Merge branch 'linus'

Conflicts:

	drivers/scsi/ipr.c

Signed-off-by: James Bottomley <James.Bottomley@SteelEye.com>

21 files changed, 1502 insertions, 105 deletions

diff --git a/Documentation/HOWTO b/Documentation/HOWTO
index 8d51c148f721..48123dba5e6a 100644
--- a/Documentation/HOWTO
+++ b/Documentation/HOWTO
@@ -30,6 +30,7 @@ are not a good substitute for a solid C education and/or years of
 experience, the following books are good for, if anything, reference:
  - "The C Programming Language" by Kernighan and Ritchie [Prentice Hall]
  - "Practical C Programming" by Steve Oualline [O'Reilly]
+ - "C:  A Reference Manual" by Harbison and Steele [Prentice Hall]
 
 The kernel is written using GNU C and the GNU toolchain.  While it
 adheres to the ISO C89 standard, it uses a number of extensions that are
diff --git a/Documentation/crypto/api-intro.txt b/Documentation/crypto/api-intro.txt
index 5a03a2801d67..e41a79aa71ce 100644
--- a/Documentation/crypto/api-intro.txt
+++ b/Documentation/crypto/api-intro.txt
@@ -193,6 +193,7 @@ Original developers of the crypto algorithms:
   Kartikey Mahendra Bhatt (CAST6)
   Jon Oberheide (ARC4)
   Jouni Malinen (Michael MIC)
+  NTT(Nippon Telegraph and Telephone Corporation) (Camellia)
 
 SHA1 algorithm contributors:
   Jean-Francois Dive
@@ -246,6 +247,9 @@ Tiger algorithm contributors:
 VIA PadLock contributors:
   Michal Ludvig
 
+Camellia algorithm contributors:
+  NTT(Nippon Telegraph and Telephone Corporation) (Camellia)
+
 Generic scatterwalk code by Adam J. Richter <adam@yggdrasil.com>
 
