+	Lists the ALSA native device mappings.
+
+meminfo
+	Shows the status of allocated pages via ALSA drivers.
+	Appears only when CONFIG_SND_DEBUG=y.
+
+hwdep
+	Lists the currently available hwdep devices in format of
+	<card>-<device>: <name>
+
+pcm
+	Lists the currently available PCM devices in format of
+	<card>-<device>: <id>: <name> : <sub-streams>
+
+timer
+	Lists the currently available timer devices
+
+
+oss/devices
+	Lists the OSS device mappings.
+
+oss/sndstat
+	Provides the output compatible with /dev/sndstat.
+	You can symlink this to /dev/sndstat.
+
+
+Card Specific Files
+-------------------
+
+The card-specific files are found in /proc/asound/card* directories.
+Some drivers (e.g. cmipci) have their own proc entries for the
+register dump, etc (e.g. /proc/asound/card*/cmipci shows the register
+dump).  These files would be really helpful for debugging.
+
+When PCM devices are available on this card, you can see directories
+like pcm0p or pcm1c.  They hold the PCM information for each PCM
+stream.  The number after 'pcm' is the PCM device number from 0, and
+the last 'p' or 'c' means playback or capture direction.  The files in
+this subtree is described later.
+
+The status of MIDI I/O is found in midi* files.  It shows the device
+name and the received/transmitted bytes through the MIDI device.
+
+When the card is equipped with AC97 codecs, there are codec97#*
+subdirectories (described later).
+
+When the OSS mixer emulation is enabled (and the module is loaded),
+oss_mixer file appears here, too.  This shows the current mapping of
+OSS mixer elements to the ALSA control elements.  You can change the
+mapping by writing to this device.  Read OSS-Emulation.txt for
+details.
+
+
+PCM Proc Files
+--------------
+
+card*/pcm*/info
+	The general information of this PCM device: card #, device #,
+	substreams, etc.
+
+card*/pcm*/xrun_debug
+	This file appears when CONFIG_SND_DEBUG=y and
+	CONFIG_PCM_XRUN_DEBUG=y.
+	This shows the status of xrun (= buffer overrun/xrun) and
+	invalid PCM position debug/check of ALSA PCM middle layer.
+	It takes an integer value, can be changed by writing to this
+	file, such as
+
+		 # echo 5 > /proc/asound/card0/pcm0p/xrun_debug
+
+	The value consists of the following bit flags:
+	  bit 0 = Enable XRUN/jiffies debug messages
+	  bit 1 = Show stack trace at XRUN / jiffies check
+	  bit 2 = Enable additional jiffies check
+	  bit 3 = Log hwptr update at each period interrupt
+	  bit 4 = Log hwptr update at each snd_pcm_update_hw_ptr()
+
+	When the bit 0 is set, the driver will show the messages to
+	kernel log when an xrun is detected.  The debug message is
+	shown also when the invalid H/W pointer is detected at the
+	update of periods (usually called from the interrupt
+	handler).
+
+	When the bit 1 is set, the driver will show the stack trace
+	additionally.  This may help the debugging.
+
+	Since 2.6.30, this option can enable the hwptr check using
+	jiffies.  This detects spontaneous invalid pointer callback
+	values, but can be lead to too much corrections for a (mostly
+	buggy) hardware that doesn't give smooth pointer updates.
+	This feature is enabled via the bit 2.
+
+	Bits 3 and 4 are for logging the hwptr records.  Note that
+	these will give flood of kernel messages.
+
+card*/pcm*/sub*/info
+	The general information of this PCM sub-stream.
+
+card*/pcm*/sub*/status
+	The current status of this PCM sub-stream, elapsed time,
+	H/W position, etc.
+
+card*/pcm*/sub*/hw_params
+	The hardware parameters set for this sub-stream.
+
+card*/pcm*/sub*/sw_params
+	The soft parameters set for this sub-stream.
+
+card*/pcm*/sub*/prealloc
+	The buffer pre-allocation information.
+
+
+AC97 Codec Information
+----------------------
+
+card*/codec97#*/ac97#?-?
+	Shows the general information of this AC97 codec chip, such as
+	name, capabilities, set up.
+
+card*/codec97#0/ac97#?-?+regs
+	Shows the AC97 register dump.  Useful for debugging.
+
+	When CONFIG_SND_DEBUG is enabled, you can write to this file for
+	changing an AC97 register directly.  Pass two hex numbers.
+	For example,
+
+	# echo 02 9f1f > /proc/asound/card0/codec97#0/ac97#0-0+regs
+
+
+USB Audio Streams
+-----------------
+
+card*/stream*
+	Shows the assignment and the current status of each audio stream
+	of the given card.  This information is very useful for debugging.
+
+
+HD-Audio Codecs
+---------------
+
+card*/codec#*
+	Shows the general codec information and the attribute of each
+	widget node.
+
+card*/eld#*
+	Available for HDMI or DisplayPort interfaces.
+	Shows ELD(EDID Like Data) info retrieved from the attached HDMI sink,
+	and describes its audio capabilities and configurations.
+
+	Some ELD fields may be modified by doing `echo name hex_value > eld#*`.
+	Only do this if you are sure the HDMI sink provided value is wrong.
+	And if that makes your HDMI audio work, please report to us so that we
+	can fix it in future kernel releases.
+
+
+Sequencer Information
+---------------------
+
+seq/drivers
+	Lists the currently available ALSA sequencer drivers.
+
+seq/clients
+	Shows the list of currently available sequencer clients and
+	ports.  The connection status and the running status are shown
+	in this file, too.
+
+seq/queues
+	Lists the currently allocated/running sequencer queues.
+
+seq/timer
+	Lists the currently allocated/running sequencer timers.
+
+seq/oss
+	Lists the OSS-compatible sequencer stuffs.
+
+
+Help For Debugging?
+-------------------
+
+When the problem is related with PCM, first try to turn on xrun_debug
+mode.  This will give you the kernel messages when and where xrun
+happened.
+
+If it's really a bug, report it with the following information:
+
+  - the name of the driver/card, show in /proc/asound/cards
+  - the register dump, if available (e.g. card*/cmipci)
+
+when it's a PCM problem,
+
+  - set-up of PCM, shown in hw_parms, sw_params, and status in the PCM
+    sub-stream directory
+
+when it's a mixer problem,
+
+  - AC97 proc files, codec97#*/* files
+
+for USB audio/midi,
+
+  - output of lsusb -v
+  - stream* files in card directory
+
+
+The ALSA bug-tracking system is found at:
+
+    https://bugtrack.alsa-project.org/alsa-bug/
diff --git a/Documentation/sound/alsa/README.maya44 b/Documentation/sound/alsa/README.maya44
new file mode 100644
index 0000000..0e41576
--- /dev/null
+++ b/Documentation/sound/alsa/README.maya44
@@ -0,0 +1,163 @@
+NOTE: The following is the original document of Rainer's patch that the
+current maya44 code based on.  Some contents might be obsoleted, but I
+keep here as reference -- tiwai
+
+----------------------------------------------------------------
+ 
+STATE OF DEVELOPMENT:
+
+This driver is being developed on the initiative of Piotr Makowski (oponek@gmail.com) and financed by Lars Bergmann.
+Development is carried out by Rainer Zimmermann (mail@lightshed.de).
+
+ESI provided a sample Maya44 card for the development work.
+
+However, unfortunately it has turned out difficult to get detailed programming information, so I (Rainer Zimmermann) had to find out some card-specific information by experiment and conjecture. Some information (in particular, several GPIO bits) is still missing.
+
+This is the first testing version of the Maya44 driver released to the alsa-devel mailing list (Feb 5, 2008).
+
+
+The following functions work, as tested by Rainer Zimmermann and Piotr Makowski:
+
+- playback and capture at all sampling rates
+- input/output level
+- crossmixing
+- line/mic switch
+- phantom power switch
+- analogue monitor a.k.a bypass
+
+
+The following functions *should* work, but are not fully tested:
+
+- Channel 3+4 analogue - S/PDIF input switching
+- S/PDIF output
+- all inputs/outputs on the M/IO/DIO extension card
+- internal/external clock selection
+
+
+*In particular, we would appreciate testing of these functions by anyone who has access to an M/IO/DIO extension card.*
+
+
+Things that do not seem to work:
+
+- The level meters ("multi track") in 'alsamixer' do not seem to react to signals in (if this is a bug, it would probably be in the existing ICE1724 code).
+
+- Ardour 2.1 seems to work only via JACK, not using ALSA directly or via OSS. This still needs to be tracked down.
+
+
+DRIVER DETAILS:
+
+the following files were added:
+
+pci/ice1724/maya44.c        - Maya44 specific code
+pci/ice1724/maya44.h
+pci/ice1724/ice1724.patch
+pci/ice1724/ice1724.h.patch - PROPOSED patch to ice1724.h (see SAMPLING RATES)
+i2c/other/wm8776.c  - low-level access routines for Wolfson WM8776 codecs 
+include/wm8776.h
+
+
+Note that the wm8776.c code is meant to be card-independent and does not actually register the codec with the ALSA infrastructure.
+This is done in maya44.c, mainly because some of the WM8776 controls are used in Maya44-specific ways, and should be named appropriately.
+
+
+the following files were created in pci/ice1724, simply #including the corresponding file from the alsa-kernel tree:
+
+wtm.h
+vt1720_mobo.h
+revo.h
+prodigy192.h
+pontis.h
+phase.h
+maya44.h
+juli.h
+aureon.h
+amp.h
+envy24ht.h
+se.h
+prodigy_hifi.h
+
+
+*I hope this is the correct way to do things.*
+
+
+SAMPLING RATES:
+
+The Maya44 card (or more exactly, the Wolfson WM8776 codecs) allow a maximum sampling rate of 192 kHz for playback and 92 kHz for capture.
+
+As the ICE1724 chip only allows one global sampling rate, this is handled as follows:
+
+* setting the sampling rate on any open PCM device on the maya44 card will always set the *global* sampling rate for all playback and capture channels.
+
+* In the current state of the driver, setting rates of up to 192 kHz is permitted even for capture devices.
+
+*AVOID CAPTURING AT RATES ABOVE 96kHz*, even though it may appear to work. The codec cannot actually capture at such rates, meaning poor quality.
+
+
+I propose some additional code for limiting the sampling rate when setting on a capture pcm device. However because of the global sampling rate, this logic would be somewhat problematic.
+
+The proposed code (currently deactivated) is in ice1712.h.patch, ice1724.c and maya44.c (in pci/ice1712).
+
+
+SOUND DEVICES:
+
+PCM devices correspond to inputs/outputs as follows (assuming Maya44 is card #0):
+
+hw:0,0 input - stereo, analog input 1+2
+hw:0,0 output - stereo, analog output 1+2
+hw:0,1 input - stereo, analog input 3+4 OR S/PDIF input
+hw:0,1 output - stereo, analog output 3+4 (and SPDIF out)
+
+
+NAMING OF MIXER CONTROLS:
+
+(for more information about the signal flow, please refer to the block diagram on p.24 of the ESI Maya44 manual, or in the ESI windows software).
+
+
+PCM: (digital) output level for channel 1+2
+PCM 1: same for channel 3+4
+
+Mic Phantom+48V: switch for +48V phantom power for electrostatic microphones on input 1/2.
+    Make sure this is not turned on while any other source is connected to input 1/2.
+    It might damage the source and/or the maya44 card.
+
+Mic/Line input: if switch is is on, input jack 1/2 is microphone input (mono), otherwise line input (stereo).
+
+Bypass: analogue bypass from ADC input to output for channel 1+2. Same as "Monitor" in the windows driver.
+Bypass 1: same for channel 3+4.
+
+Crossmix: cross-mixer from channels 1+2 to channels 3+4
+Crossmix 1: cross-mixer from channels 3+4 to channels 1+2
+
+IEC958 Output: switch for S/PDIF output.
+    This is not supported by the ESI windows driver.
+    S/PDIF should output the same signal as channel 3+4. [untested!]
+
+
+Digitial output selectors:
+
+    These switches allow a direct digital routing from the ADCs to the DACs.
+    Each switch determines where the digital input data to one of the DACs comes from.
+    They are not supported by the ESI windows driver.
+    For normal operation, they should all be set to "PCM out".
+
+H/W: Output source channel 1
+H/W 1: Output source channel 2
+H/W 2: Output source channel 3
+H/W 3: Output source channel 4
+
+H/W 4 ... H/W 9: unknown function, left in to enable testing.
+    Possibly some of these control S/PDIF output(s).
+    If these turn out to be unused, they will go away in later driver versions.
+
+Selectable values for each of the digital output selectors are:
+   "PCM out" -> DAC output of the corresponding channel (default setting)
+   "Input 1"...
+   "Input 4" -> direct routing from ADC output of the selected input channel
+
+
+--------
+
+Feb 14, 2008
+Rainer Zimmermann
+mail@lightshed.de
+
diff --git a/Documentation/sound/alsa/SB-Live-mixer.txt b/Documentation/sound/alsa/SB-Live-mixer.txt
new file mode 100644
index 0000000..f5639d4
--- /dev/null
+++ b/Documentation/sound/alsa/SB-Live-mixer.txt
@@ -0,0 +1,356 @@
+
+		Sound Blaster Live mixer / default DSP code
+		===========================================
+
+
+The EMU10K1 chips have a DSP part which can be programmed to support
+various ways of sample processing, which is described here.
+(This article does not deal with the overall functionality of the 
+EMU10K1 chips. See the manuals section for further details.)
+
+The ALSA driver programs this portion of chip by default code
+(can be altered later) which offers the following functionality:
+
+
+1) IEC958 (S/PDIF) raw PCM
+--------------------------
+
+This PCM device (it's the 4th PCM device (index 3!) and first subdevice
+(index 0) for a given card) allows to forward 48kHz, stereo, 16-bit
+little endian streams without any modifications to the digital output
+(coaxial or optical). The universal interface allows the creation of up
+to 8 raw PCM devices operating at 48kHz, 16-bit little endian. It would
+be easy to add support for multichannel devices to the current code,
+but the conversion routines exist only for stereo (2-channel streams)
+at the time. 
+
+Look to tram_poke routines in lowlevel/emu10k1/emufx.c for more details.
+
+
+2) Digital mixer controls
+-------------------------
+
+These controls are built using the DSP instructions. They offer extended
+functionality. Only the default build-in code in the ALSA driver is described
+here. Note that the controls work as attenuators: the maximum value is the 
+neutral position leaving the signal unchanged. Note that if the  same destination 
+is mentioned in multiple controls, the signal is accumulated and can be wrapped 
+(set to maximal or minimal value without checking of overflow).
+
+
+Explanation of used abbreviations:
+
+DAC    - digital to analog converter
+ADC    - analog to digital converter
+I2S    - one-way three wire serial bus for digital sound by Philips Semiconductors
+         (this standard is used for connecting standalone DAC and ADC converters)
+LFE    - low frequency effects (subwoofer signal)
+AC97   - a chip containing an analog mixer, DAC and ADC converters
+IEC958 - S/PDIF
+FX-bus - the EMU10K1 chip has an effect bus containing 16 accumulators.
+         Each of the synthesizer voices can feed its output to these accumulators
+         and the DSP microcontroller can operate with the resulting sum.
+
+
+name='Wave Playback Volume',index=0
+
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples.
+The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
+
+name='Wave Surround Playback Volume',index=0
+
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples.
+The result samples are forwarded to the rear I2S DACs. These DACs operates
+separately (they are not inside the AC97 codec).
+
+name='Wave Center Playback Volume',index=0
+
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples.
+The result is mixed to mono signal (single channel) and forwarded to
+the ??rear?? right DAC PCM slot of the AC97 codec.
+
+name='Wave LFE Playback Volume',index=0
+
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM.
+The result is mixed to mono signal (single channel) and forwarded to
+the ??rear?? left DAC PCM slot of the AC97 codec.
+
+name='Wave Capture Volume',index=0
+name='Wave Capture Switch',index=0
+
+These controls are used to attenuate samples for left and right PCM FX-bus
+accumulator. ALSA uses accumulators 0 and 1 for left and right PCM.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Music Playback Volume',index=0
+
+This control is used to attenuate samples for left and right MIDI FX-bus
+accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples.
+The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
+
+name='Music Capture Volume',index=0
+name='Music Capture Switch',index=0
+
+These controls are used to attenuate samples for left and right MIDI FX-bus
+accumulator. ALSA uses accumulators 4 and 5 for left and right PCM.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Surround Playback Volume',index=0
+
+This control is used to attenuate samples for left and right rear PCM FX-bus
+accumulators. ALSA uses accumulators 2 and 3 for left and right rear PCM samples.
+The result samples are forwarded to the rear I2S DACs. These DACs operate
+separately (they are not inside the AC97 codec).
+
+name='Surround Capture Volume',index=0
+name='Surround Capture Switch',index=0
+
+These controls are used to attenuate samples for left and right rear PCM FX-bus
+accumulators. ALSA uses accumulators 2 and 3 for left and right rear PCM samples.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Center Playback Volume',index=0
+
+This control is used to attenuate sample for center PCM FX-bus accumulator.
+ALSA uses accumulator 6 for center PCM sample. The result sample is forwarded
+to the ??rear?? right DAC PCM slot of the AC97 codec.
+
+name='LFE Playback Volume',index=0
+
+This control is used to attenuate sample for center PCM FX-bus accumulator.
+ALSA uses accumulator 6 for center PCM sample. The result sample is forwarded
+to the ??rear?? left DAC PCM slot of the AC97 codec.
+
+name='AC97 Playback Volume',index=0
+
+This control is used to attenuate samples for left and right front ADC PCM slots
+of the AC97 codec. The result samples are forwarded to the front DAC PCM
+slots of the AC97 codec.
+********************************************************************************
+*** Note: This control should be zero for the standard operations, otherwise ***
+*** a digital loopback is activated.                                         ***
+********************************************************************************
+
+name='AC97 Capture Volume',index=0
+
+This control is used to attenuate samples for left and right front ADC PCM slots
+of the AC97 codec. The result is forwarded to the ADC capture FIFO (thus to
+the standard capture PCM device).
+********************************************************************************
+*** Note: This control should be 100 (maximal value), otherwise no analog    ***
+*** inputs of the AC97 codec can be captured (recorded).                     ***
+********************************************************************************
+
+name='IEC958 TTL Playback Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 TTL
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the front DAC PCM slots of the AC97 codec.
+
+name='IEC958 TTL Capture Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 TTL
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the ADC capture FIFO (thus to the standard capture PCM device).
+
+name='Zoom Video Playback Volume',index=0
+
+This control is used to attenuate samples from left and right zoom video
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the front DAC PCM slots of the AC97 codec.
+
+name='Zoom Video Capture Volume',index=0
+
+This control is used to attenuate samples from left and right zoom video
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the ADC capture FIFO (thus to the standard capture PCM device).
+
+name='IEC958 LiveDrive Playback Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 optical
+digital input. The result samples are forwarded to the front DAC PCM slots
+of the AC97 codec.
+
+name='IEC958 LiveDrive Capture Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 optical
+digital inputs. The result samples are forwarded to the ADC capture FIFO
+(thus to the standard capture PCM device).
+
+name='IEC958 Coaxial Playback Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 coaxial
+digital inputs. The result samples are forwarded to the front DAC PCM slots
+of the AC97 codec.
+
+name='IEC958 Coaxial Capture Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 coaxial
+digital inputs. The result samples are forwarded to the ADC capture FIFO
+(thus to the standard capture PCM device).
+
+name='Line LiveDrive Playback Volume',index=0
+name='Line LiveDrive Playback Volume',index=1
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the LiveDrive). The result samples are forwarded to the front
+DAC PCM slots of the AC97 codec.
+
+name='Line LiveDrive Capture Volume',index=1
+name='Line LiveDrive Capture Volume',index=1
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the LiveDrive). The result samples are forwarded to the ADC
+capture FIFO (thus to the standard capture PCM device).
+
+name='Tone Control - Switch',index=0
+
+This control turns the tone control on or off. The samples for front, rear
+and center / LFE outputs are affected.
+
+name='Tone Control - Bass',index=0
+
+This control sets the bass intensity. There is no neutral value!!
+When the tone control code is activated, the samples are always modified.
+The closest value to pure signal is 20.
+
+name='Tone Control - Treble',index=0
+
+This control sets the treble intensity. There is no neutral value!!
+When the tone control code is activated, the samples are always modified.
+The closest value to pure signal is 20.
+
+name='IEC958 Optical Raw Playback Switch',index=0
+
+If this switch is on, then the samples for the IEC958 (S/PDIF) digital
+output are taken only from the raw FX8010 PCM, otherwise standard front
+PCM samples are taken.
+
+name='Headphone Playback Volume',index=1
+
+This control attenuates the samples for the headphone output.
+
+name='Headphone Center Playback Switch',index=1
+
+If this switch is on, then the sample for the center PCM is put to the
+left headphone output (useful for SB Live cards without separate center/LFE
+output).
+
+name='Headphone LFE Playback Switch',index=1
+
+If this switch is on, then the sample for the center PCM is put to the
+right headphone output (useful for SB Live cards without separate center/LFE
+output).
+
+
+3) PCM stream related controls
+------------------------------
+
+name='EMU10K1 PCM Volume',index 0-31
+
+Channel volume attenuation in range 0-0xffff. The maximum value (no
+attenuation) is default. The channel mapping for three values is
+as follows:
+
+	0 - mono, default 0xffff (no attenuation)
+	1 - left, default 0xffff (no attenuation)
+	2 - right, default 0xffff (no attenuation)
+
+name='EMU10K1 PCM Send Routing',index 0-31
+
+This control specifies the destination - FX-bus accumulators. There are
+twelve values with this mapping:
+
+	 0 -  mono, A destination (FX-bus 0-15), default 0
+	 1 -  mono, B destination (FX-bus 0-15), default 1
+	 2 -  mono, C destination (FX-bus 0-15), default 2
+	 3 -  mono, D destination (FX-bus 0-15), default 3
+	 4 -  left, A destination (FX-bus 0-15), default 0
+	 5 -  left, B destination (FX-bus 0-15), default 1
+	 6 -  left, C destination (FX-bus 0-15), default 2
+	 7 -  left, D destination (FX-bus 0-15), default 3
+	 8 - right, A destination (FX-bus 0-15), default 0
+	 9 - right, B destination (FX-bus 0-15), default 1
+	10 - right, C destination (FX-bus 0-15), default 2
+	11 - right, D destination (FX-bus 0-15), default 3
+
+Don't forget that it's illegal to assign a channel to the same FX-bus accumulator 
+more than once (it means 0=0 && 1=0 is an invalid combination).
+ 
+name='EMU10K1 PCM Send Volume',index 0-31
+
+It specifies the attenuation (amount) for given destination in range 0-255.
+The channel mapping is following:
+
+	 0 -  mono, A destination attn, default 255 (no attenuation)
+	 1 -  mono, B destination attn, default 255 (no attenuation)
+	 2 -  mono, C destination attn, default 0 (mute)
+	 3 -  mono, D destination attn, default 0 (mute)
+	 4 -  left, A destination attn, default 255 (no attenuation)
+	 5 -  left, B destination attn, default 0 (mute)
+	 6 -  left, C destination attn, default 0 (mute)
+	 7 -  left, D destination attn, default 0 (mute)
+	 8 - right, A destination attn, default 0 (mute)
+	 9 - right, B destination attn, default 255 (no attenuation)
+	10 - right, C destination attn, default 0 (mute)
+	11 - right, D destination attn, default 0 (mute)
+
+
+
+4) MANUALS/PATENTS:
+-------------------
+
+ftp://opensource.creative.com/pub/doc
+-------------------------------------
+
+        Files:
+        LM4545.pdf      AC97 Codec
+
+        m2049.pdf       The EMU10K1 Digital Audio Processor
+
+        hog63.ps        FX8010 - A DSP Chip Architecture for Audio Effects
+
+
+WIPO Patents
+------------
+        Patent numbers:
+        WO 9901813 (A1) Audio Effects Processor with multiple asynchronous (Jan. 14, 1999)
+                        streams
+
+        WO 9901814 (A1) Processor with Instruction Set for Audio Effects (Jan. 14, 1999)
+
+        WO 9901953 (A1) Audio Effects Processor having Decoupled Instruction
+                        Execution and Audio Data Sequencing (Jan. 14, 1999)
+
+
+US Patents (http://www.uspto.gov/)
+----------------------------------
+
+        US 5925841      Digital Sampling Instrument employing cache memory (Jul. 20, 1999)
+
+        US 5928342      Audio Effects Processor integrated on a single chip (Jul. 27, 1999)
+                        with a multiport memory onto which multiple asynchronous
+                        digital sound samples can be concurrently loaded
+
+        US 5930158      Processor with Instruction Set for Audio Effects (Jul. 27, 1999)
+
+        US 6032235      Memory initialization circuit (Tram) (Feb. 29, 2000)
+
+        US 6138207      Interpolation looping of audio samples in cache connected to    (Oct. 24, 2000)
+                        system bus with prioritization and modification of bus transfers
+                        in accordance with loop ends and minimum block sizes
+
+        US 6151670      Method for conserving memory storage using a (Nov. 21, 2000)
+                        pool of  short term memory registers
+
+        US 6195715      Interrupt control for multiple programs communicating with      (Feb. 27, 2001)
+                        a common interrupt by associating programs to GP registers,
+                        defining interrupt register, polling GP registers, and invoking
+                        callback routine associated with defined interrupt register
diff --git a/Documentation/sound/alsa/VIA82xx-mixer.txt b/Documentation/sound/alsa/VIA82xx-mixer.txt
new file mode 100644
index 0000000..1b0ac06
--- /dev/null
+++ b/Documentation/sound/alsa/VIA82xx-mixer.txt
@@ -0,0 +1,8 @@
+
+				VIA82xx mixer
+				=============
+
+On many VIA82xx boards, the 'Input Source Select' mixer control does not work.
+Setting it to 'Input2' on such boards will cause recording to hang, or fail
+with EIO (input/output error) via OSS emulation.  This control should be left
+at 'Input1' for such cards.
diff --git a/Documentation/sound/alsa/emu10k1-jack.txt b/Documentation/sound/alsa/emu10k1-jack.txt
new file mode 100644
index 0000000..751d450
--- /dev/null
+++ b/Documentation/sound/alsa/emu10k1-jack.txt
@@ -0,0 +1,74 @@
+This document is a guide to using the emu10k1 based devices with JACK for low
+latency, multichannel recording functionality.  All of my recent work to allow
+Linux users to use the full capabilities of their hardware has been inspired 
+by the kX Project.  Without their work I never would have discovered the true
+power of this hardware.
+
+	http://www.kxproject.com
+						- Lee Revell, 2005.03.30
+
+Low latency, multichannel audio with JACK and the emu10k1/emu10k2
+-----------------------------------------------------------------
+
+Until recently, emu10k1 users on Linux did not have access to the same low
+latency, multichannel features offered by the "kX ASIO" feature of their
+Windows driver.  As of ALSA 1.0.9 this is no more!
+
+For those unfamiliar with kX ASIO, this consists of 16 capture and 16 playback
+channels.  With a post 2.6.9 Linux kernel, latencies down to 64 (1.33 ms) or
+even 32 (0.66ms) frames should work well.
+
+The configuration is slightly more involved than on Windows, as you have to
+select the correct device for JACK to use.  Actually, for qjackctl users it's
+fairly self explanatory - select Duplex, then for capture and playback select
+the multichannel devices, set the in and out channels to 16, and the sample
+rate to 48000Hz.  The command line looks like this:
+
+/usr/local/bin/jackd -R -dalsa -r48000 -p64 -n2 -D -Chw:0,2 -Phw:0,3 -S
+
+This will give you 16 input ports and 16 output ports.
+
+The 16 output ports map onto the 16 FX buses (or the first 16 of 64, for the
+Audigy).  The mapping from FX bus to physical output is described in
+SB-Live-mixer.txt (or Audigy-mixer.txt).
+
+The 16 input ports are connected to the 16 physical inputs.  Contrary to
+popular belief, all emu10k1 cards are multichannel cards.  Which of these
+input channels have physical inputs connected to them depends on the card
+model.  Trial and error is highly recommended; the pinout diagrams
+for the card have been reverse engineered by some enterprising kX users and are 
+available on the internet.  Meterbridge is helpful here, and the kX forums are
+packed with useful information.
+
+Each input port will either correspond to a digital (SPDIF) input, an analog
+input, or nothing.  The one exception is the SBLive! 5.1.  On these devices,
+the second and third input ports are wired to the center/LFE output.  You will
+still see 16 capture channels, but only 14 are available for recording inputs.
+
+This chart, borrowed from kxfxlib/da_asio51.cpp, describes the mapping of JACK
+ports to FXBUS2 (multitrack recording input) and EXTOUT (physical output)
+channels.
+
+/*JACK (& ASIO) mappings on 10k1 5.1 SBLive cards:
+--------------------------------------------
+JACK		Epilog		FXBUS2(nr)
+--------------------------------------------
+capture_1	asio14		FXBUS2(0xe)
+capture_2	asio15		FXBUS2(0xf)
+capture_3	asio0		FXBUS2(0x0)	
+~capture_4	Center		EXTOUT(0x11)	// mapped to by Center
+~capture_5	LFE		EXTOUT(0x12)	// mapped to by LFE
+capture_6	asio3		FXBUS2(0x3)
+capture_7	asio4		FXBUS2(0x4)
+capture_8	asio5		FXBUS2(0x5)
+capture_9	asio6		FXBUS2(0x6)
+capture_10	asio7		FXBUS2(0x7)
+capture_11	asio8		FXBUS2(0x8)
+capture_12	asio9		FXBUS2(0x9)
+capture_13	asio10		FXBUS2(0xa)
+capture_14	asio11		FXBUS2(0xb)
+capture_15	asio12		FXBUS2(0xc)
+capture_16	asio13		FXBUS2(0xd)
+*/
+
+TODO: describe use of ld10k1/qlo10k1 in conjunction with JACK
diff --git a/Documentation/sound/alsa/hda_codec.txt b/Documentation/sound/alsa/hda_codec.txt
new file mode 100644
index 0000000..de8efbc
--- /dev/null
+++ b/Documentation/sound/alsa/hda_codec.txt
@@ -0,0 +1,322 @@
+Notes on Universal Interface for Intel High Definition Audio Codec
+------------------------------------------------------------------
+
+Takashi Iwai <tiwai@suse.de>
+
+
+[Still a draft version]
+
+
+General
+=======
+
+The snd-hda-codec module supports the generic access function for the
+High Definition (HD) audio codecs.  It's designed to be independent
+from the controller code like ac97 codec module.  The real accessors
+from/to the controller must be implemented in the lowlevel driver.
+
+The structure of this module is similar with ac97_codec module.
+Each codec chip belongs to a bus class which communicates with the
+controller.
+
+
+Initialization of Bus Instance
+==============================
+
+The card driver has to create struct hda_bus at first.  The template
+struct should be filled and passed to the constructor:
+
+struct hda_bus_template {
+	void *private_data;
+	struct pci_dev *pci;
+	const char *modelname;
+	struct hda_bus_ops ops;
+};
+
+The card driver can set and use the private_data field to retrieve its
+own data in callback functions.  The pci field is used when the patch
+needs to check the PCI subsystem IDs, so on.  For non-PCI system, it
+doesn't have to be set, of course.
+The modelname field specifies the board's specific configuration.  The
+string is passed to the codec parser, and it depends on the parser how
+the string is used.
+These fields, private_data, pci and modelname are all optional.
+
+The ops field contains the callback functions as the following:
+
+struct hda_bus_ops {
+	int (*command)(struct hda_codec *codec, hda_nid_t nid, int direct,
+		       unsigned int verb, unsigned int parm);
+	unsigned int (*get_response)(struct hda_codec *codec);
+	void (*private_free)(struct hda_bus *);
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+	void (*pm_notify)(struct hda_codec *codec);
+#endif
+};
+
+The command callback is called when the codec module needs to send a
+VERB to the controller.  It's always a single command.
+The get_response callback is called when the codec requires the answer
+for the last command.  These two callbacks are mandatory and have to
+be given.
+The third, private_free callback, is optional.  It's called in the
+destructor to release any necessary data in the lowlevel driver.
+
+The pm_notify callback is available only with
+CONFIG_SND_HDA_POWER_SAVE kconfig.  It's called when the codec needs
+to power up or may power down.  The controller should check the all
+belonging codecs on the bus whether they are actually powered off
+(check codec->power_on), and optionally the driver may power down the
+controller side, too.
+
+The bus instance is created via snd_hda_bus_new().  You need to pass
+the card instance, the template, and the pointer to store the
+resultant bus instance.
+
+int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp,
+		    struct hda_bus **busp);
+
+It returns zero if successful.  A negative return value means any
+error during creation.
+
+
+Creation of Codec Instance
+==========================
+
+Each codec chip on the board is then created on the BUS instance.
+To create a codec instance, call snd_hda_codec_new().
+
+int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
+		      struct hda_codec **codecp);
+
+The first argument is the BUS instance, the second argument is the
+address of the codec, and the last one is the pointer to store the
+resultant codec instance (can be NULL if not needed).
+
+The codec is stored in a linked list of bus instance.  You can follow
+the codec list like:
+
+	struct hda_codec *codec;
+	list_for_each_entry(codec, &bus->codec_list, list) {
+		...
+	}
+
+The codec isn't initialized at this stage properly.  The
+initialization sequence is called when the controls are built later.
+
+
+Codec Access
+============
+
+To access codec, use snd_hda_codec_read() and snd_hda_codec_write().
+snd_hda_param_read() is for reading parameters.
+For writing a sequence of verbs, use snd_hda_sequence_write().
+
+There are variants of cached read/write, snd_hda_codec_write_cache(),
+snd_hda_sequence_write_cache().  These are used for recording the
+register states for the power-management resume.  When no PM is needed,
+these are equivalent with non-cached version.
+
+To retrieve the number of sub nodes connected to the given node, use
+snd_hda_get_sub_nodes().  The connection list can be obtained via
+snd_hda_get_connections() call.
+
+When an unsolicited event happens, pass the event via
+snd_hda_queue_unsol_event() so that the codec routines will process it
+later.
+
+
+(Mixer) Controls
+================
+
+To create mixer controls of all codecs, call
+snd_hda_build_controls().  It then builds the mixers and does
+initialization stuff on each codec.
+
+
+PCM Stuff
+=========
+
+snd_hda_build_pcms() gives the necessary information to create PCM
+streams.  When it's called, each codec belonging to the bus stores 
+codec->num_pcms and codec->pcm_info fields.  The num_pcms indicates
+the number of elements in pcm_info array.  The card driver is supposed
+to traverse the codec linked list, read the pcm information in
+pcm_info array, and build pcm instances according to them. 
+
+The pcm_info array contains the following record:
+
+/* PCM information for each substream */
+struct hda_pcm_stream {
+	unsigned int substreams;	/* number of substreams, 0 = not exist */
+	unsigned int channels_min;	/* min. number of channels */
+	unsigned int channels_max;	/* max. number of channels */
+	hda_nid_t nid;	/* default NID to query rates/formats/bps, or set up */
+	u32 rates;	/* supported rates */
+	u64 formats;	/* supported formats (SNDRV_PCM_FMTBIT_) */
+	unsigned int maxbps;	/* supported max. bit per sample */
+	struct hda_pcm_ops ops;
+};
+
+/* for PCM creation */
+struct hda_pcm {
+	char *name;
+	struct hda_pcm_stream stream[2];
+};
+
+The name can be passed to snd_pcm_new().  The stream field contains
+the information  for playback (SNDRV_PCM_STREAM_PLAYBACK = 0) and
+capture (SNDRV_PCM_STREAM_CAPTURE = 1) directions.  The card driver
+should pass substreams to snd_pcm_new() for the number of substreams
+to create.
+
+The channels_min, channels_max, rates and formats should be copied to
+runtime->hw record.  They and maxbps fields are used also to compute
+the format value for the HDA codec and controller.  Call
+snd_hda_calc_stream_format() to get the format value.
+
+The ops field contains the following callback functions:
+
+struct hda_pcm_ops {
+	int (*open)(struct hda_pcm_stream *info, struct hda_codec *codec,
+		    struct snd_pcm_substream *substream);
+	int (*close)(struct hda_pcm_stream *info, struct hda_codec *codec,
+		     struct snd_pcm_substream *substream);
+	int (*prepare)(struct hda_pcm_stream *info, struct hda_codec *codec,
+		       unsigned int stream_tag, unsigned int format,
+		       struct snd_pcm_substream *substream);
+	int (*cleanup)(struct hda_pcm_stream *info, struct hda_codec *codec,
+		       struct snd_pcm_substream *substream);
+};
+
+All are non-NULL, so you can call them safely without NULL check.
+
+The open callback should be called in PCM open after runtime->hw is
+set up.  It may override some setting and constraints additionally.
+Similarly, the close callback should be called in the PCM close.
+
+The prepare callback should be called in PCM prepare.  This will set
+up the codec chip properly for the operation.  The cleanup should be
+called in hw_free to clean up the configuration.
+
+The caller should check the return value, at least for open and
+prepare callbacks.  When a negative value is returned, some error
+occurred.
+
+
+Proc Files
+==========
+
+Each codec dumps the widget node information in
+/proc/asound/card*/codec#* file.  This information would be really
+helpful for debugging.  Please provide its contents together with the
+bug report.
+
+
+Power Management
+================
+
+It's simple:
+Call snd_hda_suspend() in the PM suspend callback.
+Call snd_hda_resume() in the PM resume callback.
+
+
+Codec Preset (Patch)
+====================
+
+To set up and handle the codec functionality fully, each codec may
+have a codec preset (patch).  It's defined in struct hda_codec_preset:
+
+	struct hda_codec_preset {
+		unsigned int id;
+		unsigned int mask;
+		unsigned int subs;
+		unsigned int subs_mask;
+		unsigned int rev;
+		const char *name;
+		int (*patch)(struct hda_codec *codec);
+	};
+
+When the codec id and codec subsystem id match with the given id and
+subs fields bitwise (with bitmask mask and subs_mask), the callback
+patch is called.  The patch callback should initialize the codec and
+set the codec->patch_ops field.  This is defined as below:
+
+	struct hda_codec_ops {
+		int (*build_controls)(struct hda_codec *codec);
+		int (*build_pcms)(struct hda_codec *codec);
+		int (*init)(struct hda_codec *codec);
+		void (*free)(struct hda_codec *codec);
+		void (*unsol_event)(struct hda_codec *codec, unsigned int res);
+	#ifdef CONFIG_PM
+		int (*suspend)(struct hda_codec *codec, pm_message_t state);
+		int (*resume)(struct hda_codec *codec);
+	#endif
+	#ifdef CONFIG_SND_HDA_POWER_SAVE
+		int (*check_power_status)(struct hda_codec *codec,
+					  hda_nid_t nid);
+	#endif
+	};
+
+The build_controls callback is called from snd_hda_build_controls().
+Similarly, the build_pcms callback is called from
+snd_hda_build_pcms().  The init callback is called after
+build_controls to initialize the hardware.
+The free callback is called as a destructor.
+
+The unsol_event callback is called when an unsolicited event is
+received.
+
+The suspend and resume callbacks are for power management.
+They can be NULL if no special sequence is required.  When the resume
+callback is NULL, the driver calls the init callback and resumes the
+registers from the cache.  If other handling is needed, you'd need to
+write your own resume callback.  There, the amp values can be resumed
+via
+	void snd_hda_codec_resume_amp(struct hda_codec *codec);
+and the other codec registers via
+	void snd_hda_codec_resume_cache(struct hda_codec *codec);
+
+The check_power_status callback is called when the amp value of the
+given widget NID is changed.  The codec code can turn on/off the power
+appropriately from this information.
+
+Each entry can be NULL if not necessary to be called.
+
+
+Generic Parser
+==============
+
+When the device doesn't match with any given presets, the widgets are
+parsed via th generic parser (hda_generic.c).  Its support is
+limited: no multi-channel support, for example.
+
+
+Digital I/O
+===========
+
+Call snd_hda_create_spdif_out_ctls() from the patch to create controls
+related with SPDIF out.
+
+
+Helper Functions
+================
+
+snd_hda_get_codec_name() stores the codec name on the given string.
+
+snd_hda_check_board_config() can be used to obtain the configuration
+information matching with the device.  Define the model string table
+and the table with struct snd_pci_quirk entries (zero-terminated),
+and pass it to the function.  The function checks the modelname given
+as a module parameter, and PCI subsystem IDs.  If the matching entry
+is found, it returns the config field value.
+
+snd_hda_add_new_ctls() can be used to create and add control entries.
+Pass the zero-terminated array of struct snd_kcontrol_new
+
+Macros HDA_CODEC_VOLUME(), HDA_CODEC_MUTE() and their variables can be
+used for the entry of struct snd_kcontrol_new.
+
+The input MUX helper callbacks for such a control are provided, too:
+snd_hda_input_mux_info() and snd_hda_input_mux_put().  See
+patch_realtek.c for example.
diff --git a/Documentation/sound/alsa/hdspm.txt b/Documentation/sound/alsa/hdspm.txt
new file mode 100644
index 0000000..7a67ff7
--- /dev/null
+++ b/Documentation/sound/alsa/hdspm.txt
@@ -0,0 +1,362 @@
+Software Interface ALSA-DSP MADI Driver 
+
+(translated from German, so no good English ;-), 
+2004 - winfried ritsch
+
+
+
+ Full functionality has been added to the driver. Since some of
+ the Controls and startup-options  are ALSA-Standard and only the
+ special Controls are described and discussed below.
+
+
+ hardware functionality:
+
+   
+   Audio transmission:
+
+     number of channels --  depends on transmission mode
+
+		The number of channels chosen is from 1..Nmax. The reason to
+		use for a lower number of channels is only resource allocation,
+		since unused DMA channels are disabled and less memory is
+		allocated. So also the throughput of the PCI system can be
+		scaled. (Only important for low performance boards).
+
+       Single Speed -- 1..64 channels 
+
+		 (Note: Choosing the 56channel mode for transmission or as
+		 receiver, only 56 are transmitted/received over the MADI, but
+		 all 64 channels are available for the mixer, so channel count
+		 for the driver)
+
+       Double Speed -- 1..32 channels
+
+		 Note: Choosing the 56-channel mode for
+		 transmission/receive-mode , only 28 are transmitted/received
+		 over the MADI, but all 32 channels are available for the mixer,
+		 so channel count for the driver
+
+
+       Quad Speed -- 1..16 channels 
+
+		 Note: Choosing the 56-channel mode for
+		 transmission/receive-mode , only 14 are transmitted/received
+		 over the MADI, but all 16 channels are available for the mixer,
+		 so channel count for the driver
+
+     Format -- signed 32 Bit Little Endian (SNDRV_PCM_FMTBIT_S32_LE)
+
+     Sample Rates --
+
+       Single Speed -- 32000, 44100, 48000
+
+       Double Speed -- 64000, 88200, 96000 (untested)
+
+       Quad Speed -- 128000, 176400, 192000 (untested)
+
+     access-mode -- MMAP (memory mapped), Not interleaved
+     (PCM_NON-INTERLEAVED)
+
+     buffer-sizes -- 64,128,256,512,1024,2048,8192 Samples
+
+     fragments -- 2
+
+     Hardware-pointer -- 2 Modi
+
+
+		 The Card supports the readout of the actual Buffer-pointer,
+		 where DMA reads/writes. Since of the bulk mode of PCI it is only
+		 64 Byte accurate. SO it is not really usable for the
+		 ALSA-mid-level functions (here the buffer-ID gives a better
+		 result), but if MMAP is used by the application. Therefore it
+		 can be configured at load-time with the parameter
+		 precise-pointer.
+
+
+		 (Hint: Experimenting I found that the pointer is maximum 64 to
+		 large never to small. So if you subtract 64 you always have a
+		 safe pointer for writing, which is used on this mode inside
+		 ALSA. In theory now you can get now a latency as low as 16
+		 Samples, which is a quarter of the interrupt possibilities.)
+
+       Precise Pointer -- off
+					interrupt used for pointer-calculation
+
+       Precise Pointer -- on
+					hardware pointer used.
+
+   Controller:
+
+
+	  Since DSP-MADI-Mixer has 8152 Fader, it does not make sense to
+	  use the standard mixer-controls, since this would break most of
+	  (especially graphic) ALSA-Mixer GUIs. So Mixer control has be
+	  provided by a 2-dimensional controller using the
+	  hwdep-interface. 
+
+     Also all 128+256 Peak and RMS-Meter can be accessed via the
+     hwdep-interface. Since it could be a performance problem always
+     copying and converting Peak and RMS-Levels even if you just need
+     one, I decided to export the hardware structure, so that of
+     needed some driver-guru can implement a memory-mapping of mixer
+     or peak-meters over ioctl, or also to do only copying and no
+     conversion. A test-application shows the usage of the controller.
+
+    Latency Controls --- not implemented !!!
+
+
+	   Note: Within the windows-driver the latency is accessible of a
+	   control-panel, but buffer-sizes are controlled with ALSA from
+	   hwparams-calls and should not be changed in run-state, I did not
+	   implement it here.
+
+
+    System Clock -- suspended !!!!
+
+        Name -- "System Clock Mode"
+
+        Access -- Read Write
+
+        Values -- "Master" "Slave"
+
+
+		  !!!! This is a hardware-function but is in conflict with the
+		  Clock-source controller, which is a kind of ALSA-standard. I
+		  makes sense to set the card to a special mode (master at some
+		  frequency or slave), since even not using an Audio-application
+		  a studio should have working synchronisations setup. So use
+		  Clock-source-controller instead !!!!
+
+    Clock Source  
+
+       Name -- "Sample Clock Source"
+
+       Access -- Read Write
+
+       Values -- "AutoSync", "Internal 32.0 kHz", "Internal 44.1 kHz",
+       "Internal 48.0 kHz", "Internal 64.0 kHz", "Internal 88.2 kHz",
+       "Internal 96.0 kHz"
+
+		 Choose between Master at a specific Frequency and so also the
+		 Speed-mode or Slave (Autosync). Also see  "Preferred Sync Ref"
+
+
+       !!!! This is no pure hardware function but was implemented by
+       ALSA by some ALSA-drivers before, so I use it also. !!!
+
+
+    Preferred Sync Ref
+
+       Name -- "Preferred Sync Reference"
+
+       Access -- Read Write
+
+       Values -- "Word" "MADI"
+
+
+		 Within the Auto-sync-Mode the preferred Sync Source can be
+		 chosen. If it is not available another is used if possible.
+
+		 Note: Since MADI has a much higher bit-rate than word-clock, the
+		 card should synchronise better in MADI Mode. But since the
+		 RME-PLL is very good, there are almost no problems with
+		 word-clock too. I never found a difference.
+
+
+    TX 64 channel --- 
+
+       Name -- "TX 64 channels mode"
+
+       Access -- Read Write
+
+       Values -- 0 1
+
+		 Using 64-channel-modus (1) or 56-channel-modus for
+		 MADI-transmission (0).
+
+
+		 Note: This control is for output only. Input-mode is detected
+		 automatically from hardware sending MADI.
+
+
+    Clear TMS ---
+
+       Name -- "Clear Track Marker"
+
+       Access -- Read Write
+
+       Values -- 0 1
+
+
+		 Don't use to lower 5 Audio-bits on AES as additional Bits.
+        
+
+    Safe Mode oder Auto Input --- 
+
+       Name -- "Safe Mode"
+
+       Access -- Read Write
+
+       Values -- 0 1
+
+       (default on)
+
+		 If on (1), then if either the optical or coaxial connection
+		 has a failure, there is a takeover to the working one, with no
+		 sample failure. Its only useful if you use the second as a
+		 backup connection.
+
+    Input --- 
+
+       Name -- "Input Select"
+
+       Access -- Read Write
+
+       Values -- optical coaxial
+
+
+		 Choosing the Input, optical or coaxial. If Safe-mode is active,
+		 this is the preferred Input.
+
+-------------- Mixer ----------------------
+
+    Mixer
+
+       Name -- "Mixer"
+
+       Access -- Read Write
+
+       Values - <channel-number 0-127> <Value 0-65535>
+
+
+		 Here as a first value the channel-index is taken to get/set the
+		 corresponding mixer channel, where 0-63 are the input to output
+		 fader and 64-127 the playback to outputs fader. Value 0
+		 is channel muted 0 and 32768 an amplification of  1.
+
+    Chn 1-64
+
+       fast mixer for the ALSA-mixer utils. The diagonal of the
+       mixer-matrix is implemented from playback to output.
+       
+
+    Line Out
+
+       Name  -- "Line Out"
+
+       Access -- Read Write
+
+       Values -- 0 1
+
+		 Switching on and off the analog out, which has nothing to do
+		 with mixing or routing. the analog outs reflects channel 63,64.
+
+
+--- information (only read access):
+ 
+    Sample Rate
+
+       Name -- "System Sample Rate"
+
+       Access -- Read-only
+
+		 getting the sample rate.
+
+
+    External Rate measured
+
+       Name -- "External Rate"
+
+       Access -- Read only
+
+
+		 Should be "Autosync Rate", but Name used is
+		 ALSA-Scheme. External Sample frequency liked used on Autosync is
+		 reported.
+
+
+    MADI Sync Status
+
+       Name -- "MADI Sync Lock Status"
+
+       Access -- Read
+
+       Values -- 0,1,2
+
+       MADI-Input is 0=Unlocked, 1=Locked, or 2=Synced.
+
+
+    Word Clock Sync Status
+
+       Name -- "Word Clock Lock Status"
+
+       Access -- Read
+
+       Values -- 0,1,2
+
+       Word Clock Input is 0=Unlocked, 1=Locked, or 2=Synced.
+
+    AutoSync
+
+       Name -- "AutoSync Reference"
+
+       Access -- Read
+
+       Values -- "WordClock", "MADI", "None"
+
+		 Sync-Reference is either "WordClock", "MADI" or none.
+
+   RX 64ch --- noch nicht implementiert
+
+       MADI-Receiver is in 64 channel mode oder 56 channel mode.
+
+
+   AB_inp   --- not tested 
+
+		 Used input for Auto-Input.
+
+
+   actual Buffer Position --- not implemented
+
+	   !!! this is a ALSA internal function, so no control is used !!!
+
+
+
+Calling Parameter:
+
+   index int array (min = 1, max = 8), 
+     "Index value for RME HDSPM interface." card-index within ALSA
+
+     note: ALSA-standard
+
+   id string array (min = 1, max = 8), 
+     "ID string for RME HDSPM interface."
+
+     note: ALSA-standard
+
+   enable int array (min = 1, max = 8), 
+     "Enable/disable specific HDSPM sound-cards."
+
+     note: ALSA-standard
+
+   precise_ptr int array (min = 1, max = 8), 
+     "Enable precise pointer, or disable."
+
+     note: Use only when the application supports this (which is a special case).
+
+   line_outs_monitor int array (min = 1, max = 8), 
+     "Send playback streams to analog outs by default."
+
+
+	  note: each playback channel is mixed to the same numbered output
+	  channel (routed). This is against the ALSA-convention, where all
+	  channels have to be muted on after loading the driver, but was
+	  used before on other cards, so i historically use it again)
+
+
+
+   enable_monitor int array (min = 1, max = 8), 
+     "Enable Analog Out on Channel 63/64 by default."
+
+      note: here the analog output is enabled (but not routed).
\ No newline at end of file
diff --git a/Documentation/sound/alsa/powersave.txt b/Documentation/sound/alsa/powersave.txt
new file mode 100644
index 0000000..9657e80
--- /dev/null
+++ b/Documentation/sound/alsa/powersave.txt
@@ -0,0 +1,41 @@
+Notes on Power-Saving Mode
+==========================
+
+AC97 and HD-audio drivers have the automatic power-saving mode.
