Merge branch 'next' into for-linus

7 files changed, 244 insertions, 23 deletions

diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 012858d2b11..4252697a95d 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -460,6 +460,25 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     The power-management is supported.
 
+  Module snd-ctxfi
+  ----------------
+
+    Module for Creative Sound Blaster X-Fi boards (20k1 / 20k2 chips)
+	* Creative Sound Blaster X-Fi Titanium Fatal1ty Champion Series
+	* Creative Sound Blaster X-Fi Titanium Fatal1ty Professional Series
+	* Creative Sound Blaster X-Fi Titanium Professional Audio
+	* Creative Sound Blaster X-Fi Titanium
+	* Creative Sound Blaster X-Fi Elite Pro
+	* Creative Sound Blaster X-Fi Platinum
+	* Creative Sound Blaster X-Fi Fatal1ty
+	* Creative Sound Blaster X-Fi XtremeGamer
+	* Creative Sound Blaster X-Fi XtremeMusic
+
+    reference_rate	- reference sample rate, 44100 or 48000 (default)
+    multiple		- multiple to ref. sample rate, 1 or 2 (default)
+
+    This module supports multiple cards.
+
   Module snd-darla20
   ------------------
 
@@ -754,7 +773,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     single_cmd  - Use single immediate commands to communicate with
 		codecs (for debugging only)
     enable_msi	- Enable Message Signaled Interrupt (MSI) (default = off)
-    power_save	- Automatic power-saving timtout (in second, 0 =
+    power_save	- Automatic power-saving timeout (in second, 0 =
 		disable)
     power_save_controller - Reset HD-audio controller in power-saving mode
 		(default = on)
@@ -925,6 +944,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 			* Onkyo SE-90PCI
 			* Onkyo SE-200PCI
 			* ESI Juli@
+			* ESI Maya44
 			* Hercules Fortissimo IV
 			* EGO-SYS WaveTerminal 192M
 
@@ -933,7 +953,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 		  prodigy71xt, prodigy71hifi, prodigyhd2, prodigy192,
 		  juli, aureon51, aureon71, universe, ap192, k8x800,
 		  phase22, phase28, ms300, av710, se200pci, se90pci,
-		  fortissimo4, sn25p, WT192M
+		  fortissimo4, sn25p, WT192M, maya44
 
     This module supports multiple cards and autoprobe.
 
@@ -1093,6 +1113,13 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     This module supports multiple cards.
     The driver requires the firmware loader support on kernel.
 
+  Module snd-lx6464es
+  -------------------
+
+    Module for Digigram LX6464ES boards
+
+    This module supports multiple cards.
+
   Module snd-maestro3
   -------------------
 
@@ -1543,13 +1570,15 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
   Module snd-sc6000
   -----------------
 
-    Module for Gallant SC-6000 soundcard.
+    Module for Gallant SC-6000 soundcard and later models: SC-6600
+    and SC-7000.
 
     port	- Port # (0x220 or 0x240)
     mss_port	- MSS Port # (0x530 or 0xe80)
     irq		- IRQ # (5,7,9,10,11)
     mpu_irq	- MPU-401 IRQ # (5,7,9,10) ,0 - no MPU-401 irq
     dma		- DMA # (1,3,0)
+    joystick	- Enable gameport - 0 = disable (default), 1 = enable
 
     This module supports multiple cards.
 
@@ -1859,7 +1888,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
   -------------------
 
     Module for sound cards based on the Asus AV100/AV200 chips,
-    i.e., Xonar D1, DX, D2, D2X, HDAV1.3 (Deluxe), and Essence STX.
+    i.e., Xonar D1, DX, D2, D2X, HDAV1.3 (Deluxe), Essence ST
+    (Deluxe) and Essence STX.
 
     This module supports autoprobe and multiple cards.
 
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index 322869fc8a9..939a3dd5814 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -36,6 +36,7 @@ ALC260
   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
@@ -85,10 +86,11 @@ ALC269
   eeepc-p703	ASUS Eeepc P703 P900A
   eeepc-p901	ASUS Eeepc P901 S101
   fujitsu	FSC Amilo
+  lifebook	Fujitsu Lifebook S6420
   auto		auto-config reading BIOS (default)
 
-ALC662/663
-==========
+ALC662/663/272
+==============
   3stack-dig	3-stack (2-channel) with SPDIF
   3stack-6ch	 3-stack (6-channel)
   3stack-6ch-dig 3-stack (6-channel) with SPDIF
@@ -107,6 +109,9 @@ ALC662/663
   asus-mode4	ASUS
   asus-mode5	ASUS
   asus-mode6	ASUS
+  dell		Dell with ALC272
+  dell-zm1	Dell ZM1 with ALC272
+  samsung-nc10	Samsung NC10 mini notebook
   auto		auto-config reading BIOS (default)
 