 Please send any credits updates or corrections to:
diff --git a/Documentation/driver-model/devres.txt b/Documentation/driver-model/devres.txt
new file mode 100644
index 000000000000..5163b85308f5
--- /dev/null
+++ b/Documentation/driver-model/devres.txt
@@ -0,0 +1,268 @@
+Devres - Managed Device Resource
+================================
+
+Tejun Heo	<teheo@suse.de>
+
+First draft	10 January 2007
+
+
+1. Intro			: Huh? Devres?
+2. Devres			: Devres in a nutshell
+3. Devres Group			: Group devres'es and release them together
+4. Details			: Life time rules, calling context, ...
+5. Overhead			: How much do we have to pay for this?
+6. List of managed interfaces	: Currently implemented managed interfaces
+
+
+  1. Intro
+  --------
+
+devres came up while trying to convert libata to use iomap.  Each
+iomapped address should be kept and unmapped on driver detach.  For
+example, a plain SFF ATA controller (that is, good old PCI IDE) in
+native mode makes use of 5 PCI BARs and all of them should be
+maintained.
+
+As with many other device drivers, libata low level drivers have
+sufficient bugs in ->remove and ->probe failure path.  Well, yes,
+that's probably because libata low level driver developers are lazy
+bunch, but aren't all low level driver developers?  After spending a
+day fiddling with braindamaged hardware with no document or
+braindamaged document, if it's finally working, well, it's working.
+
+For one reason or another, low level drivers don't receive as much
+attention or testing as core code, and bugs on driver detach or
+initilaization failure doesn't happen often enough to be noticeable.
+Init failure path is worse because it's much less travelled while
+needs to handle multiple entry points.
+
+So, many low level drivers end up leaking resources on driver detach
+and having half broken failure path implementation in ->probe() which
+would leak resources or even cause oops when failure occurs.  iomap
+adds more to this mix.  So do msi and msix.
+
+
+  2. Devres
+  ---------
+
+devres is basically linked list of arbitrarily sized memory areas
+associated with a struct device.  Each devres entry is associated with
+a release function.  A devres can be released in several ways.  No
+matter what, all devres entries are released on driver detach.  On
+release, the associated release function is invoked and then the
+devres entry is freed.
+
+Managed interface is created for resources commonly used by device
+drivers using devres.  For example, coherent DMA memory is acquired
+using dma_alloc_coherent().  The managed version is called
+dmam_alloc_coherent().  It is identical to dma_alloc_coherent() except
+for the DMA memory allocated using it is managed and will be
+automatically released on driver detach.  Implementation looks like
+the following.
+
+  struct dma_devres {
+	size_t		size;
+	void		*vaddr;
+	dma_addr_t	dma_handle;
+  };
+
+  static void dmam_coherent_release(struct device *dev, void *res)
+  {
+	struct dma_devres *this = res;
+
+	dma_free_coherent(dev, this->size, this->vaddr, this->dma_handle);
+  }
+
+  dmam_alloc_coherent(dev, size, dma_handle, gfp)
+  {
+	struct dma_devres *dr;
+	void *vaddr;
+
+	dr = devres_alloc(dmam_coherent_release, sizeof(*dr), gfp);
+	...
+
+	/* alloc DMA memory as usual */
+	vaddr = dma_alloc_coherent(...);
+	...
+
+	/* record size, vaddr, dma_handle in dr */
+	dr->vaddr = vaddr;
+	...
+
+	devres_add(dev, dr);
+
+	return vaddr;
+  }
+
+If a driver uses dmam_alloc_coherent(), the area is guaranteed to be
+freed whether initialization fails half-way or the device gets
+detached.  If most resources are acquired using managed interface, a
+driver can have much simpler init and exit code.  Init path basically
+looks like the following.
+
+  my_init_one()
+  {
+	struct mydev *d;
+
+	d = devm_kzalloc(dev, sizeof(*d), GFP_KERNEL);
+	if (!d)
+		return -ENOMEM;
+
+	d->ring = dmam_alloc_coherent(...);
+	if (!d->ring)
+		return -ENOMEM;
+
+	if (check something)
+		return -EINVAL;
+	...
+
+	return register_to_upper_layer(d);
+  }
+
+And exit path,
+
+  my_remove_one()
+  {
+	unregister_from_upper_layer(d);
+	shutdown_my_hardware();
+  }
+
+As shown above, low level drivers can be simplified a lot by using
+devres.  Complexity is shifted from less maintained low level drivers
+to better maintained higher layer.  Also, as init failure path is
+shared with exit path, both can get more testing.
+
+
+  3. Devres group
+  ---------------
+
+Devres entries can be grouped using devres group.  When a group is
+released, all contained normal devres entries and properly nested
+groups are released.  One usage is to rollback series of acquired
+resources on failure.  For example,
+
+  if (!devres_open_group(dev, NULL, GFP_KERNEL))
+	return -ENOMEM;
+
+  acquire A;
+  if (failed)
+	goto err;
+
+  acquire B;
+  if (failed)
+	goto err;
+  ...
+
+  devres_remove_group(dev, NULL);
+  return 0;
+
+ err:
+  devres_release_group(dev, NULL);
+  return err_code;
+
+As resource acquision failure usually means probe failure, constructs
+like above are usually useful in midlayer driver (e.g. libata core
+layer) where interface function shouldn't have side effect on failure.
+For LLDs, just returning error code suffices in most cases.
+
+Each group is identified by void *id.  It can either be explicitly
+specified by @id argument to devres_open_group() or automatically
+created by passing NULL as @id as in the above example.  In both
+cases, devres_open_group() returns the group's id.  The returned id
+can be passed to other devres functions to select the target group.
+If NULL is given to those functions, the latest open group is
+selected.
+
+For example, you can do something like the following.
+
+  int my_midlayer_create_something()
+  {
+	if (!devres_open_group(dev, my_midlayer_create_something, GFP_KERNEL))
+		return -ENOMEM;
+
+	...
+
+	devres_close_group(dev, my_midlayer_something);
+	return 0;
+  }
+
+  void my_midlayer_destroy_something()
+  {
+	devres_release_group(dev, my_midlayer_create_soemthing);
+  }
+
+
+  4. Details
+  ----------
+
+Lifetime of a devres entry begins on devres allocation and finishes
+when it is released or destroyed (removed and freed) - no reference
+counting.
+
+devres core guarantees atomicity to all basic devres operations and
+has support for single-instance devres types (atomic
+lookup-and-add-if-not-found).  Other than that, synchronizing
+concurrent accesses to allocated devres data is caller's
+responsibility.  This is usually non-issue because bus ops and
+resource allocations already do the job.
+
+For an example of single-instance devres type, read pcim_iomap_table()
+in lib/iomap.c.
+
+All devres interface functions can be called without context if the
+right gfp mask is given.
+
+
+  5. Overhead
+  -----------
+
+Each devres bookkeeping info is allocated together with requested data
+area.  With debug option turned off, bookkeeping info occupies 16
+bytes on 32bit machines and 24 bytes on 64bit (three pointers rounded
+up to ull alignment).  If singly linked list is used, it can be
+reduced to two pointers (8 bytes on 32bit, 16 bytes on 64bit).
+
+Each devres group occupies 8 pointers.  It can be reduced to 6 if
+singly linked list is used.
+
+Memory space overhead on ahci controller with two ports is between 300
+and 400 bytes on 32bit machine after naive conversion (we can
+certainly invest a bit more effort into libata core layer).
+
+
+  6. List of managed interfaces
+  -----------------------------
+
+IO region
+  devm_request_region()
+  devm_request_mem_region()
+  devm_release_region()
+  devm_release_mem_region()
+
+IRQ
+  devm_request_irq()
+  devm_free_irq()
+
+DMA
+  dmam_alloc_coherent()
+  dmam_free_coherent()
+  dmam_alloc_noncoherent()
+  dmam_free_noncoherent()
+  dmam_declare_coherent_memory()
+  dmam_pool_create()
+  dmam_pool_destroy()
+
+PCI
+  pcim_enable_device()	: after success, all PCI ops become managed
+  pcim_pin_device()	: keep PCI device enabled after release
+
+IOMAP
+  devm_ioport_map()
+  devm_ioport_unmap()
+  devm_ioremap()
+  devm_ioremap_nocache()
+  devm_iounmap()
+  pcim_iomap()
+  pcim_iounmap()
+  pcim_iomap_table()	: array of mapped addresses indexed by BAR
+  pcim_iomap_regions()	: do request_region() and iomap() on multiple BARs
diff --git a/Documentation/feature-removal-schedule.txt b/Documentation/feature-removal-schedule.txt
index 0ba6af02cdaf..fa844fd7bded 100644
--- a/Documentation/feature-removal-schedule.txt
+++ b/Documentation/feature-removal-schedule.txt
@@ -50,22 +50,6 @@ Who:	Dan Dennedy <dan@dennedy.org>, Stefan Richter <stefanr@s5r6.in-berlin.de>
 
