Patched in Tegra support.

11 files changed, 187 insertions, 480 deletions

diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index b9cfd339a6f..89757012c7f 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -860,14 +860,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     [Multiple options for each card instance]
     model       - force the model name
-    position_fix - Fix DMA pointer
-                  -1 = system default: choose appropriate one per controller
-                        hardware
-                  0 = auto: falls back to LPIB when POSBUF doesn't work
-                  1 = use LPIB
-                  2 = POSBUF: use position buffer
-                  3 = VIACOMBO: VIA-specific workaround for capture
-                  4 = COMBO: use LPIB for playback, auto for capture stream
+    position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF)
     probe_mask  - Bitmask to probe codecs (default = -1, meaning all slots)
                   When the bit 8 (0x100) is set, the lower 8 bits are used
                   as the "fixed" codec slots; i.e. the driver probes the
@@ -881,7 +874,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
                 setup before initializing the codecs.  This option is
                 available only when CONFIG_SND_HDA_PATCH_LOADER=y is set.
                 See HD-Audio.txt for details.
-    beep_mode   - Selects the beep registration mode (0=off, 1=on); default
+    beep_mode   - Selects the beep registration mode (0=off, 1=on, 2=
+                dynamic registration via mute switch on/off); the default
                 value is set via CONFIG_SND_HDA_INPUT_BEEP_MODE kconfig.
     
     [Single (global) options]
@@ -892,12 +886,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
                 disable)
     power_save_controller - Reset HD-audio controller in power-saving mode
                 (default = on)
-    align_buffer_size - Force rounding of buffer/period sizes to multiples
-                      of 128 bytes. This is more efficient in terms of memory
-                      access but isn't required by the HDA spec and prevents
-                      users from specifying exact period/buffer sizes.
-                      (default = on)
-    snoop       - Enable/disable snooping (default = on)
 
     This module supports multiple cards and autoprobe.
     
@@ -931,11 +919,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
             (Usually SD_LPIB register is more accurate than the
             position buffer.)
 
-            position_fix=3 is specific to VIA devices.  The position
-            of the capture stream is checked from both LPIB and POSBUF
-            values.  position_fix=4 is a combination mode, using LPIB
-            for playback and POSBUF for capture.
-
     NB: If you get many "azx_get_response timeout" messages at
     loading, it's likely a problem of interrupts (e.g. ACPI irq
     routing).  Try to boot with options like "pci=noacpi".  Also, you
@@ -1550,7 +1533,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     Module for sound cards based on the C-Media CMI8786/8787/8788 chip:
     * Asound A-8788
-    * Asus Xonar DG/DGX
+    * Asus Xonar DG
     * AuzenTech X-Meridian
     * AuzenTech X-Meridian 2G
     * Bgears b-Enspirer
@@ -1599,7 +1582,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     Module supports autoprobe a chip.
 
-    Note: the driver may have problems regarding endianness.
+    Note: the driver may have problems regarding endianess.
 
     The power-management is supported.
 
@@ -1905,6 +1888,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     vid             - Vendor ID for the device (optional)
     pid             - Product ID for the device (optional)
     nrpacks         - Max. number of packets per URB (default: 8)
+    async_unlink    - Use async unlink mode (default: yes)
     device_setup    - Device specific magic number (optional)
                     - Influence depends on the device
                     - Default: 0x0000 
@@ -1916,6 +1900,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     NB: nrpacks parameter can be modified dynamically via sysfs.
         Don't put the value over 20.  Changing via sysfs has no sanity
         check.
+    NB: async_unlink=0 would cause Oops.  It remains just for
+        debugging purpose (if any).
     NB: ignore_ctl_error=1 may help when you get an error at accessing
         the mixer element such as URB error -22.  This happens on some
         buggy USB device or the controller.
@@ -2046,7 +2032,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     Install the necessary firmware files in alsa-firmware package.
     When no hotplug fw loader is available, you need to load the
     firmware via vxloader utility in alsa-tools package.  To invoke
-    vxloader automatically, add the following to /etc/modprobe.d/alsa.conf
+    vxloader automatically, add the following to /etc/modprobe.conf
 
         install snd-vx222 /sbin/modprobe --first-time -i snd-vx222 && /usr/bin/vxloader
 
@@ -2170,10 +2156,10 @@ corresponds to the card index of ALSA.  Usually, define this
 as the same card module.
 
 An example configuration for a single emu10k1 card is like below:
------ /etc/modprobe.d/alsa.conf
+----- /etc/modprobe.conf
 alias snd-card-0 snd-emu10k1
 alias sound-slot-0 snd-emu10k1
------ /etc/modprobe.d/alsa.conf
+----- /etc/modprobe.conf
 
 The available number of auto-loaded sound cards depends on the module
 option "cards_limit" of snd module.  As default it's set to 1.
@@ -2186,7 +2172,7 @@ cards is kept consistent.
 