 ALC882/885
@@ -118,6 +123,7 @@ ALC882/885
   asus-a7j	ASUS A7J
   asus-a7m	ASUS A7M
   macpro	MacPro support
+  mb5		Macbook 5,1
   mbp3		Macbook Pro rev3
   imac24	iMac 24'' with jack detection
   w2jc		ASUS W2JC
@@ -133,10 +139,13 @@ ALC883/888
   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-8930g Acer Aspire 8930G
   medion	Medion Laptops
   medion-md2	Medion MD2
   targa-dig	Targa/MSI
-  targa-2ch-dig	Targs/MSI with 2-channel
+  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
@@ -150,6 +159,9 @@ ALC883/888
   fujitsu-pi2515 Fujitsu AMILO Pi2515
   fujitsu-xa3530 Fujitsu AMILO XA3530
   3stack-6ch-intel Intel DG33* boards
+  asus-p5q	ASUS P5Q-EM boards
+  mb31		MacBook 3,1
+  sony-vaio-tt  Sony VAIO TT
   auto		auto-config reading BIOS (default)
 
 ALC861/660
@@ -228,6 +240,7 @@ AD1986A
   laptop-automute 2-channel with EAPD and HP-automute (Lenovo N100)
   ultra		2-channel with EAPD (Samsung Ultra tablet PC)
   samsung	2-channel with EAPD (Samsung R65)
+  samsung-p50	2-channel with HP-automute (Samsung P50)
 
 AD1988/AD1988B/AD1989A/AD1989B
 ==============================
@@ -348,6 +361,7 @@ STAC92HD71B*
   hp-m4		HP mini 1000
   hp-dv5	HP dv series
   hp-hdx	HP HDX series
+  hp-dv4-1222nr	HP dv4-1222nr (with LED support)
   auto		BIOS setup (default)
 
 STAC92HD73*
diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt
index 88b7433d2f1..71ac995b191 100644
--- a/Documentation/sound/alsa/HD-Audio.txt
+++ b/Documentation/sound/alsa/HD-Audio.txt
@@ -16,7 +16,7 @@ methods for the	HD-audio hardware.
 The HD-audio component consists of two parts: the controller chip and 
 the codec chips on the HD-audio bus.  Linux provides a single driver
 for all controllers, snd-hda-intel.  Although the driver name contains
-a word of a well-known harware vendor, it's not specific to it but for
+a word of a well-known hardware vendor, it's not specific to it but for
 all controller chips by other companies.  Since the HD-audio
 controllers are supposed to be compatible, the single snd-hda-driver
 should work in most cases.  But, not surprisingly, there are known
diff --git a/Documentation/sound/alsa/Procfile.txt b/Documentation/sound/alsa/Procfile.txt
index cfac20cf9e3..719a819f8cc 100644
--- a/Documentation/sound/alsa/Procfile.txt
+++ b/Documentation/sound/alsa/Procfile.txt
@@ -88,26 +88,39 @@ card*/pcm*/info
 	substreams, etc.
 
 card*/pcm*/xrun_debug
-	This file appears when CONFIG_SND_DEBUG=y.
-	This shows the status of xrun (= buffer overrun/xrun) debug of
-	ALSA PCM middle layer, as an integer from 0 to 2.  The value
-	can be changed by writing to this file, such as
-
-		 # cat 2 > /proc/asound/card0/pcm0p/xrun_debug
-
-	When this value is greater than 0, the driver will show the
-	messages to kernel log when an xrun is detected.  The debug
-	message is shown also when the invalid H/W pointer is detected
-	at the update of periods (usually called from the interrupt
+	This file appears when CONFIG_SND_DEBUG=y and
+	CONFIG_PCM_XRUN_DEBUG=y.
+	This shows the status of xrun (= buffer overrun/xrun) and
+	invalid PCM position debug/check of ALSA PCM middle layer.
+	It takes an integer value, can be changed by writing to this
+	file, such as
+
+		 # cat 5 > /proc/asound/card0/pcm0p/xrun_debug
+
+	The value consists of the following bit flags:
+	  bit 0 = Enable XRUN/jiffies debug messages
+	  bit 1 = Show stack trace at XRUN / jiffies check
+	  bit 2 = Enable additional jiffies check
+	  bit 3 = Log hwptr update at each period interrupt
+	  bit 4 = Log hwptr update at each snd_pcm_update_hw_ptr()
+
+	When the bit 0 is set, the driver will show the messages to
+	kernel log when an xrun is detected.  The debug message is
+	shown also when the invalid H/W pointer is detected at the
+	update of periods (usually called from the interrupt
 	handler).
 