 ---------------------------
 
-What:	ieee1394 core's unused exports (CONFIG_IEEE1394_EXPORT_FULL_API)
-When:	January 2007
-Why:	There are no projects known to use these exported symbols, except
-	dfg1394 (uses one symbol whose functionality is core-internal now).
-Who:	Stefan Richter <stefanr@s5r6.in-berlin.de>
-
----------------------------
-
-What:	ieee1394's *_oui sysfs attributes (CONFIG_IEEE1394_OUI_DB)
-When:	January 2007
-Files:	drivers/ieee1394/: oui.db, oui2c.sh
-Why:	big size, little value
-Who:	Stefan Richter <stefanr@s5r6.in-berlin.de>
-
----------------------------
-
 What:	Video4Linux API 1 ioctls and video_decoder.h from Video devices.
 When:	December 2006
 Why:	V4L1 AP1 was replaced by V4L2 API. during migration from 2.4 to 2.6
@@ -186,18 +170,6 @@ Who:	Greg Kroah-Hartman <gregkh@suse.de>
 
 ---------------------------
 
-What:	find_trylock_page
-When:	January 2007
-Why:	The interface no longer has any callers left in the kernel. It
-	is an odd interface (compared with other find_*_page functions), in
-	that it does not take a refcount to the page, only the page lock.
-	It should be replaced with find_get_page or find_lock_page if possible.
-	This feature removal can be reevaluated if users of the interface
-	cannot cleanly use something else.
-Who:	Nick Piggin <npiggin@suse.de>
-
----------------------------
-
 What:	Interrupt only SA_* flags
 When:	Januar 2007
 Why:	The interrupt related SA_* flags are replaced by IRQF_* to move them
@@ -274,6 +246,7 @@ Who:	Venkatesh Pallipadi <venkatesh.pallipadi@intel.com>
 
 ---------------------------
 
+<<<<<<< test:Documentation/feature-removal-schedule.txt
 What:	ACPI hotkey driver (CONFIG_ACPI_HOTKEY)
 When:	2.6.21
 Why:	hotkey.c was an attempt to consolidate multiple drivers that use
@@ -306,11 +279,18 @@ Why:	The ACPI namespace is effectively the symbol list for
 	the BIOS can be extracted and disassembled with acpidump
 	and iasl as documented in the pmtools package here:
 	http://ftp.kernel.org/pub/linux/kernel/people/lenb/acpi/utils
-
 Who:	Len Brown <len.brown@intel.com>
 
 ---------------------------
 
+What:	ACPI procfs interface
+When:	July 2007
+Why:	After ACPI sysfs conversion, ACPI attributes will be duplicated
+	in sysfs and the ACPI procfs interface should be removed.
+Who:	Zhang Rui <rui.zhang@intel.com>
+
+---------------------------
+
 What:	/proc/acpi/button
 When:	August 2007
 Why:	/proc/acpi/button has been replaced by events to the input layer
@@ -325,3 +305,10 @@ Why:	Unmaintained for years, superceded by JFFS2 for years.
 Who:	Jeff Garzik <jeff@garzik.org>
 
 ---------------------------
+
+What:   sk98lin network driver
+When:   July 2007
+Why:    In kernel tree version of driver is unmaintained. Sk98lin driver
+	replaced by the skge driver. 
+Who:    Stephen Hemminger <shemminger@osdl.org>
+
diff --git a/Documentation/s390/Debugging390.txt b/Documentation/s390/Debugging390.txt
index 3f9ddbc23b27..0993969609cf 100644
--- a/Documentation/s390/Debugging390.txt
+++ b/Documentation/s390/Debugging390.txt
@@ -480,7 +480,7 @@ r2       argument 0 / return value 0                call-clobbered
 r3       argument 1 / return value 1 (if long long) call-clobbered
 r4       argument 2                                 call-clobbered
 r5       argument 3                                 call-clobbered
-r6	 argument 5                                 saved
+r6	 argument 4				    saved
 r7       pointer-to arguments 5 to ...              saved      
 r8       this & that                                saved
 r9       this & that                                saved
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 9fef210ab50a..c30ff1bb2d10 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -242,6 +242,12 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     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.
@@ -779,6 +785,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  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
@@ -800,14 +807,18 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	ALC262
 	  fujitsu	Fujitsu Laptop
 	  hp-bpc	HP xw4400/6400/8400/9400 laptops
+	  hp-bpc-d7000	HP BPC D7000
 	  benq		Benq ED8
+	  hippo		Hippo (ATI) with jack detection, Sony UX-90s
+	  hippo_1	Hippo (Benq) with jack detection
 	  basic		fixed pin assignment w/o SPDIF
 	  auto		auto-config reading BIOS (default)
 
 	ALC882/885
 	  3stack-dig	3-jack with SPDIF I/O
-	  6stck-dig	6-jack digital with SPDIF I/O
+	  6stack-dig	6-jack digital with SPDIF I/O
 	  arima		Arima W820Di1
+	  macpro	MacPro support
 	  auto		auto-config reading BIOS (default)
 
 	ALC883/888
@@ -817,6 +828,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  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)
+	  medion	Medion Laptops
+	  targa-dig	Targa/MSI
+	  targa-2ch-dig	Targs/MSI with 2-channel
+	  laptop-eapd   3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE)
 	  auto		auto-config reading BIOS (default)
 
 	ALC861/660
@@ -825,6 +840,16 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  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)
 	  auto		auto-config reading BIOS (default)
 
 	CMI9880
@@ -845,6 +870,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  3stack	3-stack, shared surrounds
 	  laptop	2-channel only (FSC V2060, Samsung M50)
 	  laptop-eapd	2-channel with EAPD (Samsung R65, ASUS A6J)
+	  ultra		2-channel with EAPD (Samsung Ultra tablet PC)
 
 	AD1988
 	  6stack	6-jack
@@ -854,12 +880,31 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 	  laptop	3-jack with hp-jack automute
 	  laptop-dig	ditto with SPDIF
 	  auto		auto-config reading BIOS (default)
+	
+	Conexant 5045
+	  laptop	Laptop config 
+	  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
 