+This feature is enabled via Kconfig CONFIG_SND_AC97_POWER_SAVE
+and CONFIG_SND_HDA_POWER_SAVE options, respectively.
+
+With the automatic power-saving, the driver turns off the codec power
+appropriately when no operation is required.  When no applications use
+the device and/or no analog loopback is set, the power disablement is
+done fully or partially.  It'll save a certain power consumption, thus
+good for laptops (even for desktops).
+
+The time-out for automatic power-off can be specified via power_save
+module option of snd-ac97-codec and snd-hda-intel modules.  Specify
+the time-out value in seconds.  0 means to disable the automatic
+power-saving.  The default value of timeout is given via
+CONFIG_SND_AC97_POWER_SAVE_DEFAULT and
+CONFIG_SND_HDA_POWER_SAVE_DEFAULT Kconfig options.  Setting this to 1
+(the minimum value) isn't recommended because many applications try to
+reopen the device frequently.  10 would be a good choice for normal
+operations.
+
+The power_save option is exported as writable.  This means you can
+adjust the value via sysfs on the fly.  For example, to turn on the
+automatic power-save mode with 10 seconds, write to
+/sys/modules/snd_ac97_codec/parameters/power_save (usually as root):
+
+	# echo 10 > /sys/modules/snd_ac97_codec/parameters/power_save
+
+
+Note that you might hear click noise/pop when changing the power
+state.  Also, it often takes certain time to wake up from the
+power-down to the active state.  These are often hardly to fix, so
+don't report extra bug reports unless you have a fix patch ;-)
+
+For HD-audio interface, there is another module option,
+power_save_controller.  This enables/disables the power-save mode of
+the controller side.  Setting this on may reduce a bit more power
+consumption, but might result in longer wake-up time and click noise.
+Try to turn it off when you experience such a thing too often.
diff --git a/Documentation/sound/alsa/seq_oss.html b/Documentation/sound/alsa/seq_oss.html
new file mode 100644
index 0000000..d9776cf
--- /dev/null
+++ b/Documentation/sound/alsa/seq_oss.html
@@ -0,0 +1,409 @@
+<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
+<HTML>
+<HEAD>
+   <TITLE>OSS Sequencer Emulation on ALSA</TITLE>
+</HEAD>
+<BODY>
+
+<CENTER>
+<H1>
+
+<HR WIDTH="100%"></H1></CENTER>
+
+<CENTER>
+<H1>
+OSS Sequencer Emulation on ALSA</H1></CENTER>
+
+<HR WIDTH="100%">
+<P>Copyright (c) 1998,1999 by Takashi Iwai
+<TT><A HREF="mailto:iwai@ww.uni-erlangen.de">&lt;iwai@ww.uni-erlangen.de></A></TT>
+<P>ver.0.1.8; Nov. 16, 1999
+<H2>
+
+<HR WIDTH="100%"></H2>
+
+<H2>
+1. Description</H2>
+This directory contains the OSS sequencer emulation driver on ALSA. Note
+that this program is still in the development state.
+<P>What this does - it provides the emulation of the OSS sequencer, access
+via
+<TT>/dev/sequencer</TT> and <TT>/dev/music</TT> devices.
+The most of applications using OSS can run if the appropriate ALSA
+sequencer is prepared.
+<P>The following features are emulated by this driver:
+<UL>
+<LI>
+Normal sequencer and MIDI events:</LI>
+
+<BR>They are converted to the ALSA sequencer events, and sent to the corresponding
+port.
+<LI>
+Timer events:</LI>
+
+<BR>The timer is not selectable by ioctl. The control rate is fixed to
+100 regardless of HZ. That is, even on Alpha system, a tick is always
+1/100 second. The base rate and tempo can be changed in <TT>/dev/music</TT>.
+
+<LI>
+Patch loading:</LI>
+
+<BR>It purely depends on the synth drivers whether it's supported since
+the patch loading is realized by callback to the synth driver.
+<LI>
+I/O controls:</LI>
+
+<BR>Most of controls are accepted. Some controls
+are dependent on the synth driver, as well as even on original OSS.</UL>
+Furthermore, you can find the following advanced features:
+<UL>
+<LI>
+Better queue mechanism:</LI>
+
+<BR>The events are queued before processing them.
+<LI>
+Multiple applications:</LI>
+
+<BR>You can run two or more applications simultaneously (even for OSS sequencer)!
+However, each MIDI device is exclusive - that is, if a MIDI device is opened
+once by some application, other applications can't use it. No such a restriction
+in synth devices.
+<LI>
+Real-time event processing:</LI>
+
+<BR>The events can be processed in real time without using out of bound
+ioctl. To switch to real-time mode, send ABSTIME 0 event. The followed
+events will be processed in real-time without queued. To switch off the
+real-time mode, send RELTIME 0 event.
+<LI>
+<TT>/proc</TT> interface:</LI>
+
+<BR>The status of applications and devices can be shown via <TT>/proc/asound/seq/oss</TT>
+at any time. In the later version, configuration will be changed via <TT>/proc</TT>
+interface, too.</UL>
+
+<H2>
+2. Installation</H2>
+Run configure script with both sequencer support (<TT>--with-sequencer=yes</TT>)
+and OSS emulation (<TT>--with-oss=yes</TT>) options. A module <TT>snd-seq-oss.o</TT>
+will be created. If the synth module of your sound card supports for OSS
+emulation (so far, only Emu8000 driver), this module will be loaded automatically.
+Otherwise, you need to load this module manually.
+<P>At beginning, this module probes all the MIDI ports which have been
+already connected to the sequencer. Once after that, the creation and deletion
+of ports are watched by announcement mechanism of ALSA sequencer.
+<P>The available synth and MIDI devices can be found in proc interface.
+Run "<TT>cat /proc/asound/seq/oss</TT>", and check the devices. For example,
+if you use an AWE64 card, you'll see like the following:
+<PRE>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; OSS sequencer emulation version 0.1.8
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ALSA client number 63
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ALSA receiver port 0
+
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Number of applications: 0
+
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Number of synth devices: 1
+
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; synth 0: [EMU8000]
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; type 0x1 : subtype 0x20 : voices 32
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; capabilties : ioctl enabled / load_patch enabled
+
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Number of MIDI devices: 3
+
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; midi 0: [Emu8000 Port-0] ALSA port 65:0
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; capability write / opened none
+
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; midi 1: [Emu8000 Port-1] ALSA port 65:1
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; capability write / opened none
+
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; midi 2: [0: MPU-401 (UART)] ALSA port 64:0
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; capability read/write / opened none</PRE>
+Note that the device number may be different from the information of
+<TT>/proc/asound/oss-devices</TT>
+or ones of the original OSS driver. Use the device number listed in <TT>/proc/asound/seq/oss</TT>
+to play via OSS sequencer emulation.
+<H2>
+3. Using Synthesizer Devices</H2>
+Run your favorite program. I've tested playmidi-2.4, awemidi-0.4.3, gmod-3.1
+and xmp-1.1.5. You can load samples via <TT>/dev/sequencer</TT> like sfxload,
+too.
+<P>If the lowlevel driver supports multiple access to synth devices (like
+Emu8000 driver), two or more applications are allowed to run at the same
+time.
+<H2>
+4. Using MIDI Devices</H2>
+So far, only MIDI output was tested. MIDI input was not checked at all,
+but hopefully it will work. Use the device number listed in <TT>/proc/asound/seq/oss</TT>.
+Be aware that these numbers are mostly different from the list in
+<TT>/proc/asound/oss-devices</TT>.
+<H2>
+5. Module Options</H2>
+The following module options are available:
+<UL>
+<LI>
+<TT>maxqlen</TT></LI>
+
+<BR>specifies the maximum read/write queue length. This queue is private
+for OSS sequencer, so that it is independent from the queue length of ALSA
+sequencer. Default value is 1024.
+<LI>
+<TT>seq_oss_debug</TT></LI>
+
+<BR>specifies the debug level and accepts zero (= no debug message) or
+positive integer. Default value is 0.</UL>
+
+<H2>
+6. Queue Mechanism</H2>
+OSS sequencer emulation uses an ALSA priority queue. The
+events from <TT>/dev/sequencer</TT> are processed and put onto the queue
+specified by module option.
+<P>All the events from <TT>/dev/sequencer</TT> are parsed at beginning.
+The timing events are also parsed at this moment, so that the events may
+be processed in real-time. Sending an event ABSTIME 0 switches the operation
+mode to real-time mode, and sending an event RELTIME 0 switches it off.
+In the real-time mode, all events are dispatched immediately.
+<P>The queued events are dispatched to the corresponding ALSA sequencer
+ports after scheduled time by ALSA sequencer dispatcher.
+<P>If the write-queue is full, the application sleeps until a certain amount
+(as default one half) becomes empty in blocking mode. The synchronization
+to write timing was implemented, too.
+<P>The input from MIDI devices or echo-back events are stored on read FIFO
+queue. If application reads <TT>/dev/sequencer</TT> in blocking mode, the
+process will be awaked.
+
+<H2>
+7. Interface to Synthesizer Device</H2>
+
+<H3>
+7.1. Registration</H3>
+To register an OSS synthesizer device, use <TT>snd_seq_oss_synth_register</TT>
+function.
+<PRE>int snd_seq_oss_synth_register(char *name, int type, int subtype, int nvoices,
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; snd_seq_oss_callback_t *oper, void *private_data)</PRE>
+The arguments <TT>name</TT>, <TT>type</TT>, <TT>subtype</TT> and
+<TT>nvoices</TT>
+are used for making the appropriate synth_info structure for ioctl. The
+return value is an index number of this device. This index must be remembered
+for unregister. If registration is failed, -errno will be returned.
+<P>To release this device, call <TT>snd_seq_oss_synth_unregister function</TT>:
+<PRE>int snd_seq_oss_synth_unregister(int index),</PRE>
+where the <TT>index</TT> is the index number returned by register function.
+<H3>
+7.2. Callbacks</H3>
+OSS synthesizer devices have capability for sample downloading and ioctls
+like sample reset. In OSS emulation, these special features are realized
+by using callbacks. The registration argument oper is used to specify these
+callbacks. The following callback functions must be defined:
+<PRE>snd_seq_oss_callback_t:
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int (*open)(snd_seq_oss_arg_t *p, void *closure);
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int (*close)(snd_seq_oss_arg_t *p);
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int (*ioctl)(snd_seq_oss_arg_t *p, unsigned int cmd, unsigned long arg);
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int (*load_patch)(snd_seq_oss_arg_t *p, int format, const char *buf, int offs, int count);
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int (*reset)(snd_seq_oss_arg_t *p);
+Except for <TT>open</TT> and <TT>close</TT> callbacks, they are allowed
+to be NULL.
+<P>Each callback function takes the argument type snd_seq_oss_arg_t as the
+first argument.
+<PRE>struct snd_seq_oss_arg_t {
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int app_index;
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int file_mode;
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int seq_mode;
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; snd_seq_addr_t addr;
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; void *private_data;
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int event_passing;
+};</PRE>
+The first three fields, <TT>app_index</TT>, <TT>file_mode</TT> and
+<TT>seq_mode</TT>
+are initialized by OSS sequencer. The <TT>app_index</TT> is the application
+index which is unique to each application opening OSS sequencer. The
+<TT>file_mode</TT>
+is bit-flags indicating the file operation mode. See
+<TT>seq_oss.h</TT>
+for its meaning. The <TT>seq_mode</TT> is sequencer operation mode. In
+the current version, only <TT>SND_OSSSEQ_MODE_SYNTH</TT> is used.
+<P>The next two fields, <TT>addr</TT> and <TT>private_data</TT>, must be
+filled by the synth driver at open callback. The <TT>addr</TT> contains
+the address of ALSA sequencer port which is assigned to this device. If
+the driver allocates memory for <TT>private_data</TT>, it must be released
+in close callback by itself.
+<P>The last field, <TT>event_passing</TT>, indicates how to translate note-on
+/ off events. In <TT>PROCESS_EVENTS</TT> mode, the note 255 is regarded
+as velocity change, and key pressure event is passed to the port. In <TT>PASS_EVENTS</TT>
+mode, all note on/off events are passed to the port without modified. <TT>PROCESS_KEYPRESS</TT>
+mode checks the note above 128 and regards it as key pressure event (mainly
+for Emu8000 driver).
+<H4>
+7.2.1. Open Callback</H4>
+The <TT>open</TT> is called at each time this device is opened by an application
+using OSS sequencer. This must not be NULL. Typically, the open callback
+does the following procedure:
+<OL>
+<LI>
+Allocate private data record.</LI>
+
+<LI>
+Create an ALSA sequencer port.</LI>
+
+<LI>
+Set the new port address on arg->addr.</LI>
+
+<LI>
+Set the private data record pointer on arg->private_data.</LI>
+</OL>
+Note that the type bit-flags in port_info of this synth port must NOT contain
+<TT>TYPE_MIDI_GENERIC</TT>
+bit. Instead, <TT>TYPE_SPECIFIC</TT> should be used. Also, <TT>CAP_SUBSCRIPTION</TT>
+bit should NOT be included, too. This is necessary to tell it from other
+normal MIDI devices. If the open procedure succeeded, return zero. Otherwise,
+return -errno.
+<H4>
+7.2.2 Ioctl Callback</H4>
+The <TT>ioctl</TT> callback is called when the sequencer receives device-specific
+ioctls. The following two ioctls should be processed by this callback:
+<UL>
+<LI>
+<TT>IOCTL_SEQ_RESET_SAMPLES</TT></LI>
+
+<BR>reset all samples on memory -- return 0
+<LI>
+<TT>IOCTL_SYNTH_MEMAVL</TT></LI>
+
+<BR>return the available memory size
+<LI>
+<TT>FM_4OP_ENABLE</TT></LI>
+
+<BR>can be ignored usually</UL>
+The other ioctls are processed inside the sequencer without passing to
+the lowlevel driver.
+<H4>
+7.2.3 Load_Patch Callback</H4>
+The <TT>load_patch</TT> callback is used for sample-downloading. This callback
+must read the data on user-space and transfer to each device. Return 0
+if succeeded, and -errno if failed. The format argument is the patch key
+in patch_info record. The buf is user-space pointer where patch_info record
+is stored. The offs can be ignored. The count is total data size of this
+sample data.
+<H4>
+7.2.4 Close Callback</H4>
+The <TT>close</TT> callback is called when this device is closed by the
+applicaion. If any private data was allocated in open callback, it must
+be released in the close callback. The deletion of ALSA port should be
+done here, too. This callback must not be NULL.
+<H4>
+7.2.5 Reset Callback</H4>
+The <TT>reset</TT> callback is called when sequencer device is reset or
+closed by applications. The callback should turn off the sounds on the
+relevant port immediately, and initialize the status of the port. If this
+callback is undefined, OSS seq sends a <TT>HEARTBEAT</TT> event to the
+port.
+<H3>
+7.3 Events</H3>
+Most of the events are processed by sequencer and translated to the adequate
+ALSA sequencer events, so that each synth device can receive by input_event
+callback of ALSA sequencer port. The following ALSA events should be implemented
+by the driver:
+<BR>&nbsp;
+<TABLE BORDER WIDTH="75%" NOSAVE >
+<TR NOSAVE>
+<TD NOSAVE><B>ALSA event</B></TD>
+
+<TD><B>Original OSS events</B></TD>
+</TR>
+
+<TR>
+<TD>NOTEON</TD>
+
+<TD>SEQ_NOTEON
+<BR>MIDI_NOTEON</TD>
+</TR>
+
+<TR>
+<TD>NOTE</TD>
+
+<TD>SEQ_NOTEOFF
+<BR>MIDI_NOTEOFF</TD>
+</TR>
+
+<TR NOSAVE>
+<TD NOSAVE>KEYPRESS</TD>
+
+<TD>MIDI_KEY_PRESSURE</TD>
+</TR>
+
+<TR NOSAVE>
+<TD>CHANPRESS</TD>
+
+<TD NOSAVE>SEQ_AFTERTOUCH
+<BR>MIDI_CHN_PRESSURE</TD>
+</TR>
+
+<TR NOSAVE>
+<TD NOSAVE>PGMCHANGE</TD>
+
+<TD NOSAVE>SEQ_PGMCHANGE
+<BR>MIDI_PGM_CHANGE</TD>
+</TR>
+
+<TR>
+<TD>PITCHBEND</TD>
+
+<TD>SEQ_CONTROLLER(CTRL_PITCH_BENDER)
+<BR>MIDI_PITCH_BEND</TD>
+</TR>
+
+<TR>
+<TD>CONTROLLER</TD>
+
+<TD>MIDI_CTL_CHANGE
+<BR>SEQ_BALANCE (with CTL_PAN)</TD>
+</TR>
+
+<TR>
+<TD>CONTROL14</TD>
+
+<TD>SEQ_CONTROLLER</TD>
+</TR>
+
+<TR>
+<TD>REGPARAM</TD>
+
+<TD>SEQ_CONTROLLER(CTRL_PITCH_BENDER_RANGE)</TD>
+</TR>
+
+<TR>
+<TD>SYSEX</TD>
+
+<TD>SEQ_SYSEX</TD>
+</TR>
+</TABLE>
+
+<P>The most of these behavior can be realized by MIDI emulation driver
+included in the Emu8000 lowlevel driver. In the future release, this module
+will be independent.
+<P>Some OSS events (<TT>SEQ_PRIVATE</TT> and <TT>SEQ_VOLUME</TT> events) are passed as event
+type SND_SEQ_OSS_PRIVATE.  The OSS sequencer passes these event 8 byte
+packets without any modification. The lowlevel driver should process these
+events appropriately.
+<H2>
+8. Interface to MIDI Device</H2>
+Since the OSS emulation probes the creation and deletion of ALSA MIDI sequencer
+ports automatically by receiving announcement from ALSA sequencer, the
+MIDI devices don't need to be registered explicitly like synth devices.
+However, the MIDI port_info registered to ALSA sequencer must include a group
+name <TT>SND_SEQ_GROUP_DEVICE</TT> and a capability-bit <TT>CAP_READ</TT> or
+<TT>CAP_WRITE</TT>. Also, subscription capabilities, <TT>CAP_SUBS_READ</TT> or <TT>CAP_SUBS_WRITE</TT>,
+must be defined, too. If these conditions are not satisfied, the port is not
+registered as OSS sequencer MIDI device.
+<P>The events via MIDI devices are parsed in OSS sequencer and converted
+to the corresponding ALSA sequencer events. The input from MIDI sequencer
+is also converted to MIDI byte events by OSS sequencer. This works just
+a reverse way of seq_midi module.
+<H2>
+9. Known Problems / TODO's</H2>
+
+<UL>
+<LI>
+Patch loading via ALSA instrument layer is not implemented yet.</LI>
+</UL>
+
+</BODY>
+</HTML>
diff --git a/Documentation/sound/alsa/serial-u16550.txt b/Documentation/sound/alsa/serial-u16550.txt
new file mode 100644
index 0000000..c191955
--- /dev/null
+++ b/Documentation/sound/alsa/serial-u16550.txt
@@ -0,0 +1,88 @@
+
+			Serial UART 16450/16550 MIDI driver
+			===================================
+
+The adaptor module parameter allows you to select either:
+
+  0 - Roland Soundcanvas support (default)
+  1 - Midiator MS-124T support (1)
+  2 - Midiator MS-124W S/A mode (2)
+  3 - MS-124W M/B mode support (3)
+  4 - Generic device with multiple input support (4)
+
+For the Midiator MS-124W, you must set the physical M-S and A-B
+switches on the Midiator to match the driver mode you select.
+
+In Roland Soundcanvas mode, multiple ALSA raw MIDI substreams are supported
+(midiCnD0-midiCnD15).  Whenever you write to a different substream, the driver
+sends the nonstandard MIDI command sequence F5 NN, where NN is the substream
+number plus 1.  Roland modules use this command to switch between different
+"parts", so this feature lets you treat each part as a distinct raw MIDI
+substream. The driver provides no way to send F5 00 (no selection) or to not
+send the F5 NN command sequence at all; perhaps it ought to.
+
+Usage example for simple serial converter:
+
+	/sbin/setserial /dev/ttyS0 uart none
+	/sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 speed=115200
+
+Usage example for Roland SoundCanvas with 4 MIDI ports:
+
+	/sbin/setserial /dev/ttyS0 uart none
+	/sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 outs=4
+
+In MS-124T mode, one raw MIDI substream is supported (midiCnD0); the outs
+module parameter is automatically set to 1. The driver sends the same data to
+all four MIDI Out connectors.  Set the A-B switch and the speed module
+parameter to match (A=19200, B=9600).
+
+Usage example for MS-124T, with A-B switch in A position:
+
+	/sbin/setserial /dev/ttyS0 uart none
+	/sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=1 \
+			speed=19200
+
+In MS-124W S/A mode, one raw MIDI substream is supported (midiCnD0);
+the outs module parameter is automatically set to 1. The driver sends
+the same data to all four MIDI Out connectors at full MIDI speed.
+
+Usage example for S/A mode:
+
+	/sbin/setserial /dev/ttyS0 uart none
+	/sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=2
+
+In MS-124W M/B mode, the driver supports 16 ALSA raw MIDI substreams;
+the outs module parameter is automatically set to 16.  The substream
+number gives a bitmask of which MIDI Out connectors the data should be
+sent to, with midiCnD1 sending to Out 1, midiCnD2 to Out 2, midiCnD4 to
+Out 3, and midiCnD8 to Out 4.  Thus midiCnD15 sends the data to all 4 ports.
+As a special case, midiCnD0 also sends to all ports, since it is not useful
+to send the data to no ports.  M/B mode has extra overhead to select the MIDI
+Out for each byte, so the aggregate data rate across all four MIDI Outs is
+at most one byte every 520 us, as compared with the full MIDI data rate of
+one byte every 320 us per port.
+
+Usage example for M/B mode:
+
+	/sbin/setserial /dev/ttyS0 uart none
+	/sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=3
+
+The MS-124W hardware's M/A mode is currently not supported. This mode allows
+the MIDI Outs to act independently at double the aggregate throughput of M/B,
+but does not allow sending the same byte simultaneously to multiple MIDI Outs. 
+The M/A protocol requires the driver to twiddle the modem control lines under
+timing constraints, so it would be a bit more complicated to implement than
+the other modes.
+
+Midiator models other than MS-124W and MS-124T are currently not supported. 
+Note that the suffix letter is significant; the MS-124 and MS-124B are not
+compatible, nor are the other known models MS-101, MS-101B, MS-103, and MS-114.
+I do have documentation (tim.mann@compaq.com) that partially covers these models,
+but no units to experiment with.  The MS-124W support is tested with a real unit.
+The MS-124T support is untested, but should work.
+
+The Generic driver supports multiple input and output substreams over a single
+serial port.  Similar to Roland Soundcanvas mode, F5 NN is used to select the
+appropriate input or output stream (depending on the data direction).
+Additionally, the CTS signal is used to regulate the data flow.  The number of
+inputs is specified by the ins parameter.
diff --git a/Documentation/sound/alsa/soc/DAI.txt b/Documentation/sound/alsa/soc/DAI.txt
new file mode 100644
index 0000000..0ebd7ea
--- /dev/null
+++ b/Documentation/sound/alsa/soc/DAI.txt
@@ -0,0 +1,56 @@
+ASoC currently supports the three main Digital Audio Interfaces (DAI) found on
+SoC controllers and portable audio CODECs today, namely AC97, I2S and PCM.
+
+
+AC97
+====
+
+  AC97 is a five wire interface commonly found on many PC sound cards. It is
+now also popular in many portable devices. This DAI has a reset line and time
+multiplexes its data on its SDATA_OUT (playback) and SDATA_IN (capture) lines.
+The bit clock (BCLK) is always driven by the CODEC (usually 12.288MHz) and the
+frame (FRAME) (usually 48kHz) is always driven by the controller. Each AC97
+frame is 21uS long and is divided into 13 time slots.
+
+The AC97 specification can be found at :-
+http://www.intel.com/design/chipsets/audio/ac97_r23.pdf
+
+
+I2S
+===
+
+ I2S is a common 4 wire DAI used in HiFi, STB and portable devices. The Tx and
+Rx lines are used for audio transmission, whilst the bit clock (BCLK) and
+left/right clock (LRC) synchronise the link. I2S is flexible in that either the
+controller or CODEC can drive (master) the BCLK and LRC clock lines. Bit clock
+usually varies depending on the sample rate and the master system clock
+(SYSCLK). LRCLK is the same as the sample rate. A few devices support separate
+ADC and DAC LRCLKs, this allows for simultaneous capture and playback at
+different sample rates.
+
+I2S has several different operating modes:-
+
+ o I2S - MSB is transmitted on the falling edge of the first BCLK after LRC
+         transition.
+
+ o Left Justified - MSB is transmitted on transition of LRC.
+
+ o Right Justified - MSB is transmitted sample size BCLKs before LRC
+                     transition.
+
+PCM
+===
+
+PCM is another 4 wire interface, very similar to I2S, which can support a more
+flexible protocol. It has bit clock (BCLK) and sync (SYNC) lines that are used
+to synchronise the link whilst the Tx and Rx lines are used to transmit and
+receive the audio data. Bit clock usually varies depending on sample rate
+whilst sync runs at the sample rate. PCM also supports Time Division
+Multiplexing (TDM) in that several devices can use the bus simultaneously (this
+is sometimes referred to as network mode).
+
+Common PCM operating modes:-
+
+ o Mode A - MSB is transmitted on falling edge of first BCLK after FRAME/SYNC.
+
+ o Mode B - MSB is transmitted on rising edge of FRAME/SYNC.
diff --git a/Documentation/sound/alsa/soc/clocking.txt b/Documentation/sound/alsa/soc/clocking.txt
new file mode 100644
index 0000000..b130016
--- /dev/null
+++ b/Documentation/sound/alsa/soc/clocking.txt
@@ -0,0 +1,51 @@
+Audio Clocking
+==============
+
+This text describes the audio clocking terms in ASoC and digital audio in
+general. Note: Audio clocking can be complex!
+
+
+Master Clock
+------------
+
+Every audio subsystem is driven by a master clock (sometimes referred to as MCLK
+or SYSCLK). This audio master clock can be derived from a number of sources
+(e.g. crystal, PLL, CPU clock) and is responsible for producing the correct
+audio playback and capture sample rates.
+
+Some master clocks (e.g. PLLs and CPU based clocks) are configurable in that
+their speed can be altered by software (depending on the system use and to save
+power). Other master clocks are fixed at a set frequency (i.e. crystals).
+
+
+DAI Clocks
+----------
+The Digital Audio Interface is usually driven by a Bit Clock (often referred to
+as BCLK). This clock is used to drive the digital audio data across the link
+between the codec and CPU.
+
+The DAI also has a frame clock to signal the start of each audio frame. This
+clock is sometimes referred to as LRC (left right clock) or FRAME. This clock
+runs at exactly the sample rate (LRC = Rate).
+
+Bit Clock can be generated as follows:-
+
+BCLK = MCLK / x
+
+ or
+
+BCLK = LRC * x
+
+ or
+
+BCLK = LRC * Channels * Word Size
+
+This relationship depends on the codec or SoC CPU in particular. In general
+it is best to configure BCLK to the lowest possible speed (depending on your
+rate, number of channels and word size) to save on power.
+
+It is also desirable to use the codec (if possible) to drive (or master) the
+audio clocks as it usually gives more accurate sample rates than the CPU.
+
+
+
diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt
new file mode 100644
index 0000000..1e95342
--- /dev/null
+++ b/Documentation/sound/alsa/soc/codec.txt
@@ -0,0 +1,198 @@
+ASoC Codec Driver
+=================
+
+The codec driver is generic and hardware independent code that configures the
+codec to provide audio capture and playback. It should contain no code that is
+specific to the target platform or machine. All platform and machine specific
+code should be added to the platform and machine drivers respectively.
+
+Each codec driver *must* provide the following features:-
+
+ 1) Codec DAI and PCM configuration
+ 2) Codec control IO - using I2C, 3 Wire(SPI) or both APIs
+ 3) Mixers and audio controls
+ 4) Codec audio operations
+
+Optionally, codec drivers can also provide:-
+
+ 5) DAPM description.
+ 6) DAPM event handler.
+ 7) DAC Digital mute control.
+
+Its probably best to use this guide in conjunction with the existing codec
+driver code in sound/soc/codecs/
+
+ASoC Codec driver breakdown
+===========================
+
+1 - Codec DAI and PCM configuration
+-----------------------------------
+Each codec driver must have a struct snd_soc_codec_dai to define its DAI and
+PCM capabilities and operations. This struct is exported so that it can be
+registered with the core by your machine driver.
+
+e.g.
+
+struct snd_soc_codec_dai wm8731_dai = {
+	.name = "WM8731",
+	/* playback capabilities */
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8731_RATES,
+		.formats = WM8731_FORMATS,},
+	/* capture capabilities */
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8731_RATES,
+		.formats = WM8731_FORMATS,},
+	/* pcm operations - see section 4 below */
+	.ops = {
+		.prepare = wm8731_pcm_prepare,
+		.hw_params = wm8731_hw_params,
+		.shutdown = wm8731_shutdown,
+	},
+	/* DAI operations - see DAI.txt */
+	.dai_ops = {
+		.digital_mute = wm8731_mute,
+		.set_sysclk = wm8731_set_dai_sysclk,
+		.set_fmt = wm8731_set_dai_fmt,
+	}
+};
+EXPORT_SYMBOL_GPL(wm8731_dai);
+
+
+2 - Codec control IO
+--------------------
+The codec can usually be controlled via an I2C or SPI style interface
+(AC97 combines control with data in the DAI). The codec drivers provide
+functions to read and write the codec registers along with supplying a
+register cache:-
+
+	/* IO control data and register cache */
+	void *control_data; /* codec control (i2c/3wire) data */
+	void *reg_cache;
+
+Codec read/write should do any data formatting and call the hardware
+read write below to perform the IO. These functions are called by the
+core and ALSA when performing DAPM or changing the mixer:-
+
+    unsigned int (*read)(struct snd_soc_codec *, unsigned int);
+    int (*write)(struct snd_soc_codec *, unsigned int, unsigned int);
+
+Codec hardware IO functions - usually points to either the I2C, SPI or AC97
+read/write:-
+
+	hw_write_t hw_write;
+	hw_read_t hw_read;
+
+
+3 - Mixers and audio controls
+-----------------------------
+All the codec mixers and audio controls can be defined using the convenience
+macros defined in soc.h.
+
+    #define SOC_SINGLE(xname, reg, shift, mask, invert)
+
+Defines a single control as follows:-
+
+  xname = Control name e.g. "Playback Volume"
+  reg = codec register
+  shift = control bit(s) offset in register
+  mask = control bit size(s) e.g. mask of 7 = 3 bits
+  invert = the control is inverted
+
+Other macros include:-
+
+    #define SOC_DOUBLE(xname, reg, shift_left, shift_right, mask, invert)
+
+A stereo control
+
+    #define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, mask, invert)
+
+A stereo control spanning 2 registers
+
+    #define SOC_ENUM_SINGLE(xreg, xshift, xmask, xtexts)
+
+Defines an single enumerated control as follows:-
+
+   xreg = register
+   xshift = control bit(s) offset in register
+   xmask = control bit(s) size
+   xtexts = pointer to array of strings that describe each setting
+
+   #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts)
+
+Defines a stereo enumerated control
+
+
+4 - Codec Audio Operations
+--------------------------
+The codec driver also supports the following ALSA operations:-
+
+/* SoC audio ops */
+struct snd_soc_ops {
+	int (*startup)(struct snd_pcm_substream *);
+	void (*shutdown)(struct snd_pcm_substream *);
+	int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *);
+	int (*hw_free)(struct snd_pcm_substream *);
+	int (*prepare)(struct snd_pcm_substream *);
+};
+
+Please refer to the ALSA driver PCM documentation for details.
+http://www.alsa-project.org/~iwai/writing-an-alsa-driver/c436.htm
+
+
+5 - DAPM description.
+---------------------
+The Dynamic Audio Power Management description describes the codec power
+components and their relationships and registers to the ASoC core.
+Please read dapm.txt for details of building the description.
+
+Please also see the examples in other codec drivers.
+
+
+6 - DAPM event handler
+----------------------
+This function is a callback that handles codec domain PM calls and system
+domain PM calls (e.g. suspend and resume). It is used to put the codec
+to sleep when not in use.
+
+Power states:-
+
+	SNDRV_CTL_POWER_D0: /* full On */
+	/* vref/mid, clk and osc on, active */
+
+	SNDRV_CTL_POWER_D1: /* partial On */
+	SNDRV_CTL_POWER_D2: /* partial On */
+
+	SNDRV_CTL_POWER_D3hot: /* Off, with power */
+	/* everything off except vref/vmid, inactive */
+
+	SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */
+
+
+7 - Codec DAC digital mute control
+----------------------------------
+Most codecs have a digital mute before the DACs that can be used to
+minimise any system noise.  The mute stops any digital data from
+entering the DAC.
+
+A callback can be created that is called by the core for each codec DAI
+when the mute is applied or freed.
+
+i.e.
+
+static int wm8974_mute(struct snd_soc_codec *codec,
+	struct snd_soc_codec_dai *dai, int mute)
+{
+	u16 mute_reg = wm8974_read_reg_cache(codec, WM8974_DAC) & 0xffbf;
+	if(mute)
+		wm8974_write(codec, WM8974_DAC, mute_reg | 0x40);
+	else
+		wm8974_write(codec, WM8974_DAC, mute_reg);
+	return 0;
+}
diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt
new file mode 100644
index 0000000..9ac842b
--- /dev/null
+++ b/Documentation/sound/alsa/soc/dapm.txt
@@ -0,0 +1,294 @@
+Dynamic Audio Power Management for Portable Devices
+===================================================
+
+1. Description
+==============
+
+Dynamic Audio Power Management (DAPM) is designed to allow portable
+Linux devices to use the minimum amount of power within the audio
+subsystem at all times. It is independent of other kernel PM and as
+such, can easily co-exist with the other PM systems.
+
+DAPM is also completely transparent to all user space applications as
+all power switching is done within the ASoC core. No code changes or
+recompiling are required for user space applications. DAPM makes power
+switching decisions based upon any audio stream (capture/playback)
+activity and audio mixer settings within the device.
+
+DAPM spans the whole machine. It covers power control within the entire
+audio subsystem, this includes internal codec power blocks and machine
+level power systems.
+
+There are 4 power domains within DAPM
+
+   1. Codec domain - VREF, VMID (core codec and audio power)
+      Usually controlled at codec probe/remove and suspend/resume, although
+      can be set at stream time if power is not needed for sidetone, etc.
+
+   2. Platform/Machine domain - physically connected inputs and outputs
+      Is platform/machine and user action specific, is configured by the
+      machine driver and responds to asynchronous events e.g when HP
+      are inserted
+
+   3. Path domain - audio susbsystem signal paths
+      Automatically set when mixer and mux settings are changed by the user.
+      e.g. alsamixer, amixer.
+
+   4. Stream domain - DACs and ADCs.
+      Enabled and disabled when stream playback/capture is started and
+      stopped respectively. e.g. aplay, arecord.
+
+All DAPM power switching decisions are made automatically by consulting an audio
+routing map of the whole machine. This map is specific to each machine and
+consists of the interconnections between every audio component (including
+internal codec components). All audio components that effect power are called
+widgets hereafter.
+
+
+2. DAPM Widgets
+===============
+
+Audio DAPM widgets fall into a number of types:-
+
+ o Mixer      - Mixes several analog signals into a single analog signal.
+ o Mux        - An analog switch that outputs only one of many inputs.
+ o PGA        - A programmable gain amplifier or attenuation widget.
+ o ADC        - Analog to Digital Converter
+ o DAC        - Digital to Analog Converter
+ o Switch     - An analog switch
+ o Input      - A codec input pin
+ o Output     - A codec output pin
+ o Headphone  - Headphone (and optional Jack)
+ o Mic        - Mic (and optional Jack)
+ o Line       - Line Input/Output (and optional Jack)
+ o Speaker    - Speaker
+ o Supply     - Power or clock supply widget used by other widgets.
+ o Pre        - Special PRE widget (exec before all others)
+ o Post       - Special POST widget (exec after all others)
+
+(Widgets are defined in include/sound/soc-dapm.h)
+
+Widgets are usually added in the codec driver and the machine driver. There are
+convenience macros defined in soc-dapm.h that can be used to quickly build a
+list of widgets of the codecs and machines DAPM widgets.
+
+Most widgets have a name, register, shift and invert. Some widgets have extra
+parameters for stream name and kcontrols.
+
+
+2.1 Stream Domain Widgets
+-------------------------
+
+Stream Widgets relate to the stream power domain and only consist of ADCs
+(analog to digital converters) and DACs (digital to analog converters).
+
+Stream widgets have the following format:-
+
+SND_SOC_DAPM_DAC(name, stream name, reg, shift, invert),
+
+NOTE: the stream name must match the corresponding stream name in your codec
+snd_soc_codec_dai.
+
+e.g. stream widgets for HiFi playback and capture
+
+SND_SOC_DAPM_DAC("HiFi DAC", "HiFi Playback", REG, 3, 1),
+SND_SOC_DAPM_ADC("HiFi ADC", "HiFi Capture", REG, 2, 1),
+
+
+2.2 Path Domain Widgets
+-----------------------
+
+Path domain widgets have a ability to control or affect the audio signal or
+audio paths within the audio subsystem. They have the following form:-
+
+SND_SOC_DAPM_PGA(name, reg, shift, invert, controls, num_controls)
+
+Any widget kcontrols can be set using the controls and num_controls members.
+
+e.g. Mixer widget (the kcontrols are declared first)
+
+/* Output Mixer */
+static const snd_kcontrol_new_t wm8731_output_mixer_controls[] = {
+SOC_DAPM_SINGLE("Line Bypass Switch", WM8731_APANA, 3, 1, 0),
+SOC_DAPM_SINGLE("Mic Sidetone Switch", WM8731_APANA, 5, 1, 0),
+SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0),
+};
+
+SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, wm8731_output_mixer_controls,
+	ARRAY_SIZE(wm8731_output_mixer_controls)),
+
+If you dont want the mixer elements prefixed with the name of the mixer widget,
+you can use SND_SOC_DAPM_MIXER_NAMED_CTL instead. the parameters are the same
+as for SND_SOC_DAPM_MIXER.
+
+2.3 Platform/Machine domain Widgets
+-----------------------------------
+
+Machine widgets are different from codec widgets in that they don't have a
+codec register bit associated with them. A machine widget is assigned to each
+machine audio component (non codec) that can be independently powered. e.g.
+
+ o Speaker Amp
+ o Microphone Bias
+ o Jack connectors
+
+A machine widget can have an optional call back.
+
+e.g. Jack connector widget for an external Mic that enables Mic Bias
+when the Mic is inserted:-
+
+static int spitz_mic_bias(struct snd_soc_dapm_widget* w, int event)
+{
+	gpio_set_value(SPITZ_GPIO_MIC_BIAS, SND_SOC_DAPM_EVENT_ON(event));
+	return 0;
+}
+
+SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias),
+
+
+2.4 Codec Domain
+----------------
+
+The codec power domain has no widgets and is handled by the codecs DAPM event
+handler. This handler is called when the codec powerstate is changed wrt to any
+stream event or by kernel PM events.
+
+
+2.5 Virtual Widgets
+-------------------
+
+Sometimes widgets exist in the codec or machine audio map that don't have any
+corresponding soft power control. In this case it is necessary to create
+a virtual widget - a widget with no control bits e.g.
+
+SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_DAPM_NOPM, 0, 0, NULL, 0),
+
+This can be used to merge to signal paths together in software.
+
+After all the widgets have been defined, they can then be added to the DAPM
+subsystem individually with a call to snd_soc_dapm_new_control().
+
+
+3. Codec Widget Interconnections
+================================
+
+Widgets are connected to each other within the codec and machine by audio paths
+(called interconnections). Each interconnection must be defined in order to
+create a map of all audio paths between widgets.
+
+This is easiest with a diagram of the codec (and schematic of the machine audio
+system), as it requires joining widgets together via their audio signal paths.
+
+e.g., from the WM8731 output mixer (wm8731.c)
+
+The WM8731 output mixer has 3 inputs (sources)
+
+ 1. Line Bypass Input
+ 2. DAC (HiFi playback)
+ 3. Mic Sidetone Input
+
+Each input in this example has a kcontrol associated with it (defined in example
+above) and is connected to the output mixer via it's kcontrol name. We can now
+connect the destination widget (wrt audio signal) with it's source widgets.
+
+	/* output mixer */
+	{"Output Mixer", "Line Bypass Switch", "Line Input"},
+	{"Output Mixer", "HiFi Playback Switch", "DAC"},
+	{"Output Mixer", "Mic Sidetone Switch", "Mic Bias"},
+
+So we have :-
+
+	Destination Widget  <=== Path Name <=== Source Widget
+
+Or:-
+
+	Sink, Path, Source
+
+Or :-
+
+	"Output Mixer" is connected to the "DAC" via the "HiFi Playback Switch".
+
+When there is no path name connecting widgets (e.g. a direct connection) we
+pass NULL for the path name.
+
+Interconnections are created with a call to:-
+
+snd_soc_dapm_connect_input(codec, sink, path, source);
+
+Finally, snd_soc_dapm_new_widgets(codec) must be called after all widgets and
+interconnections have been registered with the core. This causes the core to
+scan the codec and machine so that the internal DAPM state matches the
+physical state of the machine.
+
+
+3.1 Machine Widget Interconnections
+-----------------------------------
+Machine widget interconnections are created in the same way as codec ones and
+directly connect the codec pins to machine level widgets.
+
+e.g. connects the speaker out codec pins to the internal speaker.
+
+	/* ext speaker connected to codec pins LOUT2, ROUT2  */
+	{"Ext Spk", NULL , "ROUT2"},
+	{"Ext Spk", NULL , "LOUT2"},
+
+This allows the DAPM to power on and off pins that are connected (and in use)
+and pins that are NC respectively.
+
+
+4 Endpoint Widgets
+===================
+An endpoint is a start or end point (widget) of an audio signal within the
+machine and includes the codec. e.g.
+
+ o Headphone Jack
+ o Internal Speaker
+ o Internal Mic
+ o Mic Jack
+ o Codec Pins
+
+When a codec pin is NC it can be marked as not used with a call to
+
+snd_soc_dapm_set_endpoint(codec, "Widget Name", 0);
+
+The last argument is 0 for inactive and 1 for active. This way the pin and its
+input widget will never be powered up and consume power.
+
+This also applies to machine widgets. e.g. if a headphone is connected to a
+jack then the jack can be marked active. If the headphone is removed, then
+the headphone jack can be marked inactive.
+
+
+5 DAPM Widget Events
+====================
+
+Some widgets can register their interest with the DAPM core in PM events.
+e.g. A Speaker with an amplifier registers a widget so the amplifier can be
+powered only when the spk is in use.
+
+/* turn speaker amplifier on/off depending on use */
+static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event)
+{
+	gpio_set_value(CORGI_GPIO_APM_ON, SND_SOC_DAPM_EVENT_ON(event));
+	return 0;
+}
+
+/* corgi machine dapm widgets */
+static const struct snd_soc_dapm_widget wm8731_dapm_widgets =
+	SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event);
+
+Please see soc-dapm.h for all other widgets that support events.
+
+
+5.1 Event types
+---------------
+
+The following event types are supported by event widgets.
+
+/* dapm event types */
+#define SND_SOC_DAPM_PRE_PMU	0x1 	/* before widget power up */
+#define SND_SOC_DAPM_POST_PMU	0x2		/* after widget power up */
+#define SND_SOC_DAPM_PRE_PMD	0x4 	/* before widget power down */
+#define SND_SOC_DAPM_POST_PMD	0x8		/* after widget power down */
+#define SND_SOC_DAPM_PRE_REG	0x10	/* before audio path setup */
+#define SND_SOC_DAPM_POST_REG	0x20	/* after audio path setup */
diff --git a/Documentation/sound/alsa/soc/jack.txt b/Documentation/sound/alsa/soc/jack.txt
new file mode 100644
index 0000000..fcf82a4
--- /dev/null
+++ b/Documentation/sound/alsa/soc/jack.txt
@@ -0,0 +1,71 @@
+ASoC jack detection
+===================
+
+ALSA has a standard API for representing physical jacks to user space,
+the kernel side of which can be seen in include/sound/jack.h.  ASoC
+provides a version of this API adding two additional features:
+
+ - It allows more than one jack detection method to work together on one
+   user visible jack.  In embedded systems it is common for multiple
+   to be present on a single jack but handled by separate bits of
+   hardware.
+
+ - Integration with DAPM, allowing DAPM endpoints to be updated
+   automatically based on the detected jack status (eg, turning off the
+   headphone outputs if no headphones are present).
+
+This is done by splitting the jacks up into three things working
+together: the jack itself represented by a struct snd_soc_jack, sets of
+snd_soc_jack_pins representing DAPM endpoints to update and blocks of
+code providing jack reporting mechanisms.
+
+For example, a system may have a stereo headset jack with two reporting
+mechanisms, one for the headphone and one for the microphone.  Some
+systems won't be able to use their speaker output while a headphone is
+connected and so will want to make sure to update both speaker and
+headphone when the headphone jack status changes.
+
+The jack - struct snd_soc_jack
+==============================
+
+This represents a physical jack on the system and is what is visible to
+user space.  The jack itself is completely passive, it is set up by the
+machine driver and updated by jack detection methods.
+
+Jacks are created by the machine driver calling snd_soc_jack_new().
+
+snd_soc_jack_pin
+================
+
+These represent a DAPM pin to update depending on some of the status
+bits supported by the jack.  Each snd_soc_jack has zero or more of these
+which are updated automatically.  They are created by the machine driver
+and associated with the jack using snd_soc_jack_add_pins().  The status
+of the endpoint may configured to be the opposite of the jack status if
+required (eg, enabling a built in microphone if a microphone is not
+connected via a jack).
+
+Jack detection methods
+======================
+
+Actual jack detection is done by code which is able to monitor some
+input to the system and update a jack by calling snd_soc_jack_report(),
+specifying a subset of bits to update.  The jack detection code should
+be set up by the machine driver, taking configuration for the jack to
+update and the set of things to report when the jack is connected.
+
+Often this is done based on the status of a GPIO - a handler for this is
+provided by the snd_soc_jack_add_gpio() function.  Other methods are
+also available, for example integrated into CODECs.  One example of
+CODEC integrated jack detection can be see in the WM8350 driver.
+
+Each jack may have multiple reporting mechanisms, though it will need at
+least one to be useful.
+
+Machine drivers
+===============
+
+These are all hooked together by the machine driver depending on the
+system hardware.  The machine driver will set up the snd_soc_jack and
+the list of pins to update then set up one or more jack detection
+mechanisms to update that jack based on their current status.
diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt
new file mode 100644
index 0000000..bab7711
--- /dev/null
+++ b/Documentation/sound/alsa/soc/machine.txt
@@ -0,0 +1,113 @@
+ASoC Machine Driver
+===================
+
+The ASoC machine (or board) driver is the code that glues together the platform
+and codec drivers.
+
+The machine driver can contain codec and platform specific code. It registers
+the audio subsystem with the kernel as a platform device and is represented by
+the following struct:-
+
+/* SoC machine */
+struct snd_soc_card {
+	char *name;
+
+	int (*probe)(struct platform_device *pdev);
+	int (*remove)(struct platform_device *pdev);
+
+	/* the pre and post PM functions are used to do any PM work before and
+	 * after the codec and DAIs do any PM work. */
+	int (*suspend_pre)(struct platform_device *pdev, pm_message_t state);
+	int (*suspend_post)(struct platform_device *pdev, pm_message_t state);
+	int (*resume_pre)(struct platform_device *pdev);
+	int (*resume_post)(struct platform_device *pdev);
+
+	/* machine stream operations */
+	struct snd_soc_ops *ops;
+
+	/* CPU <--> Codec DAI links  */
+	struct snd_soc_dai_link *dai_link;
+	int num_links;
+};
+
+probe()/remove()
+----------------
+probe/remove are optional. Do any machine specific probe here.
+
+
+suspend()/resume()
+------------------
+The machine driver has pre and post versions of suspend and resume to take care
+of any machine audio tasks that have to be done before or after the codec, DAIs
+and DMA is suspended and resumed. Optional.
+
+
+Machine operations
+------------------
+The machine specific audio operations can be set here. Again this is optional.
+
+
+Machine DAI Configuration
+-------------------------
+The machine DAI configuration glues all the codec and CPU DAIs together. It can
+also be used to set up the DAI system clock and for any machine related DAI
+initialisation e.g. the machine audio map can be connected to the codec audio
+map, unconnected codec pins can be set as such. Please see corgi.c, spitz.c
+for examples.
+
+struct snd_soc_dai_link is used to set up each DAI in your machine. e.g.
+
+/* corgi digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link corgi_dai = {
+	.name = "WM8731",
+	.stream_name = "WM8731",
+	.cpu_dai = &pxa_i2s_dai,
+	.codec_dai = &wm8731_dai,
+	.init = corgi_wm8731_init,
+	.ops = &corgi_ops,
+};
+
+struct snd_soc_card then sets up the machine with it's DAIs. e.g.
+
+/* corgi audio machine driver */
+static struct snd_soc_card snd_soc_corgi = {
+	.name = "Corgi",
+	.dai_link = &corgi_dai,
+	.num_links = 1,
+};
+
+
+Machine Audio Subsystem
+-----------------------
+
+The machine soc device glues the platform, machine and codec driver together.
+Private data can also be set here. e.g.
+
+/* corgi audio private data */
+static struct wm8731_setup_data corgi_wm8731_setup = {
+	.i2c_address = 0x1b,
+};
+
+/* corgi audio subsystem */
+static struct snd_soc_device corgi_snd_devdata = {
+	.machine = &snd_soc_corgi,
+	.platform = &pxa2xx_soc_platform,
+	.codec_dev = &soc_codec_dev_wm8731,
+	.codec_data = &corgi_wm8731_setup,
+};
+
+
+Machine Power Map
+-----------------
+
+The machine driver can optionally extend the codec power map and to become an
+audio power map of the audio subsystem. This allows for automatic power up/down
+of speaker/HP amplifiers, etc. Codec pins can be connected to the machines jack
+sockets in the machine init function. See soc/pxa/spitz.c and dapm.txt for
+details.
+
+
+Machine Controls
+----------------
+
+Machine specific audio mixer controls can be added in the DAI init function.
diff --git a/Documentation/sound/alsa/soc/overview.txt b/Documentation/sound/alsa/soc/overview.txt
new file mode 100644
index 0000000..1e4c6d3
--- /dev/null
+++ b/Documentation/sound/alsa/soc/overview.txt
@@ -0,0 +1,86 @@
+ALSA SoC Layer
+==============
+
+The overall project goal of the ALSA System on Chip (ASoC) layer is to
+provide better ALSA support for embedded system-on-chip processors (e.g.
+pxa2xx, au1x00, iMX, etc) and portable audio codecs.  Prior to the ASoC
+subsystem there was some support in the kernel for SoC audio, however it
+had some limitations:-
+
+  * Codec drivers were often tightly coupled to the underlying SoC
+    CPU. This is not ideal and leads to code duplication - for example,
+    Linux had different wm8731 drivers for 4 different SoC platforms.
+
+  * There was no standard method to signal user initiated audio events (e.g.
+    Headphone/Mic insertion, Headphone/Mic detection after an insertion
+    event). These are quite common events on portable devices and often require
+    machine specific code to re-route audio, enable amps, etc., after such an
+    event.
+
+  * Drivers tended to power up the entire codec when playing (or
+    recording) audio. This is fine for a PC, but tends to waste a lot of
+    power on portable devices. There was also no support for saving
+    power via changing codec oversampling rates, bias currents, etc.