 An example configuration for two sound cards is like below:
 
------ /etc/modprobe.d/alsa.conf
+----- /etc/modprobe.conf
 # ALSA portion
 options snd cards_limit=2
 alias snd-card-0 snd-interwave
@@ -2196,7 +2182,7 @@ options snd-ens1371 index=1
 # OSS/Free portion
 alias sound-slot-0 snd-interwave
 alias sound-slot-1 snd-ens1371
------ /etc/modprobe.d/alsa.conf
+----- /etc/modprobe.conf
 
 In this example, the interwave card is always loaded as the first card
 (index 0) and ens1371 as the second (index 1).
diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt
index 654dd3b694a..a4c53d8961e 100644
--- a/Documentation/sound/alsa/Audiophile-Usb.txt
+++ b/Documentation/sound/alsa/Audiophile-Usb.txt
@@ -232,7 +232,7 @@ The parameter can be given:
    # modprobe snd-usb-audio index=1 device_setup=0x09
 
  * Or while configuring the modules options in your modules configuration file
-   (tipically a .conf file in /etc/modprobe.d/ directory:
+   - For Fedora distributions, edit the /etc/modprobe.conf file:
        alias snd-card-1 snd-usb-audio
        options snd-usb-audio index=1 device_setup=0x09
 
@@ -253,7 +253,7 @@ CAUTION when initializing the device
    - first turn off the device
    - de-register the snd-usb-audio module (modprobe -r)
    - change the device_setup parameter by changing the device_setup
-     option in /etc/modprobe.d/*.conf
+     option in /etc/modprobe.conf 
    - turn on the device
  * A workaround for this last issue has been applied to kernel 2.6.23, but it may not
    be enough to ensure the 'stability' of the device initialization.
diff --git a/Documentation/sound/alsa/Channel-Mapping-API.txt b/Documentation/sound/alsa/Channel-Mapping-API.txt
deleted file mode 100644
index 3c43d1a4ca0..00000000000
--- a/Documentation/sound/alsa/Channel-Mapping-API.txt
+++ /dev/null
@@ -1,153 +0,0 @@
-ALSA PCM channel-mapping API
-============================
-                                        Takashi Iwai <tiwai@suse.de>
-
-GENERAL
--------
-
-The channel mapping API allows user to query the possible channel maps
-and the current channel map, also optionally to modify the channel map
-of the current stream.
-
-A channel map is an array of position for each PCM channel.
-Typically, a stereo PCM stream has a channel map of
-  { front_left, front_right }
-while a 4.0 surround PCM stream has a channel map of
-  { front left, front right, rear left, rear right }.
-
-The problem, so far, was that we had no standard channel map
-explicitly, and applications had no way to know which channel
-corresponds to which (speaker) position.  Thus, applications applied
-wrong channels for 5.1 outputs, and you hear suddenly strange sound
-from rear.  Or, some devices secretly assume that center/LFE is the
-third/fourth channels while others that C/LFE as 5th/6th channels.
-
-Also, some devices such as HDMI are configurable for different speaker
-positions even with the same number of total channels.  However, there
-was no way to specify this because of lack of channel map
-specification.  These are the main motivations for the new channel
-mapping API.
-
-
-DESIGN
-------
-
-Actually, "the channel mapping API" doesn't introduce anything new in
-the kernel/user-space ABI perspective.  It uses only the existing
-control element features.
-
-As a ground design, each PCM substream may contain a control element
-providing the channel mapping information and configuration.  This
-element is specified by:
-        iface = SNDRV_CTL_ELEM_IFACE_PCM
-        name = "Playback Channel Map" or "Capture Channel Map"
-        device = the same device number for the assigned PCM substream
-        index = the same index number for the assigned PCM substream
-
-Note the name is different depending on the PCM substream direction.
-
-Each control element provides at least the TLV read operation and the
-read operation.  Optionally, the write operation can be provided to
-allow user to change the channel map dynamically.
-
-* TLV
-
-The TLV operation gives the list of available channel
-maps.  A list item of a channel map is usually a TLV of
-        type data-bytes ch0 ch1 ch2...
-where type is the TLV type value, the second argument is the total
-bytes (not the numbers) of channel values, and the rest are the
-position value for each channel.
-
-As a TLV type, either SNDRV_CTL_TLVT_CHMAP_FIXED,
-SNDRV_CTL_TLV_CHMAP_VAR or SNDRV_CTL_TLVT_CHMAP_PAIRED can be used.
-The _FIXED type is for a channel map with the fixed channel position
-while the latter two are for flexible channel positions.  _VAR type is
-for a channel map where all channels are freely swappable and _PAIRED
-type is where pair-wise channels are swappable.  For example, when you
-have {FL/FR/RL/RR} channel map, _PAIRED type would allow you to swap
-only {RL/RR/FL/FR} while _VAR type would allow even swapping FL and
-RR.
-
-These new TLV types are defined in sound/tlv.h.
-
-The available channel position values are defined in sound/asound.h,
-here is a cut:
-
-/* channel positions */
-enum {
-        SNDRV_CHMAP_UNKNOWN = 0,
-        SNDRV_CHMAP_NA,         /* N/A, silent */
-        SNDRV_CHMAP_MONO,       /* mono stream */
-        /* this follows the alsa-lib mixer channel value + 3 */
-        SNDRV_CHMAP_FL,         /* front left */
-        SNDRV_CHMAP_FR,         /* front right */
-        SNDRV_CHMAP_RL,         /* rear left */
-        SNDRV_CHMAP_RR,         /* rear right */
-        SNDRV_CHMAP_FC,         /* front center */
-        SNDRV_CHMAP_LFE,        /* LFE */
-        SNDRV_CHMAP_SL,         /* side left */
-        SNDRV_CHMAP_SR,         /* side right */
-        SNDRV_CHMAP_RC,         /* rear center */
-        /* new definitions */
-        SNDRV_CHMAP_FLC,        /* front left center */
-        SNDRV_CHMAP_FRC,        /* front right center */
-        SNDRV_CHMAP_RLC,        /* rear left center */
-        SNDRV_CHMAP_RRC,        /* rear right center */
-        SNDRV_CHMAP_FLW,        /* front left wide */
-        SNDRV_CHMAP_FRW,        /* front right wide */
-        SNDRV_CHMAP_FLH,        /* front left high */
-        SNDRV_CHMAP_FCH,        /* front center high */
-        SNDRV_CHMAP_FRH,        /* front right high */
-        SNDRV_CHMAP_TC,         /* top center */
-        SNDRV_CHMAP_TFL,        /* top front left */
-        SNDRV_CHMAP_TFR,        /* top front right */
-        SNDRV_CHMAP_TFC,        /* top front center */
-        SNDRV_CHMAP_TRL,        /* top rear left */
-        SNDRV_CHMAP_TRR,        /* top rear right */
-        SNDRV_CHMAP_TRC,        /* top rear center */
-        SNDRV_CHMAP_LAST = SNDRV_CHMAP_TRC,
-};
-
-When a PCM stream can provide more than one channel map, you can
-provide multiple channel maps in a TLV container type.  The TLV data
-to be returned will contain such as:
-        SNDRV_CTL_TLVT_CONTAINER 96
-            SNDRV_CTL_TLVT_CHMAP_FIXED 4 SNDRV_CHMAP_FC
-            SNDRV_CTL_TLVT_CHMAP_FIXED 8 SNDRV_CHMAP_FL SNDRV_CHMAP_FR
-            SNDRV_CTL_TLVT_CHMAP_FIXED 16 NDRV_CHMAP_FL SNDRV_CHMAP_FR \
-                SNDRV_CHMAP_RL SNDRV_CHMAP_RR
-
-The channel position is provided in LSB 16bits.  The upper bits are
-used for bit flags.
-
-#define SNDRV_CHMAP_POSITION_MASK       0xffff
-#define SNDRV_CHMAP_PHASE_INVERSE       (0x01 << 16)
-#define SNDRV_CHMAP_DRIVER_SPEC         (0x02 << 16)
-
-SNDRV_CHMAP_PHASE_INVERSE indicates the channel is phase inverted,
-(thus summing left and right channels would result in almost silence).
-Some digital mic devices have this.
-
-When SNDRV_CHMAP_DRIVER_SPEC is set, all the channel position values
-don't follow the standard definition above but driver-specific.
-
-* READ OPERATION
-
-The control read operation is for providing the current channel map of
-the given stream.  The control element returns an integer array
-containing the position of each channel.
-
-When this is performed before the number of the channel is specified
-(i.e. hw_params is set), it should return all channels set to
-UNKNOWN.
-
-* WRITE OPERATION
-
-The control write operation is optional, and only for devices that can
-change the channel configuration on the fly, such as HDMI.  User needs
-to pass an integer value containing the valid channel positions for
-all channels of the assigned PCM substream.
-
-This operation is allowed only at PCM PREPARED state.  When called in
-other states, it shall return an error.
diff --git a/Documentation/sound/alsa/HD-Audio-Controls.txt b/Documentation/sound/alsa/HD-Audio-Controls.txt
index e9621e349e1..1482035243e 100644
--- a/Documentation/sound/alsa/HD-Audio-Controls.txt
+++ b/Documentation/sound/alsa/HD-Audio-Controls.txt
@@ -98,19 +98,3 @@ Conexant codecs
 