 	STAC9200/9205/9220/9221/9254
 	  ref		Reference board
 	  3stack	D945 3stack
 	  5stack	D945 5stack + SPDIF
 
+	STAC9202/9250/9251
+	  ref		Reference board, base config
+	  m2-2		Some Gateway MX series laptops
+	  m6		Some Gateway NX series laptops
+
 	STAC9227/9228/9229/927x
 	  ref		Reference board
 	  3stack	D965 3stack
@@ -974,6 +1019,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     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
@@ -993,7 +1039,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     model       - Use the given board model, one of the following:
 		  revo51, revo71, amp2000, prodigy71, prodigy71lt,
-		  prodigy192, aureon51, aureon71, universe,
+		  prodigy192, aureon51, aureon71, universe, ap192,
 		  k8x800, phase22, phase28, ms300, av710
 
     This module supports multiple cards and autoprobe.
@@ -1049,6 +1095,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     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.
 
@@ -1371,6 +1420,13 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     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)
   ---------------------------------
 
diff --git a/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl b/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl
index 1f3ae3e32d69..c4d2e3507af9 100644
--- a/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl
+++ b/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl
@@ -36,7 +36,7 @@
   </bookinfo>
 
   <chapter><title>Management of Cards and Devices</title>
-     <sect1><title>Card Managment</title>
+     <sect1><title>Card Management</title>
 !Esound/core/init.c
      </sect1>
      <sect1><title>Device Components</title>
@@ -59,7 +59,7 @@
      <sect1><title>PCM Format Helpers</title>
 !Esound/core/pcm_misc.c
      </sect1>
-     <sect1><title>PCM Memory Managment</title>
+     <sect1><title>PCM Memory Management</title>
 !Esound/core/pcm_memory.c
      </sect1>
   </chapter>
diff --git a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
index ccd0a953953d..74d3a35b59bc 100644
--- a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
+++ b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
@@ -1360,8 +1360,7 @@
         <informalexample>
           <programlisting>
 <![CDATA[
-  static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id,
-                                          struct pt_regs *regs)
+  static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id)
   {
           struct mychip *chip = dev_id;
           ....
@@ -2127,7 +2126,7 @@
 	accessible via <constant>substream-&gt;runtime</constant>.
 	This runtime pointer holds the various information; it holds
 	the copy of hw_params and sw_params configurations, the buffer
-	pointers, mmap records, spinlocks, etc.  Almost everyhing you
+	pointers, mmap records, spinlocks, etc.  Almost everything you
 	need for controlling the PCM can be found there.
 	</para>
 
@@ -2340,7 +2339,7 @@ struct _snd_pcm_runtime {
 
 	<para>
 	  When the PCM substreams can be synchronized (typically,
-	synchorinized start/stop of a playback and a capture streams),
+	synchronized start/stop of a playback and a capture streams),
 	you can give <constant>SNDRV_PCM_INFO_SYNC_START</constant>,
 	too.  In this case, you'll need to check the linked-list of
 	PCM substreams in the trigger callback.  This will be
@@ -3062,8 +3061,7 @@ struct _snd_pcm_runtime {
 	    <title>Interrupt Handler Case #1</title>
             <programlisting>
 <![CDATA[
-  static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id,
-                                          struct pt_regs *regs)
+  static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id)
   {
           struct mychip *chip = dev_id;
           spin_lock(&chip->lock);
@@ -3106,8 +3104,7 @@ struct _snd_pcm_runtime {
 	    <title>Interrupt Handler Case #2</title>
             <programlisting>
 <![CDATA[
-  static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id,
-                                          struct pt_regs *regs)
+  static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id)
   {
           struct mychip *chip = dev_id;
           spin_lock(&chip->lock);
@@ -3247,7 +3244,7 @@ struct _snd_pcm_runtime {
         You can even define your own constraint rules.
         For example, let's suppose my_chip can manage a substream of 1 channel
         if and only if the format is S16_LE, otherwise it supports any format
-        specified in the <structname>snd_pcm_hardware</structname> stucture (or in any
+        specified in the <structname>snd_pcm_hardware</structname> structure (or in any
         other constraint_list). You can build a rule like this:
 
         <example>
@@ -3691,16 +3688,6 @@ struct _snd_pcm_runtime {
         </para>
 
         <para>
-          Here, the chip instance is retrieved via
-        <function>snd_kcontrol_chip()</function> macro.  This macro
-        just accesses to kcontrol-&gt;private_data. The
-        kcontrol-&gt;private_data field is 
-        given as the argument of <function>snd_ctl_new()</function>
-        (see the later subsection
-        <link linkend="control-interface-constructor"><citetitle>Constructor</citetitle></link>).
-        </para>
-
-        <para>
 	The <structfield>value</structfield> field is depending on
         the type of control as well as on info callback.  For example,
 	the sb driver uses this field to store the register offset,
@@ -3780,7 +3767,7 @@ struct _snd_pcm_runtime {
         <para>
 	Like <structfield>get</structfield> callback,
 	when the control has more than one elements,
-	all elemehts must be evaluated in this callback, too.
+	all elements must be evaluated in this callback, too.
         </para>
       </section>
 