+
+
+ASoC Design
+===========
+
+The ASoC layer is designed to address these issues and provide the following
+features :-
+
+  * Codec independence. Allows reuse of codec drivers on other platforms
+    and machines.
+
+  * Easy I2S/PCM audio interface setup between codec and SoC. Each SoC
+    interface and codec registers it's audio interface capabilities with the
+    core and are subsequently matched and configured when the application
+    hardware parameters are known.
+
+  * Dynamic Audio Power Management (DAPM). DAPM automatically sets the codec to
+    its minimum power state at all times. This includes powering up/down
+    internal power blocks depending on the internal codec audio routing and any
+    active streams.
+
+  * Pop and click reduction. Pops and clicks can be reduced by powering the
+    codec up/down in the correct sequence (including using digital mute). ASoC
+    signals the codec when to change power states.
+
+  * Machine specific controls: Allow machines to add controls to the sound card
+    (e.g. volume control for speaker amplifier).
+
+To achieve all this, ASoC basically splits an embedded audio system into 3
+components :-
+
+  * Codec driver: The codec driver is platform independent and contains audio
+    controls, audio interface capabilities, codec DAPM definition and codec IO
+    functions.
+
+  * Platform driver: The platform driver contains the audio DMA engine and audio
+    interface drivers (e.g. I2S, AC97, PCM) for that platform.
+
+  * Machine driver: The machine driver handles any machine specific controls and
+    audio events (e.g. turning on an amp at start of playback).
+
+
+Documentation
+=============
+
+The documentation is spilt into the following sections:-
+
+overview.txt: This file.
+
+codec.txt: Codec driver internals.
+
+DAI.txt: Description of Digital Audio Interface standards and how to configure
+a DAI within your codec and CPU DAI drivers.
+
+dapm.txt: Dynamic Audio Power Management
+
+platform.txt: Platform audio DMA and DAI.
+
+machine.txt: Machine driver internals.
+
+pop_clicks.txt: How to minimise audio artifacts.
+
+clocking.txt: ASoC clocking for best power performance.
diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/alsa/soc/platform.txt
new file mode 100644
index 0000000..b681d17
--- /dev/null
+++ b/Documentation/sound/alsa/soc/platform.txt
@@ -0,0 +1,58 @@
+ASoC Platform Driver
+====================
+
+An ASoC platform driver can be divided into audio DMA and SoC DAI configuration
+and control. The platform drivers only target the SoC CPU and must have no board
+specific code.
+
+Audio DMA
+=========
+
+The platform DMA driver optionally supports the following ALSA operations:-
+
+/* SoC audio ops */
+struct snd_soc_ops {
+	int (*startup)(struct snd_pcm_substream *);
+	void (*shutdown)(struct snd_pcm_substream *);
+	int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *);
+	int (*hw_free)(struct snd_pcm_substream *);
+	int (*prepare)(struct snd_pcm_substream *);
+	int (*trigger)(struct snd_pcm_substream *, int);
+};
+
+The platform driver exports its DMA functionality via struct snd_soc_platform:-
+
+struct snd_soc_platform {
+	char *name;
+
+	int (*probe)(struct platform_device *pdev);
+	int (*remove)(struct platform_device *pdev);
+	int (*suspend)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai);
+	int (*resume)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai);
+
+	/* pcm creation and destruction */
+	int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *, struct snd_pcm *);
+	void (*pcm_free)(struct snd_pcm *);
+
+	/* platform stream ops */
+	struct snd_pcm_ops *pcm_ops;
+};
+
+Please refer to the ALSA driver documentation for details of audio DMA.
+http://www.alsa-project.org/~iwai/writing-an-alsa-driver/c436.htm
+
+An example DMA driver is soc/pxa/pxa2xx-pcm.c
+
+
+SoC DAI Drivers
+===============
+
+Each SoC DAI driver must provide the following features:-
+
+ 1) Digital audio interface (DAI) description
+ 2) Digital audio interface configuration
+ 3) PCM's description
+ 4) SYSCLK configuration
+ 5) Suspend and resume (optional)
+
+Please see codec.txt for a description of items 1 - 4.
diff --git a/Documentation/sound/alsa/soc/pops_clicks.txt b/Documentation/sound/alsa/soc/pops_clicks.txt
new file mode 100644
index 0000000..e1e74da
--- /dev/null
+++ b/Documentation/sound/alsa/soc/pops_clicks.txt
@@ -0,0 +1,52 @@
+Audio Pops and Clicks
+=====================
+
+Pops and clicks are unwanted audio artifacts caused by the powering up and down
+of components within the audio subsystem. This is noticeable on PCs when an
+audio module is either loaded or unloaded (at module load time the sound card is
+powered up and causes a popping noise on the speakers).
+
+Pops and clicks can be more frequent on portable systems with DAPM. This is
+because the components within the subsystem are being dynamically powered
+depending on the audio usage and this can subsequently cause a small pop or
+click every time a component power state is changed.
+
+
+Minimising Playback Pops and Clicks
+===================================
+
+Playback pops in portable audio subsystems cannot be completely eliminated
+currently, however future audio codec hardware will have better pop and click
+suppression.  Pops can be reduced within playback by powering the audio
+components in a specific order. This order is different for startup and
+shutdown and follows some basic rules:-
+
+ Startup Order :- DAC --> Mixers --> Output PGA --> Digital Unmute
+
+ Shutdown Order :- Digital Mute --> Output PGA --> Mixers --> DAC
+
+This assumes that the codec PCM output path from the DAC is via a mixer and then
+a PGA (programmable gain amplifier) before being output to the speakers.
+
+
+Minimising Capture Pops and Clicks
+==================================
+
+Capture artifacts are somewhat easier to get rid as we can delay activating the
+ADC until all the pops have occurred. This follows similar power rules to
+playback in that components are powered in a sequence depending upon stream
+startup or shutdown.
+
+ Startup Order - Input PGA --> Mixers --> ADC
+
+ Shutdown Order - ADC --> Mixers --> Input PGA
+
+
+Zipper Noise
+============
+An unwanted zipper noise can occur within the audio playback or capture stream
+when a volume control is changed near its maximum gain value. The zipper noise
+is heard when the gain increase or decrease changes the mean audio signal
+amplitude too quickly. It can be minimised by enabling the zero cross setting
+for each volume control. The ZC forces the gain change to occur when the signal
+crosses the zero amplitude line.
diff --git a/include/sound/Kbuild b/include/sound/Kbuild
new file mode 100644
index 0000000..e9dd936
--- /dev/null
+++ b/include/sound/Kbuild
@@ -0,0 +1,9 @@
+header-y += asound_fm.h
+header-y += hdsp.h
+header-y += hdspm.h
+header-y += sfnt_info.h
+
+unifdef-y += asequencer.h
+unifdef-y += asound.h
+unifdef-y += emu10k1.h
+unifdef-y += sb16_csp.h
diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h
new file mode 100644
index 0000000..4940045
--- /dev/null
+++ b/include/sound/ac97_codec.h
@@ -0,0 +1,654 @@
+#ifndef __SOUND_AC97_CODEC_H
+#define __SOUND_AC97_CODEC_H
+
+/*
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *  Universal interface for Audio Codec '97
+ *
+ *  For more details look to AC '97 component specification revision 2.1
+ *  by Intel Corporation (http://developer.intel.com).
+ *
+ *
+ *   This program is free software; you can redistribute it and/or modify
+ *   it under the terms of the GNU General Public License as published by
+ *   the Free Software Foundation; either version 2 of the License, or
+ *   (at your option) any later version.
+ *
+ *   This program is distributed in the hope that it will be useful,
+ *   but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *   GNU General Public License for more details.
+ *
+ *   You should have received a copy of the GNU General Public License
+ *   along with this program; if not, write to the Free Software
+ *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ */
+
+#include <linux/bitops.h>
+#include <linux/device.h>
+#include <linux/workqueue.h>
+#include "pcm.h"
+#include "control.h"
+#include "info.h"
+
+/* maximum number of devices on the AC97 bus */
+#define	AC97_BUS_MAX_DEVICES	4
+
+/*
+ *  AC'97 codec registers
+ */
+
+#define AC97_RESET		0x00	/* Reset */
+#define AC97_MASTER		0x02	/* Master Volume */
+#define AC97_HEADPHONE		0x04	/* Headphone Volume (optional) */
+#define AC97_MASTER_MONO	0x06	/* Master Volume Mono (optional) */
+#define AC97_MASTER_TONE	0x08	/* Master Tone (Bass & Treble) (optional) */
+#define AC97_PC_BEEP		0x0a	/* PC Beep Volume (optinal) */
+#define AC97_PHONE		0x0c	/* Phone Volume (optional) */
+#define AC97_MIC		0x0e	/* MIC Volume */
+#define AC97_LINE		0x10	/* Line In Volume */
+#define AC97_CD			0x12	/* CD Volume */
+#define AC97_VIDEO		0x14	/* Video Volume (optional) */
+#define AC97_AUX		0x16	/* AUX Volume (optional) */
+#define AC97_PCM		0x18	/* PCM Volume */
+#define AC97_REC_SEL		0x1a	/* Record Select */
+#define AC97_REC_GAIN		0x1c	/* Record Gain */
+#define AC97_REC_GAIN_MIC	0x1e	/* Record Gain MIC (optional) */
+#define AC97_GENERAL_PURPOSE	0x20	/* General Purpose (optional) */
+#define AC97_3D_CONTROL		0x22	/* 3D Control (optional) */
+#define AC97_INT_PAGING		0x24	/* Audio Interrupt & Paging (AC'97 2.3) */
+#define AC97_POWERDOWN		0x26	/* Powerdown control / status */
+/* range 0x28-0x3a - AUDIO AC'97 2.0 extensions */
+#define AC97_EXTENDED_ID	0x28	/* Extended Audio ID */
+#define AC97_EXTENDED_STATUS	0x2a	/* Extended Audio Status and Control */
+#define AC97_PCM_FRONT_DAC_RATE 0x2c	/* PCM Front DAC Rate */
+#define AC97_PCM_SURR_DAC_RATE	0x2e	/* PCM Surround DAC Rate */
+#define AC97_PCM_LFE_DAC_RATE	0x30	/* PCM LFE DAC Rate */
+#define AC97_PCM_LR_ADC_RATE	0x32	/* PCM LR ADC Rate */
+#define AC97_PCM_MIC_ADC_RATE	0x34	/* PCM MIC ADC Rate */
+#define AC97_CENTER_LFE_MASTER	0x36	/* Center + LFE Master Volume */
+#define AC97_SURROUND_MASTER	0x38	/* Surround (Rear) Master Volume */
+#define AC97_SPDIF		0x3a	/* S/PDIF control */
+/* range 0x3c-0x58 - MODEM */
+#define AC97_EXTENDED_MID	0x3c	/* Extended Modem ID */
+#define AC97_EXTENDED_MSTATUS	0x3e	/* Extended Modem Status and Control */
+#define AC97_LINE1_RATE		0x40	/* Line1 DAC/ADC Rate */
+#define AC97_LINE2_RATE		0x42	/* Line2 DAC/ADC Rate */
+#define AC97_HANDSET_RATE	0x44	/* Handset DAC/ADC Rate */
+#define AC97_LINE1_LEVEL	0x46	/* Line1 DAC/ADC Level */
+#define AC97_LINE2_LEVEL	0x48	/* Line2 DAC/ADC Level */
+#define AC97_HANDSET_LEVEL	0x4a	/* Handset DAC/ADC Level */
+#define AC97_GPIO_CFG		0x4c	/* GPIO Configuration */
+#define AC97_GPIO_POLARITY	0x4e	/* GPIO Pin Polarity/Type, 0=low, 1=high active */
+#define AC97_GPIO_STICKY	0x50	/* GPIO Pin Sticky, 0=not, 1=sticky */
+#define AC97_GPIO_WAKEUP	0x52	/* GPIO Pin Wakeup, 0=no int, 1=yes int */
+#define AC97_GPIO_STATUS	0x54	/* GPIO Pin Status, slot 12 */
+#define AC97_MISC_AFE		0x56	/* Miscellaneous Modem AFE Status and Control */
+/* range 0x5a-0x7b - Vendor Specific */
+#define AC97_VENDOR_ID1		0x7c	/* Vendor ID1 */
+#define AC97_VENDOR_ID2		0x7e	/* Vendor ID2 / revision */
+/* range 0x60-0x6f (page 1) - extended codec registers */
+#define AC97_CODEC_CLASS_REV	0x60	/* Codec Class/Revision */
+#define AC97_PCI_SVID		0x62	/* PCI Subsystem Vendor ID */
+#define AC97_PCI_SID		0x64	/* PCI Subsystem ID */
+#define AC97_FUNC_SELECT	0x66	/* Function Select */
+#define AC97_FUNC_INFO		0x68	/* Function Information */
+#define AC97_SENSE_INFO		0x6a	/* Sense Details */
+
+/* slot allocation */
+#define AC97_SLOT_TAG		0
+#define AC97_SLOT_CMD_ADDR	1
+#define AC97_SLOT_CMD_DATA	2
+#define AC97_SLOT_PCM_LEFT	3
+#define AC97_SLOT_PCM_RIGHT	4
+#define AC97_SLOT_MODEM_LINE1	5
+#define AC97_SLOT_PCM_CENTER	6
+#define AC97_SLOT_MIC		6	/* input */
+#define AC97_SLOT_SPDIF_LEFT1	6
+#define AC97_SLOT_PCM_SLEFT	7	/* surround left */
+#define AC97_SLOT_PCM_LEFT_0	7	/* double rate operation */
+#define AC97_SLOT_SPDIF_LEFT	7
+#define AC97_SLOT_PCM_SRIGHT	8	/* surround right */
+#define AC97_SLOT_PCM_RIGHT_0	8	/* double rate operation */
+#define AC97_SLOT_SPDIF_RIGHT	8
+#define AC97_SLOT_LFE		9
+#define AC97_SLOT_SPDIF_RIGHT1	9
+#define AC97_SLOT_MODEM_LINE2	10
+#define AC97_SLOT_PCM_LEFT_1	10	/* double rate operation */
+#define AC97_SLOT_SPDIF_LEFT2	10
+#define AC97_SLOT_HANDSET	11	/* output */
+#define AC97_SLOT_PCM_RIGHT_1	11	/* double rate operation */
+#define AC97_SLOT_SPDIF_RIGHT2	11
+#define AC97_SLOT_MODEM_GPIO	12	/* modem GPIO */
+#define AC97_SLOT_PCM_CENTER_1	12	/* double rate operation */
+
+/* basic capabilities (reset register) */
+#define AC97_BC_DEDICATED_MIC	0x0001	/* Dedicated Mic PCM In Channel */
+#define AC97_BC_RESERVED1	0x0002	/* Reserved (was Modem Line Codec support) */
+#define AC97_BC_BASS_TREBLE	0x0004	/* Bass & Treble Control */
+#define AC97_BC_SIM_STEREO	0x0008	/* Simulated stereo */
+#define AC97_BC_HEADPHONE	0x0010	/* Headphone Out Support */
+#define AC97_BC_LOUDNESS	0x0020	/* Loudness (bass boost) Support */
+#define AC97_BC_16BIT_DAC	0x0000	/* 16-bit DAC resolution */
+#define AC97_BC_18BIT_DAC	0x0040	/* 18-bit DAC resolution */
+#define AC97_BC_20BIT_DAC	0x0080	/* 20-bit DAC resolution */
+#define AC97_BC_DAC_MASK	0x00c0
+#define AC97_BC_16BIT_ADC	0x0000	/* 16-bit ADC resolution */
+#define AC97_BC_18BIT_ADC	0x0100	/* 18-bit ADC resolution */
+#define AC97_BC_20BIT_ADC	0x0200	/* 20-bit ADC resolution */
+#define AC97_BC_ADC_MASK	0x0300
+
+/* general purpose */
+#define AC97_GP_DRSS_MASK	0x0c00	/* double rate slot select */
+#define AC97_GP_DRSS_1011	0x0000	/* LR(C) 10+11(+12) */
+#define AC97_GP_DRSS_78		0x0400	/* LR 7+8 */
+
+/* powerdown bits */
+#define AC97_PD_ADC_STATUS	0x0001	/* ADC status (RO) */
+#define AC97_PD_DAC_STATUS	0x0002	/* DAC status (RO) */
+#define AC97_PD_MIXER_STATUS	0x0004	/* Analog mixer status (RO) */
+#define AC97_PD_VREF_STATUS	0x0008	/* Vref status (RO) */
+#define AC97_PD_PR0		0x0100	/* Power down PCM ADCs and input MUX */
+#define AC97_PD_PR1		0x0200	/* Power down PCM front DAC */
+#define AC97_PD_PR2		0x0400	/* Power down Mixer (Vref still on) */
+#define AC97_PD_PR3		0x0800	/* Power down Mixer (Vref off) */
+#define AC97_PD_PR4		0x1000	/* Power down AC-Link */
+#define AC97_PD_PR5		0x2000	/* Disable internal clock usage */
+#define AC97_PD_PR6		0x4000	/* Headphone amplifier */
+#define AC97_PD_EAPD		0x8000	/* External Amplifer Power Down (EAPD) */
+
+/* extended audio ID bit defines */
+#define AC97_EI_VRA		0x0001	/* Variable bit rate supported */
+#define AC97_EI_DRA		0x0002	/* Double rate supported */
+#define AC97_EI_SPDIF		0x0004	/* S/PDIF out supported */
+#define AC97_EI_VRM		0x0008	/* Variable bit rate supported for MIC */
+#define AC97_EI_DACS_SLOT_MASK	0x0030	/* DACs slot assignment */
+#define AC97_EI_DACS_SLOT_SHIFT	4
+#define AC97_EI_CDAC		0x0040	/* PCM Center DAC available */
+#define AC97_EI_SDAC		0x0080	/* PCM Surround DACs available */
+#define AC97_EI_LDAC		0x0100	/* PCM LFE DAC available */
+#define AC97_EI_AMAP		0x0200	/* indicates optional slot/DAC mapping based on codec ID */
+#define AC97_EI_REV_MASK	0x0c00	/* AC'97 revision mask */
+#define AC97_EI_REV_22		0x0400	/* AC'97 revision 2.2 */
+#define AC97_EI_REV_23		0x0800	/* AC'97 revision 2.3 */
+#define AC97_EI_REV_SHIFT	10
+#define AC97_EI_ADDR_MASK	0xc000	/* physical codec ID (address) */
+#define AC97_EI_ADDR_SHIFT	14
+
+/* extended audio status and control bit defines */
+#define AC97_EA_VRA		0x0001	/* Variable bit rate enable bit */
+#define AC97_EA_DRA		0x0002	/* Double-rate audio enable bit */
+#define AC97_EA_SPDIF		0x0004	/* S/PDIF out enable bit */
+#define AC97_EA_VRM		0x0008	/* Variable bit rate for MIC enable bit */
+#define AC97_EA_SPSA_SLOT_MASK	0x0030	/* Mask for slot assignment bits */
+#define AC97_EA_SPSA_SLOT_SHIFT 4
+#define AC97_EA_SPSA_3_4	0x0000	/* Slot assigned to 3 & 4 */
+#define AC97_EA_SPSA_7_8	0x0010	/* Slot assigned to 7 & 8 */
+#define AC97_EA_SPSA_6_9	0x0020	/* Slot assigned to 6 & 9 */
+#define AC97_EA_SPSA_10_11	0x0030	/* Slot assigned to 10 & 11 */
+#define AC97_EA_CDAC		0x0040	/* PCM Center DAC is ready (Read only) */
+#define AC97_EA_SDAC		0x0080	/* PCM Surround DACs are ready (Read only) */
+#define AC97_EA_LDAC		0x0100	/* PCM LFE DAC is ready (Read only) */
+#define AC97_EA_MDAC		0x0200	/* MIC ADC is ready (Read only) */
+#define AC97_EA_SPCV		0x0400	/* S/PDIF configuration valid (Read only) */
+#define AC97_EA_PRI		0x0800	/* Turns the PCM Center DAC off */
+#define AC97_EA_PRJ		0x1000	/* Turns the PCM Surround DACs off */
+#define AC97_EA_PRK		0x2000	/* Turns the PCM LFE DAC off */
+#define AC97_EA_PRL		0x4000	/* Turns the MIC ADC off */
+
+/* S/PDIF control bit defines */
+#define AC97_SC_PRO		0x0001	/* Professional status */
+#define AC97_SC_NAUDIO		0x0002	/* Non audio stream */
+#define AC97_SC_COPY		0x0004	/* Copyright status */
+#define AC97_SC_PRE		0x0008	/* Preemphasis status */
+#define AC97_SC_CC_MASK		0x07f0	/* Category Code mask */
+#define AC97_SC_CC_SHIFT	4
+#define AC97_SC_L		0x0800	/* Generation Level status */
+#define AC97_SC_SPSR_MASK	0x3000	/* S/PDIF Sample Rate bits */
+#define AC97_SC_SPSR_SHIFT	12
+#define AC97_SC_SPSR_44K	0x0000	/* Use 44.1kHz Sample rate */
+#define AC97_SC_SPSR_48K	0x2000	/* Use 48kHz Sample rate */
+#define AC97_SC_SPSR_32K	0x3000	/* Use 32kHz Sample rate */
+#define AC97_SC_DRS		0x4000	/* Double Rate S/PDIF */
+#define AC97_SC_V		0x8000	/* Validity status */
+
+/* Interrupt and Paging bit defines (AC'97 2.3) */
+#define AC97_PAGE_MASK		0x000f	/* Page Selector */
+#define AC97_PAGE_VENDOR	0	/* Vendor-specific registers */
+#define AC97_PAGE_1		1	/* Extended Codec Registers page 1 */
+#define AC97_INT_ENABLE		0x0800	/* Interrupt Enable */
+#define AC97_INT_SENSE		0x1000	/* Sense Cycle */
+#define AC97_INT_CAUSE_SENSE	0x2000	/* Sense Cycle Completed (RO) */
+#define AC97_INT_CAUSE_GPIO	0x4000	/* GPIO bits changed (RO) */
+#define AC97_INT_STATUS		0x8000	/* Interrupt Status */
+
+/* extended modem ID bit defines */
+#define AC97_MEI_LINE1		0x0001	/* Line1 present */
+#define AC97_MEI_LINE2		0x0002	/* Line2 present */
+#define AC97_MEI_HANDSET	0x0004	/* Handset present */
+#define AC97_MEI_CID1		0x0008	/* caller ID decode for Line1 is supported */
+#define AC97_MEI_CID2		0x0010	/* caller ID decode for Line2 is supported */
+#define AC97_MEI_ADDR_MASK	0xc000	/* physical codec ID (address) */
+#define AC97_MEI_ADDR_SHIFT	14
+
+/* extended modem status and control bit defines */
+#define AC97_MEA_GPIO		0x0001	/* GPIO is ready (ro) */
+#define AC97_MEA_MREF		0x0002	/* Vref is up to nominal level (ro) */
+#define AC97_MEA_ADC1		0x0004	/* ADC1 operational (ro) */
+#define AC97_MEA_DAC1		0x0008	/* DAC1 operational (ro) */
+#define AC97_MEA_ADC2		0x0010	/* ADC2 operational (ro) */
+#define AC97_MEA_DAC2		0x0020	/* DAC2 operational (ro) */
+#define AC97_MEA_HADC		0x0040	/* HADC operational (ro) */
+#define AC97_MEA_HDAC		0x0080	/* HDAC operational (ro) */
+#define AC97_MEA_PRA		0x0100	/* GPIO power down (high) */
+#define AC97_MEA_PRB		0x0200	/* reserved */
+#define AC97_MEA_PRC		0x0400	/* ADC1 power down (high) */
+#define AC97_MEA_PRD		0x0800	/* DAC1 power down (high) */
+#define AC97_MEA_PRE		0x1000	/* ADC2 power down (high) */
+#define AC97_MEA_PRF		0x2000	/* DAC2 power down (high) */
+#define AC97_MEA_PRG		0x4000	/* HADC power down (high) */
+#define AC97_MEA_PRH		0x8000	/* HDAC power down (high) */
+
+/* modem gpio status defines */
+#define AC97_GPIO_LINE1_OH      0x0001  /* Off Hook Line1 */
+#define AC97_GPIO_LINE1_RI      0x0002  /* Ring Detect Line1 */
+#define AC97_GPIO_LINE1_CID     0x0004  /* Caller ID path enable Line1 */
+#define AC97_GPIO_LINE1_LCS     0x0008  /* Loop Current Sense Line1 */
+#define AC97_GPIO_LINE1_PULSE   0x0010  /* Opt./ Pulse Dial Line1 (out) */
+#define AC97_GPIO_LINE1_HL1R    0x0020  /* Opt./ Handset to Line1 relay control (out) */
+#define AC97_GPIO_LINE1_HOHD    0x0040  /* Opt./ Handset off hook detect Line1 (in) */
+#define AC97_GPIO_LINE12_AC     0x0080  /* Opt./ Int.bit 1 / Line1/2 AC (out) */
+#define AC97_GPIO_LINE12_DC     0x0100  /* Opt./ Int.bit 2 / Line1/2 DC (out) */
+#define AC97_GPIO_LINE12_RS     0x0200  /* Opt./ Int.bit 3 / Line1/2 RS (out) */
+#define AC97_GPIO_LINE2_OH      0x0400  /* Off Hook Line2 */
+#define AC97_GPIO_LINE2_RI      0x0800  /* Ring Detect Line2 */
+#define AC97_GPIO_LINE2_CID     0x1000  /* Caller ID path enable Line2 */
+#define AC97_GPIO_LINE2_LCS     0x2000  /* Loop Current Sense Line2 */
+#define AC97_GPIO_LINE2_PULSE   0x4000  /* Opt./ Pulse Dial Line2 (out) */
+#define AC97_GPIO_LINE2_HL1R    0x8000  /* Opt./ Handset to Line2 relay control (out) */
+
+/* specific - SigmaTel */
+#define AC97_SIGMATEL_OUTSEL	0x64	/* Output Select, STAC9758 */
+#define AC97_SIGMATEL_INSEL	0x66	/* Input Select, STAC9758 */
+#define AC97_SIGMATEL_IOMISC	0x68	/* STAC9758 */
+#define AC97_SIGMATEL_ANALOG	0x6c	/* Analog Special */
+#define AC97_SIGMATEL_DAC2INVERT 0x6e
+#define AC97_SIGMATEL_BIAS1	0x70
+#define AC97_SIGMATEL_BIAS2	0x72
+#define AC97_SIGMATEL_VARIOUS	0x72	/* STAC9758 */
+#define AC97_SIGMATEL_MULTICHN	0x74	/* Multi-Channel programming */
+#define AC97_SIGMATEL_CIC1	0x76
+#define AC97_SIGMATEL_CIC2	0x78
+
+/* specific - Analog Devices */
+#define AC97_AD_TEST		0x5a	/* test register */
+#define AC97_AD_TEST2		0x5c	/* undocumented test register 2 */
+#define AC97_AD_HPFD_SHIFT	12	/* High Pass Filter Disable */
+#define AC97_AD_CODEC_CFG	0x70	/* codec configuration */
+#define AC97_AD_JACK_SPDIF	0x72	/* Jack Sense & S/PDIF */
+#define AC97_AD_SERIAL_CFG	0x74	/* Serial Configuration */
+#define AC97_AD_MISC		0x76	/* Misc Control Bits */
+#define AC97_AD_VREFD_SHIFT	2	/* V_REFOUT Disable (AD1888) */
+
+/* specific - Cirrus Logic */
+#define AC97_CSR_ACMODE		0x5e	/* AC Mode Register */
+#define AC97_CSR_MISC_CRYSTAL	0x60	/* Misc Crystal Control */
+#define AC97_CSR_SPDIF		0x68	/* S/PDIF Register */
+#define AC97_CSR_SERIAL		0x6a	/* Serial Port Control */
+#define AC97_CSR_SPECF_ADDR	0x6c	/* Special Feature Address */
+#define AC97_CSR_SPECF_DATA	0x6e	/* Special Feature Data */
+#define AC97_CSR_BDI_STATUS	0x7a	/* BDI Status */
+
+/* specific - Conexant */
+#define AC97_CXR_AUDIO_MISC	0x5c
+#define AC97_CXR_SPDIFEN	(1<<3)
+#define AC97_CXR_COPYRGT	(1<<2)
+#define AC97_CXR_SPDIF_MASK	(3<<0)
+#define AC97_CXR_SPDIF_PCM	0x0
+#define AC97_CXR_SPDIF_AC3	0x2
+
+/* specific - ALC */
+#define AC97_ALC650_SPDIF_INPUT_STATUS1	0x60
+/* S/PDIF input status 1 bit defines */
+#define AC97_ALC650_PRO             0x0001  /* Professional status */
+#define AC97_ALC650_NAUDIO          0x0002  /* Non audio stream */
+#define AC97_ALC650_COPY            0x0004  /* Copyright status */
+#define AC97_ALC650_PRE             0x0038  /* Preemphasis status */
+#define AC97_ALC650_PRE_SHIFT       3
+#define AC97_ALC650_MODE            0x00C0  /* Preemphasis status */
+#define AC97_ALC650_MODE_SHIFT      6
+#define AC97_ALC650_CC_MASK         0x7f00  /* Category Code mask */
+#define AC97_ALC650_CC_SHIFT        8
+#define AC97_ALC650_L               0x8000  /* Generation Level status */
+
+#define AC97_ALC650_SPDIF_INPUT_STATUS2	0x62
+/* S/PDIF input status 2 bit defines */
+#define AC97_ALC650_SOUCE_MASK      0x000f  /* Source number */
+#define AC97_ALC650_CHANNEL_MASK    0x00f0  /* Channel number */
+#define AC97_ALC650_CHANNEL_SHIFT   4 
+#define AC97_ALC650_SPSR_MASK       0x0f00  /* S/PDIF Sample Rate bits */
+#define AC97_ALC650_SPSR_SHIFT      8
+#define AC97_ALC650_SPSR_44K        0x0000  /* Use 44.1kHz Sample rate */
+#define AC97_ALC650_SPSR_48K        0x0200  /* Use 48kHz Sample rate */
+#define AC97_ALC650_SPSR_32K        0x0300  /* Use 32kHz Sample rate */
+#define AC97_ALC650_CLOCK_ACCURACY  0x3000  /* Clock accuracy */
+#define AC97_ALC650_CLOCK_SHIFT     12
+#define AC97_ALC650_CLOCK_LOCK      0x4000  /* Clock locked status */
+#define AC97_ALC650_V               0x8000  /* Validity status */
+
+#define AC97_ALC650_SURR_DAC_VOL	0x64
+#define AC97_ALC650_LFE_DAC_VOL		0x66
+#define AC97_ALC650_UNKNOWN1		0x68
+#define AC97_ALC650_MULTICH		0x6a
+#define AC97_ALC650_UNKNOWN2		0x6c
+#define AC97_ALC650_REVISION		0x6e
+#define AC97_ALC650_UNKNOWN3		0x70
+#define AC97_ALC650_UNKNOWN4		0x72
+#define AC97_ALC650_MISC		0x74
+#define AC97_ALC650_GPIO_SETUP		0x76
+#define AC97_ALC650_GPIO_STATUS		0x78
+#define AC97_ALC650_CLOCK		0x7a
+
+/* specific - Yamaha YMF7x3 */
+#define AC97_YMF7X3_DIT_CTRL	0x66	/* DIT Control (YMF743) / 2 (YMF753) */
+#define AC97_YMF7X3_3D_MODE_SEL	0x68	/* 3D Mode Select */
+
+/* specific - C-Media */
+#define AC97_CM9738_VENDOR_CTRL	0x5a
+#define AC97_CM9739_MULTI_CHAN	0x64
+#define AC97_CM9739_SPDIF_IN_STATUS	0x68 /* 32bit */
+#define AC97_CM9739_SPDIF_CTRL	0x6c
+
+/* specific - wolfson */
+#define AC97_WM97XX_FMIXER_VOL  0x72
+#define AC97_WM9704_RMIXER_VOL  0x74
+#define AC97_WM9704_TEST        0x5a
+#define AC97_WM9704_RPCM_VOL    0x70
+#define AC97_WM9711_OUT3VOL     0x16
+
+
+/* ac97->scaps */
+#define AC97_SCAP_AUDIO		(1<<0)	/* audio codec 97 */
+#define AC97_SCAP_MODEM		(1<<1)	/* modem codec 97 */
+#define AC97_SCAP_SURROUND_DAC	(1<<2)	/* surround L&R DACs are present */
+#define AC97_SCAP_CENTER_LFE_DAC (1<<3)	/* center and LFE DACs are present */
+#define AC97_SCAP_SKIP_AUDIO	(1<<4)	/* skip audio part of codec */
+#define AC97_SCAP_SKIP_MODEM	(1<<5)	/* skip modem part of codec */
+#define AC97_SCAP_INDEP_SDIN	(1<<6)	/* independent SDIN */
+#define AC97_SCAP_INV_EAPD	(1<<7)	/* inverted EAPD */
+#define AC97_SCAP_DETECT_BY_VENDOR (1<<8) /* use vendor registers for read tests */
+#define AC97_SCAP_NO_SPDIF	(1<<9)	/* don't build SPDIF controls */
+#define AC97_SCAP_EAPD_LED	(1<<10)	/* EAPD as mute LED */
+#define AC97_SCAP_POWER_SAVE	(1<<11)	/* capable for aggresive power-saving */
+
+/* ac97->flags */
+#define AC97_HAS_PC_BEEP	(1<<0)	/* force PC Speaker usage */
+#define AC97_AD_MULTI		(1<<1)	/* Analog Devices - multi codecs */
+#define AC97_CS_SPDIF		(1<<2)	/* Cirrus Logic uses funky SPDIF */
+#define AC97_CX_SPDIF		(1<<3)	/* Conexant's spdif interface */
+#define AC97_STEREO_MUTES	(1<<4)	/* has stereo mute bits */
+#define AC97_DOUBLE_RATE	(1<<5)	/* supports double rate playback */
+#define AC97_HAS_NO_MASTER_VOL	(1<<6)	/* no Master volume */
+#define AC97_HAS_NO_PCM_VOL	(1<<7)	/* no PCM volume */
+#define AC97_DEFAULT_POWER_OFF	(1<<8)	/* no RESET write */
+#define AC97_MODEM_PATCH	(1<<9)	/* modem patch */
+#define AC97_HAS_NO_REC_GAIN	(1<<10) /* no Record gain */
+#define AC97_HAS_NO_PHONE	(1<<11) /* no PHONE volume */
+#define AC97_HAS_NO_PC_BEEP	(1<<12) /* no PC Beep volume */
+#define AC97_HAS_NO_VIDEO	(1<<13) /* no Video volume */
+#define AC97_HAS_NO_CD		(1<<14) /* no CD volume */
+#define AC97_HAS_NO_MIC	(1<<15) /* no MIC volume */
+#define AC97_HAS_NO_TONE	(1<<16) /* no Tone volume */
+#define AC97_HAS_NO_STD_PCM	(1<<17)	/* no standard AC97 PCM volume and mute */
+#define AC97_HAS_NO_AUX		(1<<18) /* no standard AC97 AUX volume and mute */
+#define AC97_HAS_8CH		(1<<19) /* supports 8-channel output */
+
+/* rates indexes */
+#define AC97_RATES_FRONT_DAC	0
+#define AC97_RATES_SURR_DAC	1
+#define AC97_RATES_LFE_DAC	2
+#define AC97_RATES_ADC		3
+#define AC97_RATES_MIC_ADC	4
+#define AC97_RATES_SPDIF	5
+
+/*
+ *
+ */
+
+struct snd_ac97;
+
+struct snd_ac97_build_ops {
+	int (*build_3d) (struct snd_ac97 *ac97);
+	int (*build_specific) (struct snd_ac97 *ac97);
+	int (*build_spdif) (struct snd_ac97 *ac97);
+	int (*build_post_spdif) (struct snd_ac97 *ac97);
+#ifdef CONFIG_PM
+	void (*suspend) (struct snd_ac97 *ac97);
+	void (*resume) (struct snd_ac97 *ac97);
+#endif
+	void (*update_jacks) (struct snd_ac97 *ac97);	/* for jack-sharing */
+};
+
+struct snd_ac97_bus_ops {
+	void (*reset) (struct snd_ac97 *ac97);
+	void (*warm_reset)(struct snd_ac97 *ac97);
+	void (*write) (struct snd_ac97 *ac97, unsigned short reg, unsigned short val);
+	unsigned short (*read) (struct snd_ac97 *ac97, unsigned short reg);
+	void (*wait) (struct snd_ac97 *ac97);
+	void (*init) (struct snd_ac97 *ac97);
+};
+
+struct snd_ac97_bus {
+	/* -- lowlevel (hardware) driver specific -- */
+	struct snd_ac97_bus_ops *ops;
+	void *private_data;
+	void (*private_free) (struct snd_ac97_bus *bus);
+	/* --- */
+	struct snd_card *card;
+	unsigned short num;	/* bus number */
+	unsigned short no_vra: 1, /* bridge doesn't support VRA */
+		       dra: 1,	/* bridge supports double rate */
+		       isdin: 1;/* independent SDIN */
+	unsigned int clock;	/* AC'97 base clock (usually 48000Hz) */
+	spinlock_t bus_lock;	/* used mainly for slot allocation */
+	unsigned short used_slots[2][4]; /* actually used PCM slots */
+	unsigned short pcms_count; /* count of PCMs */
+	struct ac97_pcm *pcms;
+	struct snd_ac97 *codec[4];
+	struct snd_info_entry *proc;
+};
+
+/* static resolution table */
+struct snd_ac97_res_table {
+	unsigned short reg;	/* register */
+	unsigned short bits;	/* resolution bitmask */
+};
+
+struct snd_ac97_template {
+	void *private_data;
+	void (*private_free) (struct snd_ac97 *ac97);
+	struct pci_dev *pci;	/* assigned PCI device - used for quirks */
+	unsigned short num;	/* number of codec: 0 = primary, 1 = secondary */
+	unsigned short addr;	/* physical address of codec [0-3] */
+	unsigned int scaps;	/* driver capabilities */
+	const struct snd_ac97_res_table *res_table;	/* static resolution */
+};
+
+struct snd_ac97 {
+	/* -- lowlevel (hardware) driver specific -- */
+	struct snd_ac97_build_ops * build_ops;
+	void *private_data;
+	void (*private_free) (struct snd_ac97 *ac97);
+	/* --- */
+	struct snd_ac97_bus *bus;
+	struct pci_dev *pci;	/* assigned PCI device - used for quirks */
+	struct snd_info_entry *proc;
+	struct snd_info_entry *proc_regs;
+	unsigned short subsystem_vendor;
+	unsigned short subsystem_device;
+	struct mutex reg_mutex;
+	struct mutex page_mutex;	/* mutex for AD18xx multi-codecs and paging (2.3) */
+	unsigned short num;	/* number of codec: 0 = primary, 1 = secondary */
+	unsigned short addr;	/* physical address of codec [0-3] */
+	unsigned int id;	/* identification of codec */
+	unsigned short caps;	/* capabilities (register 0) */
+	unsigned short ext_id;	/* extended feature identification (register 28) */
+	unsigned short ext_mid;	/* extended modem ID (register 3C) */
+	const struct snd_ac97_res_table *res_table;	/* static resolution */
+	unsigned int scaps;	/* driver capabilities */
+	unsigned int flags;	/* specific code */
+	unsigned int rates[6];	/* see AC97_RATES_* defines */
+	unsigned int spdif_status;
+	unsigned short regs[0x80]; /* register cache */
+	DECLARE_BITMAP(reg_accessed, 0x80); /* bit flags */
+	union {			/* vendor specific code */
+		struct {
+			unsigned short unchained[3];	// 0 = C34, 1 = C79, 2 = C69
+			unsigned short chained[3];	// 0 = C34, 1 = C79, 2 = C69
+			unsigned short id[3];		// codec IDs (lower 16-bit word)
+			unsigned short pcmreg[3];	// PCM registers
+			unsigned short codec_cfg[3];	// CODEC_CFG bits
+			unsigned char swap_mic_linein;	// AD1986/AD1986A only
+			unsigned char lo_as_master;	/* LO as master */
+		} ad18xx;
+		unsigned int dev_flags;		/* device specific */
+	} spec;
+	/* jack-sharing info */
+	unsigned char indep_surround;
+	unsigned char channel_mode;
+
+#ifdef CONFIG_SND_AC97_POWER_SAVE
+	unsigned int power_up;	/* power states */
+	struct delayed_work power_work;
+#endif
+	struct device dev;
+};
+
+#define to_ac97_t(d) container_of(d, struct snd_ac97, dev)
+
+/* conditions */
+static inline int ac97_is_audio(struct snd_ac97 * ac97)
+{
+	return (ac97->scaps & AC97_SCAP_AUDIO);
+}
+static inline int ac97_is_modem(struct snd_ac97 * ac97)
+{
+	return (ac97->scaps & AC97_SCAP_MODEM);
+}
+static inline int ac97_is_rev22(struct snd_ac97 * ac97)
+{
+	return (ac97->ext_id & AC97_EI_REV_MASK) >= AC97_EI_REV_22;
+}
+static inline int ac97_can_amap(struct snd_ac97 * ac97)
+{
+	return (ac97->ext_id & AC97_EI_AMAP) != 0;
+}
+static inline int ac97_can_spdif(struct snd_ac97 * ac97)
+{
+	return (ac97->ext_id & AC97_EI_SPDIF) != 0;
+}
+
+/* functions */
+/* create new AC97 bus */
+int snd_ac97_bus(struct snd_card *card, int num, struct snd_ac97_bus_ops *ops,
+		 void *private_data, struct snd_ac97_bus **rbus);
+/* create mixer controls */
+int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template,
+		   struct snd_ac97 **rac97);
+const char *snd_ac97_get_short_name(struct snd_ac97 *ac97);
+
+void snd_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short value);
+unsigned short snd_ac97_read(struct snd_ac97 *ac97, unsigned short reg);
+void snd_ac97_write_cache(struct snd_ac97 *ac97, unsigned short reg, unsigned short value);
+int snd_ac97_update(struct snd_ac97 *ac97, unsigned short reg, unsigned short value);
+int snd_ac97_update_bits(struct snd_ac97 *ac97, unsigned short reg, unsigned short mask, unsigned short value);
+#ifdef CONFIG_SND_AC97_POWER_SAVE
+int snd_ac97_update_power(struct snd_ac97 *ac97, int reg, int powerup);
+#else
+static inline int snd_ac97_update_power(struct snd_ac97 *ac97, int reg,
+					int powerup)
+{
+	return 0;
+}
+#endif
+#ifdef CONFIG_PM
+void snd_ac97_suspend(struct snd_ac97 *ac97);
+void snd_ac97_resume(struct snd_ac97 *ac97);
+#endif
+
+/* quirk types */
+enum {
+	AC97_TUNE_DEFAULT = -1,	/* use default from quirk list (not valid in list) */
+	AC97_TUNE_NONE = 0,	/* nothing extra to do */
+	AC97_TUNE_HP_ONLY,	/* headphone (true line-out) control as master only */
+	AC97_TUNE_SWAP_HP,	/* swap headphone and master controls */
+	AC97_TUNE_SWAP_SURROUND, /* swap master and surround controls */
+	AC97_TUNE_AD_SHARING,	/* for AD1985, turn on OMS bit and use headphone */
+	AC97_TUNE_ALC_JACK,	/* for Realtek, enable JACK detection */
+	AC97_TUNE_INV_EAPD,	/* inverted EAPD implementation */
+	AC97_TUNE_MUTE_LED,	/* EAPD bit works as mute LED */
+	AC97_TUNE_HP_MUTE_LED,  /* EAPD bit works as mute LED, use headphone control as master */
+};
+
+struct ac97_quirk {
+	unsigned short subvendor; /* PCI subsystem vendor id */
+	unsigned short subdevice; /* PCI subsystem device id */
+	unsigned short mask;	/* device id bit mask, 0 = accept all */
+	unsigned int codec_id;	/* codec id (if any), 0 = accept all */
+	const char *name;	/* name shown as info */
+	int type;		/* quirk type above */
+};
+
+int snd_ac97_tune_hardware(struct snd_ac97 *ac97, struct ac97_quirk *quirk, const char *override);
+int snd_ac97_set_rate(struct snd_ac97 *ac97, int reg, unsigned int rate);
+
+/*
+ * PCM allocation
+ */
+
+enum ac97_pcm_cfg {
+	AC97_PCM_CFG_FRONT = 2,
+	AC97_PCM_CFG_REAR = 10,		/* alias surround */
+	AC97_PCM_CFG_LFE = 11,		/* center + lfe */
+	AC97_PCM_CFG_40 = 4,		/* front + rear */
+	AC97_PCM_CFG_51 = 6,		/* front + rear + center/lfe */
+	AC97_PCM_CFG_SPDIF = 20
+};
+
+struct ac97_pcm {
+	struct snd_ac97_bus *bus;
+	unsigned int stream: 1,	   	   /* stream type: 1 = capture */
+		     exclusive: 1,	   /* exclusive mode, don't override with other pcms */
+		     copy_flag: 1,	   /* lowlevel driver must fill all entries */
+		     spdif: 1;		   /* spdif pcm */
+	unsigned short aslots;		   /* active slots */
+	unsigned short cur_dbl;		   /* current double-rate state */
+	unsigned int rates;		   /* available rates */
+	struct {
+		unsigned short slots;	   /* driver input: requested AC97 slot numbers */
+		unsigned short rslots[4];  /* allocated slots per codecs */
+		unsigned char rate_table[4];
+		struct snd_ac97 *codec[4];	   /* allocated codecs */
+	} r[2];				   /* 0 = standard rates, 1 = double rates */
+	unsigned long private_value;	   /* used by the hardware driver */
+};
+
+int snd_ac97_pcm_assign(struct snd_ac97_bus *ac97,
+			unsigned short pcms_count,
+			const struct ac97_pcm *pcms);
+int snd_ac97_pcm_open(struct ac97_pcm *pcm, unsigned int rate,
+		      enum ac97_pcm_cfg cfg, unsigned short slots);
+int snd_ac97_pcm_close(struct ac97_pcm *pcm);
+int snd_ac97_pcm_double_rate_rules(struct snd_pcm_runtime *runtime);
+
+/* ad hoc AC97 device driver access */
+extern struct bus_type ac97_bus_type;
+
+/* AC97 platform_data adding function */
+static inline void snd_ac97_dev_add_pdata(struct snd_ac97 *ac97, void *data)
+{
+	ac97->dev.platform_data = data;
+}
+
+#endif /* __SOUND_AC97_CODEC_H */
diff --git a/include/sound/aci.h b/include/sound/aci.h
new file mode 100644
index 0000000..ee639d3
--- /dev/null
+++ b/include/sound/aci.h
@@ -0,0 +1,90 @@
+#ifndef _ACI_H_
+#define _ACI_H_
+
+#define ACI_REG_COMMAND		0	/* write register offset */
+#define ACI_REG_STATUS		1	/* read register offset */
+#define ACI_REG_BUSY		2	/* busy register offset */
+#define ACI_REG_RDS		2	/* PCM20: RDS register offset */
+#define ACI_MINTIME		500	/* ACI time out limit */
+
+#define ACI_SET_MUTE		0x0d
+#define ACI_SET_POWERAMP	0x0f
+#define ACI_SET_TUNERMUTE	0xa3
+#define ACI_SET_TUNERMONO	0xa4
+#define ACI_SET_IDE		0xd0
+#define ACI_SET_WSS		0xd1
+#define ACI_SET_SOLOMODE	0xd2
+#define ACI_SET_PREAMP		0x03
+#define ACI_GET_PREAMP		0x21
+#define ACI_WRITE_TUNE		0xa7
+#define ACI_READ_TUNERSTEREO	0xa8
+#define ACI_READ_TUNERSTATION	0xa9
+#define ACI_READ_VERSION	0xf1
+#define ACI_READ_IDCODE		0xf2
+#define ACI_INIT		0xff
+#define ACI_STATUS		0xf0
+#define ACI_S_GENERAL		0x00
+#define ACI_ERROR_OP		0xdf
+
+/* ACI Mixer */
+
+/* These are the values for the right channel GET registers.
+   Add an offset of 0x01 for the left channel register.
+   (left=right+0x01) */
+
+#define ACI_GET_MASTER		0x03
+#define ACI_GET_MIC		0x05
+#define ACI_GET_LINE		0x07
+#define ACI_GET_CD		0x09
+#define ACI_GET_SYNTH		0x0b
+#define ACI_GET_PCM		0x0d
+#define ACI_GET_LINE1		0x10	/* Radio on PCM20 */
+#define ACI_GET_LINE2		0x12
+
+#define ACI_GET_EQ1		0x22	/* from Bass ... */
+#define ACI_GET_EQ2		0x24
+#define ACI_GET_EQ3		0x26
+#define ACI_GET_EQ4		0x28
+#define ACI_GET_EQ5		0x2a
+#define ACI_GET_EQ6		0x2c
+#define ACI_GET_EQ7		0x2e	/* ... to Treble */
+
+/* And these are the values for the right channel SET registers.
+   For left channel access you have to add an offset of 0x08.
+   MASTER is an exception, which needs an offset of 0x01 */
+
+#define ACI_SET_MASTER		0x00
+#define ACI_SET_MIC		0x30
+#define ACI_SET_LINE		0x31
+#define ACI_SET_CD		0x34
+#define ACI_SET_SYNTH		0x33
+#define ACI_SET_PCM		0x32
+#define ACI_SET_LINE1		0x35	/* Radio on PCM20 */
+#define ACI_SET_LINE2		0x36
+
+#define ACI_SET_EQ1		0x40	/* from Bass ... */
+#define ACI_SET_EQ2		0x41
+#define ACI_SET_EQ3		0x42
+#define ACI_SET_EQ4		0x43
+#define ACI_SET_EQ5		0x44
+#define ACI_SET_EQ6		0x45
+#define ACI_SET_EQ7		0x46	/* ... to Treble */
+
+struct snd_miro_aci {
+	unsigned long aci_port;
+	int aci_vendor;
+	int aci_product;
+	int aci_version;
+	int aci_amp;
+	int aci_preamp;
+	int aci_solomode;
+
+	struct mutex aci_mutex;
+};
+
+int snd_aci_cmd(struct snd_miro_aci *aci, int write1, int write2, int write3);
+
+struct snd_miro_aci *snd_aci_get_aci(void);
+
+#endif  /* _ACI_H_ */
+
diff --git a/include/sound/ad1816a.h b/include/sound/ad1816a.h
new file mode 100644
index 0000000..d010858
--- /dev/null
+++ b/include/sound/ad1816a.h
@@ -0,0 +1,175 @@
+#ifndef __SOUND_AD1816A_H
+#define __SOUND_AD1816A_H
+
+/*
+    ad1816a.h - definitions for ADI SoundPort AD1816A chip.