 * Auto-Mute Mode
   See Reatek codecs.
-
-
-Analog codecs
---------------
-
-* Channel Mode
-  This is an enum control to change the surround-channel setup,
-  appears only when the surround channels are available.
-  It gives the number of channels to be used, "2ch", "4ch" and "6ch".
-  According to the configuration, this also controls the
-  jack-retasking of multi-I/O jacks.
-
-* Independent HP
-  When this enum control is enabled, the headphone output is routed
-  from an individual stream (the third PCM such as hw:0,2) instead of
-  the primary stream.
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index 16dfe57f173..f8961402a85 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -8,31 +8,101 @@ ALC880
   5stack-digout 5-jack in back, 2-jack in front, a SPDIF out
   6stack        6-jack in back, 2-jack in front
   6stack-digout 6-jack with a SPDIF out
+  w810          3-jack
+  z71v          3-jack (HP shared SPDIF)
+  asus          3-jack (ASUS Mobo)
+  asus-w1v      ASUS W1V
+  asus-dig      ASUS with SPDIF out
+  asus-dig2     ASUS with SPDIF out (using GPIO2)
+  uniwill       3-jack
+  fujitsu       Fujitsu Laptops (Pi1536)
+  F1734         2-jack
+  lg            LG laptop (m1 express dual)
+  lg-lw         LG LW20/LW25 laptop
+  tcl           TCL S700
+  clevo         Clevo laptops (m520G, m665n)
+  medion        Medion Rim 2150
+  test          for testing/debugging purpose, almost all controls can be
+                adjusted.  Appearing only when compiled with
+                $CONFIG_SND_DEBUG=y
+  auto          auto-config reading BIOS (default)
 
