Merge rsync://rsync.kernel.org/pub/scm/linux/kernel/git/torvalds/linux-2.6

12 files changed, 986 insertions, 36 deletions

diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 9fef210ab50..c30ff1bb2d1 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 1f3ae3e32d6..c4d2e3507af 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 077fbe25ebf..74d3a35b59b 100644
--- a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
+++ b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
@@ -927,7 +927,7 @@
           <informalexample>
             <programlisting>
 <![CDATA[
-  struct mychip *chip = (struct mychip *)card->private_data;
+  struct mychip *chip = card->private_data;
 ]]>
             </programlisting>
           </informalexample>
@@ -1095,7 +1095,7 @@
 
           /* release the irq */
           if (chip->irq >= 0)
-                  free_irq(chip->irq, (void *)chip);
+                  free_irq(chip->irq, chip);
           /* release the i/o ports & memory */
           pci_release_regions(chip->pci);
           /* disable the PCI entry */
@@ -1148,7 +1148,7 @@
           }
           chip->port = pci_resource_start(pci, 0);
           if (request_irq(pci->irq, snd_mychip_interrupt,
-                          IRQF_DISABLED|IRQF_SHARED, "My Chip", chip)) {
+                          IRQF_SHARED, "My Chip", chip)) {
                   printk(KERN_ERR "cannot grab irq %d\n", pci->irq);
                   snd_mychip_free(chip);
                   return -EBUSY;
@@ -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;
           ....
@@ -1387,7 +1386,7 @@
           <programlisting>
 <![CDATA[
   if (chip->irq >= 0)
-          free_irq(chip->irq, (void *)chip);
+          free_irq(chip->irq, chip);
 ]]>
           </programlisting>
         </informalexample>
@@ -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 0be57ed8130..4eaae2a4553 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 00000000000..58cbfd01ea8
--- /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 00000000000..e93960d53a1
--- /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 00000000000..48983c75aad
--- /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 00000000000..c11877f5b4a
--- /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 00000000000..72bd222f2a2
--- /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 00000000000..753c5cc5984
--- /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 00000000000..e95b16d5a53
--- /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 00000000000..2cf7ee5b3d7
--- /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.