-	When this value is greater than 1, the driver will show the
-	stack trace additionally.  This may help the debugging.
+	When the bit 1 is set, the driver will show the stack trace
+	additionally.  This may help the debugging.
 
-	Since 2.6.30, this option also enables the hwptr check using
+	Since 2.6.30, this option can enable the hwptr check using
 	jiffies.  This detects spontaneous invalid pointer callback
 	values, but can be lead to too much corrections for a (mostly
 	buggy) hardware that doesn't give smooth pointer updates.
+	This feature is enabled via the bit 2.
+
+	Bits 3 and 4 are for logging the hwptr records.  Note that
+	these will give flood of kernel messages.
 
 card*/pcm*/sub*/info
 	The general information of this PCM sub-stream.
diff --git a/Documentation/sound/alsa/README.maya44 b/Documentation/sound/alsa/README.maya44
new file mode 100644
index 00000000000..0e41576fa13
--- /dev/null
+++ b/Documentation/sound/alsa/README.maya44
@@ -0,0 +1,163 @@
+NOTE: The following is the original document of Rainer's patch that the
+current maya44 code based on.  Some contents might be obsoleted, but I
+keep here as reference -- tiwai
+
+----------------------------------------------------------------
+ 
+STATE OF DEVELOPMENT:
+
+This driver is being developed on the initiative of Piotr Makowski (oponek@gmail.com) and financed by Lars Bergmann.
+Development is carried out by Rainer Zimmermann (mail@lightshed.de).
+
+ESI provided a sample Maya44 card for the development work.
+
+However, unfortunately it has turned out difficult to get detailed programming information, so I (Rainer Zimmermann) had to find out some card-specific information by experiment and conjecture. Some information (in particular, several GPIO bits) is still missing.
+
+This is the first testing version of the Maya44 driver released to the alsa-devel mailing list (Feb 5, 2008).
+
+
+The following functions work, as tested by Rainer Zimmermann and Piotr Makowski:
+
+- playback and capture at all sampling rates
+- input/output level
+- crossmixing
+- line/mic switch
+- phantom power switch
+- analogue monitor a.k.a bypass
+
+
+The following functions *should* work, but are not fully tested:
+
+- Channel 3+4 analogue - S/PDIF input switching
+- S/PDIF output
+- all inputs/outputs on the M/IO/DIO extension card
+- internal/external clock selection
+
+
+*In particular, we would appreciate testing of these functions by anyone who has access to an M/IO/DIO extension card.*
+
+
+Things that do not seem to work:
+
+- The level meters ("multi track") in 'alsamixer' do not seem to react to signals in (if this is a bug, it would probably be in the existing ICE1724 code).
+
+- Ardour 2.1 seems to work only via JACK, not using ALSA directly or via OSS. This still needs to be tracked down.
+
+
+DRIVER DETAILS:
+
+the following files were added:
+
+pci/ice1724/maya44.c        - Maya44 specific code
+pci/ice1724/maya44.h
+pci/ice1724/ice1724.patch
+pci/ice1724/ice1724.h.patch - PROPOSED patch to ice1724.h (see SAMPLING RATES)
+i2c/other/wm8776.c  - low-level access routines for Wolfson WM8776 codecs 
+include/wm8776.h
+
+
+Note that the wm8776.c code is meant to be card-independent and does not actually register the codec with the ALSA infrastructure.
+This is done in maya44.c, mainly because some of the WM8776 controls are used in Maya44-specific ways, and should be named appropriately.
+
+
+the following files were created in pci/ice1724, simply #including the corresponding file from the alsa-kernel tree:
+
+wtm.h
+vt1720_mobo.h
+revo.h
+prodigy192.h
+pontis.h
+phase.h
+maya44.h
+juli.h
+aureon.h
+amp.h
+envy24ht.h
+se.h
+prodigy_hifi.h
+
+
+*I hope this is the correct way to do things.*
+
+
+SAMPLING RATES:
+
+The Maya44 card (or more exactly, the Wolfson WM8776 codecs) allow a maximum sampling rate of 192 kHz for playback and 92 kHz for capture.
+
+As the ICE1724 chip only allows one global sampling rate, this is handled as follows:
+
+* setting the sampling rate on any open PCM device on the maya44 card will always set the *global* sampling rate for all playback and capture channels.