 ALC260
 ======
-  N/A
+  hp            HP machines
+  hp-3013       HP machines (3013-variant)
+  hp-dc7600     HP DC7600
+  fujitsu       Fujitsu S7020
+  acer          Acer TravelMate
+  will          Will laptops (PB V7900)
+  replacer      Replacer 672V
+  favorit100    Maxdata Favorit 100XS
+  basic         fixed pin assignment (old default model)
+  test          for testing/debugging purpose, almost all controls can
+                adjusted.  Appearing only when compiled with
+                $CONFIG_SND_DEBUG=y
+  auto          auto-config reading BIOS (default)
 
 ALC262
 ======
-  inv-dmic      Inverted internal mic workaround
+  fujitsu       Fujitsu Laptop
+  hp-bpc        HP xw4400/6400/8400/9400 laptops
+  hp-bpc-d7000  HP BPC D7000
+  hp-tc-t5735   HP Thin Client T5735
+  hp-rp5700     HP RP5700
+  benq          Benq ED8
+  benq-t31      Benq T31
+  hippo         Hippo (ATI) with jack detection, Sony UX-90s
+  hippo_1       Hippo (Benq) with jack detection
+  sony-assamd   Sony ASSAMD
+  toshiba-s06   Toshiba S06
+  toshiba-rx1   Toshiba RX1
+  tyan          Tyan Thunder n6650W (S2915-E)
+  ultra         Samsung Q1 Ultra Vista model
+  lenovo-3000   Lenovo 3000 y410
+  nec           NEC Versa S9100
+  basic         fixed pin assignment w/o SPDIF
+  auto          auto-config reading BIOS (default)
 
 ALC267/268
 ==========
-  inv-dmic      Inverted internal mic workaround
+  quanta-il1    Quanta IL1 mini-notebook
+  3stack        3-stack model
+  toshiba       Toshiba A205
+  acer          Acer laptops
+  acer-dmic     Acer laptops with digital-mic
+  acer-aspire   Acer Aspire One
+  dell          Dell OEM laptops (Vostro 1200)
+  zepto         Zepto laptops
+  test          for testing/debugging purpose, almost all controls can
+                adjusted.  Appearing only when compiled with
+                $CONFIG_SND_DEBUG=y
+  auto          auto-config reading BIOS (default)
 
-ALC269/270/275/276/280/282
+ALC269
 ======
+  basic         Basic preset
+  quanta        Quanta FL1
   laptop-amic   Laptops with analog-mic input
   laptop-dmic   Laptops with digital-mic input
-  alc269-dmic   Enable ALC269(VA) digital mic workaround
-  alc271-dmic   Enable ALC271X digital mic workaround
-  inv-dmic      Inverted internal mic workaround
-  lenovo-dock   Enables docking station I/O for some Lenovos
+  fujitsu       FSC Amilo
+  lifebook      Fujitsu Lifebook S6420
+  auto          auto-config reading BIOS (default)
 
 ALC662/663/272
 ==============
-  mario         Chromebook mario model fixup
+  3stack-dig    3-stack (2-channel) with SPDIF
+  3stack-6ch     3-stack (6-channel)
+  3stack-6ch-dig 3-stack (6-channel) with SPDIF
+  5stack-dig     5-stack with SPDIF
+  lenovo-101e    Lenovo laptop
+  eeepc-p701    ASUS Eeepc P701
+  eeepc-ep20    ASUS Eeepc EP20
+  ecs           ECS/Foxconn mobo
+  m51va         ASUS M51VA
+  g71v          ASUS G71V
+  h13           ASUS H13
+  g50v          ASUS G50V
   asus-mode1    ASUS
   asus-mode2    ASUS
   asus-mode3    ASUS
@@ -41,27 +111,92 @@ ALC662/663/272
   asus-mode6    ASUS
   asus-mode7    ASUS
   asus-mode8    ASUS
-  inv-dmic      Inverted internal mic workaround
+  dell          Dell with ALC272
+  dell-zm1      Dell ZM1 with ALC272
+  samsung-nc10  Samsung NC10 mini notebook
+  auto          auto-config reading BIOS (default)
 
 ALC680
 ======
-  N/A
+  base          Base model (ASUS NX90)
+  auto          auto-config reading BIOS (default)
 