@@ -5541,12 +5528,12 @@ struct _snd_pcm_runtime {
   #ifdef CONFIG_PM
   static int snd_my_suspend(struct pci_dev *pci, pm_message_t state)
   {
-          .... /* do things for suspsend */
+          .... /* do things for suspend */
           return 0;
   }
   static int snd_my_resume(struct pci_dev *pci)
   {
-          .... /* do things for suspsend */
+          .... /* do things for suspend */
           return 0;
   }
   #endif
@@ -6111,7 +6098,7 @@ struct _snd_pcm_runtime {
 <!-- ****************************************************** -->
 <!-- Acknowledgments  -->
 <!-- ****************************************************** -->
-  <chapter id="acknowledments">
+  <chapter id="acknowledgments">
     <title>Acknowledgments</title>
     <para>
       I would like to thank Phil Kerr for his help for improvement and
diff --git a/Documentation/sound/alsa/hda_codec.txt b/Documentation/sound/alsa/hda_codec.txt
index 0be57ed81302..4eaae2a45534 100644
--- a/Documentation/sound/alsa/hda_codec.txt
+++ b/Documentation/sound/alsa/hda_codec.txt
@@ -277,11 +277,11 @@ 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 table with struct
-hda_board_config 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.
+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.  The same array
diff --git a/Documentation/sound/alsa/soc/DAI.txt b/Documentation/sound/alsa/soc/DAI.txt
new file mode 100644
index 000000000000..58cbfd01ea8f
--- /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 transmision, 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 LRCLK's, this allows for similtanious 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 BCLK's before LRC
+                     transition.
+
+PCM
+===
+
+PCM is another 4 wire interface, very similar to I2S, that 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 similtaniuosly (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 000000000000..e93960d53a1e
--- /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 refered 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. PLL's and CPU based clocks) are configuarble in that
+their speed can be altered by software (depending on the system use and to save
+power). Other master clocks are fixed at at 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's best to configure BCLK to the lowest possible speed (depending on your
+rate, number of channels and wordsize) to save on power.
+
+It's also desireable to use the codec (if possible) to drive (or master) the
+audio clocks as it's 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 000000000000..48983c75aad9
--- /dev/null
+++ b/Documentation/sound/alsa/soc/codec.txt
@@ -0,0 +1,197 @@
+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 API's
+ 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.
+
+It's probably best to use this guide in conjuction 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 it's DAI and
+PCM's capablities 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 ususally be controlled via an I2C or SPI style interface (AC97
+combines control with data in the DAI). The codec drivers will have to 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's power
+components, 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's 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 DAC's 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 000000000000..c11877f5b4a1
--- /dev/null
+++ b/Documentation/sound/alsa/soc/dapm.txt
@@ -0,0 +1,297 @@
+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 descisions 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 - DAC's and ADC's.
+      Enabled and disabled when stream playback/capture is started and
+      stopped respectively. e.g. aplay, arecord.
+
+All DAPM power switching descisons 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 1 of it's 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 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
+convience 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 ADC's
+(analog to digital converters) and DAC's (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 codecs
+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 effect 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)),
+
+
+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)
+{
+	if(SND_SOC_DAPM_EVENT_ON(event))
+		set_scoop_gpio(&spitzscoop2_device.dev, SPITZ_SCP2_MIC_BIAS);
+	else
+		reset_scoop_gpio(&spitzscoop2_device.dev, SPITZ_SCP2_MIC_BIAS);
+
+	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 register bit for power control. In this case it's 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.
+
+i.e. from the WM8731 codec's 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)
+{
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON);
+	else
+		reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON);
+
+	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/machine.txt b/Documentation/sound/alsa/soc/machine.txt
new file mode 100644
index 000000000000..72bd222f2a21
--- /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_machine {
+	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 DAI's 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, DAI's
+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 DAI's 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, unconnnected 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_machine then sets up the machine with it's DAI's. e.g.
+
+/* corgi audio machine driver */
+static struct snd_soc_machine snd_soc_machine_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_machine_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.
\ No newline at end of file
diff --git a/Documentation/sound/alsa/soc/overview.txt b/Documentation/sound/alsa/soc/overview.txt
new file mode 100644
index 000000000000..753c5cc5984a
--- /dev/null
+++ b/Documentation/sound/alsa/soc/overview.txt
@@ -0,0 +1,83 @@
+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 procesors (e.g. pxa2xx, au1x00,
+iMX, etc) and portable audio codecs. Currently there is some support in the
+kernel for SoC audio, however it has some limitations:-
+
+  * Currently, codec drivers are often tightly coupled to the underlying SoC
+    cpu. This is not ideal and leads to code duplication i.e. Linux now has 4
+    different wm8731 drivers for 4 different SoC platforms.
+
+  * There is 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 ofter require
+    machine specific code to re route audio, enable amps etc after such an event.
+
+  * Current drivers tend 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 is 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 hw params are known.
+
+  * Dynamic Audio Power Management (DAPM). DAPM automatically sets the codec to
+    it's 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 amp.
+
+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. i.e. turing 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.
\ No newline at end of file
diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/alsa/soc/platform.txt
new file mode 100644
index 000000000000..e95b16d5a53b
--- /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 it's DMA functionailty 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 000000000000..2cf7ee5b3d74
--- /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 noticable on PC's 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 atm,
+however future audio codec hardware will have better pop and click supression.
+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 occured. 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/Documentation/sysrq.txt b/Documentation/sysrq.txt
index e0188a23fd5e..61613166981b 100644
--- a/Documentation/sysrq.txt
+++ b/Documentation/sysrq.txt
@@ -1,6 +1,6 @@
 Linux Magic System Request Key Hacks
-Documentation for sysrq.c version 1.15
-Last update: $Date: 2001/01/28 10:15:59 $
+Documentation for sysrq.c
+Last update: 2007-JAN-06
 
 *  What is the magic SysRq key?
 ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
@@ -35,7 +35,7 @@ You can set the value in the file by the following command:
 
 Note that the value of /proc/sys/kernel/sysrq influences only the invocation
 via a keyboard. Invocation of any operation via /proc/sysrq-trigger is always
-allowed.
+allowed (by a user with admin privileges).
 