+    Copyright (C) 1999-2000 by Massimo Piccioni <dafastidio@libero.it>
+
+    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., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+*/
+
+#include "control.h"
+#include "pcm.h"
+#include "timer.h"
+
+#define AD1816A_REG(r)			(chip->port + r)
+
+#define AD1816A_CHIP_STATUS		0x00
+#define AD1816A_INDIR_ADDR		0x00
+#define AD1816A_INTERRUPT_STATUS	0x01
+#define AD1816A_INDIR_DATA_LOW		0x02
+#define AD1816A_INDIR_DATA_HIGH		0x03
+#define AD1816A_PIO_DEBUG		0x04
+#define AD1816A_PIO_STATUS		0x05
+#define AD1816A_PIO_DATA		0x06
+#define AD1816A_RESERVED_7		0x07
+#define AD1816A_PLAYBACK_CONFIG		0x08
+#define AD1816A_CAPTURE_CONFIG		0x09
+#define AD1816A_RESERVED_10		0x0a
+#define AD1816A_RESERVED_11		0x0b
+#define AD1816A_JOYSTICK_RAW_DATA	0x0c
+#define AD1816A_JOYSTICK_CTRL		0x0d
+#define AD1816A_JOY_POS_DATA_LOW	0x0e
+#define AD1816A_JOY_POS_DATA_HIGH	0x0f
+
+#define AD1816A_LOW_BYTE_TMP		0x00
+#define AD1816A_INTERRUPT_ENABLE	0x01
+#define AD1816A_EXTERNAL_CTRL		0x01
+#define AD1816A_PLAYBACK_SAMPLE_RATE	0x02
+#define AD1816A_CAPTURE_SAMPLE_RATE	0x03
+#define AD1816A_VOICE_ATT		0x04
+#define AD1816A_FM_ATT			0x05
+#define AD1816A_I2S_1_ATT		0x06
+#define AD1816A_I2S_0_ATT		0x07
+#define AD1816A_PLAYBACK_BASE_COUNT	0x08
+#define AD1816A_PLAYBACK_CURR_COUNT	0x09
+#define AD1816A_CAPTURE_BASE_COUNT	0x0a
+#define AD1816A_CAPTURE_CURR_COUNT	0x0b
+#define AD1816A_TIMER_BASE_COUNT	0x0c
+#define AD1816A_TIMER_CURR_COUNT	0x0d
+#define AD1816A_MASTER_ATT		0x0e
+#define AD1816A_CD_GAIN_ATT		0x0f
+#define AD1816A_SYNTH_GAIN_ATT		0x10
+#define AD1816A_VID_GAIN_ATT		0x11
+#define AD1816A_LINE_GAIN_ATT		0x12
+#define AD1816A_MIC_GAIN_ATT		0x13
+#define AD1816A_PHONE_IN_GAIN_ATT	0x13
+#define AD1816A_ADC_SOURCE_SEL		0x14
+#define AD1816A_ADC_PGA			0x14
+#define AD1816A_CHIP_CONFIG		0x20
+#define AD1816A_DSP_CONFIG		0x21
+#define AD1816A_FM_SAMPLE_RATE		0x22
+#define AD1816A_I2S_1_SAMPLE_RATE	0x23
+#define AD1816A_I2S_0_SAMPLE_RATE	0x24
+#define AD1816A_RESERVED_37		0x25
+#define AD1816A_PROGRAM_CLOCK_RATE	0x26
+#define AD1816A_3D_PHAT_CTRL		0x27
+#define AD1816A_PHONE_OUT_ATT		0x27
+#define AD1816A_RESERVED_40		0x28
+#define AD1816A_HW_VOL_BUT		0x29
+#define AD1816A_DSP_MAILBOX_0		0x2a
+#define AD1816A_DSP_MAILBOX_1		0x2b
+#define AD1816A_POWERDOWN_CTRL		0x2c
+#define AD1816A_TIMER_CTRL		0x2c
+#define AD1816A_VERSION_ID		0x2d
+#define AD1816A_RESERVED_46		0x2e
+
+#define AD1816A_READY			0x80
+
+#define AD1816A_PLAYBACK_IRQ_PENDING	0x80
+#define AD1816A_CAPTURE_IRQ_PENDING	0x40
+#define AD1816A_TIMER_IRQ_PENDING	0x20
+
+#define AD1816A_PLAYBACK_ENABLE		0x01
+#define AD1816A_PLAYBACK_PIO		0x02
+#define AD1816A_CAPTURE_ENABLE		0x01
+#define AD1816A_CAPTURE_PIO		0x02
+
+#define AD1816A_FMT_LINEAR_8		0x00
+#define AD1816A_FMT_ULAW_8		0x08
+#define AD1816A_FMT_LINEAR_16_LIT	0x10
+#define AD1816A_FMT_ALAW_8		0x18
+#define AD1816A_FMT_LINEAR_16_BIG	0x30
+#define AD1816A_FMT_ALL			0x38
+#define AD1816A_FMT_STEREO		0x04
+
+#define AD1816A_PLAYBACK_IRQ_ENABLE	0x8000
+#define AD1816A_CAPTURE_IRQ_ENABLE	0x4000
+#define AD1816A_TIMER_IRQ_ENABLE	0x2000
+#define AD1816A_TIMER_ENABLE		0x0080
+
+#define AD1816A_SRC_LINE		0x00
+#define AD1816A_SRC_OUT			0x10
+#define AD1816A_SRC_CD			0x20
+#define AD1816A_SRC_SYNTH		0x30
+#define AD1816A_SRC_VIDEO		0x40
+#define AD1816A_SRC_MIC			0x50
+#define AD1816A_SRC_MONO		0x50
+#define AD1816A_SRC_PHONE_IN		0x60
+#define AD1816A_SRC_MASK		0x70
+
+#define AD1816A_CAPTURE_NOT_EQUAL	0x1000
+#define AD1816A_WSS_ENABLE		0x8000
+
+struct snd_ad1816a {
+	unsigned long port;
+	struct resource *res_port;
+	int irq;
+	int dma1;
+	int dma2;
+
+	unsigned short hardware;
+	unsigned short version;
+
+	spinlock_t lock;
+
+	unsigned short mode;
+	unsigned int clock_freq;
+
+	struct snd_card *card;
+	struct snd_pcm *pcm;
+
+	struct snd_pcm_substream *playback_substream;
+	struct snd_pcm_substream *capture_substream;
+	unsigned int p_dma_size;
+	unsigned int c_dma_size;
+
+	struct snd_timer *timer;
+};
+
+
+#define AD1816A_HW_AUTO		0
+#define AD1816A_HW_AD1816A	1
+#define AD1816A_HW_AD1815	2
+#define AD1816A_HW_AD18MAX10	3
+
+#define AD1816A_MODE_PLAYBACK	0x01
+#define AD1816A_MODE_CAPTURE	0x02
+#define AD1816A_MODE_TIMER	0x04
+#define AD1816A_MODE_OPEN	(AD1816A_MODE_PLAYBACK |	\
+				AD1816A_MODE_CAPTURE |		\
+				AD1816A_MODE_TIMER)
+
+
+extern int snd_ad1816a_create(struct snd_card *card, unsigned long port,
+			      int irq, int dma1, int dma2,
+			      struct snd_ad1816a **chip);
+
+extern int snd_ad1816a_pcm(struct snd_ad1816a *chip, int device, struct snd_pcm **rpcm);
+extern int snd_ad1816a_mixer(struct snd_ad1816a *chip);
+extern int snd_ad1816a_timer(struct snd_ad1816a *chip, int device,
+			     struct snd_timer **rtimer);
+
+#endif	/* __SOUND_AD1816A_H */
diff --git a/include/sound/ad1843.h b/include/sound/ad1843.h
new file mode 100644
index 0000000..b236a9d
--- /dev/null
+++ b/include/sound/ad1843.h
@@ -0,0 +1,46 @@
+/*
+ * This file is subject to the terms and conditions of the GNU General Public
+ * License.  See the file "COPYING" in the main directory of this archive
+ * for more details.
+ *
+ * Copyright 2003 Vivien Chappelier <vivien.chappelier@linux-mips.org>
+ * Copyright 2008 Thomas Bogendoerfer <tsbogend@franken.de>
+ */
+
+#ifndef __SOUND_AD1843_H
+#define __SOUND_AD1843_H
+
+struct snd_ad1843 {
+	void *chip;
+	int (*read)(void *chip, int reg);
+	int (*write)(void *chip, int reg, int val);
+};
+
+#define AD1843_GAIN_RECLEV 0
+#define AD1843_GAIN_LINE   1
+#define AD1843_GAIN_LINE_2 2
+#define AD1843_GAIN_MIC    3
+#define AD1843_GAIN_PCM_0  4
+#define AD1843_GAIN_PCM_1  5
+#define AD1843_GAIN_SIZE   (AD1843_GAIN_PCM_1+1)
+
+int ad1843_get_gain_max(struct snd_ad1843 *ad1843, int id);
+int ad1843_get_gain(struct snd_ad1843 *ad1843, int id);
+int ad1843_set_gain(struct snd_ad1843 *ad1843, int id, int newval);
+int ad1843_get_recsrc(struct snd_ad1843 *ad1843);
+int ad1843_set_recsrc(struct snd_ad1843 *ad1843, int newsrc);
+void ad1843_setup_dac(struct snd_ad1843 *ad1843,
+		      unsigned int id,
+		      unsigned int framerate,
+		      snd_pcm_format_t fmt,
+		      unsigned int channels);
+void ad1843_shutdown_dac(struct snd_ad1843 *ad1843,
+			 unsigned int id);
+void ad1843_setup_adc(struct snd_ad1843 *ad1843,
+		      unsigned int framerate,
+		      snd_pcm_format_t fmt,
+		      unsigned int channels);
+void ad1843_shutdown_adc(struct snd_ad1843 *ad1843);
+int ad1843_init(struct snd_ad1843 *ad1843);
+
+#endif /* __SOUND_AD1843_H */
diff --git a/include/sound/ak4113.h b/include/sound/ak4113.h
new file mode 100644
index 0000000..2609048
--- /dev/null
+++ b/include/sound/ak4113.h
@@ -0,0 +1,321 @@
+#ifndef __SOUND_AK4113_H
+#define __SOUND_AK4113_H
+
+/*
+ *  Routines for Asahi Kasei AK4113
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
+ *  Copyright (c) by Pavel Hofman <pavel.hofman@ivitera.com>,
+ *
+ *
+ *   This program is free software; you can redistribute it and/or modify
+ *   it under the terms of the GNU General Public License as published by
+ *   the Free Software Foundation; either version 2 of the License, or
+ *   (at your option) any later version.
+ *
+ *   This program is distributed in the hope that it will be useful,
+ *   but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *   GNU General Public License for more details.
+ *
+ *   You should have received a copy of the GNU General Public License
+ *   along with this program; if not, write to the Free Software
+ *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ */
+
+/* AK4113 registers */
+/* power down */
+#define AK4113_REG_PWRDN	0x00
+/* format control */
+#define AK4113_REG_FORMAT	0x01
+/* input/output control */
+#define AK4113_REG_IO0		0x02
+/* input/output control */
+#define AK4113_REG_IO1		0x03
+/* interrupt0 mask */
+#define AK4113_REG_INT0_MASK	0x04
+/* interrupt1 mask */
+#define AK4113_REG_INT1_MASK	0x05
+/* DAT mask & DTS select */
+#define AK4113_REG_DATDTS	0x06
+/* receiver status 0 */
+#define AK4113_REG_RCS0		0x07
+/* receiver status 1 */
+#define AK4113_REG_RCS1		0x08
+/* receiver status 2 */
+#define AK4113_REG_RCS2		0x09
+/* RX channel status byte 0 */
+#define AK4113_REG_RXCSB0	0x0a
+/* RX channel status byte 1 */
+#define AK4113_REG_RXCSB1	0x0b
+/* RX channel status byte 2 */
+#define AK4113_REG_RXCSB2	0x0c
+/* RX channel status byte 3 */
+#define AK4113_REG_RXCSB3	0x0d
+/* RX channel status byte 4 */
+#define AK4113_REG_RXCSB4	0x0e
+/* burst preamble Pc byte 0 */
+#define AK4113_REG_Pc0		0x0f
+/* burst preamble Pc byte 1 */
+#define AK4113_REG_Pc1		0x10
+/* burst preamble Pd byte 0 */
+#define AK4113_REG_Pd0		0x11
+/* burst preamble Pd byte 1 */
+#define AK4113_REG_Pd1		0x12
+/* Q-subcode address + control */
+#define AK4113_REG_QSUB_ADDR	0x13
+/* Q-subcode track */
+#define AK4113_REG_QSUB_TRACK	0x14
+/* Q-subcode index */
+#define AK4113_REG_QSUB_INDEX	0x15
+/* Q-subcode minute */
+#define AK4113_REG_QSUB_MINUTE	0x16
+/* Q-subcode second */
+#define AK4113_REG_QSUB_SECOND	0x17
+/* Q-subcode frame */
+#define AK4113_REG_QSUB_FRAME	0x18
+/* Q-subcode zero */
+#define AK4113_REG_QSUB_ZERO	0x19
+/* Q-subcode absolute minute */
+#define AK4113_REG_QSUB_ABSMIN	0x1a
+/* Q-subcode absolute second */
+#define AK4113_REG_QSUB_ABSSEC	0x1b
+/* Q-subcode absolute frame */
+#define AK4113_REG_QSUB_ABSFRM	0x1c
+
+/* sizes */
+#define AK4113_REG_RXCSB_SIZE	((AK4113_REG_RXCSB4-AK4113_REG_RXCSB0)+1)
+#define AK4113_REG_QSUB_SIZE	((AK4113_REG_QSUB_ABSFRM-AK4113_REG_QSUB_ADDR)\
+		+1)
+
+#define AK4113_WRITABLE_REGS	(AK4113_REG_DATDTS + 1)
+
+/* AK4113_REG_PWRDN bits */
+/* Channel Status Select */
+#define AK4113_CS12		(1<<7)
+/* Block Start & C/U Output Mode */
+#define AK4113_BCU		(1<<6)
+/* Master Clock Operation Select */
+#define AK4113_CM1		(1<<5)
+/* Master Clock Operation Select */
+#define AK4113_CM0		(1<<4)
+/* Master Clock Frequency Select */
+#define AK4113_OCKS1		(1<<3)
+/* Master Clock Frequency Select */
+#define AK4113_OCKS0		(1<<2)
+/* 0 = power down, 1 = normal operation */
+#define AK4113_PWN		(1<<1)
+/* 0 = reset & initialize (except thisregister), 1 = normal operation */
+#define AK4113_RST		(1<<0)
+
+/* AK4113_REQ_FORMAT bits */
+/* V/TX Output select: 0 = Validity Flag Output, 1 = TX */
+#define AK4113_VTX		(1<<7)
+/* Audio Data Control */
+#define AK4113_DIF2		(1<<6)
+/* Audio Data Control */
+#define AK4113_DIF1		(1<<5)
+/* Audio Data Control */
+#define AK4113_DIF0		(1<<4)
+/* Deemphasis Autodetect Enable (1 = enable) */
+#define AK4113_DEAU		(1<<3)
+/* 32kHz-48kHz Deemphasis Control */
+#define AK4113_DEM1		(1<<2)
+/* 32kHz-48kHz Deemphasis Control */
+#define AK4113_DEM0		(1<<1)
+#define AK4113_DEM_OFF		(AK4113_DEM0)
+#define AK4113_DEM_44KHZ	(0)
+#define AK4113_DEM_48KHZ	(AK4113_DEM1)
+#define AK4113_DEM_32KHZ	(AK4113_DEM0|AK4113_DEM1)
+/* STDO: 16-bit, right justified */
+#define AK4113_DIF_16R		(0)
+/* STDO: 18-bit, right justified */
+#define AK4113_DIF_18R		(AK4113_DIF0)
+/* STDO: 20-bit, right justified */
+#define AK4113_DIF_20R		(AK4113_DIF1)
+/* STDO: 24-bit, right justified */
+#define AK4113_DIF_24R		(AK4113_DIF1|AK4113_DIF0)
+/* STDO: 24-bit, left justified */
+#define AK4113_DIF_24L		(AK4113_DIF2)
+/* STDO: I2S */
+#define AK4113_DIF_24I2S	(AK4113_DIF2|AK4113_DIF0)
+/* STDO: 24-bit, left justified; LRCLK, BICK = Input */
+#define AK4113_DIF_I24L		(AK4113_DIF2|AK4113_DIF1)
+/* STDO: I2S;  LRCLK, BICK = Input */
+#define AK4113_DIF_I24I2S	(AK4113_DIF2|AK4113_DIF1|AK4113_DIF0)
+
+/* AK4113_REG_IO0 */
+/* XTL1=0,XTL0=0 -> 11.2896Mhz; XTL1=0,XTL0=1 -> 12.288Mhz */
+#define AK4113_XTL1		(1<<6)
+/* XTL1=1,XTL0=0 -> 24.576Mhz; XTL1=1,XTL0=1 -> use channel status */
+#define AK4113_XTL0		(1<<5)
+/* Block Start Signal Output: 0 = U-bit, 1 = C-bit (req. BCU = 1) */
+#define AK4113_UCE		(1<<4)
+/* TX Output Enable (1 = enable) */
+#define AK4113_TXE		(1<<3)
+/* Output Through Data Selector for TX pin */
+#define AK4113_OPS2		(1<<2)
+/* Output Through Data Selector for TX pin */
+#define AK4113_OPS1		(1<<1)
+/* Output Through Data Selector for TX pin */
+#define AK4113_OPS0		(1<<0)
+/* 11.2896 MHz ref. Xtal freq. */
+#define AK4113_XTL_11_2896M	(0)
+/* 12.288 MHz ref. Xtal freq. */
+#define AK4113_XTL_12_288M	(AK4113_XTL0)
+/* 24.576 MHz ref. Xtal freq. */
+#define AK4113_XTL_24_576M	(AK4113_XTL1)
+
+/* AK4113_REG_IO1 */
+/* Interrupt 0 pin Hold */
+#define AK4113_EFH1		(1<<7)
+/* Interrupt 0 pin Hold */
+#define AK4113_EFH0		(1<<6)
+#define AK4113_EFH_512LRCLK	(0)
+#define AK4113_EFH_1024LRCLK	(AK4113_EFH0)
+#define AK4113_EFH_2048LRCLK	(AK4113_EFH1)
+#define AK4113_EFH_4096LRCLK	(AK4113_EFH1|AK4113_EFH0)
+/* PLL Lock Time: 0 = 384/fs, 1 = 1/fs */
+#define AK4113_FAST		(1<<5)
+/* MCKO2 Output Select: 0 = CMx/OCKSx, 1 = Xtal */
+#define AK4113_XMCK		(1<<4)
+/* MCKO2 Output Freq. Select: 0 = x1, 1 = x0.5  (req. XMCK = 1) */
+#define AK4113_DIV		(1<<3)
+/* Input Recovery Data Select */
+#define AK4113_IPS2		(1<<2)
+/* Input Recovery Data Select */
+#define AK4113_IPS1		(1<<1)
+/* Input Recovery Data Select */
+#define AK4113_IPS0		(1<<0)
+#define AK4113_IPS(x)		((x)&7)
+
+/* AK4113_REG_INT0_MASK && AK4113_REG_INT1_MASK*/
+/* mask enable for QINT bit */
+#define AK4113_MQI		(1<<7)
+/* mask enable for AUTO bit */
+#define AK4113_MAUT		(1<<6)
+/* mask enable for CINT bit */
+#define AK4113_MCIT		(1<<5)
+/* mask enable for UNLOCK bit */
+#define AK4113_MULK		(1<<4)
+/* mask enable for V bit */
+#define AK4113_V		(1<<3)
+/* mask enable for STC bit */
+#define AK4113_STC		(1<<2)
+/* mask enable for AUDN bit */
+#define AK4113_MAN		(1<<1)
+/* mask enable for PAR bit */
+#define AK4113_MPR		(1<<0)
+
+/* AK4113_REG_DATDTS */
+/* DAT Start ID Counter */
+#define AK4113_DCNT		(1<<4)
+/* DTS-CD 16-bit Sync Word Detect */
+#define AK4113_DTS16		(1<<3)
+/* DTS-CD 14-bit Sync Word Detect */
+#define AK4113_DTS14		(1<<2)
+/* mask enable for DAT bit (if 1, no INT1 effect */
+#define AK4113_MDAT1		(1<<1)
+/* mask enable for DAT bit (if 1, no INT0 effect */
+#define AK4113_MDAT0		(1<<0)
+
+/* AK4113_REG_RCS0 */
+/* Q-subcode buffer interrupt, 0 = no change, 1 = changed */
+#define AK4113_QINT		(1<<7)
+/* Non-PCM or DTS stream auto detection, 0 = no detect, 1 = detect */
+#define AK4113_AUTO		(1<<6)
+/* channel status buffer interrupt, 0 = no change, 1 = change */
+#define AK4113_CINT		(1<<5)
+/* PLL lock status, 0 = lock, 1 = unlock */
+#define AK4113_UNLCK		(1<<4)
+/* Validity bit, 0 = valid, 1 = invalid */
+#define AK4113_V		(1<<3)
+/* sampling frequency or Pre-emphasis change, 0 = no detect, 1 = detect */
+#define AK4113_STC		(1<<2)
+/* audio bit output, 0 = audio, 1 = non-audio */
+#define AK4113_AUDION		(1<<1)
+/* parity error or biphase error status, 0 = no error, 1 = error */
+#define AK4113_PAR		(1<<0)
+
+/* AK4113_REG_RCS1 */
+/* sampling frequency detection */
+#define AK4113_FS3		(1<<7)
+#define AK4113_FS2		(1<<6)
+#define AK4113_FS1		(1<<5)
+#define AK4113_FS0		(1<<4)
+/* Pre-emphasis detect, 0 = OFF, 1 = ON */
+#define AK4113_PEM		(1<<3)
+/* DAT Start ID Detect, 0 = no detect, 1 = detect */
+#define AK4113_DAT		(1<<2)
+/* DTS-CD bit audio stream detect, 0 = no detect, 1 = detect */
+#define AK4113_DTSCD		(1<<1)
+/* Non-PCM bit stream detection, 0 = no detect, 1 = detect */
+#define AK4113_NPCM		(1<<0)
+#define AK4113_FS_8000HZ	(AK4113_FS3|AK4113_FS0)
+#define AK4113_FS_11025HZ	(AK4113_FS2|AK4113_FS0)
+#define AK4113_FS_16000HZ	(AK4113_FS2|AK4113_FS1|AK4113_FS0)
+#define AK4113_FS_22050HZ	(AK4113_FS2)
+#define AK4113_FS_24000HZ	(AK4113_FS2|AK4113_FS1)
+#define AK4113_FS_32000HZ	(AK4113_FS1|AK4113_FS0)
+#define AK4113_FS_44100HZ	(0)
+#define AK4113_FS_48000HZ	(AK4113_FS1)
+#define AK4113_FS_64000HZ	(AK4113_FS3|AK4113_FS1|AK4113_FS0)
+#define AK4113_FS_88200HZ	(AK4113_FS3)
+#define AK4113_FS_96000HZ	(AK4113_FS3|AK4113_FS1)
+#define AK4113_FS_176400HZ	(AK4113_FS3|AK4113_FS2)
+#define AK4113_FS_192000HZ	(AK4113_FS3|AK4113_FS2|AK4113_FS1)
+
+/* AK4113_REG_RCS2 */
+/* CRC for Q-subcode, 0 = no error, 1 = error */
+#define AK4113_QCRC		(1<<1)
+/* CRC for channel status, 0 = no error, 1 = error */
+#define AK4113_CCRC		(1<<0)
+
+/* flags for snd_ak4113_check_rate_and_errors() */
+#define AK4113_CHECK_NO_STAT	(1<<0)	/* no statistics */
+#define AK4113_CHECK_NO_RATE	(1<<1)	/* no rate check */
+
+#define AK4113_CONTROLS		13
+
+typedef void (ak4113_write_t)(void *private_data, unsigned char addr,
+		unsigned char data);
+typedef unsigned char (ak4113_read_t)(void *private_data, unsigned char addr);
+
+struct ak4113 {
+	struct snd_card *card;
+	ak4113_write_t *write;
+	ak4113_read_t *read;
+	void *private_data;
+	unsigned int init:1;
+	spinlock_t lock;
+	unsigned char regmap[AK4113_WRITABLE_REGS];
+	struct snd_kcontrol *kctls[AK4113_CONTROLS];
+	struct snd_pcm_substream *substream;
+	unsigned long parity_errors;
+	unsigned long v_bit_errors;
+	unsigned long qcrc_errors;
+	unsigned long ccrc_errors;
+	unsigned char rcs0;
+	unsigned char rcs1;
+	unsigned char rcs2;
+	struct delayed_work work;
+	unsigned int check_flags;
+	void *change_callback_private;
+	void (*change_callback)(struct ak4113 *ak4113, unsigned char c0,
+			unsigned char c1);
+};
+
+int snd_ak4113_create(struct snd_card *card, ak4113_read_t *read,
+		ak4113_write_t *write,
+		const unsigned char *pgm,
+		void *private_data, struct ak4113 **r_ak4113);
+void snd_ak4113_reg_write(struct ak4113 *ak4113, unsigned char reg,
+		unsigned char mask, unsigned char val);
+void snd_ak4113_reinit(struct ak4113 *ak4113);
+int snd_ak4113_build(struct ak4113 *ak4113,
+		struct snd_pcm_substream *capture_substream);
+int snd_ak4113_external_rate(struct ak4113 *ak4113);
+int snd_ak4113_check_rate_and_errors(struct ak4113 *ak4113, unsigned int flags);
+
+#endif /* __SOUND_AK4113_H */
+
diff --git a/include/sound/ak4114.h b/include/sound/ak4114.h
new file mode 100644
index 0000000..3ce69fd
--- /dev/null
+++ b/include/sound/ak4114.h
@@ -0,0 +1,203 @@
+#ifndef __SOUND_AK4114_H
+#define __SOUND_AK4114_H
+
+/*
+ *  Routines for Asahi Kasei AK4114
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
+ *
+ *
+ *   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., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ */
+
+/* AK4114 registers */
+#define AK4114_REG_PWRDN	0x00	/* power down */
+#define AK4114_REG_FORMAT	0x01	/* format control */
+#define AK4114_REG_IO0		0x02	/* input/output control */
+#define AK4114_REG_IO1		0x03	/* input/output control */
+#define AK4114_REG_INT0_MASK	0x04	/* interrupt0 mask */
+#define AK4114_REG_INT1_MASK	0x05	/* interrupt1 mask */
+#define AK4114_REG_RCS0		0x06	/* receiver status 0 */
+#define AK4114_REG_RCS1		0x07	/* receiver status 1 */
+#define AK4114_REG_RXCSB0	0x08	/* RX channel status byte 0 */
+#define AK4114_REG_RXCSB1	0x09	/* RX channel status byte 1 */
+#define AK4114_REG_RXCSB2	0x0a	/* RX channel status byte 2 */
+#define AK4114_REG_RXCSB3	0x0b	/* RX channel status byte 3 */
+#define AK4114_REG_RXCSB4	0x0c	/* RX channel status byte 4 */
+#define AK4114_REG_TXCSB0	0x0d	/* TX channel status byte 0 */
+#define AK4114_REG_TXCSB1	0x0e	/* TX channel status byte 1 */
+#define AK4114_REG_TXCSB2	0x0f	/* TX channel status byte 2 */
+#define AK4114_REG_TXCSB3	0x10	/* TX channel status byte 3 */
+#define AK4114_REG_TXCSB4	0x11	/* TX channel status byte 4 */
+#define AK4114_REG_Pc0		0x12	/* burst preamble Pc byte 0 */
+#define AK4114_REG_Pc1		0x13	/* burst preamble Pc byte 1 */
+#define AK4114_REG_Pd0		0x14	/* burst preamble Pd byte 0 */
+#define AK4114_REG_Pd1		0x15	/* burst preamble Pd byte 1 */
+#define AK4114_REG_QSUB_ADDR	0x16	/* Q-subcode address + control */
+#define AK4114_REG_QSUB_TRACK	0x17	/* Q-subcode track */
+#define AK4114_REG_QSUB_INDEX	0x18	/* Q-subcode index */
+#define AK4114_REG_QSUB_MINUTE	0x19	/* Q-subcode minute */
+#define AK4114_REG_QSUB_SECOND	0x1a	/* Q-subcode second */
+#define AK4114_REG_QSUB_FRAME	0x1b	/* Q-subcode frame */
+#define AK4114_REG_QSUB_ZERO	0x1c	/* Q-subcode zero */
+#define AK4114_REG_QSUB_ABSMIN	0x1d	/* Q-subcode absolute minute */
+#define AK4114_REG_QSUB_ABSSEC	0x1e	/* Q-subcode absolute second */
+#define AK4114_REG_QSUB_ABSFRM	0x1f	/* Q-subcode absolute frame */
+
+/* sizes */
+#define AK4114_REG_RXCSB_SIZE	((AK4114_REG_RXCSB4-AK4114_REG_RXCSB0)+1)
+#define AK4114_REG_TXCSB_SIZE	((AK4114_REG_TXCSB4-AK4114_REG_TXCSB0)+1)
+#define AK4114_REG_QSUB_SIZE	((AK4114_REG_QSUB_ABSFRM-AK4114_REG_QSUB_ADDR)+1)
+
+/* AK4117_REG_PWRDN bits */
+#define AK4114_CS12		(1<<7)	/* Channel Status Select */
+#define AK4114_BCU		(1<<6)	/* Block Start & C/U Output Mode */
+#define AK4114_CM1		(1<<5)	/* Master Clock Operation Select */
+#define AK4114_CM0		(1<<4)	/* Master Clock Operation Select */
+#define AK4114_OCKS1		(1<<3)	/* Master Clock Frequency Select */
+#define AK4114_OCKS0		(1<<2)	/* Master Clock Frequency Select */
+#define AK4114_PWN		(1<<1)	/* 0 = power down, 1 = normal operation */
+#define AK4114_RST		(1<<0)	/* 0 = reset & initialize (except this register), 1 = normal operation */
+
+/* AK4114_REQ_FORMAT bits */
+#define AK4114_MONO		(1<<7)	/* Double Sampling Frequency Mode: 0 = stereo, 1 = mono */
+#define AK4114_DIF2		(1<<6)	/* Audio Data Control */
+#define AK4114_DIF1		(1<<5)	/* Audio Data Control */
+#define AK4114_DIF0		(1<<4)	/* Audio Data Control */
+#define AK4114_DIF_16R		(0)				/* STDO: 16-bit, right justified */
+#define AK4114_DIF_18R		(AK4114_DIF0)			/* STDO: 18-bit, right justified */
+#define AK4114_DIF_20R		(AK4114_DIF1)			/* STDO: 20-bit, right justified */
+#define AK4114_DIF_24R		(AK4114_DIF1|AK4114_DIF0)	/* STDO: 24-bit, right justified */
+#define AK4114_DIF_24L		(AK4114_DIF2)			/* STDO: 24-bit, left justified */
+#define AK4114_DIF_24I2S	(AK4114_DIF2|AK4114_DIF0)	/* STDO: I2S */
+#define AK4114_DIF_I24L		(AK4114_DIF2|AK4114_DIF1)	/* STDO: 24-bit, left justified; LRCLK, BICK = Input */
+#define AK4114_DIF_I24I2S	(AK4114_DIF2|AK4114_DIF1|AK4114_DIF0) /* STDO: I2S;  LRCLK, BICK = Input */
+#define AK4114_DEAU		(1<<3)	/* Deemphasis Autodetect Enable (1 = enable) */
+#define AK4114_DEM1		(1<<2)	/* 32kHz-48kHz Deemphasis Control */
+#define AK4114_DEM0		(1<<1)	/* 32kHz-48kHz Deemphasis Control */
+#define AK4114_DEM_44KHZ	(0)
+#define AK4114_DEM_48KHZ	(AK4114_DEM1)
+#define AK4114_DEM_32KHZ	(AK4114_DEM0|AK4114_DEM1)
+#define AK4114_DEM_96KHZ	(AK4114_DEM1)	/* DFS must be set */
+#define AK4114_DFS		(1<<0)	/* 96kHz Deemphasis Control */
+
+/* AK4114_REG_IO0 */
+#define AK4114_TX1E		(1<<7)	/* TX1 Output Enable (1 = enable) */
+#define AK4114_OPS12		(1<<6)	/* Output Data Selector for TX1 pin */
+#define AK4114_OPS11		(1<<5)	/* Output Data Selector for TX1 pin */
+#define AK4114_OPS10		(1<<4)	/* Output Data Selector for TX1 pin */
+#define AK4114_TX0E		(1<<3)	/* TX0 Output Enable (1 = enable) */
+#define AK4114_OPS02		(1<<2)	/* Output Data Selector for TX0 pin */
+#define AK4114_OPS01		(1<<1)	/* Output Data Selector for TX0 pin */
+#define AK4114_OPS00		(1<<0)	/* Output Data Selector for TX0 pin */
+
+/* AK4114_REG_IO1 */
+#define AK4114_EFH1		(1<<7)	/* Interrupt 0 pin Hold */
+#define AK4114_EFH0		(1<<6)	/* Interrupt 0 pin Hold */
+#define AK4114_EFH_512		(0)
+#define AK4114_EFH_1024		(AK4114_EFH0)
+#define AK4114_EFH_2048		(AK4114_EFH1)
+#define AK4114_EFH_4096		(AK4114_EFH1|AK4114_EFH0)
+#define AK4114_UDIT		(1<<5)	/* U-bit Control for DIT (0 = fixed '0', 1 = recovered) */
+#define AK4114_TLR		(1<<4)	/* Double Sampling Frequency Select for DIT (0 = L channel, 1 = R channel) */
+#define AK4114_DIT		(1<<3)	/* TX1 out: 0 = Through Data (RX data), 1 = Transmit Data (DAUX data) */
+#define AK4114_IPS2		(1<<2)	/* Input Recovery Data Select */
+#define AK4114_IPS1		(1<<1)	/* Input Recovery Data Select */
+#define AK4114_IPS0		(1<<0)	/* Input Recovery Data Select */
+#define AK4114_IPS(x)		((x)&7)
+
+/* AK4114_REG_INT0_MASK && AK4114_REG_INT1_MASK*/
+#define AK4117_MQI              (1<<7)  /* mask enable for QINT bit */
+#define AK4117_MAT              (1<<6)  /* mask enable for AUTO bit */
+#define AK4117_MCI              (1<<5)  /* mask enable for CINT bit */
+#define AK4117_MUL              (1<<4)  /* mask enable for UNLOCK bit */
+#define AK4117_MDTS             (1<<3)  /* mask enable for DTSCD bit */
+#define AK4117_MPE              (1<<2)  /* mask enable for PEM bit */
+#define AK4117_MAN              (1<<1)  /* mask enable for AUDN bit */
+#define AK4117_MPR              (1<<0)  /* mask enable for PAR bit */
+
+/* AK4114_REG_RCS0 */
+#define AK4114_QINT		(1<<7)	/* Q-subcode buffer interrupt, 0 = no change, 1 = changed */
+#define AK4114_AUTO		(1<<6)	/* Non-PCM or DTS stream auto detection, 0 = no detect, 1 = detect */
+#define AK4114_CINT		(1<<5)	/* channel status buffer interrupt, 0 = no change, 1 = change */
+#define AK4114_UNLCK		(1<<4)	/* PLL lock status, 0 = lock, 1 = unlock */
+#define AK4114_DTSCD		(1<<3)	/* DTS-CD Detect, 0 = No detect, 1 = Detect */
+#define AK4114_PEM		(1<<2)	/* Pre-emphasis Detect, 0 = OFF, 1 = ON */
+#define AK4114_AUDION		(1<<1)	/* audio bit output, 0 = audio, 1 = non-audio */
+#define AK4114_PAR		(1<<0)	/* parity error or biphase error status, 0 = no error, 1 = error */
+
+/* AK4114_REG_RCS1 */
+#define AK4114_FS3		(1<<7)	/* sampling frequency detection */
+#define AK4114_FS2		(1<<6)
+#define AK4114_FS1		(1<<5)
+#define AK4114_FS0		(1<<4)
+#define AK4114_FS_44100HZ	(0)
+#define AK4114_FS_48000HZ	(AK4114_FS1)
+#define AK4114_FS_32000HZ	(AK4114_FS1|AK4114_FS0)
+#define AK4114_FS_88200HZ	(AK4114_FS3)
+#define AK4114_FS_96000HZ	(AK4114_FS3|AK4114_FS1)
+#define AK4114_FS_176400HZ	(AK4114_FS3|AK4114_FS2)
+#define AK4114_FS_192000HZ	(AK4114_FS3|AK4114_FS2|AK4114_FS1)
+#define AK4114_V		(1<<3)	/* Validity of Channel Status, 0 = Valid, 1 = Invalid */
+#define AK4114_QCRC		(1<<1)	/* CRC for Q-subcode, 0 = no error, 1 = error */
+#define AK4114_CCRC		(1<<0)	/* CRC for channel status, 0 = no error, 1 = error */
+
+/* flags for snd_ak4114_check_rate_and_errors() */
+#define AK4114_CHECK_NO_STAT	(1<<0)	/* no statistics */
+#define AK4114_CHECK_NO_RATE	(1<<1)	/* no rate check */
+
+#define AK4114_CONTROLS		15
+
+typedef void (ak4114_write_t)(void *private_data, unsigned char addr, unsigned char data);
+typedef unsigned char (ak4114_read_t)(void *private_data, unsigned char addr);
+
+struct ak4114 {
+	struct snd_card *card;
+	ak4114_write_t * write;
+	ak4114_read_t * read;
+	void * private_data;
+	unsigned int init: 1;
+	spinlock_t lock;
+	unsigned char regmap[7];
+	unsigned char txcsb[5];
+	struct snd_kcontrol *kctls[AK4114_CONTROLS];
+	struct snd_pcm_substream *playback_substream;
+	struct snd_pcm_substream *capture_substream;
+	unsigned long parity_errors;
+	unsigned long v_bit_errors;
+	unsigned long qcrc_errors;
+	unsigned long ccrc_errors;
+	unsigned char rcs0;
+	unsigned char rcs1;
+	struct delayed_work work;
+	unsigned int check_flags;
+	void *change_callback_private;
+	void (*change_callback)(struct ak4114 *ak4114, unsigned char c0, unsigned char c1);
+};
+
+int snd_ak4114_create(struct snd_card *card,
+		      ak4114_read_t *read, ak4114_write_t *write,
+		      const unsigned char pgm[7], const unsigned char txcsb[5],
+		      void *private_data, struct ak4114 **r_ak4114);
+void snd_ak4114_reg_write(struct ak4114 *ak4114, unsigned char reg, unsigned char mask, unsigned char val);
+void snd_ak4114_reinit(struct ak4114 *ak4114);
+int snd_ak4114_build(struct ak4114 *ak4114,
+		     struct snd_pcm_substream *playback_substream,
+                     struct snd_pcm_substream *capture_substream);
+int snd_ak4114_external_rate(struct ak4114 *ak4114);
+int snd_ak4114_check_rate_and_errors(struct ak4114 *ak4114, unsigned int flags);
+
+#endif /* __SOUND_AK4114_H */
+
diff --git a/include/sound/ak4117.h b/include/sound/ak4117.h
new file mode 100644
index 0000000..1e81781
--- /dev/null
+++ b/include/sound/ak4117.h
@@ -0,0 +1,189 @@
+#ifndef __SOUND_AK4117_H
+#define __SOUND_AK4117_H
+
+/*
+ *  Routines for Asahi Kasei AK4117
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
+ *
+ *
+ *   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., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ */
+
+#define AK4117_REG_PWRDN	0x00	/* power down */
+#define AK4117_REG_CLOCK	0x01	/* clock control */
+#define AK4117_REG_IO		0x02	/* input/output control */
+#define AK4117_REG_INT0_MASK	0x03	/* interrupt0 mask */
+#define AK4117_REG_INT1_MASK	0x04	/* interrupt1 mask */
+#define AK4117_REG_RCS0		0x05	/* receiver status 0 */
+#define AK4117_REG_RCS1		0x06	/* receiver status 1 */
+#define AK4117_REG_RCS2		0x07	/* receiver status 2 */
+#define AK4117_REG_RXCSB0	0x08	/* RX channel status byte 0 */
+#define AK4117_REG_RXCSB1	0x09	/* RX channel status byte 1 */
+#define AK4117_REG_RXCSB2	0x0a	/* RX channel status byte 2 */
+#define AK4117_REG_RXCSB3	0x0b	/* RX channel status byte 3 */
+#define AK4117_REG_RXCSB4	0x0c	/* RX channel status byte 4 */
+#define AK4117_REG_Pc0		0x0d	/* burst preamble Pc byte 0 */
+#define AK4117_REG_Pc1		0x0e	/* burst preamble Pc byte 1 */
+#define AK4117_REG_Pd0		0x0f	/* burst preamble Pd byte 0 */
+#define AK4117_REG_Pd1		0x10	/* burst preamble Pd byte 1 */
+#define AK4117_REG_QSUB_ADDR	0x11	/* Q-subcode address + control */
+#define AK4117_REG_QSUB_TRACK	0x12	/* Q-subcode track */
+#define AK4117_REG_QSUB_INDEX	0x13	/* Q-subcode index */
+#define AK4117_REG_QSUB_MINUTE	0x14	/* Q-subcode minute */
+#define AK4117_REG_QSUB_SECOND	0x15	/* Q-subcode second */
+#define AK4117_REG_QSUB_FRAME	0x16	/* Q-subcode frame */
+#define AK4117_REG_QSUB_ZERO	0x17	/* Q-subcode zero */
+#define AK4117_REG_QSUB_ABSMIN	0x18	/* Q-subcode absolute minute */
+#define AK4117_REG_QSUB_ABSSEC	0x19	/* Q-subcode absolute second */
+#define AK4117_REG_QSUB_ABSFRM	0x1a	/* Q-subcode absolute frame */
+
+/* sizes */
+#define AK4117_REG_RXCSB_SIZE	((AK4117_REG_RXCSB4-AK4117_REG_RXCSB0)+1)
+#define AK4117_REG_QSUB_SIZE	((AK4117_REG_QSUB_ABSFRM-AK4117_REG_QSUB_ADDR)+1)
+
+/* AK4117_REG_PWRDN bits */
+#define AK4117_EXCT		(1<<4)	/* 0 = X'tal mode, 1 = external clock mode */
+#define AK4117_XTL1		(1<<3)	/* XTL1=0,XTL0=0 -> 11.2896Mhz; XTL1=0,XTL0=1 -> 12.288Mhz */
+#define AK4117_XTL0		(1<<2)	/* XTL1=1,XTL0=0 -> 24.576Mhz; XTL1=1,XTL0=1 -> use channel status */
+#define AK4117_XTL_11_2896M	(0)
+#define AK4117_XTL_12_288M	AK4117_XTL0
+#define AK4117_XTL_24_576M	AK4117_XTL1
+#define AK4117_XTL_EXT		(AK4117_XTL1|AK4117_XTL0)
+#define AK4117_PWN		(1<<1)	/* 0 = power down, 1 = normal operation */
+#define AK4117_RST		(1<<0)	/* 0 = reset & initialize (except this register), 1 = normal operation */
+
+/* AK4117_REQ_CLOCK bits */
+#define AK4117_LP		(1<<7)	/* 0 = normal mode, 1 = low power mode (Fs up to 48kHz only) */
+#define AK4117_PKCS1		(1<<6)	/* master clock frequency at PLL mode (when LP == 0) */
+#define AK4117_PKCS0		(1<<5)
+#define AK4117_PKCS_512fs	(0)
+#define AK4117_PKCS_256fs	AK4117_PKCS0
+#define AK4117_PKCS_128fs	AK4117_PKCS1
+#define AK4117_DIV		(1<<4)	/* 0 = MCKO == Fs, 1 = MCKO == Fs / 2; X'tal mode only */
+#define AK4117_XCKS1		(1<<3)	/* master clock frequency at X'tal mode */
+#define AK4117_XCKS0		(1<<2)
+#define AK4117_XCKS_128fs	(0)
+#define AK4117_XCKS_256fs	AK4117_XCKS0
+#define AK4117_XCKS_512fs	AK4117_XCKS1
+#define AK4117_XCKS_1024fs	(AK4117_XCKS1|AK4117_XCKS0)
+#define AK4117_CM1		(1<<1)	/* MCKO operation mode select */
+#define AK4117_CM0		(1<<0)
+#define AK4117_CM_PLL		(0)		/* use RX input as master clock */
+#define AK4117_CM_XTAL		(AK4117_CM0)	/* use X'tal as master clock */
+#define AK4117_CM_PLL_XTAL	(AK4117_CM1)	/* use Rx input but X'tal when PLL loses lock */
+#define AK4117_CM_MONITOR	(AK4117_CM0|AK4117_CM1) /* use X'tal as master clock, but use PLL for monitoring */
+
+/* AK4117_REG_IO */
+#define AK4117_IPS		(1<<7)	/* Input Recovery Data Select, 0 = RX0, 1 = RX1 */
+#define AK4117_UOUTE		(1<<6)	/* U-bit output enable to UOUT, 0 = disable, 1 = enable */
+#define AK4117_CS12		(1<<5)	/* channel status select, 0 = channel1, 1 = channel2 */
+#define AK4117_EFH2		(1<<4)	/* INT0 pin hold count select */
+#define AK4117_EFH1		(1<<3)
+#define AK4117_EFH_512LRCLK	(0)
+#define AK4117_EFH_1024LRCLK	(AK4117_EFH1)
+#define AK4117_EFH_2048LRCLK	(AK4117_EFH2)
+#define AK4117_EFH_4096LRCLK	(AK4117_EFH1|AK4117_EFH2)
+#define AK4117_DIF2		(1<<2)	/* audio data format control */
+#define AK4117_DIF1		(1<<1)
+#define AK4117_DIF0		(1<<0)
+#define AK4117_DIF_16R		(0)				/* STDO: 16-bit, right justified */
+#define AK4117_DIF_18R		(AK4117_DIF0)			/* STDO: 18-bit, right justified */
+#define AK4117_DIF_20R		(AK4117_DIF1)			/* STDO: 20-bit, right justified */
+#define AK4117_DIF_24R		(AK4117_DIF1|AK4117_DIF0)	/* STDO: 24-bit, right justified */
+#define AK4117_DIF_24L		(AK4117_DIF2)			/* STDO: 24-bit, left justified */
+#define AK4117_DIF_24I2S	(AK4117_DIF2|AK4117_DIF0)	/* STDO: I2S */
+
+/* AK4117_REG_INT0_MASK & AK4117_REG_INT1_MASK */
+#define AK4117_MULK		(1<<7)	/* mask enable for UNLOCK bit */
+#define AK4117_MPAR		(1<<6)	/* mask enable for PAR bit */
+#define AK4117_MAUTO		(1<<5)	/* mask enable for AUTO bit */
+#define AK4117_MV		(1<<4)	/* mask enable for V bit */
+#define AK4117_MAUD		(1<<3)	/* mask enable for AUDION bit */
+#define AK4117_MSTC		(1<<2)	/* mask enable for STC bit */
+#define AK4117_MCIT		(1<<1)	/* mask enable for CINT bit */
+#define AK4117_MQIT		(1<<0)	/* mask enable for QINT bit */
+
+/* AK4117_REG_RCS0 */
+#define AK4117_UNLCK		(1<<7)	/* PLL lock status, 0 = lock, 1 = unlock */
+#define AK4117_PAR		(1<<6)	/* parity error or biphase error status, 0 = no error, 1 = error */
+#define AK4117_AUTO		(1<<5)	/* Non-PCM or DTS stream auto detection, 0 = no detect, 1 = detect */
+#define AK4117_V		(1<<4)	/* Validity bit, 0 = valid, 1 = invalid */
+#define AK4117_AUDION		(1<<3)	/* audio bit output, 0 = audio, 1 = non-audio */
+#define AK4117_STC		(1<<2)	/* sampling frequency or Pre-emphasis change, 0 = no detect, 1 = detect */
+#define AK4117_CINT		(1<<1)	/* channel status buffer interrupt, 0 = no change, 1 = change */
+#define AK4117_QINT		(1<<0)	/* Q-subcode buffer interrupt, 0 = no change, 1 = changed */
+
+/* AK4117_REG_RCS1 */
+#define AK4117_DTSCD		(1<<6)	/* DTS-CD bit audio stream detect, 0 = no detect, 1 = detect */
+#define AK4117_NPCM		(1<<5)	/* Non-PCM bit stream detection, 0 = no detect, 1 = detect */
+#define AK4117_PEM		(1<<4)	/* Pre-emphasis detect, 0 = OFF, 1 = ON */
+#define AK4117_FS3		(1<<3)	/* sampling frequency detection */
+#define AK4117_FS2		(1<<2)
+#define AK4117_FS1		(1<<1)
+#define AK4117_FS0		(1<<0)
+#define AK4117_FS_44100HZ	(0)
+#define AK4117_FS_48000HZ	(AK4117_FS1)
+#define AK4117_FS_32000HZ	(AK4117_FS1|AK4117_FS0)
+#define AK4117_FS_88200HZ	(AK4117_FS3)
+#define AK4117_FS_96000HZ	(AK4117_FS3|AK4117_FS1)
+#define AK4117_FS_176400HZ	(AK4117_FS3|AK4117_FS2)
+#define AK4117_FS_192000HZ	(AK4117_FS3|AK4117_FS2|AK4117_FS1)
+
+/* AK4117_REG_RCS2 */
+#define AK4117_CCRC		(1<<1)	/* CRC for channel status, 0 = no error, 1 = error */
+#define AK4117_QCRC		(1<<0)	/* CRC for Q-subcode, 0 = no error, 1 = error */
+
+/* flags for snd_ak4117_check_rate_and_errors() */
+#define AK4117_CHECK_NO_STAT	(1<<0)	/* no statistics */
+#define AK4117_CHECK_NO_RATE	(1<<1)	/* no rate check */
+
+#define AK4117_CONTROLS		13
+
+typedef void (ak4117_write_t)(void *private_data, unsigned char addr, unsigned char data);
+typedef unsigned char (ak4117_read_t)(void *private_data, unsigned char addr);
+
+struct ak4117 {
+	struct snd_card *card;
+	ak4117_write_t * write;
+	ak4117_read_t * read;
+	void * private_data;
+	unsigned int init: 1;
+	spinlock_t lock;
+	unsigned char regmap[5];
+	struct snd_kcontrol *kctls[AK4117_CONTROLS];
+	struct snd_pcm_substream *substream;
+	unsigned long parity_errors;
+	unsigned long v_bit_errors;
+	unsigned long qcrc_errors;
+	unsigned long ccrc_errors;
+	unsigned char rcs0;
+	unsigned char rcs1;
+	unsigned char rcs2;
+	struct timer_list timer;	/* statistic timer */
+	void *change_callback_private;
+	void (*change_callback)(struct ak4117 *ak4117, unsigned char c0, unsigned char c1);
+};
+
+int snd_ak4117_create(struct snd_card *card, ak4117_read_t *read, ak4117_write_t *write,
+		      const unsigned char pgm[5], void *private_data, struct ak4117 **r_ak4117);
+void snd_ak4117_reg_write(struct ak4117 *ak4117, unsigned char reg, unsigned char mask, unsigned char val);
+void snd_ak4117_reinit(struct ak4117 *ak4117);
+int snd_ak4117_build(struct ak4117 *ak4117, struct snd_pcm_substream *capture_substream);
+int snd_ak4117_external_rate(struct ak4117 *ak4117);
+int snd_ak4117_check_rate_and_errors(struct ak4117 *ak4117, unsigned int flags);
+
+#endif /* __SOUND_AK4117_H */
+
diff --git a/include/sound/ak4531_codec.h b/include/sound/ak4531_codec.h
new file mode 100644
index 0000000..575296c
--- /dev/null
+++ b/include/sound/ak4531_codec.h
@@ -0,0 +1,85 @@
+#ifndef __SOUND_AK4531_CODEC_H
+#define __SOUND_AK4531_CODEC_H
+
+/*
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *  Universal interface for Audio Codec '97
+ *
+ *  For more details look to AC '97 component specification revision 2.1
+ *  by Intel Corporation (http://developer.intel.com).
+ *
+ *
+ *   This program is free software; you can redistribute it and/or modify
+ *   it under the terms of the GNU General Public License as published by
+ *   the Free Software Foundation; either version 2 of the License, or
+ *   (at your option) any later version.
+ *
+ *   This program is distributed in the hope that it will be useful,
+ *   but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *   GNU General Public License for more details.