 ALC882/883/885/888/889
 ======================
-  acer-aspire-4930g     Acer Aspire 4930G/5930G/6530G/6930G/7730G
-  acer-aspire-8930g     Acer Aspire 8330G/6935G
-  acer-aspire           Acer Aspire others
-  inv-dmic      Inverted internal mic workaround
-  no-primary-hp         VAIO Z workaround (for fixed speaker DAC)
+  3stack-dig    3-jack with SPDIF I/O
+  6stack-dig    6-jack digital with SPDIF I/O
+  arima         Arima W820Di1
+  targa         Targa T8, MSI-1049 T8
+  asus-a7j      ASUS A7J
+  asus-a7m      ASUS A7M
+  macpro        MacPro support
+  mb5           Macbook 5,1
+  macmini3      Macmini 3,1
+  mba21         Macbook Air 2,1
+  mbp3          Macbook Pro rev3
+  imac24        iMac 24'' with jack detection
+  imac91        iMac 9,1
+  w2jc          ASUS W2JC
+  3stack-2ch-dig        3-jack with SPDIF I/O (ALC883)
+  alc883-6stack-dig     6-jack digital with SPDIF I/O (ALC883)
+  3stack-6ch    3-jack 6-channel
+  3stack-6ch-dig 3-jack 6-channel with SPDIF I/O
+  6stack-dig-demo  6-jack digital for Intel demo board
+  acer          Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc)
+  acer-aspire   Acer Aspire 9810
+  acer-aspire-4930g Acer Aspire 4930G
+  acer-aspire-6530g Acer Aspire 6530G
+  acer-aspire-7730g Acer Aspire 7730G
+  acer-aspire-8930g Acer Aspire 8930G
+  medion        Medion Laptops
+  targa-dig     Targa/MSI
+  targa-2ch-dig Targa/MSI with 2-channel
+  targa-8ch-dig Targa/MSI with 8-channel (MSI GX620)
+  laptop-eapd   3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE)
+  lenovo-101e   Lenovo 101E
+  lenovo-nb0763 Lenovo NB0763
+  lenovo-ms7195-dig Lenovo MS7195
+  lenovo-sky    Lenovo Sky
+  haier-w66     Haier W66
+  3stack-hp     HP machines with 3stack (Lucknow, Samba boards)
+  6stack-dell   Dell machines with 6stack (Inspiron 530)
+  mitac         Mitac 8252D
+  clevo-m540r   Clevo M540R (6ch + digital)
+  clevo-m720    Clevo M720 laptop series
+  fujitsu-pi2515 Fujitsu AMILO Pi2515
+  fujitsu-xa3530 Fujitsu AMILO XA3530
+  3stack-6ch-intel Intel DG33* boards
+  intel-alc889a Intel IbexPeak with ALC889A
+  intel-x58     Intel DX58 with ALC889
+  asus-p5q      ASUS P5Q-EM boards
+  mb31          MacBook 3,1
+  sony-vaio-tt  Sony VAIO TT
+  auto          auto-config reading BIOS (default)
 
 ALC861/660
 ==========
-  N/A
+  3stack        3-jack
+  3stack-dig    3-jack with SPDIF I/O
+  6stack-dig    6-jack with SPDIF I/O
+  3stack-660    3-jack (for ALC660)
+  uniwill-m31   Uniwill M31 laptop
+  toshiba       Toshiba laptop support
+  asus          Asus laptop support
+  asus-laptop   ASUS F2/F3 laptops
+  auto          auto-config reading BIOS (default)
 
 ALC861VD/660VD
 ==============
-  N/A
+  3stack        3-jack
+  3stack-dig    3-jack with SPDIF OUT
+  6stack-dig    6-jack with SPDIF OUT
+  3stack-660    3-jack (for ALC660VD)
+  3stack-660-digout 3-jack with SPDIF OUT (for ALC660VD)
+  lenovo        Lenovo 3000 C200
+  dallas        Dallas laptops
+  hp            HP TX1000
+  asus-v1s      ASUS V1Sn
+  auto          auto-config reading BIOS (default)
 
 CMI9880
 =======
@@ -74,8 +209,7 @@ CMI9880
 
 AD1882 / AD1882A
 ================
-  3stack        3-stack mode
-  3stack-automute 3-stack with automute front HP (default)
+  3stack        3-stack mode (default)
   6stack        6-stack mode
 
 AD1884A / AD1883 / AD1984A / AD1984B
@@ -155,6 +289,7 @@ Conexant 5051
   hp-dv6736     HP dv6736
   hp-f700       HP Compaq Presario F700
   ideapad       Lenovo IdeaPad laptop
+  lenovo-x200   Lenovo X200 laptop
   toshiba       Toshiba Satellite M300
 
 Conexant 5066
@@ -274,11 +409,8 @@ STAC92HD83*
   mic-ref       Reference board with power management for ports
   dell-s14      Dell laptop
   dell-vostro-3500      Dell Vostro 3500 laptop
+  hp            HP laptops with (inverted) mute-LED
   hp-dv7-4000   HP dv-7 4000
-  hp_cNB11_intquad HP CNB models with 4 speakers
-  hp-zephyr     HP Zephyr
-  hp-led        HP with broken BIOS for mute LED
-  hp-inv-led    HP with broken BIOS for inverted mute LED
   auto          BIOS setup (default)
 
 STAC9872
diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt
index 7813c06a5c7..c82beb00763 100644
--- a/Documentation/sound/alsa/HD-Audio.txt
+++ b/Documentation/sound/alsa/HD-Audio.txt
@@ -59,12 +59,7 @@ a case, you can change the default method via `position_fix` option.
 `position_fix=1` means to use LPIB method explicitly.
 `position_fix=2` means to use the position-buffer.
 `position_fix=3` means to use a combination of both methods, needed
-for some VIA controllers.  The capture stream position is corrected
-by comparing both LPIB and position-buffer values.
-`position_fix=4` is another combination available for all controllers,
-and uses LPIB for the playback and the position-buffer for the capture
-streams.
-0 is the default value for all other
+for some VIA and ATI controllers.  0 is the default value for all other
 controllers, the automatic check and fallback to LPIB as described in
 the above.  If you get a problem of repeated sounds, this option might
 help.
@@ -452,10 +447,7 @@ The file needs to have a line `[codec]`.  The next line should contain
 three numbers indicating the codec vendor-id (0x12345678 in the
 example), the codec subsystem-id (0xabcd1234) and the address (2) of
 the codec.  The rest patch entries are applied to this specified codec
-until another codec entry is given.  Passing 0 or a negative number to
-the first or the second value will make the check of the corresponding
-field be skipped.  It'll be useful for really broken devices that don't
-initialize SSID properly.
+until another codec entry is given.
 