 *  How do I use the magic SysRq key?
 ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
@@ -58,7 +58,7 @@ On PowerPC - Press 'ALT - Print Screen (or F13) - <command key>,
 On other - If you know of the key combos for other architectures, please
            let me know so I can add them to this section.
 
-On all -  write a character to /proc/sysrq-trigger.  eg:
+On all -  write a character to /proc/sysrq-trigger.  e.g.:
 
 		echo t > /proc/sysrq-trigger
 
@@ -74,6 +74,8 @@ On all -  write a character to /proc/sysrq-trigger.  eg:
 
 'c'	- Will perform a kexec reboot in order to take a crashdump.
 
+'d'	- Shows all locks that are held.
+
 'o'     - Will shut your system off (if configured and supported).
 
 's'     - Will attempt to sync all mounted filesystems.
@@ -87,38 +89,43 @@ On all -  write a character to /proc/sysrq-trigger.  eg:
 
 'm'     - Will dump current memory info to your console.
 
+'n'	- Used to make RT tasks nice-able
+
 'v'	- Dumps Voyager SMP processor info to your console.
 
+'w'	- Dumps tasks that are in uninterruptable (blocked) state.
+
+'x'	- Used by xmon interface on ppc/powerpc platforms.
+
 '0'-'9' - Sets the console log level, controlling which kernel messages
           will be printed to your console. ('0', for example would make
           it so that only emergency messages like PANICs or OOPSes would
           make it to your console.)
 
-'f'	- Will call oom_kill to kill a memory hog process
+'f'	- Will call oom_kill to kill a memory hog process.
 
 'e'     - Send a SIGTERM to all processes, except for init.
 
-'i'     - Send a SIGKILL to all processes, except for init.
+'g'	- Used by kgdb on ppc platforms.
 
-'l'     - Send a SIGKILL to all processes, INCLUDING init. (Your system
-          will be non-functional after this.)
+'i'     - Send a SIGKILL to all processes, except for init.
 
-'h'     - Will display help ( actually any other key than those listed
+'h'     - Will display help (actually any other key than those listed
           above will display help. but 'h' is easy to remember :-)
 
 *  Okay, so what can I use them for?
 ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
 Well, un'R'aw is very handy when your X server or a svgalib program crashes.
 
-sa'K' (Secure Access Key) is useful when you want to be sure there are no
-trojan program is running at console and which could grab your password
-when you would try to login. It will kill all programs on given console
-and thus letting you make sure that the login prompt you see is actually
+sa'K' (Secure Access Key) is useful when you want to be sure there is no
+trojan program running at console which could grab your password
+when you would try to login. It will kill all programs on given console,
+thus letting you make sure that the login prompt you see is actually
 the one from init, not some trojan program.
 IMPORTANT: In its true form it is not a true SAK like the one in a :IMPORTANT
 IMPORTANT: c2 compliant system, and it should not be mistaken as   :IMPORTANT
 IMPORTANT: such.                                                   :IMPORTANT
-       It seems other find it useful as (System Attention Key) which is
+       It seems others find it useful as (System Attention Key) which is
 useful when you want to exit a program that will not let you switch consoles.
 (For example, X or a svgalib program.)
 
@@ -139,8 +146,8 @@ OK or Done message...)
 Again, the unmount (remount read-only) hasn't taken place until you see the
 "OK" and "Done" message appear on the screen.
 
-The loglevel'0'-'9' is useful when your console is being flooded with
-kernel messages you do not want to see. Setting '0' will prevent all but
+The loglevels '0'-'9' are useful when your console is being flooded with
+kernel messages you do not want to see. Selecting '0' will prevent all but
 the most urgent kernel messages from reaching your console. (They will
 still be logged if syslogd/klogd are alive, though.)
 
@@ -152,7 +159,7 @@ processes.
 ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
 That happens to me, also. I've found that tapping shift, alt, and control
 on both sides of the keyboard, and hitting an invalid sysrq sequence again
-will fix the problem. (ie, something like alt-sysrq-z). Switching to another
+will fix the problem. (i.e., something like alt-sysrq-z). Switching to another
 virtual console (ALT+Fn) and then back again should also help.
 
 *  I hit SysRq, but nothing seems to happen, what's wrong?
@@ -174,11 +181,11 @@ handler function you will use, B) a help_msg string, that will print when SysRQ
 prints help, and C) an action_msg string, that will print right before your
 handler is called. Your handler must conform to the prototype in 'sysrq.h'.
 
-After the sysrq_key_op is created, you can call the macro 
-register_sysrq_key(int key, struct sysrq_key_op *op_p) that is defined in
-sysrq.h, this will register the operation pointed to by 'op_p' at table
-key 'key', if that slot in the table is blank. At module unload time, you must
-call the macro unregister_sysrq_key(int key, struct sysrq_key_op *op_p), which
+After the sysrq_key_op is created, you can call the kernel function
+register_sysrq_key(int key, struct sysrq_key_op *op_p); this will
+register the operation pointed to by 'op_p' at table key 'key',
+if that slot in the table is blank. At module unload time, you must call
+the function unregister_sysrq_key(int key, struct sysrq_key_op *op_p), which
 will remove the key op pointed to by 'op_p' from the key 'key', if and only if
 it is currently registered in that slot. This is in case the slot has been
 overwritten since you registered it.
@@ -186,15 +193,12 @@ overwritten since you registered it.
 The Magic SysRQ system works by registering key operations against a key op
 lookup table, which is defined in 'drivers/char/sysrq.c'. This key table has
 a number of operations registered into it at compile time, but is mutable,
-and 4 functions are exported for interface to it: __sysrq_lock_table,
-__sysrq_unlock_table, __sysrq_get_key_op, and __sysrq_put_key_op. The
-functions __sysrq_swap_key_ops and __sysrq_swap_key_ops_nolock are defined
-in the header itself, and the REGISTER and UNREGISTER macros are built from
-these. More complex (and dangerous!) manipulations of the table are possible
-using these functions, but you must be careful to always lock the table before
-you read or write from it, and to unlock it again when you are done. (And of
-course, to never ever leave an invalid pointer in the table). Null pointers in
-the table are always safe :)
+and 2 functions are exported for interface to it:
+	register_sysrq_key and unregister_sysrq_key.
+Of course, never ever leave an invalid pointer in the table. I.e., when
+your module that called register_sysrq_key() exits, it must call
+unregister_sysrq_key() to clean up the sysrq key table entry that it used.
+Null pointers in the table are always safe. :)
 