+ *
+ *   You should have received a copy of the GNU General Public License
+ *   along with this program; if not, write to the Free Software
+ *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ */
+
+#include "info.h"
+#include "control.h"
+
+/*
+ *  ASAHI KASEI - AK4531 codec
+ *  - not really AC'97 codec, but it uses very similar interface as AC'97
+ */
+
+/*
+ *  AK4531 codec registers
+ */
+
+#define AK4531_LMASTER  0x00	/* master volume left */
+#define AK4531_RMASTER  0x01	/* master volume right */
+#define AK4531_LVOICE   0x02	/* channel volume left */
+#define AK4531_RVOICE   0x03	/* channel volume right */
+#define AK4531_LFM      0x04	/* FM volume left */
+#define AK4531_RFM      0x05	/* FM volume right */
+#define AK4531_LCD      0x06	/* CD volume left */
+#define AK4531_RCD      0x07	/* CD volume right */
+#define AK4531_LLINE    0x08	/* LINE volume left */
+#define AK4531_RLINE    0x09	/* LINE volume right */
+#define AK4531_LAUXA    0x0a	/* AUXA volume left */
+#define AK4531_RAUXA    0x0b	/* AUXA volume right */
+#define AK4531_MONO1    0x0c	/* MONO1 volume left */
+#define AK4531_MONO2    0x0d	/* MONO1 volume right */
+#define AK4531_MIC      0x0e	/* MIC volume */
+#define AK4531_MONO_OUT 0x0f	/* Mono-out volume */
+#define AK4531_OUT_SW1  0x10	/* Output mixer switch 1 */
+#define AK4531_OUT_SW2  0x11	/* Output mixer switch 2 */
+#define AK4531_LIN_SW1  0x12	/* Input left mixer switch 1 */
+#define AK4531_RIN_SW1  0x13	/* Input right mixer switch 1 */
+#define AK4531_LIN_SW2  0x14	/* Input left mixer switch 2 */
+#define AK4531_RIN_SW2  0x15	/* Input right mixer switch 2 */
+#define AK4531_RESET    0x16	/* Reset & power down */
+#define AK4531_CLOCK    0x17	/* Clock select */
+#define AK4531_AD_IN    0x18	/* AD input select */
+#define AK4531_MIC_GAIN 0x19	/* MIC amplified gain */
+
+struct snd_ak4531 {
+	void (*write) (struct snd_ak4531 *ak4531, unsigned short reg,
+		       unsigned short val);
+	void *private_data;
+	void (*private_free) (struct snd_ak4531 *ak4531);
+	/* --- */
+	unsigned char regs[0x20];
+	struct mutex reg_mutex;
+};
+
+int snd_ak4531_mixer(struct snd_card *card, struct snd_ak4531 *_ak4531,
+		     struct snd_ak4531 **rak4531);
+
+#ifdef CONFIG_PM
+void snd_ak4531_suspend(struct snd_ak4531 *ak4531);
+void snd_ak4531_resume(struct snd_ak4531 *ak4531);
+#endif
+
+#endif /* __SOUND_AK4531_CODEC_H */
diff --git a/include/sound/ak4xxx-adda.h b/include/sound/ak4xxx-adda.h
new file mode 100644
index 0000000..030b87c
--- /dev/null
+++ b/include/sound/ak4xxx-adda.h
@@ -0,0 +1,99 @@
+#ifndef __SOUND_AK4XXX_ADDA_H
+#define __SOUND_AK4XXX_ADDA_H
+
+/*
+ *   ALSA driver for AK4524 / AK4528 / AK4529 / AK4355 / AK4381
+ *   AD and DA converters
+ *
+ *	Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
+ *
+ *   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., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ */      
+
+#ifndef AK4XXX_MAX_CHIPS
+#define AK4XXX_MAX_CHIPS	4
+#endif
+
+struct snd_akm4xxx;
+
+struct snd_ak4xxx_ops {
+	void (*lock)(struct snd_akm4xxx *ak, int chip);
+	void (*unlock)(struct snd_akm4xxx *ak, int chip);
+	void (*write)(struct snd_akm4xxx *ak, int chip, unsigned char reg,
+		      unsigned char val);
+	void (*set_rate_val)(struct snd_akm4xxx *ak, unsigned int rate);
+};
+
+#define AK4XXX_IMAGE_SIZE	(AK4XXX_MAX_CHIPS * 16)	/* 64 bytes */
+
+/* DAC label and channels */
+struct snd_akm4xxx_dac_channel {
+	char *name;		/* mixer volume name */
+	unsigned int num_channels;
+	char *switch_name;		/* mixer switch*/
+};
+
+/* ADC labels and channels */
+struct snd_akm4xxx_adc_channel {
+	char *name;		/* capture gain volume label */
+	char *switch_name;	/* capture switch */
+	unsigned int num_channels;
+	char *selector_name;	/* capture source select label */
+	const char **input_names; /* capture source names (NULL terminated) */
+};
+
+struct snd_akm4xxx {
+	struct snd_card *card;
+	unsigned int num_adcs;			/* AK4524 or AK4528 ADCs */
+	unsigned int num_dacs;			/* AK4524 or AK4528 DACs */
+	unsigned char images[AK4XXX_IMAGE_SIZE]; /* saved register image */
+	unsigned char volumes[AK4XXX_IMAGE_SIZE]; /* saved volume values */
+	unsigned long private_value[AK4XXX_MAX_CHIPS];	/* helper for driver */
+	void *private_data[AK4XXX_MAX_CHIPS];		/* helper for driver */
+	/* template should fill the following fields */
+	unsigned int idx_offset;		/* control index offset */
+	enum {
+		SND_AK4524, SND_AK4528, SND_AK4529,
+		SND_AK4355, SND_AK4358, SND_AK4381,
+		SND_AK5365, SND_AK4620,
+	} type;
+
+	/* (array) information of combined codecs */
+	const struct snd_akm4xxx_dac_channel *dac_info;
+	const struct snd_akm4xxx_adc_channel *adc_info;
+
+	struct snd_ak4xxx_ops ops;
+	unsigned int num_chips;
+	unsigned int total_regs;
+	const char *name;
+};
+
+void snd_akm4xxx_write(struct snd_akm4xxx *ak, int chip, unsigned char reg,
+		       unsigned char val);
+void snd_akm4xxx_reset(struct snd_akm4xxx *ak, int state);
+void snd_akm4xxx_init(struct snd_akm4xxx *ak);
+int snd_akm4xxx_build_controls(struct snd_akm4xxx *ak);
+
+#define snd_akm4xxx_get(ak,chip,reg) \
+	(ak)->images[(chip) * 16 + (reg)]
+#define snd_akm4xxx_set(ak,chip,reg,val) \
+	((ak)->images[(chip) * 16 + (reg)] = (val))
+#define snd_akm4xxx_get_vol(ak,chip,reg) \
+	(ak)->volumes[(chip) * 16 + (reg)]
+#define snd_akm4xxx_set_vol(ak,chip,reg,val) \
+	((ak)->volumes[(chip) * 16 + (reg)] = (val))
+
+#endif /* __SOUND_AK4XXX_ADDA_H */
diff --git a/include/sound/asequencer.h b/include/sound/asequencer.h
new file mode 100644
index 0000000..1505e6d
--- /dev/null
+++ b/include/sound/asequencer.h
@@ -0,0 +1,678 @@
+/*
+ *  Main header file for the ALSA sequencer
+ *  Copyright (c) 1998-1999 by Frank van de Pol <fvdpol@coil.demon.nl>
+ *            (c) 1998-1999 by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ *   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., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ */
+#ifndef __SOUND_ASEQUENCER_H
+#define __SOUND_ASEQUENCER_H
+
+#ifdef __KERNEL__
+#include <linux/ioctl.h>
+#include <sound/asound.h>
+#endif
+
+/** version of the sequencer */
+#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 1)
+
+/**
+ * definition of sequencer event types
+ */
+
+/** system messages
+ * event data type = #snd_seq_result
+ */
+#define SNDRV_SEQ_EVENT_SYSTEM		0
+#define SNDRV_SEQ_EVENT_RESULT		1
+
+/** note messages (channel specific)
+ * event data type = #snd_seq_ev_note
+ */
+#define SNDRV_SEQ_EVENT_NOTE		5
+#define SNDRV_SEQ_EVENT_NOTEON		6
+#define SNDRV_SEQ_EVENT_NOTEOFF		7
+#define SNDRV_SEQ_EVENT_KEYPRESS	8
+	
+/** control messages (channel specific)
+ * event data type = #snd_seq_ev_ctrl
+ */
+#define SNDRV_SEQ_EVENT_CONTROLLER	10
+#define SNDRV_SEQ_EVENT_PGMCHANGE	11
+#define SNDRV_SEQ_EVENT_CHANPRESS	12
+#define SNDRV_SEQ_EVENT_PITCHBEND	13	/**< from -8192 to 8191 */
+#define SNDRV_SEQ_EVENT_CONTROL14	14	/**< 14 bit controller value */
+#define SNDRV_SEQ_EVENT_NONREGPARAM	15	/**< 14 bit NRPN address + 14 bit unsigned value */
+#define SNDRV_SEQ_EVENT_REGPARAM	16	/**< 14 bit RPN address + 14 bit unsigned value */
+
+/** synchronisation messages
+ * event data type = #snd_seq_ev_ctrl
+ */
+#define SNDRV_SEQ_EVENT_SONGPOS		20	/* Song Position Pointer with LSB and MSB values */
+#define SNDRV_SEQ_EVENT_SONGSEL		21	/* Song Select with song ID number */
+#define SNDRV_SEQ_EVENT_QFRAME		22	/* midi time code quarter frame */
+#define SNDRV_SEQ_EVENT_TIMESIGN	23	/* SMF Time Signature event */
+#define SNDRV_SEQ_EVENT_KEYSIGN		24	/* SMF Key Signature event */
+	        
+/** timer messages
+ * event data type = snd_seq_ev_queue_control
+ */
+#define SNDRV_SEQ_EVENT_START		30	/* midi Real Time Start message */
+#define SNDRV_SEQ_EVENT_CONTINUE	31	/* midi Real Time Continue message */
+#define SNDRV_SEQ_EVENT_STOP		32	/* midi Real Time Stop message */	
+#define	SNDRV_SEQ_EVENT_SETPOS_TICK	33	/* set tick queue position */
+#define SNDRV_SEQ_EVENT_SETPOS_TIME	34	/* set realtime queue position */
+#define SNDRV_SEQ_EVENT_TEMPO		35	/* (SMF) Tempo event */
+#define SNDRV_SEQ_EVENT_CLOCK		36	/* midi Real Time Clock message */
+#define SNDRV_SEQ_EVENT_TICK		37	/* midi Real Time Tick message */
+#define SNDRV_SEQ_EVENT_QUEUE_SKEW	38	/* skew queue tempo */
+
+/** others
+ * event data type = none
+ */
+#define SNDRV_SEQ_EVENT_TUNE_REQUEST	40	/* tune request */
+#define SNDRV_SEQ_EVENT_RESET		41	/* reset to power-on state */
+#define SNDRV_SEQ_EVENT_SENSING		42	/* "active sensing" event */
+
+/** echo back, kernel private messages
+ * event data type = any type
+ */
+#define SNDRV_SEQ_EVENT_ECHO		50	/* echo event */
+#define SNDRV_SEQ_EVENT_OSS		51	/* OSS raw event */
+
+/** system status messages (broadcast for subscribers)
+ * event data type = snd_seq_addr
+ */
+#define SNDRV_SEQ_EVENT_CLIENT_START	60	/* new client has connected */
+#define SNDRV_SEQ_EVENT_CLIENT_EXIT	61	/* client has left the system */
+#define SNDRV_SEQ_EVENT_CLIENT_CHANGE	62	/* client status/info has changed */
+#define SNDRV_SEQ_EVENT_PORT_START	63	/* new port was created */
+#define SNDRV_SEQ_EVENT_PORT_EXIT	64	/* port was deleted from system */
+#define SNDRV_SEQ_EVENT_PORT_CHANGE	65	/* port status/info has changed */
+
+/** port connection changes
+ * event data type = snd_seq_connect
+ */
+#define SNDRV_SEQ_EVENT_PORT_SUBSCRIBED	66	/* ports connected */
+#define SNDRV_SEQ_EVENT_PORT_UNSUBSCRIBED 67	/* ports disconnected */
+
+/* 70-89:  synthesizer events - obsoleted */
+
+/** user-defined events with fixed length
+ * event data type = any
+ */
+#define SNDRV_SEQ_EVENT_USR0		90
+#define SNDRV_SEQ_EVENT_USR1		91
+#define SNDRV_SEQ_EVENT_USR2		92
+#define SNDRV_SEQ_EVENT_USR3		93
+#define SNDRV_SEQ_EVENT_USR4		94
+#define SNDRV_SEQ_EVENT_USR5		95
+#define SNDRV_SEQ_EVENT_USR6		96
+#define SNDRV_SEQ_EVENT_USR7		97
+#define SNDRV_SEQ_EVENT_USR8		98
+#define SNDRV_SEQ_EVENT_USR9		99
+
+/* 100-118: instrument layer - obsoleted */
+/* 119-129: reserved */
+
+/* 130-139: variable length events
+ * event data type = snd_seq_ev_ext
+ * (SNDRV_SEQ_EVENT_LENGTH_VARIABLE must be set)
+ */
+#define SNDRV_SEQ_EVENT_SYSEX		130	/* system exclusive data (variable length) */
+#define SNDRV_SEQ_EVENT_BOUNCE		131	/* error event */
+/* 132-134: reserved */
+#define SNDRV_SEQ_EVENT_USR_VAR0	135
+#define SNDRV_SEQ_EVENT_USR_VAR1	136
+#define SNDRV_SEQ_EVENT_USR_VAR2	137
+#define SNDRV_SEQ_EVENT_USR_VAR3	138
+#define SNDRV_SEQ_EVENT_USR_VAR4	139
+
+/* 150-151: kernel events with quote - DO NOT use in user clients */
+#define SNDRV_SEQ_EVENT_KERNEL_ERROR	150
+#define SNDRV_SEQ_EVENT_KERNEL_QUOTE	151	/* obsolete */
+
+/* 152-191: reserved */
+
+/* 192-254: hardware specific events */
+
+/* 255: special event */
+#define SNDRV_SEQ_EVENT_NONE		255
+
+
+typedef unsigned char snd_seq_event_type_t;
+
+/** event address */
+struct snd_seq_addr {
+	unsigned char client;	/**< Client number:         0..255, 255 = broadcast to all clients */
+	unsigned char port;	/**< Port within client:    0..255, 255 = broadcast to all ports */
+};
+
+/** port connection */
+struct snd_seq_connect {
+	struct snd_seq_addr sender;
+	struct snd_seq_addr dest;
+};
+
+
+#define SNDRV_SEQ_ADDRESS_UNKNOWN	253	/* unknown source */
+#define SNDRV_SEQ_ADDRESS_SUBSCRIBERS	254	/* send event to all subscribed ports */
+#define SNDRV_SEQ_ADDRESS_BROADCAST	255	/* send event to all queues/clients/ports/channels */
+#define SNDRV_SEQ_QUEUE_DIRECT		253	/* direct dispatch */
+
+	/* event mode flag - NOTE: only 8 bits available! */
+#define SNDRV_SEQ_TIME_STAMP_TICK	(0<<0) /* timestamp in clock ticks */
+#define SNDRV_SEQ_TIME_STAMP_REAL	(1<<0) /* timestamp in real time */
+#define SNDRV_SEQ_TIME_STAMP_MASK	(1<<0)
+
+#define SNDRV_SEQ_TIME_MODE_ABS		(0<<1)	/* absolute timestamp */
+#define SNDRV_SEQ_TIME_MODE_REL		(1<<1)	/* relative to current time */
+#define SNDRV_SEQ_TIME_MODE_MASK	(1<<1)
+
+#define SNDRV_SEQ_EVENT_LENGTH_FIXED	(0<<2)	/* fixed event size */
+#define SNDRV_SEQ_EVENT_LENGTH_VARIABLE	(1<<2)	/* variable event size */
+#define SNDRV_SEQ_EVENT_LENGTH_VARUSR	(2<<2)	/* variable event size - user memory space */
+#define SNDRV_SEQ_EVENT_LENGTH_MASK	(3<<2)
+
+#define SNDRV_SEQ_PRIORITY_NORMAL	(0<<4)	/* normal priority */
+#define SNDRV_SEQ_PRIORITY_HIGH		(1<<4)	/* event should be processed before others */
+#define SNDRV_SEQ_PRIORITY_MASK		(1<<4)
+
+
+	/* note event */
+struct snd_seq_ev_note {
+	unsigned char channel;
+	unsigned char note;
+	unsigned char velocity;
+	unsigned char off_velocity;	/* only for SNDRV_SEQ_EVENT_NOTE */
+	unsigned int duration;		/* only for SNDRV_SEQ_EVENT_NOTE */
+};
+
+	/* controller event */
+struct snd_seq_ev_ctrl {
+	unsigned char channel;
+	unsigned char unused1, unused2, unused3;	/* pad */
+	unsigned int param;
+	signed int value;
+};
+
+	/* generic set of bytes (12x8 bit) */
+struct snd_seq_ev_raw8 {
+	unsigned char d[12];	/* 8 bit value */
+};
+
+	/* generic set of integers (3x32 bit) */
+struct snd_seq_ev_raw32 {
+	unsigned int d[3];	/* 32 bit value */
+};
+
+	/* external stored data */
+struct snd_seq_ev_ext {
+	unsigned int len;	/* length of data */
+	void *ptr;		/* pointer to data (note: maybe 64-bit) */
+} __attribute__((packed));
+
+struct snd_seq_result {
+	int event;		/* processed event type */
+	int result;
+};
+
+
+struct snd_seq_real_time {
+	unsigned int tv_sec;	/* seconds */
+	unsigned int tv_nsec;	/* nanoseconds */
+};
+
+typedef unsigned int snd_seq_tick_time_t;	/* midi ticks */
+
+union snd_seq_timestamp {
+	snd_seq_tick_time_t tick;
+	struct snd_seq_real_time time;
+};
+
+struct snd_seq_queue_skew {
+	unsigned int value;
+	unsigned int base;
+};
+
+	/* queue timer control */
+struct snd_seq_ev_queue_control {
+	unsigned char queue;			/* affected queue */
+	unsigned char pad[3];			/* reserved */
+	union {
+		signed int value;		/* affected value (e.g. tempo) */
+		union snd_seq_timestamp time;	/* time */
+		unsigned int position;		/* sync position */
+		struct snd_seq_queue_skew skew;
+		unsigned int d32[2];
+		unsigned char d8[8];
+	} param;
+};
+
+	/* quoted event - inside the kernel only */
+struct snd_seq_ev_quote {
+	struct snd_seq_addr origin;		/* original sender */
+	unsigned short value;		/* optional data */
+	struct snd_seq_event *event;		/* quoted event */
+} __attribute__((packed));
+
+
+	/* sequencer event */
+struct snd_seq_event {
+	snd_seq_event_type_t type;	/* event type */
+	unsigned char flags;		/* event flags */
+	char tag;
+	
+	unsigned char queue;		/* schedule queue */
+	union snd_seq_timestamp time;	/* schedule time */
+
+
+	struct snd_seq_addr source;	/* source address */
+	struct snd_seq_addr dest;	/* destination address */
+
+	union {				/* event data... */
+		struct snd_seq_ev_note note;
+		struct snd_seq_ev_ctrl control;
+		struct snd_seq_ev_raw8 raw8;
+		struct snd_seq_ev_raw32 raw32;
+		struct snd_seq_ev_ext ext;
+		struct snd_seq_ev_queue_control queue;
+		union snd_seq_timestamp time;
+		struct snd_seq_addr addr;
+		struct snd_seq_connect connect;
+		struct snd_seq_result result;
+		struct snd_seq_ev_quote quote;
+	} data;
+};
+
+
+/*
+ * bounce event - stored as variable size data
+ */
+struct snd_seq_event_bounce {
+	int err;
+	struct snd_seq_event event;
+	/* external data follows here. */
+};
+
+#ifdef __KERNEL__
+
+/* helper macro */
+#define snd_seq_event_bounce_ext_data(ev) ((void*)((char *)(ev)->data.ext.ptr + sizeof(struct snd_seq_event_bounce)))
+
+/*
+ * type check macros
+ */
+/* result events: 0-4 */
+#define snd_seq_ev_is_result_type(ev)	((ev)->type < 5)
+/* channel specific events: 5-19 */
+#define snd_seq_ev_is_channel_type(ev)	((ev)->type >= 5 && (ev)->type < 20)
+/* note events: 5-9 */
+#define snd_seq_ev_is_note_type(ev)	((ev)->type >= 5 && (ev)->type < 10)
+/* control events: 10-19 */
+#define snd_seq_ev_is_control_type(ev)	((ev)->type >= 10 && (ev)->type < 20)
+/* queue control events: 30-39 */
+#define snd_seq_ev_is_queue_type(ev)	((ev)->type >= 30 && (ev)->type < 40)
+/* system status messages */
+#define snd_seq_ev_is_message_type(ev)	((ev)->type >= 60 && (ev)->type < 69)
+/* sample messages */
+#define snd_seq_ev_is_sample_type(ev)	((ev)->type >= 70 && (ev)->type < 79)
+/* user-defined messages */
+#define snd_seq_ev_is_user_type(ev)	((ev)->type >= 90 && (ev)->type < 99)
+/* fixed length events: 0-99 */
+#define snd_seq_ev_is_fixed_type(ev)	((ev)->type < 100)
+/* variable length events: 130-139 */
+#define snd_seq_ev_is_variable_type(ev)	((ev)->type >= 130 && (ev)->type < 140)
+/* reserved for kernel */
+#define snd_seq_ev_is_reserved(ev)	((ev)->type >= 150)
+
+/* direct dispatched events */
+#define snd_seq_ev_is_direct(ev)	((ev)->queue == SNDRV_SEQ_QUEUE_DIRECT)
+
+/*
+ * macros to check event flags
+ */
+/* prior events */
+#define snd_seq_ev_is_prior(ev)		(((ev)->flags & SNDRV_SEQ_PRIORITY_MASK) == SNDRV_SEQ_PRIORITY_HIGH)
+
+/* event length type */
+#define snd_seq_ev_length_type(ev)	((ev)->flags & SNDRV_SEQ_EVENT_LENGTH_MASK)
+#define snd_seq_ev_is_fixed(ev)		(snd_seq_ev_length_type(ev) == SNDRV_SEQ_EVENT_LENGTH_FIXED)
+#define snd_seq_ev_is_variable(ev)	(snd_seq_ev_length_type(ev) == SNDRV_SEQ_EVENT_LENGTH_VARIABLE)
+#define snd_seq_ev_is_varusr(ev)	(snd_seq_ev_length_type(ev) == SNDRV_SEQ_EVENT_LENGTH_VARUSR)
+
+/* time-stamp type */
+#define snd_seq_ev_timestamp_type(ev)	((ev)->flags & SNDRV_SEQ_TIME_STAMP_MASK)
+#define snd_seq_ev_is_tick(ev)		(snd_seq_ev_timestamp_type(ev) == SNDRV_SEQ_TIME_STAMP_TICK)
+#define snd_seq_ev_is_real(ev)		(snd_seq_ev_timestamp_type(ev) == SNDRV_SEQ_TIME_STAMP_REAL)
+
+/* time-mode type */
+#define snd_seq_ev_timemode_type(ev)	((ev)->flags & SNDRV_SEQ_TIME_MODE_MASK)
+#define snd_seq_ev_is_abstime(ev)	(snd_seq_ev_timemode_type(ev) == SNDRV_SEQ_TIME_MODE_ABS)
+#define snd_seq_ev_is_reltime(ev)	(snd_seq_ev_timemode_type(ev) == SNDRV_SEQ_TIME_MODE_REL)
+
+/* queue sync port */
+#define snd_seq_queue_sync_port(q)	((q) + 16)
+
+#endif /* __KERNEL__ */
+
+	/* system information */
+struct snd_seq_system_info {
+	int queues;			/* maximum queues count */
+	int clients;			/* maximum clients count */
+	int ports;			/* maximum ports per client */
+	int channels;			/* maximum channels per port */
+	int cur_clients;		/* current clients */
+	int cur_queues;			/* current queues */
+	char reserved[24];
+};
+
+
+	/* system running information */
+struct snd_seq_running_info {
+	unsigned char client;		/* client id */
+	unsigned char big_endian;	/* 1 = big-endian */
+	unsigned char cpu_mode;		/* 4 = 32bit, 8 = 64bit */
+	unsigned char pad;		/* reserved */
+	unsigned char reserved[12];
+};
+
+
+	/* known client numbers */
+#define SNDRV_SEQ_CLIENT_SYSTEM		0
+	/* internal client numbers */
+#define SNDRV_SEQ_CLIENT_DUMMY		14	/* midi through */
+#define SNDRV_SEQ_CLIENT_OSS		15	/* oss sequencer emulator */
+
+
+	/* client types */
+typedef int __bitwise snd_seq_client_type_t;
+#define	NO_CLIENT	((__force snd_seq_client_type_t) 0)
+#define	USER_CLIENT	((__force snd_seq_client_type_t) 1)
+#define	KERNEL_CLIENT	((__force snd_seq_client_type_t) 2)
+                        
+	/* event filter flags */
+#define SNDRV_SEQ_FILTER_BROADCAST	(1<<0)	/* accept broadcast messages */
+#define SNDRV_SEQ_FILTER_MULTICAST	(1<<1)	/* accept multicast messages */
+#define SNDRV_SEQ_FILTER_BOUNCE		(1<<2)	/* accept bounce event in error */
+#define SNDRV_SEQ_FILTER_USE_EVENT	(1<<31)	/* use event filter */
+
+struct snd_seq_client_info {
+	int client;			/* client number to inquire */
+	snd_seq_client_type_t type;	/* client type */
+	char name[64];			/* client name */
+	unsigned int filter;		/* filter flags */
+	unsigned char multicast_filter[8]; /* multicast filter bitmap */
+	unsigned char event_filter[32];	/* event filter bitmap */
+	int num_ports;			/* RO: number of ports */
+	int event_lost;			/* number of lost events */
+	char reserved[64];		/* for future use */
+};
+
+
+/* client pool size */
+struct snd_seq_client_pool {
+	int client;			/* client number to inquire */
+	int output_pool;		/* outgoing (write) pool size */
+	int input_pool;			/* incoming (read) pool size */
+	int output_room;		/* minimum free pool size for select/blocking mode */
+	int output_free;		/* unused size */
+	int input_free;			/* unused size */
+	char reserved[64];
+};
+
+
+/* Remove events by specified criteria */
+
+#define SNDRV_SEQ_REMOVE_INPUT		(1<<0)	/* Flush input queues */
+#define SNDRV_SEQ_REMOVE_OUTPUT		(1<<1)	/* Flush output queues */
+#define SNDRV_SEQ_REMOVE_DEST		(1<<2)	/* Restrict by destination q:client:port */
+#define SNDRV_SEQ_REMOVE_DEST_CHANNEL	(1<<3)	/* Restrict by channel */
+#define SNDRV_SEQ_REMOVE_TIME_BEFORE	(1<<4)	/* Restrict to before time */
+#define SNDRV_SEQ_REMOVE_TIME_AFTER	(1<<5)	/* Restrict to time or after */
+#define SNDRV_SEQ_REMOVE_TIME_TICK	(1<<6)	/* Time is in ticks */
+#define SNDRV_SEQ_REMOVE_EVENT_TYPE	(1<<7)	/* Restrict to event type */
+#define SNDRV_SEQ_REMOVE_IGNORE_OFF 	(1<<8)	/* Do not flush off events */
+#define SNDRV_SEQ_REMOVE_TAG_MATCH 	(1<<9)	/* Restrict to events with given tag */
+
+struct snd_seq_remove_events {
+	unsigned int  remove_mode;	/* Flags that determine what gets removed */
+
+	union snd_seq_timestamp time;
+
+	unsigned char queue;	/* Queue for REMOVE_DEST */
+	struct snd_seq_addr dest;	/* Address for REMOVE_DEST */
+	unsigned char channel;	/* Channel for REMOVE_DEST */
+
+	int  type;	/* For REMOVE_EVENT_TYPE */
+	char  tag;	/* Tag for REMOVE_TAG */
+
+	int  reserved[10];	/* To allow for future binary compatibility */
+
+};
+
+
+	/* known port numbers */
+#define SNDRV_SEQ_PORT_SYSTEM_TIMER	0
+#define SNDRV_SEQ_PORT_SYSTEM_ANNOUNCE	1
+
+	/* port capabilities (32 bits) */
+#define SNDRV_SEQ_PORT_CAP_READ		(1<<0)	/* readable from this port */
+#define SNDRV_SEQ_PORT_CAP_WRITE	(1<<1)	/* writable to this port */
+
+#define SNDRV_SEQ_PORT_CAP_SYNC_READ	(1<<2)
+#define SNDRV_SEQ_PORT_CAP_SYNC_WRITE	(1<<3)
+
+#define SNDRV_SEQ_PORT_CAP_DUPLEX	(1<<4)
+
+#define SNDRV_SEQ_PORT_CAP_SUBS_READ	(1<<5)	/* allow read subscription */
+#define SNDRV_SEQ_PORT_CAP_SUBS_WRITE	(1<<6)	/* allow write subscription */
+#define SNDRV_SEQ_PORT_CAP_NO_EXPORT	(1<<7)	/* routing not allowed */
+
+	/* port type */
+#define SNDRV_SEQ_PORT_TYPE_SPECIFIC	(1<<0)	/* hardware specific */
+#define SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC (1<<1)	/* generic MIDI device */
+#define SNDRV_SEQ_PORT_TYPE_MIDI_GM	(1<<2)	/* General MIDI compatible device */
+#define SNDRV_SEQ_PORT_TYPE_MIDI_GS	(1<<3)	/* GS compatible device */
+#define SNDRV_SEQ_PORT_TYPE_MIDI_XG	(1<<4)	/* XG compatible device */
+#define SNDRV_SEQ_PORT_TYPE_MIDI_MT32	(1<<5)	/* MT-32 compatible device */
+#define SNDRV_SEQ_PORT_TYPE_MIDI_GM2	(1<<6)	/* General MIDI 2 compatible device */
+
+/* other standards...*/
+#define SNDRV_SEQ_PORT_TYPE_SYNTH	(1<<10)	/* Synth device (no MIDI compatible - direct wavetable) */
+#define SNDRV_SEQ_PORT_TYPE_DIRECT_SAMPLE (1<<11)	/* Sampling device (support sample download) */
+#define SNDRV_SEQ_PORT_TYPE_SAMPLE	(1<<12)	/* Sampling device (sample can be downloaded at any time) */
+/*...*/
+#define SNDRV_SEQ_PORT_TYPE_HARDWARE	(1<<16)	/* driver for a hardware device */
+#define SNDRV_SEQ_PORT_TYPE_SOFTWARE	(1<<17)	/* implemented in software */
+#define SNDRV_SEQ_PORT_TYPE_SYNTHESIZER	(1<<18)	/* generates sound */
+#define SNDRV_SEQ_PORT_TYPE_PORT	(1<<19)	/* connects to other device(s) */
+#define SNDRV_SEQ_PORT_TYPE_APPLICATION	(1<<20)	/* application (sequencer/editor) */
+
+/* misc. conditioning flags */
+#define SNDRV_SEQ_PORT_FLG_GIVEN_PORT	(1<<0)
+#define SNDRV_SEQ_PORT_FLG_TIMESTAMP	(1<<1)
+#define SNDRV_SEQ_PORT_FLG_TIME_REAL	(1<<2)
+
+struct snd_seq_port_info {
+	struct snd_seq_addr addr;	/* client/port numbers */
+	char name[64];			/* port name */
+
+	unsigned int capability;	/* port capability bits */
+	unsigned int type;		/* port type bits */
+	int midi_channels;		/* channels per MIDI port */
+	int midi_voices;		/* voices per MIDI port */
+	int synth_voices;		/* voices per SYNTH port */
+
+	int read_use;			/* R/O: subscribers for output (from this port) */
+	int write_use;			/* R/O: subscribers for input (to this port) */
+
+	void *kernel;			/* reserved for kernel use (must be NULL) */
+	unsigned int flags;		/* misc. conditioning */
+	unsigned char time_queue;	/* queue # for timestamping */
+	char reserved[59];		/* for future use */
+};
+
+
+/* queue flags */
+#define SNDRV_SEQ_QUEUE_FLG_SYNC	(1<<0)	/* sync enabled */
+
+/* queue information */
+struct snd_seq_queue_info {
+	int queue;		/* queue id */
+
+	/*
+	 *  security settings, only owner of this queue can start/stop timer
+	 *  etc. if the queue is locked for other clients
+	 */
+	int owner;		/* client id for owner of the queue */
+	unsigned locked:1;	/* timing queue locked for other queues */
+	char name[64];		/* name of this queue */
+	unsigned int flags;	/* flags */
+	char reserved[60];	/* for future use */
+
+};
+
+/* queue info/status */
+struct snd_seq_queue_status {
+	int queue;			/* queue id */
+	int events;			/* read-only - queue size */
+	snd_seq_tick_time_t tick;	/* current tick */
+	struct snd_seq_real_time time;	/* current time */
+	int running;			/* running state of queue */
+	int flags;			/* various flags */
+	char reserved[64];		/* for the future */
+};
+
+
+/* queue tempo */
+struct snd_seq_queue_tempo {
+	int queue;			/* sequencer queue */
+	unsigned int tempo;		/* current tempo, us/tick */
+	int ppq;			/* time resolution, ticks/quarter */
+	unsigned int skew_value;	/* queue skew */
+	unsigned int skew_base;		/* queue skew base */
+	char reserved[24];		/* for the future */
+};
+
+
+/* sequencer timer sources */
+#define SNDRV_SEQ_TIMER_ALSA		0	/* ALSA timer */
+#define SNDRV_SEQ_TIMER_MIDI_CLOCK	1	/* Midi Clock (CLOCK event) */
+#define SNDRV_SEQ_TIMER_MIDI_TICK	2	/* Midi Timer Tick (TICK event) */
+
+/* queue timer info */
+struct snd_seq_queue_timer {
+	int queue;			/* sequencer queue */
+	int type;			/* source timer type */
+	union {
+		struct {
+			struct snd_timer_id id;	/* ALSA's timer ID */
+			unsigned int resolution;	/* resolution in Hz */
+		} alsa;
+	} u;
+	char reserved[64];		/* for the future use */
+};
+
+
+struct snd_seq_queue_client {
+	int queue;		/* sequencer queue */
+	int client;		/* sequencer client */
+	int used;		/* queue is used with this client
+				   (must be set for accepting events) */
+	/* per client watermarks */
+	char reserved[64];	/* for future use */
+};
+
+
+#define SNDRV_SEQ_PORT_SUBS_EXCLUSIVE	(1<<0)	/* exclusive connection */
+#define SNDRV_SEQ_PORT_SUBS_TIMESTAMP	(1<<1)
+#define SNDRV_SEQ_PORT_SUBS_TIME_REAL	(1<<2)
+
+struct snd_seq_port_subscribe {
+	struct snd_seq_addr sender;	/* sender address */
+	struct snd_seq_addr dest;	/* destination address */
+	unsigned int voices;		/* number of voices to be allocated (0 = don't care) */
+	unsigned int flags;		/* modes */
+	unsigned char queue;		/* input time-stamp queue (optional) */
+	unsigned char pad[3];		/* reserved */
+	char reserved[64];
+};
+
+/* type of query subscription */
+#define SNDRV_SEQ_QUERY_SUBS_READ	0
+#define SNDRV_SEQ_QUERY_SUBS_WRITE	1
+
+struct snd_seq_query_subs {
+	struct snd_seq_addr root;	/* client/port id to be searched */
+	int type;		/* READ or WRITE */
+	int index;		/* 0..N-1 */
+	int num_subs;		/* R/O: number of subscriptions on this port */
+	struct snd_seq_addr addr;	/* R/O: result */
+	unsigned char queue;	/* R/O: result */
+	unsigned int flags;	/* R/O: result */
+	char reserved[64];	/* for future use */
+};
+
+
+/*
+ *  IOCTL commands
+ */
+
+#define SNDRV_SEQ_IOCTL_PVERSION	_IOR ('S', 0x00, int)
+#define SNDRV_SEQ_IOCTL_CLIENT_ID	_IOR ('S', 0x01, int)
+#define SNDRV_SEQ_IOCTL_SYSTEM_INFO	_IOWR('S', 0x02, struct snd_seq_system_info)
+#define SNDRV_SEQ_IOCTL_RUNNING_MODE	_IOWR('S', 0x03, struct snd_seq_running_info)
+
+#define SNDRV_SEQ_IOCTL_GET_CLIENT_INFO	_IOWR('S', 0x10, struct snd_seq_client_info)
+#define SNDRV_SEQ_IOCTL_SET_CLIENT_INFO	_IOW ('S', 0x11, struct snd_seq_client_info)
+
+#define SNDRV_SEQ_IOCTL_CREATE_PORT	_IOWR('S', 0x20, struct snd_seq_port_info)
+#define SNDRV_SEQ_IOCTL_DELETE_PORT	_IOW ('S', 0x21, struct snd_seq_port_info)
+#define SNDRV_SEQ_IOCTL_GET_PORT_INFO	_IOWR('S', 0x22, struct snd_seq_port_info)
+#define SNDRV_SEQ_IOCTL_SET_PORT_INFO	_IOW ('S', 0x23, struct snd_seq_port_info)
+
+#define SNDRV_SEQ_IOCTL_SUBSCRIBE_PORT	_IOW ('S', 0x30, struct snd_seq_port_subscribe)
+#define SNDRV_SEQ_IOCTL_UNSUBSCRIBE_PORT _IOW ('S', 0x31, struct snd_seq_port_subscribe)
+
+#define SNDRV_SEQ_IOCTL_CREATE_QUEUE	_IOWR('S', 0x32, struct snd_seq_queue_info)
+#define SNDRV_SEQ_IOCTL_DELETE_QUEUE	_IOW ('S', 0x33, struct snd_seq_queue_info)
+#define SNDRV_SEQ_IOCTL_GET_QUEUE_INFO	_IOWR('S', 0x34, struct snd_seq_queue_info)
+#define SNDRV_SEQ_IOCTL_SET_QUEUE_INFO	_IOWR('S', 0x35, struct snd_seq_queue_info)
+#define SNDRV_SEQ_IOCTL_GET_NAMED_QUEUE	_IOWR('S', 0x36, struct snd_seq_queue_info)
+#define SNDRV_SEQ_IOCTL_GET_QUEUE_STATUS _IOWR('S', 0x40, struct snd_seq_queue_status)
+#define SNDRV_SEQ_IOCTL_GET_QUEUE_TEMPO	_IOWR('S', 0x41, struct snd_seq_queue_tempo)
+#define SNDRV_SEQ_IOCTL_SET_QUEUE_TEMPO	_IOW ('S', 0x42, struct snd_seq_queue_tempo)
+#define SNDRV_SEQ_IOCTL_GET_QUEUE_OWNER	_IOWR('S', 0x43, struct snd_seq_queue_owner)
+#define SNDRV_SEQ_IOCTL_SET_QUEUE_OWNER	_IOW ('S', 0x44, struct snd_seq_queue_owner)
+#define SNDRV_SEQ_IOCTL_GET_QUEUE_TIMER	_IOWR('S', 0x45, struct snd_seq_queue_timer)
+#define SNDRV_SEQ_IOCTL_SET_QUEUE_TIMER	_IOW ('S', 0x46, struct snd_seq_queue_timer)
+/* XXX
+#define SNDRV_SEQ_IOCTL_GET_QUEUE_SYNC	_IOWR('S', 0x53, struct snd_seq_queue_sync)
+#define SNDRV_SEQ_IOCTL_SET_QUEUE_SYNC	_IOW ('S', 0x54, struct snd_seq_queue_sync)
+*/
+#define SNDRV_SEQ_IOCTL_GET_QUEUE_CLIENT	_IOWR('S', 0x49, struct snd_seq_queue_client)
+#define SNDRV_SEQ_IOCTL_SET_QUEUE_CLIENT	_IOW ('S', 0x4a, struct snd_seq_queue_client)
+#define SNDRV_SEQ_IOCTL_GET_CLIENT_POOL	_IOWR('S', 0x4b, struct snd_seq_client_pool)
+#define SNDRV_SEQ_IOCTL_SET_CLIENT_POOL	_IOW ('S', 0x4c, struct snd_seq_client_pool)
+#define SNDRV_SEQ_IOCTL_REMOVE_EVENTS	_IOW ('S', 0x4e, struct snd_seq_remove_events)
+#define SNDRV_SEQ_IOCTL_QUERY_SUBS	_IOWR('S', 0x4f, struct snd_seq_query_subs)
+#define SNDRV_SEQ_IOCTL_GET_SUBSCRIPTION	_IOWR('S', 0x50, struct snd_seq_port_subscribe)
+#define SNDRV_SEQ_IOCTL_QUERY_NEXT_CLIENT	_IOWR('S', 0x51, struct snd_seq_client_info)
+#define SNDRV_SEQ_IOCTL_QUERY_NEXT_PORT	_IOWR('S', 0x52, struct snd_seq_port_info)
+
+#endif /* __SOUND_ASEQUENCER_H */
diff --git a/include/sound/asound.h b/include/sound/asound.h
new file mode 100644
index 0000000..0985955
--- /dev/null
+++ b/include/sound/asound.h
@@ -0,0 +1,911 @@
+/*
+ *  Advanced Linux Sound Architecture - ALSA - Driver
+ *  Copyright (c) 1994-2003 by Jaroslav Kysela <perex@perex.cz>,
+ *                             Abramo Bagnara <abramo@alsa-project.org>
+ *
+ *
+ *   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., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ */
+
+#ifndef __SOUND_ASOUND_H
+#define __SOUND_ASOUND_H
+
+#include <linux/types.h>
+
+#ifdef __KERNEL__
+#include <linux/ioctl.h>
+#include <linux/time.h>
+#include <asm/byteorder.h>
+
+#ifdef  __LITTLE_ENDIAN
+#define SNDRV_LITTLE_ENDIAN
+#else
+#ifdef __BIG_ENDIAN
+#define SNDRV_BIG_ENDIAN
+#else
+#error "Unsupported endian..."