 The `[model]` line allows to change the model name of the each codec.
 In the example above, it will be changed to model=auto.
@@ -499,7 +491,7 @@ Also, the codec chip name can be rewritten via `[chip_name]` line.
 The hd-audio driver reads the file via request_firmware().  Thus,
 a patch file has to be located on the appropriate firmware path,
 typically, /lib/firmware.  For example, when you pass the option
-`patch=hda-init.fw`, the file /lib/firmware/hda-init.fw must be
+`patch=hda-init.fw`, the file /lib/firmware/hda-init-fw must be
 present.
 
 The patch module option is specific to each card instance, and you
@@ -532,59 +524,11 @@ power-saving.  See /sys/module/snd_hda_intel/parameters/power_save to
 check the current value.  If it's non-zero, the feature is turned on.
 
 
-Tracepoints
-~~~~~~~~~~~
-The hd-audio driver gives a few basic tracepoints.
-`hda:hda_send_cmd` traces each CORB write while `hda:hda_get_response`
-traces the response from RIRB (only when read from the codec driver).
-`hda:hda_bus_reset` traces the bus-reset due to fatal error, etc,
-`hda:hda_unsol_event` traces the unsolicited events, and
-`hda:hda_power_down` and `hda:hda_power_up` trace the power down/up
-via power-saving behavior.
-
-Enabling all tracepoints can be done like
-------------------------------------------------------------------------
-  # echo 1 > /sys/kernel/debug/tracing/events/hda/enable
-------------------------------------------------------------------------
-then after some commands, you can traces from
-/sys/kernel/debug/tracing/trace file.  For example, when you want to
-trace what codec command is sent, enable the tracepoint like:
-------------------------------------------------------------------------
-  # cat /sys/kernel/debug/tracing/trace
-  # tracer: nop
-  #
-  #       TASK-PID    CPU#    TIMESTAMP  FUNCTION
-  #          | |       |          |         |
-         <...>-7807  [002] 105147.774889: hda_send_cmd: [0:0] val=e3a019
-         <...>-7807  [002] 105147.774893: hda_send_cmd: [0:0] val=e39019
-         <...>-7807  [002] 105147.999542: hda_send_cmd: [0:0] val=e3a01a
-         <...>-7807  [002] 105147.999543: hda_send_cmd: [0:0] val=e3901a
-         <...>-26764 [001] 349222.837143: hda_send_cmd: [0:0] val=e3a019
-         <...>-26764 [001] 349222.837148: hda_send_cmd: [0:0] val=e39019
-         <...>-26764 [001] 349223.058539: hda_send_cmd: [0:0] val=e3a01a
-         <...>-26764 [001] 349223.058541: hda_send_cmd: [0:0] val=e3901a
-------------------------------------------------------------------------
-Here `[0:0]` indicates the card number and the codec address, and
-`val` shows the value sent to the codec, respectively.  The value is
-a packed value, and you can decode it via hda-decode-verb program
-included in hda-emu package below.  For example, the value e3a019 is
-to set the left output-amp value to 25.
-------------------------------------------------------------------------
-  % hda-decode-verb 0xe3a019
-  raw value = 0x00e3a019
-  cid = 0, nid = 0x0e, verb = 0x3a0, parm = 0x19
-  raw value: verb = 0x3a0, parm = 0x19
-  verbname = set_amp_gain_mute
-  amp raw val = 0xa019
-  output, left, idx=0, mute=0, val=25
-------------------------------------------------------------------------
-
-
 Development Tree
 ~~~~~~~~~~~~~~~~
 The latest development codes for HD-audio are found on sound git tree:
 
-- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git
+- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git
 
 The master branch or for-next branches can be used as the main
 development branches in general while the HD-audio specific patches
@@ -599,7 +543,7 @@ is, installed via the usual spells: configure, make and make
 install(-modules).  See INSTALL in the package.  The snapshot tarballs
 are found at:
 
-- ftp://ftp.suse.com/pub/people/tiwai/snapshot/
+- ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/
 
 
 Sending a Bug Report
@@ -701,7 +645,7 @@ via hda-verb won't change the mixer value.
 
 The hda-verb program is found in the ftp directory:
 
-- ftp://ftp.suse.com/pub/people/tiwai/misc/
+- ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/misc/
 
 Also a git repository is available:
 
@@ -769,7 +713,7 @@ operation, the jack plugging simulation, etc.
 
 The package is found in:
 
-- ftp://ftp.suse.com/pub/people/tiwai/misc/
+- ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/misc/
 
 A git repository is available:
 
diff --git a/Documentation/sound/alsa/MIXART.txt b/Documentation/sound/alsa/MIXART.txt
index 4ee35b4fbe4..ef42c44fa1f 100644
--- a/Documentation/sound/alsa/MIXART.txt
+++ b/Documentation/sound/alsa/MIXART.txt
@@ -76,9 +76,9 @@ FIRMWARE
  when CONFIG_FW_LOADER is set.  The mixartloader is necessary only
  for older versions or when you build the driver into kernel.]
  
-For loading the firmware automatically after the module is loaded, use a
-install command.  For example, add the following entry to
-/etc/modprobe.d/mixart.conf for miXart driver:
+For loading the firmware automatically after the module is loaded, use
+the post-install command.  For example, add the following entry to
+/etc/modprobe.conf for miXart driver:
 
         install snd-mixart /sbin/modprobe --first-time -i snd-mixart && \
                            /usr/bin/mixartloader
diff --git a/Documentation/sound/alsa/OSS-Emulation.txt b/Documentation/sound/alsa/OSS-Emulation.txt
index 152ca2a3f1b..022aaeb0e9d 100644