+
+* In the current state of the driver, setting rates of up to 192 kHz is permitted even for capture devices.
+
+*AVOID CAPTURING AT RATES ABOVE 96kHz*, even though it may appear to work. The codec cannot actually capture at such rates, meaning poor quality.
+
+
+I propose some additional code for limiting the sampling rate when setting on a capture pcm device. However because of the global sampling rate, this logic would be somewhat problematic.
+
+The proposed code (currently deactivated) is in ice1712.h.patch, ice1724.c and maya44.c (in pci/ice1712).
+
+
+SOUND DEVICES:
+
+PCM devices correspond to inputs/outputs as follows (assuming Maya44 is card #0):
+
+hw:0,0 input - stereo, analog input 1+2
+hw:0,0 output - stereo, analog output 1+2
+hw:0,1 input - stereo, analog input 3+4 OR S/PDIF input
+hw:0,1 output - stereo, analog output 3+4 (and SPDIF out)
+
+
+NAMING OF MIXER CONTROLS:
+
+(for more information about the signal flow, please refer to the block diagram on p.24 of the ESI Maya44 manual, or in the ESI windows software).
+
+
+PCM: (digital) output level for channel 1+2
+PCM 1: same for channel 3+4
+
+Mic Phantom+48V: switch for +48V phantom power for electrostatic microphones on input 1/2.
+    Make sure this is not turned on while any other source is connected to input 1/2.
+    It might damage the source and/or the maya44 card.
+
+Mic/Line input: if switch is is on, input jack 1/2 is microphone input (mono), otherwise line input (stereo).
+
+Bypass: analogue bypass from ADC input to output for channel 1+2. Same as "Monitor" in the windows driver.
+Bypass 1: same for channel 3+4.
+
+Crossmix: cross-mixer from channels 1+2 to channels 3+4
+Crossmix 1: cross-mixer from channels 3+4 to channels 1+2
+
+IEC958 Output: switch for S/PDIF output.
+    This is not supported by the ESI windows driver.
+    S/PDIF should output the same signal as channel 3+4. [untested!]
+
+
+Digitial output selectors:
+
+    These switches allow a direct digital routing from the ADCs to the DACs.
+    Each switch determines where the digital input data to one of the DACs comes from.
+    They are not supported by the ESI windows driver.
+    For normal operation, they should all be set to "PCM out".
+
+H/W: Output source channel 1
+H/W 1: Output source channel 2
+H/W 2: Output source channel 3
+H/W 3: Output source channel 4
+
+H/W 4 ... H/W 9: unknown function, left in to enable testing.
+    Possibly some of these control S/PDIF output(s).
+    If these turn out to be unused, they will go away in later driver versions.
+
+Selectable values for each of the digital output selectors are:
+   "PCM out" -> DAC output of the corresponding channel (default setting)
+   "Input 1"...
+   "Input 4" -> direct routing from ADC output of the selected input channel
+
+
+--------
+
+Feb 14, 2008
+Rainer Zimmermann
+mail@lightshed.de
+
diff --git a/Documentation/sound/alsa/hda_codec.txt b/Documentation/sound/alsa/hda_codec.txt
index 34e87ec1379..de8efbc7e4b 100644
--- a/Documentation/sound/alsa/hda_codec.txt
+++ b/Documentation/sound/alsa/hda_codec.txt
@@ -114,7 +114,7 @@ For writing a sequence of verbs, use snd_hda_sequence_write().
 
 There are variants of cached read/write, snd_hda_codec_write_cache(),
 snd_hda_sequence_write_cache().  These are used for recording the
-register states for the power-mangement resume.  When no PM is needed,
+register states for the power-management resume.  When no PM is needed,
 these are equivalent with non-cached version.
 
 To retrieve the number of sub nodes connected to the given node, use
diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt
index 9e6763264a2..9ac842be9b4 100644
--- a/Documentation/sound/alsa/soc/dapm.txt
+++ b/Documentation/sound/alsa/soc/dapm.txt
@@ -62,6 +62,7 @@ Audio DAPM widgets fall into a number of types:-
  o Mic        - Mic (and optional Jack)
  o Line       - Line Input/Output (and optional Jack)
  o Speaker    - Speaker
+ o Supply     - Power or clock supply widget used by other widgets.
  o Pre        - Special PRE widget (exec before all others)
  o Post       - Special POST widget (exec after all others)