+#endif
+#endif
+
+#endif /* __KERNEL__ **/
+
+/*
+ *  protocol version
+ */
+
+#define SNDRV_PROTOCOL_VERSION(major, minor, subminor) (((major)<<16)|((minor)<<8)|(subminor))
+#define SNDRV_PROTOCOL_MAJOR(version) (((version)>>16)&0xffff)
+#define SNDRV_PROTOCOL_MINOR(version) (((version)>>8)&0xff)
+#define SNDRV_PROTOCOL_MICRO(version) ((version)&0xff)
+#define SNDRV_PROTOCOL_INCOMPATIBLE(kversion, uversion) \
+	(SNDRV_PROTOCOL_MAJOR(kversion) != SNDRV_PROTOCOL_MAJOR(uversion) || \
+	 (SNDRV_PROTOCOL_MAJOR(kversion) == SNDRV_PROTOCOL_MAJOR(uversion) && \
+	   SNDRV_PROTOCOL_MINOR(kversion) != SNDRV_PROTOCOL_MINOR(uversion)))
+
+/****************************************************************************
+ *                                                                          *
+ *        Digital audio interface					    *
+ *                                                                          *
+ ****************************************************************************/
+
+struct snd_aes_iec958 {
+	unsigned char status[24];	/* AES/IEC958 channel status bits */
+	unsigned char subcode[147];	/* AES/IEC958 subcode bits */
+	unsigned char pad;		/* nothing */
+	unsigned char dig_subframe[4];	/* AES/IEC958 subframe bits */
+};
+
+/****************************************************************************
+ *                                                                          *
+ *      Section for driver hardware dependent interface - /dev/snd/hw?      *
+ *                                                                          *
+ ****************************************************************************/
+
+#define SNDRV_HWDEP_VERSION		SNDRV_PROTOCOL_VERSION(1, 0, 1)
+
+enum {
+	SNDRV_HWDEP_IFACE_OPL2 = 0,
+	SNDRV_HWDEP_IFACE_OPL3,
+	SNDRV_HWDEP_IFACE_OPL4,
+	SNDRV_HWDEP_IFACE_SB16CSP,	/* Creative Signal Processor */
+	SNDRV_HWDEP_IFACE_EMU10K1,	/* FX8010 processor in EMU10K1 chip */
+	SNDRV_HWDEP_IFACE_YSS225,	/* Yamaha FX processor */
+	SNDRV_HWDEP_IFACE_ICS2115,	/* Wavetable synth */
+	SNDRV_HWDEP_IFACE_SSCAPE,	/* Ensoniq SoundScape ISA card (MC68EC000) */
+	SNDRV_HWDEP_IFACE_VX,		/* Digigram VX cards */
+	SNDRV_HWDEP_IFACE_MIXART,	/* Digigram miXart cards */
+	SNDRV_HWDEP_IFACE_USX2Y,	/* Tascam US122, US224 & US428 usb */
+	SNDRV_HWDEP_IFACE_EMUX_WAVETABLE, /* EmuX wavetable */	
+	SNDRV_HWDEP_IFACE_BLUETOOTH,	/* Bluetooth audio */
+	SNDRV_HWDEP_IFACE_USX2Y_PCM,	/* Tascam US122, US224 & US428 rawusb pcm */
+	SNDRV_HWDEP_IFACE_PCXHR,	/* Digigram PCXHR */
+	SNDRV_HWDEP_IFACE_SB_RC,	/* SB Extigy/Audigy2NX remote control */
+	SNDRV_HWDEP_IFACE_HDA,		/* HD-audio */
+	SNDRV_HWDEP_IFACE_USB_STREAM,	/* direct access to usb stream */
+
+	/* Don't forget to change the following: */
+	SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_USB_STREAM
+};
+
+struct snd_hwdep_info {
+	unsigned int device;		/* WR: device number */
+	int card;			/* R: card number */
+	unsigned char id[64];		/* ID (user selectable) */
+	unsigned char name[80];		/* hwdep name */
+	int iface;			/* hwdep interface */
+	unsigned char reserved[64];	/* reserved for future */
+};
+
+/* generic DSP loader */
+struct snd_hwdep_dsp_status {
+	unsigned int version;		/* R: driver-specific version */
+	unsigned char id[32];		/* R: driver-specific ID string */
+	unsigned int num_dsps;		/* R: number of DSP images to transfer */
+	unsigned int dsp_loaded;	/* R: bit flags indicating the loaded DSPs */
+	unsigned int chip_ready;	/* R: 1 = initialization finished */
+	unsigned char reserved[16];	/* reserved for future use */
+};
+
+struct snd_hwdep_dsp_image {
+	unsigned int index;		/* W: DSP index */
+	unsigned char name[64];		/* W: ID (e.g. file name) */
+	unsigned char __user *image;	/* W: binary image */
+	size_t length;			/* W: size of image in bytes */
+	unsigned long driver_data;	/* W: driver-specific data */
+};
+
+#define SNDRV_HWDEP_IOCTL_PVERSION	_IOR ('H', 0x00, int)
+#define SNDRV_HWDEP_IOCTL_INFO		_IOR ('H', 0x01, struct snd_hwdep_info)
+#define SNDRV_HWDEP_IOCTL_DSP_STATUS	_IOR('H', 0x02, struct snd_hwdep_dsp_status)
+#define SNDRV_HWDEP_IOCTL_DSP_LOAD	_IOW('H', 0x03, struct snd_hwdep_dsp_image)
+
+/*****************************************************************************
+ *                                                                           *
+ *             Digital Audio (PCM) interface - /dev/snd/pcm??                *
+ *                                                                           *
+ *****************************************************************************/
+
+#define SNDRV_PCM_VERSION		SNDRV_PROTOCOL_VERSION(2, 0, 10)
+
+typedef unsigned long snd_pcm_uframes_t;
+typedef signed long snd_pcm_sframes_t;
+
+enum {
+	SNDRV_PCM_CLASS_GENERIC = 0,	/* standard mono or stereo device */
+	SNDRV_PCM_CLASS_MULTI,		/* multichannel device */
+	SNDRV_PCM_CLASS_MODEM,		/* software modem class */
+	SNDRV_PCM_CLASS_DIGITIZER,	/* digitizer class */
+	/* Don't forget to change the following: */
+	SNDRV_PCM_CLASS_LAST = SNDRV_PCM_CLASS_DIGITIZER,
+};
+
+enum {
+	SNDRV_PCM_SUBCLASS_GENERIC_MIX = 0, /* mono or stereo subdevices are mixed together */
+	SNDRV_PCM_SUBCLASS_MULTI_MIX,	/* multichannel subdevices are mixed together */
+	/* Don't forget to change the following: */
+	SNDRV_PCM_SUBCLASS_LAST = SNDRV_PCM_SUBCLASS_MULTI_MIX,
+};
+
+enum {
+	SNDRV_PCM_STREAM_PLAYBACK = 0,
+	SNDRV_PCM_STREAM_CAPTURE,
+	SNDRV_PCM_STREAM_LAST = SNDRV_PCM_STREAM_CAPTURE,
+};
+
+typedef int __bitwise snd_pcm_access_t;
+#define	SNDRV_PCM_ACCESS_MMAP_INTERLEAVED	((__force snd_pcm_access_t) 0) /* interleaved mmap */
+#define	SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED	((__force snd_pcm_access_t) 1) /* noninterleaved mmap */
+#define	SNDRV_PCM_ACCESS_MMAP_COMPLEX		((__force snd_pcm_access_t) 2) /* complex mmap */
+#define	SNDRV_PCM_ACCESS_RW_INTERLEAVED		((__force snd_pcm_access_t) 3) /* readi/writei */
+#define	SNDRV_PCM_ACCESS_RW_NONINTERLEAVED	((__force snd_pcm_access_t) 4) /* readn/writen */
+#define	SNDRV_PCM_ACCESS_LAST		SNDRV_PCM_ACCESS_RW_NONINTERLEAVED
+
+typedef int __bitwise snd_pcm_format_t;
+#define	SNDRV_PCM_FORMAT_S8	((__force snd_pcm_format_t) 0)
+#define	SNDRV_PCM_FORMAT_U8	((__force snd_pcm_format_t) 1)
+#define	SNDRV_PCM_FORMAT_S16_LE	((__force snd_pcm_format_t) 2)
+#define	SNDRV_PCM_FORMAT_S16_BE	((__force snd_pcm_format_t) 3)
+#define	SNDRV_PCM_FORMAT_U16_LE	((__force snd_pcm_format_t) 4)
+#define	SNDRV_PCM_FORMAT_U16_BE	((__force snd_pcm_format_t) 5)
+#define	SNDRV_PCM_FORMAT_S24_LE	((__force snd_pcm_format_t) 6) /* low three bytes */
+#define	SNDRV_PCM_FORMAT_S24_BE	((__force snd_pcm_format_t) 7) /* low three bytes */
+#define	SNDRV_PCM_FORMAT_U24_LE	((__force snd_pcm_format_t) 8) /* low three bytes */
+#define	SNDRV_PCM_FORMAT_U24_BE	((__force snd_pcm_format_t) 9) /* low three bytes */
+#define	SNDRV_PCM_FORMAT_S32_LE	((__force snd_pcm_format_t) 10)
+#define	SNDRV_PCM_FORMAT_S32_BE	((__force snd_pcm_format_t) 11)
+#define	SNDRV_PCM_FORMAT_U32_LE	((__force snd_pcm_format_t) 12)
+#define	SNDRV_PCM_FORMAT_U32_BE	((__force snd_pcm_format_t) 13)
+#define	SNDRV_PCM_FORMAT_FLOAT_LE	((__force snd_pcm_format_t) 14) /* 4-byte float, IEEE-754 32-bit, range -1.0 to 1.0 */
+#define	SNDRV_PCM_FORMAT_FLOAT_BE	((__force snd_pcm_format_t) 15) /* 4-byte float, IEEE-754 32-bit, range -1.0 to 1.0 */
+#define	SNDRV_PCM_FORMAT_FLOAT64_LE	((__force snd_pcm_format_t) 16) /* 8-byte float, IEEE-754 64-bit, range -1.0 to 1.0 */
+#define	SNDRV_PCM_FORMAT_FLOAT64_BE	((__force snd_pcm_format_t) 17) /* 8-byte float, IEEE-754 64-bit, range -1.0 to 1.0 */
+#define	SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE ((__force snd_pcm_format_t) 18) /* IEC-958 subframe, Little Endian */
+#define	SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE ((__force snd_pcm_format_t) 19) /* IEC-958 subframe, Big Endian */
+#define	SNDRV_PCM_FORMAT_MU_LAW		((__force snd_pcm_format_t) 20)
+#define	SNDRV_PCM_FORMAT_A_LAW		((__force snd_pcm_format_t) 21)
+#define	SNDRV_PCM_FORMAT_IMA_ADPCM	((__force snd_pcm_format_t) 22)
+#define	SNDRV_PCM_FORMAT_MPEG		((__force snd_pcm_format_t) 23)
+#define	SNDRV_PCM_FORMAT_GSM		((__force snd_pcm_format_t) 24)
+#define	SNDRV_PCM_FORMAT_SPECIAL	((__force snd_pcm_format_t) 31)
+#define	SNDRV_PCM_FORMAT_S24_3LE	((__force snd_pcm_format_t) 32)	/* in three bytes */
+#define	SNDRV_PCM_FORMAT_S24_3BE	((__force snd_pcm_format_t) 33)	/* in three bytes */
+#define	SNDRV_PCM_FORMAT_U24_3LE	((__force snd_pcm_format_t) 34)	/* in three bytes */
+#define	SNDRV_PCM_FORMAT_U24_3BE	((__force snd_pcm_format_t) 35)	/* in three bytes */
+#define	SNDRV_PCM_FORMAT_S20_3LE	((__force snd_pcm_format_t) 36)	/* in three bytes */
+#define	SNDRV_PCM_FORMAT_S20_3BE	((__force snd_pcm_format_t) 37)	/* in three bytes */
+#define	SNDRV_PCM_FORMAT_U20_3LE	((__force snd_pcm_format_t) 38)	/* in three bytes */
+#define	SNDRV_PCM_FORMAT_U20_3BE	((__force snd_pcm_format_t) 39)	/* in three bytes */
+#define	SNDRV_PCM_FORMAT_S18_3LE	((__force snd_pcm_format_t) 40)	/* in three bytes */
+#define	SNDRV_PCM_FORMAT_S18_3BE	((__force snd_pcm_format_t) 41)	/* in three bytes */
+#define	SNDRV_PCM_FORMAT_U18_3LE	((__force snd_pcm_format_t) 42)	/* in three bytes */
+#define	SNDRV_PCM_FORMAT_U18_3BE	((__force snd_pcm_format_t) 43)	/* in three bytes */
+#define	SNDRV_PCM_FORMAT_LAST		SNDRV_PCM_FORMAT_U18_3BE
+
+#ifdef SNDRV_LITTLE_ENDIAN
+#define	SNDRV_PCM_FORMAT_S16		SNDRV_PCM_FORMAT_S16_LE
+#define	SNDRV_PCM_FORMAT_U16		SNDRV_PCM_FORMAT_U16_LE
+#define	SNDRV_PCM_FORMAT_S24		SNDRV_PCM_FORMAT_S24_LE
+#define	SNDRV_PCM_FORMAT_U24		SNDRV_PCM_FORMAT_U24_LE
+#define	SNDRV_PCM_FORMAT_S32		SNDRV_PCM_FORMAT_S32_LE
+#define	SNDRV_PCM_FORMAT_U32		SNDRV_PCM_FORMAT_U32_LE
+#define	SNDRV_PCM_FORMAT_FLOAT		SNDRV_PCM_FORMAT_FLOAT_LE
+#define	SNDRV_PCM_FORMAT_FLOAT64	SNDRV_PCM_FORMAT_FLOAT64_LE
+#define	SNDRV_PCM_FORMAT_IEC958_SUBFRAME SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE
+#endif
+#ifdef SNDRV_BIG_ENDIAN
+#define	SNDRV_PCM_FORMAT_S16		SNDRV_PCM_FORMAT_S16_BE
+#define	SNDRV_PCM_FORMAT_U16		SNDRV_PCM_FORMAT_U16_BE
+#define	SNDRV_PCM_FORMAT_S24		SNDRV_PCM_FORMAT_S24_BE
+#define	SNDRV_PCM_FORMAT_U24		SNDRV_PCM_FORMAT_U24_BE
+#define	SNDRV_PCM_FORMAT_S32		SNDRV_PCM_FORMAT_S32_BE
+#define	SNDRV_PCM_FORMAT_U32		SNDRV_PCM_FORMAT_U32_BE
+#define	SNDRV_PCM_FORMAT_FLOAT		SNDRV_PCM_FORMAT_FLOAT_BE
+#define	SNDRV_PCM_FORMAT_FLOAT64	SNDRV_PCM_FORMAT_FLOAT64_BE
+#define	SNDRV_PCM_FORMAT_IEC958_SUBFRAME SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE
+#endif
+
+typedef int __bitwise snd_pcm_subformat_t;
+#define	SNDRV_PCM_SUBFORMAT_STD		((__force snd_pcm_subformat_t) 0)
+#define	SNDRV_PCM_SUBFORMAT_LAST	SNDRV_PCM_SUBFORMAT_STD
+
+#define SNDRV_PCM_INFO_MMAP		0x00000001	/* hardware supports mmap */
+#define SNDRV_PCM_INFO_MMAP_VALID	0x00000002	/* period data are valid during transfer */
+#define SNDRV_PCM_INFO_DOUBLE		0x00000004	/* Double buffering needed for PCM start/stop */
+#define SNDRV_PCM_INFO_BATCH		0x00000010	/* double buffering */
+#define SNDRV_PCM_INFO_INTERLEAVED	0x00000100	/* channels are interleaved */
+#define SNDRV_PCM_INFO_NONINTERLEAVED	0x00000200	/* channels are not interleaved */
+#define SNDRV_PCM_INFO_COMPLEX		0x00000400	/* complex frame organization (mmap only) */
+#define SNDRV_PCM_INFO_BLOCK_TRANSFER	0x00010000	/* hardware transfer block of samples */
+#define SNDRV_PCM_INFO_OVERRANGE	0x00020000	/* hardware supports ADC (capture) overrange detection */
+#define SNDRV_PCM_INFO_RESUME		0x00040000	/* hardware supports stream resume after suspend */
+#define SNDRV_PCM_INFO_PAUSE		0x00080000	/* pause ioctl is supported */
+#define SNDRV_PCM_INFO_HALF_DUPLEX	0x00100000	/* only half duplex */
+#define SNDRV_PCM_INFO_JOINT_DUPLEX	0x00200000	/* playback and capture stream are somewhat correlated */
+#define SNDRV_PCM_INFO_SYNC_START	0x00400000	/* pcm support some kind of sync go */
+#define SNDRV_PCM_INFO_FIFO_IN_FRAMES	0x80000000	/* internal kernel flag - FIFO size is in frames */
+
+typedef int __bitwise snd_pcm_state_t;
+#define	SNDRV_PCM_STATE_OPEN		((__force snd_pcm_state_t) 0) /* stream is open */
+#define	SNDRV_PCM_STATE_SETUP		((__force snd_pcm_state_t) 1) /* stream has a setup */
+#define	SNDRV_PCM_STATE_PREPARED	((__force snd_pcm_state_t) 2) /* stream is ready to start */
+#define	SNDRV_PCM_STATE_RUNNING		((__force snd_pcm_state_t) 3) /* stream is running */
+#define	SNDRV_PCM_STATE_XRUN		((__force snd_pcm_state_t) 4) /* stream reached an xrun */
+#define	SNDRV_PCM_STATE_DRAINING	((__force snd_pcm_state_t) 5) /* stream is draining */
+#define	SNDRV_PCM_STATE_PAUSED		((__force snd_pcm_state_t) 6) /* stream is paused */
+#define	SNDRV_PCM_STATE_SUSPENDED	((__force snd_pcm_state_t) 7) /* hardware is suspended */
+#define	SNDRV_PCM_STATE_DISCONNECTED	((__force snd_pcm_state_t) 8) /* hardware is disconnected */
+#define	SNDRV_PCM_STATE_LAST		SNDRV_PCM_STATE_DISCONNECTED
+
+enum {
+	SNDRV_PCM_MMAP_OFFSET_DATA = 0x00000000,
+	SNDRV_PCM_MMAP_OFFSET_STATUS = 0x80000000,
+	SNDRV_PCM_MMAP_OFFSET_CONTROL = 0x81000000,
+};
+
+union snd_pcm_sync_id {
+	unsigned char id[16];
+	unsigned short id16[8];
+	unsigned int id32[4];
+};
+
+struct snd_pcm_info {
+	unsigned int device;		/* RO/WR (control): device number */
+	unsigned int subdevice;		/* RO/WR (control): subdevice number */
+	int stream;			/* RO/WR (control): stream direction */
+	int card;			/* R: card number */
+	unsigned char id[64];		/* ID (user selectable) */
+	unsigned char name[80];		/* name of this device */
+	unsigned char subname[32];	/* subdevice name */
+	int dev_class;			/* SNDRV_PCM_CLASS_* */
+	int dev_subclass;		/* SNDRV_PCM_SUBCLASS_* */
+	unsigned int subdevices_count;
+	unsigned int subdevices_avail;
+	union snd_pcm_sync_id sync;	/* hardware synchronization ID */
+	unsigned char reserved[64];	/* reserved for future... */
+};
+
+typedef int snd_pcm_hw_param_t;
+#define	SNDRV_PCM_HW_PARAM_ACCESS	0	/* Access type */
+#define	SNDRV_PCM_HW_PARAM_FORMAT	1	/* Format */
+#define	SNDRV_PCM_HW_PARAM_SUBFORMAT	2	/* Subformat */
+#define	SNDRV_PCM_HW_PARAM_FIRST_MASK	SNDRV_PCM_HW_PARAM_ACCESS
+#define	SNDRV_PCM_HW_PARAM_LAST_MASK	SNDRV_PCM_HW_PARAM_SUBFORMAT
+
+#define	SNDRV_PCM_HW_PARAM_SAMPLE_BITS	8	/* Bits per sample */
+#define	SNDRV_PCM_HW_PARAM_FRAME_BITS	9	/* Bits per frame */
+#define	SNDRV_PCM_HW_PARAM_CHANNELS	10	/* Channels */
+#define	SNDRV_PCM_HW_PARAM_RATE		11	/* Approx rate */
+#define	SNDRV_PCM_HW_PARAM_PERIOD_TIME	12	/* Approx distance between
+						 * interrupts in us
+						 */
+#define	SNDRV_PCM_HW_PARAM_PERIOD_SIZE	13	/* Approx frames between
+						 * interrupts
+						 */
+#define	SNDRV_PCM_HW_PARAM_PERIOD_BYTES	14	/* Approx bytes between
+						 * interrupts
+						 */
+#define	SNDRV_PCM_HW_PARAM_PERIODS	15	/* Approx interrupts per
+						 * buffer
+						 */
+#define	SNDRV_PCM_HW_PARAM_BUFFER_TIME	16	/* Approx duration of buffer
+						 * in us
+						 */
+#define	SNDRV_PCM_HW_PARAM_BUFFER_SIZE	17	/* Size of buffer in frames */
+#define	SNDRV_PCM_HW_PARAM_BUFFER_BYTES	18	/* Size of buffer in bytes */
+#define	SNDRV_PCM_HW_PARAM_TICK_TIME	19	/* Approx tick duration in us */
+#define	SNDRV_PCM_HW_PARAM_FIRST_INTERVAL	SNDRV_PCM_HW_PARAM_SAMPLE_BITS
+#define	SNDRV_PCM_HW_PARAM_LAST_INTERVAL	SNDRV_PCM_HW_PARAM_TICK_TIME
+
+#define SNDRV_PCM_HW_PARAMS_NORESAMPLE	(1<<0)	/* avoid rate resampling */
+
+struct snd_interval {
+	unsigned int min, max;
+	unsigned int openmin:1,
+		     openmax:1,
+		     integer:1,
+		     empty:1;
+};
+
+#define SNDRV_MASK_MAX	256
+
+struct snd_mask {
+	__u32 bits[(SNDRV_MASK_MAX+31)/32];
+};
+
+struct snd_pcm_hw_params {
+	unsigned int flags;
+	struct snd_mask masks[SNDRV_PCM_HW_PARAM_LAST_MASK - 
+			       SNDRV_PCM_HW_PARAM_FIRST_MASK + 1];
+	struct snd_mask mres[5];	/* reserved masks */
+	struct snd_interval intervals[SNDRV_PCM_HW_PARAM_LAST_INTERVAL -
+				        SNDRV_PCM_HW_PARAM_FIRST_INTERVAL + 1];
+	struct snd_interval ires[9];	/* reserved intervals */
+	unsigned int rmask;		/* W: requested masks */
+	unsigned int cmask;		/* R: changed masks */
+	unsigned int info;		/* R: Info flags for returned setup */
+	unsigned int msbits;		/* R: used most significant bits */
+	unsigned int rate_num;		/* R: rate numerator */
+	unsigned int rate_den;		/* R: rate denominator */
+	snd_pcm_uframes_t fifo_size;	/* R: chip FIFO size in frames */
+	unsigned char reserved[64];	/* reserved for future */
+};
+
+enum {
+	SNDRV_PCM_TSTAMP_NONE = 0,
+	SNDRV_PCM_TSTAMP_ENABLE,
+	SNDRV_PCM_TSTAMP_LAST = SNDRV_PCM_TSTAMP_ENABLE,
+};
+
+struct snd_pcm_sw_params {
+	int tstamp_mode;			/* timestamp mode */
+	unsigned int period_step;
+	unsigned int sleep_min;			/* min ticks to sleep */
+	snd_pcm_uframes_t avail_min;		/* min avail frames for wakeup */
+	snd_pcm_uframes_t xfer_align;		/* obsolete: xfer size need to be a multiple */
+	snd_pcm_uframes_t start_threshold;	/* min hw_avail frames for automatic start */
+	snd_pcm_uframes_t stop_threshold;	/* min avail frames for automatic stop */
+	snd_pcm_uframes_t silence_threshold;	/* min distance from noise for silence filling */
+	snd_pcm_uframes_t silence_size;		/* silence block size */
+	snd_pcm_uframes_t boundary;		/* pointers wrap point */
+	unsigned char reserved[64];		/* reserved for future */
+};
+
+struct snd_pcm_channel_info {
+	unsigned int channel;
+	__kernel_off_t offset;		/* mmap offset */
+	unsigned int first;		/* offset to first sample in bits */
+	unsigned int step;		/* samples distance in bits */
+};
+
+struct snd_pcm_status {
+	snd_pcm_state_t state;		/* stream state */
+	struct timespec trigger_tstamp;	/* time when stream was started/stopped/paused */
+	struct timespec tstamp;		/* reference timestamp */
+	snd_pcm_uframes_t appl_ptr;	/* appl ptr */
+	snd_pcm_uframes_t hw_ptr;	/* hw ptr */
+	snd_pcm_sframes_t delay;	/* current delay in frames */
+	snd_pcm_uframes_t avail;	/* number of frames available */
+	snd_pcm_uframes_t avail_max;	/* max frames available on hw since last status */
+	snd_pcm_uframes_t overrange;	/* count of ADC (capture) overrange detections from last status */
+	snd_pcm_state_t suspended_state; /* suspended stream state */
+	unsigned char reserved[60];	/* must be filled with zero */
+};
+
+struct snd_pcm_mmap_status {
+	snd_pcm_state_t state;		/* RO: state - SNDRV_PCM_STATE_XXXX */
+	int pad1;			/* Needed for 64 bit alignment */
+	snd_pcm_uframes_t hw_ptr;	/* RO: hw ptr (0...boundary-1) */
+	struct timespec tstamp;		/* Timestamp */
+	snd_pcm_state_t suspended_state; /* RO: suspended stream state */
+};
+
+struct snd_pcm_mmap_control {
+	snd_pcm_uframes_t appl_ptr;	/* RW: appl ptr (0...boundary-1) */
+	snd_pcm_uframes_t avail_min;	/* RW: min available frames for wakeup */
+};
+
+#define SNDRV_PCM_SYNC_PTR_HWSYNC	(1<<0)	/* execute hwsync */
+#define SNDRV_PCM_SYNC_PTR_APPL		(1<<1)	/* get appl_ptr from driver (r/w op) */
+#define SNDRV_PCM_SYNC_PTR_AVAIL_MIN	(1<<2)	/* get avail_min from driver */
+
+struct snd_pcm_sync_ptr {
+	unsigned int flags;
+	union {
+		struct snd_pcm_mmap_status status;
+		unsigned char reserved[64];
+	} s;
+	union {
+		struct snd_pcm_mmap_control control;
+		unsigned char reserved[64];
+	} c;
+};
+
+struct snd_xferi {
+	snd_pcm_sframes_t result;
+	void __user *buf;
+	snd_pcm_uframes_t frames;
+};
+
+struct snd_xfern {
+	snd_pcm_sframes_t result;
+	void __user * __user *bufs;
+	snd_pcm_uframes_t frames;
+};
+
+enum {
+	SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0,	/* gettimeofday equivalent */
+	SNDRV_PCM_TSTAMP_TYPE_MONOTONIC,	/* posix_clock_monotonic equivalent */
+	SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC,
+};
+
+#define SNDRV_PCM_IOCTL_PVERSION	_IOR('A', 0x00, int)
+#define SNDRV_PCM_IOCTL_INFO		_IOR('A', 0x01, struct snd_pcm_info)
+#define SNDRV_PCM_IOCTL_TSTAMP		_IOW('A', 0x02, int)
+#define SNDRV_PCM_IOCTL_TTSTAMP		_IOW('A', 0x03, int)
+#define SNDRV_PCM_IOCTL_HW_REFINE	_IOWR('A', 0x10, struct snd_pcm_hw_params)
+#define SNDRV_PCM_IOCTL_HW_PARAMS	_IOWR('A', 0x11, struct snd_pcm_hw_params)
+#define SNDRV_PCM_IOCTL_HW_FREE		_IO('A', 0x12)
+#define SNDRV_PCM_IOCTL_SW_PARAMS	_IOWR('A', 0x13, struct snd_pcm_sw_params)
+#define SNDRV_PCM_IOCTL_STATUS		_IOR('A', 0x20, struct snd_pcm_status)
+#define SNDRV_PCM_IOCTL_DELAY		_IOR('A', 0x21, snd_pcm_sframes_t)
+#define SNDRV_PCM_IOCTL_HWSYNC		_IO('A', 0x22)
+#define SNDRV_PCM_IOCTL_SYNC_PTR	_IOWR('A', 0x23, struct snd_pcm_sync_ptr)
+#define SNDRV_PCM_IOCTL_CHANNEL_INFO	_IOR('A', 0x32, struct snd_pcm_channel_info)
+#define SNDRV_PCM_IOCTL_PREPARE		_IO('A', 0x40)
+#define SNDRV_PCM_IOCTL_RESET		_IO('A', 0x41)
+#define SNDRV_PCM_IOCTL_START		_IO('A', 0x42)
+#define SNDRV_PCM_IOCTL_DROP		_IO('A', 0x43)
+#define SNDRV_PCM_IOCTL_DRAIN		_IO('A', 0x44)
+#define SNDRV_PCM_IOCTL_PAUSE		_IOW('A', 0x45, int)
+#define SNDRV_PCM_IOCTL_REWIND		_IOW('A', 0x46, snd_pcm_uframes_t)
+#define SNDRV_PCM_IOCTL_RESUME		_IO('A', 0x47)
+#define SNDRV_PCM_IOCTL_XRUN		_IO('A', 0x48)
+#define SNDRV_PCM_IOCTL_FORWARD		_IOW('A', 0x49, snd_pcm_uframes_t)
+#define SNDRV_PCM_IOCTL_WRITEI_FRAMES	_IOW('A', 0x50, struct snd_xferi)
+#define SNDRV_PCM_IOCTL_READI_FRAMES	_IOR('A', 0x51, struct snd_xferi)
+#define SNDRV_PCM_IOCTL_WRITEN_FRAMES	_IOW('A', 0x52, struct snd_xfern)
+#define SNDRV_PCM_IOCTL_READN_FRAMES	_IOR('A', 0x53, struct snd_xfern)
+#define SNDRV_PCM_IOCTL_LINK		_IOW('A', 0x60, int)
+#define SNDRV_PCM_IOCTL_UNLINK		_IO('A', 0x61)
+
+/*****************************************************************************
+ *                                                                           *
+ *                            MIDI v1.0 interface                            *
+ *                                                                           *
+ *****************************************************************************/
+
+/*
+ *  Raw MIDI section - /dev/snd/midi??
+ */
+
+#define SNDRV_RAWMIDI_VERSION		SNDRV_PROTOCOL_VERSION(2, 0, 0)
+
+enum {
+	SNDRV_RAWMIDI_STREAM_OUTPUT = 0,
+	SNDRV_RAWMIDI_STREAM_INPUT,
+	SNDRV_RAWMIDI_STREAM_LAST = SNDRV_RAWMIDI_STREAM_INPUT,
+};
+
+#define SNDRV_RAWMIDI_INFO_OUTPUT		0x00000001
+#define SNDRV_RAWMIDI_INFO_INPUT		0x00000002
+#define SNDRV_RAWMIDI_INFO_DUPLEX		0x00000004
+
+struct snd_rawmidi_info {
+	unsigned int device;		/* RO/WR (control): device number */
+	unsigned int subdevice;		/* RO/WR (control): subdevice number */
+	int stream;			/* WR: stream */
+	int card;			/* R: card number */
+	unsigned int flags;		/* SNDRV_RAWMIDI_INFO_XXXX */
+	unsigned char id[64];		/* ID (user selectable) */
+	unsigned char name[80];		/* name of device */
+	unsigned char subname[32];	/* name of active or selected subdevice */
+	unsigned int subdevices_count;
+	unsigned int subdevices_avail;
+	unsigned char reserved[64];	/* reserved for future use */
+};
+
+struct snd_rawmidi_params {
+	int stream;
+	size_t buffer_size;		/* queue size in bytes */
+	size_t avail_min;		/* minimum avail bytes for wakeup */
+	unsigned int no_active_sensing: 1; /* do not send active sensing byte in close() */
+	unsigned char reserved[16];	/* reserved for future use */
+};
+
+struct snd_rawmidi_status {
+	int stream;
+	struct timespec tstamp;		/* Timestamp */
+	size_t avail;			/* available bytes */
+	size_t xruns;			/* count of overruns since last status (in bytes) */
+	unsigned char reserved[16];	/* reserved for future use */
+};
+
+#define SNDRV_RAWMIDI_IOCTL_PVERSION	_IOR('W', 0x00, int)
+#define SNDRV_RAWMIDI_IOCTL_INFO	_IOR('W', 0x01, struct snd_rawmidi_info)
+#define SNDRV_RAWMIDI_IOCTL_PARAMS	_IOWR('W', 0x10, struct snd_rawmidi_params)
+#define SNDRV_RAWMIDI_IOCTL_STATUS	_IOWR('W', 0x20, struct snd_rawmidi_status)
+#define SNDRV_RAWMIDI_IOCTL_DROP	_IOW('W', 0x30, int)
+#define SNDRV_RAWMIDI_IOCTL_DRAIN	_IOW('W', 0x31, int)
+
+/*
+ *  Timer section - /dev/snd/timer
+ */
+
+#define SNDRV_TIMER_VERSION		SNDRV_PROTOCOL_VERSION(2, 0, 6)
+
+enum {
+	SNDRV_TIMER_CLASS_NONE = -1,
+	SNDRV_TIMER_CLASS_SLAVE = 0,
+	SNDRV_TIMER_CLASS_GLOBAL,
+	SNDRV_TIMER_CLASS_CARD,
+	SNDRV_TIMER_CLASS_PCM,
+	SNDRV_TIMER_CLASS_LAST = SNDRV_TIMER_CLASS_PCM,
+};
+
+/* slave timer classes */
+enum {
+	SNDRV_TIMER_SCLASS_NONE = 0,
+	SNDRV_TIMER_SCLASS_APPLICATION,
+	SNDRV_TIMER_SCLASS_SEQUENCER,		/* alias */
+	SNDRV_TIMER_SCLASS_OSS_SEQUENCER,	/* alias */
+	SNDRV_TIMER_SCLASS_LAST = SNDRV_TIMER_SCLASS_OSS_SEQUENCER,
+};
+
+/* global timers (device member) */
+#define SNDRV_TIMER_GLOBAL_SYSTEM	0
+#define SNDRV_TIMER_GLOBAL_RTC		1
+#define SNDRV_TIMER_GLOBAL_HPET		2
+#define SNDRV_TIMER_GLOBAL_HRTIMER	3
+
+/* info flags */
+#define SNDRV_TIMER_FLG_SLAVE		(1<<0)	/* cannot be controlled */
+
+struct snd_timer_id {
+	int dev_class;	
+	int dev_sclass;
+	int card;
+	int device;
+	int subdevice;
+};
+
+struct snd_timer_ginfo {
+	struct snd_timer_id tid;	/* requested timer ID */
+	unsigned int flags;		/* timer flags - SNDRV_TIMER_FLG_* */
+	int card;			/* card number */
+	unsigned char id[64];		/* timer identification */
+	unsigned char name[80];		/* timer name */
+	unsigned long reserved0;	/* reserved for future use */
+	unsigned long resolution;	/* average period resolution in ns */
+	unsigned long resolution_min;	/* minimal period resolution in ns */
+	unsigned long resolution_max;	/* maximal period resolution in ns */
+	unsigned int clients;		/* active timer clients */
+	unsigned char reserved[32];
+};
+
+struct snd_timer_gparams {
+	struct snd_timer_id tid;	/* requested timer ID */
+	unsigned long period_num;	/* requested precise period duration (in seconds) - numerator */
+	unsigned long period_den;	/* requested precise period duration (in seconds) - denominator */
+	unsigned char reserved[32];
+};
+
+struct snd_timer_gstatus {
+	struct snd_timer_id tid;	/* requested timer ID */
+	unsigned long resolution;	/* current period resolution in ns */
+	unsigned long resolution_num;	/* precise current period resolution (in seconds) - numerator */
+	unsigned long resolution_den;	/* precise current period resolution (in seconds) - denominator */
+	unsigned char reserved[32];
+};
+
+struct snd_timer_select {
+	struct snd_timer_id id;	/* bind to timer ID */
+	unsigned char reserved[32];	/* reserved */
+};
+
+struct snd_timer_info {
+	unsigned int flags;		/* timer flags - SNDRV_TIMER_FLG_* */
+	int card;			/* card number */
+	unsigned char id[64];		/* timer identificator */
+	unsigned char name[80];		/* timer name */
+	unsigned long reserved0;	/* reserved for future use */
+	unsigned long resolution;	/* average period resolution in ns */
+	unsigned char reserved[64];	/* reserved */
+};
+
+#define SNDRV_TIMER_PSFLG_AUTO		(1<<0)	/* auto start, otherwise one-shot */
+#define SNDRV_TIMER_PSFLG_EXCLUSIVE	(1<<1)	/* exclusive use, precise start/stop/pause/continue */
+#define SNDRV_TIMER_PSFLG_EARLY_EVENT	(1<<2)	/* write early event to the poll queue */
+
+struct snd_timer_params {
+	unsigned int flags;		/* flags - SNDRV_MIXER_PSFLG_* */
+	unsigned int ticks;		/* requested resolution in ticks */
+	unsigned int queue_size;	/* total size of queue (32-1024) */
+	unsigned int reserved0;		/* reserved, was: failure locations */
+	unsigned int filter;		/* event filter (bitmask of SNDRV_TIMER_EVENT_*) */
+	unsigned char reserved[60];	/* reserved */
+};
+
+struct snd_timer_status {
+	struct timespec tstamp;		/* Timestamp - last update */
+	unsigned int resolution;	/* current period resolution in ns */
+	unsigned int lost;		/* counter of master tick lost */
+	unsigned int overrun;		/* count of read queue overruns */
+	unsigned int queue;		/* used queue size */
+	unsigned char reserved[64];	/* reserved */
+};
+
+#define SNDRV_TIMER_IOCTL_PVERSION	_IOR('T', 0x00, int)
+#define SNDRV_TIMER_IOCTL_NEXT_DEVICE	_IOWR('T', 0x01, struct snd_timer_id)
+#define SNDRV_TIMER_IOCTL_TREAD		_IOW('T', 0x02, int)
+#define SNDRV_TIMER_IOCTL_GINFO		_IOWR('T', 0x03, struct snd_timer_ginfo)
+#define SNDRV_TIMER_IOCTL_GPARAMS	_IOW('T', 0x04, struct snd_timer_gparams)
+#define SNDRV_TIMER_IOCTL_GSTATUS	_IOWR('T', 0x05, struct snd_timer_gstatus)
+#define SNDRV_TIMER_IOCTL_SELECT	_IOW('T', 0x10, struct snd_timer_select)
+#define SNDRV_TIMER_IOCTL_INFO		_IOR('T', 0x11, struct snd_timer_info)
+#define SNDRV_TIMER_IOCTL_PARAMS	_IOW('T', 0x12, struct snd_timer_params)
+#define SNDRV_TIMER_IOCTL_STATUS	_IOR('T', 0x14, struct snd_timer_status)
+/* The following four ioctls are changed since 1.0.9 due to confliction */
+#define SNDRV_TIMER_IOCTL_START		_IO('T', 0xa0)
+#define SNDRV_TIMER_IOCTL_STOP		_IO('T', 0xa1)
+#define SNDRV_TIMER_IOCTL_CONTINUE	_IO('T', 0xa2)
+#define SNDRV_TIMER_IOCTL_PAUSE		_IO('T', 0xa3)
+
+struct snd_timer_read {
+	unsigned int resolution;
+	unsigned int ticks;
+};
+
+enum {
+	SNDRV_TIMER_EVENT_RESOLUTION = 0,	/* val = resolution in ns */
+	SNDRV_TIMER_EVENT_TICK,			/* val = ticks */
+	SNDRV_TIMER_EVENT_START,		/* val = resolution in ns */
+	SNDRV_TIMER_EVENT_STOP,			/* val = 0 */
+	SNDRV_TIMER_EVENT_CONTINUE,		/* val = resolution in ns */
+	SNDRV_TIMER_EVENT_PAUSE,		/* val = 0 */
+	SNDRV_TIMER_EVENT_EARLY,		/* val = 0, early event */
+	SNDRV_TIMER_EVENT_SUSPEND,		/* val = 0 */
+	SNDRV_TIMER_EVENT_RESUME,		/* val = resolution in ns */
+	/* master timer events for slave timer instances */
+	SNDRV_TIMER_EVENT_MSTART = SNDRV_TIMER_EVENT_START + 10,
+	SNDRV_TIMER_EVENT_MSTOP = SNDRV_TIMER_EVENT_STOP + 10,
+	SNDRV_TIMER_EVENT_MCONTINUE = SNDRV_TIMER_EVENT_CONTINUE + 10,
+	SNDRV_TIMER_EVENT_MPAUSE = SNDRV_TIMER_EVENT_PAUSE + 10,
+	SNDRV_TIMER_EVENT_MSUSPEND = SNDRV_TIMER_EVENT_SUSPEND + 10,
+	SNDRV_TIMER_EVENT_MRESUME = SNDRV_TIMER_EVENT_RESUME + 10,
+};
+
+struct snd_timer_tread {
+	int event;
+	struct timespec tstamp;
+	unsigned int val;
+};
+
+/****************************************************************************
+ *                                                                          *
+ *        Section for driver control interface - /dev/snd/control?          *
+ *                                                                          *
+ ****************************************************************************/
+
+#define SNDRV_CTL_VERSION		SNDRV_PROTOCOL_VERSION(2, 0, 6)
+
+struct snd_ctl_card_info {
+	int card;			/* card number */
+	int pad;			/* reserved for future (was type) */
+	unsigned char id[16];		/* ID of card (user selectable) */
+	unsigned char driver[16];	/* Driver name */
+	unsigned char name[32];		/* Short name of soundcard */
+	unsigned char longname[80];	/* name + info text about soundcard */
+	unsigned char reserved_[16];	/* reserved for future (was ID of mixer) */
+	unsigned char mixername[80];	/* visual mixer identification */
+	unsigned char components[128];	/* card components / fine identification, delimited with one space (AC97 etc..) */
+};
+
+typedef int __bitwise snd_ctl_elem_type_t;
+#define	SNDRV_CTL_ELEM_TYPE_NONE	((__force snd_ctl_elem_type_t) 0) /* invalid */
+#define	SNDRV_CTL_ELEM_TYPE_BOOLEAN	((__force snd_ctl_elem_type_t) 1) /* boolean type */
+#define	SNDRV_CTL_ELEM_TYPE_INTEGER	((__force snd_ctl_elem_type_t) 2) /* integer type */
+#define	SNDRV_CTL_ELEM_TYPE_ENUMERATED	((__force snd_ctl_elem_type_t) 3) /* enumerated type */
+#define	SNDRV_CTL_ELEM_TYPE_BYTES	((__force snd_ctl_elem_type_t) 4) /* byte array */
+#define	SNDRV_CTL_ELEM_TYPE_IEC958	((__force snd_ctl_elem_type_t) 5) /* IEC958 (S/PDIF) setup */
+#define	SNDRV_CTL_ELEM_TYPE_INTEGER64	((__force snd_ctl_elem_type_t) 6) /* 64-bit integer type */
+#define	SNDRV_CTL_ELEM_TYPE_LAST	SNDRV_CTL_ELEM_TYPE_INTEGER64
+
+typedef int __bitwise snd_ctl_elem_iface_t;
+#define	SNDRV_CTL_ELEM_IFACE_CARD	((__force snd_ctl_elem_iface_t) 0) /* global control */
+#define	SNDRV_CTL_ELEM_IFACE_HWDEP	((__force snd_ctl_elem_iface_t) 1) /* hardware dependent device */
+#define	SNDRV_CTL_ELEM_IFACE_MIXER	((__force snd_ctl_elem_iface_t) 2) /* virtual mixer device */
+#define	SNDRV_CTL_ELEM_IFACE_PCM	((__force snd_ctl_elem_iface_t) 3) /* PCM device */
+#define	SNDRV_CTL_ELEM_IFACE_RAWMIDI	((__force snd_ctl_elem_iface_t) 4) /* RawMidi device */
+#define	SNDRV_CTL_ELEM_IFACE_TIMER	((__force snd_ctl_elem_iface_t) 5) /* timer device */
+#define	SNDRV_CTL_ELEM_IFACE_SEQUENCER	((__force snd_ctl_elem_iface_t) 6) /* sequencer client */
+#define	SNDRV_CTL_ELEM_IFACE_LAST	SNDRV_CTL_ELEM_IFACE_SEQUENCER
+
+#define SNDRV_CTL_ELEM_ACCESS_READ		(1<<0)
+#define SNDRV_CTL_ELEM_ACCESS_WRITE		(1<<1)
+#define SNDRV_CTL_ELEM_ACCESS_READWRITE		(SNDRV_CTL_ELEM_ACCESS_READ|SNDRV_CTL_ELEM_ACCESS_WRITE)
+#define SNDRV_CTL_ELEM_ACCESS_VOLATILE		(1<<2)	/* control value may be changed without a notification */
+#define SNDRV_CTL_ELEM_ACCESS_TIMESTAMP		(1<<3)	/* when was control changed */
+#define SNDRV_CTL_ELEM_ACCESS_TLV_READ		(1<<4)	/* TLV read is possible */
+#define SNDRV_CTL_ELEM_ACCESS_TLV_WRITE		(1<<5)	/* TLV write is possible */
+#define SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE	(SNDRV_CTL_ELEM_ACCESS_TLV_READ|SNDRV_CTL_ELEM_ACCESS_TLV_WRITE)
+#define SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND	(1<<6)	/* TLV command is possible */
+#define SNDRV_CTL_ELEM_ACCESS_INACTIVE		(1<<8)	/* control does actually nothing, but may be updated */
+#define SNDRV_CTL_ELEM_ACCESS_LOCK		(1<<9)	/* write lock */
+#define SNDRV_CTL_ELEM_ACCESS_OWNER		(1<<10)	/* write lock owner */
+#define SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK	(1<<28)	/* kernel use a TLV callback */ 
+#define SNDRV_CTL_ELEM_ACCESS_USER		(1<<29) /* user space element */
+/* bits 30 and 31 are obsoleted (for indirect access) */
+
+/* for further details see the ACPI and PCI power management specification */
+#define SNDRV_CTL_POWER_D0		0x0000	/* full On */
+#define SNDRV_CTL_POWER_D1		0x0100	/* partial On */
+#define SNDRV_CTL_POWER_D2		0x0200	/* partial On */
+#define SNDRV_CTL_POWER_D3		0x0300	/* Off */
+#define SNDRV_CTL_POWER_D3hot		(SNDRV_CTL_POWER_D3|0x0000)	/* Off, with power */
+#define SNDRV_CTL_POWER_D3cold		(SNDRV_CTL_POWER_D3|0x0001)	/* Off, without power */
+
+struct snd_ctl_elem_id {
+	unsigned int numid;		/* numeric identifier, zero = invalid */
+	snd_ctl_elem_iface_t iface;	/* interface identifier */
+	unsigned int device;		/* device/client number */
+	unsigned int subdevice;		/* subdevice (substream) number */
+        unsigned char name[44];		/* ASCII name of item */
+	unsigned int index;		/* index of item */
+};
+
+struct snd_ctl_elem_list {
+	unsigned int offset;		/* W: first element ID to get */
+	unsigned int space;		/* W: count of element IDs to get */
+	unsigned int used;		/* R: count of element IDs set */
+	unsigned int count;		/* R: count of all elements */
+	struct snd_ctl_elem_id __user *pids; /* R: IDs */
+	unsigned char reserved[50];
+};
+
+struct snd_ctl_elem_info {
+	struct snd_ctl_elem_id id;	/* W: element ID */
+	snd_ctl_elem_type_t type;	/* R: value type - SNDRV_CTL_ELEM_TYPE_* */
+	unsigned int access;		/* R: value access (bitmask) - SNDRV_CTL_ELEM_ACCESS_* */
+	unsigned int count;		/* count of values */
+	__kernel_pid_t owner;		/* owner's PID of this control */
+	union {
+		struct {
+			long min;		/* R: minimum value */
+			long max;		/* R: maximum value */
+			long step;		/* R: step (0 variable) */
+		} integer;
+		struct {
+			long long min;		/* R: minimum value */
+			long long max;		/* R: maximum value */
+			long long step;		/* R: step (0 variable) */
+		} integer64;
+		struct {
+			unsigned int items;	/* R: number of items */
+			unsigned int item;	/* W: item number */
+			char name[64];		/* R: value name */
+		} enumerated;
+		unsigned char reserved[128];
+	} value;
+	union {
+		unsigned short d[4];		/* dimensions */
+		unsigned short *d_ptr;		/* indirect - obsoleted */
+	} dimen;
+	unsigned char reserved[64-4*sizeof(unsigned short)];
+};
+
+struct snd_ctl_elem_value {
+	struct snd_ctl_elem_id id;	/* W: element ID */
+	unsigned int indirect: 1;	/* W: indirect access - obsoleted */
+        union {
+		union {
+			long value[128];
+			long *value_ptr;	/* obsoleted */
+		} integer;
+		union {
+			long long value[64];
+			long long *value_ptr;	/* obsoleted */
+		} integer64;
+		union {
+			unsigned int item[128];
+			unsigned int *item_ptr;	/* obsoleted */
+		} enumerated;
+		union {
+			unsigned char data[512];
+			unsigned char *data_ptr;	/* obsoleted */
+		} bytes;
+		struct snd_aes_iec958 iec958;
+        } value;                /* RO */
+	struct timespec tstamp;
+        unsigned char reserved[128-sizeof(struct timespec)];
+};
+
+struct snd_ctl_tlv {
+        unsigned int numid;	/* control element numeric identification */
+        unsigned int length;	/* in bytes aligned to 4 */
+        unsigned int tlv[0];	/* first TLV */
+};
+
+#define SNDRV_CTL_IOCTL_PVERSION	_IOR('U', 0x00, int)
+#define SNDRV_CTL_IOCTL_CARD_INFO	_IOR('U', 0x01, struct snd_ctl_card_info)
+#define SNDRV_CTL_IOCTL_ELEM_LIST	_IOWR('U', 0x10, struct snd_ctl_elem_list)
+#define SNDRV_CTL_IOCTL_ELEM_INFO	_IOWR('U', 0x11, struct snd_ctl_elem_info)
+#define SNDRV_CTL_IOCTL_ELEM_READ	_IOWR('U', 0x12, struct snd_ctl_elem_value)
+#define SNDRV_CTL_IOCTL_ELEM_WRITE	_IOWR('U', 0x13, struct snd_ctl_elem_value)
+#define SNDRV_CTL_IOCTL_ELEM_LOCK	_IOW('U', 0x14, struct snd_ctl_elem_id)
+#define SNDRV_CTL_IOCTL_ELEM_UNLOCK	_IOW('U', 0x15, struct snd_ctl_elem_id)
+#define SNDRV_CTL_IOCTL_SUBSCRIBE_EVENTS _IOWR('U', 0x16, int)
+#define SNDRV_CTL_IOCTL_ELEM_ADD	_IOWR('U', 0x17, struct snd_ctl_elem_info)
+#define SNDRV_CTL_IOCTL_ELEM_REPLACE	_IOWR('U', 0x18, struct snd_ctl_elem_info)
+#define SNDRV_CTL_IOCTL_ELEM_REMOVE	_IOWR('U', 0x19, struct snd_ctl_elem_id)
+#define SNDRV_CTL_IOCTL_TLV_READ	_IOWR('U', 0x1a, struct snd_ctl_tlv)
+#define SNDRV_CTL_IOCTL_TLV_WRITE	_IOWR('U', 0x1b, struct snd_ctl_tlv)
+#define SNDRV_CTL_IOCTL_TLV_COMMAND	_IOWR('U', 0x1c, struct snd_ctl_tlv)
+#define SNDRV_CTL_IOCTL_HWDEP_NEXT_DEVICE _IOWR('U', 0x20, int)
+#define SNDRV_CTL_IOCTL_HWDEP_INFO	_IOR('U', 0x21, struct snd_hwdep_info)
+#define SNDRV_CTL_IOCTL_PCM_NEXT_DEVICE	_IOR('U', 0x30, int)
+#define SNDRV_CTL_IOCTL_PCM_INFO	_IOWR('U', 0x31, struct snd_pcm_info)
+#define SNDRV_CTL_IOCTL_PCM_PREFER_SUBDEVICE _IOW('U', 0x32, int)
+#define SNDRV_CTL_IOCTL_RAWMIDI_NEXT_DEVICE _IOWR('U', 0x40, int)
+#define SNDRV_CTL_IOCTL_RAWMIDI_INFO	_IOWR('U', 0x41, struct snd_rawmidi_info)
+#define SNDRV_CTL_IOCTL_RAWMIDI_PREFER_SUBDEVICE _IOW('U', 0x42, int)
+#define SNDRV_CTL_IOCTL_POWER		_IOWR('U', 0xd0, int)
+#define SNDRV_CTL_IOCTL_POWER_STATE	_IOR('U', 0xd1, int)
+
+/*
+ *  Read interface.
+ */
+
+enum sndrv_ctl_event_type {
+	SNDRV_CTL_EVENT_ELEM = 0,
+	SNDRV_CTL_EVENT_LAST = SNDRV_CTL_EVENT_ELEM,
+};
+
+#define SNDRV_CTL_EVENT_MASK_VALUE	(1<<0)	/* element value was changed */
+#define SNDRV_CTL_EVENT_MASK_INFO	(1<<1)	/* element info was changed */
+#define SNDRV_CTL_EVENT_MASK_ADD	(1<<2)	/* element was added */
+#define SNDRV_CTL_EVENT_MASK_TLV	(1<<3)	/* element TLV tree was changed */
+#define SNDRV_CTL_EVENT_MASK_REMOVE	(~0U)	/* element was removed */
+
+struct snd_ctl_event {
+	int type;	/* event type - SNDRV_CTL_EVENT_* */
+	union {
+		struct {
+			unsigned int mask;
+			struct snd_ctl_elem_id id;
+		} elem;
+                unsigned char data8[60];
+        } data;
+};
+
+/*
+ *  Control names
+ */
+
+#define SNDRV_CTL_NAME_NONE				""
+#define SNDRV_CTL_NAME_PLAYBACK				"Playback "
+#define SNDRV_CTL_NAME_CAPTURE				"Capture "
+
+#define SNDRV_CTL_NAME_IEC958_NONE			""
+#define SNDRV_CTL_NAME_IEC958_SWITCH			"Switch"
+#define SNDRV_CTL_NAME_IEC958_VOLUME			"Volume"
+#define SNDRV_CTL_NAME_IEC958_DEFAULT			"Default"
+#define SNDRV_CTL_NAME_IEC958_MASK			"Mask"
+#define SNDRV_CTL_NAME_IEC958_CON_MASK			"Con Mask"
+#define SNDRV_CTL_NAME_IEC958_PRO_MASK			"Pro Mask"
+#define SNDRV_CTL_NAME_IEC958_PCM_STREAM		"PCM Stream"
+#define SNDRV_CTL_NAME_IEC958(expl,direction,what)	"IEC958 " expl SNDRV_CTL_NAME_##direction SNDRV_CTL_NAME_IEC958_##what
+
+#endif /* __SOUND_ASOUND_H */
diff --git a/include/sound/asound_fm.h b/include/sound/asound_fm.h
new file mode 100644
index 0000000..c2a4b96
--- /dev/null
+++ b/include/sound/asound_fm.h
@@ -0,0 +1,134 @@
+#ifndef __SOUND_ASOUND_FM_H
+#define __SOUND_ASOUND_FM_H
+
+/*
+ *  Advanced Linux Sound Architecture - ALSA
+ *
+ *  Interface file between ALSA driver & user space
+ *  Copyright (c) 1994-98 by Jaroslav Kysela <perex@perex.cz>,
+ *                           4Front Technologies
+ *
+ *  Direct FM control
+ *
+ *   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., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ */
+
+#define SNDRV_DM_FM_MODE_OPL2	0x00
+#define SNDRV_DM_FM_MODE_OPL3	0x01
+
+struct snd_dm_fm_info {
+	unsigned char fm_mode;		/* OPL mode, see SNDRV_DM_FM_MODE_XXX */
+	unsigned char rhythm;		/* percussion mode flag */
+};
+
+/*
+ *  Data structure composing an FM "note" or sound event.
+ */
+
+struct snd_dm_fm_voice {
+	unsigned char op;		/* operator cell (0 or 1) */
+	unsigned char voice;		/* FM voice (0 to 17) */
+
+	unsigned char am;		/* amplitude modulation */
+	unsigned char vibrato;		/* vibrato effect */
+	unsigned char do_sustain;	/* sustain phase */
+	unsigned char kbd_scale;	/* keyboard scaling */
+	unsigned char harmonic;		/* 4 bits: harmonic and multiplier */
+	unsigned char scale_level;	/* 2 bits: decrease output freq rises */
+	unsigned char volume;		/* 6 bits: volume */
+
+	unsigned char attack;		/* 4 bits: attack rate */
+	unsigned char decay;		/* 4 bits: decay rate */
+	unsigned char sustain;		/* 4 bits: sustain level */
+	unsigned char release;		/* 4 bits: release rate */
+
+	unsigned char feedback;		/* 3 bits: feedback for op0 */
+	unsigned char connection;	/* 0 for serial, 1 for parallel */
+	unsigned char left;		/* stereo left */
+	unsigned char right;		/* stereo right */
+	unsigned char waveform;		/* 3 bits: waveform shape */
+};
+
+/*
+ *  This describes an FM note by its voice, octave, frequency number (10bit)
+ *  and key on/off.