--- a/Documentation/sound/alsa/OSS-Emulation.txt
+++ b/Documentation/sound/alsa/OSS-Emulation.txt
@@ -19,7 +19,7 @@ the card number and the minor unit number.  Usually you don't have to
 define these aliases by yourself.
 
 Only necessary step for auto-loading of OSS modules is to define the
-card alias in /etc/modprobe.d/alsa.conf, such as
+card alias in /etc/modprobe.conf, such as
 
         alias sound-slot-0 snd-emu10k1
 
diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt
deleted file mode 100644
index 90e9b3a11ab..00000000000
--- a/Documentation/sound/alsa/compress_offload.txt
+++ /dev/null
@@ -1,188 +0,0 @@
-                compress_offload.txt
-                =====================
-        Pierre-Louis.Bossart <pierre-louis.bossart@linux.intel.com>
-                Vinod Koul <vinod.koul@linux.intel.com>
-
-Overview
-
-Since its early days, the ALSA API was defined with PCM support or
-constant bitrates payloads such as IEC61937 in mind. Arguments and
-returned values in frames are the norm, making it a challenge to
-extend the existing API to compressed data streams.
-
-In recent years, audio digital signal processors (DSP) were integrated
-in system-on-chip designs, and DSPs are also integrated in audio
-codecs. Processing compressed data on such DSPs results in a dramatic
-reduction of power consumption compared to host-based
-processing. Support for such hardware has not been very good in Linux,
-mostly because of a lack of a generic API available in the mainline
-kernel.
-
-Rather than requiring a compatibility break with an API change of the
-ALSA PCM interface, a new 'Compressed Data' API is introduced to
-provide a control and data-streaming interface for audio DSPs.
-
-The design of this API was inspired by the 2-year experience with the
-Intel Moorestown SOC, with many corrections required to upstream the
-API in the mainline kernel instead of the staging tree and make it
-usable by others.
-
-Requirements
-
-The main requirements are:
-
-- separation between byte counts and time. Compressed formats may have
-  a header per file, per frame, or no header at all. The payload size
-  may vary from frame-to-frame. As a result, it is not possible to
-  estimate reliably the duration of audio buffers when handling
-  compressed data. Dedicated mechanisms are required to allow for
-  reliable audio-video synchronization, which requires precise
-  reporting of the number of samples rendered at any given time.
-
-- Handling of multiple formats. PCM data only requires a specification
-  of the sampling rate, number of channels and bits per sample. In
-  contrast, compressed data comes in a variety of formats. Audio DSPs
-  may also provide support for a limited number of audio encoders and
-  decoders embedded in firmware, or may support more choices through
-  dynamic download of libraries.
-
-- Focus on main formats. This API provides support for the most
-  popular formats used for audio and video capture and playback. It is
-  likely that as audio compression technology advances, new formats
-  will be added.
-
-- Handling of multiple configurations. Even for a given format like
-  AAC, some implementations may support AAC multichannel but HE-AAC
-  stereo. Likewise WMA10 level M3 may require too much memory and cpu
-  cycles. The new API needs to provide a generic way of listing these
-  formats.
-
-- Rendering/Grabbing only. This API does not provide any means of
-  hardware acceleration, where PCM samples are provided back to
-  user-space for additional processing. This API focuses instead on
-  streaming compressed data to a DSP, with the assumption that the
-  decoded samples are routed to a physical output or logical back-end.
-
- - Complexity hiding. Existing user-space multimedia frameworks all
-  have existing enums/structures for each compressed format. This new
-  API assumes the existence of a platform-specific compatibility layer
-  to expose, translate and make use of the capabilities of the audio
-  DSP, eg. Android HAL or PulseAudio sinks. By construction, regular
-  applications are not supposed to make use of this API.
-
-
-Design
-
-The new API shares a number of concepts with with the PCM API for flow
-control. Start, pause, resume, drain and stop commands have the same
-semantics no matter what the content is.
-
-The concept of memory ring buffer divided in a set of fragments is
-borrowed from the ALSA PCM API. However, only sizes in bytes can be
-specified.
-
-Seeks/trick modes are assumed to be handled by the host.
-
-The notion of rewinds/forwards is not supported. Data committed to the
-ring buffer cannot be invalidated, except when dropping all buffers.
-
-The Compressed Data API does not make any assumptions on how the data
-is transmitted to the audio DSP. DMA transfers from main memory to an
-embedded audio cluster or to a SPI interface for external DSPs are
-possible. As in the ALSA PCM case, a core set of routines is exposed;
-each driver implementer will have to write support for a set of
-mandatory routines and possibly make use of optional ones.
-
-The main additions are
-
-- get_caps
-This routine returns the list of audio formats supported. Querying the
-codecs on a capture stream will return encoders, decoders will be
-listed for playback streams.
-
-- get_codec_caps For each codec, this routine returns a list of
-capabilities. The intent is to make sure all the capabilities