 If for some reason you feel the need to call the handle_sysrq function from
 within a function called by handle_sysrq, you must be aware that you are in
diff --git a/Documentation/usb/proc_usb_info.txt b/Documentation/usb/proc_usb_info.txt
index 22c5331260ca..077e9032d0cd 100644
--- a/Documentation/usb/proc_usb_info.txt
+++ b/Documentation/usb/proc_usb_info.txt
@@ -213,15 +213,16 @@ C:* #Ifs=dd Cfg#=dd Atr=xx MPwr=dddmA
 
 Interface descriptor info (can be multiple per Config):
 
-I:  If#=dd Alt=dd #EPs=dd Cls=xx(sssss) Sub=xx Prot=xx Driver=ssss
-|   |      |      |       |             |      |       |__Driver name
-|   |      |      |       |             |      |          or "(none)"
-|   |      |      |       |             |      |__InterfaceProtocol
-|   |      |      |       |             |__InterfaceSubClass
-|   |      |      |       |__InterfaceClass
-|   |      |      |__NumberOfEndpoints
-|   |      |__AlternateSettingNumber
-|   |__InterfaceNumber
+I:* If#=dd Alt=dd #EPs=dd Cls=xx(sssss) Sub=xx Prot=xx Driver=ssss
+| | |      |      |       |             |      |       |__Driver name
+| | |      |      |       |             |      |          or "(none)"
+| | |      |      |       |             |      |__InterfaceProtocol
+| | |      |      |       |             |__InterfaceSubClass
+| | |      |      |       |__InterfaceClass
+| | |      |      |__NumberOfEndpoints
+| | |      |__AlternateSettingNumber
+| | |__InterfaceNumber
+| |__ "*" indicates the active altsetting (others are " ")
 |__Interface info tag
 
     A given interface may have one or more "alternate" settings.
@@ -277,7 +278,7 @@ of the USB devices on a system's root hub.  (See more below
 on how to do this.)
 
 The Interface lines can be used to determine what driver is
-being used for each device.
+being used for each device, and which altsetting it activated.
 
 The Configuration lines could be used to list maximum power
 (in milliamps) that a system's USB devices are using.
diff --git a/Documentation/usb/usbmon.txt b/Documentation/usb/usbmon.txt
index e65ec828d7aa..0f6808abd612 100644
--- a/Documentation/usb/usbmon.txt
+++ b/Documentation/usb/usbmon.txt
@@ -77,7 +77,7 @@ that the file size is not excessive for your favourite editor.
 
 The '1t' type data consists of a stream of events, such as URB submission,
 URB callback, submission error. Every event is a text line, which consists
-of whitespace separated words. The number of position of words may depend
+of whitespace separated words. The number or position of words may depend
 on the event type, but there is a set of words, common for all types.
 
 Here is the list of words, from left to right:
@@ -170,4 +170,152 @@ dd65f0e8 4128379808 C Bo:005:02 0 31 >
 