+ */
+
+struct snd_dm_fm_note {
+	unsigned char voice;	/* 0-17 voice channel */
+	unsigned char octave;	/* 3 bits: what octave to play */
+	unsigned int fnum;	/* 10 bits: frequency number */
+	unsigned char key_on;	/* set for active, clear for silent */
+};
+
+/*
+ *  FM parameters that apply globally to all voices, and thus are not "notes"
+ */
+
+struct snd_dm_fm_params {
+	unsigned char am_depth;		/* amplitude modulation depth (1=hi) */
+	unsigned char vib_depth;	/* vibrato depth (1=hi) */
+	unsigned char kbd_split;	/* keyboard split */
+	unsigned char rhythm;		/* percussion mode select */
+
+	/* This block is the percussion instrument data */
+	unsigned char bass;
+	unsigned char snare;
+	unsigned char tomtom;
+	unsigned char cymbal;
+	unsigned char hihat;
+};
+
+/*
+ *  FM mode ioctl settings
+ */
+
+#define SNDRV_DM_FM_IOCTL_INFO		_IOR('H', 0x20, struct snd_dm_fm_info)
+#define SNDRV_DM_FM_IOCTL_RESET		_IO ('H', 0x21)
+#define SNDRV_DM_FM_IOCTL_PLAY_NOTE	_IOW('H', 0x22, struct snd_dm_fm_note)
+#define SNDRV_DM_FM_IOCTL_SET_VOICE	_IOW('H', 0x23, struct snd_dm_fm_voice)
+#define SNDRV_DM_FM_IOCTL_SET_PARAMS	_IOW('H', 0x24, struct snd_dm_fm_params)
+#define SNDRV_DM_FM_IOCTL_SET_MODE	_IOW('H', 0x25, int)
+/* for OPL3 only */
+#define SNDRV_DM_FM_IOCTL_SET_CONNECTION	_IOW('H', 0x26, int)
+/* SBI patch management */
+#define SNDRV_DM_FM_IOCTL_CLEAR_PATCHES	_IO ('H', 0x40)
+
+#define SNDRV_DM_FM_OSS_IOCTL_RESET		0x20
+#define SNDRV_DM_FM_OSS_IOCTL_PLAY_NOTE		0x21
+#define SNDRV_DM_FM_OSS_IOCTL_SET_VOICE		0x22
+#define SNDRV_DM_FM_OSS_IOCTL_SET_PARAMS	0x23
+#define SNDRV_DM_FM_OSS_IOCTL_SET_MODE		0x24
+#define SNDRV_DM_FM_OSS_IOCTL_SET_OPL		0x25
+
+/*
+ * Patch Record - fixed size for write
+ */
+
+#define FM_KEY_SBI	"SBI\032"
+#define FM_KEY_2OP	"2OP\032"
+#define FM_KEY_4OP	"4OP\032"
+
+struct sbi_patch {
+	unsigned char prog;
+	unsigned char bank;
+	char key[4];
+	char name[25];
+	char extension[7];
+	unsigned char data[32];
+};
+
+#endif /* __SOUND_ASOUND_FM_H */
diff --git a/include/sound/asoundef.h b/include/sound/asoundef.h
new file mode 100644
index 0000000..20ebf32
--- /dev/null
+++ b/include/sound/asoundef.h
@@ -0,0 +1,284 @@
+#ifndef __SOUND_ASOUNDEF_H
+#define __SOUND_ASOUNDEF_H
+
+/*
+ *  Advanced Linux Sound Architecture - ALSA - Driver
+ *  Copyright (c) 1994-2000 by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ *   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., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ */
+
+/****************************************************************************
+ *                                                                          *
+ *        Digital audio interface					    *
+ *                                                                          *
+ ****************************************************************************/
+
+/* AES/IEC958 channel status bits */
+#define IEC958_AES0_PROFESSIONAL	(1<<0)	/* 0 = consumer, 1 = professional */
+#define IEC958_AES0_NONAUDIO		(1<<1)	/* 0 = audio, 1 = non-audio */
+#define IEC958_AES0_PRO_EMPHASIS	(7<<2)	/* mask - emphasis */
+#define IEC958_AES0_PRO_EMPHASIS_NOTID	(0<<2)	/* emphasis not indicated */
+#define IEC958_AES0_PRO_EMPHASIS_NONE	(1<<2)	/* none emphasis */
+#define IEC958_AES0_PRO_EMPHASIS_5015	(3<<2)	/* 50/15us emphasis */
+#define IEC958_AES0_PRO_EMPHASIS_CCITT	(7<<2)	/* CCITT J.17 emphasis */
+#define IEC958_AES0_PRO_FREQ_UNLOCKED	(1<<5)	/* source sample frequency: 0 = locked, 1 = unlocked */
+#define IEC958_AES0_PRO_FS		(3<<6)	/* mask - sample frequency */
+#define IEC958_AES0_PRO_FS_NOTID	(0<<6)	/* fs not indicated */
+#define IEC958_AES0_PRO_FS_44100	(1<<6)	/* 44.1kHz */
+#define IEC958_AES0_PRO_FS_48000	(2<<6)	/* 48kHz */
+#define IEC958_AES0_PRO_FS_32000	(3<<6)	/* 32kHz */
+#define IEC958_AES0_CON_NOT_COPYRIGHT	(1<<2)	/* 0 = copyright, 1 = not copyright */
+#define IEC958_AES0_CON_EMPHASIS	(7<<3)	/* mask - emphasis */
+#define IEC958_AES0_CON_EMPHASIS_NONE	(0<<3)	/* none emphasis */
+#define IEC958_AES0_CON_EMPHASIS_5015	(1<<3)	/* 50/15us emphasis */
+#define IEC958_AES0_CON_MODE		(3<<6)	/* mask - mode */
+#define IEC958_AES1_PRO_MODE		(15<<0)	/* mask - channel mode */
+#define IEC958_AES1_PRO_MODE_NOTID	(0<<0)	/* not indicated */
+#define IEC958_AES1_PRO_MODE_STEREOPHONIC (2<<0) /* stereophonic - ch A is left */
+#define IEC958_AES1_PRO_MODE_SINGLE	(4<<0)	/* single channel */
+#define IEC958_AES1_PRO_MODE_TWO	(8<<0)	/* two channels */
+#define IEC958_AES1_PRO_MODE_PRIMARY	(12<<0)	/* primary/secondary */
+#define IEC958_AES1_PRO_MODE_BYTE3	(15<<0)	/* vector to byte 3 */
+#define IEC958_AES1_PRO_USERBITS	(15<<4)	/* mask - user bits */
+#define IEC958_AES1_PRO_USERBITS_NOTID	(0<<4)	/* not indicated */
+#define IEC958_AES1_PRO_USERBITS_192	(8<<4)	/* 192-bit structure */
+#define IEC958_AES1_PRO_USERBITS_UDEF	(12<<4)	/* user defined application */
+#define IEC958_AES1_CON_CATEGORY	0x7f
+#define IEC958_AES1_CON_GENERAL		0x00
+#define IEC958_AES1_CON_LASEROPT_MASK	0x07
+#define IEC958_AES1_CON_LASEROPT_ID	0x01
+#define IEC958_AES1_CON_IEC908_CD	(IEC958_AES1_CON_LASEROPT_ID|0x00)
+#define IEC958_AES1_CON_NON_IEC908_CD	(IEC958_AES1_CON_LASEROPT_ID|0x08)
+#define IEC958_AES1_CON_MINI_DISC	(IEC958_AES1_CON_LASEROPT_ID|0x48)
+#define IEC958_AES1_CON_DVD		(IEC958_AES1_CON_LASEROPT_ID|0x18)
+#define IEC958_AES1_CON_LASTEROPT_OTHER	(IEC958_AES1_CON_LASEROPT_ID|0x78)
+#define IEC958_AES1_CON_DIGDIGCONV_MASK 0x07
+#define IEC958_AES1_CON_DIGDIGCONV_ID	0x02
+#define IEC958_AES1_CON_PCM_CODER	(IEC958_AES1_CON_DIGDIGCONV_ID|0x00)
+#define IEC958_AES1_CON_MIXER		(IEC958_AES1_CON_DIGDIGCONV_ID|0x10)
+#define IEC958_AES1_CON_RATE_CONVERTER	(IEC958_AES1_CON_DIGDIGCONV_ID|0x18)
+#define IEC958_AES1_CON_SAMPLER		(IEC958_AES1_CON_DIGDIGCONV_ID|0x20)
+#define IEC958_AES1_CON_DSP		(IEC958_AES1_CON_DIGDIGCONV_ID|0x28)
+#define IEC958_AES1_CON_DIGDIGCONV_OTHER (IEC958_AES1_CON_DIGDIGCONV_ID|0x78)
+#define IEC958_AES1_CON_MAGNETIC_MASK	0x07
+#define IEC958_AES1_CON_MAGNETIC_ID	0x03
+#define IEC958_AES1_CON_DAT		(IEC958_AES1_CON_MAGNETIC_ID|0x00)
+#define IEC958_AES1_CON_VCR		(IEC958_AES1_CON_MAGNETIC_ID|0x08)
+#define IEC958_AES1_CON_DCC		(IEC958_AES1_CON_MAGNETIC_ID|0x40)
+#define IEC958_AES1_CON_MAGNETIC_DISC	(IEC958_AES1_CON_MAGNETIC_ID|0x18)
+#define IEC958_AES1_CON_MAGNETIC_OTHER	(IEC958_AES1_CON_MAGNETIC_ID|0x78)
+#define IEC958_AES1_CON_BROADCAST1_MASK 0x07
+#define IEC958_AES1_CON_BROADCAST1_ID	0x04
+#define IEC958_AES1_CON_DAB_JAPAN	(IEC958_AES1_CON_BROADCAST1_ID|0x00)
+#define IEC958_AES1_CON_DAB_EUROPE	(IEC958_AES1_CON_BROADCAST1_ID|0x08)
+#define IEC958_AES1_CON_DAB_USA		(IEC958_AES1_CON_BROADCAST1_ID|0x60)
+#define IEC958_AES1_CON_SOFTWARE	(IEC958_AES1_CON_BROADCAST1_ID|0x40)
+#define IEC958_AES1_CON_IEC62105	(IEC958_AES1_CON_BROADCAST1_ID|0x20)
+#define IEC958_AES1_CON_BROADCAST1_OTHER (IEC958_AES1_CON_BROADCAST1_ID|0x78)
+#define IEC958_AES1_CON_BROADCAST2_MASK 0x0f
+#define IEC958_AES1_CON_BROADCAST2_ID	0x0e
+#define IEC958_AES1_CON_MUSICAL_MASK	0x07
+#define IEC958_AES1_CON_MUSICAL_ID	0x05
+#define IEC958_AES1_CON_SYNTHESIZER	(IEC958_AES1_CON_MUSICAL_ID|0x00)
+#define IEC958_AES1_CON_MICROPHONE	(IEC958_AES1_CON_MUSICAL_ID|0x08)
+#define IEC958_AES1_CON_MUSICAL_OTHER	(IEC958_AES1_CON_MUSICAL_ID|0x78)
+#define IEC958_AES1_CON_ADC_MASK	0x1f
+#define IEC958_AES1_CON_ADC_ID		0x06
+#define IEC958_AES1_CON_ADC		(IEC958_AES1_CON_ADC_ID|0x00)
+#define IEC958_AES1_CON_ADC_OTHER	(IEC958_AES1_CON_ADC_ID|0x60)
+#define IEC958_AES1_CON_ADC_COPYRIGHT_MASK 0x1f
+#define IEC958_AES1_CON_ADC_COPYRIGHT_ID 0x16
+#define IEC958_AES1_CON_ADC_COPYRIGHT	(IEC958_AES1_CON_ADC_COPYRIGHT_ID|0x00)
+#define IEC958_AES1_CON_ADC_COPYRIGHT_OTHER (IEC958_AES1_CON_ADC_COPYRIGHT_ID|0x60)
+#define IEC958_AES1_CON_SOLIDMEM_MASK	0x0f
+#define IEC958_AES1_CON_SOLIDMEM_ID	0x08
+#define IEC958_AES1_CON_SOLIDMEM_DIGITAL_RECORDER_PLAYER (IEC958_AES1_CON_SOLIDMEM_ID|0x00)
+#define IEC958_AES1_CON_SOLIDMEM_OTHER	(IEC958_AES1_CON_SOLIDMEM_ID|0x70)
+#define IEC958_AES1_CON_EXPERIMENTAL	0x40
+#define IEC958_AES1_CON_ORIGINAL	(1<<7)	/* this bits depends on the category code */
+#define IEC958_AES2_PRO_SBITS		(7<<0)	/* mask - sample bits */
+#define IEC958_AES2_PRO_SBITS_20	(2<<0)	/* 20-bit - coordination */
+#define IEC958_AES2_PRO_SBITS_24	(4<<0)	/* 24-bit - main audio */
+#define IEC958_AES2_PRO_SBITS_UDEF	(6<<0)	/* user defined application */
+#define IEC958_AES2_PRO_WORDLEN		(7<<3)	/* mask - source word length */
+#define IEC958_AES2_PRO_WORDLEN_NOTID	(0<<3)	/* not indicated */
+#define IEC958_AES2_PRO_WORDLEN_22_18	(2<<3)	/* 22-bit or 18-bit */
+#define IEC958_AES2_PRO_WORDLEN_23_19	(4<<3)	/* 23-bit or 19-bit */
+#define IEC958_AES2_PRO_WORDLEN_24_20	(5<<3)	/* 24-bit or 20-bit */
+#define IEC958_AES2_PRO_WORDLEN_20_16	(6<<3)	/* 20-bit or 16-bit */
+#define IEC958_AES2_CON_SOURCE		(15<<0)	/* mask - source number */
+#define IEC958_AES2_CON_SOURCE_UNSPEC	(0<<0)	/* unspecified */
+#define IEC958_AES2_CON_CHANNEL		(15<<4)	/* mask - channel number */
+#define IEC958_AES2_CON_CHANNEL_UNSPEC	(0<<4)	/* unspecified */
+#define IEC958_AES3_CON_FS		(15<<0)	/* mask - sample frequency */
+#define IEC958_AES3_CON_FS_44100	(0<<0)	/* 44.1kHz */
+#define IEC958_AES3_CON_FS_NOTID	(1<<0)	/* non indicated */
+#define IEC958_AES3_CON_FS_48000	(2<<0)	/* 48kHz */
+#define IEC958_AES3_CON_FS_32000	(3<<0)	/* 32kHz */
+#define IEC958_AES3_CON_FS_22050	(4<<0)	/* 22.05kHz */
+#define IEC958_AES3_CON_FS_24000	(6<<0)	/* 24kHz */
+#define IEC958_AES3_CON_FS_88200	(8<<0)	/* 88.2kHz */
+#define IEC958_AES3_CON_FS_768000	(9<<0)	/* 768kHz */
+#define IEC958_AES3_CON_FS_96000	(10<<0)	/* 96kHz */
+#define IEC958_AES3_CON_FS_176400	(12<<0)	/* 176.4kHz */
+#define IEC958_AES3_CON_FS_192000	(14<<0)	/* 192kHz */
+#define IEC958_AES3_CON_CLOCK		(3<<4)	/* mask - clock accuracy */
+#define IEC958_AES3_CON_CLOCK_1000PPM	(0<<4)	/* 1000 ppm */
+#define IEC958_AES3_CON_CLOCK_50PPM	(1<<4)	/* 50 ppm */
+#define IEC958_AES3_CON_CLOCK_VARIABLE	(2<<4)	/* variable pitch */
+#define IEC958_AES4_CON_MAX_WORDLEN_24	(1<<0)	/* 0 = 20-bit, 1 = 24-bit */
+#define IEC958_AES4_CON_WORDLEN		(7<<1)	/* mask - sample word length */
+#define IEC958_AES4_CON_WORDLEN_NOTID	(0<<1)	/* not indicated */
+#define IEC958_AES4_CON_WORDLEN_20_16	(1<<1)	/* 20-bit or 16-bit */
+#define IEC958_AES4_CON_WORDLEN_22_18	(2<<1)	/* 22-bit or 18-bit */
+#define IEC958_AES4_CON_WORDLEN_23_19	(4<<1)	/* 23-bit or 19-bit */
+#define IEC958_AES4_CON_WORDLEN_24_20	(5<<1)	/* 24-bit or 20-bit */
+#define IEC958_AES4_CON_WORDLEN_21_17	(6<<1)	/* 21-bit or 17-bit */
+#define IEC958_AES4_CON_ORIGFS		(15<<4)	/* mask - original sample frequency */
+#define IEC958_AES4_CON_ORIGFS_NOTID	(0<<4)	/* not indicated */
+#define IEC958_AES4_CON_ORIGFS_192000	(1<<4)	/* 192kHz */
+#define IEC958_AES4_CON_ORIGFS_12000	(2<<4)	/* 12kHz */
+#define IEC958_AES4_CON_ORIGFS_176400	(3<<4)	/* 176.4kHz */
+#define IEC958_AES4_CON_ORIGFS_96000	(5<<4)	/* 96kHz */
+#define IEC958_AES4_CON_ORIGFS_8000	(6<<4)	/* 8kHz */
+#define IEC958_AES4_CON_ORIGFS_88200	(7<<4)	/* 88.2kHz */
+#define IEC958_AES4_CON_ORIGFS_16000	(8<<4)	/* 16kHz */
+#define IEC958_AES4_CON_ORIGFS_24000	(9<<4)	/* 24kHz */
+#define IEC958_AES4_CON_ORIGFS_11025	(10<<4)	/* 11.025kHz */
+#define IEC958_AES4_CON_ORIGFS_22050	(11<<4)	/* 22.05kHz */
+#define IEC958_AES4_CON_ORIGFS_32000	(12<<4)	/* 32kHz */
+#define IEC958_AES4_CON_ORIGFS_48000	(13<<4)	/* 48kHz */
+#define IEC958_AES4_CON_ORIGFS_44100	(15<<4)	/* 44.1kHz */
+#define IEC958_AES5_CON_CGMSA		(3<<0)	/* mask - CGMS-A */
+#define IEC958_AES5_CON_CGMSA_COPYFREELY (0<<0)	/* copying is permitted without restriction */
+#define IEC958_AES5_CON_CGMSA_COPYONCE	(1<<0)	/* one generation of copies may be made */
+#define IEC958_AES5_CON_CGMSA_COPYNOMORE (2<<0)	/* condition not be used */
+#define IEC958_AES5_CON_CGMSA_COPYNEVER	(3<<0)	/* no copying is permitted */
+
+/*****************************************************************************
+ *                                                                           *
+ *                            MIDI v1.0 interface                            *
+ *                                                                           *
+ *****************************************************************************/
+
+#define MIDI_CHANNELS			16
+#define MIDI_GM_DRUM_CHANNEL		(10-1)
+
+/*
+ *  MIDI commands
+ */
+
+#define MIDI_CMD_NOTE_OFF		0x80
+#define MIDI_CMD_NOTE_ON		0x90
+#define MIDI_CMD_NOTE_PRESSURE		0xa0
+#define MIDI_CMD_CONTROL		0xb0
+#define MIDI_CMD_PGM_CHANGE		0xc0
+#define MIDI_CMD_CHANNEL_PRESSURE	0xd0
+#define MIDI_CMD_BENDER			0xe0
+
+#define MIDI_CMD_COMMON_SYSEX		0xf0
+#define MIDI_CMD_COMMON_MTC_QUARTER	0xf1
+#define MIDI_CMD_COMMON_SONG_POS	0xf2
+#define MIDI_CMD_COMMON_SONG_SELECT	0xf3
+#define MIDI_CMD_COMMON_TUNE_REQUEST	0xf6
+#define MIDI_CMD_COMMON_SYSEX_END	0xf7
+#define MIDI_CMD_COMMON_CLOCK		0xf8
+#define MIDI_CMD_COMMON_START		0xfa
+#define MIDI_CMD_COMMON_CONTINUE	0xfb
+#define MIDI_CMD_COMMON_STOP		0xfc
+#define MIDI_CMD_COMMON_SENSING		0xfe
+#define MIDI_CMD_COMMON_RESET		0xff
+
+/*
+ *  MIDI controllers
+ */
+
+#define MIDI_CTL_MSB_BANK		0x00
+#define MIDI_CTL_MSB_MODWHEEL         	0x01
+#define MIDI_CTL_MSB_BREATH           	0x02
+#define MIDI_CTL_MSB_FOOT             	0x04
+#define MIDI_CTL_MSB_PORTAMENTO_TIME 	0x05
+#define MIDI_CTL_MSB_DATA_ENTRY		0x06
+#define MIDI_CTL_MSB_MAIN_VOLUME      	0x07
+#define MIDI_CTL_MSB_BALANCE          	0x08
+#define MIDI_CTL_MSB_PAN              	0x0a
+#define MIDI_CTL_MSB_EXPRESSION       	0x0b
+#define MIDI_CTL_MSB_EFFECT1		0x0c
+#define MIDI_CTL_MSB_EFFECT2		0x0d
+#define MIDI_CTL_MSB_GENERAL_PURPOSE1 	0x10
+#define MIDI_CTL_MSB_GENERAL_PURPOSE2 	0x11
+#define MIDI_CTL_MSB_GENERAL_PURPOSE3 	0x12
+#define MIDI_CTL_MSB_GENERAL_PURPOSE4 	0x13
+#define MIDI_CTL_LSB_BANK		0x20
+#define MIDI_CTL_LSB_MODWHEEL        	0x21
+#define MIDI_CTL_LSB_BREATH           	0x22
+#define MIDI_CTL_LSB_FOOT             	0x24
+#define MIDI_CTL_LSB_PORTAMENTO_TIME 	0x25
+#define MIDI_CTL_LSB_DATA_ENTRY		0x26
+#define MIDI_CTL_LSB_MAIN_VOLUME      	0x27
+#define MIDI_CTL_LSB_BALANCE          	0x28
+#define MIDI_CTL_LSB_PAN              	0x2a
+#define MIDI_CTL_LSB_EXPRESSION       	0x2b
+#define MIDI_CTL_LSB_EFFECT1		0x2c
+#define MIDI_CTL_LSB_EFFECT2		0x2d
+#define MIDI_CTL_LSB_GENERAL_PURPOSE1 	0x30
+#define MIDI_CTL_LSB_GENERAL_PURPOSE2 	0x31
+#define MIDI_CTL_LSB_GENERAL_PURPOSE3 	0x32
+#define MIDI_CTL_LSB_GENERAL_PURPOSE4 	0x33
+#define MIDI_CTL_SUSTAIN              	0x40
+#define MIDI_CTL_PORTAMENTO           	0x41
+#define MIDI_CTL_SOSTENUTO            	0x42
+#define MIDI_CTL_SOFT_PEDAL           	0x43
+#define MIDI_CTL_LEGATO_FOOTSWITCH	0x44
+#define MIDI_CTL_HOLD2                	0x45
+#define MIDI_CTL_SC1_SOUND_VARIATION	0x46
+#define MIDI_CTL_SC2_TIMBRE		0x47
+#define MIDI_CTL_SC3_RELEASE_TIME	0x48
+#define MIDI_CTL_SC4_ATTACK_TIME	0x49
+#define MIDI_CTL_SC5_BRIGHTNESS		0x4a
+#define MIDI_CTL_SC6			0x4b
+#define MIDI_CTL_SC7			0x4c
+#define MIDI_CTL_SC8			0x4d
+#define MIDI_CTL_SC9			0x4e
+#define MIDI_CTL_SC10			0x4f
+#define MIDI_CTL_GENERAL_PURPOSE5     	0x50
+#define MIDI_CTL_GENERAL_PURPOSE6     	0x51
+#define MIDI_CTL_GENERAL_PURPOSE7     	0x52
+#define MIDI_CTL_GENERAL_PURPOSE8     	0x53
+#define MIDI_CTL_PORTAMENTO_CONTROL	0x54
+#define MIDI_CTL_E1_REVERB_DEPTH	0x5b
+#define MIDI_CTL_E2_TREMOLO_DEPTH	0x5c
+#define MIDI_CTL_E3_CHORUS_DEPTH	0x5d
+#define MIDI_CTL_E4_DETUNE_DEPTH	0x5e
+#define MIDI_CTL_E5_PHASER_DEPTH	0x5f
+#define MIDI_CTL_DATA_INCREMENT       	0x60
+#define MIDI_CTL_DATA_DECREMENT       	0x61
+#define MIDI_CTL_NONREG_PARM_NUM_LSB  	0x62
+#define MIDI_CTL_NONREG_PARM_NUM_MSB  	0x63
+#define MIDI_CTL_REGIST_PARM_NUM_LSB  	0x64
+#define MIDI_CTL_REGIST_PARM_NUM_MSB	0x65
+#define MIDI_CTL_ALL_SOUNDS_OFF		0x78
+#define MIDI_CTL_RESET_CONTROLLERS	0x79
+#define MIDI_CTL_LOCAL_CONTROL_SWITCH	0x7a
+#define MIDI_CTL_ALL_NOTES_OFF		0x7b
+#define MIDI_CTL_OMNI_OFF		0x7c
+#define MIDI_CTL_OMNI_ON		0x7d
+#define MIDI_CTL_MONO1			0x7e
+#define MIDI_CTL_MONO2			0x7f
+
+#endif /* __SOUND_ASOUNDEF_H */
diff --git a/include/sound/atmel-abdac.h b/include/sound/atmel-abdac.h
new file mode 100644
index 0000000..edff6a8
--- /dev/null
+++ b/include/sound/atmel-abdac.h
@@ -0,0 +1,23 @@
+/*
+ * Driver for the Atmel Audio Bitstream DAC (ABDAC)
+ *
+ * Copyright (C) 2009 Atmel Corporation
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ */
+#ifndef __INCLUDE_SOUND_ATMEL_ABDAC_H
+#define __INCLUDE_SOUND_ATMEL_ABDAC_H
+
+#include <linux/dw_dmac.h>
+
+/**
+ * struct atmel_abdac_pdata - board specific ABDAC configuration
+ * @dws: DMA slave interface to use for sound playback.
+ */
+struct atmel_abdac_pdata {
+	struct dw_dma_slave	dws;
+};
+
+#endif /* __INCLUDE_SOUND_ATMEL_ABDAC_H */
diff --git a/include/sound/atmel-ac97c.h b/include/sound/atmel-ac97c.h
new file mode 100644
index 0000000..e6aabdb
--- /dev/null
+++ b/include/sound/atmel-ac97c.h
@@ -0,0 +1,40 @@
+/*
+ * Driver for the Atmel AC97C controller
+ *
+ * Copyright (C) 2005-2009 Atmel Corporation
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ */
+#ifndef __INCLUDE_SOUND_ATMEL_AC97C_H
+#define __INCLUDE_SOUND_ATMEL_AC97C_H
+
+#include <linux/dw_dmac.h>
+
+#define AC97C_CAPTURE	0x01
+#define AC97C_PLAYBACK	0x02
+#define AC97C_BOTH	(AC97C_CAPTURE | AC97C_PLAYBACK)
+
+/**
+ * struct atmel_ac97c_pdata - board specific AC97C configuration
+ * @rx_dws: DMA slave interface to use for sound capture.
+ * @tx_dws: DMA slave interface to use for sound playback.
+ * @reset_pin: GPIO pin wired to the reset input on the external AC97 codec,
+ *             optional to use, set to -ENODEV if not in use. AC97 layer will
+ *             try to do a software reset of the external codec anyway.
+ * @flags: Flags for which directions should be enabled.
+ *
+ * If the user do not want to use a DMA channel for playback or capture, i.e.
+ * only one feature is required on the board. The slave for playback or capture
+ * can be set to NULL. The AC97C driver will take use of this when setting up
+ * the sound streams.
+ */
+struct ac97c_platform_data {
+	struct dw_dma_slave	rx_dws;
+	struct dw_dma_slave	tx_dws;
+	unsigned int 		flags;
+	int			reset_pin;
+};
+
+#endif /* __INCLUDE_SOUND_ATMEL_AC97C_H */
diff --git a/include/sound/control.h b/include/sound/control.h
new file mode 100644
index 0000000..112374d
--- /dev/null
+++ b/include/sound/control.h
@@ -0,0 +1,225 @@
+#ifndef __SOUND_CONTROL_H
+#define __SOUND_CONTROL_H
+
+/*
+ *  Header file for control interface
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ *   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., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ */
+
+#include <sound/asound.h>
+
+#define snd_kcontrol_chip(kcontrol) ((kcontrol)->private_data)
+
+struct snd_kcontrol;
+typedef int (snd_kcontrol_info_t) (struct snd_kcontrol * kcontrol, struct snd_ctl_elem_info * uinfo);
+typedef int (snd_kcontrol_get_t) (struct snd_kcontrol * kcontrol, struct snd_ctl_elem_value * ucontrol);
+typedef int (snd_kcontrol_put_t) (struct snd_kcontrol * kcontrol, struct snd_ctl_elem_value * ucontrol);
+typedef int (snd_kcontrol_tlv_rw_t)(struct snd_kcontrol *kcontrol,
+				    int op_flag, /* 0=read,1=write,-1=command */
+				    unsigned int size,
+				    unsigned int __user *tlv);
+
+
+struct snd_kcontrol_new {
+	snd_ctl_elem_iface_t iface;	/* interface identifier */
+	unsigned int device;		/* device/client number */
+	unsigned int subdevice;		/* subdevice (substream) number */
+	unsigned char *name;		/* ASCII name of item */
+	unsigned int index;		/* index of item */
+	unsigned int access;		/* access rights */
+	unsigned int count;		/* count of same elements */
+	snd_kcontrol_info_t *info;
+	snd_kcontrol_get_t *get;
+	snd_kcontrol_put_t *put;
+	union {
+		snd_kcontrol_tlv_rw_t *c;
+		const unsigned int *p;
+	} tlv;
+	unsigned long private_value;
+};
+
+struct snd_kcontrol_volatile {
+	struct snd_ctl_file *owner;	/* locked */
+	unsigned int access;	/* access rights */
+};
+
+struct snd_kcontrol {
+	struct list_head list;		/* list of controls */
+	struct snd_ctl_elem_id id;
+	unsigned int count;		/* count of same elements */
+	snd_kcontrol_info_t *info;
+	snd_kcontrol_get_t *get;
+	snd_kcontrol_put_t *put;
+	union {
+		snd_kcontrol_tlv_rw_t *c;
+		const unsigned int *p;
+	} tlv;
+	unsigned long private_value;
+	void *private_data;
+	void (*private_free)(struct snd_kcontrol *kcontrol);
+	struct snd_kcontrol_volatile vd[0];	/* volatile data */
+};
+
+#define snd_kcontrol(n) list_entry(n, struct snd_kcontrol, list)
+
+struct snd_kctl_event {
+	struct list_head list;	/* list of events */
+	struct snd_ctl_elem_id id;
+	unsigned int mask;
+};
+
+#define snd_kctl_event(n) list_entry(n, struct snd_kctl_event, list)
+
+struct pid;
+
+struct snd_ctl_file {
+	struct list_head list;		/* list of all control files */
+	struct snd_card *card;
+	struct pid *pid;
+	int prefer_pcm_subdevice;
+	int prefer_rawmidi_subdevice;
+	wait_queue_head_t change_sleep;
+	spinlock_t read_lock;
+	struct fasync_struct *fasync;
+	int subscribed;			/* read interface is activated */
+	struct list_head events;	/* waiting events for read */
+};
+
+#define snd_ctl_file(n) list_entry(n, struct snd_ctl_file, list)
+
+typedef int (*snd_kctl_ioctl_func_t) (struct snd_card * card,
+				      struct snd_ctl_file * control,
+				      unsigned int cmd, unsigned long arg);
+
+void snd_ctl_notify(struct snd_card * card, unsigned int mask, struct snd_ctl_elem_id * id);
+
+struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new * kcontrolnew, void * private_data);
+void snd_ctl_free_one(struct snd_kcontrol * kcontrol);
+int snd_ctl_add(struct snd_card * card, struct snd_kcontrol * kcontrol);
+int snd_ctl_remove(struct snd_card * card, struct snd_kcontrol * kcontrol);
+int snd_ctl_remove_id(struct snd_card * card, struct snd_ctl_elem_id *id);
+int snd_ctl_rename_id(struct snd_card * card, struct snd_ctl_elem_id *src_id, struct snd_ctl_elem_id *dst_id);
+struct snd_kcontrol *snd_ctl_find_numid(struct snd_card * card, unsigned int numid);
+struct snd_kcontrol *snd_ctl_find_id(struct snd_card * card, struct snd_ctl_elem_id *id);
+
+int snd_ctl_create(struct snd_card *card);
+
+int snd_ctl_register_ioctl(snd_kctl_ioctl_func_t fcn);
+int snd_ctl_unregister_ioctl(snd_kctl_ioctl_func_t fcn);
+#ifdef CONFIG_COMPAT
+int snd_ctl_register_ioctl_compat(snd_kctl_ioctl_func_t fcn);
+int snd_ctl_unregister_ioctl_compat(snd_kctl_ioctl_func_t fcn);
+#else
+#define snd_ctl_register_ioctl_compat(fcn)
+#define snd_ctl_unregister_ioctl_compat(fcn)
+#endif
+
+static inline unsigned int snd_ctl_get_ioffnum(struct snd_kcontrol *kctl, struct snd_ctl_elem_id *id)
+{
+	return id->numid - kctl->id.numid;
+}
+
+static inline unsigned int snd_ctl_get_ioffidx(struct snd_kcontrol *kctl, struct snd_ctl_elem_id *id)
+{
+	return id->index - kctl->id.index;
+}
+
+static inline unsigned int snd_ctl_get_ioff(struct snd_kcontrol *kctl, struct snd_ctl_elem_id *id)
+{
+	if (id->numid) {
+		return snd_ctl_get_ioffnum(kctl, id);
+	} else {
+		return snd_ctl_get_ioffidx(kctl, id);
+	}
+}
+
+static inline struct snd_ctl_elem_id *snd_ctl_build_ioff(struct snd_ctl_elem_id *dst_id,
+						    struct snd_kcontrol *src_kctl,
+						    unsigned int offset)
+{
+	*dst_id = src_kctl->id;
+	dst_id->index += offset;
+	dst_id->numid += offset;
+	return dst_id;
+}
+
+/*
+ * Frequently used control callbacks
+ */
+int snd_ctl_boolean_mono_info(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_info *uinfo);
+int snd_ctl_boolean_stereo_info(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_info *uinfo);
+
+/*
+ * virtual master control
+ */
+struct snd_kcontrol *snd_ctl_make_virtual_master(char *name,
+						 const unsigned int *tlv);
+int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave,
+		       unsigned int flags);
+/* optional flags for slave */
+#define SND_CTL_SLAVE_NEED_UPDATE	(1 << 0)
+
+/**
+ * snd_ctl_add_slave - Add a virtual slave control
+ * @master: vmaster element
+ * @slave: slave element to add
+ *
+ * Add a virtual slave control to the given master element created via
+ * snd_ctl_create_virtual_master() beforehand.
+ * Returns zero if successful or a negative error code.
+ *
+ * All slaves must be the same type (returning the same information
+ * via info callback).  The fucntion doesn't check it, so it's your
+ * responsibility.
+ *
+ * Also, some additional limitations:
+ * at most two channels,
+ * logarithmic volume control (dB level) thus no linear volume,
+ * master can only attenuate the volume without gain
+ */
+static inline int
+snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave)
+{
+	return _snd_ctl_add_slave(master, slave, 0);
+}
+
+/**
+ * snd_ctl_add_slave_uncached - Add a virtual slave control
+ * @master: vmaster element
+ * @slave: slave element to add
+ *
+ * Add a virtual slave control to the given master.
+ * Unlike snd_ctl_add_slave(), the element added via this function
+ * is supposed to have volatile values, and get callback is called
+ * at each time quried from the master.
+ *
+ * When the control peeks the hardware values directly and the value
+ * can be changed by other means than the put callback of the element,
+ * this function should be used to keep the value always up-to-date.
+ */
+static inline int
+snd_ctl_add_slave_uncached(struct snd_kcontrol *master,
+			   struct snd_kcontrol *slave)
+{
+	return _snd_ctl_add_slave(master, slave, SND_CTL_SLAVE_NEED_UPDATE);
+}
+
+#endif	/* __SOUND_CONTROL_H */
diff --git a/include/sound/core.h b/include/sound/core.h
new file mode 100644
index 0000000..89e0ac1
--- /dev/null
+++ b/include/sound/core.h
@@ -0,0 +1,465 @@
+#ifndef __SOUND_CORE_H
+#define __SOUND_CORE_H
+
+/*
+ *  Main header file for the ALSA driver
+ *  Copyright (c) 1994-2001 by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ *   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., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/sched.h>		/* wake_up() */
+#include <linux/mutex.h>		/* struct mutex */
+#include <linux/rwsem.h>		/* struct rw_semaphore */
+#include <linux/pm.h>			/* pm_message_t */
+#include <linux/device.h>
+#include <linux/stringify.h>
+
+/* number of supported soundcards */
+#ifdef CONFIG_SND_DYNAMIC_MINORS
+#define SNDRV_CARDS 32
+#else
+#define SNDRV_CARDS 8		/* don't change - minor numbers */
+#endif
+
+#define CONFIG_SND_MAJOR	116	/* standard configuration */
+
+/* forward declarations */
+#ifdef CONFIG_PCI
+struct pci_dev;
+#endif
+
+/* device allocation stuff */
+
+#define SNDRV_DEV_TYPE_RANGE_SIZE		0x1000
+
+typedef int __bitwise snd_device_type_t;
+#define	SNDRV_DEV_TOPLEVEL	((__force snd_device_type_t) 0)
+#define	SNDRV_DEV_CONTROL	((__force snd_device_type_t) 1)
+#define	SNDRV_DEV_LOWLEVEL_PRE	((__force snd_device_type_t) 2)
+#define	SNDRV_DEV_LOWLEVEL_NORMAL ((__force snd_device_type_t) 0x1000)
+#define	SNDRV_DEV_PCM		((__force snd_device_type_t) 0x1001)
+#define	SNDRV_DEV_RAWMIDI	((__force snd_device_type_t) 0x1002)
+#define	SNDRV_DEV_TIMER		((__force snd_device_type_t) 0x1003)
+#define	SNDRV_DEV_SEQUENCER	((__force snd_device_type_t) 0x1004)
+#define	SNDRV_DEV_HWDEP		((__force snd_device_type_t) 0x1005)
+#define	SNDRV_DEV_INFO		((__force snd_device_type_t) 0x1006)
+#define	SNDRV_DEV_BUS		((__force snd_device_type_t) 0x1007)
+#define	SNDRV_DEV_CODEC		((__force snd_device_type_t) 0x1008)
+#define	SNDRV_DEV_JACK          ((__force snd_device_type_t) 0x1009)
+#define	SNDRV_DEV_LOWLEVEL	((__force snd_device_type_t) 0x2000)
+
+typedef int __bitwise snd_device_state_t;
+#define	SNDRV_DEV_BUILD		((__force snd_device_state_t) 0)
+#define	SNDRV_DEV_REGISTERED	((__force snd_device_state_t) 1)
+#define	SNDRV_DEV_DISCONNECTED	((__force snd_device_state_t) 2)
+
+typedef int __bitwise snd_device_cmd_t;
+#define	SNDRV_DEV_CMD_PRE	((__force snd_device_cmd_t) 0)
+#define	SNDRV_DEV_CMD_NORMAL	((__force snd_device_cmd_t) 1)	
+#define	SNDRV_DEV_CMD_POST	((__force snd_device_cmd_t) 2)
+
+struct snd_device;
+
+struct snd_device_ops {
+	int (*dev_free)(struct snd_device *dev);
+	int (*dev_register)(struct snd_device *dev);
+	int (*dev_disconnect)(struct snd_device *dev);
+};
+
+struct snd_device {
+	struct list_head list;		/* list of registered devices */
+	struct snd_card *card;		/* card which holds this device */
+	snd_device_state_t state;	/* state of the device */
+	snd_device_type_t type;		/* device type */
+	void *device_data;		/* device structure */
+	struct snd_device_ops *ops;	/* operations */
+};
+
+#define snd_device(n) list_entry(n, struct snd_device, list)
+
+/* main structure for soundcard */
+
+struct snd_card {
+	int number;			/* number of soundcard (index to
+								snd_cards) */
+
+	char id[16];			/* id string of this card */
+	char driver[16];		/* driver name */
+	char shortname[32];		/* short name of this soundcard */
+	char longname[80];		/* name of this soundcard */
+	char mixername[80];		/* mixer name */
+	char components[128];		/* card components delimited with
+								space */
+	struct module *module;		/* top-level module */
+
+	void *private_data;		/* private data for soundcard */
+	void (*private_free) (struct snd_card *card); /* callback for freeing of
+								private data */
+	struct list_head devices;	/* devices */
+
+	unsigned int last_numid;	/* last used numeric ID */
+	struct rw_semaphore controls_rwsem;	/* controls list lock */
+	rwlock_t ctl_files_rwlock;	/* ctl_files list lock */
+	int controls_count;		/* count of all controls */
+	int user_ctl_count;		/* count of all user controls */
+	struct list_head controls;	/* all controls for this card */
+	struct list_head ctl_files;	/* active control files */
+
+	struct snd_info_entry *proc_root;	/* root for soundcard specific files */
+	struct snd_info_entry *proc_id;	/* the card id */
+	struct proc_dir_entry *proc_root_link;	/* number link to real id */
+
+	struct list_head files_list;	/* all files associated to this card */
+	struct snd_shutdown_f_ops *s_f_ops; /* file operations in the shutdown
+								state */
+	spinlock_t files_lock;		/* lock the files for this card */
+	int shutdown;			/* this card is going down */
+	int free_on_last_close;		/* free in context of file_release */
+	wait_queue_head_t shutdown_sleep;
+	struct device *dev;		/* device assigned to this card */
+#ifndef CONFIG_SYSFS_DEPRECATED
+	struct device *card_dev;	/* cardX object for sysfs */
+#endif
+
+#ifdef CONFIG_PM
+	unsigned int power_state;	/* power state */
+	struct mutex power_lock;	/* power lock */
+	wait_queue_head_t power_sleep;
+#endif
+
+#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE)
+	struct snd_mixer_oss *mixer_oss;
+	int mixer_oss_change_count;
+#endif
+};
+
+#ifdef CONFIG_PM
+static inline void snd_power_lock(struct snd_card *card)
+{
+	mutex_lock(&card->power_lock);
+}
+
+static inline void snd_power_unlock(struct snd_card *card)
+{
+	mutex_unlock(&card->power_lock);
+}
+
+static inline unsigned int snd_power_get_state(struct snd_card *card)
+{
+	return card->power_state;
+}
+
+static inline void snd_power_change_state(struct snd_card *card, unsigned int state)
+{
+	card->power_state = state;
+	wake_up(&card->power_sleep);
+}
+
+/* init.c */
+int snd_power_wait(struct snd_card *card, unsigned int power_state);
+
+#else /* ! CONFIG_PM */
+
+#define snd_power_lock(card)		do { (void)(card); } while (0)
+#define snd_power_unlock(card)		do { (void)(card); } while (0)
+static inline int snd_power_wait(struct snd_card *card, unsigned int state) { return 0; }
+#define snd_power_get_state(card)	SNDRV_CTL_POWER_D0
+#define snd_power_change_state(card, state)	do { (void)(card); } while (0)
+
+#endif /* CONFIG_PM */
+
+struct snd_minor {
+	int type;			/* SNDRV_DEVICE_TYPE_XXX */
+	int card;			/* card number */
+	int device;			/* device number */
+	const struct file_operations *f_ops;	/* file operations */
+	void *private_data;		/* private data for f_ops->open */
+	struct device *dev;		/* device for sysfs */
+};
+
+/* return a device pointer linked to each sound device as a parent */
+static inline struct device *snd_card_get_device_link(struct snd_card *card)
+{
+#ifdef CONFIG_SYSFS_DEPRECATED
+	return card ? card->dev : NULL;
+#else
+	return card ? card->card_dev : NULL;
+#endif
+}
+
+/* sound.c */
+
+extern int snd_major;
+extern int snd_ecards_limit;
+extern struct class *sound_class;
+
+void snd_request_card(int card);
+
+int snd_register_device_for_dev(int type, struct snd_card *card,
+				int dev,
+				const struct file_operations *f_ops,
+				void *private_data,
+				const char *name,
+				struct device *device);
+
+/**
+ * snd_register_device - Register the ALSA device file for the card
+ * @type: the device type, SNDRV_DEVICE_TYPE_XXX
+ * @card: the card instance
+ * @dev: the device index
+ * @f_ops: the file operations
+ * @private_data: user pointer for f_ops->open()
+ * @name: the device file name
+ *
+ * Registers an ALSA device file for the given card.
+ * The operators have to be set in reg parameter.
+ *
+ * This function uses the card's device pointer to link to the
+ * correct &struct device.
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ */
+static inline int snd_register_device(int type, struct snd_card *card, int dev,
+				      const struct file_operations *f_ops,
+				      void *private_data,
+				      const char *name)
+{
+	return snd_register_device_for_dev(type, card, dev, f_ops,
+					   private_data, name,
+					   snd_card_get_device_link(card));
+}
+
+int snd_unregister_device(int type, struct snd_card *card, int dev);
+void *snd_lookup_minor_data(unsigned int minor, int type);
+int snd_add_device_sysfs_file(int type, struct snd_card *card, int dev,
+			      struct device_attribute *attr);
+
+#ifdef CONFIG_SND_OSSEMUL
+int snd_register_oss_device(int type, struct snd_card *card, int dev,
+			    const struct file_operations *f_ops, void *private_data,
+			    const char *name);
+int snd_unregister_oss_device(int type, struct snd_card *card, int dev);
+void *snd_lookup_oss_minor_data(unsigned int minor, int type);
+#endif
+
+int snd_minor_info_init(void);
+int snd_minor_info_done(void);
+
+/* sound_oss.c */
+
+#ifdef CONFIG_SND_OSSEMUL
+int snd_minor_info_oss_init(void);
+int snd_minor_info_oss_done(void);
+#else
+static inline int snd_minor_info_oss_init(void) { return 0; }
+static inline int snd_minor_info_oss_done(void) { return 0; }
+#endif
+
+/* memory.c */
+
+int copy_to_user_fromio(void __user *dst, const volatile void __iomem *src, size_t count);
+int copy_from_user_toio(volatile void __iomem *dst, const void __user *src, size_t count);
+
+/* init.c */
+
+extern struct snd_card *snd_cards[SNDRV_CARDS];
+int snd_card_locked(int card);
+#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE)
+#define SND_MIXER_OSS_NOTIFY_REGISTER	0
+#define SND_MIXER_OSS_NOTIFY_DISCONNECT	1
+#define SND_MIXER_OSS_NOTIFY_FREE	2
+extern int (*snd_mixer_oss_notify_callback)(struct snd_card *card, int cmd);
+#endif
+
+int snd_card_create(int idx, const char *id,
+		    struct module *module, int extra_size,
+		    struct snd_card **card_ret);
+
+int snd_card_disconnect(struct snd_card *card);
+int snd_card_free(struct snd_card *card);
+int snd_card_free_when_closed(struct snd_card *card);
+void snd_card_set_id(struct snd_card *card, const char *id);
+int snd_card_register(struct snd_card *card);
+int snd_card_info_init(void);
+int snd_card_info_done(void);
+int snd_component_add(struct snd_card *card, const char *component);
+int snd_card_file_add(struct snd_card *card, struct file *file);
+int snd_card_file_remove(struct snd_card *card, struct file *file);
+
+#define snd_card_set_dev(card, devptr) ((card)->dev = (devptr))
+
+/* device.c */
+
+int snd_device_new(struct snd_card *card, snd_device_type_t type,
+		   void *device_data, struct snd_device_ops *ops);
+int snd_device_register(struct snd_card *card, void *device_data);
+int snd_device_register_all(struct snd_card *card);
+int snd_device_disconnect(struct snd_card *card, void *device_data);
+int snd_device_disconnect_all(struct snd_card *card);
+int snd_device_free(struct snd_card *card, void *device_data);
+int snd_device_free_all(struct snd_card *card, snd_device_cmd_t cmd);
+
+/* isadma.c */
+
+#ifdef CONFIG_ISA_DMA_API
+#define DMA_MODE_NO_ENABLE	0x0100
+
+void snd_dma_program(unsigned long dma, unsigned long addr, unsigned int size, unsigned short mode);
+void snd_dma_disable(unsigned long dma);
+unsigned int snd_dma_pointer(unsigned long dma, unsigned int size);
+#endif
+
+/* misc.c */
+struct resource;
+void release_and_free_resource(struct resource *res);
+
+/* --- */
+
+#if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK)
+void __snd_printk(unsigned int level, const char *file, int line,
+		  const char *format, ...)
+     __attribute__ ((format (printf, 4, 5)));
+#else
+#define __snd_printk(level, file, line, format, args...) \
+	printk(format, ##args)
+#endif
+
+/**
+ * snd_printk - printk wrapper
+ * @fmt: format string
+ *
+ * Works like printk() but prints the file and the line of the caller
+ * when configured with CONFIG_SND_VERBOSE_PRINTK.
+ */
+#define snd_printk(fmt, args...) \
+	__snd_printk(0, __FILE__, __LINE__, fmt, ##args)
+
+#ifdef CONFIG_SND_DEBUG
+/**
+ * snd_printd - debug printk
+ * @fmt: format string
+ *
+ * Works like snd_printk() for debugging purposes.
+ * Ignored when CONFIG_SND_DEBUG is not set.
+ */
+#define snd_printd(fmt, args...) \
+	__snd_printk(1, __FILE__, __LINE__, fmt, ##args)
+
+/**
+ * snd_BUG - give a BUG warning message and stack trace
+ *
+ * Calls WARN() if CONFIG_SND_DEBUG is set.
+ * Ignored when CONFIG_SND_DEBUG is not set.
+ */
+#define snd_BUG()		WARN(1, "BUG?\n")
+
+/**
+ * snd_BUG_ON - debugging check macro
+ * @cond: condition to evaluate
+ *
+ * When CONFIG_SND_DEBUG is set, this macro evaluates the given condition,
+ * and call WARN() and returns the value if it's non-zero.
+ * 
+ * When CONFIG_SND_DEBUG is not set, this just returns zero, and the given
+ * condition is ignored.
+ *
+ * NOTE: the argument won't be evaluated at all when CONFIG_SND_DEBUG=n.
+ * Thus, don't put any statement that influences on the code behavior,
+ * such as pre/post increment, to the argument of this macro.
+ * If you want to evaluate and give a warning, use standard WARN_ON().
+ */
+#define snd_BUG_ON(cond)	WARN((cond), "BUG? (%s)\n", __stringify(cond))
+
+#else /* !CONFIG_SND_DEBUG */
+
+#define snd_printd(fmt, args...)	do { } while (0)
+#define snd_BUG()			do { } while (0)
+static inline int __snd_bug_on(int cond)
+{
+	return 0;
+}
+#define snd_BUG_ON(cond)	__snd_bug_on(0 && (cond))  /* always false */
+
+#endif /* CONFIG_SND_DEBUG */
+
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+/**
+ * snd_printdd - debug printk
+ * @format: format string
+ *
+ * Works like snd_printk() for debugging purposes.
+ * Ignored when CONFIG_SND_DEBUG_VERBOSE is not set.