-correspond to valid settings, and to minimize the risks of
-configuration failures. For example, for a complex codec such as AAC,
-the number of channels supported may depend on a specific profile. If
-the capabilities were exposed with a single descriptor, it may happen
-that a specific combination of profiles/channels/formats may not be
-supported. Likewise, embedded DSPs have limited memory and cpu cycles,
-it is likely that some implementations make the list of capabilities
-dynamic and dependent on existing workloads. In addition to codec
-settings, this routine returns the minimum buffer size handled by the
-implementation. This information can be a function of the DMA buffer
-sizes, the number of bytes required to synchronize, etc, and can be
-used by userspace to define how much needs to be written in the ring
-buffer before playback can start.
-
-- set_params
-This routine sets the configuration chosen for a specific codec. The
-most important field in the parameters is the codec type; in most
-cases decoders will ignore other fields, while encoders will strictly
-comply to the settings
-
-- get_params
-This routines returns the actual settings used by the DSP. Changes to
-the settings should remain the exception.
-
-- get_timestamp
-The timestamp becomes a multiple field structure. It lists the number
-of bytes transferred, the number of samples processed and the number
-of samples rendered/grabbed. All these values can be used to determine
-the avarage bitrate, figure out if the ring buffer needs to be
-refilled or the delay due to decoding/encoding/io on the DSP.
-
-Note that the list of codecs/profiles/modes was derived from the
-OpenMAX AL specification instead of reinventing the wheel.
-Modifications include:
-- Addition of FLAC and IEC formats
-- Merge of encoder/decoder capabilities
-- Profiles/modes listed as bitmasks to make descriptors more compact
-- Addition of set_params for decoders (missing in OpenMAX AL)
-- Addition of AMR/AMR-WB encoding modes (missing in OpenMAX AL)
-- Addition of format information for WMA
-- Addition of encoding options when required (derived from OpenMAX IL)
-- Addition of rateControlSupported (missing in OpenMAX AL)
-
-Not supported:
-
-- Support for VoIP/circuit-switched calls is not the target of this
-  API. Support for dynamic bit-rate changes would require a tight
-  coupling between the DSP and the host stack, limiting power savings.
-
-- Packet-loss concealment is not supported. This would require an
-  additional interface to let the decoder synthesize data when frames
-  are lost during transmission. This may be added in the future.
-
-- Volume control/routing is not handled by this API. Devices exposing a
-  compressed data interface will be considered as regular ALSA devices;
-  volume changes and routing information will be provided with regular
-  ALSA kcontrols.
-
-- Embedded audio effects. Such effects should be enabled in the same
-  manner, no matter if the input was PCM or compressed.
-
-- multichannel IEC encoding. Unclear if this is required.
-
-- Encoding/decoding acceleration is not supported as mentioned
-  above. It is possible to route the output of a decoder to a capture
-  stream, or even implement transcoding capabilities. This routing
-  would be enabled with ALSA kcontrols.
-
-- Audio policy/resource management. This API does not provide any
-  hooks to query the utilization of the audio DSP, nor any premption
-  mechanisms.
-
-- No notion of underun/overrun. Since the bytes written are compressed
-  in nature and data written/read doesn't translate directly to
-  rendered output in time, this does not deal with underrun/overun and
-  maybe dealt in user-library
-
-Credits:
-- Mark Brown and Liam Girdwood for discussions on the need for this API
-- Harsha Priya for her work on intel_sst compressed API
-- Rakesh Ughreja for valuable feedback
-- Sing Nallasellan, Sikkandar Madar and Prasanna Samaga for
-  demonstrating and quantifying the benefits of audio offload on a
-  real platform.
diff --git a/Documentation/sound/alsa/hdspm.txt b/Documentation/sound/alsa/hdspm.txt
index 7ba31948dea..7a67ff71a9f 100644
--- a/Documentation/sound/alsa/hdspm.txt
+++ b/Documentation/sound/alsa/hdspm.txt
@@ -359,4 +359,4 @@ Calling Parameter:
    enable_monitor int array (min = 1, max = 8), 
      "Enable Analog Out on Channel 63/64 by default."
 
-      note: here the analog output is enabled (but not routed).
+      note: here the analog output is enabled (but not routed).
\ No newline at end of file
diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt
index d50c14df341..3e2ec9cbf39 100644
--- a/Documentation/sound/alsa/soc/machine.txt
+++ b/Documentation/sound/alsa/soc/machine.txt
@@ -50,7 +50,8 @@ Machine DAI Configuration
 The machine DAI configuration glues all the codec and CPU DAIs together. It can
 also be used to set up the DAI system clock and for any machine related DAI
 initialisation e.g. the machine audio map can be connected to the codec audio
-map, unconnected codec pins can be set as such.
+map, unconnected codec pins can be set as such. Please see corgi.c, spitz.c
+for examples.
 
 struct snd_soc_dai_link is used to set up each DAI in your machine. e.g.
 
@@ -82,7 +83,8 @@ 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.
+sockets in the machine init function. See soc/pxa/spitz.c and dapm.txt for
+details.
 
 
 Machine Controls