 * Raw binary format and API
 
-TBD
+The overall architecture of the API is about the same as the one above,
+only the events are delivered in binary format. Each event is sent in
+the following structure (its name is made up, so that we can refer to it):
+
+struct usbmon_packet {
+	u64 id;			/*  0: URB ID - from submission to callback */
+	unsigned char type;	/*  8: Same as text; extensible. */
+	unsigned char xfer_type; /*    ISO (0), Intr, Control, Bulk (3) */
+	unsigned char epnum;	/*     Endpoint number and transfer direction */
+	unsigned char devnum;	/*     Device address */
+	u16 busnum;		/* 12: Bus number */
+	char flag_setup;	/* 14: Same as text */
+	char flag_data;		/* 15: Same as text; Binary zero is OK. */
+	s64 ts_sec;		/* 16: gettimeofday */
+	s32 ts_usec;		/* 24: gettimeofday */
+	int status;		/* 28: */
+	unsigned int length;	/* 32: Length of data (submitted or actual) */
+	unsigned int len_cap;	/* 36: Delivered length */
+	unsigned char setup[8];	/* 40: Only for Control 'S' */
+};				/* 48 bytes total */
+
+These events can be received from a character device by reading with read(2),
+with an ioctl(2), or by accessing the buffer with mmap.
+
+The character device is usually called /dev/usbmonN, where N is the USB bus
+number. Number zero (/dev/usbmon0) is special and means "all buses".
+However, this feature is not implemented yet. Note that specific naming
+policy is set by your Linux distribution.
+
+If you create /dev/usbmon0 by hand, make sure that it is owned by root
+and has mode 0600. Otherwise, unpriviledged users will be able to snoop
+keyboard traffic.
+
+The following ioctl calls are available, with MON_IOC_MAGIC 0x92:
+
+ MON_IOCQ_URB_LEN, defined as _IO(MON_IOC_MAGIC, 1)
+
+This call returns the length of data in the next event. Note that majority of
+events contain no data, so if this call returns zero, it does not mean that
+no events are available.
+
+ MON_IOCG_STATS, defined as _IOR(MON_IOC_MAGIC, 3, struct mon_bin_stats)
+
+The argument is a pointer to the following structure:
+
+struct mon_bin_stats {
+	u32 queued;
+	u32 dropped;
+};
+
+The member "queued" refers to the number of events currently queued in the
+buffer (and not to the number of events processed since the last reset).
+
+The member "dropped" is the number of events lost since the last call
+to MON_IOCG_STATS.
+
+ MON_IOCT_RING_SIZE, defined as _IO(MON_IOC_MAGIC, 4)
+
+This call sets the buffer size. The argument is the size in bytes.
+The size may be rounded down to the next chunk (or page). If the requested
+size is out of [unspecified] bounds for this kernel, the call fails with
+-EINVAL.
+
+ MON_IOCQ_RING_SIZE, defined as _IO(MON_IOC_MAGIC, 5)
+
+This call returns the current size of the buffer in bytes.
+
+ MON_IOCX_GET, defined as _IOW(MON_IOC_MAGIC, 6, struct mon_get_arg)
+
+This call waits for events to arrive if none were in the kernel buffer,
+then returns the first event. Its argument is a pointer to the following
+structure:
+
+struct mon_get_arg {
+	struct usbmon_packet *hdr;
+	void *data;
+	size_t alloc;		/* Length of data (can be zero) */
+};
+
+Before the call, hdr, data, and alloc should be filled. Upon return, the area
+pointed by hdr contains the next event structure, and the data buffer contains
+the data, if any. The event is removed from the kernel buffer.
+
+ MON_IOCX_MFETCH, defined as _IOWR(MON_IOC_MAGIC, 7, struct mon_mfetch_arg)
+
+This ioctl is primarily used when the application accesses the buffer
+with mmap(2). Its argument is a pointer to the following structure:
+
+struct mon_mfetch_arg {
+	uint32_t *offvec;	/* Vector of events fetched */
+	uint32_t nfetch;	/* Number of events to fetch (out: fetched) */
+	uint32_t nflush;	/* Number of events to flush */
+};
+
+The ioctl operates in 3 stages.
+
+First, it removes and discards up to nflush events from the kernel buffer.
+The actual number of events discarded is returned in nflush.
+
+Second, it waits for an event to be present in the buffer, unless the pseudo-
+device is open with O_NONBLOCK.
+
+Third, it extracts up to nfetch offsets into the mmap buffer, and stores
+them into the offvec. The actual number of event offsets is stored into
+the nfetch.
+
+ MON_IOCH_MFLUSH, defined as _IO(MON_IOC_MAGIC, 8)
+
+This call removes a number of events from the kernel buffer. Its argument
+is the number of events to remove. If the buffer contains fewer events
+than requested, all events present are removed, and no error is reported.
+This works when no events are available too.
+
+ FIONBIO
+
+The ioctl FIONBIO may be implemented in the future, if there's a need.
+
+In addition to ioctl(2) and read(2), the special file of binary API can
+be polled with select(2) and poll(2). But lseek(2) does not work.
+
+* Memory-mapped access of the kernel buffer for the binary API
+
+The basic idea is simple:
+
+To prepare, map the buffer by getting the current size, then using mmap(2).
+Then, execute a loop similar to the one written in pseudo-code below:
+
+   struct mon_mfetch_arg fetch;
+   struct usbmon_packet *hdr;
+   int nflush = 0;
+   for (;;) {
+      fetch.offvec = vec; // Has N 32-bit words
+      fetch.nfetch = N;   // Or less than N
+      fetch.nflush = nflush;
+      ioctl(fd, MON_IOCX_MFETCH, &fetch);   // Process errors, too
+      nflush = fetch.nfetch;       // This many packets to flush when done
+      for (i = 0; i < nflush; i++) {
+         hdr = (struct ubsmon_packet *) &mmap_area[vec[i]];
+         if (hdr->type == '@')     // Filler packet
+            continue;
+         caddr_t data = &mmap_area[vec[i]] + 64;
+         process_packet(hdr, data);
+      }
+   }
+
+Thus, the main idea is to execute only one ioctl per N events.
+
+Although the buffer is circular, the returned headers and data do not cross
+the end of the buffer, so the above pseudo-code does not need any gathering.
diff --git a/Documentation/video-output.txt b/Documentation/video-output.txt
new file mode 100644
index 000000000000..e517011be4f9
--- /dev/null
+++ b/Documentation/video-output.txt
@@ -0,0 +1,34 @@
+
+		Video Output Switcher Control
+		~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+		2006 luming.yu@intel.com
+
+The output sysfs class driver provides an abstract video output layer that
+can be used to hook platform specific methods to enable/disable video output
+device through common sysfs interface. For example, on my IBM ThinkPad T42
+laptop, The ACPI video driver registered its output devices and read/write
+method for 'state' with output sysfs class. The user interface under sysfs is:
+
+linux:/sys/class/video_output # tree .
+.
+|-- CRT0
+|   |-- device -> ../../../devices/pci0000:00/0000:00:01.0
+|   |-- state
+|   |-- subsystem -> ../../../class/video_output
+|   `-- uevent
+|-- DVI0
+|   |-- device -> ../../../devices/pci0000:00/0000:00:01.0
+|   |-- state
+|   |-- subsystem -> ../../../class/video_output
+|   `-- uevent
+|-- LCD0
+|   |-- device -> ../../../devices/pci0000:00/0000:00:01.0
+|   |-- state
+|   |-- subsystem -> ../../../class/video_output
+|   `-- uevent
+`-- TV0
+   |-- device -> ../../../devices/pci0000:00/0000:00:01.0
+   |-- state
+   |-- subsystem -> ../../../class/video_output
+   `-- uevent
+