+ */
+#define snd_printdd(format, args...) \
+	__snd_printk(2, __FILE__, __LINE__, format, ##args)
+#else
+#define snd_printdd(format, args...)	do { } while (0)
+#endif
+
+
+#define SNDRV_OSS_VERSION         ((3<<16)|(8<<8)|(1<<4)|(0))	/* 3.8.1a */
+
+/* for easier backward-porting */
+#if defined(CONFIG_GAMEPORT) || defined(CONFIG_GAMEPORT_MODULE)
+#define gameport_set_dev_parent(gp,xdev) ((gp)->dev.parent = (xdev))
+#define gameport_set_port_data(gp,r) ((gp)->port_data = (r))
+#define gameport_get_port_data(gp) (gp)->port_data
+#endif
+
+/* PCI quirk list helper */
+struct snd_pci_quirk {
+	unsigned short subvendor;	/* PCI subvendor ID */
+	unsigned short subdevice;	/* PCI subdevice ID */
+	unsigned short subdevice_mask;	/* bitmask to match */
+	int value;			/* value */
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+	const char *name;		/* name of the device (optional) */
+#endif
+};
+
+#define _SND_PCI_QUIRK_ID_MASK(vend, mask, dev)	\
+	.subvendor = (vend), .subdevice = (dev), .subdevice_mask = (mask)
+#define _SND_PCI_QUIRK_ID(vend, dev) \
+	_SND_PCI_QUIRK_ID_MASK(vend, 0xffff, dev)
+#define SND_PCI_QUIRK_ID(vend,dev) {_SND_PCI_QUIRK_ID(vend, dev)}
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+#define SND_PCI_QUIRK(vend,dev,xname,val) \
+	{_SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname)}
+#define SND_PCI_QUIRK_VENDOR(vend, xname, val)			\
+	{_SND_PCI_QUIRK_ID_MASK(vend, 0, 0), .value = (val), .name = (xname)}
+#define SND_PCI_QUIRK_MASK(vend, mask, dev, xname, val)			\
+	{_SND_PCI_QUIRK_ID_MASK(vend, mask, dev),			\
+			.value = (val), .name = (xname)}
+#else
+#define SND_PCI_QUIRK(vend,dev,xname,val) \
+	{_SND_PCI_QUIRK_ID(vend, dev), .value = (val)}
+#define SND_PCI_QUIRK_MASK(vend, mask, dev, xname, val)			\
+	{_SND_PCI_QUIRK_ID_MASK(vend, mask, dev), .value = (val)}
+#define SND_PCI_QUIRK_VENDOR(vend, xname, val)			\
+	{_SND_PCI_QUIRK_ID_MASK(vend, 0, 0), .value = (val)}
+#endif
+
+const struct snd_pci_quirk *
+snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list);
+
+const struct snd_pci_quirk *
+snd_pci_quirk_lookup_id(u16 vendor, u16 device,
+			const struct snd_pci_quirk *list);
+
+#endif /* __SOUND_CORE_H */
diff --git a/include/sound/cs4231-regs.h b/include/sound/cs4231-regs.h
new file mode 100644
index 0000000..66d28c2
--- /dev/null
+++ b/include/sound/cs4231-regs.h
@@ -0,0 +1,187 @@
+#ifndef __SOUND_CS4231_REGS_H
+#define __SOUND_CS4231_REGS_H
+
+/*
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *  Definitions for CS4231 & InterWave chips & compatible chips registers
+ *
+ *
+ *   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., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ */
+
+/* IO ports */
+
+#define CS4231P(x)		(c_d_c_CS4231##x)
+
+#define c_d_c_CS4231REGSEL	0
+#define c_d_c_CS4231REG		1
+#define c_d_c_CS4231STATUS	2
+#define c_d_c_CS4231PIO		3
+
+/* codec registers */
+
+#define CS4231_LEFT_INPUT	0x00	/* left input control */
+#define CS4231_RIGHT_INPUT	0x01	/* right input control */
+#define CS4231_AUX1_LEFT_INPUT	0x02	/* left AUX1 input control */
+#define CS4231_AUX1_RIGHT_INPUT	0x03	/* right AUX1 input control */
+#define CS4231_AUX2_LEFT_INPUT	0x04	/* left AUX2 input control */
+#define CS4231_AUX2_RIGHT_INPUT	0x05	/* right AUX2 input control */
+#define CS4231_LEFT_OUTPUT	0x06	/* left output control register */
+#define CS4231_RIGHT_OUTPUT	0x07	/* right output control register */
+#define CS4231_PLAYBK_FORMAT	0x08	/* clock and data format - playback - bits 7-0 MCE */
+#define CS4231_IFACE_CTRL	0x09	/* interface control - bits 7-2 MCE */
+#define CS4231_PIN_CTRL		0x0a	/* pin control */
+#define CS4231_TEST_INIT	0x0b	/* test and initialization */
+#define CS4231_MISC_INFO	0x0c	/* miscellaneous information */
+#define CS4231_LOOPBACK		0x0d	/* loopback control */
+#define CS4231_PLY_UPR_CNT	0x0e	/* playback upper base count */
+#define CS4231_PLY_LWR_CNT	0x0f	/* playback lower base count */
+#define CS4231_ALT_FEATURE_1	0x10	/* alternate #1 feature enable */
+#define AD1845_AF1_MIC_LEFT	0x10	/* alternate #1 feature + MIC left */
+#define CS4231_ALT_FEATURE_2	0x11	/* alternate #2 feature enable */
+#define AD1845_AF2_MIC_RIGHT	0x11	/* alternate #2 feature + MIC right */
+#define CS4231_LEFT_LINE_IN	0x12	/* left line input control */
+#define CS4231_RIGHT_LINE_IN	0x13	/* right line input control */
+#define CS4231_TIMER_LOW	0x14	/* timer low byte */
+#define CS4231_TIMER_HIGH	0x15	/* timer high byte */
+#define CS4231_LEFT_MIC_INPUT	0x16	/* left MIC input control register (InterWave only) */
+#define AD1845_UPR_FREQ_SEL	0x16	/* upper byte of frequency select */
+#define CS4231_RIGHT_MIC_INPUT	0x17	/* right MIC input control register (InterWave only) */
+#define AD1845_LWR_FREQ_SEL	0x17	/* lower byte of frequency select */
+#define CS4236_EXT_REG		0x17	/* extended register access */
+#define CS4231_IRQ_STATUS	0x18	/* irq status register */
+#define CS4231_LINE_LEFT_OUTPUT	0x19	/* left line output control register (InterWave only) */
+#define CS4231_VERSION		0x19	/* CS4231(A) - version values */
+#define CS4231_MONO_CTRL	0x1a	/* mono input/output control */
+#define CS4231_LINE_RIGHT_OUTPUT 0x1b	/* right line output control register (InterWave only) */
+#define AD1845_PWR_DOWN		0x1b	/* power down control */
+#define CS4235_LEFT_MASTER	0x1b	/* left master output control */
+#define CS4231_REC_FORMAT	0x1c	/* clock and data format - record - bits 7-0 MCE */
+#define AD1845_CLOCK		0x1d	/* crystal clock select and total power down */
+#define CS4235_RIGHT_MASTER	0x1d	/* right master output control */
+#define CS4231_REC_UPR_CNT	0x1e	/* record upper count */
+#define CS4231_REC_LWR_CNT	0x1f	/* record lower count */
+
+/* definitions for codec register select port - CODECP( REGSEL ) */
+
+#define CS4231_INIT		0x80	/* CODEC is initializing */
+#define CS4231_MCE		0x40	/* mode change enable */
+#define CS4231_TRD		0x20	/* transfer request disable */
+
+/* definitions for codec status register - CODECP( STATUS ) */
+
+#define CS4231_GLOBALIRQ	0x01	/* IRQ is active */
+
+/* definitions for codec irq status */
+
+#define CS4231_PLAYBACK_IRQ	0x10
+#define CS4231_RECORD_IRQ	0x20
+#define CS4231_TIMER_IRQ	0x40
+#define CS4231_ALL_IRQS		0x70
+#define CS4231_REC_UNDERRUN	0x08
+#define CS4231_REC_OVERRUN	0x04
+#define CS4231_PLY_OVERRUN	0x02
+#define CS4231_PLY_UNDERRUN	0x01
+
+/* definitions for CS4231_LEFT_INPUT and CS4231_RIGHT_INPUT registers */
+
+#define CS4231_ENABLE_MIC_GAIN	0x20
+
+#define CS4231_MIXS_LINE	0x00
+#define CS4231_MIXS_AUX1	0x40
+#define CS4231_MIXS_MIC		0x80
+#define CS4231_MIXS_ALL		0xc0
+
+/* definitions for clock and data format register - CS4231_PLAYBK_FORMAT */
+
+#define CS4231_LINEAR_8		0x00	/* 8-bit unsigned data */
+#define CS4231_ALAW_8		0x60	/* 8-bit A-law companded */
+#define CS4231_ULAW_8		0x20	/* 8-bit U-law companded */
+#define CS4231_LINEAR_16	0x40	/* 16-bit twos complement data - little endian */
+#define CS4231_LINEAR_16_BIG	0xc0	/* 16-bit twos complement data - big endian */
+#define CS4231_ADPCM_16		0xa0	/* 16-bit ADPCM */
+#define CS4231_STEREO		0x10	/* stereo mode */
+/* bits 3-1 define frequency divisor */
+#define CS4231_XTAL1		0x00	/* 24.576 crystal */
+#define CS4231_XTAL2		0x01	/* 16.9344 crystal */
+
+/* definitions for interface control register - CS4231_IFACE_CTRL */
+
+#define CS4231_RECORD_PIO	0x80	/* record PIO enable */
+#define CS4231_PLAYBACK_PIO	0x40	/* playback PIO enable */
+#define CS4231_CALIB_MODE	0x18	/* calibration mode bits */
+#define CS4231_AUTOCALIB	0x08	/* auto calibrate */
+#define CS4231_SINGLE_DMA	0x04	/* use single DMA channel */
+#define CS4231_RECORD_ENABLE	0x02	/* record enable */
+#define CS4231_PLAYBACK_ENABLE	0x01	/* playback enable */
+
+/* definitions for pin control register - CS4231_PIN_CTRL */
+
+#define CS4231_IRQ_ENABLE	0x02	/* enable IRQ */
+#define CS4231_XCTL1		0x40	/* external control #1 */
+#define CS4231_XCTL0		0x80	/* external control #0 */
+
+/* definitions for test and init register - CS4231_TEST_INIT */
+
+#define CS4231_CALIB_IN_PROGRESS 0x20	/* auto calibrate in progress */
+#define CS4231_DMA_REQUEST	0x10	/* DMA request in progress */
+
+/* definitions for misc control register - CS4231_MISC_INFO */
+
+#define CS4231_MODE2		0x40	/* MODE 2 */
+#define CS4231_IW_MODE3		0x6c	/* MODE 3 - InterWave enhanced mode */
+#define CS4231_4236_MODE3	0xe0	/* MODE 3 - CS4236+ enhanced mode */
+
+/* definitions for alternate feature 1 register - CS4231_ALT_FEATURE_1 */
+
+#define	CS4231_DACZ		0x01	/* zero DAC when underrun */
+#define CS4231_TIMER_ENABLE	0x40	/* codec timer enable */
+#define CS4231_OLB		0x80	/* output level bit */
+
+/* definitions for Extended Registers - CS4236+ */
+
+#define CS4236_REG(i23val)	(((i23val << 2) & 0x10) | ((i23val >> 4) & 0x0f))
+#define CS4236_I23VAL(reg)	((((reg)&0xf) << 4) | (((reg)&0x10) >> 2) | 0x8)
+
+#define CS4236_LEFT_LINE	0x08	/* left LINE alternate volume */
+#define CS4236_RIGHT_LINE	0x18	/* right LINE alternate volume */
+#define CS4236_LEFT_MIC		0x28	/* left MIC volume */
+#define CS4236_RIGHT_MIC	0x38	/* right MIC volume */
+#define CS4236_LEFT_MIX_CTRL	0x48	/* synthesis and left input mixer control */
+#define CS4236_RIGHT_MIX_CTRL	0x58	/* right input mixer control */
+#define CS4236_LEFT_FM		0x68	/* left FM volume */
+#define CS4236_RIGHT_FM		0x78	/* right FM volume */
+#define CS4236_LEFT_DSP		0x88	/* left DSP serial port volume */
+#define CS4236_RIGHT_DSP	0x98	/* right DSP serial port volume */
+#define CS4236_RIGHT_LOOPBACK	0xa8	/* right loopback monitor volume */
+#define CS4236_DAC_MUTE		0xb8	/* DAC mute and IFSE enable */
+#define CS4236_ADC_RATE		0xc8	/* indenpendent ADC sample frequency */
+#define CS4236_DAC_RATE		0xd8	/* indenpendent DAC sample frequency */
+#define CS4236_LEFT_MASTER	0xe8	/* left master digital audio volume */
+#define CS4236_RIGHT_MASTER	0xf8	/* right master digital audio volume */
+#define CS4236_LEFT_WAVE	0x0c	/* left wavetable serial port volume */
+#define CS4236_RIGHT_WAVE	0x1c	/* right wavetable serial port volume */
+#define CS4236_VERSION		0x9c	/* chip version and ID */
+
+/* definitions for extended registers - OPTI93X */
+#define OPTi931_AUX_LEFT_INPUT	0x10
+#define OPTi931_AUX_RIGHT_INPUT	0x11
+#define OPTi93X_MIC_LEFT_INPUT	0x14
+#define OPTi93X_MIC_RIGHT_INPUT	0x15
+#define OPTi93X_OUT_LEFT	0x16
+#define OPTi93X_OUT_RIGHT	0x17
+
+#endif /* __SOUND_CS4231_REGS_H */
diff --git a/include/sound/cs46xx.h b/include/sound/cs46xx.h
new file mode 100644
index 0000000..e3005a6
--- /dev/null
+++ b/include/sound/cs46xx.h
@@ -0,0 +1,1745 @@
+#ifndef __SOUND_CS46XX_H
+#define __SOUND_CS46XX_H
+
+/*
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
+ *		     Cirrus Logic, Inc.
+ *  Definitions for Cirrus Logic CS46xx chips
+ *
+ *
+ *   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., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ */
+
+#include "pcm.h"
+#include "pcm-indirect.h"
+#include "rawmidi.h"
+#include "ac97_codec.h"
+#include "cs46xx_dsp_spos.h"
+
+/*
+ *  Direct registers
+ */
+
+/*
+ *  The following define the offsets of the registers accessed via base address
+ *  register zero on the CS46xx part.
+ */
+#define BA0_HISR				0x00000000
+#define BA0_HSR0                                0x00000004
+#define BA0_HICR                                0x00000008
+#define BA0_DMSR                                0x00000100
+#define BA0_HSAR                                0x00000110
+#define BA0_HDAR                                0x00000114
+#define BA0_HDMR                                0x00000118
+#define BA0_HDCR                                0x0000011C
+#define BA0_PFMC                                0x00000200
+#define BA0_PFCV1                               0x00000204
+#define BA0_PFCV2                               0x00000208
+#define BA0_PCICFG00                            0x00000300
+#define BA0_PCICFG04                            0x00000304
+#define BA0_PCICFG08                            0x00000308
+#define BA0_PCICFG0C                            0x0000030C
+#define BA0_PCICFG10                            0x00000310
+#define BA0_PCICFG14                            0x00000314
+#define BA0_PCICFG18                            0x00000318
+#define BA0_PCICFG1C                            0x0000031C
+#define BA0_PCICFG20                            0x00000320
+#define BA0_PCICFG24                            0x00000324
+#define BA0_PCICFG28                            0x00000328
+#define BA0_PCICFG2C                            0x0000032C
+#define BA0_PCICFG30                            0x00000330
+#define BA0_PCICFG34                            0x00000334
+#define BA0_PCICFG38                            0x00000338
+#define BA0_PCICFG3C                            0x0000033C
+#define BA0_CLKCR1                              0x00000400
+#define BA0_CLKCR2                              0x00000404
+#define BA0_PLLM                                0x00000408
+#define BA0_PLLCC                               0x0000040C
+#define BA0_FRR                                 0x00000410 
+#define BA0_CFL1                                0x00000414
+#define BA0_CFL2                                0x00000418
+#define BA0_SERMC1                              0x00000420
+#define BA0_SERMC2                              0x00000424
+#define BA0_SERC1                               0x00000428
+#define BA0_SERC2                               0x0000042C
+#define BA0_SERC3                               0x00000430
+#define BA0_SERC4                               0x00000434
+#define BA0_SERC5                               0x00000438
+#define BA0_SERBSP                              0x0000043C
+#define BA0_SERBST                              0x00000440
+#define BA0_SERBCM                              0x00000444
+#define BA0_SERBAD                              0x00000448
+#define BA0_SERBCF                              0x0000044C
+#define BA0_SERBWP                              0x00000450
+#define BA0_SERBRP                              0x00000454
+#ifndef NO_CS4612
+#define BA0_ASER_FADDR                          0x00000458
+#endif
+#define BA0_ACCTL                               0x00000460
+#define BA0_ACSTS                               0x00000464
+#define BA0_ACOSV                               0x00000468
+#define BA0_ACCAD                               0x0000046C
+#define BA0_ACCDA                               0x00000470
+#define BA0_ACISV                               0x00000474
+#define BA0_ACSAD                               0x00000478
+#define BA0_ACSDA                               0x0000047C
+#define BA0_JSPT                                0x00000480
+#define BA0_JSCTL                               0x00000484
+#define BA0_JSC1                                0x00000488
+#define BA0_JSC2                                0x0000048C
+#define BA0_MIDCR                               0x00000490
+#define BA0_MIDSR                               0x00000494
+#define BA0_MIDWP                               0x00000498
+#define BA0_MIDRP                               0x0000049C
+#define BA0_JSIO                                0x000004A0
+#ifndef NO_CS4612
+#define BA0_ASER_MASTER                         0x000004A4
+#endif
+#define BA0_CFGI                                0x000004B0
+#define BA0_SSVID                               0x000004B4
+#define BA0_GPIOR                               0x000004B8
+#ifndef NO_CS4612
+#define BA0_EGPIODR                             0x000004BC
+#define BA0_EGPIOPTR                            0x000004C0
+#define BA0_EGPIOTR                             0x000004C4
+#define BA0_EGPIOWR                             0x000004C8
+#define BA0_EGPIOSR                             0x000004CC
+#define BA0_SERC6                               0x000004D0
+#define BA0_SERC7                               0x000004D4
+#define BA0_SERACC                              0x000004D8
+#define BA0_ACCTL2                              0x000004E0
+#define BA0_ACSTS2                              0x000004E4
+#define BA0_ACOSV2                              0x000004E8
+#define BA0_ACCAD2                              0x000004EC
+#define BA0_ACCDA2                              0x000004F0
+#define BA0_ACISV2                              0x000004F4
+#define BA0_ACSAD2                              0x000004F8
+#define BA0_ACSDA2                              0x000004FC
+#define BA0_IOTAC0                              0x00000500
+#define BA0_IOTAC1                              0x00000504
+#define BA0_IOTAC2                              0x00000508
+#define BA0_IOTAC3                              0x0000050C
+#define BA0_IOTAC4                              0x00000510
+#define BA0_IOTAC5                              0x00000514
+#define BA0_IOTAC6                              0x00000518
+#define BA0_IOTAC7                              0x0000051C
+#define BA0_IOTAC8                              0x00000520
+#define BA0_IOTAC9                              0x00000524
+#define BA0_IOTAC10                             0x00000528
+#define BA0_IOTAC11                             0x0000052C
+#define BA0_IOTFR0                              0x00000540
+#define BA0_IOTFR1                              0x00000544
+#define BA0_IOTFR2                              0x00000548
+#define BA0_IOTFR3                              0x0000054C
+#define BA0_IOTFR4                              0x00000550
+#define BA0_IOTFR5                              0x00000554
+#define BA0_IOTFR6                              0x00000558
+#define BA0_IOTFR7                              0x0000055C
+#define BA0_IOTFIFO                             0x00000580
+#define BA0_IOTRRD                              0x00000584
+#define BA0_IOTFP                               0x00000588
+#define BA0_IOTCR                               0x0000058C
+#define BA0_DPCID                               0x00000590
+#define BA0_DPCIA                               0x00000594
+#define BA0_DPCIC                               0x00000598
+#define BA0_PCPCIR                              0x00000600
+#define BA0_PCPCIG                              0x00000604
+#define BA0_PCPCIEN                             0x00000608
+#define BA0_EPCIPMC                             0x00000610
+#endif
+
+/*
+ *  The following define the offsets of the registers and memories accessed via
+ *  base address register one on the CS46xx part.
+ */
+#define BA1_SP_DMEM0                            0x00000000
+#define BA1_SP_DMEM1                            0x00010000
+#define BA1_SP_PMEM                             0x00020000
+#define BA1_SP_REG				0x00030000
+#define BA1_SPCR                                0x00030000
+#define BA1_DREG                                0x00030004
+#define BA1_DSRWP                               0x00030008
+#define BA1_TWPR                                0x0003000C
+#define BA1_SPWR                                0x00030010
+#define BA1_SPIR                                0x00030014
+#define BA1_FGR1                                0x00030020
+#define BA1_SPCS                                0x00030028
+#define BA1_SDSR                                0x0003002C
+#define BA1_FRMT                                0x00030030
+#define BA1_FRCC                                0x00030034
+#define BA1_FRSC                                0x00030038
+#define BA1_OMNI_MEM                            0x000E0000
+
+
+/*
+ *  The following defines are for the flags in the host interrupt status
+ *  register.
+ */
+#define HISR_VC_MASK                            0x0000FFFF
+#define HISR_VC0                                0x00000001
+#define HISR_VC1                                0x00000002
+#define HISR_VC2                                0x00000004
+#define HISR_VC3                                0x00000008
+#define HISR_VC4                                0x00000010
+#define HISR_VC5                                0x00000020
+#define HISR_VC6                                0x00000040
+#define HISR_VC7                                0x00000080
+#define HISR_VC8                                0x00000100
+#define HISR_VC9                                0x00000200
+#define HISR_VC10                               0x00000400
+#define HISR_VC11                               0x00000800
+#define HISR_VC12                               0x00001000
+#define HISR_VC13                               0x00002000
+#define HISR_VC14                               0x00004000
+#define HISR_VC15                               0x00008000
+#define HISR_INT0                               0x00010000
+#define HISR_INT1                               0x00020000
+#define HISR_DMAI                               0x00040000
+#define HISR_FROVR                              0x00080000
+#define HISR_MIDI                               0x00100000
+#ifdef NO_CS4612
+#define HISR_RESERVED                           0x0FE00000
+#else
+#define HISR_SBINT                              0x00200000
+#define HISR_RESERVED                           0x0FC00000
+#endif
+#define HISR_H0P                                0x40000000
+#define HISR_INTENA                             0x80000000
+
+/*
+ *  The following defines are for the flags in the host signal register 0.
+ */
+#define HSR0_VC_MASK                            0xFFFFFFFF
+#define HSR0_VC16                               0x00000001
+#define HSR0_VC17                               0x00000002
+#define HSR0_VC18                               0x00000004
+#define HSR0_VC19                               0x00000008
+#define HSR0_VC20                               0x00000010
+#define HSR0_VC21                               0x00000020
+#define HSR0_VC22                               0x00000040
+#define HSR0_VC23                               0x00000080
+#define HSR0_VC24                               0x00000100
+#define HSR0_VC25                               0x00000200
+#define HSR0_VC26                               0x00000400
+#define HSR0_VC27                               0x00000800
+#define HSR0_VC28                               0x00001000
+#define HSR0_VC29                               0x00002000
+#define HSR0_VC30                               0x00004000
+#define HSR0_VC31                               0x00008000
+#define HSR0_VC32                               0x00010000
+#define HSR0_VC33                               0x00020000
+#define HSR0_VC34                               0x00040000
+#define HSR0_VC35                               0x00080000
+#define HSR0_VC36                               0x00100000
+#define HSR0_VC37                               0x00200000
+#define HSR0_VC38                               0x00400000
+#define HSR0_VC39                               0x00800000
+#define HSR0_VC40                               0x01000000
+#define HSR0_VC41                               0x02000000
+#define HSR0_VC42                               0x04000000
+#define HSR0_VC43                               0x08000000
+#define HSR0_VC44                               0x10000000
+#define HSR0_VC45                               0x20000000
+#define HSR0_VC46                               0x40000000
+#define HSR0_VC47                               0x80000000
+
+/*
+ *  The following defines are for the flags in the host interrupt control
+ *  register.
+ */
+#define HICR_IEV                                0x00000001
+#define HICR_CHGM                               0x00000002
+
+/*
+ *  The following defines are for the flags in the DMA status register.
+ */
+#define DMSR_HP                                 0x00000001
+#define DMSR_HR                                 0x00000002
+#define DMSR_SP                                 0x00000004
+#define DMSR_SR                                 0x00000008
+
+/*
+ *  The following defines are for the flags in the host DMA source address
+ *  register.
+ */
+#define HSAR_HOST_ADDR_MASK                     0xFFFFFFFF
+#define HSAR_DSP_ADDR_MASK                      0x0000FFFF
+#define HSAR_MEMID_MASK                         0x000F0000
+#define HSAR_MEMID_SP_DMEM0                     0x00000000
+#define HSAR_MEMID_SP_DMEM1                     0x00010000
+#define HSAR_MEMID_SP_PMEM                      0x00020000
+#define HSAR_MEMID_SP_DEBUG                     0x00030000
+#define HSAR_MEMID_OMNI_MEM                     0x000E0000
+#define HSAR_END                                0x40000000
+#define HSAR_ERR                                0x80000000
+
+/*
+ *  The following defines are for the flags in the host DMA destination address
+ *  register.
+ */
+#define HDAR_HOST_ADDR_MASK                     0xFFFFFFFF
+#define HDAR_DSP_ADDR_MASK                      0x0000FFFF
+#define HDAR_MEMID_MASK                         0x000F0000
+#define HDAR_MEMID_SP_DMEM0                     0x00000000
+#define HDAR_MEMID_SP_DMEM1                     0x00010000
+#define HDAR_MEMID_SP_PMEM                      0x00020000
+#define HDAR_MEMID_SP_DEBUG                     0x00030000
+#define HDAR_MEMID_OMNI_MEM                     0x000E0000
+#define HDAR_END                                0x40000000
+#define HDAR_ERR                                0x80000000
+
+/*
+ *  The following defines are for the flags in the host DMA control register.
+ */
+#define HDMR_AC_MASK                            0x0000F000
+#define HDMR_AC_8_16                            0x00001000
+#define HDMR_AC_M_S                             0x00002000
+#define HDMR_AC_B_L                             0x00004000
+#define HDMR_AC_S_U                             0x00008000
+
+/*
+ *  The following defines are for the flags in the host DMA control register.
+ */
+#define HDCR_COUNT_MASK                         0x000003FF
+#define HDCR_DONE                               0x00004000
+#define HDCR_OPT                                0x00008000
+#define HDCR_WBD                                0x00400000
+#define HDCR_WBS                                0x00800000
+#define HDCR_DMS_MASK                           0x07000000
+#define HDCR_DMS_LINEAR                         0x00000000
+#define HDCR_DMS_16_DWORDS                      0x01000000
+#define HDCR_DMS_32_DWORDS                      0x02000000
+#define HDCR_DMS_64_DWORDS                      0x03000000
+#define HDCR_DMS_128_DWORDS                     0x04000000
+#define HDCR_DMS_256_DWORDS                     0x05000000
+#define HDCR_DMS_512_DWORDS                     0x06000000
+#define HDCR_DMS_1024_DWORDS                    0x07000000
+#define HDCR_DH                                 0x08000000
+#define HDCR_SMS_MASK                           0x70000000
+#define HDCR_SMS_LINEAR                         0x00000000
+#define HDCR_SMS_16_DWORDS                      0x10000000
+#define HDCR_SMS_32_DWORDS                      0x20000000
+#define HDCR_SMS_64_DWORDS                      0x30000000
+#define HDCR_SMS_128_DWORDS                     0x40000000
+#define HDCR_SMS_256_DWORDS                     0x50000000
+#define HDCR_SMS_512_DWORDS                     0x60000000
+#define HDCR_SMS_1024_DWORDS                    0x70000000
+#define HDCR_SH                                 0x80000000
+#define HDCR_COUNT_SHIFT                        0
+
+/*
+ *  The following defines are for the flags in the performance monitor control
+ *  register.
+ */
+#define PFMC_C1SS_MASK                          0x0000001F
+#define PFMC_C1EV                               0x00000020
+#define PFMC_C1RS                               0x00008000
+#define PFMC_C2SS_MASK                          0x001F0000
+#define PFMC_C2EV                               0x00200000
+#define PFMC_C2RS                               0x80000000
+#define PFMC_C1SS_SHIFT                         0
+#define PFMC_C2SS_SHIFT                         16
+#define PFMC_BUS_GRANT                          0
+#define PFMC_GRANT_AFTER_REQ                    1
+#define PFMC_TRANSACTION                        2
+#define PFMC_DWORD_TRANSFER                     3
+#define PFMC_SLAVE_READ                         4
+#define PFMC_SLAVE_WRITE                        5
+#define PFMC_PREEMPTION                         6
+#define PFMC_DISCONNECT_RETRY                   7
+#define PFMC_INTERRUPT                          8
+#define PFMC_BUS_OWNERSHIP                      9
+#define PFMC_TRANSACTION_LAG                    10
+#define PFMC_PCI_CLOCK                          11
+#define PFMC_SERIAL_CLOCK                       12
+#define PFMC_SP_CLOCK                           13
+
+/*
+ *  The following defines are for the flags in the performance counter value 1
+ *  register.
+ */
+#define PFCV1_PC1V_MASK                         0xFFFFFFFF
+#define PFCV1_PC1V_SHIFT                        0
+
+/*
+ *  The following defines are for the flags in the performance counter value 2
+ *  register.
+ */
+#define PFCV2_PC2V_MASK                         0xFFFFFFFF
+#define PFCV2_PC2V_SHIFT                        0
+
+/*
+ *  The following defines are for the flags in the clock control register 1.
+ */
+#define CLKCR1_OSCS                             0x00000001
+#define CLKCR1_OSCP                             0x00000002
+#define CLKCR1_PLLSS_MASK                       0x0000000C
+#define CLKCR1_PLLSS_SERIAL                     0x00000000
+#define CLKCR1_PLLSS_CRYSTAL                    0x00000004
+#define CLKCR1_PLLSS_PCI                        0x00000008
+#define CLKCR1_PLLSS_RESERVED                   0x0000000C
+#define CLKCR1_PLLP                             0x00000010
+#define CLKCR1_SWCE                             0x00000020
+#define CLKCR1_PLLOS                            0x00000040
+
+/*
+ *  The following defines are for the flags in the clock control register 2.
+ */
+#define CLKCR2_PDIVS_MASK                       0x0000000F
+#define CLKCR2_PDIVS_1                          0x00000001
+#define CLKCR2_PDIVS_2                          0x00000002
+#define CLKCR2_PDIVS_4                          0x00000004
+#define CLKCR2_PDIVS_7                          0x00000007
+#define CLKCR2_PDIVS_8                          0x00000008
+#define CLKCR2_PDIVS_16                         0x00000000
+
+/*
+ *  The following defines are for the flags in the PLL multiplier register.
+ */
+#define PLLM_MASK                               0x000000FF
+#define PLLM_SHIFT                              0
+
+/*
+ *  The following defines are for the flags in the PLL capacitor coefficient
+ *  register.
+ */
+#define PLLCC_CDR_MASK                          0x00000007
+#ifndef NO_CS4610
+#define PLLCC_CDR_240_350_MHZ                   0x00000000
+#define PLLCC_CDR_184_265_MHZ                   0x00000001
+#define PLLCC_CDR_144_205_MHZ                   0x00000002
+#define PLLCC_CDR_111_160_MHZ                   0x00000003
+#define PLLCC_CDR_87_123_MHZ                    0x00000004
+#define PLLCC_CDR_67_96_MHZ                     0x00000005
+#define PLLCC_CDR_52_74_MHZ                     0x00000006
+#define PLLCC_CDR_45_58_MHZ                     0x00000007
+#endif
+#ifndef NO_CS4612
+#define PLLCC_CDR_271_398_MHZ                   0x00000000
+#define PLLCC_CDR_227_330_MHZ                   0x00000001
+#define PLLCC_CDR_167_239_MHZ                   0x00000002
+#define PLLCC_CDR_150_215_MHZ                   0x00000003
+#define PLLCC_CDR_107_154_MHZ                   0x00000004
+#define PLLCC_CDR_98_140_MHZ                    0x00000005
+#define PLLCC_CDR_73_104_MHZ                    0x00000006
+#define PLLCC_CDR_63_90_MHZ                     0x00000007
+#endif
+#define PLLCC_LPF_MASK                          0x000000F8
+#ifndef NO_CS4610
+#define PLLCC_LPF_23850_60000_KHZ               0x00000000
+#define PLLCC_LPF_7960_26290_KHZ                0x00000008
+#define PLLCC_LPF_4160_10980_KHZ                0x00000018
+#define PLLCC_LPF_1740_4580_KHZ                 0x00000038
+#define PLLCC_LPF_724_1910_KHZ                  0x00000078
+#define PLLCC_LPF_317_798_KHZ                   0x000000F8
+#endif
+#ifndef NO_CS4612
+#define PLLCC_LPF_25580_64530_KHZ               0x00000000
+#define PLLCC_LPF_14360_37270_KHZ               0x00000008
+#define PLLCC_LPF_6100_16020_KHZ                0x00000018
+#define PLLCC_LPF_2540_6690_KHZ                 0x00000038
+#define PLLCC_LPF_1050_2780_KHZ                 0x00000078
+#define PLLCC_LPF_450_1160_KHZ                  0x000000F8
+#endif
+
+/*
+ *  The following defines are for the flags in the feature reporting register.
+ */
+#define FRR_FAB_MASK                            0x00000003
+#define FRR_MASK_MASK                           0x0000001C
+#ifdef NO_CS4612
+#define FRR_CFOP_MASK                           0x000000E0
+#else
+#define FRR_CFOP_MASK                           0x00000FE0
+#endif
+#define FRR_CFOP_NOT_DVD                        0x00000020
+#define FRR_CFOP_A3D                            0x00000040
+#define FRR_CFOP_128_PIN                        0x00000080
+#ifndef NO_CS4612
+#define FRR_CFOP_CS4280                         0x00000800
+#endif
+#define FRR_FAB_SHIFT                           0
+#define FRR_MASK_SHIFT                          2
+#define FRR_CFOP_SHIFT                          5
+
+/*
+ *  The following defines are for the flags in the configuration load 1
+ *  register.
+ */
+#define CFL1_CLOCK_SOURCE_MASK                  0x00000003
+#define CFL1_CLOCK_SOURCE_CS423X                0x00000000
+#define CFL1_CLOCK_SOURCE_AC97                  0x00000001
+#define CFL1_CLOCK_SOURCE_CRYSTAL               0x00000002
+#define CFL1_CLOCK_SOURCE_DUAL_AC97             0x00000003
+#define CFL1_VALID_DATA_MASK                    0x000000FF
+
+/*
+ *  The following defines are for the flags in the configuration load 2
+ *  register.
+ */
+#define CFL2_VALID_DATA_MASK                    0x000000FF
+
+/*
+ *  The following defines are for the flags in the serial port master control
+ *  register 1.
+ */
+#define SERMC1_MSPE                             0x00000001
+#define SERMC1_PTC_MASK                         0x0000000E
+#define SERMC1_PTC_CS423X                       0x00000000
+#define SERMC1_PTC_AC97                         0x00000002
+#define SERMC1_PTC_DAC                          0x00000004
+#define SERMC1_PLB                              0x00000010
+#define SERMC1_XLB                              0x00000020
+
+/*
+ *  The following defines are for the flags in the serial port master control
+ *  register 2.
+ */
+#define SERMC2_LROE                             0x00000001
+#define SERMC2_MCOE                             0x00000002
+#define SERMC2_MCDIV                            0x00000004
+
+/*
+ *  The following defines are for the flags in the serial port 1 configuration
+ *  register.
+ */
+#define SERC1_SO1EN                             0x00000001
+#define SERC1_SO1F_MASK                         0x0000000E
+#define SERC1_SO1F_CS423X                       0x00000000
+#define SERC1_SO1F_AC97                         0x00000002
+#define SERC1_SO1F_DAC                          0x00000004
+#define SERC1_SO1F_SPDIF                        0x00000006
+
+/*
+ *  The following defines are for the flags in the serial port 2 configuration
+ *  register.
+ */
+#define SERC2_SI1EN                             0x00000001
+#define SERC2_SI1F_MASK                         0x0000000E
+#define SERC2_SI1F_CS423X                       0x00000000
+#define SERC2_SI1F_AC97                         0x00000002
+#define SERC2_SI1F_ADC                          0x00000004
+#define SERC2_SI1F_SPDIF                        0x00000006
+
+/*
+ *  The following defines are for the flags in the serial port 3 configuration
+ *  register.
+ */
+#define SERC3_SO2EN                             0x00000001
+#define SERC3_SO2F_MASK                         0x00000006
+#define SERC3_SO2F_DAC                          0x00000000
+#define SERC3_SO2F_SPDIF                        0x00000002
+
+/*
+ *  The following defines are for the flags in the serial port 4 configuration
+ *  register.
+ */
+#define SERC4_SO3EN                             0x00000001
+#define SERC4_SO3F_MASK                         0x00000006
+#define SERC4_SO3F_DAC                          0x00000000
+#define SERC4_SO3F_SPDIF                        0x00000002
+
+/*
+ *  The following defines are for the flags in the serial port 5 configuration
+ *  register.
+ */
+#define SERC5_SI2EN                             0x00000001
+#define SERC5_SI2F_MASK                         0x00000006
+#define SERC5_SI2F_ADC                          0x00000000
+#define SERC5_SI2F_SPDIF                        0x00000002
+
+/*
+ *  The following defines are for the flags in the serial port backdoor sample
+ *  pointer register.
+ */
+#define SERBSP_FSP_MASK                         0x0000000F
+#define SERBSP_FSP_SHIFT                        0
+
+/*
+ *  The following defines are for the flags in the serial port backdoor status
+ *  register.
+ */
+#define SERBST_RRDY                             0x00000001
+#define SERBST_WBSY                             0x00000002
+
+/*
+ *  The following defines are for the flags in the serial port backdoor command
+ *  register.
+ */
+#define SERBCM_RDC                              0x00000001
+#define SERBCM_WRC                              0x00000002
+
+/*
+ *  The following defines are for the flags in the serial port backdoor address
+ *  register.
+ */
+#ifdef NO_CS4612
+#define SERBAD_FAD_MASK                         0x000000FF
+#else
+#define SERBAD_FAD_MASK                         0x000001FF
+#endif
+#define SERBAD_FAD_SHIFT                        0
+
+/*
+ *  The following defines are for the flags in the serial port backdoor
+ *  configuration register.
+ */
+#define SERBCF_HBP                              0x00000001
+
+/*
+ *  The following defines are for the flags in the serial port backdoor write
+ *  port register.
+ */
+#define SERBWP_FWD_MASK                         0x000FFFFF
+#define SERBWP_FWD_SHIFT                        0
+
+/*
+ *  The following defines are for the flags in the serial port backdoor read
+ *  port register.
+ */
+#define SERBRP_FRD_MASK                         0x000FFFFF
+#define SERBRP_FRD_SHIFT                        0
+
+/*
+ *  The following defines are for the flags in the async FIFO address register.
+ */
+#ifndef NO_CS4612
+#define ASER_FADDR_A1_MASK                      0x000001FF
+#define ASER_FADDR_EN1                          0x00008000
+#define ASER_FADDR_A2_MASK                      0x01FF0000
+#define ASER_FADDR_EN2                          0x80000000
+#define ASER_FADDR_A1_SHIFT                     0
+#define ASER_FADDR_A2_SHIFT                     16
+#endif
+
+/*
+ *  The following defines are for the flags in the AC97 control register.
+ */
+#define ACCTL_RSTN                              0x00000001
+#define ACCTL_ESYN                              0x00000002
+#define ACCTL_VFRM                              0x00000004
+#define ACCTL_DCV                               0x00000008
+#define ACCTL_CRW                               0x00000010
+#define ACCTL_ASYN                              0x00000020
+#ifndef NO_CS4612
+#define ACCTL_TC                                0x00000040
+#endif
+
+/*
+ *  The following defines are for the flags in the AC97 status register.
+ */
+#define ACSTS_CRDY                              0x00000001
+#define ACSTS_VSTS                              0x00000002
+#ifndef NO_CS4612
+#define ACSTS_WKUP                              0x00000004
+#endif
+
+/*
+ *  The following defines are for the flags in the AC97 output slot valid
+ *  register.
+ */
+#define ACOSV_SLV3                              0x00000001
+#define ACOSV_SLV4                              0x00000002
+#define ACOSV_SLV5                              0x00000004
+#define ACOSV_SLV6                              0x00000008
+#define ACOSV_SLV7                              0x00000010
+#define ACOSV_SLV8                              0x00000020
+#define ACOSV_SLV9                              0x00000040
+#define ACOSV_SLV10                             0x00000080
+#define ACOSV_SLV11                             0x00000100
+#define ACOSV_SLV12                             0x00000200
+
+/*
+ *  The following defines are for the flags in the AC97 command address
+ *  register.
+ */
+#define ACCAD_CI_MASK                           0x0000007F
+#define ACCAD_CI_SHIFT                          0
+
+/*
+ *  The following defines are for the flags in the AC97 command data register.
+ */
+#define ACCDA_CD_MASK                           0x0000FFFF
+#define ACCDA_CD_SHIFT                          0
+
+/*
+ *  The following defines are for the flags in the AC97 input slot valid
+ *  register.
+ */
+#define ACISV_ISV3                              0x00000001
+#define ACISV_ISV4                              0x00000002
+#define ACISV_ISV5                              0x00000004
+#define ACISV_ISV6                              0x00000008
+#define ACISV_ISV7                              0x00000010
+#define ACISV_ISV8                              0x00000020
+#define ACISV_ISV9                              0x00000040
+#define ACISV_ISV10                             0x00000080
+#define ACISV_ISV11                             0x00000100
+#define ACISV_ISV12                             0x00000200
+
+/*
+ *  The following defines are for the flags in the AC97 status address
+ *  register.
+ */
+#define ACSAD_SI_MASK                           0x0000007F
+#define ACSAD_SI_SHIFT                          0
+
+/*
+ *  The following defines are for the flags in the AC97 status data register.
+ */
+#define ACSDA_SD_MASK                           0x0000FFFF
+#define ACSDA_SD_SHIFT                          0
+
+/*
+ *  The following defines are for the flags in the joystick poll/trigger
+ *  register.
+ */
+#define JSPT_CAX                                0x00000001
+#define JSPT_CAY                                0x00000002
+#define JSPT_CBX                                0x00000004
+#define JSPT_CBY                                0x00000008
+#define JSPT_BA1                                0x00000010
+#define JSPT_BA2                                0x00000020
+#define JSPT_BB1                                0x00000040
+#define JSPT_BB2                                0x00000080
+
+/*
+ *  The following defines are for the flags in the joystick control register.
+ */
+#define JSCTL_SP_MASK                           0x00000003
+#define JSCTL_SP_SLOW                           0x00000000
+#define JSCTL_SP_MEDIUM_SLOW                    0x00000001
+#define JSCTL_SP_MEDIUM_FAST                    0x00000002
+#define JSCTL_SP_FAST                           0x00000003
+#define JSCTL_ARE                               0x00000004
+
+/*
+ *  The following defines are for the flags in the joystick coordinate pair 1
+ *  readback register.
+ */
+#define JSC1_Y1V_MASK                           0x0000FFFF
+#define JSC1_X1V_MASK                           0xFFFF0000
+#define JSC1_Y1V_SHIFT                          0
+#define JSC1_X1V_SHIFT                          16
+
+/*
+ *  The following defines are for the flags in the joystick coordinate pair 2
+ *  readback register.
+ */
+#define JSC2_Y2V_MASK                           0x0000FFFF
+#define JSC2_X2V_MASK                           0xFFFF0000
+#define JSC2_Y2V_SHIFT                          0
+#define JSC2_X2V_SHIFT                          16
+
+/*
+ *  The following defines are for the flags in the MIDI control register.
+ */
+#define MIDCR_TXE                               0x00000001	/* Enable transmitting. */
+#define MIDCR_RXE                               0x00000002	/* Enable receiving. */
+#define MIDCR_RIE                               0x00000004	/* Interrupt upon tx ready. */
+#define MIDCR_TIE                               0x00000008	/* Interrupt upon rx ready. */
+#define MIDCR_MLB                               0x00000010	/* Enable midi loopback. */
+#define MIDCR_MRST                              0x00000020	/* Reset interface. */
+
+/*
+ *  The following defines are for the flags in the MIDI status register.
+ */
+#define MIDSR_TBF                               0x00000001	/* Tx FIFO is full. */
+#define MIDSR_RBE                               0x00000002	/* Rx FIFO is empty. */
+
+/*
+ *  The following defines are for the flags in the MIDI write port register.
+ */
+#define MIDWP_MWD_MASK                          0x000000FF
+#define MIDWP_MWD_SHIFT                         0
+
+/*
+ *  The following defines are for the flags in the MIDI read port register.
+ */
+#define MIDRP_MRD_MASK                          0x000000FF
+#define MIDRP_MRD_SHIFT                         0
+
+/*
+ *  The following defines are for the flags in the joystick GPIO register.
+ */
+#define JSIO_DAX                                0x00000001
+#define JSIO_DAY                                0x00000002
+#define JSIO_DBX                                0x00000004
+#define JSIO_DBY                                0x00000008
+#define JSIO_AXOE                               0x00000010
+#define JSIO_AYOE                               0x00000020
+#define JSIO_BXOE                               0x00000040
+#define JSIO_BYOE                               0x00000080
+
+/*
+ *  The following defines are for the flags in the master async/sync serial
+ *  port enable register.
+ */
+#ifndef NO_CS4612
+#define ASER_MASTER_ME                          0x00000001
+#endif
+
+/*
+ *  The following defines are for the flags in the configuration interface
+ *  register.
+ */
+#define CFGI_CLK                                0x00000001
+#define CFGI_DOUT                               0x00000002
+#define CFGI_DIN_EEN                            0x00000004
+#define CFGI_EELD                               0x00000008
+
+/*
+ *  The following defines are for the flags in the subsystem ID and vendor ID
+ *  register.
+ */
+#define SSVID_VID_MASK                          0x0000FFFF
+#define SSVID_SID_MASK                          0xFFFF0000
+#define SSVID_VID_SHIFT                         0
+#define SSVID_SID_SHIFT                         16
+
+/*
+ *  The following defines are for the flags in the GPIO pin interface register.
+ */
+#define GPIOR_VOLDN                             0x00000001
+#define GPIOR_VOLUP                             0x00000002
+#define GPIOR_SI2D                              0x00000004
+#define GPIOR_SI2OE                             0x00000008
+
+/*
+ *  The following defines are for the flags in the extended GPIO pin direction
+ *  register.
+ */
+#ifndef NO_CS4612
+#define EGPIODR_GPOE0                           0x00000001
+#define EGPIODR_GPOE1                           0x00000002
+#define EGPIODR_GPOE2                           0x00000004
+#define EGPIODR_GPOE3                           0x00000008
+#define EGPIODR_GPOE4                           0x00000010
+#define EGPIODR_GPOE5                           0x00000020
+#define EGPIODR_GPOE6                           0x00000040
+#define EGPIODR_GPOE7                           0x00000080
+#define EGPIODR_GPOE8                           0x00000100
+#endif
+
+/*
+ *  The following defines are for the flags in the extended GPIO pin polarity/
+ *  type register.
+ */
+#ifndef NO_CS4612
+#define EGPIOPTR_GPPT0                          0x00000001
+#define EGPIOPTR_GPPT1                          0x00000002
+#define EGPIOPTR_GPPT2                          0x00000004
+#define EGPIOPTR_GPPT3                          0x00000008
+#define EGPIOPTR_GPPT4                          0x00000010
+#define EGPIOPTR_GPPT5                          0x00000020
+#define EGPIOPTR_GPPT6                          0x00000040
+#define EGPIOPTR_GPPT7                          0x00000080
+#define EGPIOPTR_GPPT8                          0x00000100
+#endif
+
+/*
+ *  The following defines are for the flags in the extended GPIO pin sticky
+ *  register.
+ */
+#ifndef NO_CS4612
+#define EGPIOTR_GPS0                            0x00000001
+#define EGPIOTR_GPS1                            0x00000002
+#define EGPIOTR_GPS2                            0x00000004
+#define EGPIOTR_GPS3                            0x00000008
+#define EGPIOTR_GPS4                            0x00000010
+#define EGPIOTR_GPS5                            0x00000020
+#define EGPIOTR_GPS6                            0x00000040
+#define EGPIOTR_GPS7                            0x00000080
+#define EGPIOTR_GPS8                            0x00000100
+#endif
+
+/*
+ *  The following defines are for the flags in the extended GPIO ping wakeup
+ *  register.
+ */
+#ifndef NO_CS4612
+#define EGPIOWR_GPW0                            0x00000001
+#define EGPIOWR_GPW1                            0x00000002
+#define EGPIOWR_GPW2                            0x00000004
+#define EGPIOWR_GPW3                            0x00000008
+#define EGPIOWR_GPW4                            0x00000010
+#define EGPIOWR_GPW5                            0x00000020
+#define EGPIOWR_GPW6                            0x00000040
+#define EGPIOWR_GPW7                            0x00000080
+#define EGPIOWR_GPW8                            0x00000100
+#endif
+
+/*
+ *  The following defines are for the flags in the extended GPIO pin status
+ *  register.
+ */
+#ifndef NO_CS4612
+#define EGPIOSR_GPS0                            0x00000001
+#define EGPIOSR_GPS1                            0x00000002
+#define EGPIOSR_GPS2                            0x00000004
+#define EGPIOSR_GPS3                            0x00000008
+#define EGPIOSR_GPS4                            0x00000010
+#define EGPIOSR_GPS5                            0x00000020
+#define EGPIOSR_GPS6                            0x00000040
+#define EGPIOSR_GPS7                            0x00000080
+#define EGPIOSR_GPS8                            0x00000100
+#endif
+
+/*
+ *  The following defines are for the flags in the serial port 6 configuration
+ *  register.
+ */
+#ifndef NO_CS4612
+#define SERC6_ASDO2EN                           0x00000001
+#endif
+
+/*
+ *  The following defines are for the flags in the serial port 7 configuration
+ *  register.
+ */
+#ifndef NO_CS4612
+#define SERC7_ASDI2EN                           0x00000001
+#define SERC7_POSILB                            0x00000002
+#define SERC7_SIPOLB                            0x00000004
+#define SERC7_SOSILB                            0x00000008
+#define SERC7_SISOLB                            0x00000010
+#endif
+
+/*
+ *  The following defines are for the flags in the serial port AC link
+ *  configuration register.
+ */
+#ifndef NO_CS4612
+#define SERACC_CHIP_TYPE_MASK                  0x00000001
+#define SERACC_CHIP_TYPE_1_03                  0x00000000
+#define SERACC_CHIP_TYPE_2_0                   0x00000001
+#define SERACC_TWO_CODECS                      0x00000002
+#define SERACC_MDM                             0x00000004
+#define SERACC_HSP                             0x00000008
+#define SERACC_ODT                             0x00000010 /* only CS4630 */
+#endif
+
+/*
+ *  The following defines are for the flags in the AC97 control register 2.
+ */
+#ifndef NO_CS4612
+#define ACCTL2_RSTN                             0x00000001
+#define ACCTL2_ESYN                             0x00000002
+#define ACCTL2_VFRM                             0x00000004
+#define ACCTL2_DCV                              0x00000008
+#define ACCTL2_CRW                              0x00000010
+#define ACCTL2_ASYN                             0x00000020
+#endif
+
+/*
+ *  The following defines are for the flags in the AC97 status register 2.
+ */
+#ifndef NO_CS4612
+#define ACSTS2_CRDY                             0x00000001
+#define ACSTS2_VSTS                             0x00000002
+#endif
+
+/*
+ *  The following defines are for the flags in the AC97 output slot valid
+ *  register 2.
+ */
+#ifndef NO_CS4612
+#define ACOSV2_SLV3                             0x00000001
+#define ACOSV2_SLV4                             0x00000002
+#define ACOSV2_SLV5                             0x00000004
+#define ACOSV2_SLV6                             0x00000008
+#define ACOSV2_SLV7                             0x00000010
+#define ACOSV2_SLV8                             0x00000020
+#define ACOSV2_SLV9                             0x00000040
+#define ACOSV2_SLV10                            0x00000080
+#define ACOSV2_SLV11                            0x00000100
+#define ACOSV2_SLV12                            0x00000200
+#endif
+
+/*
+ *  The following defines are for the flags in the AC97 command address
+ *  register 2.
+ */
+#ifndef NO_CS4612
+#define ACCAD2_CI_MASK                          0x0000007F
+#define ACCAD2_CI_SHIFT                         0
+#endif
+
+/*
+ *  The following defines are for the flags in the AC97 command data register
+ *  2.
+ */
+#ifndef NO_CS4612
+#define ACCDA2_CD_MASK                          0x0000FFFF
+#define ACCDA2_CD_SHIFT                         0  
+#endif
+
+/*
+ *  The following defines are for the flags in the AC97 input slot valid
+ *  register 2.
+ */
+#ifndef NO_CS4612
+#define ACISV2_ISV3                             0x00000001
+#define ACISV2_ISV4                             0x00000002
+#define ACISV2_ISV5                             0x00000004
+#define ACISV2_ISV6                             0x00000008
+#define ACISV2_ISV7                             0x00000010
+#define ACISV2_ISV8                             0x00000020
+#define ACISV2_ISV9                             0x00000040
+#define ACISV2_ISV10                            0x00000080
+#define ACISV2_ISV11                            0x00000100
+#define ACISV2_ISV12                            0x00000200
+#endif
+
+/*
+ *  The following defines are for the flags in the AC97 status address
+ *  register 2.
+ */
+#ifndef NO_CS4612
+#define ACSAD2_SI_MASK                          0x0000007F
+#define ACSAD2_SI_SHIFT                         0
+#endif
+
+/*
+ *  The following defines are for the flags in the AC97 status data register 2.
+ */
+#ifndef NO_CS4612
+#define ACSDA2_SD_MASK                          0x0000FFFF
+#define ACSDA2_SD_SHIFT                         0
+#endif
+
+/*
+ *  The following defines are for the flags in the I/O trap address and control
+ *  registers (all 12).
+ */
+#ifndef NO_CS4612
+#define IOTAC_SA_MASK                           0x0000FFFF
+#define IOTAC_MSK_MASK                          0x000F0000
+#define IOTAC_IODC_MASK                         0x06000000
+#define IOTAC_IODC_16_BIT                       0x00000000
+#define IOTAC_IODC_10_BIT                       0x02000000
+#define IOTAC_IODC_12_BIT                       0x04000000
+#define IOTAC_WSPI                              0x08000000
+#define IOTAC_RSPI                              0x10000000
+#define IOTAC_WSE                               0x20000000
+#define IOTAC_WE                                0x40000000
+#define IOTAC_RE                                0x80000000
+#define IOTAC_SA_SHIFT                          0
+#define IOTAC_MSK_SHIFT                         16
+#endif
+
+/*
+ *  The following defines are for the flags in the I/O trap fast read registers
+ *  (all 8).
+ */
+#ifndef NO_CS4612
+#define IOTFR_D_MASK                            0x0000FFFF
+#define IOTFR_A_MASK                            0x000F0000
+#define IOTFR_R_MASK                            0x0F000000
+#define IOTFR_ALL